res_rtp_asterisk.c File Reference

Supports RTP and RTCP with Symmetric RTP support for NAT traversal. More...

#include "asterisk.h"
#include <arpa/nameser.h>
#include "asterisk/dns_core.h"
#include "asterisk/dns_internal.h"
#include "asterisk/dns_recurring.h"
#include <sys/time.h>
#include <signal.h>
#include <fcntl.h>
#include <math.h>
#include "asterisk/conversions.h"
#include "asterisk/options.h"
#include "asterisk/logger_category.h"
#include "asterisk/stun.h"
#include "asterisk/pbx.h"
#include "asterisk/frame.h"
#include "asterisk/format_cache.h"
#include "asterisk/channel.h"
#include "asterisk/acl.h"
#include "asterisk/config.h"
#include "asterisk/lock.h"
#include "asterisk/utils.h"
#include "asterisk/cli.h"
#include "asterisk/manager.h"
#include "asterisk/unaligned.h"
#include "asterisk/module.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/smoother.h"
#include "asterisk/uuid.h"
#include "asterisk/test.h"
#include "asterisk/data_buffer.h"

Include dependency graph for res_rtp_asterisk.c:

Go to the source code of this file.

Data Structures
struct	ast_rtcp
	Structure defining an RTCP session. More...
struct	ast_rtp
	RTP session description. More...
struct	ast_rtp_rtcp_nack_payload
	Structure for storing RTP packets for retransmission. More...
struct	frame_list
struct	optional_ts
struct	rtp_learning_info
	RTP learning mode tracking information. More...
struct	rtp_red
struct	rtp_ssrc_mapping
	Structure used for mapping an incoming SSRC to an RTP instance. More...
struct	rtp_transport_wide_cc_packet_statistics
	Packet statistics (used for transport-cc) More...
struct	rtp_transport_wide_cc_statistics
	Statistics information (used for transport-cc) More...

Macros
#define	CALC_LEARNING_MIN_DURATION(count)   (((count) - 1) * 9 - 5)
	Calculate the min learning duration in ms.
#define	DEFAULT_DTLS_MTU   1200
#define	DEFAULT_DTMF_TIMEOUT   (150 * (8000 / 1000))
#define	DEFAULT_ICESUPPORT   1
#define	DEFAULT_LEARNING_MIN_DURATION   CALC_LEARNING_MIN_DURATION(DEFAULT_LEARNING_MIN_SEQUENTIAL)
#define	DEFAULT_LEARNING_MIN_SEQUENTIAL   4
#define	DEFAULT_RTP_END   31000
#define	DEFAULT_RTP_RECV_BUFFER_SIZE   20
#define	DEFAULT_RTP_SEND_BUFFER_SIZE   250
#define	DEFAULT_RTP_START   5000
#define	DEFAULT_SRTP_REPLAY_PROTECTION   1
#define	DEFAULT_STRICT_RTP   STRICT_RTP_YES
#define	DEFAULT_STUN_SOFTWARE_ATTRIBUTE   1
#define	DEFAULT_TURN_PORT   3478
#define	FLAG_3389_WARNING   (1 << 0)
#define	FLAG_DTMF_COMPENSATE   (1 << 4)
#define	FLAG_NAT_ACTIVE   (3 << 1)
#define	FLAG_NAT_INACTIVE   (0 << 1)
#define	FLAG_NAT_INACTIVE_NOWARN   (1 << 1)
#define	FLAG_NEED_MARKER_BIT   (1 << 3)
#define	FLAG_REQ_LOCAL_BRIDGE_BIT   (1 << 5)
#define	MAX_TIMESTAMP_SKEW   640
#define	MAXIMUM_RTP_PORT   65535
#define	MAXIMUM_RTP_RECV_BUFFER_SIZE   (DEFAULT_RTP_RECV_BUFFER_SIZE + 20)
#define	MAXIMUM_RTP_SEND_BUFFER_SIZE   (DEFAULT_RTP_SEND_BUFFER_SIZE + 200)
#define	MINIMUM_RTP_PORT   1024
#define	MISSING_SEQNOS_ADDED_TRIGGER   2
#define	OLD_PACKET_COUNT   1000
#define	RESCALE(in, inmin, inmax, outmin, outmax)   ((((in - inmin)/(inmax-inmin))*(outmax-outmin))+outmin)
#define	RTCP_DEFAULT_INTERVALMS   5000
#define	RTCP_FB_NACK_BLOCK_WORD_LENGTH   2
#define	RTCP_FB_REMB_BLOCK_WORD_LENGTH   4
#define	RTCP_HEADER_SSRC_LENGTH   2
#define	RTCP_LENGTH_MASK   0xFFFF
#define	RTCP_LENGTH_SHIFT   0
#define	RTCP_MAX_INTERVALMS   60000
#define	RTCP_MIN_INTERVALMS   500
#define	RTCP_PADDING_MASK   0x01
#define	RTCP_PADDING_SHIFT   29
#define	RTCP_PAYLOAD_TYPE_MASK   0xFF
#define	RTCP_PAYLOAD_TYPE_SHIFT   16
#define	RTCP_PT_APP   204
#define	RTCP_PT_BYE   203
#define	RTCP_PT_FUR   192
#define	RTCP_PT_PSFB   AST_RTP_RTCP_PSFB
#define	RTCP_PT_RR   AST_RTP_RTCP_RR
#define	RTCP_PT_SDES   202
#define	RTCP_PT_SR   AST_RTP_RTCP_SR
#define	RTCP_REPORT_COUNT_MASK   0x1F
#define	RTCP_REPORT_COUNT_SHIFT   24
#define	RTCP_RR_BLOCK_WORD_LENGTH   6
#define	RTCP_SR_BLOCK_WORD_LENGTH   5
#define	RTCP_VALID_MASK   (RTCP_VERSION_MASK_SHIFTED | (((RTCP_PAYLOAD_TYPE_MASK & ~0x1)) << RTCP_PAYLOAD_TYPE_SHIFT))
#define	RTCP_VALID_VALUE   (RTCP_VERSION_SHIFTED | (RTCP_PT_SR << RTCP_PAYLOAD_TYPE_SHIFT))
#define	RTCP_VERSION   2U
#define	RTCP_VERSION_MASK   0x03
#define	RTCP_VERSION_MASK_SHIFTED   (RTCP_VERSION_MASK << RTCP_VERSION_SHIFT)
#define	RTCP_VERSION_SHIFT   30
#define	RTCP_VERSION_SHIFTED   (RTCP_VERSION << RTCP_VERSION_SHIFT)
#define	RTP_DTLS_ESTABLISHED   -37
#define	RTP_IGNORE_FIRST_PACKETS_COUNT   15
#define	RTP_MTU   1200
#define	RTP_SEQ_MOD   (1<<16)
#define	SEQNO_CYCLE_OVER   65536
#define	SRTP_MASTER_KEY_LEN   16
#define	SRTP_MASTER_LEN   (SRTP_MASTER_KEY_LEN + SRTP_MASTER_SALT_LEN)
#define	SRTP_MASTER_SALT_LEN   14
#define	SSRC_MAPPING_ELEM_CMP(elem, value)   ((elem).instance == (value))
	SSRC mapping comparator for AST_VECTOR_REMOVE_CMP_UNORDERED()
#define	STRICT_RTP_LEARN_TIMEOUT   5000
	Strict RTP learning timeout time in milliseconds.
#define	TRANSPORT_SOCKET_RTCP   1
#define	TRANSPORT_SOCKET_RTP   0
#define	TRANSPORT_TURN_RTCP   3
#define	TRANSPORT_TURN_RTP   2
#define	TURN_STATE_WAIT_TIME   2000
#define	ZFONE_PROFILE_ID   0x505a

Enumerations
enum	strict_rtp_mode { STRICT_RTP_NO = 0 , STRICT_RTP_YES , STRICT_RTP_SEQNO }
enum	strict_rtp_state { STRICT_RTP_OPEN = 0 , STRICT_RTP_LEARN , STRICT_RTP_CLOSED }

Functions
static void	__reg_module (void)
static struct ast_rtp_instance *	__rtp_find_instance_by_ssrc (struct ast_rtp_instance *instance, struct ast_rtp *rtp, unsigned int ssrc, int source)
static int	__rtp_recvfrom (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int rtcp)
static int	__rtp_sendto (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int rtcp, int *via_ice, int use_srtp)
static void	__unreg_module (void)
struct ast_module *	AST_MODULE_SELF_SYM (void)
static unsigned int	ast_rtcp_calc_interval (struct ast_rtp *rtp)
static int	ast_rtcp_calculate_sr_rr_statistics (struct ast_rtp_instance *instance, struct ast_rtp_rtcp_report *rtcp_report, struct ast_sockaddr remote_address, int ice, int sr)
static int	ast_rtcp_generate_compound_prefix (struct ast_rtp_instance *instance, unsigned char *rtcpheader, struct ast_rtp_rtcp_report *report, int *sr)
static int	ast_rtcp_generate_nack (struct ast_rtp_instance *instance, unsigned char *rtcpheader)
static int	ast_rtcp_generate_report (struct ast_rtp_instance *instance, unsigned char *rtcpheader, struct ast_rtp_rtcp_report *rtcp_report, int *sr)
static int	ast_rtcp_generate_sdes (struct ast_rtp_instance *instance, unsigned char *rtcpheader, struct ast_rtp_rtcp_report *rtcp_report)
static struct ast_frame *	ast_rtcp_interpret (struct ast_rtp_instance *instance, struct ast_srtp *srtp, const unsigned char *rtcpdata, size_t size, struct ast_sockaddr *addr)
static struct ast_frame *	ast_rtcp_read (struct ast_rtp_instance *instance)
static int	ast_rtcp_write (const void *data)
	Write a RTCP packet to the far end.
static int	ast_rtp_bundle (struct ast_rtp_instance *child, struct ast_rtp_instance *parent)
static void	ast_rtp_change_source (struct ast_rtp_instance *instance)
static int	ast_rtp_destroy (struct ast_rtp_instance *instance)
static int	ast_rtp_dtmf_begin (struct ast_rtp_instance *instance, char digit)
static int	ast_rtp_dtmf_compatible (struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1)
static int	ast_rtp_dtmf_continuation (struct ast_rtp_instance *instance)
static int	ast_rtp_dtmf_end (struct ast_rtp_instance *instance, char digit)
static int	ast_rtp_dtmf_end_with_duration (struct ast_rtp_instance *instance, char digit, unsigned int duration)
static enum ast_rtp_dtmf_mode	ast_rtp_dtmf_mode_get (struct ast_rtp_instance *instance)
static int	ast_rtp_dtmf_mode_set (struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
static int	ast_rtp_extension_enable (struct ast_rtp_instance *instance, enum ast_rtp_extension extension)
static int	ast_rtp_fd (struct ast_rtp_instance *instance, int rtcp)
static const char *	ast_rtp_get_cname (struct ast_rtp_instance *instance)
static unsigned int	ast_rtp_get_ssrc (struct ast_rtp_instance *instance)
static int	ast_rtp_get_stat (struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
static struct ast_frame *	ast_rtp_interpret (struct ast_rtp_instance *instance, struct ast_srtp *srtp, const struct ast_sockaddr *remote_address, unsigned char *read_area, int length, int prev_seqno, unsigned int bundled)
static int	ast_rtp_local_bridge (struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1)
static int	ast_rtp_new (struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
static void	ast_rtp_prop_set (struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
static int	ast_rtp_qos_set (struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
static struct ast_frame *	ast_rtp_read (struct ast_rtp_instance *instance, int rtcp)
static void	ast_rtp_remote_address_set (struct ast_rtp_instance *instance, struct ast_sockaddr *addr)
static int	ast_rtp_rtcp_handle_nack (struct ast_rtp_instance *instance, unsigned int *nackdata, unsigned int position, unsigned int length)
static int	ast_rtp_rtcp_handle_nack_locking (struct ast_rtp_instance *instance, struct ast_rtp_instance *transport, unsigned int *nackdata, unsigned int position, unsigned int length)
static int	ast_rtp_sendcng (struct ast_rtp_instance *instance, int level)
	generate comfort noice (CNG)
static void	ast_rtp_set_remote_ssrc (struct ast_rtp_instance *instance, unsigned int ssrc)
static void	ast_rtp_set_stream_num (struct ast_rtp_instance *instance, int stream_num)
static void	ast_rtp_stop (struct ast_rtp_instance *instance)
static void	ast_rtp_stun_request (struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username)
static void	ast_rtp_update_source (struct ast_rtp_instance *instance)
static int	ast_rtp_write (struct ast_rtp_instance *instance, struct ast_frame *frame)
static int	bridge_p2p_rtp_write (struct ast_rtp_instance *instance, struct ast_rtp_instance *instance1, unsigned int *rtpheader, int len, int hdrlen)
static void	calc_mean_and_standard_deviation (double new_sample, double *mean, double *std_dev, unsigned int *count)
static double	calc_media_experience_score (struct ast_rtp_instance *instance, double normdevrtt, double normdev_rxjitter, double stdev_rxjitter, double normdev_rxlost)
	Calculate a "media experience score" based on given data.
static void	calc_rxstamp_and_jitter (struct timeval *tv, struct ast_rtp *rtp, unsigned int rx_rtp_ts, int mark)
static unsigned int	calc_txstamp (struct ast_rtp *rtp, struct timeval *delivery)
static void	calculate_lost_packet_statistics (struct ast_rtp *rtp, unsigned int *lost_packets, int *fraction_lost)
static int	compare_by_value (int elem, int value)
	Helper function to compare an elem in a vector by value.
static struct ast_frame *	create_dtmf_frame (struct ast_rtp_instance *instance, enum ast_frame_type type, int compensate)
static int	create_new_socket (const char *type, struct ast_sockaddr *bind_addr)
static int	find_by_value (int elem, int value)
	Helper function to find an elem in a vector by value.
static char *	handle_cli_rtcp_set_debug (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	handle_cli_rtcp_set_stats (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	handle_cli_rtp_set_debug (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	handle_cli_rtp_settings (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static int	load_module (void)
static void	ntp2timeval (unsigned int msw, unsigned int lsw, struct timeval *tv)
static struct ast_frame *	process_cn_rfc3389 (struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, int payloadtype, int mark)
static struct ast_frame *	process_dtmf_cisco (struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, int payloadtype, int mark)
static void	process_dtmf_rfc2833 (struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, int payloadtype, int mark, struct frame_list *frames)
static void	put_unaligned_time24 (void *p, uint32_t time_msw, uint32_t time_lsw)
static struct ast_frame *	red_t140_to_red (struct rtp_red *red)
static int	red_write (const void *data)
	Write t140 redundancy frame.
static int	reload_module (void)
static int	rtcp_debug_test_addr (struct ast_sockaddr *addr)
static char *	rtcp_do_debug_ip (struct ast_cli_args *a)
static int	rtcp_mux (struct ast_rtp *rtp, const unsigned char *packet)
static const char *	rtcp_payload_subtype2str (unsigned int pt, unsigned int subtype)
static const char *	rtcp_payload_type2str (unsigned int pt)
static int	rtcp_recvfrom (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa)
static int	rtcp_sendto (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int *ice)
static int	rtp_allocate_transport (struct ast_rtp_instance *instance, struct ast_rtp *rtp)
static void	rtp_deallocate_transport (struct ast_rtp_instance *instance, struct ast_rtp *rtp)
static int	rtp_debug_test_addr (struct ast_sockaddr *addr)
static char *	rtp_do_debug_ip (struct ast_cli_args *a)
static struct ast_rtp_instance *	rtp_find_instance_by_media_source_ssrc (struct ast_rtp_instance *instance, struct ast_rtp *rtp, unsigned int ssrc)
static struct ast_rtp_instance *	rtp_find_instance_by_packet_source_ssrc (struct ast_rtp_instance *instance, struct ast_rtp *rtp, unsigned int ssrc)
static void	rtp_instance_parse_extmap_extensions (struct ast_rtp_instance *instance, struct ast_rtp *rtp, unsigned char *extension, int len)
static void	rtp_instance_parse_transport_wide_cc (struct ast_rtp_instance *instance, struct ast_rtp *rtp, unsigned char *data, int len)
static void	rtp_instance_unlock (struct ast_rtp_instance *instance)
static int	rtp_learning_rtp_seq_update (struct rtp_learning_info *info, uint16_t seq)
static void	rtp_learning_seq_init (struct rtp_learning_info *info, uint16_t seq)
static void	rtp_learning_start (struct ast_rtp *rtp)
	Start the strictrtp learning mode.
static int	rtp_raw_write (struct ast_rtp_instance *instance, struct ast_frame *frame, int codec)
static int	rtp_recvfrom (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa)
static int	rtp_red_buffer (struct ast_rtp_instance *instance, struct ast_frame *frame)
static int	rtp_red_init (struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
static int	rtp_reload (int reload, int by_external_config)
static int	rtp_sendto (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int *ice)
static int	rtp_transport_wide_cc_feedback_produce (const void *data)
static void	rtp_transport_wide_cc_feedback_status_append (unsigned char *rtcpheader, int *packet_len, int *status_vector_chunk_bits, uint16_t *status_vector_chunk, int *run_length_chunk_count, int *run_length_chunk_status, int status)
static void	rtp_transport_wide_cc_feedback_status_vector_append (unsigned char *rtcpheader, int *packet_len, int *status_vector_chunk_bits, uint16_t *status_vector_chunk, int status)
static int	rtp_transport_wide_cc_packet_statistics_cmp (struct rtp_transport_wide_cc_packet_statistics a, struct rtp_transport_wide_cc_packet_statistics b)
static void	rtp_write_rtcp_fir (struct ast_rtp_instance *instance, struct ast_rtp *rtp, struct ast_sockaddr *remote_address)
static void	rtp_write_rtcp_psfb (struct ast_rtp_instance *instance, struct ast_rtp *rtp, struct ast_frame *frame, struct ast_sockaddr *remote_address)
static void	timeval2ntp (struct timeval tv, unsigned int *msw, unsigned int *lsw)
static int	unload_module (void)
static void	update_jitter_stats (struct ast_rtp *rtp, unsigned int ia_jitter)
static void	update_local_mes_stats (struct ast_rtp *rtp)
static void	update_lost_stats (struct ast_rtp *rtp, unsigned int lost_packets)
static void	update_reported_mes_stats (struct ast_rtp *rtp)
static int	update_rtt_stats (struct ast_rtp *rtp, unsigned int lsr, unsigned int dlsr)

Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Asterisk RTP Stack" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .reload = reload_module, .load_pri = AST_MODPRI_CHANNEL_DEPEND, #ifdef HAVE_PJPROJECT .requires = "res_pjproject", #endif }
static const struct ast_module_info *	ast_module_info = &__mod_info
static struct ast_rtp_engine	asterisk_rtp_engine
static struct ast_cli_entry	cli_rtp []
static int	dtmftimeout = DEFAULT_DTMF_TIMEOUT
static int	learning_min_duration = DEFAULT_LEARNING_MIN_DURATION
static int	learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL
struct ast_srtp_res *	res_srtp
struct ast_srtp_policy_res *	res_srtp_policy
static struct ast_sockaddr	rtcpdebugaddr
static int	rtcpdebugport
static int	rtcpinterval = RTCP_DEFAULT_INTERVALMS
static int	rtcpstats
static struct ast_sockaddr	rtpdebugaddr
static int	rtpdebugport
static int	rtpend = DEFAULT_RTP_END
static int	rtpstart = DEFAULT_RTP_START
static int	srtp_replay_protection = DEFAULT_SRTP_REPLAY_PROTECTION
static int	strictrtp = DEFAULT_STRICT_RTP

Detailed Description

Supports RTP and RTCP with Symmetric RTP support for NAT traversal.

Author

Mark Spencer marks.nosp@m.ter@.nosp@m.digiu.nosp@m.m.co.nosp@m.m

Note

RTP is defined in RFC 3550.

Definition in file res_rtp_asterisk.c.

Macro Definition Documentation

CALC_LEARNING_MIN_DURATION

#define CALC_LEARNING_MIN_DURATION

(

count

)

(((count) - 1) * 9 - 5)

Calculate the min learning duration in ms.

The min supported packet size represents 10 ms and we need to account for some jitter and fast clocks while learning. Some messed up devices have very bad jitter for a small packet sample size. Jitter can also be introduced by the network itself.

So we'll allow packets to come in every 9ms on average for fast clocking with the last one coming in 5ms early for jitter.

Definition at line 159 of file res_rtp_asterisk.c.

DEFAULT_DTLS_MTU

#define DEFAULT_DTLS_MTU   1200

Definition at line 193 of file res_rtp_asterisk.c.

DEFAULT_DTMF_TIMEOUT

#define DEFAULT_DTMF_TIMEOUT   (150 * (8000 / 1000))

samples

Definition at line 142 of file res_rtp_asterisk.c.

DEFAULT_ICESUPPORT

#define DEFAULT_ICESUPPORT   1

Definition at line 191 of file res_rtp_asterisk.c.

DEFAULT_LEARNING_MIN_DURATION

#define DEFAULT_LEARNING_MIN_DURATION   CALC_LEARNING_MIN_DURATION(DEFAULT_LEARNING_MIN_SEQUENTIAL)

Definition at line 160 of file res_rtp_asterisk.c.

DEFAULT_LEARNING_MIN_SEQUENTIAL

#define DEFAULT_LEARNING_MIN_SEQUENTIAL   4

Definition at line 146 of file res_rtp_asterisk.c.

DEFAULT_RTP_END

#define DEFAULT_RTP_END   31000

Default maximum port number to end allocating RTP ports at

Definition at line 106 of file res_rtp_asterisk.c.

DEFAULT_RTP_RECV_BUFFER_SIZE

#define DEFAULT_RTP_RECV_BUFFER_SIZE   20

The initial size of the RTP receiver buffer

Definition at line 117 of file res_rtp_asterisk.c.

DEFAULT_RTP_SEND_BUFFER_SIZE

#define DEFAULT_RTP_SEND_BUFFER_SIZE   250

The initial size of the RTP send buffer

Definition at line 115 of file res_rtp_asterisk.c.

DEFAULT_RTP_START

#define DEFAULT_RTP_START   5000

Default port number to start allocating RTP ports from

Definition at line 105 of file res_rtp_asterisk.c.

DEFAULT_SRTP_REPLAY_PROTECTION

#define DEFAULT_SRTP_REPLAY_PROTECTION   1

Definition at line 190 of file res_rtp_asterisk.c.

DEFAULT_STRICT_RTP

#define DEFAULT_STRICT_RTP   STRICT_RTP_YES

Enabled by default

Definition at line 189 of file res_rtp_asterisk.c.

DEFAULT_STUN_SOFTWARE_ATTRIBUTE

#define DEFAULT_STUN_SOFTWARE_ATTRIBUTE   1

Definition at line 192 of file res_rtp_asterisk.c.

DEFAULT_TURN_PORT

#define DEFAULT_TURN_PORT   3478

Definition at line 111 of file res_rtp_asterisk.c.

FLAG_3389_WARNING

#define FLAG_3389_WARNING   (1 << 0)

Definition at line 308 of file res_rtp_asterisk.c.

FLAG_DTMF_COMPENSATE

#define FLAG_DTMF_COMPENSATE   (1 << 4)

Definition at line 313 of file res_rtp_asterisk.c.

FLAG_NAT_ACTIVE

#define FLAG_NAT_ACTIVE   (3 << 1)

Definition at line 309 of file res_rtp_asterisk.c.

FLAG_NAT_INACTIVE

#define FLAG_NAT_INACTIVE   (0 << 1)

Definition at line 310 of file res_rtp_asterisk.c.

FLAG_NAT_INACTIVE_NOWARN

#define FLAG_NAT_INACTIVE_NOWARN   (1 << 1)

Definition at line 311 of file res_rtp_asterisk.c.

FLAG_NEED_MARKER_BIT

#define FLAG_NEED_MARKER_BIT   (1 << 3)

Definition at line 312 of file res_rtp_asterisk.c.

FLAG_REQ_LOCAL_BRIDGE_BIT

#define FLAG_REQ_LOCAL_BRIDGE_BIT   (1 << 5)

Definition at line 314 of file res_rtp_asterisk.c.

MAX_TIMESTAMP_SKEW

#define MAX_TIMESTAMP_SKEW   640

Definition at line 98 of file res_rtp_asterisk.c.

MAXIMUM_RTP_PORT

#define MAXIMUM_RTP_PORT   65535

Maximum port number to accept

Definition at line 109 of file res_rtp_asterisk.c.

MAXIMUM_RTP_RECV_BUFFER_SIZE

#define MAXIMUM_RTP_RECV_BUFFER_SIZE   (DEFAULT_RTP_RECV_BUFFER_SIZE + 20)

Maximum RTP receive buffer size

Definition at line 118 of file res_rtp_asterisk.c.

MAXIMUM_RTP_SEND_BUFFER_SIZE

#define MAXIMUM_RTP_SEND_BUFFER_SIZE   (DEFAULT_RTP_SEND_BUFFER_SIZE + 200)

Maximum RTP send buffer size

Definition at line 116 of file res_rtp_asterisk.c.

MINIMUM_RTP_PORT

#define MINIMUM_RTP_PORT   1024

Minimum port number to accept

Definition at line 108 of file res_rtp_asterisk.c.

MISSING_SEQNOS_ADDED_TRIGGER

#define MISSING_SEQNOS_ADDED_TRIGGER   2

The number of immediate missing packets that will trigger an immediate NACK

Definition at line 120 of file res_rtp_asterisk.c.

OLD_PACKET_COUNT

#define OLD_PACKET_COUNT   1000

The number of previous packets that are considered old

Definition at line 119 of file res_rtp_asterisk.c.

RESCALE

#define RESCALE	(		in,
			inmin,
			inmax,
			outmin,
			outmax
	)		((((in - inmin)/(inmax-inmin))*(outmax-outmin))+outmin)

Definition at line 6434 of file res_rtp_asterisk.c.

RTCP_DEFAULT_INTERVALMS

#define RTCP_DEFAULT_INTERVALMS   5000

Default milli-seconds between RTCP reports we send

Definition at line 101 of file res_rtp_asterisk.c.

RTCP_FB_NACK_BLOCK_WORD_LENGTH

#define RTCP_FB_NACK_BLOCK_WORD_LENGTH   2

Definition at line 6861 of file res_rtp_asterisk.c.

RTCP_FB_REMB_BLOCK_WORD_LENGTH

#define RTCP_FB_REMB_BLOCK_WORD_LENGTH   4

Definition at line 6860 of file res_rtp_asterisk.c.

RTCP_HEADER_SSRC_LENGTH

#define RTCP_HEADER_SSRC_LENGTH   2

Definition at line 6859 of file res_rtp_asterisk.c.

RTCP_LENGTH_MASK

#define RTCP_LENGTH_MASK   0xFFFF

Definition at line 6824 of file res_rtp_asterisk.c.

RTCP_LENGTH_SHIFT

#define RTCP_LENGTH_SHIFT   0

Definition at line 6833 of file res_rtp_asterisk.c.

RTCP_MAX_INTERVALMS

#define RTCP_MAX_INTERVALMS   60000

Max milli-seconds between RTCP reports we send

Definition at line 103 of file res_rtp_asterisk.c.

RTCP_MIN_INTERVALMS

#define RTCP_MIN_INTERVALMS   500

Min milli-seconds between RTCP reports we send

Definition at line 102 of file res_rtp_asterisk.c.

RTCP_PADDING_MASK

#define RTCP_PADDING_MASK   0x01

Definition at line 6827 of file res_rtp_asterisk.c.

RTCP_PADDING_SHIFT

#define RTCP_PADDING_SHIFT   29

Definition at line 6836 of file res_rtp_asterisk.c.

RTCP_PAYLOAD_TYPE_MASK

#define RTCP_PAYLOAD_TYPE_MASK   0xFF

Definition at line 6825 of file res_rtp_asterisk.c.

RTCP_PAYLOAD_TYPE_SHIFT

#define RTCP_PAYLOAD_TYPE_SHIFT   16

Definition at line 6834 of file res_rtp_asterisk.c.

RTCP_PT_APP

#define RTCP_PT_APP   204

Application defined (From RFC3550)

Definition at line 135 of file res_rtp_asterisk.c.

RTCP_PT_BYE

#define RTCP_PT_BYE   203

Goodbye (To remove SSRC's from tables) (From RFC3550)

Definition at line 133 of file res_rtp_asterisk.c.

RTCP_PT_FUR

#define RTCP_PT_FUR   192

Full INTRA-frame Request / Fast Update Request (From RFC2032)

Definition at line 125 of file res_rtp_asterisk.c.

RTCP_PT_PSFB

#define RTCP_PT_PSFB   AST_RTP_RTCP_PSFB

Payload Specific Feed Back (From RFC4585 also RFC5104)

Definition at line 138 of file res_rtp_asterisk.c.

RTCP_PT_RR

#define RTCP_PT_RR   AST_RTP_RTCP_RR

Receiver Report (From RFC3550)

Definition at line 129 of file res_rtp_asterisk.c.

RTCP_PT_SDES

#define RTCP_PT_SDES   202

Source Description (From RFC3550)

Definition at line 131 of file res_rtp_asterisk.c.

RTCP_PT_SR

#define RTCP_PT_SR   AST_RTP_RTCP_SR

Sender Report (From RFC3550)

Definition at line 127 of file res_rtp_asterisk.c.

RTCP_REPORT_COUNT_MASK

#define RTCP_REPORT_COUNT_MASK   0x1F

Definition at line 6826 of file res_rtp_asterisk.c.

RTCP_REPORT_COUNT_SHIFT

#define RTCP_REPORT_COUNT_SHIFT   24

Definition at line 6835 of file res_rtp_asterisk.c.

RTCP_RR_BLOCK_WORD_LENGTH

#define RTCP_RR_BLOCK_WORD_LENGTH   6

Definition at line 6858 of file res_rtp_asterisk.c.

RTCP_SR_BLOCK_WORD_LENGTH

#define RTCP_SR_BLOCK_WORD_LENGTH   5

Definition at line 6857 of file res_rtp_asterisk.c.

RTCP_VALID_MASK

#define RTCP_VALID_MASK   (RTCP_VERSION_MASK_SHIFTED | (((RTCP_PAYLOAD_TYPE_MASK & ~0x1)) << RTCP_PAYLOAD_TYPE_SHIFT))

Definition at line 6854 of file res_rtp_asterisk.c.

RTCP_VALID_VALUE

#define RTCP_VALID_VALUE   (RTCP_VERSION_SHIFTED | (RTCP_PT_SR << RTCP_PAYLOAD_TYPE_SHIFT))

Definition at line 6855 of file res_rtp_asterisk.c.

RTCP_VERSION

#define RTCP_VERSION   2U

Definition at line 6839 of file res_rtp_asterisk.c.

RTCP_VERSION_MASK

#define RTCP_VERSION_MASK   0x03

Definition at line 6828 of file res_rtp_asterisk.c.

RTCP_VERSION_MASK_SHIFTED

#define RTCP_VERSION_MASK_SHIFTED   (RTCP_VERSION_MASK << RTCP_VERSION_SHIFT)

Definition at line 6841 of file res_rtp_asterisk.c.

RTCP_VERSION_SHIFT

#define RTCP_VERSION_SHIFT   30

Definition at line 6837 of file res_rtp_asterisk.c.

RTCP_VERSION_SHIFTED

#define RTCP_VERSION_SHIFTED   (RTCP_VERSION << RTCP_VERSION_SHIFT)

Definition at line 6840 of file res_rtp_asterisk.c.

RTP_DTLS_ESTABLISHED

#define RTP_DTLS_ESTABLISHED   -37

Definition at line 166 of file res_rtp_asterisk.c.

RTP_IGNORE_FIRST_PACKETS_COUNT

#define RTP_IGNORE_FIRST_PACKETS_COUNT   15

Because both ends usually don't start sending RTP at the same time, some of the calculations like rtt and jitter will probably be unstable for a while so we'll skip some received packets before starting analyzing. This just affects analyzing; we still process the RTP as normal.

Definition at line 203 of file res_rtp_asterisk.c.

RTP_MTU

#define RTP_MTU   1200

Definition at line 140 of file res_rtp_asterisk.c.

RTP_SEQ_MOD

#define RTP_SEQ_MOD   (1<<16)

A sequence number can't be more than 16 bits

Definition at line 100 of file res_rtp_asterisk.c.

SEQNO_CYCLE_OVER

#define SEQNO_CYCLE_OVER   65536

The number after the maximum allowed sequence number

Definition at line 122 of file res_rtp_asterisk.c.

SRTP_MASTER_KEY_LEN

#define SRTP_MASTER_KEY_LEN   16

Definition at line 162 of file res_rtp_asterisk.c.

SRTP_MASTER_LEN

#define SRTP_MASTER_LEN   (SRTP_MASTER_KEY_LEN + SRTP_MASTER_SALT_LEN)

Definition at line 164 of file res_rtp_asterisk.c.

SRTP_MASTER_SALT_LEN

#define SRTP_MASTER_SALT_LEN   14

Definition at line 163 of file res_rtp_asterisk.c.

SSRC_MAPPING_ELEM_CMP

#define SSRC_MAPPING_ELEM_CMP	(		elem,
			value
	)		((elem).instance == (value))

SSRC mapping comparator for AST_VECTOR_REMOVE_CMP_UNORDERED()

Parameters

elem	Element to compare against
value	Value to compare with the vector element.

Return values

0	if element does not match.
Non-zero	if element matches.

Definition at line 4385 of file res_rtp_asterisk.c.

STRICT_RTP_LEARN_TIMEOUT

#define STRICT_RTP_LEARN_TIMEOUT   5000

Strict RTP learning timeout time in milliseconds.

Note

Set to 5 seconds to allow reinvite chains for direct media to settle before media actually starts to arrive. There may be a reinvite collision involved on the other leg.

Definition at line 187 of file res_rtp_asterisk.c.

TRANSPORT_SOCKET_RTCP

#define TRANSPORT_SOCKET_RTCP   1

Definition at line 317 of file res_rtp_asterisk.c.

TRANSPORT_SOCKET_RTP

#define TRANSPORT_SOCKET_RTP   0

Definition at line 316 of file res_rtp_asterisk.c.

TRANSPORT_TURN_RTCP

#define TRANSPORT_TURN_RTCP   3

Definition at line 319 of file res_rtp_asterisk.c.

TRANSPORT_TURN_RTP

#define TRANSPORT_TURN_RTP   2

Definition at line 318 of file res_rtp_asterisk.c.

TURN_STATE_WAIT_TIME

#define TURN_STATE_WAIT_TIME   2000

Definition at line 113 of file res_rtp_asterisk.c.

ZFONE_PROFILE_ID

#define ZFONE_PROFILE_ID   0x505a

Definition at line 144 of file res_rtp_asterisk.c.

Enumeration Type Documentation

strict_rtp_mode

enum strict_rtp_mode

Enumerator
STRICT_RTP_NO
STRICT_RTP_YES	Don't adhere to any strict RTP rules
STRICT_RTP_SEQNO	Strict RTP that restricts packets based on time and sequence number

Definition at line 174 of file res_rtp_asterisk.c.

                     {
    STRICT_RTP_NO = 0,  /*! Don't adhere to any strict RTP rules */
    STRICT_RTP_YES,     /*! Strict RTP that restricts packets based on time and sequence number */
    STRICT_RTP_SEQNO,   /*! Strict RTP that restricts packets based on sequence number */
};

strict_rtp_state

enum strict_rtp_state

Enumerator
STRICT_RTP_OPEN
STRICT_RTP_LEARN	No RTP packets should be dropped, all sources accepted
STRICT_RTP_CLOSED	Accept next packet as source

Definition at line 168 of file res_rtp_asterisk.c.

                      {
    STRICT_RTP_OPEN = 0, /*! No RTP packets should be dropped, all sources accepted */
    STRICT_RTP_LEARN,    /*! Accept next packet as source */
    STRICT_RTP_CLOSED,   /*! Drop all RTP packets not coming from source that was learned */
};

Function Documentation

__reg_module()

static void __reg_module ( void  )

static

Definition at line 10798 of file res_rtp_asterisk.c.

__rtp_find_instance_by_ssrc()

static struct ast_rtp_instance * __rtp_find_instance_by_ssrc ( struct ast_rtp_instance *  instance, struct ast_rtp *  rtp, unsigned int  ssrc, int  source  )

static

Precondition

instance is locked

Definition at line 6595 of file res_rtp_asterisk.c.

{
    int index;
 
    if (!AST_VECTOR_SIZE(&rtp->ssrc_mapping)) {
        /* This instance is not bundled */
        return instance;
    }
 
    /* Find the bundled child instance */
    for (index = 0; index < AST_VECTOR_SIZE(&rtp->ssrc_mapping); ++index) {
        struct rtp_ssrc_mapping *mapping = AST_VECTOR_GET_ADDR(&rtp->ssrc_mapping, index);
        unsigned int mapping_ssrc = source ? ast_rtp_get_ssrc(mapping->instance) : mapping->ssrc;
 
        if (mapping->ssrc_valid && mapping_ssrc == ssrc) {
            return mapping->instance;
        }
    }
 
    /* Does the SSRC match the bundled parent? */
    if (rtp->themssrc_valid && rtp->themssrc == ssrc) {
        return instance;
    }
    return NULL;
}

References ast_rtp_get_ssrc(), AST_VECTOR_GET_ADDR, AST_VECTOR_SIZE, rtp_ssrc_mapping::instance, NULL, rtp_ssrc_mapping::ssrc, ast_rtp::ssrc_mapping, rtp_ssrc_mapping::ssrc_valid, ast_rtp::themssrc, and ast_rtp::themssrc_valid.

Referenced by rtp_find_instance_by_media_source_ssrc(), and rtp_find_instance_by_packet_source_ssrc().

__rtp_recvfrom()

static int __rtp_recvfrom ( struct ast_rtp_instance *  instance, void *  buf, size_t  size, int  flags, struct ast_sockaddr *  sa, int  rtcp  )

static

Precondition

instance is locked

Definition at line 3237 of file res_rtp_asterisk.c.

{
    int len;
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
    char *in = buf;
#endif
#ifdef HAVE_PJPROJECT
    struct ast_sockaddr *loop = rtcp ? &rtp->rtcp_loop : &rtp->rtp_loop;
#endif
#ifdef TEST_FRAMEWORK
    struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
#endif
 
    if ((len = ast_recvfrom(rtcp ? rtp->rtcp->s : rtp->s, buf, size, flags, sa)) < 0) {
        return len;
    }
 
#ifdef TEST_FRAMEWORK
    if (test && test->packets_to_drop > 0) {
        test->packets_to_drop--;
        return 0;
    }
#endif
 
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
    /* If this is an SSL packet pass it to OpenSSL for processing. RFC section for first byte value:
     * https://tools.ietf.org/html/rfc5764#section-5.1.2 */
    if ((*in >= 20) && (*in <= 63)) {
        struct dtls_details *dtls = !rtcp ? &rtp->dtls : &rtp->rtcp->dtls;
        int res = 0;
 
        /* If no SSL session actually exists terminate things */
        if (!dtls->ssl) {
            ast_log(LOG_ERROR, "Received SSL traffic on RTP instance '%p' without an SSL session\n",
                instance);
            return -1;
        }
 
        ast_debug_dtls(3, "(%p) DTLS - __rtp_recvfrom rtp=%p - Got SSL packet '%d'\n", instance, rtp, *in);
 
#ifdef HAVE_PJPROJECT
        /* If this packet arrived via TURN/ICE loopback re-injection,
         * substitute the real remote address before the candidate check
         * otherwise the DTLS check will see 127.0.0.1 and drop the packet.
         */
        if (!ast_sockaddr_isnull(&rtp->rtp_loop) && !ast_sockaddr_cmp(&rtp->rtp_loop, sa)) {
            ast_rtp_instance_get_remote_address(instance, sa);
        } else if (rtcp && !ast_sockaddr_isnull(&rtp->rtcp_loop) && !ast_sockaddr_cmp(&rtp->rtcp_loop, sa)) {
            ast_sockaddr_copy(sa, &rtp->rtcp->them);
        }
#endif
 
        /*
         * If ICE is in use, we can prevent a possible DOS attack
         * by allowing DTLS protocol messages (client hello, etc)
         * only from sources that are in the active remote
         * candidates list.
         */
 
#ifdef HAVE_PJPROJECT
        if (rtp->ice) {
            int pass_src_check = 0;
            int ix = 0;
 
            /*
             * You'd think that this check would cause a "deadlock"
             * because ast_rtp_ice_start_media calls dtls_perform_handshake
             * before it sets ice_media_started = 1 so how can we do a
             * handshake if we're dropping packets before we send them
             * to openssl.  Fortunately, dtls_perform_handshake just sets
             * up openssl to do the handshake and doesn't actually perform it
             * itself and the locking prevents __rtp_recvfrom from
             * running before the ice_media_started flag is set.  So only
             * unexpected DTLS packets can get dropped here.
             */
            if (!rtp->ice_media_started) {
                ast_log(LOG_WARNING, "%s: DTLS packet from %s dropped. ICE not completed yet.\n",
                    ast_rtp_instance_get_channel_id(instance),
                    ast_sockaddr_stringify(sa));
                return 0;
            }
 
            /*
             * If we got this far, then there have to be candidates.
             * We have to use pjproject's rcands because they may have
             * peer reflexive candidates that our ice_active_remote_candidates
             * won't.
             */
            for (ix = 0; ix < rtp->ice->real_ice->rcand_cnt; ix++) {
                pj_ice_sess_cand *rcand = &rtp->ice->real_ice->rcand[ix];
                if (ast_sockaddr_pj_sockaddr_cmp(sa, &rcand->addr) == 0) {
                    pass_src_check = 1;
                    break;
                }
            }
 
            if (!pass_src_check) {
                ast_log(LOG_WARNING, "%s: DTLS packet from %s dropped. Source not in ICE active candidate list.\n",
                    ast_rtp_instance_get_channel_id(instance),
                    ast_sockaddr_stringify(sa));
                return 0;
            }
        }
#endif
 
        /*
         * A race condition is prevented between dtls_perform_handshake()
         * and this function because both functions have to get the
         * instance lock before they can do anything.  The
         * dtls_perform_handshake() function needs to start the timer
         * before we stop it below.
         */
 
        /* Before we feed data into OpenSSL ensure that the timeout timer is either stopped or completed */
        ao2_unlock(instance);
        dtls_srtp_stop_timeout_timer(instance, rtp, rtcp);
        ao2_lock(instance);
 
        /* If we don't yet know if we are active or passive and we receive a packet... we are obviously passive */
        if (dtls->dtls_setup == AST_RTP_DTLS_SETUP_ACTPASS) {
            dtls->dtls_setup = AST_RTP_DTLS_SETUP_PASSIVE;
            SSL_set_accept_state(dtls->ssl);
        }
 
        BIO_write(dtls->read_bio, buf, len);
 
        len = SSL_read(dtls->ssl, buf, len);
 
        if ((len < 0) && (SSL_get_error(dtls->ssl, len) == SSL_ERROR_SSL)) {
            unsigned long error = ERR_get_error();
            ast_log(LOG_ERROR, "DTLS failure occurred on RTP instance '%p' due to reason '%s', terminating\n",
                instance, ERR_reason_error_string(error));
            return -1;
        }
 
        if (SSL_is_init_finished(dtls->ssl)) {
            /* Any further connections will be existing since this is now established */
            dtls->connection = AST_RTP_DTLS_CONNECTION_EXISTING;
            /* Use the keying material to set up key/salt information */
            if ((res = dtls_srtp_setup(rtp, instance, rtcp))) {
                return res;
            }
            /* Notify that dtls has been established */
            res = RTP_DTLS_ESTABLISHED;
 
            ast_debug_dtls(3, "(%p) DTLS - __rtp_recvfrom rtp=%p - established'\n", instance, rtp);
        } else {
            /* Since we've sent additional traffic start the timeout timer for retransmission */
            dtls_srtp_start_timeout_timer(instance, rtp, rtcp);
        }
 
        return res;
    }
#endif
 
#ifdef HAVE_PJPROJECT
    if (!ast_sockaddr_isnull(loop) && !ast_sockaddr_cmp(loop, sa)) {
        /* ICE traffic will have been handled in the TURN callback, so skip it but update the address
         * so it reflects the actual source and not the loopback
         */
        if (rtcp) {
            ast_sockaddr_copy(sa, &rtp->rtcp->them);
        } else {
            ast_rtp_instance_get_remote_address(instance, sa);
        }
    } else if (rtp->ice) {
        pj_str_t combined = pj_str(ast_sockaddr_stringify(sa));
        pj_sockaddr address;
        pj_status_t status;
        struct ice_wrap *ice;
 
        pj_thread_register_check();
 
        pj_sockaddr_parse(pj_AF_UNSPEC(), 0, &combined, &address);
 
        /* Release the instance lock to avoid deadlock with PJPROJECT group lock */
        ice = rtp->ice;
        ao2_ref(ice, +1);
        ao2_unlock(instance);
        status = pj_ice_sess_on_rx_pkt(ice->real_ice,
            rtcp ? AST_RTP_ICE_COMPONENT_RTCP : AST_RTP_ICE_COMPONENT_RTP,
            rtcp ? TRANSPORT_SOCKET_RTCP : TRANSPORT_SOCKET_RTP, buf, len, &address,
            pj_sockaddr_get_len(&address));
        ao2_ref(ice, -1);
        ao2_lock(instance);
        if (status != PJ_SUCCESS) {
            char err_buf[100];
 
            pj_strerror(status, err_buf, sizeof(err_buf));
            ast_log(LOG_WARNING, "PJ ICE Rx error status code: %d '%s'.\n",
                (int)status, err_buf);
            return -1;
        }
        if (!rtp->passthrough) {
            /* If a unidirectional ICE negotiation occurs then lock on to the source of the
             * ICE traffic and use it as the target. This will occur if the remote side only
             * wants to receive media but never send to us.
             */
            if (!rtp->ice_active_remote_candidates && !rtp->ice_proposed_remote_candidates) {
                if (rtcp) {
                    ast_sockaddr_copy(&rtp->rtcp->them, sa);
                } else {
                    ast_rtp_instance_set_remote_address(instance, sa);
                }
            }
            return 0;
        }
        rtp->passthrough = 0;
    }
#endif
 
    return len;
}

References ao2_lock, ao2_ref, ao2_unlock, ast_debug_dtls, ast_log, ast_recvfrom(), AST_RTP_DTLS_CONNECTION_EXISTING, AST_RTP_DTLS_SETUP_ACTPASS, AST_RTP_DTLS_SETUP_PASSIVE, AST_RTP_ICE_COMPONENT_RTCP, AST_RTP_ICE_COMPONENT_RTP, ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_rtp_instance_set_remote_address, ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_isnull(), ast_sockaddr_pj_sockaddr_cmp(), ast_sockaddr_stringify(), buf, error(), in, len(), LOG_ERROR, LOG_WARNING, ast_rtp::rtcp, RTP_DTLS_ESTABLISHED, ast_rtp::s, ast_rtcp::s, status, ast_rtcp::them, TRANSPORT_SOCKET_RTCP, and TRANSPORT_SOCKET_RTP.

Referenced by rtcp_recvfrom(), and rtp_recvfrom().

__rtp_sendto()

static int __rtp_sendto ( struct ast_rtp_instance *  instance, void *  buf, size_t  size, int  flags, struct ast_sockaddr *  sa, int  rtcp, int *  via_ice, int  use_srtp  )

static

Precondition

instance is locked

Definition at line 3465 of file res_rtp_asterisk.c.

{
    int len = size;
    void *temp = buf;
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_rtp_instance *transport = rtp->bundled ? rtp->bundled : instance;
    struct ast_rtp *transport_rtp = ast_rtp_instance_get_data(transport);
    struct ast_srtp *srtp = ast_rtp_instance_get_srtp(transport, rtcp);
    int res;
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
    char *out = buf;
    struct dtls_details *dtls = (!rtcp || rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_MUX) ? &rtp->dtls : &rtp->rtcp->dtls;
 
    /* Don't send RTP if DTLS hasn't finished yet */
    if (dtls->ssl && ((*out < 20) || (*out > 63)) && dtls->connection == AST_RTP_DTLS_CONNECTION_NEW) {
        *via_ice = 0;
        return 0;
    }
#endif
 
    *via_ice = 0;
 
    if (use_srtp && res_srtp && srtp && res_srtp->protect(srtp, &temp, &len, rtcp) < 0) {
        return -1;
    }
 
#ifdef HAVE_PJPROJECT
    if (transport_rtp->ice) {
        enum ast_rtp_ice_component_type component = rtcp ? AST_RTP_ICE_COMPONENT_RTCP : AST_RTP_ICE_COMPONENT_RTP;
        pj_status_t status;
        struct ice_wrap *ice;
 
        /* If RTCP is sharing the same socket then use the same component */
        if (rtcp && rtp->rtcp->s == rtp->s) {
            component = AST_RTP_ICE_COMPONENT_RTP;
        }
 
        pj_thread_register_check();
 
        /* Release the instance lock to avoid deadlock with PJPROJECT group lock */
        ice = transport_rtp->ice;
        ao2_ref(ice, +1);
        ao2_ref(transport, +1);
        ao2_unlock(instance);
        status = pj_ice_sess_send_data(ice->real_ice, component, temp, len);
        ao2_ref(ice, -1);
        ao2_lock(instance);
        if (status == PJ_SUCCESS) {
            *via_ice = 1;
            ao2_ref(transport, -1);
            return len;
        }
        if (transport != (rtp->bundled ? rtp->bundled : instance)) {
            /*
             * In case the transport was bundled or un-bundled while we were unlocked don't
             * fall through to sending using the transport instance as we may no longer be
             * associated with it.
             */
            ao2_ref(transport, -1);
            return 0;
        }
        ao2_ref(transport, -1);
    }
#endif
 
    res = ast_sendto(rtcp ? transport_rtp->rtcp->s : transport_rtp->s, temp, len, flags, sa);
    if (res > 0) {
        ast_rtp_instance_set_last_tx(instance, time(NULL));
    }
 
    return res;
}

References ao2_lock, ao2_ref, ao2_unlock, AST_RTP_DTLS_CONNECTION_NEW, AST_RTP_ICE_COMPONENT_RTCP, AST_RTP_ICE_COMPONENT_RTP, ast_rtp_instance_get_data(), ast_rtp_instance_get_srtp(), AST_RTP_INSTANCE_RTCP_MUX, ast_rtp_instance_set_last_tx(), ast_sendto(), buf, ast_rtp::bundled, len(), NULL, out, ast_srtp_res::protect, res_srtp, ast_rtp::rtcp, ast_rtp::s, ast_rtcp::s, status, and ast_rtcp::type.

Referenced by rtcp_sendto(), and rtp_sendto().

__unreg_module()

static void __unreg_module ( void  )

static

Definition at line 10798 of file res_rtp_asterisk.c.

AST_MODULE_SELF_SYM()

struct ast_module * AST_MODULE_SELF_SYM

(

void

)

Definition at line 10798 of file res_rtp_asterisk.c.

ast_rtcp_calc_interval()

static unsigned int ast_rtcp_calc_interval ( struct ast_rtp *  rtp )

static

Todo:

XXX Do a more reasonable calculation on this one Look in RFC 3550 Section A.7 for an example

Definition at line 3559 of file res_rtp_asterisk.c.

{
    unsigned int interval;
    /*! \todo XXX Do a more reasonable calculation on this one
     * Look in RFC 3550 Section A.7 for an example*/
    interval = rtcpinterval;
    return interval;
}

References rtcpinterval.

Referenced by ast_rtp_interpret(), and rtp_raw_write().

ast_rtcp_calculate_sr_rr_statistics()

static int ast_rtcp_calculate_sr_rr_statistics ( struct ast_rtp_instance *  instance, struct ast_rtp_rtcp_report *  rtcp_report, struct ast_sockaddr  remote_address, int  ice, int  sr  )

static

Definition at line 4969 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_rtp_rtcp_report_block *report_block = NULL;
    RAII_VAR(struct ast_json *, message_blob, NULL, ast_json_unref);
 
    if (!rtp || !rtp->rtcp) {
        return 0;
    }
 
    if (ast_sockaddr_isnull(&rtp->rtcp->them)) {
        return 0;
    }
 
    if (!rtcp_report) {
        return -1;
    }
 
    report_block = rtcp_report->report_block[0];
 
    if (sr) {
        rtp->rtcp->txlsr = rtcp_report->sender_information.ntp_timestamp;
        rtp->rtcp->sr_count++;
        rtp->rtcp->lastsrtxcount = rtp->txcount;
    } else {
        rtp->rtcp->rr_count++;
    }
 
    if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
        ast_verbose("* Sent RTCP %s to %s%s\n", sr ? "SR" : "RR",
                ast_sockaddr_stringify(&remote_address), ice ? " (via ICE)" : "");
        ast_verbose("  Our SSRC: %u\n", rtcp_report->ssrc);
        if (sr) {
            ast_verbose("  Sent(NTP): %u.%06u\n",
                (unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_sec,
                (unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_usec);
            ast_verbose("  Sent(RTP): %u\n", rtcp_report->sender_information.rtp_timestamp);
            ast_verbose("  Sent packets: %u\n", rtcp_report->sender_information.packet_count);
            ast_verbose("  Sent octets: %u\n", rtcp_report->sender_information.octet_count);
        }
        if (report_block) {
            int rate = ast_rtp_get_rate(rtp->f.subclass.format);
            ast_verbose("  Report block:\n");
            ast_verbose("    Their SSRC: %u\n", report_block->source_ssrc);
            ast_verbose("    Fraction lost: %d\n", report_block->lost_count.fraction);
            ast_verbose("    Cumulative loss: %u\n", report_block->lost_count.packets);
            ast_verbose("    Highest seq no: %u\n", report_block->highest_seq_no);
            ast_verbose("    IA jitter (samp): %u\n", report_block->ia_jitter);
            ast_verbose("    IA jitter (secs): %.6f\n", ast_samp2sec(report_block->ia_jitter, rate));
            ast_verbose("    Their last SR: %u\n", report_block->lsr);
            ast_verbose("    DLSR: %4.4f (sec)\n\n", (double)(report_block->dlsr / 65536.0));
        }
    }
 
    message_blob = ast_json_pack("{s: s, s: s, s: f}",
            "to", ast_sockaddr_stringify(&remote_address),
            "from", rtp->rtcp->local_addr_str,
            "mes", rtp->rxmes);
 
    ast_rtp_publish_rtcp_message(instance, ast_rtp_rtcp_sent_type(),
            rtcp_report, message_blob);
 
    return 1;
}

References ast_json_pack(), ast_json_unref(), ast_rtp_get_rate(), ast_rtp_instance_get_data(), ast_rtp_publish_rtcp_message(), ast_rtp_rtcp_sent_type(), ast_samp2sec(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_verbose, ast_rtp_rtcp_report_block::dlsr, ast_rtp::f, ast_frame_subclass::format, ast_rtp_rtcp_report_block::fraction, ast_rtp_rtcp_report_block::highest_seq_no, ast_rtp_rtcp_report_block::ia_jitter, ast_rtcp::lastsrtxcount, ast_rtcp::local_addr_str, ast_rtp_rtcp_report_block::lost_count, ast_rtp_rtcp_report_block::lsr, ast_rtp_rtcp_report::ntp_timestamp, NULL, ast_rtp_rtcp_report::octet_count, ast_rtp_rtcp_report::packet_count, ast_rtp_rtcp_report_block::packets, RAII_VAR, ast_rtp_rtcp_report::report_block, ast_rtcp::rr_count, ast_rtp::rtcp, rtcp_debug_test_addr(), ast_rtp_rtcp_report::rtp_timestamp, ast_rtp::rxmes, ast_rtp_rtcp_report::sender_information, ast_rtp_rtcp_report_block::source_ssrc, ast_rtcp::sr_count, ast_rtp_rtcp_report::ssrc, ast_frame::subclass, ast_rtcp::them, ast_rtp::txcount, and ast_rtcp::txlsr.

Referenced by ast_rtcp_write(), ast_rtp_read(), rtp_write_rtcp_fir(), and rtp_write_rtcp_psfb().

ast_rtcp_generate_compound_prefix()

static int ast_rtcp_generate_compound_prefix ( struct ast_rtp_instance *  instance, unsigned char *  rtcpheader, struct ast_rtp_rtcp_report *  report, int *  sr  )

static

Definition at line 5093 of file res_rtp_asterisk.c.

{
    int packet_len = 0;
    int res;
 
    /* Every RTCP packet needs to be sent out with a SR/RR and SDES prefixing it.
     * At the end of this function, rtcpheader should contain both of those packets,
     * and will return the length of the overall packet. This can be used to determine
     * where further packets can be inserted in the compound packet.
     */
    res = ast_rtcp_generate_report(instance, rtcpheader, report, sr);
 
    if (res == 0 || res == 1) {
        ast_debug_rtcp(1, "(%p) RTCP failed to generate %s report!\n", instance, sr ? "SR" : "RR");
        return 0;
    }
 
    packet_len += res;
 
    res = ast_rtcp_generate_sdes(instance, rtcpheader + packet_len, report);
 
    if (res == 0 || res == 1) {
        ast_debug_rtcp(1, "(%p) RTCP failed to generate SDES!\n", instance);
        return 0;
    }
 
    return packet_len + res;
}

References ast_debug_rtcp, ast_rtcp_generate_report(), and ast_rtcp_generate_sdes().

Referenced by ast_rtcp_write(), ast_rtp_read(), rtp_write_rtcp_fir(), and rtp_write_rtcp_psfb().

ast_rtcp_generate_nack()

static int ast_rtcp_generate_nack ( struct ast_rtp_instance *  instance, unsigned char *  rtcpheader  )

static

Definition at line 5123 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    int packet_len;
    int blp_index = -1;
    int current_seqno;
    unsigned int fci = 0;
    size_t remaining_missing_seqno;
 
    if (!rtp || !rtp->rtcp) {
        return 0;
    }
 
    if (ast_sockaddr_isnull(&rtp->rtcp->them)) {
        return 0;
    }
 
    current_seqno = rtp->expectedrxseqno;
    remaining_missing_seqno = AST_VECTOR_SIZE(&rtp->missing_seqno);
    packet_len = 12; /* The header length is 12 (version line, packet source SSRC, media source SSRC) */
 
    /* If there are no missing sequence numbers then don't bother sending a NACK needlessly */
    if (!remaining_missing_seqno) {
        return 0;
    }
 
    /* This iterates through the possible forward sequence numbers seeing which ones we
     * have no packet for, adding it to the NACK until we are out of missing packets.
     */
    while (remaining_missing_seqno) {
        int *missing_seqno;
 
        /* On the first entry to this loop blp_index will be -1, so this will become 0
         * and the sequence number will be placed into the packet as the PID.
         */
        blp_index++;
 
        missing_seqno = AST_VECTOR_GET_CMP(&rtp->missing_seqno, current_seqno,
                find_by_value);
        if (missing_seqno) {
            /* We hit the max blp size, reset */
            if (blp_index >= 17) {
                put_unaligned_uint32(rtcpheader + packet_len, htonl(fci));
                fci = 0;
                blp_index = 0;
                packet_len += 4;
            }
 
            if (blp_index == 0) {
                fci |= (current_seqno << 16);
            } else {
                fci |= (1 << (blp_index - 1));
            }
 
            /* Since we've used a missing sequence number, we're down one */
            remaining_missing_seqno--;
        }
 
        /* Handle cycling of the sequence number */
        current_seqno++;
        if (current_seqno == SEQNO_CYCLE_OVER) {
            current_seqno = 0;
        }
    }
 
    put_unaligned_uint32(rtcpheader + packet_len, htonl(fci));
    packet_len += 4;
 
    /* Length MUST be 2+n, where n is the number of NACKs. Same as length in words minus 1 */
    put_unaligned_uint32(rtcpheader, htonl((2 << 30) | (AST_RTP_RTCP_FMT_NACK << 24)
                | (AST_RTP_RTCP_RTPFB << 16) | ((packet_len / 4) - 1)));
    put_unaligned_uint32(rtcpheader + 4, htonl(rtp->ssrc));
    put_unaligned_uint32(rtcpheader + 8, htonl(rtp->themssrc));
 
    return packet_len;
}

References ast_rtp_instance_get_data(), AST_RTP_RTCP_FMT_NACK, AST_RTP_RTCP_RTPFB, ast_sockaddr_isnull(), AST_VECTOR_GET_CMP, AST_VECTOR_SIZE, ast_rtp::expectedrxseqno, find_by_value(), ast_rtp::missing_seqno, put_unaligned_uint32(), ast_rtp::rtcp, SEQNO_CYCLE_OVER, ast_rtp::ssrc, ast_rtcp::them, and ast_rtp::themssrc.

Referenced by ast_rtp_read().

ast_rtcp_generate_report()

static int ast_rtcp_generate_report ( struct ast_rtp_instance *  instance, unsigned char *  rtcpheader, struct ast_rtp_rtcp_report *  rtcp_report, int *  sr  )

static

Definition at line 4876 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    int len = 0;
    struct timeval now;
    unsigned int now_lsw;
    unsigned int now_msw;
    unsigned int lost_packets;
    int fraction_lost;
    struct timeval dlsr = { 0, };
    struct ast_rtp_rtcp_report_block *report_block = NULL;
 
    if (!rtp || !rtp->rtcp) {
        return 0;
    }
 
    if (ast_sockaddr_isnull(&rtp->rtcp->them)) { /* This'll stop rtcp for this rtp session */
        /* RTCP was stopped. */
        return 0;
    }
 
    if (!rtcp_report) {
        return 1;
    }
 
    *sr = rtp->txcount > rtp->rtcp->lastsrtxcount ? 1 : 0;
 
    /* Compute statistics */
    calculate_lost_packet_statistics(rtp, &lost_packets, &fraction_lost);
    /*
     * update_local_mes_stats must be called AFTER
     * calculate_lost_packet_statistics
     */
    update_local_mes_stats(rtp);
 
    gettimeofday(&now, NULL);
    rtcp_report->reception_report_count = rtp->themssrc_valid ? 1 : 0;
    rtcp_report->ssrc = rtp->ssrc;
    rtcp_report->type = *sr ? RTCP_PT_SR : RTCP_PT_RR;
    if (*sr) {
        rtcp_report->sender_information.ntp_timestamp = now;
        rtcp_report->sender_information.rtp_timestamp = rtp->lastts;
        rtcp_report->sender_information.packet_count = rtp->txcount;
        rtcp_report->sender_information.octet_count = rtp->txoctetcount;
    }
 
    if (rtp->themssrc_valid) {
        report_block = ast_calloc(1, sizeof(*report_block));
        if (!report_block) {
            return 1;
        }
 
        rtcp_report->report_block[0] = report_block;
        report_block->source_ssrc = rtp->themssrc;
        report_block->lost_count.fraction = (fraction_lost & 0xff);
        report_block->lost_count.packets = (lost_packets & 0xffffff);
        report_block->highest_seq_no = (rtp->cycles | (rtp->lastrxseqno & 0xffff));
        report_block->ia_jitter = (unsigned int)rtp->rxjitter_samples;
        report_block->lsr = rtp->rtcp->themrxlsr;
        /* If we haven't received an SR report, DLSR should be 0 */
        if (!ast_tvzero(rtp->rtcp->rxlsr)) {
            timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
            report_block->dlsr = (((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000;
        }
    }
    timeval2ntp(rtcp_report->sender_information.ntp_timestamp, &now_msw, &now_lsw);
    put_unaligned_uint32(rtcpheader + 4, htonl(rtcp_report->ssrc)); /* Our SSRC */
    len += 8;
    if (*sr) {
        put_unaligned_uint32(rtcpheader + len, htonl(now_msw)); /* now, MSW. gettimeofday() + SEC_BETWEEN_1900_AND_1970 */
        put_unaligned_uint32(rtcpheader + len + 4, htonl(now_lsw)); /* now, LSW */
        put_unaligned_uint32(rtcpheader + len + 8, htonl(rtcp_report->sender_information.rtp_timestamp));
        put_unaligned_uint32(rtcpheader + len + 12, htonl(rtcp_report->sender_information.packet_count));
        put_unaligned_uint32(rtcpheader + len + 16, htonl(rtcp_report->sender_information.octet_count));
        len += 20;
    }
    if (report_block) {
        put_unaligned_uint32(rtcpheader + len, htonl(report_block->source_ssrc)); /* Their SSRC */
        put_unaligned_uint32(rtcpheader + len + 4, htonl((report_block->lost_count.fraction << 24) | report_block->lost_count.packets));
        put_unaligned_uint32(rtcpheader + len + 8, htonl(report_block->highest_seq_no));
        put_unaligned_uint32(rtcpheader + len + 12, htonl(report_block->ia_jitter));
        put_unaligned_uint32(rtcpheader + len + 16, htonl(report_block->lsr));
        put_unaligned_uint32(rtcpheader + len + 20, htonl(report_block->dlsr));
        len += 24;
    }
 
    put_unaligned_uint32(rtcpheader, htonl((2 << 30) | (rtcp_report->reception_report_count << 24)
                | ((*sr ? RTCP_PT_SR : RTCP_PT_RR) << 16) | ((len/4)-1)));
 
    return len;
}

References ast_calloc, ast_rtp_instance_get_data(), ast_sockaddr_isnull(), ast_tvzero(), calculate_lost_packet_statistics(), ast_rtp::cycles, ast_rtp_rtcp_report_block::dlsr, ast_rtp_rtcp_report_block::fraction, ast_rtp_rtcp_report_block::highest_seq_no, ast_rtp_rtcp_report_block::ia_jitter, ast_rtp::lastrxseqno, ast_rtcp::lastsrtxcount, ast_rtp::lastts, len(), ast_rtp_rtcp_report_block::lost_count, ast_rtp_rtcp_report_block::lsr, ast_rtp_rtcp_report::ntp_timestamp, NULL, ast_rtp_rtcp_report::octet_count, ast_rtp_rtcp_report::packet_count, ast_rtp_rtcp_report_block::packets, put_unaligned_uint32(), ast_rtp_rtcp_report::reception_report_count, ast_rtp_rtcp_report::report_block, ast_rtp::rtcp, RTCP_PT_RR, RTCP_PT_SR, ast_rtp_rtcp_report::rtp_timestamp, ast_rtp::rxjitter_samples, ast_rtcp::rxlsr, ast_rtp_rtcp_report::sender_information, ast_rtp_rtcp_report_block::source_ssrc, ast_rtp_rtcp_report::ssrc, ast_rtp::ssrc, ast_rtcp::them, ast_rtcp::themrxlsr, ast_rtp::themssrc, ast_rtp::themssrc_valid, timersub(), timeval2ntp(), ast_rtp::txcount, ast_rtp::txoctetcount, ast_rtp_rtcp_report::type, and update_local_mes_stats().

Referenced by ast_rtcp_generate_compound_prefix().

ast_rtcp_generate_sdes()

static int ast_rtcp_generate_sdes ( struct ast_rtp_instance *  instance, unsigned char *  rtcpheader, struct ast_rtp_rtcp_report *  rtcp_report  )

static

Definition at line 5035 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    int len = 0;
    uint16_t sdes_packet_len_bytes;
    uint16_t sdes_packet_len_rounded;
 
    if (!rtp || !rtp->rtcp) {
        return 0;
    }
 
    if (ast_sockaddr_isnull(&rtp->rtcp->them)) {
        return 0;
    }
 
    if (!rtcp_report) {
        return -1;
    }
 
    sdes_packet_len_bytes =
        4 + /* RTCP Header */
        4 + /* SSRC */
        1 + /* Type (CNAME) */
        1 + /* Text Length */
        AST_UUID_STR_LEN /* Text and NULL terminator */
        ;
 
    /* Round to 32 bit boundary */
    sdes_packet_len_rounded = (sdes_packet_len_bytes + 3) & ~0x3;
 
    put_unaligned_uint32(rtcpheader, htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | ((sdes_packet_len_rounded / 4) - 1)));
    put_unaligned_uint32(rtcpheader + 4, htonl(rtcp_report->ssrc));
    rtcpheader[8] = 0x01; /* CNAME */
    rtcpheader[9] = AST_UUID_STR_LEN - 1; /* Number of bytes of text */
    memcpy(rtcpheader + 10, rtp->cname, AST_UUID_STR_LEN);
    len += 10 + AST_UUID_STR_LEN;
 
    /* Padding - Note that we don't set the padded bit on the packet. From
     * RFC 3550 Section 6.5:
     *
     *   No length octet follows the null item type octet, but additional null
     *   octets MUST be included if needd to pad until the next 32-bit
     *   boundary. Note that this padding is separate from that indicated by
     *   the P bit in the RTCP header.
     *
     * These bytes will already be zeroed out during array initialization.
     */
    len += (sdes_packet_len_rounded - sdes_packet_len_bytes);
 
    return len;
}

References ast_rtp_instance_get_data(), ast_sockaddr_isnull(), AST_UUID_STR_LEN, ast_rtp::cname, len(), put_unaligned_uint32(), ast_rtp::rtcp, RTCP_PT_SDES, ast_rtp_rtcp_report::ssrc, and ast_rtcp::them.

Referenced by ast_rtcp_generate_compound_prefix().

ast_rtcp_interpret()

static struct ast_frame * ast_rtcp_interpret ( struct ast_rtp_instance *  instance, struct ast_srtp *  srtp, const unsigned char *  rtcpdata, size_t  size, struct ast_sockaddr *  addr  )

static

True if we have seen an acceptable SSRC to learn the remote RTCP address

True if the ssrc value we have is valid and not garbage because it doesn't exist.

Always use packet source SSRC to find the rtp instance unless explicitly told not to.

Definition at line 6863 of file res_rtp_asterisk.c.

{
    struct ast_rtp_instance *transport = instance;
    struct ast_rtp *transport_rtp = ast_rtp_instance_get_data(instance);
    int len = size;
    unsigned int *rtcpheader = (unsigned int *)(rtcpdata);
    unsigned int packetwords;
    unsigned int position;
    unsigned int first_word;
    /*! True if we have seen an acceptable SSRC to learn the remote RTCP address */
    unsigned int ssrc_seen;
    struct ast_rtp_rtcp_report_block *report_block;
    struct ast_frame *f = &ast_null_frame;
#ifdef TEST_FRAMEWORK
    struct ast_rtp_engine_test *test_engine;
#endif
 
    /* If this is encrypted then decrypt the payload */
    if ((*rtcpheader & 0xC0) && res_srtp && srtp && res_srtp->unprotect(
            srtp, rtcpheader, &len, 1 | (srtp_replay_protection << 1)) < 0) {
       return &ast_null_frame;
    }
 
    packetwords = len / 4;
 
    ast_debug_rtcp(2, "(%s) RTCP got report of %d bytes from %s\n",
        ast_rtp_instance_get_channel_id(instance),
        len, ast_sockaddr_stringify(addr));
 
    /*
     * Validate the RTCP packet according to an adapted and slightly
     * modified RFC3550 validation algorithm.
     */
    if (packetwords < RTCP_HEADER_SSRC_LENGTH) {
        ast_debug_rtcp(2, "(%s) RTCP %p -- from %s: Frame size (%u words) is too short\n",
            ast_rtp_instance_get_channel_id(instance),
            transport_rtp, ast_sockaddr_stringify(addr), packetwords);
        return &ast_null_frame;
    }
    position = 0;
    first_word = ntohl(rtcpheader[position]);
    if ((first_word & RTCP_VALID_MASK) != RTCP_VALID_VALUE) {
        ast_debug_rtcp(2, "(%s) RTCP %p -- from %s: Failed first packet validity check\n",
            ast_rtp_instance_get_channel_id(instance),
            transport_rtp, ast_sockaddr_stringify(addr));
        return &ast_null_frame;
    }
    do {
        position += ((first_word >> RTCP_LENGTH_SHIFT) & RTCP_LENGTH_MASK) + 1;
        if (packetwords <= position) {
            break;
        }
        first_word = ntohl(rtcpheader[position]);
    } while ((first_word & RTCP_VERSION_MASK_SHIFTED) == RTCP_VERSION_SHIFTED);
    if (position != packetwords) {
        ast_debug_rtcp(2, "(%s) RTCP %p -- from %s: Failed packet version or length check\n",
            ast_rtp_instance_get_channel_id(instance),
            transport_rtp, ast_sockaddr_stringify(addr));
        return &ast_null_frame;
    }
 
    /*
     * Note: RFC3605 points out that true NAT (vs NAPT) can cause RTCP
     * to have a different IP address and port than RTP.  Otherwise, when
     * strictrtp is enabled we could reject RTCP packets not coming from
     * the learned RTP IP address if it is available.
     */
 
    /*
     * strictrtp safety needs SSRC to match before we use the
     * sender's address for symmetrical RTP to send our RTCP
     * reports.
     *
     * If strictrtp is not enabled then claim to have already seen
     * a matching SSRC so we'll accept this packet's address for
     * symmetrical RTP.
     */
    ssrc_seen = transport_rtp->strict_rtp_state == STRICT_RTP_OPEN;
 
    position = 0;
    while (position < packetwords) {
        unsigned int i;
        unsigned int pt;
        unsigned int rc;
        unsigned int ssrc;
        /*! True if the ssrc value we have is valid and not garbage because it doesn't exist. */
        unsigned int ssrc_valid;
        unsigned int length;
        unsigned int min_length;
        /*! Always use packet source SSRC to find the rtp instance unless explicitly told not to. */
        unsigned int use_packet_source = 1;
 
        struct ast_json *message_blob;
        RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report, NULL, ao2_cleanup);
        struct ast_rtp_instance *child;
        struct ast_rtp *rtp;
        struct ast_rtp_rtcp_feedback *feedback;
 
        i = position;
        first_word = ntohl(rtcpheader[i]);
        pt = (first_word >> RTCP_PAYLOAD_TYPE_SHIFT) & RTCP_PAYLOAD_TYPE_MASK;
        rc = (first_word >> RTCP_REPORT_COUNT_SHIFT) & RTCP_REPORT_COUNT_MASK;
        /* RFC3550 says 'length' is the number of words in the packet - 1 */
        length = ((first_word >> RTCP_LENGTH_SHIFT) & RTCP_LENGTH_MASK) + 1;
 
        /* Check expected RTCP packet record length */
        min_length = RTCP_HEADER_SSRC_LENGTH;
        switch (pt) {
        case RTCP_PT_SR:
            min_length += RTCP_SR_BLOCK_WORD_LENGTH;
            /* fall through */
        case RTCP_PT_RR:
            min_length += (rc * RTCP_RR_BLOCK_WORD_LENGTH);
            use_packet_source = 0;
            break;
        case RTCP_PT_FUR:
            break;
        case AST_RTP_RTCP_RTPFB:
            switch (rc) {
            case AST_RTP_RTCP_FMT_NACK:
                min_length += RTCP_FB_NACK_BLOCK_WORD_LENGTH;
                break;
            default:
                break;
            }
            use_packet_source = 0;
            break;
        case RTCP_PT_PSFB:
            switch (rc) {
            case AST_RTP_RTCP_FMT_REMB:
                min_length += RTCP_FB_REMB_BLOCK_WORD_LENGTH;
                break;
            default:
                break;
            }
            break;
        case RTCP_PT_SDES:
        case RTCP_PT_BYE:
            /*
             * There may not be a SSRC/CSRC present.  The packet is
             * useless but still valid if it isn't present.
             *
             * We don't know what min_length should be so disable the check
             */
            min_length = length;
            break;
        default:
            ast_debug_rtcp(1, "(%p) RTCP %p -- from %s: %u(%s) skipping record\n",
                instance, transport_rtp, ast_sockaddr_stringify(addr), pt, rtcp_payload_type2str(pt));
            if (rtcp_debug_test_addr(addr)) {
                ast_verbose("\n");
                ast_verbose("RTCP from %s: %u(%s) skipping record\n",
                    ast_sockaddr_stringify(addr), pt, rtcp_payload_type2str(pt));
            }
            position += length;
            continue;
        }
        if (length < min_length) {
            ast_debug_rtcp(1, "(%p) RTCP %p -- from %s: %u(%s) length field less than expected minimum.  Min:%u Got:%u\n",
                instance, transport_rtp, ast_sockaddr_stringify(addr), pt, rtcp_payload_type2str(pt),
                min_length - 1, length - 1);
            return &ast_null_frame;
        }
 
        /* Get the RTCP record SSRC if defined for the record */
        ssrc_valid = 1;
        switch (pt) {
        case RTCP_PT_SR:
        case RTCP_PT_RR:
            rtcp_report = ast_rtp_rtcp_report_alloc(rc);
            if (!rtcp_report) {
                return &ast_null_frame;
            }
            rtcp_report->reception_report_count = rc;
 
            ssrc = ntohl(rtcpheader[i + 2]);
            rtcp_report->ssrc = ssrc;
            break;
        case RTCP_PT_FUR:
        case RTCP_PT_PSFB:
            ssrc = ntohl(rtcpheader[i + 1]);
            break;
        case AST_RTP_RTCP_RTPFB:
            ssrc = ntohl(rtcpheader[i + 2]);
            break;
        case RTCP_PT_SDES:
        case RTCP_PT_BYE:
        default:
            ssrc = 0;
            ssrc_valid = 0;
            break;
        }
 
        if (rtcp_debug_test_addr(addr)) {
            const char *subtype = rtcp_payload_subtype2str(pt, rc);
 
            ast_verbose("\n");
            ast_verbose("RTCP from %s\n", ast_sockaddr_stringify(addr));
            ast_verbose("PT: %u (%s)\n", pt, rtcp_payload_type2str(pt));
            if (subtype) {
                ast_verbose("Packet Subtype: %u (%s)\n", rc, subtype);
            } else {
                ast_verbose("Reception reports: %u\n", rc);
            }
            ast_verbose("SSRC of sender: %u\n", ssrc);
        }
 
        /* Determine the appropriate instance for this */
        if (ssrc_valid) {
            /*
             * Depending on the payload type, either the packet source or media source
             * SSRC is used.
             */
            if (use_packet_source) {
                child = rtp_find_instance_by_packet_source_ssrc(transport, transport_rtp, ssrc);
            } else {
                child = rtp_find_instance_by_media_source_ssrc(transport, transport_rtp, ssrc);
            }
            if (child && child != transport) {
                /*
                 * It is safe to hold the child lock while holding the parent lock.
                 * We guarantee that the locking order is always parent->child or
                 * that the child lock is not held when acquiring the parent lock.
                 */
                ao2_lock(child);
                instance = child;
                rtp = ast_rtp_instance_get_data(instance);
            } else {
                /* The child is the parent! We don't need to unlock it. */
                child = NULL;
                rtp = transport_rtp;
            }
        } else {
            child = NULL;
            rtp = transport_rtp;
        }
 
        if (ssrc_valid && rtp->themssrc_valid) {
            /*
             * If the SSRC is 1, we still need to handle RTCP since this could be a
             * special case. For example, if we have a unidirectional video stream, the
             * SSRC may be set to 1 by the browser (in the case of chromium), and requests
             * will still need to be processed so that video can flow as expected. This
             * should only be done for PLI and FUR, since there is not a way to get the
             * appropriate rtp instance when the SSRC is 1.
             */
            int exception = (ssrc == 1 && !((pt == RTCP_PT_PSFB && rc == AST_RTP_RTCP_FMT_PLI) || pt == RTCP_PT_FUR));
            if ((ssrc != rtp->themssrc && use_packet_source && ssrc != 1)
                    || exception) {
                /*
                 * Skip over this RTCP record as it does not contain the
                 * correct SSRC.  We should not act upon RTCP records
                 * for a different stream.
                 */
                position += length;
                ast_debug_rtcp(1, "(%p) RTCP %p -- from %s: Skipping record, received SSRC '%u' != expected '%u'\n",
                    instance, rtp, ast_sockaddr_stringify(addr), ssrc, rtp->themssrc);
                if (child) {
                    ao2_unlock(child);
                }
                continue;
            }
            ssrc_seen = 1;
        }
 
        if (ssrc_seen && ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
            /* Send to whoever sent to us */
            if (ast_sockaddr_cmp(&rtp->rtcp->them, addr)) {
                ast_sockaddr_copy(&rtp->rtcp->them, addr);
                if (ast_debug_rtp_packet_is_allowed) {
                    ast_debug(0, "(%p) RTCP NAT: Got RTCP from other end. Now sending to address %s\n",
                        instance, ast_sockaddr_stringify(addr));
                }
            }
        }
 
        i += RTCP_HEADER_SSRC_LENGTH; /* Advance past header and ssrc */
        switch (pt) {
        case RTCP_PT_SR:
            gettimeofday(&rtp->rtcp->rxlsr, NULL);
            rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16);
            rtp->rtcp->spc = ntohl(rtcpheader[i + 3]);
            rtp->rtcp->soc = ntohl(rtcpheader[i + 4]);
 
            rtcp_report->type = RTCP_PT_SR;
            rtcp_report->sender_information.packet_count = rtp->rtcp->spc;
            rtcp_report->sender_information.octet_count = rtp->rtcp->soc;
            ntp2timeval((unsigned int)ntohl(rtcpheader[i]),
                    (unsigned int)ntohl(rtcpheader[i + 1]),
                    &rtcp_report->sender_information.ntp_timestamp);
            rtcp_report->sender_information.rtp_timestamp = ntohl(rtcpheader[i + 2]);
            if (rtcp_debug_test_addr(addr)) {
                ast_verbose("NTP timestamp: %u.%06u\n",
                        (unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_sec,
                        (unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_usec);
                ast_verbose("RTP timestamp: %u\n", rtcp_report->sender_information.rtp_timestamp);
                ast_verbose("SPC: %u\tSOC: %u\n",
                        rtcp_report->sender_information.packet_count,
                        rtcp_report->sender_information.octet_count);
            }
            i += RTCP_SR_BLOCK_WORD_LENGTH;
            /* Intentional fall through */
        case RTCP_PT_RR:
            if (rtcp_report->type != RTCP_PT_SR) {
                rtcp_report->type = RTCP_PT_RR;
            }
 
            if (rc > 0) {
                /* Don't handle multiple reception reports (rc > 1) yet */
                report_block = ast_calloc(1, sizeof(*report_block));
                if (!report_block) {
                    if (child) {
                        ao2_unlock(child);
                    }
                    return &ast_null_frame;
                }
                rtcp_report->report_block[0] = report_block;
                report_block->source_ssrc = ntohl(rtcpheader[i]);
                report_block->lost_count.packets = ntohl(rtcpheader[i + 1]) & 0x00ffffff;
                report_block->lost_count.fraction = ((ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24);
                report_block->highest_seq_no = ntohl(rtcpheader[i + 2]);
                report_block->ia_jitter =  ntohl(rtcpheader[i + 3]);
                report_block->lsr = ntohl(rtcpheader[i + 4]);
                report_block->dlsr = ntohl(rtcpheader[i + 5]);
                if (report_block->lsr) {
                    int skewed = update_rtt_stats(rtp, report_block->lsr, report_block->dlsr);
                    if (skewed && rtcp_debug_test_addr(addr)) {
                        struct timeval now;
                        unsigned int lsr_now, lsw, msw;
                        gettimeofday(&now, NULL);
                        timeval2ntp(now, &msw, &lsw);
                        lsr_now = (((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16));
                        ast_verbose("Internal RTCP NTP clock skew detected: "
                            "lsr=%u, now=%u, dlsr=%u (%u:%03ums), "
                            "diff=%u\n",
                            report_block->lsr, lsr_now, report_block->dlsr, report_block->dlsr / 65536,
                            (report_block->dlsr % 65536) * 1000 / 65536,
                            report_block->dlsr - (lsr_now - report_block->lsr));
                    }
                }
                update_jitter_stats(rtp, report_block->ia_jitter);
                update_lost_stats(rtp, report_block->lost_count.packets);
                /*
                 * update_reported_mes_stats must be called AFTER
                 * update_rtt_stats, update_jitter_stats and
                 * update_lost_stats.
                 */
                update_reported_mes_stats(rtp);
 
                if (rtcp_debug_test_addr(addr)) {
                    int rate = ast_rtp_get_rate(rtp->f.subclass.format);
 
                    ast_verbose("  Fraction lost: %d\n", report_block->lost_count.fraction);
                    ast_verbose("  Packets lost so far: %u\n", report_block->lost_count.packets);
                    ast_verbose("  Highest sequence number: %u\n", report_block->highest_seq_no & 0x0000ffff);
                    ast_verbose("  Sequence number cycles: %u\n", report_block->highest_seq_no >> 16);
                    ast_verbose("  Interarrival jitter (samp): %u\n", report_block->ia_jitter);
                    ast_verbose("  Interarrival jitter (secs): %.6f\n", ast_samp2sec(report_block->ia_jitter, rate));
                    ast_verbose("  Last SR(our NTP): %lu.%010lu\n",(unsigned long)(report_block->lsr) >> 16,((unsigned long)(report_block->lsr) << 16) * 4096);
                    ast_verbose("  DLSR: %4.4f (sec)\n",(double)report_block->dlsr / 65536.0);
                    ast_verbose("  RTT: %4.4f(sec)\n", rtp->rtcp->rtt);
                    ast_verbose("  MES: %4.1f\n", rtp->rtcp->reported_mes);
                }
            }
            /* If and when we handle more than one report block, this should occur outside
             * this loop.
             */
 
            message_blob = ast_json_pack("{s: s, s: s, s: f, s: f}",
                "from", ast_sockaddr_stringify(addr),
                "to", transport_rtp->rtcp->local_addr_str,
                "rtt", rtp->rtcp->rtt,
                "mes", rtp->rtcp->reported_mes);
            ast_rtp_publish_rtcp_message(instance, ast_rtp_rtcp_received_type(),
                    rtcp_report,
                    message_blob);
            ast_json_unref(message_blob);
 
            /* Return an AST_FRAME_RTCP frame with the ast_rtp_rtcp_report
             * object as a its data */
            transport_rtp->f.frametype = AST_FRAME_RTCP;
            transport_rtp->f.subclass.integer = pt;
            transport_rtp->f.data.ptr = rtp->rtcp->frame_buf + AST_FRIENDLY_OFFSET;
            memcpy(transport_rtp->f.data.ptr, rtcp_report, sizeof(struct ast_rtp_rtcp_report));
            transport_rtp->f.datalen = sizeof(struct ast_rtp_rtcp_report);
            if (rc > 0) {
                /* There's always a single report block stored, here */
                struct ast_rtp_rtcp_report *rtcp_report2;
                report_block = transport_rtp->f.data.ptr + transport_rtp->f.datalen + sizeof(struct ast_rtp_rtcp_report_block *);
                memcpy(report_block, rtcp_report->report_block[0], sizeof(struct ast_rtp_rtcp_report_block));
                rtcp_report2 = (struct ast_rtp_rtcp_report *)transport_rtp->f.data.ptr;
                rtcp_report2->report_block[0] = report_block;
                transport_rtp->f.datalen += sizeof(struct ast_rtp_rtcp_report_block);
            }
            transport_rtp->f.offset = AST_FRIENDLY_OFFSET;
            transport_rtp->f.samples = 0;
            transport_rtp->f.mallocd = 0;
            transport_rtp->f.delivery.tv_sec = 0;
            transport_rtp->f.delivery.tv_usec = 0;
            transport_rtp->f.src = "RTP";
            transport_rtp->f.stream_num = rtp->stream_num;
            f = &transport_rtp->f;
            break;
        case AST_RTP_RTCP_RTPFB:
            switch (rc) {
            case AST_RTP_RTCP_FMT_NACK:
                /* If retransmissions are not enabled ignore this message */
                if (!rtp->send_buffer) {
                    break;
                }
 
                if (rtcp_debug_test_addr(addr)) {
                    ast_verbose("Received generic RTCP NACK message\n");
                }
 
                ast_rtp_rtcp_handle_nack_locking(instance,
                    child ? transport : NULL, rtcpheader, position, length);
 
                break;
            default:
                break;
            }
            break;
        case RTCP_PT_FUR:
            /* Handle RTCP FUR as FIR by setting the format to 4 */
            rc = AST_RTP_RTCP_FMT_FIR;
        case RTCP_PT_PSFB:
            switch (rc) {
            case AST_RTP_RTCP_FMT_PLI:
            case AST_RTP_RTCP_FMT_FIR:
                if (rtcp_debug_test_addr(addr)) {
                    ast_verbose("Received an RTCP Fast Update Request\n");
                }
                transport_rtp->f.frametype = AST_FRAME_CONTROL;
                transport_rtp->f.subclass.integer = AST_CONTROL_VIDUPDATE;
                transport_rtp->f.datalen = 0;
                transport_rtp->f.samples = 0;
                transport_rtp->f.mallocd = 0;
                transport_rtp->f.src = "RTP";
                f = &transport_rtp->f;
                break;
            case AST_RTP_RTCP_FMT_REMB:
                /* If REMB support is not enabled ignore this message */
                if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_REMB)) {
                    break;
                }
 
                if (rtcp_debug_test_addr(addr)) {
                    ast_verbose("Received REMB report\n");
                }
                transport_rtp->f.frametype = AST_FRAME_RTCP;
                transport_rtp->f.subclass.integer = pt;
                transport_rtp->f.stream_num = rtp->stream_num;
                transport_rtp->f.data.ptr = rtp->rtcp->frame_buf + AST_FRIENDLY_OFFSET;
                feedback = transport_rtp->f.data.ptr;
                feedback->fmt = rc;
 
                /* We don't actually care about the SSRC information in the feedback message */
                first_word = ntohl(rtcpheader[i + 2]);
                feedback->remb.br_exp = (first_word >> 18) & ((1 << 6) - 1);
                feedback->remb.br_mantissa = first_word & ((1 << 18) - 1);
 
                transport_rtp->f.datalen = sizeof(struct ast_rtp_rtcp_feedback);
                transport_rtp->f.offset = AST_FRIENDLY_OFFSET;
                transport_rtp->f.samples = 0;
                transport_rtp->f.mallocd = 0;
                transport_rtp->f.delivery.tv_sec = 0;
                transport_rtp->f.delivery.tv_usec = 0;
                transport_rtp->f.src = "RTP";
                f = &transport_rtp->f;
                break;
            default:
                break;
            }
            break;
        case RTCP_PT_SDES:
            if (rtcp_debug_test_addr(addr)) {
                ast_verbose("Received an SDES from %s\n",
                    ast_sockaddr_stringify(addr));
            }
#ifdef TEST_FRAMEWORK
            if ((test_engine = ast_rtp_instance_get_test(instance))) {
                test_engine->sdes_received = 1;
            }
#endif
            break;
        case RTCP_PT_BYE:
            if (rtcp_debug_test_addr(addr)) {
                ast_verbose("Received a BYE from %s\n",
                    ast_sockaddr_stringify(addr));
            }
            break;
        default:
            break;
        }
        position += length;
        rtp->rtcp->rtcp_info = 1;
 
        if (child) {
            ao2_unlock(child);
        }
    }
 
    return f;
}

References ao2_cleanup, ao2_lock, ao2_unlock, ast_calloc, AST_CONTROL_VIDUPDATE, ast_debug, ast_debug_rtcp, ast_debug_rtp_packet_is_allowed, AST_FRAME_CONTROL, AST_FRAME_RTCP, AST_FRIENDLY_OFFSET, ast_json_pack(), ast_json_unref(), ast_null_frame, ast_rtp_get_rate(), ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_data(), ast_rtp_instance_get_prop(), AST_RTP_PROPERTY_NAT, AST_RTP_PROPERTY_REMB, ast_rtp_publish_rtcp_message(), AST_RTP_RTCP_FMT_FIR, AST_RTP_RTCP_FMT_NACK, AST_RTP_RTCP_FMT_PLI, AST_RTP_RTCP_FMT_REMB, ast_rtp_rtcp_handle_nack_locking(), ast_rtp_rtcp_received_type(), ast_rtp_rtcp_report_alloc(), AST_RTP_RTCP_RTPFB, ast_samp2sec(), ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_stringify(), ast_verbose, ast_rtp_rtcp_feedback_remb::br_exp, ast_rtp_rtcp_feedback_remb::br_mantissa, ast_frame::data, ast_frame::datalen, ast_frame::delivery, ast_rtp_rtcp_report_block::dlsr, ast_rtp::f, ast_rtp_rtcp_feedback::fmt, ast_frame_subclass::format, ast_rtp_rtcp_report_block::fraction, ast_rtcp::frame_buf, ast_frame::frametype, ast_rtp_rtcp_report_block::highest_seq_no, ast_rtp_rtcp_report_block::ia_jitter, ast_frame_subclass::integer, len(), ast_rtcp::local_addr_str, ast_rtp_rtcp_report_block::lost_count, ast_rtp_rtcp_report_block::lsr, ast_frame::mallocd, ntp2timeval(), NULL, ast_frame::offset, ast_rtp_rtcp_report_block::packets, ast_frame::ptr, RAII_VAR, ast_rtp_rtcp_feedback::remb, ast_rtp_rtcp_report::report_block, ast_rtcp::reported_mes, res_srtp, ast_rtp::rtcp, rtcp_debug_test_addr(), RTCP_FB_NACK_BLOCK_WORD_LENGTH, RTCP_FB_REMB_BLOCK_WORD_LENGTH, RTCP_HEADER_SSRC_LENGTH, ast_rtcp::rtcp_info, RTCP_LENGTH_MASK, RTCP_LENGTH_SHIFT, rtcp_payload_subtype2str(), rtcp_payload_type2str(), RTCP_PAYLOAD_TYPE_MASK, RTCP_PAYLOAD_TYPE_SHIFT, RTCP_PT_BYE, RTCP_PT_FUR, RTCP_PT_PSFB, RTCP_PT_RR, RTCP_PT_SDES, RTCP_PT_SR, RTCP_REPORT_COUNT_MASK, RTCP_REPORT_COUNT_SHIFT, RTCP_RR_BLOCK_WORD_LENGTH, RTCP_SR_BLOCK_WORD_LENGTH, RTCP_VALID_MASK, RTCP_VALID_VALUE, RTCP_VERSION_MASK_SHIFTED, RTCP_VERSION_SHIFTED, rtp_find_instance_by_media_source_ssrc(), rtp_find_instance_by_packet_source_ssrc(), ast_rtcp::rtt, ast_rtcp::rxlsr, ast_frame::samples, ast_rtp::send_buffer, ast_rtcp::soc, ast_rtp_rtcp_report_block::source_ssrc, ast_rtcp::spc, ast_frame::src, srtp_replay_protection, ast_frame::stream_num, ast_rtp::stream_num, STRICT_RTP_OPEN, ast_rtp::strict_rtp_state, ast_frame::subclass, ast_rtcp::them, ast_rtcp::themrxlsr, ast_rtp::themssrc, ast_rtp::themssrc_valid, timeval2ntp(), ast_srtp_res::unprotect, update_jitter_stats(), update_lost_stats(), update_reported_mes_stats(), and update_rtt_stats().

Referenced by ast_rtcp_read(), and ast_rtp_read().

ast_rtcp_read()

static struct ast_frame * ast_rtcp_read ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 7371 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance, 1);
    struct ast_sockaddr addr;
    unsigned char rtcpdata[8192 + AST_FRIENDLY_OFFSET];
    unsigned char *read_area = rtcpdata + AST_FRIENDLY_OFFSET;
    size_t read_area_size = sizeof(rtcpdata) - AST_FRIENDLY_OFFSET;
    int res;
 
    /* Read in RTCP data from the socket */
    if ((res = rtcp_recvfrom(instance, read_area, read_area_size,
                0, &addr)) < 0) {
        if (res == RTP_DTLS_ESTABLISHED) {
            rtp->f.frametype = AST_FRAME_CONTROL;
            rtp->f.subclass.integer = AST_CONTROL_SRCCHANGE;
            return &rtp->f;
        }
 
        ast_assert(errno != EBADF);
        if (errno != EAGAIN) {
            ast_log(LOG_WARNING, "RTCP Read error: %s.  Hanging up.\n",
                (errno) ? strerror(errno) : "Unspecified");
            return NULL;
        }
        return &ast_null_frame;
    }
 
    /* If this was handled by the ICE session don't do anything further */
    if (!res) {
        return &ast_null_frame;
    }
 
    if (!*read_area) {
        struct sockaddr_in addr_tmp;
        struct ast_sockaddr addr_v4;
 
        if (ast_sockaddr_is_ipv4(&addr)) {
            ast_sockaddr_to_sin(&addr, &addr_tmp);
        } else if (ast_sockaddr_ipv4_mapped(&addr, &addr_v4)) {
            ast_debug_stun(2, "(%p) STUN using IPv6 mapped address %s\n",
                instance, ast_sockaddr_stringify(&addr));
            ast_sockaddr_to_sin(&addr_v4, &addr_tmp);
        } else {
            ast_debug_stun(2, "(%p) STUN cannot do for non IPv4 address %s\n",
                instance, ast_sockaddr_stringify(&addr));
            return &ast_null_frame;
        }
        if ((ast_stun_handle_packet(rtp->rtcp->s, &addr_tmp, read_area, res, NULL, NULL) == AST_STUN_ACCEPT)) {
            ast_sockaddr_from_sin(&addr, &addr_tmp);
            ast_sockaddr_copy(&rtp->rtcp->them, &addr);
        }
        return &ast_null_frame;
    }
 
    return ast_rtcp_interpret(instance, srtp, read_area, res, &addr);
}

References ast_assert, AST_CONTROL_SRCCHANGE, ast_debug_stun, AST_FRAME_CONTROL, AST_FRIENDLY_OFFSET, ast_log, ast_null_frame, ast_rtcp_interpret(), ast_rtp_instance_get_data(), ast_rtp_instance_get_srtp(), ast_sockaddr_copy(), ast_sockaddr_from_sin, ast_sockaddr_ipv4_mapped(), ast_sockaddr_is_ipv4(), ast_sockaddr_stringify(), ast_sockaddr_to_sin, AST_STUN_ACCEPT, ast_stun_handle_packet(), errno, ast_rtp::f, ast_frame::frametype, ast_frame_subclass::integer, LOG_WARNING, NULL, ast_rtp::rtcp, rtcp_recvfrom(), RTP_DTLS_ESTABLISHED, ast_rtcp::s, ast_frame::subclass, and ast_rtcp::them.

Referenced by ast_rtp_read().

ast_rtcp_write()

static int ast_rtcp_write ( const void *  data )

static

Write a RTCP packet to the far end.

Note

Decide if we are going to send an SR (with Reception Block) or RR RR is sent if we have not sent any rtp packets in the previous interval

Scheduler callback

Definition at line 5208 of file res_rtp_asterisk.c.

{
    struct ast_rtp_instance *instance = (struct ast_rtp_instance *) data;
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    int res;
    int sr = 0;
    int packet_len = 0;
    int ice;
    struct ast_sockaddr remote_address = { { 0, } };
    unsigned char *rtcpheader;
    unsigned char bdata[AST_UUID_STR_LEN + 128] = ""; /* More than enough */
    RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report, NULL, ao2_cleanup);
 
    if (!rtp || !rtp->rtcp || rtp->rtcp->schedid == -1) {
        ao2_ref(instance, -1);
        return 0;
    }
 
    ao2_lock(instance);
    rtcpheader = bdata;
    rtcp_report = ast_rtp_rtcp_report_alloc(rtp->themssrc_valid ? 1 : 0);
    res = ast_rtcp_generate_compound_prefix(instance, rtcpheader, rtcp_report, &sr);
 
    if (res == 0 || res == 1) {
        goto cleanup;
    }
 
    packet_len += res;
 
    if (rtp->bundled) {
        ast_rtp_instance_get_remote_address(instance, &remote_address);
    } else {
        ast_sockaddr_copy(&remote_address, &rtp->rtcp->them);
    }
 
    res = rtcp_sendto(instance, (unsigned int *)rtcpheader, packet_len, 0, &remote_address, &ice);
    if (res < 0) {
        ast_log(LOG_ERROR, "RTCP %s transmission error to %s, rtcp halted %s\n",
                sr ? "SR" : "RR",
                ast_sockaddr_stringify(&rtp->rtcp->them),
                strerror(errno));
        res = 0;
    } else {
        ast_rtcp_calculate_sr_rr_statistics(instance, rtcp_report, remote_address, ice, sr);
    }
 
cleanup:
    ao2_unlock(instance);
 
    if (!res) {
        /*
         * Not being rescheduled.
         */
        rtp->rtcp->schedid = -1;
        ao2_ref(instance, -1);
    }
 
    return res;
}

References ao2_cleanup, ao2_lock, ao2_ref, ao2_unlock, ast_log, ast_rtcp_calculate_sr_rr_statistics(), ast_rtcp_generate_compound_prefix(), ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_rtp_rtcp_report_alloc(), ast_sockaddr_copy(), ast_sockaddr_stringify(), AST_UUID_STR_LEN, ast_rtp::bundled, cleanup(), ast_rtp_instance::data, errno, LOG_ERROR, NULL, RAII_VAR, ast_rtp::rtcp, rtcp_sendto(), ast_rtcp::schedid, ast_rtcp::them, and ast_rtp::themssrc_valid.

Referenced by ast_rtp_interpret(), and rtp_raw_write().

ast_rtp_bundle()

static int ast_rtp_bundle ( struct ast_rtp_instance *  child, struct ast_rtp_instance *  parent  )

static

Precondition

child is locked

Definition at line 9767 of file res_rtp_asterisk.c.

{
    struct ast_rtp *child_rtp = ast_rtp_instance_get_data(child);
    struct ast_rtp *parent_rtp;
    struct rtp_ssrc_mapping mapping;
    struct ast_sockaddr them = { { 0, } };
 
    if (child_rtp->bundled == parent) {
        return 0;
    }
 
    /* If this instance was already bundled then remove the SSRC mapping */
    if (child_rtp->bundled) {
        struct ast_rtp *bundled_rtp;
 
        ao2_unlock(child);
 
        /* The child lock can't be held while accessing the parent */
        ao2_lock(child_rtp->bundled);
        bundled_rtp = ast_rtp_instance_get_data(child_rtp->bundled);
        AST_VECTOR_REMOVE_CMP_UNORDERED(&bundled_rtp->ssrc_mapping, child, SSRC_MAPPING_ELEM_CMP, AST_VECTOR_ELEM_CLEANUP_NOOP);
        ao2_unlock(child_rtp->bundled);
 
        ao2_lock(child);
        ao2_ref(child_rtp->bundled, -1);
        child_rtp->bundled = NULL;
    }
 
    if (!parent) {
        /* We transitioned away from bundle so we need our own transport resources once again */
        rtp_allocate_transport(child, child_rtp);
        return 0;
    }
 
    parent_rtp = ast_rtp_instance_get_data(parent);
 
    /* We no longer need any transport related resources as we will use our parent RTP instance instead */
    rtp_deallocate_transport(child, child_rtp);
 
    /* Children maintain a reference to the parent to guarantee that the transport doesn't go away on them */
    child_rtp->bundled = ao2_bump(parent);
 
    mapping.ssrc = child_rtp->themssrc;
    mapping.ssrc_valid = child_rtp->themssrc_valid;
    mapping.instance = child;
 
    ao2_unlock(child);
 
    ao2_lock(parent);
 
    AST_VECTOR_APPEND(&parent_rtp->ssrc_mapping, mapping);
 
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
    /* If DTLS-SRTP is already in use then add the local SSRC to it, otherwise it will get added once DTLS
     * negotiation has been completed.
     */
    if (parent_rtp->dtls.connection == AST_RTP_DTLS_CONNECTION_EXISTING) {
        dtls_srtp_add_local_ssrc(parent_rtp, parent, 0, child_rtp->ssrc, 0);
    }
#endif
 
    /* Bundle requires that RTCP-MUX be in use so only the main remote address needs to match */
    ast_rtp_instance_get_remote_address(parent, &them);
 
    ao2_unlock(parent);
 
    ao2_lock(child);
 
    ast_rtp_instance_set_remote_address(child, &them);
 
    return 0;
}

References ao2_bump, ao2_lock, ao2_ref, ao2_unlock, AST_RTP_DTLS_CONNECTION_EXISTING, ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_rtp_instance_set_remote_address, AST_VECTOR_APPEND, AST_VECTOR_ELEM_CLEANUP_NOOP, AST_VECTOR_REMOVE_CMP_UNORDERED, ast_rtp::bundled, rtp_ssrc_mapping::instance, NULL, rtp_allocate_transport(), rtp_deallocate_transport(), rtp_ssrc_mapping::ssrc, ast_rtp::ssrc, ast_rtp::ssrc_mapping, SSRC_MAPPING_ELEM_CMP, rtp_ssrc_mapping::ssrc_valid, ast_rtp::themssrc, and ast_rtp::themssrc_valid.

ast_rtp_change_source()

static void ast_rtp_change_source ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 4745 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance, 0);
    struct ast_srtp *rtcp_srtp = ast_rtp_instance_get_srtp(instance, 1);
    unsigned int ssrc = ast_random();
 
    if (rtp->lastts) {
        /* We simply set this bit so that the next packet sent will have the marker bit turned on */
        ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
    }
 
    ast_debug_rtp(3, "(%p) RTP changing ssrc from %u to %u due to a source change\n",
        instance, rtp->ssrc, ssrc);
 
    if (srtp) {
        ast_debug_rtp(3, "(%p) RTP changing ssrc for SRTP from %u to %u\n",
            instance, rtp->ssrc, ssrc);
        res_srtp->change_source(srtp, rtp->ssrc, ssrc);
        if (rtcp_srtp != srtp) {
            res_srtp->change_source(rtcp_srtp, rtp->ssrc, ssrc);
        }
    }
 
    rtp->ssrc = ssrc;
 
    /* Since the source is changing, we don't know what sequence number to expect next */
    rtp->expectedrxseqno = -1;
 
    return;
}

References ast_debug_rtp, ast_random(), ast_rtp_instance_get_data(), ast_rtp_instance_get_srtp(), ast_set_flag, ast_srtp_res::change_source, FLAG_NEED_MARKER_BIT, res_srtp, and ast_srtp::rtp.

ast_rtp_destroy()

static int ast_rtp_destroy ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 4388 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    if (rtp->bundled) {
        struct ast_rtp *bundled_rtp;
 
        /* We can't hold our instance lock while removing ourselves from the parent */
        ao2_unlock(instance);
 
        ao2_lock(rtp->bundled);
        bundled_rtp = ast_rtp_instance_get_data(rtp->bundled);
        AST_VECTOR_REMOVE_CMP_UNORDERED(&bundled_rtp->ssrc_mapping, instance, SSRC_MAPPING_ELEM_CMP, AST_VECTOR_ELEM_CLEANUP_NOOP);
        ao2_unlock(rtp->bundled);
 
        ao2_lock(instance);
        ao2_ref(rtp->bundled, -1);
    }
 
    rtp_deallocate_transport(instance, rtp);
 
    /* Destroy the smoother that was smoothing out audio if present */
    if (rtp->smoother) {
        ast_smoother_free(rtp->smoother);
    }
 
    /* Destroy RTCP if it was being used */
    if (rtp->rtcp) {
        /*
         * It is not possible for there to be an active RTCP scheduler
         * entry at this point since it holds a reference to the
         * RTP instance while it's active.
         */
        ast_free(rtp->rtcp->local_addr_str);
        ast_free(rtp->rtcp);
    }
 
    /* Destroy RED if it was being used */
    if (rtp->red) {
        ao2_unlock(instance);
        AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
        ao2_lock(instance);
        ast_free(rtp->red);
        rtp->red = NULL;
    }
 
    /* Destroy the send buffer if it was being used */
    if (rtp->send_buffer) {
        ast_data_buffer_free(rtp->send_buffer);
    }
 
    /* Destroy the recv buffer if it was being used */
    if (rtp->recv_buffer) {
        ast_data_buffer_free(rtp->recv_buffer);
    }
 
    AST_VECTOR_FREE(&rtp->transport_wide_cc.packet_statistics);
 
    ao2_cleanup(rtp->lasttxformat);
    ao2_cleanup(rtp->lastrxformat);
    ao2_cleanup(rtp->f.subclass.format);
    AST_VECTOR_FREE(&rtp->ssrc_mapping);
    AST_VECTOR_FREE(&rtp->missing_seqno);
 
    /* Finally destroy ourselves */
    rtp->owner = NULL;
    ast_free(rtp);
 
    return 0;
}

References ao2_cleanup, ao2_lock, ao2_ref, ao2_unlock, ast_data_buffer_free(), ast_free, ast_rtp_instance_get_data(), AST_SCHED_DEL, ast_smoother_free(), AST_VECTOR_ELEM_CLEANUP_NOOP, AST_VECTOR_FREE, AST_VECTOR_REMOVE_CMP_UNORDERED, ast_rtp::bundled, ast_rtp::f, ast_frame_subclass::format, ast_rtp::lastrxformat, ast_rtp::lasttxformat, ast_rtcp::local_addr_str, ast_rtp::missing_seqno, NULL, ast_rtp::owner, rtp_transport_wide_cc_statistics::packet_statistics, ast_rtp::recv_buffer, ast_rtp::red, ast_rtp::rtcp, rtp_deallocate_transport(), ast_rtp::sched, rtp_red::schedid, ast_rtp::send_buffer, ast_rtp::smoother, ast_rtp::ssrc_mapping, SSRC_MAPPING_ELEM_CMP, ast_frame::subclass, and ast_rtp::transport_wide_cc.

ast_rtp_dtmf_begin()

static int ast_rtp_dtmf_begin ( struct ast_rtp_instance *  instance, char  digit  )

static

Precondition

instance is locked

Definition at line 4475 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_sockaddr remote_address = { {0,} };
    int hdrlen = 12, res = 0, i = 0, payload = -1, sample_rate = -1;
    char data[256];
    unsigned int *rtpheader = (unsigned int*)data;
    RAII_VAR(struct ast_format *, payload_format, NULL, ao2_cleanup);
 
    ast_rtp_instance_get_remote_address(instance, &remote_address);
 
    /* If we have no remote address information bail out now */
    if (ast_sockaddr_isnull(&remote_address)) {
        return -1;
    }
 
    /* Convert given digit into what we want to transmit */
    if ((digit <= '9') && (digit >= '0')) {
        digit -= '0';
    } else if (digit == '*') {
        digit = 10;
    } else if (digit == '#') {
        digit = 11;
    } else if ((digit >= 'A') && (digit <= 'D')) {
        digit = digit - 'A' + 12;
    } else if ((digit >= 'a') && (digit <= 'd')) {
        digit = digit - 'a' + 12;
    } else {
        ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
        return -1;
    }
 
 
    /* g722 is a 16K codec that masquerades as an 8K codec within RTP. ast_rtp_get_rate was written specifically to
       handle this. If we use the actual sample rate of g722 in this scenario and there is a 16K telephone-event on
       offer, we will end up using that instead of the 8K rate telephone-event that is expected with g722. */
    if (rtp->lasttxformat == ast_format_none) {
        /* No audio frames have been written yet so we have to lookup both the preferred payload type and bitrate. */
        payload_format = ast_rtp_codecs_get_preferred_format(ast_rtp_instance_get_codecs(instance));
        if (payload_format) {
            /* If we have a preferred type, use that. Otherwise default to 8K. */
            sample_rate = ast_rtp_get_rate(payload_format);
        }
    } else {
        sample_rate = ast_rtp_get_rate(rtp->lasttxformat);
    }
 
    if (sample_rate != -1) {
        payload = ast_rtp_codecs_payload_code_tx_sample_rate(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_DTMF, sample_rate);
    }
 
    if (payload == -1 ||
        !ast_rtp_payload_mapping_tx_is_present(
            ast_rtp_instance_get_codecs(instance), ast_rtp_codecs_get_payload(ast_rtp_instance_get_codecs(instance), payload))) {
        /* Fall back to the preferred DTMF payload type and sample rate as either we couldn't find an audio codec to try and match
           sample rates with or we could, but a telephone-event matching that audio codec's sample rate was not included in the
           sdp negotiated by the far end. */
        payload = ast_rtp_codecs_get_preferred_dtmf_format_pt(ast_rtp_instance_get_codecs(instance));
        sample_rate = ast_rtp_codecs_get_preferred_dtmf_format_rate(ast_rtp_instance_get_codecs(instance));
    }
 
    /* The sdp negotiation has not yeilded a usable RFC 2833/4733 format. Try a default-rate one as a last resort. */
    if (payload == -1 || sample_rate == -1) {
        sample_rate = DEFAULT_DTMF_SAMPLE_RATE_MS;
        payload = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_DTMF);
    }
    /* Even trying a default payload has failed. We are trying to send a digit outside of what was negotiated for. */
    if (payload == -1) {
        return -1;
    }
 
    ast_test_suite_event_notify("DTMF_BEGIN", "Digit: %d\r\nPayload: %d\r\nRate: %d\r\n", digit, payload, sample_rate);
    ast_debug(1, "Sending digit '%d' at rate %d with payload %d\n", digit, sample_rate, payload);
 
    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
    rtp->send_duration = 160;
    rtp->dtmf_samplerate_ms = (sample_rate / 1000);
    rtp->lastts += calc_txstamp(rtp, NULL) * rtp->dtmf_samplerate_ms;
    rtp->lastdigitts = rtp->lastts + rtp->send_duration;
 
    /* Create the actual packet that we will be sending */
    rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
    rtpheader[1] = htonl(rtp->lastdigitts);
    rtpheader[2] = htonl(rtp->ssrc);
 
    /* Actually send the packet */
    for (i = 0; i < 2; i++) {
        int ice;
 
        rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
        res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 4, 0, &remote_address, &ice);
        if (res < 0) {
            ast_log(LOG_ERROR, "RTP Transmission error to %s: %s\n",
                ast_sockaddr_stringify(&remote_address),
                strerror(errno));
        }
        if (rtp_debug_test_addr(&remote_address)) {
            ast_verbose("Sent RTP DTMF packet to %s%s (type %-2.2d, seq %-6.6d, ts %-6.6u, len %-6.6d)\n",
                    ast_sockaddr_stringify(&remote_address),
                    ice ? " (via ICE)" : "",
                    payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
        }
        rtp->seqno++;
        rtp->send_duration += 160;
        rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
    }
 
    /* Record that we are in the process of sending a digit and information needed to continue doing so */
    rtp->sending_digit = 1;
    rtp->send_digit = digit;
    rtp->send_payload = payload;
 
    return 0;
}

References ao2_cleanup, ast_debug, ast_format_none, ast_log, ast_rtp_codecs_get_payload(), ast_rtp_codecs_get_preferred_dtmf_format_pt(), ast_rtp_codecs_get_preferred_dtmf_format_rate(), ast_rtp_codecs_get_preferred_format(), ast_rtp_codecs_payload_code_tx(), ast_rtp_codecs_payload_code_tx_sample_rate(), AST_RTP_DTMF, ast_rtp_get_rate(), ast_rtp_instance_get_codecs(), ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_rtp_payload_mapping_tx_is_present(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_test_suite_event_notify, ast_tv(), ast_tvadd(), ast_tvnow(), ast_verbose, calc_txstamp(), DEFAULT_DTMF_SAMPLE_RATE_MS, digit, ast_rtp::dtmf_samplerate_ms, ast_rtp::dtmfmute, errno, ast_rtp::lastdigitts, ast_rtp::lastts, ast_rtp::lasttxformat, LOG_ERROR, LOG_WARNING, NULL, RAII_VAR, rtp_debug_test_addr(), rtp_sendto(), ast_rtp::send_digit, ast_rtp::send_duration, ast_rtp::send_payload, ast_rtp::sending_digit, ast_rtp::seqno, and ast_rtp::ssrc.

ast_rtp_dtmf_compatible()

static int ast_rtp_dtmf_compatible ( struct ast_channel *  chan0, struct ast_rtp_instance *  instance0, struct ast_channel *  chan1, struct ast_rtp_instance *  instance1  )

static

Precondition

Neither instance0 nor instance1 are locked

Definition at line 9550 of file res_rtp_asterisk.c.

{
    /* If both sides are not using the same method of DTMF transmission
     * (ie: one is RFC2833, other is INFO... then we can not do direct media.
     * --------------------------------------------------
     * | DTMF Mode |  HAS_DTMF  |  Accepts Begin Frames |
     * |-----------|------------|-----------------------|
     * | Inband    | False      | True                  |
     * | RFC2833   | True       | True                  |
     * | SIP INFO  | False      | False                 |
     * --------------------------------------------------
     */
    return (((ast_rtp_instance_get_prop(instance0, AST_RTP_PROPERTY_DTMF) != ast_rtp_instance_get_prop(instance1, AST_RTP_PROPERTY_DTMF)) ||
         (!ast_channel_tech(chan0)->send_digit_begin != !ast_channel_tech(chan1)->send_digit_begin)) ? 0 : 1);
}

References ast_rtp_instance_get_prop(), and AST_RTP_PROPERTY_DTMF.

ast_rtp_dtmf_continuation()

static int ast_rtp_dtmf_continuation ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 4591 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_sockaddr remote_address = { {0,} };
    int hdrlen = 12, res = 0;
    char data[256];
    unsigned int *rtpheader = (unsigned int*)data;
    int ice;
 
    ast_rtp_instance_get_remote_address(instance, &remote_address);
 
    /* Make sure we know where the other side is so we can send them the packet */
    if (ast_sockaddr_isnull(&remote_address)) {
        return -1;
    }
 
    /* Actually create the packet we will be sending */
    rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
    rtpheader[1] = htonl(rtp->lastdigitts);
    rtpheader[2] = htonl(rtp->ssrc);
    rtpheader[3] = htonl((rtp->send_digit << 24) | (0xa << 16) | (rtp->send_duration));
 
    /* Boom, send it on out */
    res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 4, 0, &remote_address, &ice);
    if (res < 0) {
        ast_log(LOG_ERROR, "RTP Transmission error to %s: %s\n",
            ast_sockaddr_stringify(&remote_address),
            strerror(errno));
    }
 
    if (rtp_debug_test_addr(&remote_address)) {
        ast_verbose("Sent RTP DTMF packet to %s%s (type %-2.2d, seq %-6.6d, ts %-6.6u, len %-6.6d)\n",
                ast_sockaddr_stringify(&remote_address),
                ice ? " (via ICE)" : "",
                rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
    }
 
    /* And now we increment some values for the next time we swing by */
    rtp->seqno++;
    rtp->send_duration += 160;
    rtp->lastts += calc_txstamp(rtp, NULL) * rtp->dtmf_samplerate_ms;
 
    return 0;
}

References ast_log, ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_verbose, calc_txstamp(), ast_rtp::dtmf_samplerate_ms, errno, ast_rtp::lastdigitts, ast_rtp::lastts, LOG_ERROR, NULL, rtp_debug_test_addr(), rtp_sendto(), ast_rtp::send_digit, ast_rtp::send_duration, ast_rtp::send_payload, ast_rtp::seqno, and ast_rtp::ssrc.

Referenced by ast_rtp_interpret().

ast_rtp_dtmf_end()

static int ast_rtp_dtmf_end ( struct ast_rtp_instance *  instance, char  digit  )

static

Precondition

instance is locked

Definition at line 4727 of file res_rtp_asterisk.c.

{
    return ast_rtp_dtmf_end_with_duration(instance, digit, 0);
}

References ast_rtp_dtmf_end_with_duration(), and digit.

ast_rtp_dtmf_end_with_duration()

static int ast_rtp_dtmf_end_with_duration ( struct ast_rtp_instance *  instance, char  digit, unsigned int  duration  )

static

Precondition

instance is locked

Definition at line 4637 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_sockaddr remote_address = { {0,} };
    int hdrlen = 12, res = -1, i = 0;
    char data[256];
    unsigned int *rtpheader = (unsigned int*)data;
    unsigned int measured_samples;
 
    ast_rtp_instance_get_remote_address(instance, &remote_address);
 
    /* Make sure we know where the remote side is so we can send them the packet we construct */
    if (ast_sockaddr_isnull(&remote_address)) {
        goto cleanup;
    }
 
    /* Convert the given digit to the one we are going to send */
    if ((digit <= '9') && (digit >= '0')) {
        digit -= '0';
    } else if (digit == '*') {
        digit = 10;
    } else if (digit == '#') {
        digit = 11;
    } else if ((digit >= 'A') && (digit <= 'D')) {
        digit = digit - 'A' + 12;
    } else if ((digit >= 'a') && (digit <= 'd')) {
        digit = digit - 'a' + 12;
    } else {
        ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
        goto cleanup;
    }
 
    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
 
    if (duration > 0 && (measured_samples = duration * ast_rtp_get_rate(rtp->f.subclass.format) / 1000) > rtp->send_duration) {
        ast_debug_rtp(2, "(%p) RTP adjusting final end duration from %d to %u\n",
            instance, rtp->send_duration, measured_samples);
        rtp->send_duration = measured_samples;
    }
 
    /* Construct the packet we are going to send */
    rtpheader[1] = htonl(rtp->lastdigitts);
    rtpheader[2] = htonl(rtp->ssrc);
    rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
    rtpheader[3] |= htonl((1 << 23));
 
    /* Send it 3 times, that's the magical number */
    for (i = 0; i < 3; i++) {
        int ice;
 
        rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
 
        res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 4, 0, &remote_address, &ice);
 
        if (res < 0) {
            ast_log(LOG_ERROR, "RTP Transmission error to %s: %s\n",
                ast_sockaddr_stringify(&remote_address),
                strerror(errno));
        }
 
        if (rtp_debug_test_addr(&remote_address)) {
            ast_verbose("Sent RTP DTMF packet to %s%s (type %-2.2d, seq %-6.6d, ts %-6.6u, len %-6.6d)\n",
                    ast_sockaddr_stringify(&remote_address),
                    ice ? " (via ICE)" : "",
                    rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
        }
 
        rtp->seqno++;
    }
    res = 0;
 
    /* Oh and we can't forget to turn off the stuff that says we are sending DTMF */
    rtp->lastts += calc_txstamp(rtp, NULL) * rtp->dtmf_samplerate_ms;
 
    /* Reset the smoother as the delivery time stored in it is now out of date */
    if (rtp->smoother) {
        ast_smoother_free(rtp->smoother);
        rtp->smoother = NULL;
    }
cleanup:
    rtp->sending_digit = 0;
    rtp->send_digit = 0;
 
    /* Re-Learn expected seqno */
    rtp->expectedseqno = -1;
 
    return res;
}

References ast_debug_rtp, ast_log, ast_rtp_get_rate(), ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_smoother_free(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_tv(), ast_tvadd(), ast_tvnow(), ast_verbose, calc_txstamp(), cleanup(), digit, ast_rtp::dtmf_samplerate_ms, ast_rtp::dtmfmute, errno, ast_rtp::expectedseqno, ast_rtp::f, ast_frame_subclass::format, ast_rtp::lastdigitts, ast_rtp::lastts, LOG_ERROR, LOG_WARNING, NULL, rtp_debug_test_addr(), rtp_sendto(), ast_rtp::send_digit, ast_rtp::send_duration, ast_rtp::send_payload, ast_rtp::sending_digit, ast_rtp::seqno, ast_rtp::smoother, ast_rtp::ssrc, and ast_frame::subclass.

Referenced by ast_rtp_dtmf_end().

ast_rtp_dtmf_mode_get()

static enum ast_rtp_dtmf_mode ast_rtp_dtmf_mode_get ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 4468 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    return rtp->dtmfmode;
}

References ast_rtp_instance_get_data(), and ast_rtp::dtmfmode.

ast_rtp_dtmf_mode_set()

static int ast_rtp_dtmf_mode_set ( struct ast_rtp_instance *  instance, enum ast_rtp_dtmf_mode  dtmf_mode  )

static

Precondition

instance is locked

Definition at line 4460 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    rtp->dtmfmode = dtmf_mode;
    return 0;
}

References ast_rtp_instance_get_data(), and ast_rtp::dtmfmode.

ast_rtp_extension_enable()

static int ast_rtp_extension_enable ( struct ast_rtp_instance *  instance, enum ast_rtp_extension  extension  )

static

Definition at line 9755 of file res_rtp_asterisk.c.

{
    switch (extension) {
    case AST_RTP_EXTENSION_ABS_SEND_TIME:
    case AST_RTP_EXTENSION_TRANSPORT_WIDE_CC:
        return 1;
    default:
        return 0;
    }
}

References AST_RTP_EXTENSION_ABS_SEND_TIME, and AST_RTP_EXTENSION_TRANSPORT_WIDE_CC.

ast_rtp_fd()

static int ast_rtp_fd ( struct ast_rtp_instance *  instance, int  rtcp  )

static

Precondition

instance is locked

Definition at line 9271 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    return rtcp ? (rtp->rtcp ? rtp->rtcp->s : -1) : rtp->s;
}

References ast_rtp_instance_get_data(), ast_rtp::rtcp, ast_rtp::s, and ast_rtcp::s.

ast_rtp_get_cname()

static const char * ast_rtp_get_cname ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 9702 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    return rtp->cname;
}

References ast_rtp_instance_get_data(), and ast_rtp::cname.

ast_rtp_get_ssrc()

static unsigned int ast_rtp_get_ssrc ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 9694 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    return rtp->ssrc;
}

References ast_rtp_instance_get_data(), and ast_rtp::ssrc.

Referenced by __rtp_find_instance_by_ssrc().

ast_rtp_get_stat()

static int ast_rtp_get_stat ( struct ast_rtp_instance *  instance, struct ast_rtp_instance_stats *  stats, enum ast_rtp_instance_stat  stat  )

static

Precondition

instance is locked

Definition at line 9485 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    if (!rtp->rtcp) {
        return -1;
    }
 
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_TXCOUNT, -1, stats->txcount, rtp->txcount);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXCOUNT, -1, stats->rxcount, rtp->rxcount);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_TXOCTETCOUNT, -1, stats->txoctetcount, rtp->txoctetcount);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXOCTETCOUNT, -1, stats->rxoctetcount, rtp->rxoctetcount);
 
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_TXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->txploss, rtp->rtcp->reported_lost);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->rxploss, rtp->rtcp->expected_prior - rtp->rtcp->received_prior);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_maxrxploss, rtp->rtcp->reported_maxlost);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_minrxploss, rtp->rtcp->reported_minlost);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_normdevrxploss, rtp->rtcp->reported_normdev_lost);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_stdevrxploss, rtp->rtcp->reported_stdev_lost);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_maxrxploss, rtp->rtcp->maxrxlost);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_minrxploss, rtp->rtcp->minrxlost);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_normdevrxploss, rtp->rtcp->normdev_rxlost);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_stdevrxploss, rtp->rtcp->stdev_rxlost);
    AST_RTP_STAT_TERMINATOR(AST_RTP_INSTANCE_STAT_COMBINED_LOSS);
 
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_TXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->txjitter, rtp->rxjitter);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->rxjitter, rtp->rtcp->reported_jitter);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_maxjitter, rtp->rtcp->reported_maxjitter);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_minjitter, rtp->rtcp->reported_minjitter);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_normdevjitter, rtp->rtcp->reported_normdev_jitter);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_stdevjitter, rtp->rtcp->reported_stdev_jitter);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_maxjitter, rtp->rtcp->maxrxjitter);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_minjitter, rtp->rtcp->minrxjitter);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_normdevjitter, rtp->rtcp->normdev_rxjitter);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_stdevjitter, rtp->rtcp->stdev_rxjitter);
    AST_RTP_STAT_TERMINATOR(AST_RTP_INSTANCE_STAT_COMBINED_JITTER);
 
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->rtt, rtp->rtcp->rtt);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_MAX_RTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->maxrtt, rtp->rtcp->maxrtt);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_MIN_RTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->minrtt, rtp->rtcp->minrtt);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_NORMDEVRTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->normdevrtt, rtp->rtcp->normdevrtt);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_STDEVRTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->stdevrtt, rtp->rtcp->stdevrtt);
    AST_RTP_STAT_TERMINATOR(AST_RTP_INSTANCE_STAT_COMBINED_RTT);
 
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_TXMES, AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->txmes, rtp->rxmes);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXMES, AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->rxmes, rtp->rtcp->reported_mes);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MAXMES, AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->remote_maxmes, rtp->rtcp->reported_maxmes);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MINMES, AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->remote_minmes, rtp->rtcp->reported_minmes);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVMES, AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->remote_normdevmes, rtp->rtcp->reported_normdev_mes);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_STDEVMES, AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->remote_stdevmes, rtp->rtcp->reported_stdev_mes);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MAXMES, AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_maxmes, rtp->rtcp->maxrxmes);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MINMES, AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_minmes, rtp->rtcp->minrxmes);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVMES, AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_normdevmes, rtp->rtcp->normdev_rxmes);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVMES, AST_RTP_INSTANCE_STAT_COMBINED_MES, stats->local_stdevmes, rtp->rtcp->stdev_rxmes);
    AST_RTP_STAT_TERMINATOR(AST_RTP_INSTANCE_STAT_COMBINED_MES);
 
 
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_SSRC, -1, stats->local_ssrc, rtp->ssrc);
    AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_SSRC, -1, stats->remote_ssrc, rtp->themssrc);
    AST_RTP_STAT_STRCPY(AST_RTP_INSTANCE_STAT_CHANNEL_UNIQUEID, -1, stats->channel_uniqueid, ast_rtp_instance_get_channel_id(instance));
 
    return 0;
}

References ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_data(), AST_RTP_INSTANCE_STAT_CHANNEL_UNIQUEID, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, AST_RTP_INSTANCE_STAT_COMBINED_MES, AST_RTP_INSTANCE_STAT_COMBINED_RTT, AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER, AST_RTP_INSTANCE_STAT_LOCAL_MAXMES, AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER, AST_RTP_INSTANCE_STAT_LOCAL_MINMES, AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVMES, AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_SSRC, AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER, AST_RTP_INSTANCE_STAT_LOCAL_STDEVMES, AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_MAX_RTT, AST_RTP_INSTANCE_STAT_MIN_RTT, AST_RTP_INSTANCE_STAT_NORMDEVRTT, AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER, AST_RTP_INSTANCE_STAT_REMOTE_MAXMES, AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER, AST_RTP_INSTANCE_STAT_REMOTE_MINMES, AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVMES, AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_SSRC, AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER, AST_RTP_INSTANCE_STAT_REMOTE_STDEVMES, AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_RTT, AST_RTP_INSTANCE_STAT_RXCOUNT, AST_RTP_INSTANCE_STAT_RXJITTER, AST_RTP_INSTANCE_STAT_RXMES, AST_RTP_INSTANCE_STAT_RXOCTETCOUNT, AST_RTP_INSTANCE_STAT_RXPLOSS, AST_RTP_INSTANCE_STAT_STDEVRTT, AST_RTP_INSTANCE_STAT_TXCOUNT, AST_RTP_INSTANCE_STAT_TXJITTER, AST_RTP_INSTANCE_STAT_TXMES, AST_RTP_INSTANCE_STAT_TXOCTETCOUNT, AST_RTP_INSTANCE_STAT_TXPLOSS, AST_RTP_STAT_SET, AST_RTP_STAT_STRCPY, AST_RTP_STAT_TERMINATOR, ast_rtp_instance_stats::channel_uniqueid, ast_rtcp::expected_prior, ast_rtp_instance_stats::local_maxjitter, ast_rtp_instance_stats::local_maxmes, ast_rtp_instance_stats::local_maxrxploss, ast_rtp_instance_stats::local_minjitter, ast_rtp_instance_stats::local_minmes, ast_rtp_instance_stats::local_minrxploss, ast_rtp_instance_stats::local_normdevjitter, ast_rtp_instance_stats::local_normdevmes, ast_rtp_instance_stats::local_normdevrxploss, ast_rtp_instance_stats::local_ssrc, ast_rtp_instance_stats::local_stdevjitter, ast_rtp_instance_stats::local_stdevmes, ast_rtp_instance_stats::local_stdevrxploss, ast_rtp_instance_stats::maxrtt, ast_rtcp::maxrtt, ast_rtcp::maxrxjitter, ast_rtcp::maxrxlost, ast_rtcp::maxrxmes, ast_rtp_instance_stats::minrtt, ast_rtcp::minrtt, ast_rtcp::minrxjitter, ast_rtcp::minrxlost, ast_rtcp::minrxmes, ast_rtcp::normdev_rxjitter, ast_rtcp::normdev_rxlost, ast_rtcp::normdev_rxmes, ast_rtp_instance_stats::normdevrtt, ast_rtcp::normdevrtt, ast_rtcp::received_prior, ast_rtp_instance_stats::remote_maxjitter, ast_rtp_instance_stats::remote_maxmes, ast_rtp_instance_stats::remote_maxrxploss, ast_rtp_instance_stats::remote_minjitter, ast_rtp_instance_stats::remote_minmes, ast_rtp_instance_stats::remote_minrxploss, ast_rtp_instance_stats::remote_normdevjitter, ast_rtp_instance_stats::remote_normdevmes, ast_rtp_instance_stats::remote_normdevrxploss, ast_rtp_instance_stats::remote_ssrc, ast_rtp_instance_stats::remote_stdevjitter, ast_rtp_instance_stats::remote_stdevmes, ast_rtp_instance_stats::remote_stdevrxploss, ast_rtcp::reported_jitter, ast_rtcp::reported_lost, ast_rtcp::reported_maxjitter, ast_rtcp::reported_maxlost, ast_rtcp::reported_maxmes, ast_rtcp::reported_mes, ast_rtcp::reported_minjitter, ast_rtcp::reported_minlost, ast_rtcp::reported_minmes, ast_rtcp::reported_normdev_jitter, ast_rtcp::reported_normdev_lost, ast_rtcp::reported_normdev_mes, ast_rtcp::reported_stdev_jitter, ast_rtcp::reported_stdev_lost, ast_rtcp::reported_stdev_mes, ast_rtp::rtcp, ast_rtp_instance_stats::rtt, ast_rtcp::rtt, ast_rtp_instance_stats::rxcount, ast_rtp::rxcount, ast_rtp_instance_stats::rxjitter, ast_rtp::rxjitter, ast_rtp_instance_stats::rxmes, ast_rtp::rxmes, ast_rtp_instance_stats::rxoctetcount, ast_rtp::rxoctetcount, ast_rtp_instance_stats::rxploss, ast_rtp::ssrc, ast_rtcp::stdev_rxjitter, ast_rtcp::stdev_rxlost, ast_rtcp::stdev_rxmes, ast_rtp_instance_stats::stdevrtt, ast_rtcp::stdevrtt, ast_rtp::themssrc, ast_rtp_instance_stats::txcount, ast_rtp::txcount, ast_rtp_instance_stats::txjitter, ast_rtp_instance_stats::txmes, ast_rtp_instance_stats::txoctetcount, ast_rtp::txoctetcount, and ast_rtp_instance_stats::txploss.

ast_rtp_interpret()

static struct ast_frame * ast_rtp_interpret ( struct ast_rtp_instance *  instance, struct ast_srtp *  srtp, const struct ast_sockaddr *  remote_address, unsigned char *  read_area, int  length, int  prev_seqno, unsigned int  bundled  )

static

Definition at line 7981 of file res_rtp_asterisk.c.

{
    unsigned int *rtpheader = (unsigned int*)(read_area);
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_rtp_instance *instance1;
    int res = length, hdrlen = 12, ssrc, seqno, payloadtype, padding, mark, ext, cc;
    unsigned int timestamp;
    RAII_VAR(struct ast_rtp_payload_type *, payload, NULL, ao2_cleanup);
    struct frame_list frames;
 
    /* If this payload is encrypted then decrypt it using the given SRTP instance */
    if ((*read_area & 0xC0) && res_srtp && srtp && res_srtp->unprotect(
            srtp, read_area, &res, 0 | (srtp_replay_protection << 1)) < 0) {
        return &ast_null_frame;
    }
 
    /* If we are currently sending DTMF to the remote party send a continuation packet */
    if (rtp->sending_digit) {
        ast_rtp_dtmf_continuation(instance);
    }
 
    /* Pull out the various other fields we will need */
    ssrc = ntohl(rtpheader[2]);
    seqno = ntohl(rtpheader[0]);
    payloadtype = (seqno & 0x7f0000) >> 16;
    padding = seqno & (1 << 29);
    mark = seqno & (1 << 23);
    ext = seqno & (1 << 28);
    cc = (seqno & 0xF000000) >> 24;
    seqno &= 0xffff;
    timestamp = ntohl(rtpheader[1]);
 
    AST_LIST_HEAD_INIT_NOLOCK(&frames);
 
    /* Remove any padding bytes that may be present */
    if (padding) {
        res -= read_area[res - 1];
    }
 
    /* Skip over any CSRC fields */
    if (cc) {
        hdrlen += cc * 4;
    }
 
    /* Look for any RTP extensions, currently we do not support any */
    if (ext) {
        int extensions_size = (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
        unsigned int profile;
        profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16;
 
        if (profile == 0xbede) {
            /* We skip over the first 4 bytes as they are just for the one byte extension header */
            rtp_instance_parse_extmap_extensions(instance, rtp, read_area + hdrlen + 4, extensions_size);
        } else if (DEBUG_ATLEAST(1)) {
            if (profile == 0x505a) {
                ast_log(LOG_DEBUG, "Found Zfone extension in RTP stream - zrtp - not supported.\n");
            } else {
                /* SDP negotiated RTP extensions can not currently be output in logging */
                ast_log(LOG_DEBUG, "Found unknown RTP Extensions %x\n", profile);
            }
        }
 
        hdrlen += extensions_size;
        hdrlen += 4;
    }
 
    /* Make sure after we potentially mucked with the header length that it is once again valid */
    if (res < hdrlen) {
        ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d\n", res, hdrlen);
        return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
    }
 
    /* Only non-bundled instances can change/learn the remote's SSRC implicitly. */
    if (!bundled) {
        /* Force a marker bit and change SSRC if the SSRC changes */
        if (rtp->themssrc_valid && rtp->themssrc != ssrc) {
            struct ast_frame *f, srcupdate = {
                AST_FRAME_CONTROL,
                .subclass.integer = AST_CONTROL_SRCCHANGE,
            };
 
            if (!mark) {
                if (ast_debug_rtp_packet_is_allowed) {
                    ast_debug(0, "(%p) RTP forcing Marker bit, because SSRC has changed\n", instance);
                }
                mark = 1;
            }
 
            f = ast_frisolate(&srcupdate);
            AST_LIST_INSERT_TAIL(&frames, f, frame_list);
 
            rtp->seedrxseqno = 0;
            rtp->rxcount = 0;
            rtp->rxoctetcount = 0;
            rtp->cycles = 0;
            prev_seqno = 0;
            rtp->last_seqno = 0;
            rtp->last_end_timestamp.ts = 0;
            rtp->last_end_timestamp.is_set = 0;
            if (rtp->rtcp) {
                rtp->rtcp->expected_prior = 0;
                rtp->rtcp->received_prior = 0;
            }
        }
 
        rtp->themssrc = ssrc; /* Record their SSRC to put in future RR */
        rtp->themssrc_valid = 1;
    }
 
    rtp->rxcount++;
    rtp->rxoctetcount += (res - hdrlen);
    if (rtp->rxcount == 1) {
        rtp->seedrxseqno = seqno;
    }
 
    /* Do not schedule RR if RTCP isn't run */
    if (rtp->rtcp && !ast_sockaddr_isnull(&rtp->rtcp->them) && rtp->rtcp->schedid < 0) {
        /* Schedule transmission of Receiver Report */
        ao2_ref(instance, +1);
        rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, instance);
        if (rtp->rtcp->schedid < 0) {
            ao2_ref(instance, -1);
            ast_log(LOG_WARNING, "scheduling RTCP transmission failed.\n");
        }
    }
    if ((int)prev_seqno - (int)seqno  > 100) /* if so it would indicate that the sender cycled; allow for misordering */
        rtp->cycles += RTP_SEQ_MOD;
 
    /* If we are directly bridged to another instance send the audio directly out,
     * but only after updating core information about the received traffic so that
     * outgoing RTCP reflects it.
     */
    instance1 = ast_rtp_instance_get_bridged(instance);
    if (instance1
        && !bridge_p2p_rtp_write(instance, instance1, rtpheader, res, hdrlen)) {
        struct timeval rxtime;
        struct ast_frame *f;
 
        /* Update statistics for jitter so they are correct in RTCP */
        calc_rxstamp_and_jitter(&rxtime, rtp, timestamp, mark);
 
 
        /* When doing P2P we don't need to raise any frames about SSRC change to the core */
        while ((f = AST_LIST_REMOVE_HEAD(&frames, frame_list)) != NULL) {
            ast_frfree(f);
        }
 
        return &ast_null_frame;
    }
 
    payload = ast_rtp_codecs_get_payload(ast_rtp_instance_get_codecs(instance), payloadtype);
    if (!payload) {
        /* Unknown payload type. */
        return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
    }
 
    /* If the payload is not actually an Asterisk one but a special one pass it off to the respective handler */
    if (!payload->asterisk_format) {
        struct ast_frame *f = NULL;
        if (payload->rtp_code == AST_RTP_DTMF) {
            /* process_dtmf_rfc2833 may need to return multiple frames. We do this
             * by passing the pointer to the frame list to it so that the method
             * can append frames to the list as needed.
             */
            process_dtmf_rfc2833(instance, read_area + hdrlen, res - hdrlen, seqno, timestamp, payloadtype, mark, &frames);
        } else if (payload->rtp_code == AST_RTP_CISCO_DTMF) {
            f = process_dtmf_cisco(instance, read_area + hdrlen, res - hdrlen, seqno, timestamp, payloadtype, mark);
        } else if (payload->rtp_code == AST_RTP_CN) {
            f = process_cn_rfc3389(instance, read_area + hdrlen, res - hdrlen, seqno, timestamp, payloadtype, mark);
        } else {
            ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n",
                payloadtype,
                ast_sockaddr_stringify(remote_address));
        }
 
        if (f) {
            AST_LIST_INSERT_TAIL(&frames, f, frame_list);
        }
        /* Even if no frame was returned by one of the above methods,
         * we may have a frame to return in our frame list
         */
        return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
    }
 
    ao2_replace(rtp->lastrxformat, payload->format);
    ao2_replace(rtp->f.subclass.format, payload->format);
    switch (ast_format_get_type(rtp->f.subclass.format)) {
    case AST_MEDIA_TYPE_AUDIO:
        rtp->f.frametype = AST_FRAME_VOICE;
        break;
    case AST_MEDIA_TYPE_VIDEO:
        rtp->f.frametype = AST_FRAME_VIDEO;
        break;
    case AST_MEDIA_TYPE_TEXT:
        rtp->f.frametype = AST_FRAME_TEXT;
        break;
    case AST_MEDIA_TYPE_IMAGE:
        /* Fall through */
    default:
        ast_log(LOG_WARNING, "Unknown or unsupported media type: %s\n",
            ast_codec_media_type2str(ast_format_get_type(rtp->f.subclass.format)));
        return &ast_null_frame;
    }
 
    if (rtp->dtmf_timeout && rtp->dtmf_timeout < timestamp) {
        rtp->dtmf_timeout = 0;
 
        if (rtp->resp) {
            struct ast_frame *f;
            f = create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0);
            f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, ast_rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
            rtp->resp = 0;
            rtp->dtmf_timeout = rtp->dtmf_duration = 0;
            AST_LIST_INSERT_TAIL(&frames, f, frame_list);
            return AST_LIST_FIRST(&frames);
        }
    }
 
    rtp->f.src = "RTP";
    rtp->f.mallocd = 0;
    rtp->f.datalen = res - hdrlen;
    rtp->f.data.ptr = read_area + hdrlen;
    rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
    ast_set_flag(&rtp->f, AST_FRFLAG_HAS_SEQUENCE_NUMBER);
    rtp->f.seqno = seqno;
    rtp->f.stream_num = rtp->stream_num;
 
    if ((ast_format_cmp(rtp->f.subclass.format, ast_format_t140) == AST_FORMAT_CMP_EQUAL)
        && ((int)seqno - (prev_seqno + 1) > 0)
        && ((int)seqno - (prev_seqno + 1) < 10)) {
        unsigned char *data = rtp->f.data.ptr;
 
        memmove(rtp->f.data.ptr+3, rtp->f.data.ptr, rtp->f.datalen);
        rtp->f.datalen +=3;
        *data++ = 0xEF;
        *data++ = 0xBF;
        *data = 0xBD;
    }
 
    if (ast_format_cmp(rtp->f.subclass.format, ast_format_t140_red) == AST_FORMAT_CMP_EQUAL) {
        unsigned char *data = rtp->f.data.ptr;
        unsigned char *header_end;
        int num_generations;
        int header_length;
        int len;
        int diff =(int)seqno - (prev_seqno+1); /* if diff = 0, no drop*/
        int x;
 
        ao2_replace(rtp->f.subclass.format, ast_format_t140);
        header_end = memchr(data, ((*data) & 0x7f), rtp->f.datalen);
        if (header_end == NULL) {
            return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
        }
        header_end++;
 
        header_length = header_end - data;
        num_generations = header_length / 4;
        len = header_length;
 
        if (!diff) {
            for (x = 0; x < num_generations; x++)
                len += data[x * 4 + 3];
 
            if (!(rtp->f.datalen - len))
                return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
 
            rtp->f.data.ptr += len;
            rtp->f.datalen -= len;
        } else if (diff > num_generations && diff < 10) {
            len -= 3;
            rtp->f.data.ptr += len;
            rtp->f.datalen -= len;
 
            data = rtp->f.data.ptr;
            *data++ = 0xEF;
            *data++ = 0xBF;
            *data = 0xBD;
        } else {
            for ( x = 0; x < num_generations - diff; x++)
                len += data[x * 4 + 3];
 
            rtp->f.data.ptr += len;
            rtp->f.datalen -= len;
        }
    }
 
    if (ast_format_get_type(rtp->f.subclass.format) == AST_MEDIA_TYPE_AUDIO) {
        rtp->f.samples = ast_codec_samples_count(&rtp->f);
        if (ast_format_cache_is_slinear(rtp->f.subclass.format)) {
            ast_frame_byteswap_be(&rtp->f);
        }
        calc_rxstamp_and_jitter(&rtp->f.delivery, rtp, timestamp, mark);
        /* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
        ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
        rtp->f.ts = timestamp / (ast_rtp_get_rate(rtp->f.subclass.format) / 1000);
        rtp->f.len = rtp->f.samples / ((ast_format_get_sample_rate(rtp->f.subclass.format) / 1000));
    } else if (ast_format_get_type(rtp->f.subclass.format) == AST_MEDIA_TYPE_VIDEO) {
        /* Video -- samples is # of samples vs. 90000 */
        if (!rtp->lastividtimestamp)
            rtp->lastividtimestamp = timestamp;
        calc_rxstamp_and_jitter(&rtp->f.delivery, rtp, timestamp, mark);
        ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
        rtp->f.ts = timestamp / (ast_rtp_get_rate(rtp->f.subclass.format) / 1000);
        rtp->f.samples = timestamp - rtp->lastividtimestamp;
        rtp->lastividtimestamp = timestamp;
        rtp->f.delivery.tv_sec = 0;
        rtp->f.delivery.tv_usec = 0;
        /* Pass the RTP marker bit as bit */
        rtp->f.subclass.frame_ending = mark ? 1 : 0;
    } else if (ast_format_get_type(rtp->f.subclass.format) == AST_MEDIA_TYPE_TEXT) {
        /* TEXT -- samples is # of samples vs. 1000 */
        if (!rtp->lastitexttimestamp)
            rtp->lastitexttimestamp = timestamp;
        rtp->f.samples = timestamp - rtp->lastitexttimestamp;
        rtp->lastitexttimestamp = timestamp;
        rtp->f.delivery.tv_sec = 0;
        rtp->f.delivery.tv_usec = 0;
    } else {
        ast_log(LOG_WARNING, "Unknown or unsupported media type: %s\n",
            ast_codec_media_type2str(ast_format_get_type(rtp->f.subclass.format)));
        return &ast_null_frame;
    }
 
    AST_LIST_INSERT_TAIL(&frames, &rtp->f, frame_list);
    return AST_LIST_FIRST(&frames);
}

References ao2_cleanup, ao2_ref, ao2_replace, ast_codec_media_type2str(), ast_codec_samples_count(), AST_CONTROL_SRCCHANGE, ast_debug, ast_debug_rtp_packet_is_allowed, ast_format_cache_is_slinear(), ast_format_cmp(), AST_FORMAT_CMP_EQUAL, ast_format_get_sample_rate(), ast_format_get_type(), ast_format_t140, ast_format_t140_red, ast_frame_byteswap_be, AST_FRAME_CONTROL, AST_FRAME_DTMF_END, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, AST_FRFLAG_HAS_SEQUENCE_NUMBER, AST_FRFLAG_HAS_TIMING_INFO, ast_frfree, AST_FRIENDLY_OFFSET, ast_frisolate, AST_LIST_FIRST, AST_LIST_HEAD_INIT_NOLOCK, AST_LIST_INSERT_TAIL, AST_LIST_REMOVE_HEAD, ast_log, AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_IMAGE, AST_MEDIA_TYPE_TEXT, AST_MEDIA_TYPE_VIDEO, ast_null_frame, ast_rtcp_calc_interval(), ast_rtcp_write(), AST_RTP_CISCO_DTMF, AST_RTP_CN, ast_rtp_codecs_get_payload(), AST_RTP_DTMF, ast_rtp_dtmf_continuation(), ast_rtp_get_rate(), ast_rtp_instance_get_bridged(), ast_rtp_instance_get_codecs(), ast_rtp_instance_get_data(), ast_samp2tv(), ast_sched_add(), ast_set_flag, ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_tv(), ast_tvdiff_ms(), bridge_p2p_rtp_write(), calc_rxstamp_and_jitter(), create_dtmf_frame(), ast_rtp::cycles, ast_frame::data, ast_frame::datalen, DEBUG_ATLEAST, ast_frame::delivery, ast_rtp::dtmf_duration, ast_rtp::dtmf_timeout, ast_rtcp::expected_prior, ext, ast_rtp::f, ast_frame_subclass::format, ast_frame_subclass::frame_ending, frames, ast_frame::frametype, ast_frame_subclass::integer, optional_ts::is_set, ast_rtp::last_end_timestamp, ast_rtp::last_seqno, ast_rtp::lastitexttimestamp, ast_rtp::lastividtimestamp, ast_rtp::lastrxformat, len(), ast_frame::len, LOG_DEBUG, LOG_NOTICE, LOG_WARNING, ast_frame::mallocd, NULL, ast_frame::offset, process_cn_rfc3389(), process_dtmf_cisco(), process_dtmf_rfc2833(), ast_frame::ptr, RAII_VAR, ast_rtcp::received_prior, res_srtp, ast_rtp::resp, ast_rtp::rtcp, rtp_instance_parse_extmap_extensions(), RTP_SEQ_MOD, ast_rtp::rxcount, ast_rtp::rxoctetcount, ast_frame::samples, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::seedrxseqno, ast_rtp::sending_digit, ast_frame::seqno, ast_frame::src, srtp_replay_protection, ast_frame::stream_num, ast_rtp::stream_num, ast_frame::subclass, ast_rtcp::them, ast_rtp::themssrc, ast_rtp::themssrc_valid, ast_frame::ts, optional_ts::ts, and ast_srtp_res::unprotect.

Referenced by ast_rtp_read().

ast_rtp_local_bridge()

static int ast_rtp_local_bridge ( struct ast_rtp_instance *  instance0, struct ast_rtp_instance *  instance1  )

static

Precondition

Neither instance0 nor instance1 are locked

Definition at line 9457 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance0);
 
    ao2_lock(instance0);
    ast_set_flag(rtp, FLAG_NEED_MARKER_BIT | FLAG_REQ_LOCAL_BRIDGE_BIT);
    if (rtp->smoother) {
        ast_smoother_free(rtp->smoother);
        rtp->smoother = NULL;
    }
 
    /* We must use a new SSRC when local bridge ends */
    if (!instance1) {
        rtp->ssrc = rtp->ssrc_orig;
        rtp->ssrc_orig = 0;
        rtp->ssrc_saved = 0;
    } else if (!rtp->ssrc_saved) {
        /* In case ast_rtp_local_bridge is called multiple times, only save the ssrc from before local bridge began */
        rtp->ssrc_orig = rtp->ssrc;
        rtp->ssrc_saved = 1;
    }
 
    ao2_unlock(instance0);
 
    return 0;
}

References ao2_lock, ao2_unlock, ast_rtp_instance_get_data(), ast_set_flag, ast_smoother_free(), FLAG_NEED_MARKER_BIT, FLAG_REQ_LOCAL_BRIDGE_BIT, NULL, ast_rtp::smoother, ast_rtp::ssrc, ast_rtp::ssrc_orig, and ast_rtp::ssrc_saved.

ast_rtp_new()

static int ast_rtp_new ( struct ast_rtp_instance *  instance, struct ast_sched_context *  sched, struct ast_sockaddr *  addr, void *  data  )

static

Precondition

instance is locked

Definition at line 4332 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = NULL;
 
    /* Create a new RTP structure to hold all of our data */
    if (!(rtp = ast_calloc(1, sizeof(*rtp)))) {
        return -1;
    }
    rtp->owner = instance;
    /* Set default parameters on the newly created RTP structure */
    rtp->ssrc = ast_random();
    ast_uuid_generate_str(rtp->cname, sizeof(rtp->cname));
    rtp->seqno = ast_random() & 0xffff;
    rtp->expectedrxseqno = -1;
    rtp->expectedseqno = -1;
    rtp->rxstart = -1;
    rtp->sched = sched;
    ast_sockaddr_copy(&rtp->bind_address, addr);
    /* Transport creation operations can grab the RTP data from the instance, so set it */
    ast_rtp_instance_set_data(instance, rtp);
 
    if (rtp_allocate_transport(instance, rtp)) {
        return -1;
    }
 
    if (AST_VECTOR_INIT(&rtp->ssrc_mapping, 1)) {
        return -1;
    }
 
    if (AST_VECTOR_INIT(&rtp->transport_wide_cc.packet_statistics, 0)) {
        return -1;
    }
    rtp->transport_wide_cc.schedid = -1;
 
    rtp->f.subclass.format = ao2_bump(ast_format_none);
    rtp->lastrxformat = ao2_bump(ast_format_none);
    rtp->lasttxformat = ao2_bump(ast_format_none);
    rtp->stream_num = -1;
 
    return 0;
}

References ao2_bump, ast_calloc, ast_format_none, ast_random(), ast_rtp_instance_set_data(), ast_sockaddr_copy(), ast_uuid_generate_str(), AST_VECTOR_INIT, ast_rtp::bind_address, ast_rtp::cname, ast_rtp::expectedrxseqno, ast_rtp::expectedseqno, ast_rtp::f, ast_frame_subclass::format, ast_rtp::lastrxformat, ast_rtp::lasttxformat, NULL, ast_rtp::owner, rtp_transport_wide_cc_statistics::packet_statistics, rtp_allocate_transport(), ast_rtp::rxstart, ast_rtp::sched, rtp_transport_wide_cc_statistics::schedid, ast_rtp::seqno, ast_rtp::ssrc, ast_rtp::ssrc_mapping, ast_rtp::stream_num, ast_frame::subclass, and ast_rtp::transport_wide_cc.

ast_rtp_prop_set()

static void ast_rtp_prop_set ( struct ast_rtp_instance *  instance, enum ast_rtp_property  property, int  value  )

static

Precondition

instance is locked

Definition at line 9087 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    if (property == AST_RTP_PROPERTY_RTCP) {
        if (value) {
            struct ast_sockaddr local_addr;
 
            if (rtp->rtcp && rtp->rtcp->type == value) {
                ast_debug_rtcp(1, "(%p) RTCP ignoring duplicate property\n", instance);
                return;
            }
 
            if (!rtp->rtcp) {
                rtp->rtcp = ast_calloc(1, sizeof(*rtp->rtcp));
                if (!rtp->rtcp) {
                    return;
                }
                rtp->rtcp->s = -1;
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
                rtp->rtcp->dtls.timeout_timer = -1;
#endif
                rtp->rtcp->schedid = -1;
            }
 
            rtp->rtcp->type = value;
 
            /* Grab the IP address and port we are going to use */
            ast_rtp_instance_get_local_address(instance, &rtp->rtcp->us);
            if (value == AST_RTP_INSTANCE_RTCP_STANDARD) {
                ast_sockaddr_set_port(&rtp->rtcp->us,
                    ast_sockaddr_port(&rtp->rtcp->us) + 1);
            }
 
            ast_sockaddr_copy(&local_addr, &rtp->rtcp->us);
            if (!ast_find_ourip(&local_addr, &rtp->rtcp->us, 0)) {
                ast_sockaddr_set_port(&local_addr, ast_sockaddr_port(&rtp->rtcp->us));
            } else {
                /* Failed to get local address reset to use default. */
                ast_sockaddr_copy(&local_addr, &rtp->rtcp->us);
            }
 
            ast_free(rtp->rtcp->local_addr_str);
            rtp->rtcp->local_addr_str = ast_strdup(ast_sockaddr_stringify(&local_addr));
            if (!rtp->rtcp->local_addr_str) {
                ast_free(rtp->rtcp);
                rtp->rtcp = NULL;
                return;
            }
 
            if (value == AST_RTP_INSTANCE_RTCP_STANDARD) {
                /* We're either setting up RTCP from scratch or
                 * switching from MUX. Either way, we won't have
                 * a socket set up, and we need to set it up
                 */
                if ((rtp->rtcp->s = create_new_socket("RTCP", &rtp->rtcp->us)) < 0) {
                    ast_debug_rtcp(1, "(%p) RTCP failed to create a new socket\n", instance);
                    ast_free(rtp->rtcp->local_addr_str);
                    ast_free(rtp->rtcp);
                    rtp->rtcp = NULL;
                    return;
                }
 
                /* Try to actually bind to the IP address and port we are going to use for RTCP, if this fails we have to bail out */
                if (ast_bind(rtp->rtcp->s, &rtp->rtcp->us)) {
                    ast_debug_rtcp(1, "(%p) RTCP failed to setup RTP instance\n", instance);
                    close(rtp->rtcp->s);
                    ast_free(rtp->rtcp->local_addr_str);
                    ast_free(rtp->rtcp);
                    rtp->rtcp = NULL;
                    return;
                }
#ifdef HAVE_PJPROJECT
                if (rtp->ice) {
                    rtp_add_candidates_to_ice(instance, rtp, &rtp->rtcp->us, ast_sockaddr_port(&rtp->rtcp->us), AST_RTP_ICE_COMPONENT_RTCP, TRANSPORT_SOCKET_RTCP);
                }
#endif
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
                dtls_setup_rtcp(instance);
#endif
            } else {
                struct ast_sockaddr addr;
                /* RTCPMUX uses the same socket as RTP. If we were previously using standard RTCP
                 * then close the socket we previously created.
                 *
                 * It may seem as though there is a possible race condition here where we might try
                 * to close the RTCP socket while it is being used to send data. However, this is not
                 * a problem in practice since setting and adjusting of RTCP properties happens prior
                 * to activating RTP. It is not until RTP is activated that timers start for RTCP
                 * transmission
                 */
                if (rtp->rtcp->s > -1 && rtp->rtcp->s != rtp->s) {
                    close(rtp->rtcp->s);
                }
                rtp->rtcp->s = rtp->s;
                ast_rtp_instance_get_remote_address(instance, &addr);
                ast_sockaddr_copy(&rtp->rtcp->them, &addr);
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
                if (rtp->rtcp->dtls.ssl && rtp->rtcp->dtls.ssl != rtp->dtls.ssl) {
                    SSL_free(rtp->rtcp->dtls.ssl);
                }
                rtp->rtcp->dtls.ssl = rtp->dtls.ssl;
#endif
            }
 
            ast_debug_rtcp(1, "(%s) RTCP setup on RTP instance\n",
                ast_rtp_instance_get_channel_id(instance));
        } else {
            if (rtp->rtcp) {
                if (rtp->rtcp->schedid > -1) {
                    ao2_unlock(instance);
                    if (!ast_sched_del(rtp->sched, rtp->rtcp->schedid)) {
                        /* Successfully cancelled scheduler entry. */
                        ao2_ref(instance, -1);
                    } else {
                        /* Unable to cancel scheduler entry */
                        ast_debug_rtcp(1, "(%p) RTCP failed to tear down RTCP\n", instance);
                        ao2_lock(instance);
                        return;
                    }
                    ao2_lock(instance);
                    rtp->rtcp->schedid = -1;
                }
                if (rtp->transport_wide_cc.schedid > -1) {
                    ao2_unlock(instance);
                    if (!ast_sched_del(rtp->sched, rtp->transport_wide_cc.schedid)) {
                        ao2_ref(instance, -1);
                    } else {
                        ast_debug_rtcp(1, "(%p) RTCP failed to tear down transport-cc feedback\n", instance);
                        ao2_lock(instance);
                        return;
                    }
                    ao2_lock(instance);
                    rtp->transport_wide_cc.schedid = -1;
                }
                if (rtp->rtcp->s > -1 && rtp->rtcp->s != rtp->s) {
                    close(rtp->rtcp->s);
                }
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
                ao2_unlock(instance);
                dtls_srtp_stop_timeout_timer(instance, rtp, 1);
                ao2_lock(instance);
 
                if (rtp->rtcp->dtls.ssl && rtp->rtcp->dtls.ssl != rtp->dtls.ssl) {
                    SSL_free(rtp->rtcp->dtls.ssl);
                }
#endif
                ast_free(rtp->rtcp->local_addr_str);
                ast_free(rtp->rtcp);
                rtp->rtcp = NULL;
                ast_debug_rtcp(1, "(%s) RTCP torn down on RTP instance\n",
                    ast_rtp_instance_get_channel_id(instance));
            }
        }
    } else if (property == AST_RTP_PROPERTY_ASYMMETRIC_CODEC) {
        rtp->asymmetric_codec = value;
    } else if (property == AST_RTP_PROPERTY_RETRANS_SEND) {
        if (value) {
            if (!rtp->send_buffer) {
                rtp->send_buffer = ast_data_buffer_alloc(ast_free_ptr, DEFAULT_RTP_SEND_BUFFER_SIZE);
            }
        } else {
            if (rtp->send_buffer) {
                ast_data_buffer_free(rtp->send_buffer);
                rtp->send_buffer = NULL;
            }
        }
    } else if (property == AST_RTP_PROPERTY_RETRANS_RECV) {
        if (value) {
            if (!rtp->recv_buffer) {
                rtp->recv_buffer = ast_data_buffer_alloc(ast_free_ptr, DEFAULT_RTP_RECV_BUFFER_SIZE);
                AST_VECTOR_INIT(&rtp->missing_seqno, 0);
            }
        } else {
            if (rtp->recv_buffer) {
                ast_data_buffer_free(rtp->recv_buffer);
                rtp->recv_buffer = NULL;
                AST_VECTOR_FREE(&rtp->missing_seqno);
            }
        }
    }
}

References ao2_lock, ao2_ref, ao2_unlock, ast_bind(), ast_calloc, ast_data_buffer_alloc(), ast_data_buffer_free(), ast_debug_rtcp, ast_find_ourip(), ast_free, ast_free_ptr(), AST_RTP_ICE_COMPONENT_RTCP, ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_data(), ast_rtp_instance_get_local_address(), ast_rtp_instance_get_remote_address, AST_RTP_INSTANCE_RTCP_STANDARD, AST_RTP_PROPERTY_ASYMMETRIC_CODEC, AST_RTP_PROPERTY_RETRANS_RECV, AST_RTP_PROPERTY_RETRANS_SEND, AST_RTP_PROPERTY_RTCP, ast_sched_del(), ast_sockaddr_copy(), ast_sockaddr_port, ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_strdup, AST_VECTOR_FREE, AST_VECTOR_INIT, ast_rtp::asymmetric_codec, create_new_socket(), DEFAULT_RTP_RECV_BUFFER_SIZE, DEFAULT_RTP_SEND_BUFFER_SIZE, ast_rtcp::local_addr_str, ast_rtp::missing_seqno, NULL, ast_rtp::recv_buffer, ast_rtp::rtcp, ast_rtp::s, ast_rtcp::s, ast_rtp::sched, rtp_transport_wide_cc_statistics::schedid, ast_rtcp::schedid, ast_rtp::send_buffer, ast_rtcp::them, TRANSPORT_SOCKET_RTCP, ast_rtp::transport_wide_cc, ast_rtcp::type, ast_rtcp::us, and value.

ast_rtp_qos_set()

static int ast_rtp_qos_set ( struct ast_rtp_instance *  instance, int  tos, int  cos, const char *  desc  )

static

Precondition

instance is locked

Definition at line 9633 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    return ast_set_qos(rtp->s, tos, cos, desc);
}

References ast_rtp_instance_get_data(), ast_set_qos(), cos, desc, ast_rtp::s, and tos.

ast_rtp_read()

static struct ast_frame * ast_rtp_read ( struct ast_rtp_instance *  instance, int  rtcp  )

static

Precondition

instance is locked

Definition at line 8420 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_srtp *srtp;
    RAII_VAR(struct ast_rtp_instance *, child, NULL, rtp_instance_unlock);
    struct ast_sockaddr addr;
    int res, hdrlen = 12, version, payloadtype;
    unsigned char *read_area = rtp->rawdata + AST_FRIENDLY_OFFSET;
    size_t read_area_size = sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET;
    unsigned int *rtpheader = (unsigned int*)(read_area), seqno, ssrc, timestamp, prev_seqno;
    struct ast_sockaddr remote_address = { {0,} };
    struct frame_list frames;
    struct ast_frame *frame;
    unsigned int bundled;
 
    /* If this is actually RTCP let's hop on over and handle it */
    if (rtcp) {
        if (rtp->rtcp && rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_STANDARD) {
            return ast_rtcp_read(instance);
        }
        return &ast_null_frame;
    }
 
    /* Actually read in the data from the socket */
    if ((res = rtp_recvfrom(instance, read_area, read_area_size, 0,
                &addr)) < 0) {
        if (res == RTP_DTLS_ESTABLISHED) {
            rtp->f.frametype = AST_FRAME_CONTROL;
            rtp->f.subclass.integer = AST_CONTROL_SRCCHANGE;
            return &rtp->f;
        }
 
        ast_assert(errno != EBADF);
        if (errno != EAGAIN) {
            ast_log(LOG_WARNING, "RTP Read error: %s.  Hanging up.\n",
                (errno) ? strerror(errno) : "Unspecified");
            return NULL;
        }
        return &ast_null_frame;
    }
 
    /* If this was handled by the ICE session don't do anything */
    if (!res) {
        return &ast_null_frame;
    }
 
    /* This could be a multiplexed RTCP packet. If so, be sure to interpret it correctly */
    if (rtcp_mux(rtp, read_area)) {
        return ast_rtcp_interpret(instance, ast_rtp_instance_get_srtp(instance, 1), read_area, res, &addr);
    }
 
    /* Make sure the data that was read in is actually enough to make up an RTP packet */
    if (res < hdrlen) {
        /* If this is a keepalive containing only nulls, don't bother with a warning */
        int i;
        for (i = 0; i < res; ++i) {
            if (read_area[i] != '\0') {
                ast_log(LOG_WARNING, "RTP Read too short\n");
                return &ast_null_frame;
            }
        }
        return &ast_null_frame;
    }
 
    /* Get fields and verify this is an RTP packet */
    seqno = ntohl(rtpheader[0]);
 
    ast_rtp_instance_get_remote_address(instance, &remote_address);
 
    if (!(version = (seqno & 0xC0000000) >> 30)) {
        struct sockaddr_in addr_tmp;
        struct ast_sockaddr addr_v4;
        if (ast_sockaddr_is_ipv4(&addr)) {
            ast_sockaddr_to_sin(&addr, &addr_tmp);
        } else if (ast_sockaddr_ipv4_mapped(&addr, &addr_v4)) {
            ast_debug_stun(1, "(%p) STUN using IPv6 mapped address %s\n",
                instance, ast_sockaddr_stringify(&addr));
            ast_sockaddr_to_sin(&addr_v4, &addr_tmp);
        } else {
            ast_debug_stun(1, "(%p) STUN cannot do for non IPv4 address %s\n",
                instance, ast_sockaddr_stringify(&addr));
            return &ast_null_frame;
        }
        if ((ast_stun_handle_packet(rtp->s, &addr_tmp, read_area, res, NULL, NULL) == AST_STUN_ACCEPT) &&
            ast_sockaddr_isnull(&remote_address)) {
            ast_sockaddr_from_sin(&addr, &addr_tmp);
            ast_rtp_instance_set_remote_address(instance, &addr);
        }
        return &ast_null_frame;
    }
 
    /* If the version is not what we expected by this point then just drop the packet */
    if (version != 2) {
        return &ast_null_frame;
    }
 
    /* We use the SSRC to determine what RTP instance this packet is actually for */
    ssrc = ntohl(rtpheader[2]);
 
    /* We use the SRTP data from the provided instance that it came in on, not the child */
    srtp = ast_rtp_instance_get_srtp(instance, 0);
 
    /* Determine the appropriate instance for this */
    child = rtp_find_instance_by_packet_source_ssrc(instance, rtp, ssrc);
    if (!child) {
        /* Neither the bundled parent nor any child has this SSRC */
        return &ast_null_frame;
    }
    if (child != instance) {
        /* It is safe to hold the child lock while holding the parent lock, we guarantee that the locking order
         * is always parent->child or that the child lock is not held when acquiring the parent lock.
         */
        ao2_lock(child);
        instance = child;
        rtp = ast_rtp_instance_get_data(instance);
    } else {
        /* The child is the parent! We don't need to unlock it. */
        child = NULL;
    }
 
    /* If strict RTP protection is enabled see if we need to learn the remote address or if we need to drop the packet */
    switch (rtp->strict_rtp_state) {
    case STRICT_RTP_LEARN:
        /*
         * Scenario setup:
         * PartyA -- Ast1 -- Ast2 -- PartyB
         *
         * The learning timeout is necessary for Ast1 to handle the above
         * setup where PartyA calls PartyB and Ast2 initiates direct media
         * between Ast1 and PartyB.  Ast1 may lock onto the Ast2 stream and
         * never learn the PartyB stream when it starts.  The timeout makes
         * Ast1 stay in the learning state long enough to see and learn the
         * RTP stream from PartyB.
         *
         * To mitigate against attack, the learning state cannot switch
         * streams while there are competing streams.  The competing streams
         * interfere with each other's qualification.  Once we accept a
         * stream and reach the timeout, an attacker cannot interfere
         * anymore.
         *
         * Here are a few scenarios and each one assumes that the streams
         * are continuous:
         *
         * 1) We already have a known stream source address and the known
         * stream wants to change to a new source address.  An attacking
         * stream will block learning the new stream source.  After the
         * timeout we re-lock onto the original stream source address which
         * likely went away.  The result is one way audio.
         *
         * 2) We already have a known stream source address and the known
         * stream doesn't want to change source addresses.  An attacking
         * stream will not be able to replace the known stream.  After the
         * timeout we re-lock onto the known stream.  The call is not
         * affected.
         *
         * 3) We don't have a known stream source address.  This presumably
         * is the start of a call.  Competing streams will result in staying
         * in learning mode until a stream becomes the victor and we reach
         * the timeout.  We cannot exit learning if we have no known stream
         * to lock onto.  The result is one way audio until there is a victor.
         *
         * If we learn a stream source address before the timeout we will be
         * in scenario 1) or 2) when a competing stream starts.
         */
        if (!ast_sockaddr_isnull(&rtp->strict_rtp_address)
            && STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(), rtp->rtp_source_learn.start)) {
            ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address %s\n",
                rtp, ast_sockaddr_stringify(&rtp->strict_rtp_address));
            ast_test_suite_event_notify("STRICT_RTP_LEARN", "Source: %s",
                ast_sockaddr_stringify(&rtp->strict_rtp_address));
            rtp->strict_rtp_state = STRICT_RTP_CLOSED;
        } else {
            struct ast_sockaddr target_address;
 
            if (!ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
                /*
                 * We are open to learning a new address but have received
                 * traffic from the current address, accept it and reset
                 * the learning counts for a new source.  When no more
                 * current source packets arrive a new source can take over
                 * once sufficient traffic is received.
                 */
                rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
                break;
            }
 
            /*
             * We give preferential treatment to the requested target address
             * (negotiated SDP address) where we are to send our RTP.  However,
             * the other end has no obligation to send from that address even
             * though it is practically a requirement when NAT is involved.
             */
            ast_rtp_instance_get_requested_target_address(instance, &target_address);
            if (!ast_sockaddr_cmp(&target_address, &addr)) {
                /* Accept the negotiated target RTP stream as the source */
                ast_verb(4, "%p -- Strict RTP switching to RTP target address %s as source\n",
                    rtp, ast_sockaddr_stringify(&addr));
                ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
                rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
                break;
            }
 
            /*
             * Trying to learn a new address.  If we pass a probationary period
             * with it, that means we've stopped getting RTP from the original
             * source and we should switch to it.
             */
            if (!ast_sockaddr_cmp(&rtp->rtp_source_learn.proposed_address, &addr)) {
                if (rtp->rtp_source_learn.stream_type == AST_MEDIA_TYPE_UNKNOWN) {
                    struct ast_rtp_codecs *codecs;
 
                    codecs = ast_rtp_instance_get_codecs(instance);
                    rtp->rtp_source_learn.stream_type =
                        ast_rtp_codecs_get_stream_type(codecs);
                    ast_verb(4, "%p -- Strict RTP qualifying stream type: %s\n",
                        rtp, ast_codec_media_type2str(rtp->rtp_source_learn.stream_type));
                }
                if (!rtp_learning_rtp_seq_update(&rtp->rtp_source_learn, seqno)) {
                    /* Accept the new RTP stream */
                    ast_verb(4, "%p -- Strict RTP switching source address to %s\n",
                        rtp, ast_sockaddr_stringify(&addr));
                    ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
                    rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
                    break;
                }
                /* Not ready to accept the RTP stream candidate */
                ast_debug_rtp(1, "(%p) RTP %p -- Received packet from %s, dropping due to strict RTP protection. Will switch to it in %d packets.\n",
                    instance, rtp, ast_sockaddr_stringify(&addr), rtp->rtp_source_learn.packets);
            } else {
                /*
                 * This is either an attacking stream or
                 * the start of the expected new stream.
                 */
                ast_sockaddr_copy(&rtp->rtp_source_learn.proposed_address, &addr);
                rtp_learning_seq_init(&rtp->rtp_source_learn, seqno);
                ast_debug_rtp(1, "(%p) RTP %p -- Received packet from %s, dropping due to strict RTP protection. Qualifying new stream.\n",
                    instance, rtp, ast_sockaddr_stringify(&addr));
            }
            return &ast_null_frame;
        }
        /* Fall through */
    case STRICT_RTP_CLOSED:
        /*
         * We should not allow a stream address change if the SSRC matches
         * once strictrtp learning is closed.  Any kind of address change
         * like this should have happened while we were in the learning
         * state.  We do not want to allow the possibility of an attacker
         * interfering with the RTP stream after the learning period.
         * An attacker could manage to get an RTCP packet redirected to
         * them which can contain the SSRC value.
         */
        if (!ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
            break;
        }
        ast_debug_rtp(1, "(%p) RTP %p -- Received packet from %s, dropping due to strict RTP protection.\n",
            instance, rtp, ast_sockaddr_stringify(&addr));
#ifdef TEST_FRAMEWORK
    {
        static int strict_rtp_test_event = 1;
        if (strict_rtp_test_event) {
            ast_test_suite_event_notify("STRICT_RTP_CLOSED", "Source: %s",
                ast_sockaddr_stringify(&addr));
            strict_rtp_test_event = 0; /* Only run this event once to prevent possible spam */
        }
    }
#endif
        return &ast_null_frame;
    case STRICT_RTP_OPEN:
        break;
    }
 
    /* If symmetric RTP is enabled see if the remote side is not what we expected and change where we are sending audio */
    if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
        if (ast_sockaddr_cmp(&remote_address, &addr)) {
            /* do not update the originally given address, but only the remote */
            ast_rtp_instance_set_incoming_source_address(instance, &addr);
            ast_sockaddr_copy(&remote_address, &addr);
            if (rtp->rtcp && rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_STANDARD) {
                ast_sockaddr_copy(&rtp->rtcp->them, &addr);
                ast_sockaddr_set_port(&rtp->rtcp->them, ast_sockaddr_port(&addr) + 1);
            }
            ast_set_flag(rtp, FLAG_NAT_ACTIVE);
            if (ast_debug_rtp_packet_is_allowed)
                ast_debug(0, "(%p) RTP NAT: Got audio from other end. Now sending to address %s\n",
                    instance, ast_sockaddr_stringify(&remote_address));
        }
    }
 
    /* Pull out the various other fields we will need */
    payloadtype = (seqno & 0x7f0000) >> 16;
    seqno &= 0xffff;
    timestamp = ntohl(rtpheader[1]);
 
#ifdef AST_DEVMODE
    if (should_drop_packets(&addr)) {
        ast_debug(0, "(%p) RTP: drop received packet from %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6d)\n",
            instance, ast_sockaddr_stringify(&addr), payloadtype, seqno, timestamp, res - hdrlen);
        return &ast_null_frame;
    }
#endif
 
    if (rtp_debug_test_addr(&addr)) {
        ast_verbose("Got  RTP packet from    %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6d)\n",
                ast_sockaddr_stringify(&addr),
                payloadtype, seqno, timestamp, res - hdrlen);
    }
 
    AST_LIST_HEAD_INIT_NOLOCK(&frames);
 
    bundled = (child || AST_VECTOR_SIZE(&rtp->ssrc_mapping)) ? 1 : 0;
 
    prev_seqno = rtp->lastrxseqno;
    /* We need to save lastrxseqno for use by jitter before resetting it. */
    rtp->prevrxseqno = rtp->lastrxseqno;
    rtp->lastrxseqno = seqno;
 
    if (!rtp->recv_buffer) {
        /* If there is no receive buffer then we can pass back the frame directly */
        frame = ast_rtp_interpret(instance, srtp, &addr, read_area, res, prev_seqno, bundled);
        AST_LIST_INSERT_TAIL(&frames, frame, frame_list);
        return AST_LIST_FIRST(&frames);
    } else if (rtp->expectedrxseqno == -1 || seqno == rtp->expectedrxseqno) {
        rtp->expectedrxseqno = seqno + 1;
 
        /* We've cycled over, so go back to 0 */
        if (rtp->expectedrxseqno == SEQNO_CYCLE_OVER) {
            rtp->expectedrxseqno = 0;
        }
 
        /* If there are no buffered packets that will be placed after this frame then we can
         * return it directly without duplicating it.
         */
        if (!ast_data_buffer_count(rtp->recv_buffer)) {
            frame = ast_rtp_interpret(instance, srtp, &addr, read_area, res, prev_seqno, bundled);
            AST_LIST_INSERT_TAIL(&frames, frame, frame_list);
            return AST_LIST_FIRST(&frames);
        }
 
        if (!AST_VECTOR_REMOVE_CMP_ORDERED(&rtp->missing_seqno, seqno, find_by_value,
            AST_VECTOR_ELEM_CLEANUP_NOOP)) {
            ast_debug_rtp(2, "(%p) RTP Packet with sequence number '%d' on instance is no longer missing\n",
                instance, seqno);
        }
 
        /* If we don't have the next packet after this we can directly return the frame, as there is no
         * chance it will be overwritten.
         */
        if (!ast_data_buffer_get(rtp->recv_buffer, rtp->expectedrxseqno)) {
            frame = ast_rtp_interpret(instance, srtp, &addr, read_area, res, prev_seqno, bundled);
            AST_LIST_INSERT_TAIL(&frames, frame, frame_list);
            return AST_LIST_FIRST(&frames);
        }
 
        /* Otherwise we need to dupe the frame so that the potential processing of frames placed after
         * it do not overwrite the data. You may be thinking that we could just add the current packet
         * to the head of the frames list and avoid having to duplicate it but this would result in out
         * of order packet processing by libsrtp which we are trying to avoid.
         */
        frame = ast_frdup(ast_rtp_interpret(instance, srtp, &addr, read_area, res, prev_seqno, bundled));
        if (frame) {
            AST_LIST_INSERT_TAIL(&frames, frame, frame_list);
            prev_seqno = seqno;
        }
 
        /* Add any additional packets that we have buffered and that are available */
        while (ast_data_buffer_count(rtp->recv_buffer)) {
            struct ast_rtp_rtcp_nack_payload *payload;
 
            payload = (struct ast_rtp_rtcp_nack_payload *)ast_data_buffer_remove(rtp->recv_buffer, rtp->expectedrxseqno);
            if (!payload) {
                break;
            }
 
            frame = ast_frdup(ast_rtp_interpret(instance, srtp, &addr, payload->buf, payload->size, prev_seqno, bundled));
            ast_free(payload);
 
            if (!frame) {
                /* If this packet can't be interpreted due to being out of memory we return what we have and assume
                 * that we will determine it is a missing packet later and NACK for it.
                 */
                return AST_LIST_FIRST(&frames);
            }
 
            ast_debug_rtp(2, "(%p) RTP pulled buffered packet with sequence number '%d' to additionally return\n",
                instance, frame->seqno);
            AST_LIST_INSERT_TAIL(&frames, frame, frame_list);
            prev_seqno = rtp->expectedrxseqno;
            rtp->expectedrxseqno++;
            if (rtp->expectedrxseqno == SEQNO_CYCLE_OVER) {
                rtp->expectedrxseqno = 0;
            }
        }
 
        return AST_LIST_FIRST(&frames);
    } else if ((((seqno - rtp->expectedrxseqno) > 100) && timestamp > rtp->lastividtimestamp) ||
        ast_data_buffer_count(rtp->recv_buffer) == ast_data_buffer_max(rtp->recv_buffer)) {
        int inserted = 0;
 
        /* We have a large number of outstanding buffered packets or we've jumped far ahead in time.
         * To compensate we dump what we have in the buffer and place the current packet in a logical
         * spot. In the case of video we also require a full frame to give the decoding side a fighting
         * chance.
         */
 
        if (rtp->rtp_source_learn.stream_type == AST_MEDIA_TYPE_VIDEO) {
            ast_debug_rtp(2, "(%p) RTP source has wild gap or packet loss, sending FIR\n",
                instance);
            rtp_write_rtcp_fir(instance, rtp, &remote_address);
        }
 
        /* This works by going through the progression of the sequence number retrieving buffered packets
         * or inserting the current received packet until we've run out of packets. This ensures that the
         * packets are in the correct sequence number order.
         */
        while (ast_data_buffer_count(rtp->recv_buffer)) {
            struct ast_rtp_rtcp_nack_payload *payload;
 
            /* If the packet we received is the one we are expecting at this point then add it in */
            if (rtp->expectedrxseqno == seqno) {
                frame = ast_frdup(ast_rtp_interpret(instance, srtp, &addr, read_area, res, prev_seqno, bundled));
                if (frame) {
                    AST_LIST_INSERT_TAIL(&frames, frame, frame_list);
                    prev_seqno = seqno;
                    ast_debug_rtp(2, "(%p) RTP inserted just received packet with sequence number '%d' in correct order\n",
                        instance, seqno);
                }
                /* It is possible due to packet retransmission for this packet to also exist in the receive
                 * buffer so we explicitly remove it in case this occurs, otherwise the receive buffer will
                 * never be empty.
                 */
                payload = (struct ast_rtp_rtcp_nack_payload *)ast_data_buffer_remove(rtp->recv_buffer, seqno);
                if (payload) {
                    ast_free(payload);
                }
                rtp->expectedrxseqno++;
                if (rtp->expectedrxseqno == SEQNO_CYCLE_OVER) {
                    rtp->expectedrxseqno = 0;
                }
                inserted = 1;
                continue;
            }
 
            payload = (struct ast_rtp_rtcp_nack_payload *)ast_data_buffer_remove(rtp->recv_buffer, rtp->expectedrxseqno);
            if (payload) {
                frame = ast_frdup(ast_rtp_interpret(instance, srtp, &addr, payload->buf, payload->size, prev_seqno, bundled));
                if (frame) {
                    AST_LIST_INSERT_TAIL(&frames, frame, frame_list);
                    prev_seqno = rtp->expectedrxseqno;
                    ast_debug_rtp(2, "(%p) RTP emptying queue and returning packet with sequence number '%d'\n",
                        instance, frame->seqno);
                }
                ast_free(payload);
            }
 
            rtp->expectedrxseqno++;
            if (rtp->expectedrxseqno == SEQNO_CYCLE_OVER) {
                rtp->expectedrxseqno = 0;
            }
        }
 
        if (!inserted) {
            /* This current packet goes after them, and we assume that packets going forward will follow
             * that new sequence number increment. It is okay for this to not be duplicated as it is guaranteed
             * to be the last packet processed right now and it is also guaranteed that it will always return
             * non-NULL.
             */
            frame = ast_rtp_interpret(instance, srtp, &addr, read_area, res, prev_seqno, bundled);
            AST_LIST_INSERT_TAIL(&frames, frame, frame_list);
            rtp->expectedrxseqno = seqno + 1;
            if (rtp->expectedrxseqno == SEQNO_CYCLE_OVER) {
                rtp->expectedrxseqno = 0;
            }
 
            ast_debug_rtp(2, "(%p) RTP adding just received packet with sequence number '%d' to end of dumped queue\n",
                instance, seqno);
        }
 
        /* When we flush increase our chance for next time by growing the receive buffer when possible
         * by how many packets we missed, to give ourselves a bit more breathing room.
         */
        ast_data_buffer_resize(rtp->recv_buffer, MIN(MAXIMUM_RTP_RECV_BUFFER_SIZE,
            ast_data_buffer_max(rtp->recv_buffer) + AST_VECTOR_SIZE(&rtp->missing_seqno)));
        ast_debug_rtp(2, "(%p) RTP receive buffer is now at maximum of %zu\n", instance, ast_data_buffer_max(rtp->recv_buffer));
 
        /* As there is such a large gap we don't want to flood the order side with missing packets, so we
         * give up and start anew.
         */
        AST_VECTOR_RESET(&rtp->missing_seqno, AST_VECTOR_ELEM_CLEANUP_NOOP);
 
        return AST_LIST_FIRST(&frames);
    }
 
    /* We're finished with the frames list */
    ast_frame_free(AST_LIST_FIRST(&frames), 0);
 
    /* Determine if the received packet is from the last OLD_PACKET_COUNT (1000 by default) packets or not.
     * For the case where the received sequence number exceeds that of the expected sequence number we calculate
     * the past sequence number that would be 1000 sequence numbers ago. If the received sequence number
     * exceeds or meets that then it is within OLD_PACKET_COUNT packets ago. For example if the expected
     * sequence number is 100 and we receive 65530, then it would be considered old. This is because
     * 65535 - 1000 + 100 = 64635 which gives us the sequence number at which we would consider the packets
     * old. Since 65530 is above that, it would be considered old.
     * For the case where the received sequence number is less than the expected sequence number we can do
     * a simple subtraction to see if it is 1000 packets ago or not.
     */
    if ((seqno < rtp->expectedrxseqno && ((rtp->expectedrxseqno - seqno) <= OLD_PACKET_COUNT)) ||
        (seqno > rtp->expectedrxseqno && (seqno >= (65535 - OLD_PACKET_COUNT + rtp->expectedrxseqno)))) {
        /* If this is a packet from the past then we have received a duplicate packet, so just drop it */
        ast_debug_rtp(2, "(%p) RTP received an old packet with sequence number '%d', dropping it\n",
            instance, seqno);
        return &ast_null_frame;
    } else if (ast_data_buffer_get(rtp->recv_buffer, seqno)) {
        /* If this is a packet we already have buffered then it is a duplicate, so just drop it */
        ast_debug_rtp(2, "(%p) RTP received a duplicate transmission of packet with sequence number '%d', dropping it\n",
            instance, seqno);
        return &ast_null_frame;
    } else {
        /* This is an out of order packet from the future */
        struct ast_rtp_rtcp_nack_payload *payload;
        int missing_seqno;
        int remove_failed;
        unsigned int missing_seqnos_added = 0;
 
        ast_debug_rtp(2, "(%p) RTP received an out of order packet with sequence number '%d' while expecting '%d' from the future\n",
            instance, seqno, rtp->expectedrxseqno);
 
        payload = ast_malloc(sizeof(*payload) + res);
        if (!payload) {
            /* If the payload can't be allocated then we can't defer this packet right now.
             * Instead of dumping what we have we pretend we lost this packet. It will then
             * get NACKed later or the existing buffer will be returned entirely. Well, we may
             * try since we're seemingly out of memory. It's a bad situation all around and
             * packets are likely to get lost anyway.
             */
            return &ast_null_frame;
        }
 
        payload->size = res;
        memcpy(payload->buf, rtpheader, res);
        if (ast_data_buffer_put(rtp->recv_buffer, seqno, payload) == -1) {
            ast_free(payload);
        }
 
        /* If this sequence number is removed that means we had a gap and this packet has filled it in
         * some. Since it was part of the gap we will have already added any other missing sequence numbers
         * before it (and possibly after it) to the vector so we don't need to do that again. Note that
         * remove_failed will be set to -1 if the sequence number isn't removed, and 0 if it is.
         */
        remove_failed = AST_VECTOR_REMOVE_CMP_ORDERED(&rtp->missing_seqno, seqno, find_by_value,
            AST_VECTOR_ELEM_CLEANUP_NOOP);
        if (!remove_failed) {
            ast_debug_rtp(2, "(%p) RTP packet with sequence number '%d' is no longer missing\n",
                instance, seqno);
        }
 
        /* The missing sequence number code works by taking the sequence number of the
         * packet we've just received and going backwards until we hit the sequence number
         * of the last packet we've received. While doing so we check to make sure that the
         * sequence number is not already missing and that it is not already buffered.
         */
        missing_seqno = seqno;
        while (remove_failed) {
            missing_seqno -= 1;
 
            /* If we've cycled backwards then start back at the top */
            if (missing_seqno < 0) {
                missing_seqno = 65535;
            }
 
            /* We've gone backwards enough such that we've hit the previous sequence number */
            if (missing_seqno == prev_seqno) {
                break;
            }
 
            /* We don't want missing sequence number duplicates. If, for some reason,
             * packets are really out of order, we could end up in this scenario:
             *
             * We are expecting sequence number 100
             * We receive sequence number 105
             * Sequence numbers 100 through 104 get added to the vector
             * We receive sequence number 101 (this section is skipped)
             * We receive sequence number 103
             * Sequence number 102 is added to the vector
             *
             * This will prevent the duplicate from being added.
             */
            if (AST_VECTOR_GET_CMP(&rtp->missing_seqno, missing_seqno,
                        find_by_value)) {
                continue;
            }
 
            /* If this packet has been buffered already then don't count it amongst the
             * missing.
             */
            if (ast_data_buffer_get(rtp->recv_buffer, missing_seqno)) {
                continue;
            }
 
            ast_debug_rtp(2, "(%p) RTP added missing sequence number '%d'\n",
                instance, missing_seqno);
            AST_VECTOR_ADD_SORTED(&rtp->missing_seqno, missing_seqno,
                    compare_by_value);
            missing_seqnos_added++;
        }
 
        /* When we add a large number of missing sequence numbers we assume there was a substantial
         * gap in reception so we trigger an immediate NACK. When our data buffer is 1/4 full we
         * assume that the packets aren't just out of order but have actually been lost. At 1/2
         * full we get more aggressive and ask for retransmission when we get a new packet.
         * To get them back we construct and send a NACK causing the sender to retransmit them.
         */
        if (missing_seqnos_added >= MISSING_SEQNOS_ADDED_TRIGGER ||
            ast_data_buffer_count(rtp->recv_buffer) == ast_data_buffer_max(rtp->recv_buffer) / 4 ||
            ast_data_buffer_count(rtp->recv_buffer) >= ast_data_buffer_max(rtp->recv_buffer) / 2) {
            int packet_len = 0;
            int res = 0;
            int ice;
            int sr;
            size_t data_size = AST_UUID_STR_LEN + 128 + (AST_VECTOR_SIZE(&rtp->missing_seqno) * 4);
            RAII_VAR(unsigned char *, rtcpheader, NULL, ast_free_ptr);
            RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report,
                    ast_rtp_rtcp_report_alloc(rtp->themssrc_valid ? 1 : 0),
                    ao2_cleanup);
 
            /* Sufficient space for RTCP headers and report, SDES with CNAME, NACK header,
             * and worst case 4 bytes per missing sequence number.
             */
            rtcpheader = ast_malloc(sizeof(*rtcpheader) + data_size);
            if (!rtcpheader) {
                ast_debug_rtcp(1, "(%p) RTCP failed to allocate memory for NACK\n", instance);
                return &ast_null_frame;
            }
 
            memset(rtcpheader, 0, data_size);
 
            res = ast_rtcp_generate_compound_prefix(instance, rtcpheader, rtcp_report, &sr);
 
            if (res == 0 || res == 1) {
                return &ast_null_frame;
            }
 
            packet_len += res;
 
            res = ast_rtcp_generate_nack(instance, rtcpheader + packet_len);
 
            if (res == 0) {
                ast_debug_rtcp(1, "(%p) RTCP failed to construct NACK, stopping here\n", instance);
                return &ast_null_frame;
            }
 
            packet_len += res;
 
            res = rtcp_sendto(instance, rtcpheader, packet_len, 0, &remote_address, &ice);
            if (res < 0) {
                ast_debug_rtcp(1, "(%p) RTCP failed to send NACK request out\n", instance);
            } else {
                ast_debug_rtcp(2, "(%p) RTCP sending a NACK request to get missing packets\n", instance);
                /* Update RTCP SR/RR statistics */
                ast_rtcp_calculate_sr_rr_statistics(instance, rtcp_report, remote_address, ice, sr);
            }
        }
    }
 
    return &ast_null_frame;
}

References ao2_cleanup, ao2_lock, ast_assert, ast_codec_media_type2str(), AST_CONTROL_SRCCHANGE, ast_data_buffer_count(), ast_data_buffer_get(), ast_data_buffer_max(), ast_data_buffer_put(), ast_data_buffer_remove(), ast_data_buffer_resize(), ast_debug, ast_debug_rtcp, ast_debug_rtp, ast_debug_rtp_packet_is_allowed, ast_debug_stun, AST_FRAME_CONTROL, ast_frame_free(), ast_frdup, ast_free, ast_free_ptr(), AST_FRIENDLY_OFFSET, AST_LIST_FIRST, AST_LIST_HEAD_INIT_NOLOCK, AST_LIST_INSERT_TAIL, ast_log, ast_malloc, AST_MEDIA_TYPE_UNKNOWN, AST_MEDIA_TYPE_VIDEO, ast_null_frame, ast_rtcp_calculate_sr_rr_statistics(), ast_rtcp_generate_compound_prefix(), ast_rtcp_generate_nack(), ast_rtcp_interpret(), ast_rtcp_read(), ast_rtp_codecs_get_stream_type(), ast_rtp_instance_get_codecs(), ast_rtp_instance_get_data(), ast_rtp_instance_get_prop(), ast_rtp_instance_get_remote_address, ast_rtp_instance_get_requested_target_address(), ast_rtp_instance_get_srtp(), AST_RTP_INSTANCE_RTCP_STANDARD, ast_rtp_instance_set_incoming_source_address(), ast_rtp_instance_set_remote_address, ast_rtp_interpret(), AST_RTP_PROPERTY_NAT, ast_rtp_rtcp_report_alloc(), ast_set_flag, ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_from_sin, ast_sockaddr_ipv4_mapped(), ast_sockaddr_is_ipv4(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_sockaddr_to_sin, AST_STUN_ACCEPT, ast_stun_handle_packet(), ast_test_suite_event_notify, ast_tvdiff_ms(), ast_tvnow(), AST_UUID_STR_LEN, AST_VECTOR_ADD_SORTED, AST_VECTOR_ELEM_CLEANUP_NOOP, AST_VECTOR_GET_CMP, AST_VECTOR_REMOVE_CMP_ORDERED, AST_VECTOR_RESET, AST_VECTOR_SIZE, ast_verb, ast_verbose, ast_rtp_rtcp_nack_payload::buf, codecs, compare_by_value(), errno, ast_rtp::expectedrxseqno, ast_rtp::f, find_by_value(), FLAG_NAT_ACTIVE, frames, ast_frame::frametype, ast_frame_subclass::integer, ast_rtp::lastividtimestamp, ast_rtp::lastrxseqno, LOG_WARNING, MAXIMUM_RTP_RECV_BUFFER_SIZE, MIN, ast_rtp::missing_seqno, MISSING_SEQNOS_ADDED_TRIGGER, NULL, OLD_PACKET_COUNT, rtp_learning_info::packets, ast_rtp::prevrxseqno, rtp_learning_info::proposed_address, RAII_VAR, ast_rtp::rawdata, ast_rtp::recv_buffer, ast_rtp::rtcp, rtcp_mux(), rtcp_sendto(), rtp_debug_test_addr(), RTP_DTLS_ESTABLISHED, rtp_find_instance_by_packet_source_ssrc(), rtp_instance_unlock(), rtp_learning_rtp_seq_update(), rtp_learning_seq_init(), rtp_recvfrom(), ast_rtp::rtp_source_learn, rtp_write_rtcp_fir(), ast_rtp::s, ast_frame::seqno, SEQNO_CYCLE_OVER, ast_rtp_rtcp_nack_payload::size, ast_rtp::ssrc_mapping, rtp_learning_info::start, rtp_learning_info::stream_type, ast_rtp::strict_rtp_address, STRICT_RTP_CLOSED, STRICT_RTP_LEARN, STRICT_RTP_LEARN_TIMEOUT, STRICT_RTP_OPEN, ast_rtp::strict_rtp_state, ast_frame::subclass, ast_rtcp::them, ast_rtp::themssrc_valid, ast_rtcp::type, and version.

ast_rtp_remote_address_set()

static void ast_rtp_remote_address_set ( struct ast_rtp_instance *  instance, struct ast_sockaddr *  addr  )

static

Precondition

instance is locked

Definition at line 9279 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_sockaddr local;
    int index;
 
    ast_rtp_instance_get_local_address(instance, &local);
    if (!ast_sockaddr_isnull(addr)) {
        /* Update the local RTP address with what is being used */
        if (ast_ouraddrfor(addr, &local)) {
            /* Failed to update our address so reuse old local address */
            ast_rtp_instance_get_local_address(instance, &local);
        } else {
            ast_rtp_instance_set_local_address(instance, &local);
        }
    }
 
    if (rtp->rtcp && !ast_sockaddr_isnull(addr)) {
        ast_debug_rtcp(1, "(%p) RTCP setting address on RTP instance\n", instance);
        ast_sockaddr_copy(&rtp->rtcp->them, addr);
 
        if (rtp->rtcp->type == AST_RTP_INSTANCE_RTCP_STANDARD) {
            ast_sockaddr_set_port(&rtp->rtcp->them, ast_sockaddr_port(addr) + 1);
 
            /* Update the local RTCP address with what is being used */
            ast_sockaddr_set_port(&local, ast_sockaddr_port(&local) + 1);
        }
        ast_sockaddr_copy(&rtp->rtcp->us, &local);
 
        ast_free(rtp->rtcp->local_addr_str);
        rtp->rtcp->local_addr_str = ast_strdup(ast_sockaddr_stringify(&local));
    }
 
    /* Update any bundled RTP instances */
    for (index = 0; index < AST_VECTOR_SIZE(&rtp->ssrc_mapping); ++index) {
        struct rtp_ssrc_mapping *mapping = AST_VECTOR_GET_ADDR(&rtp->ssrc_mapping, index);
 
        ast_rtp_instance_set_remote_address(mapping->instance, addr);
    }
 
    /* Need to reset the DTMF last sequence number and the timestamp of the last END packet */
    rtp->last_seqno = 0;
    rtp->last_end_timestamp.ts = 0;
    rtp->last_end_timestamp.is_set = 0;
 
    if (strictrtp && rtp->strict_rtp_state != STRICT_RTP_OPEN
        && !ast_sockaddr_isnull(addr) && ast_sockaddr_cmp(addr, &rtp->strict_rtp_address)) {
        /* We only need to learn a new strict source address if we've been told the source is
         * changing to something different.
         */
        ast_verb(4, "%p -- Strict RTP learning after remote address set to: %s\n",
            rtp, ast_sockaddr_stringify(addr));
        rtp_learning_start(rtp);
    }
}

References ast_debug_rtcp, ast_free, ast_ouraddrfor(), ast_rtp_instance_get_data(), ast_rtp_instance_get_local_address(), AST_RTP_INSTANCE_RTCP_STANDARD, ast_rtp_instance_set_local_address(), ast_rtp_instance_set_remote_address, ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_strdup, AST_VECTOR_GET_ADDR, AST_VECTOR_SIZE, ast_verb, rtp_ssrc_mapping::instance, optional_ts::is_set, ast_rtp::last_end_timestamp, ast_rtp::last_seqno, ast_rtcp::local_addr_str, ast_rtp::rtcp, rtp_learning_start(), ast_rtp::ssrc_mapping, ast_rtp::strict_rtp_address, STRICT_RTP_OPEN, ast_rtp::strict_rtp_state, strictrtp, ast_rtcp::them, optional_ts::ts, ast_rtcp::type, and ast_rtcp::us.

ast_rtp_rtcp_handle_nack()

static int ast_rtp_rtcp_handle_nack ( struct ast_rtp_instance *  instance, unsigned int *  nackdata, unsigned int  position, unsigned int  length  )

static

Precondition

instance is locked

Definition at line 6689 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    int res = 0;
    int blp_index;
    int packet_index;
    int ice;
    struct ast_rtp_rtcp_nack_payload *payload;
    unsigned int current_word;
    unsigned int pid;   /* Packet ID which refers to seqno of lost packet */
    unsigned int blp;   /* Bitmask of following lost packets */
    struct ast_sockaddr remote_address = { {0,} };
    int abs_send_time_id;
    unsigned int now_msw = 0;
    unsigned int now_lsw = 0;
    unsigned int packets_not_found = 0;
 
    if (!rtp->send_buffer) {
        ast_debug_rtcp(1, "(%p) RTCP tried to handle NACK request, "
            "but we don't have a RTP packet storage!\n", instance);
        return res;
    }
 
    abs_send_time_id = ast_rtp_instance_extmap_get_id(instance, AST_RTP_EXTENSION_ABS_SEND_TIME);
    if (abs_send_time_id != -1) {
        timeval2ntp(ast_tvnow(), &now_msw, &now_lsw);
    }
 
    ast_rtp_instance_get_remote_address(instance, &remote_address);
 
    /*
     * We use index 3 because with feedback messages, the FCI (Feedback Control Information)
     * does not begin until after the version, packet SSRC, and media SSRC words.
     */
    for (packet_index = 3; packet_index < length; packet_index++) {
        current_word = ntohl(nackdata[position + packet_index]);
        pid = current_word >> 16;
        /* We know the remote end is missing this packet. Go ahead and send it if we still have it. */
        payload = (struct ast_rtp_rtcp_nack_payload *)ast_data_buffer_get(rtp->send_buffer, pid);
        if (payload) {
            if (abs_send_time_id != -1) {
                /* On retransmission we need to update the timestamp within the packet, as it
                 * is supposed to contain when the packet was actually sent.
                 */
                put_unaligned_time24(payload->buf + 17, now_msw, now_lsw);
            }
            res += rtp_sendto(instance, payload->buf, payload->size, 0, &remote_address, &ice);
        } else {
            ast_debug_rtcp(1, "(%p) RTCP received NACK request for RTP packet with seqno %d, "
                "but we don't have it\n", instance, pid);
            packets_not_found++;
        }
        /*
         * The bitmask. Denoting the least significant bit as 1 and its most significant bit
         * as 16, then bit i of the bitmask is set to 1 if the receiver has not received RTP
         * packet (pid+i)(modulo 2^16). Otherwise, it is set to 0. We cannot assume bits set
         * to 0 after a bit set to 1 have actually been received.
         */
        blp = current_word & 0xffff;
        blp_index = 1;
        while (blp) {
            if (blp & 1) {
                /* Packet (pid + i)(modulo 2^16) is missing too. */
                unsigned int seqno = (pid + blp_index) % 65536;
                payload = (struct ast_rtp_rtcp_nack_payload *)ast_data_buffer_get(rtp->send_buffer, seqno);
                if (payload) {
                    if (abs_send_time_id != -1) {
                        put_unaligned_time24(payload->buf + 17, now_msw, now_lsw);
                    }
                    res += rtp_sendto(instance, payload->buf, payload->size, 0, &remote_address, &ice);
                } else {
                    ast_debug_rtcp(1, "(%p) RTCP remote end also requested RTP packet with seqno %d, "
                        "but we don't have it\n", instance, seqno);
                    packets_not_found++;
                }
            }
            blp >>= 1;
            blp_index++;
        }
    }
 
    if (packets_not_found) {
        /* Grow the send buffer based on how many packets were not found in the buffer, but
         * enforce a maximum.
         */
        ast_data_buffer_resize(rtp->send_buffer, MIN(MAXIMUM_RTP_SEND_BUFFER_SIZE,
            ast_data_buffer_max(rtp->send_buffer) + packets_not_found));
        ast_debug_rtcp(2, "(%p) RTCP send buffer on RTP instance is now at maximum of %zu\n",
            instance, ast_data_buffer_max(rtp->send_buffer));
    }
 
    return res;
}

References ast_data_buffer_get(), ast_data_buffer_max(), ast_data_buffer_resize(), ast_debug_rtcp, AST_RTP_EXTENSION_ABS_SEND_TIME, ast_rtp_instance_extmap_get_id(), ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_tvnow(), ast_rtp_rtcp_nack_payload::buf, MAXIMUM_RTP_SEND_BUFFER_SIZE, MIN, put_unaligned_time24(), rtp_sendto(), ast_rtp::send_buffer, ast_rtp_rtcp_nack_payload::size, and timeval2ntp().

Referenced by ast_rtp_rtcp_handle_nack_locking().

ast_rtp_rtcp_handle_nack_locking()

static int ast_rtp_rtcp_handle_nack_locking ( struct ast_rtp_instance *  instance, struct ast_rtp_instance *  transport, unsigned int *  nackdata, unsigned int  position, unsigned int  length  )

static

Definition at line 6788 of file res_rtp_asterisk.c.

{
    int res;
 
    if (!transport || transport == instance) {
        return ast_rtp_rtcp_handle_nack(instance, nackdata, position, length);
    }
 
    ao2_ref(instance, +1);
    ao2_ref(transport, +1);
 
    /* Release child then parent; reacquire parent then child. */
    ao2_unlock(instance);
    ao2_unlock(transport);
    ao2_lock(instance);
 
    res = ast_rtp_rtcp_handle_nack(instance, nackdata, position, length);
 
    ao2_unlock(instance);
    ao2_lock(transport);
    ao2_lock(instance);
 
    ao2_ref(transport, -1);
    ao2_ref(instance, -1);
 
    return res;
}

References ao2_lock, ao2_ref, ao2_unlock, and ast_rtp_rtcp_handle_nack().

Referenced by ast_rtcp_interpret().

ast_rtp_sendcng()

static int ast_rtp_sendcng ( struct ast_rtp_instance *  instance, int  level  )

static

generate comfort noice (CNG)

Precondition

instance is locked

Definition at line 9645 of file res_rtp_asterisk.c.

{
    unsigned int *rtpheader;
    int hdrlen = 12;
    int res, payload = 0;
    char data[256];
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_sockaddr remote_address = { {0,} };
    int ice;
 
    ast_rtp_instance_get_remote_address(instance, &remote_address);
 
    if (ast_sockaddr_isnull(&remote_address)) {
        return -1;
    }
 
    payload = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_CN);
 
    level = 127 - (level & 0x7f);
 
    rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
 
    /* Get a pointer to the header */
    rtpheader = (unsigned int *)data;
    rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
    rtpheader[1] = htonl(rtp->lastts);
    rtpheader[2] = htonl(rtp->ssrc);
    data[12] = level;
 
    res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 1, 0, &remote_address, &ice);
 
    if (res < 0) {
        ast_log(LOG_ERROR, "RTP Comfort Noise Transmission error to %s: %s\n", ast_sockaddr_stringify(&remote_address), strerror(errno));
        return res;
    }
 
    if (rtp_debug_test_addr(&remote_address)) {
        ast_verbose("Sent Comfort Noise RTP packet to %s%s (type %-2.2d, seq %-6.6d, ts %-6.6u, len %-6.6d)\n",
                ast_sockaddr_stringify(&remote_address),
                ice ? " (via ICE)" : "",
                AST_RTP_CN, rtp->seqno, rtp->lastdigitts, res - hdrlen);
    }
 
    rtp->seqno++;
 
    return res;
}

References ast_log, AST_RTP_CN, ast_rtp_codecs_payload_code_tx(), ast_rtp_instance_get_codecs(), ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_tv(), ast_tvadd(), ast_tvnow(), ast_verbose, ast_rtp::dtmfmute, errno, ast_rtp::lastdigitts, ast_rtp::lastts, LOG_ERROR, NULL, rtp_debug_test_addr(), rtp_sendto(), ast_rtp::seqno, and ast_rtp::ssrc.

ast_rtp_set_remote_ssrc()

static void ast_rtp_set_remote_ssrc ( struct ast_rtp_instance *  instance, unsigned int  ssrc  )

static

Precondition

instance is locked

Definition at line 9710 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    if (rtp->themssrc_valid && rtp->themssrc == ssrc) {
        return;
    }
 
    rtp->themssrc = ssrc;
    rtp->themssrc_valid = 1;
 
    /* If this is bundled we need to update the SSRC mapping */
    if (rtp->bundled) {
        struct ast_rtp *bundled_rtp;
        int index;
 
        ao2_unlock(instance);
 
        /* The child lock can't be held while accessing the parent */
        ao2_lock(rtp->bundled);
        bundled_rtp = ast_rtp_instance_get_data(rtp->bundled);
 
        for (index = 0; index < AST_VECTOR_SIZE(&bundled_rtp->ssrc_mapping); ++index) {
            struct rtp_ssrc_mapping *mapping = AST_VECTOR_GET_ADDR(&bundled_rtp->ssrc_mapping, index);
 
            if (mapping->instance == instance) {
                mapping->ssrc = ssrc;
                mapping->ssrc_valid = 1;
                break;
            }
        }
 
        ao2_unlock(rtp->bundled);
 
        ao2_lock(instance);
    }
}

References ao2_lock, ao2_unlock, ast_rtp_instance_get_data(), AST_VECTOR_GET_ADDR, AST_VECTOR_SIZE, ast_rtp::bundled, rtp_ssrc_mapping::instance, rtp_ssrc_mapping::ssrc, ast_rtp::ssrc, ast_rtp::ssrc_mapping, rtp_ssrc_mapping::ssrc_valid, ast_rtp::themssrc, and ast_rtp::themssrc_valid.

ast_rtp_set_stream_num()

static void ast_rtp_set_stream_num ( struct ast_rtp_instance *  instance, int  stream_num  )

static

Definition at line 9748 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    rtp->stream_num = stream_num;
}

References ast_rtp_instance_get_data(), and ast_rtp::stream_num.

ast_rtp_stop()

static void ast_rtp_stop ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 9582 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_sockaddr addr = { {0,} };
 
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
    ao2_unlock(instance);
    AST_SCHED_DEL_UNREF(rtp->sched, rtp->rekeyid, ao2_ref(instance, -1));
 
    dtls_srtp_stop_timeout_timer(instance, rtp, 0);
    if (rtp->rtcp) {
        dtls_srtp_stop_timeout_timer(instance, rtp, 1);
    }
    ao2_lock(instance);
#endif
    ast_debug_rtp(1, "(%s) RTP Stop\n",
        ast_rtp_instance_get_channel_id(instance));
 
    if (rtp->rtcp && rtp->rtcp->schedid > -1) {
        ao2_unlock(instance);
        if (!ast_sched_del(rtp->sched, rtp->rtcp->schedid)) {
            /* successfully cancelled scheduler entry. */
            ao2_ref(instance, -1);
        }
        ao2_lock(instance);
        rtp->rtcp->schedid = -1;
    }
 
    if (rtp->transport_wide_cc.schedid > -1) {
        ao2_unlock(instance);
        if (!ast_sched_del(rtp->sched, rtp->transport_wide_cc.schedid)) {
            ao2_ref(instance, -1);
        }
        ao2_lock(instance);
        rtp->transport_wide_cc.schedid = -1;
        }
 
    if (rtp->red) {
        ao2_unlock(instance);
        AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
        ao2_lock(instance);
        ast_free(rtp->red);
        rtp->red = NULL;
    }
 
    ast_rtp_instance_set_remote_address(instance, &addr);
 
    ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
}

References ao2_lock, ao2_ref, ao2_unlock, ast_debug_rtp, ast_free, ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_data(), ast_rtp_instance_set_remote_address, AST_SCHED_DEL, ast_sched_del(), AST_SCHED_DEL_UNREF, ast_set_flag, FLAG_NEED_MARKER_BIT, NULL, ast_rtp::red, ast_rtp::rtcp, ast_rtp::sched, rtp_transport_wide_cc_statistics::schedid, ast_rtcp::schedid, rtp_red::schedid, and ast_rtp::transport_wide_cc.

ast_rtp_stun_request()

static void ast_rtp_stun_request ( struct ast_rtp_instance *  instance, struct ast_sockaddr *  suggestion, const char *  username  )

static

Precondition

instance is NOT locked

Definition at line 9567 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct sockaddr_in suggestion_tmp;
 
    /*
     * The instance should not be locked because we can block
     * waiting for a STUN respone.
     */
    ast_sockaddr_to_sin(suggestion, &suggestion_tmp);
    ast_stun_request(rtp->s, &suggestion_tmp, username, NULL);
    ast_sockaddr_from_sin(suggestion, &suggestion_tmp);
}

References ast_rtp_instance_get_data(), ast_sockaddr_from_sin, ast_sockaddr_to_sin, ast_stun_request(), NULL, and ast_rtp::s.

ast_rtp_update_source()

static void ast_rtp_update_source ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 4733 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    /* We simply set this bit so that the next packet sent will have the marker bit turned on */
    ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
    ast_debug_rtp(3, "(%p) RTP setting the marker bit due to a source update\n", instance);
 
    return;
}

References ast_debug_rtp, ast_rtp_instance_get_data(), ast_set_flag, and FLAG_NEED_MARKER_BIT.

ast_rtp_write()

static int ast_rtp_write ( struct ast_rtp_instance *  instance, struct ast_frame *  frame  )

static

Precondition

instance is locked

Definition at line 5694 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_sockaddr remote_address = { {0,} };
    struct ast_format *format;
    int codec;
 
    ast_rtp_instance_get_remote_address(instance, &remote_address);
 
    /* If we don't actually know the remote address don't even bother doing anything */
    if (ast_sockaddr_isnull(&remote_address)) {
        ast_debug_rtp(1, "(%p) RTP no remote address on instance, so dropping frame\n", instance);
        return 0;
    }
 
    /* VP8: is this a request to send a RTCP FIR? */
    if (frame->frametype == AST_FRAME_CONTROL && frame->subclass.integer == AST_CONTROL_VIDUPDATE) {
        rtp_write_rtcp_fir(instance, rtp, &remote_address);
        return 0;
    } else if (frame->frametype == AST_FRAME_RTCP) {
        if (frame->subclass.integer == AST_RTP_RTCP_PSFB) {
            rtp_write_rtcp_psfb(instance, rtp, frame, &remote_address);
        }
        return 0;
    }
 
    /* If there is no data length we can't very well send the packet */
    if (!frame->datalen) {
        ast_debug_rtp(1, "(%p) RTP received frame with no data for instance, so dropping frame\n", instance);
        return 0;
    }
 
    /* If the packet is not one our RTP stack supports bail out */
    if (frame->frametype != AST_FRAME_VOICE && frame->frametype != AST_FRAME_VIDEO && frame->frametype != AST_FRAME_TEXT) {
        ast_log(LOG_WARNING, "RTP can only send voice, video, and text\n");
        return -1;
    }
 
    if (rtp->red) {
        /* return 0; */
        /* no primary data or generations to send */
        if ((frame = red_t140_to_red(rtp->red)) == NULL)
            return 0;
    }
 
    /* Grab the subclass and look up the payload we are going to use */
    codec = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance),
        1, frame->subclass.format, 0);
    if (codec < 0) {
        ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n",
            ast_format_get_name(frame->subclass.format));
        return -1;
    }
 
    /* Note that we do not increase the ref count here as this pointer
     * will not be held by any thing explicitly. The format variable is
     * merely a convenience reference to frame->subclass.format */
    format = frame->subclass.format;
    if (ast_format_cmp(rtp->lasttxformat, format) == AST_FORMAT_CMP_NOT_EQUAL) {
        /* Oh dear, if the format changed we will have to set up a new smoother */
        ast_debug_rtp(3, "(%s) RTP ooh, format changed from %s to %s\n",
            ast_rtp_instance_get_channel_id(instance),
            ast_format_get_name(rtp->lasttxformat),
            ast_format_get_name(frame->subclass.format));
        ao2_replace(rtp->lasttxformat, format);
        if (rtp->smoother) {
            ast_smoother_free(rtp->smoother);
            rtp->smoother = NULL;
        }
    }
 
    /* If no smoother is present see if we have to set one up */
    if (!rtp->smoother && ast_format_can_be_smoothed(format)) {
        unsigned int smoother_flags = ast_format_get_smoother_flags(format);
        unsigned int framing_ms = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance));
 
        if (!framing_ms && (smoother_flags & AST_SMOOTHER_FLAG_FORCED)) {
            framing_ms = ast_format_get_default_ms(format);
        }
 
        if (framing_ms) {
            rtp->smoother = ast_smoother_new((framing_ms * ast_format_get_minimum_bytes(format)) / ast_format_get_minimum_ms(format));
            if (!rtp->smoother) {
                ast_log(LOG_WARNING, "Unable to create smoother: format %s ms: %u len: %u\n",
                    ast_format_get_name(format), framing_ms, ast_format_get_minimum_bytes(format));
                return -1;
            }
            ast_smoother_set_flags(rtp->smoother, smoother_flags);
        }
    }
 
    /* Feed audio frames into the actual function that will create a frame and send it */
    if (rtp->smoother) {
        struct ast_frame *f;
 
        if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
            ast_smoother_feed_be(rtp->smoother, frame);
        } else {
            ast_smoother_feed(rtp->smoother, frame);
        }
 
        while ((f = ast_smoother_read(rtp->smoother)) && (f->data.ptr)) {
                rtp_raw_write(instance, f, codec);
        }
    } else {
        int hdrlen = 12;
        struct ast_frame *f = NULL;
 
        if (frame->offset < hdrlen) {
            f = ast_frdup(frame);
        } else {
            f = frame;
        }
        if (f->data.ptr) {
            rtp_raw_write(instance, f, codec);
        }
        if (f != frame) {
            ast_frfree(f);
        }
 
    }
 
    return 0;
}

References ao2_replace, AST_CONTROL_VIDUPDATE, ast_debug_rtp, ast_format_can_be_smoothed(), ast_format_cmp(), AST_FORMAT_CMP_NOT_EQUAL, ast_format_get_default_ms(), ast_format_get_minimum_bytes(), ast_format_get_minimum_ms(), ast_format_get_name(), ast_format_get_smoother_flags(), AST_FRAME_CONTROL, AST_FRAME_RTCP, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_frdup, ast_frfree, ast_log, ast_rtp_codecs_get_framing(), ast_rtp_codecs_payload_code_tx(), ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_codecs(), ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, AST_RTP_RTCP_PSFB, ast_smoother_feed, ast_smoother_feed_be, AST_SMOOTHER_FLAG_BE, AST_SMOOTHER_FLAG_FORCED, ast_smoother_free(), ast_smoother_new(), ast_smoother_read(), ast_smoother_set_flags(), ast_smoother_test_flag(), ast_sockaddr_isnull(), ast_format::codec, ast_frame::data, ast_frame::datalen, ast_frame_subclass::format, ast_frame::frametype, ast_frame_subclass::integer, ast_rtp::lasttxformat, LOG_WARNING, NULL, ast_frame::offset, ast_frame::ptr, ast_rtp::red, red_t140_to_red(), rtp_raw_write(), rtp_write_rtcp_fir(), rtp_write_rtcp_psfb(), ast_rtp::smoother, and ast_frame::subclass.

Referenced by red_write(), and rtp_red_buffer().

bridge_p2p_rtp_write()

static int bridge_p2p_rtp_write ( struct ast_rtp_instance *  instance, struct ast_rtp_instance *  instance1, unsigned int *  rtpheader, int  len, int  hdrlen  )

static

Precondition

instance is locked

Definition at line 7430 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_rtp *bridged;
    int res = 0, payload = 0, bridged_payload = 0, mark;
    RAII_VAR(struct ast_rtp_payload_type *, payload_type, NULL, ao2_cleanup);
    int reconstruct = ntohl(rtpheader[0]);
    struct ast_sockaddr remote_address = { {0,} };
    int ice;
    unsigned int timestamp = ntohl(rtpheader[1]);
 
    /* Get fields from packet */
    payload = (reconstruct & 0x7f0000) >> 16;
    mark = (reconstruct & 0x800000) >> 23;
 
    /* Check what the payload value should be */
    payload_type = ast_rtp_codecs_get_payload(ast_rtp_instance_get_codecs(instance), payload);
    if (!payload_type) {
        return -1;
    }
 
    /* Otherwise adjust bridged payload to match */
    bridged_payload = ast_rtp_codecs_payload_code_tx_sample_rate(ast_rtp_instance_get_codecs(instance1),
        payload_type->asterisk_format, payload_type->format, payload_type->rtp_code, payload_type->sample_rate);
 
    /* If no codec could be matched between instance and instance1, then somehow things were made incompatible while we were still bridged.  Bail. */
    if (bridged_payload < 0) {
        return -1;
    }
 
    /* If the payload coming in is not one of the negotiated ones then send it to the core, this will cause formats to change and the bridge to break */
    if (ast_rtp_codecs_find_payload_code(ast_rtp_instance_get_codecs(instance1), bridged_payload) == -1) {
        ast_debug_rtp(1, "(%p, %p) RTP unsupported payload type received\n", instance, instance1);
        return -1;
    }
 
    /*
     * Even if we are no longer in dtmf, we could still be receiving
     * re-transmissions of the last dtmf end still.  Feed those to the
     * core so they can be filtered accordingly.
     */
    if (rtp->last_end_timestamp.is_set && rtp->last_end_timestamp.ts == timestamp) {
        ast_debug_rtp(1, "(%p, %p) RTP feeding packet with duplicate timestamp to core\n", instance, instance1);
        return -1;
    }
 
    if (payload_type->asterisk_format) {
        ao2_replace(rtp->lastrxformat, payload_type->format);
    }
 
    /*
     * We have now determined that we need to send the RTP packet
     * out the bridged instance to do local bridging so we must unlock
     * the receiving instance to prevent deadlock with the bridged
     * instance.
     *
     * Technically we should grab a ref to instance1 so it won't go
     * away on us.  However, we should be safe because the bridged
     * instance won't change without both channels involved being
     * locked and we currently have the channel lock for the receiving
     * instance.
     */
    ao2_unlock(instance);
    ao2_lock(instance1);
 
    /*
     * Get the peer rtp pointer now to emphasize that using it
     * must happen while instance1 is locked.
     */
    bridged = ast_rtp_instance_get_data(instance1);
 
 
    /* If bridged peer is in dtmf, feed all packets to core until it finishes to avoid infinite dtmf */
    if (bridged->sending_digit) {
        ast_debug_rtp(1, "(%p, %p) RTP Feeding packet to core until DTMF finishes\n", instance, instance1);
        ao2_unlock(instance1);
        ao2_lock(instance);
        return -1;
    }
 
    if (payload_type->asterisk_format) {
        /*
         * If bridged peer has already received rtp, perform the asymmetric codec check
         * if that feature has been activated
         */
        if (!bridged->asymmetric_codec
            && bridged->lastrxformat != ast_format_none
            && ast_format_cmp(payload_type->format, bridged->lastrxformat) == AST_FORMAT_CMP_NOT_EQUAL) {
            ast_debug_rtp(1, "(%p, %p) RTP asymmetric RTP codecs detected (TX: %s, RX: %s) sending frame to core\n",
                instance, instance1, ast_format_get_name(payload_type->format),
                ast_format_get_name(bridged->lastrxformat));
            ao2_unlock(instance1);
            ao2_lock(instance);
            return -1;
        }
 
        ao2_replace(bridged->lasttxformat, payload_type->format);
    }
 
    ast_rtp_instance_get_remote_address(instance1, &remote_address);
 
    if (ast_sockaddr_isnull(&remote_address)) {
        ast_debug_rtp(5, "(%p, %p) RTP remote address is null, most likely RTP has been stopped\n",
            instance, instance1);
        ao2_unlock(instance1);
        ao2_lock(instance);
        return 0;
    }
 
    /* If the marker bit has been explicitly set turn it on */
    if (ast_test_flag(bridged, FLAG_NEED_MARKER_BIT)) {
        mark = 1;
        ast_clear_flag(bridged, FLAG_NEED_MARKER_BIT);
    }
 
    /* Set the marker bit for the first local bridged packet which has the first bridged peer's SSRC. */
    if (ast_test_flag(bridged, FLAG_REQ_LOCAL_BRIDGE_BIT)) {
        mark = 1;
        ast_clear_flag(bridged, FLAG_REQ_LOCAL_BRIDGE_BIT);
    }
 
    /* Reconstruct part of the packet */
    reconstruct &= 0xFF80FFFF;
    reconstruct |= (bridged_payload << 16);
    reconstruct |= (mark << 23);
    rtpheader[0] = htonl(reconstruct);
 
    if (mark) {
        /* make this rtp instance aware of the new ssrc it is sending */
        bridged->ssrc = ntohl(rtpheader[2]);
    }
 
    /* Send the packet back out */
    res = rtp_sendto(instance1, (void *)rtpheader, len, 0, &remote_address, &ice);
    if (res < 0) {
        if (!ast_rtp_instance_get_prop(instance1, AST_RTP_PROPERTY_NAT) || (ast_rtp_instance_get_prop(instance1, AST_RTP_PROPERTY_NAT) && (ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
            ast_log(LOG_WARNING,
                "RTP Transmission error of packet to %s: %s\n",
                ast_sockaddr_stringify(&remote_address),
                strerror(errno));
        } else if (((ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || ast_debug_rtp_packet_is_allowed) && !ast_test_flag(bridged, FLAG_NAT_INACTIVE_NOWARN)) {
            if (ast_debug_rtp_packet_is_allowed || DEBUG_ATLEAST(1)) {
                ast_log(LOG_WARNING,
                    "RTP NAT: Can't write RTP to private "
                    "address %s, waiting for other end to "
                    "send audio...\n",
                    ast_sockaddr_stringify(&remote_address));
            }
            ast_set_flag(bridged, FLAG_NAT_INACTIVE_NOWARN);
        }
        ao2_unlock(instance1);
        ao2_lock(instance);
        return 0;
    }
 
    if (rtp_debug_test_addr(&remote_address)) {
        ast_verbose("Sent RTP P2P packet to %s%s (type %-2.2d, len %-6.6d)\n",
                ast_sockaddr_stringify(&remote_address),
                ice ? " (via ICE)" : "",
                bridged_payload, len - hdrlen);
    }
 
    ao2_unlock(instance1);
    ao2_lock(instance);
    return 0;
}

References ao2_cleanup, ao2_lock, ao2_replace, ao2_unlock, ast_clear_flag, ast_debug_rtp, ast_debug_rtp_packet_is_allowed, ast_format_cmp(), AST_FORMAT_CMP_NOT_EQUAL, ast_format_get_name(), ast_format_none, ast_log, ast_rtp_codecs_find_payload_code(), ast_rtp_codecs_get_payload(), ast_rtp_codecs_payload_code_tx_sample_rate(), ast_rtp_instance_get_codecs(), ast_rtp_instance_get_data(), ast_rtp_instance_get_prop(), ast_rtp_instance_get_remote_address, AST_RTP_PROPERTY_NAT, ast_set_flag, ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_test_flag, ast_verbose, ast_rtp::asymmetric_codec, DEBUG_ATLEAST, errno, FLAG_NAT_ACTIVE, FLAG_NAT_INACTIVE, FLAG_NAT_INACTIVE_NOWARN, FLAG_NEED_MARKER_BIT, FLAG_REQ_LOCAL_BRIDGE_BIT, optional_ts::is_set, ast_rtp::last_end_timestamp, ast_rtp::lastrxformat, ast_rtp::lasttxformat, len(), LOG_WARNING, NULL, RAII_VAR, reconstruct(), rtp_debug_test_addr(), rtp_sendto(), ast_rtp::sending_digit, ast_rtp::ssrc, and optional_ts::ts.

Referenced by ast_rtp_interpret().

calc_mean_and_standard_deviation()

static void calc_mean_and_standard_deviation ( double  new_sample, double *  mean, double *  std_dev, unsigned int *  count  )

static

Definition at line 3568 of file res_rtp_asterisk.c.

{
    double delta1;
    double delta2;
 
    /* First convert the standard deviation back into a sum of squares. */
    double last_sum_of_squares = (*std_dev) * (*std_dev) * (*count ?: 1);
 
    if (++(*count) == 0) {
        /* Avoid potential divide by zero on an overflow */
        *count = 1;
    }
 
    /*
     * Below is an implementation of Welford's online algorithm [1] for calculating
     * mean and variance in a single pass.
     *
     * [1] https://en.wikipedia.org/wiki/Algorithms_for_calculating_variance
     */
 
    delta1 = new_sample - *mean;
    *mean += (delta1 / *count);
    delta2 = new_sample - *mean;
 
    /* Now calculate the new variance, and subsequent standard deviation */
    *std_dev = sqrt((last_sum_of_squares + (delta1 * delta2)) / *count);
}

Referenced by calc_rxstamp_and_jitter(), calculate_lost_packet_statistics(), update_jitter_stats(), update_local_mes_stats(), update_lost_stats(), update_reported_mes_stats(), and update_rtt_stats().

calc_media_experience_score()

static double calc_media_experience_score ( struct ast_rtp_instance *  instance, double  normdevrtt, double  normdev_rxjitter, double  stdev_rxjitter, double  normdev_rxlost  )

static

Calculate a "media experience score" based on given data.

Technically, a mean opinion score (MOS) cannot be calculated without the involvement of human eyes (video) and ears (audio). Thus instead we'll approximate an opinion using the given parameters, and call it a media experience score.

The tallied score is based upon recommendations and formulas from ITU-T G.107, ITU-T G.109, ITU-T G.113, and other various internet sources.

Parameters

instance	RTP instance
normdevrtt	The average round trip time
normdev_rxjitter	The smoothed jitter
stdev_rxjitter	The jitter standard deviation value
normdev_rxlost	The average number of packets lost since last check

Returns

A media experience score.

Note

The calculations in this function could probably be simplified but calculating a MOS using the information available publicly, then re-scaling it to 0.0 -> 100.0 makes the process clearer and easier to troubleshoot or change.

Definition at line 6458 of file res_rtp_asterisk.c.

{
    double r_value;
    double pseudo_mos;
    double mes = 0;
 
    /*
     * While the media itself might be okay, a significant enough delay could make
     * for an unpleasant user experience.
     *
     * Calculate the effective latency by using the given round trip time, and adding
     * jitter scaled according to its standard deviation. The scaling is done in order
     * to increase jitter's weight since a higher deviation can result in poorer overall
     * quality.
     *
     * normdevrtt is the mean round trip time in seconds. The G.107's delay-impairment
     * model is based on one-way so we need to cut it in half before converting to
     * milliseconds.
     *
     * normdev_rxjitter and stdev_rxjitter are also in seconds and are converted to
     * milliseconds to match.
     */
    double effective_latency = ((normdevrtt / 2) * 1000)
        + ((normdev_rxjitter * 1000 * 2) * (stdev_rxjitter * 1000 / 3))
        + 10;
 
    /*
     * Using the defaults for the standard transmission rating factor ("R" value)
     * one arrives at 93.2 (see ITU-T G.107 for more details), so we'll use that
     * as the starting value and subtract deficiencies that could affect quality.
     *
     * Calculate the impact of the effective latency. Influence increases with
     * values over 160 as the significant "lag" can degrade user experience.
     */
    if (effective_latency < 160) {
        r_value = 93.2 - (effective_latency / 40);
    } else {
        r_value = 93.2 - (effective_latency - 120) / 10;
    }
 
    /* Next evaluate the impact of lost packets */
    r_value = r_value - (normdev_rxlost * 2.0);
 
    /*
     * Finally convert the "R" value into a opinion/quality score between 1 (really anything
     * below 3 should be considered poor) and 4.5 (the highest achievable for VOIP).
     */
    if (r_value < 0) {
        pseudo_mos = 1.0;
    } else if (r_value > 100) {
        pseudo_mos = 4.5;
    } else {
        pseudo_mos = 1 + (0.035 * r_value) + (r_value * (r_value - 60) * (100 - r_value) * 0.000007);
    }
 
    /*
     * We're going to rescale the 0.0->5.0 pseudo_mos to the 0.0->100.0 MES.
     * For those ranges, we could actually just multiply the pseudo_mos
     * by 20 but we may want to change the scale later.
     */
    mes = RESCALE(pseudo_mos, 0.0, 5.0, 0.0, 100.0);
 
    return mes;
}

References RESCALE.

Referenced by update_local_mes_stats(), and update_reported_mes_stats().

calc_rxstamp_and_jitter()

static void calc_rxstamp_and_jitter ( struct timeval *  tv, struct ast_rtp *  rtp, unsigned int  rx_rtp_ts, int  mark  )

static

Definition at line 5819 of file res_rtp_asterisk.c.

{
    int rate = ast_rtp_get_rate(rtp->f.subclass.format);
 
    double jitter = 0.0;
    double prev_jitter = 0.0;
    struct timeval now;
    struct timeval tmp;
    double rxnow;
    double arrival_sec;
    unsigned int arrival;
    int transit;
    int d;
 
    gettimeofday(&now,NULL);
 
    if (rtp->rxcount == 1 || mark) {
        rtp->rxstart = ast_tv2double(&now);
        rtp->remote_seed_rx_rtp_ts = rx_rtp_ts;
 
        /*
         * "tv" is placed in the received frame's
         * "delivered" field and when this frame is
         * sent out again on the other side, it's
         * used to calculate the timestamp on the
         * outgoing RTP packets.
         *
         * NOTE: We need to do integer math here
         * because double math rounding issues can
         * generate incorrect timestamps.
         */
        rtp->rxcore = now;
        tmp = ast_samp2tv(rx_rtp_ts, rate);
        rtp->rxcore = ast_tvsub(rtp->rxcore, tmp);
        rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 100;
        *tv = ast_tvadd(rtp->rxcore, tmp);
 
        ast_debug_rtcp(3, "%s: "
            "Seed ts: %u current time: %f\n",
            ast_rtp_instance_get_channel_id(rtp->owner)
            , rx_rtp_ts
            , rtp->rxstart
        );
 
        return;
    }
 
    tmp = ast_samp2tv(rx_rtp_ts, rate);
    /* See the comment about "tv" above. Even if
     * we don't use this received packet for jitter
     * calculations, we still need to set tv so the
     * timestamp will be correct when this packet is
     * sent out again.
     */
    *tv = ast_tvadd(rtp->rxcore, tmp);
 
    /*
     * The first few packets are generally unstable so let's
     * not use them in the calculations.
     */
    if (rtp->rxcount < RTP_IGNORE_FIRST_PACKETS_COUNT) {
        ast_debug_rtcp(3, "%s: Packet %d < %d.  Ignoring\n",
            ast_rtp_instance_get_channel_id(rtp->owner)
            , rtp->rxcount
            , RTP_IGNORE_FIRST_PACKETS_COUNT
        );
 
        return;
    }
 
    /*
     * First good packet. Capture the start time and timestamp
     * but don't actually use this packet for calculation.
     */
    if (rtp->rxcount == RTP_IGNORE_FIRST_PACKETS_COUNT) {
        rtp->rxstart_stable = ast_tv2double(&now);
        rtp->remote_seed_rx_rtp_ts_stable = rx_rtp_ts;
        rtp->last_transit_time_samples = -rx_rtp_ts;
 
        ast_debug_rtcp(3, "%s: "
            "pkt: %5u Stable Seed ts: %u current time: %f\n",
            ast_rtp_instance_get_channel_id(rtp->owner)
            , rtp->rxcount
            , rx_rtp_ts
            , rtp->rxstart_stable
        );
 
        return;
    }
 
    /*
     * If the current packet isn't in sequence, don't
     * use it in any calculations as remote_current_rx_rtp_ts
     * is not going to be correct.
     */
    if (rtp->lastrxseqno != rtp->prevrxseqno + 1) {
        ast_debug_rtcp(3, "%s: Current packet seq %d != last packet seq %d + 1.  Ignoring\n",
            ast_rtp_instance_get_channel_id(rtp->owner)
            , rtp->lastrxseqno
            , rtp->prevrxseqno
        );
 
        return;
    }
 
    /*
     * The following calculations are taken from
     * https://www.rfc-editor.org/rfc/rfc3550#appendix-A.8
     *
     * The received rtp timestamp is the random "seed"
     * timestamp chosen by the sender when they sent the
     * first packet, plus the number of samples since then.
     *
     * To get our arrival time in the same units, we
     * calculate the time difference in seconds between
     * when we received the first packet and when we
     * received this packet and convert that to samples.
     */
    rxnow = ast_tv2double(&now);
    arrival_sec = rxnow - rtp->rxstart_stable;
    arrival = ast_sec2samp(arrival_sec, rate);
 
    /*
     * Now we can use the exact formula in
     * https://www.rfc-editor.org/rfc/rfc3550#appendix-A.8 :
     *
     * int transit = arrival - r->ts;
     * int d = transit - s->transit;
     * s->transit = transit;
     * if (d < 0) d = -d;
     * s->jitter += (1./16.) * ((double)d - s->jitter);
     *
     * Our rx_rtp_ts is their r->ts.
     * Our rtp->last_transit_time_samples is their s->transit.
     * Our rtp->rxjitter is their s->jitter.
     */
    transit = arrival - rx_rtp_ts;
    d = transit - rtp->last_transit_time_samples;
 
    if (d < 0) {
        d = -d;
    }
 
    prev_jitter = rtp->rxjitter_samples;
    jitter = (1.0/16.0) * (((double)d) - prev_jitter);
    rtp->rxjitter_samples = prev_jitter + jitter;
 
    /*
     * We need to hang on to jitter in both samples and seconds.
     */
    rtp->rxjitter = ast_samp2sec(rtp->rxjitter_samples, rate);
 
    ast_debug_rtcp(3, "%s: pkt: %5u "
        "Arrival sec: %7.3f  Arrival ts: %10u  RX ts: %10u "
        "Transit samp: %6d Last transit samp: %6d d: %4d "
        "Curr jitter: %7.0f(%7.3f) Prev Jitter: %7.0f(%7.3f) New Jitter: %7.0f(%7.3f)\n",
        ast_rtp_instance_get_channel_id(rtp->owner)
        , rtp->rxcount
        , arrival_sec
        , arrival
        , rx_rtp_ts
        , transit
        , rtp->last_transit_time_samples
        , d
        , jitter
        , ast_samp2sec(jitter, rate)
        , prev_jitter
        , ast_samp2sec(prev_jitter, rate)
        , rtp->rxjitter_samples
        , rtp->rxjitter
        );
 
    rtp->last_transit_time_samples = transit;
 
    /*
     * Update all the stats.
     */
    if (rtp->rtcp) {
        if (rtp->rxjitter > rtp->rtcp->maxrxjitter)
            rtp->rtcp->maxrxjitter = rtp->rxjitter;
        if (rtp->rtcp->rxjitter_count == 1)
            rtp->rtcp->minrxjitter = rtp->rxjitter;
        if (rtp->rtcp && rtp->rxjitter < rtp->rtcp->minrxjitter)
            rtp->rtcp->minrxjitter = rtp->rxjitter;
 
        calc_mean_and_standard_deviation(rtp->rxjitter,
            &rtp->rtcp->normdev_rxjitter, &rtp->rtcp->stdev_rxjitter,
            &rtp->rtcp->rxjitter_count);
    }
 
    return;
}

References ast_debug_rtcp, ast_rtp_get_rate(), ast_rtp_instance_get_channel_id(), ast_samp2sec(), ast_samp2tv(), ast_sec2samp(), ast_tv2double(), ast_tvadd(), ast_tvsub(), calc_mean_and_standard_deviation(), d, ast_rtp::f, ast_frame_subclass::format, ast_rtp::last_transit_time_samples, ast_rtp::lastrxseqno, ast_rtcp::maxrxjitter, ast_rtcp::minrxjitter, ast_rtcp::normdev_rxjitter, NULL, ast_rtp::owner, ast_rtp::prevrxseqno, ast_rtp::remote_seed_rx_rtp_ts, ast_rtp::remote_seed_rx_rtp_ts_stable, ast_rtp::rtcp, RTP_IGNORE_FIRST_PACKETS_COUNT, ast_rtp::rxcore, ast_rtp::rxcount, ast_rtp::rxjitter, ast_rtcp::rxjitter_count, ast_rtp::rxjitter_samples, ast_rtp::rxstart, ast_rtp::rxstart_stable, ast_rtcp::stdev_rxjitter, and ast_frame::subclass.

Referenced by ast_rtp_interpret().

calc_txstamp()

static unsigned int calc_txstamp ( struct ast_rtp *  rtp, struct timeval *  delivery  )

static

Definition at line 4038 of file res_rtp_asterisk.c.

{
    struct timeval t;
    long ms;
 
    if (ast_tvzero(rtp->txcore)) {
        rtp->txcore = ast_tvnow();
        rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
    }
 
    t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
    if ((ms = ast_tvdiff_ms(t, rtp->txcore)) < 0) {
        ms = 0;
    }
    rtp->txcore = t;
 
    return (unsigned int) ms;
}

References ast_tvdiff_ms(), ast_tvnow(), ast_tvzero(), and ast_rtp::txcore.

Referenced by ast_rtp_dtmf_begin(), ast_rtp_dtmf_continuation(), ast_rtp_dtmf_end_with_duration(), and rtp_raw_write().

calculate_lost_packet_statistics()

static void calculate_lost_packet_statistics ( struct ast_rtp *  rtp, unsigned int *  lost_packets, int *  fraction_lost  )

static

Definition at line 4813 of file res_rtp_asterisk.c.

{
    unsigned int extended_seq_no;
    unsigned int expected_packets;
    unsigned int expected_interval;
    unsigned int received_interval;
    int lost_interval;
 
    /* Compute statistics */
    extended_seq_no = rtp->cycles + rtp->lastrxseqno;
    expected_packets = extended_seq_no - rtp->seedrxseqno + 1;
    if (rtp->rxcount > expected_packets) {
        expected_packets += rtp->rxcount - expected_packets;
    }
    *lost_packets = expected_packets - rtp->rxcount;
    expected_interval = expected_packets - rtp->rtcp->expected_prior;
    received_interval = rtp->rxcount - rtp->rtcp->received_prior;
    if (received_interval > expected_interval) {
        /* If we receive some late packets it is possible for the packets
         * we received in this interval to exceed the number we expected.
         * We update the expected so that the packet loss calculations
         * show that no packets are lost.
         */
        expected_interval = received_interval;
    }
    lost_interval = expected_interval - received_interval;
    if (expected_interval == 0 || lost_interval <= 0) {
        *fraction_lost = 0;
    } else {
        *fraction_lost = (lost_interval << 8) / expected_interval;
    }
 
    /* Update RTCP statistics */
    rtp->rtcp->received_prior = rtp->rxcount;
    rtp->rtcp->expected_prior = expected_packets;
 
    /*
     * While rxlost represents the number of packets lost since the last report was sent, for
     * the calculations below it should be thought of as a single sample. Thus min/max are the
     * lowest/highest sample value seen, and the mean is the average number of packets lost
     * between each report. As such rxlost_count only needs to be incremented per report.
     */
    if (lost_interval <= 0) {
        rtp->rtcp->rxlost = 0;
    } else {
        rtp->rtcp->rxlost = lost_interval;
    }
    if (rtp->rtcp->rxlost_count == 0) {
        rtp->rtcp->minrxlost = rtp->rtcp->rxlost;
    }
    if (lost_interval && lost_interval < rtp->rtcp->minrxlost) {
        rtp->rtcp->minrxlost = rtp->rtcp->rxlost;
    }
    if (lost_interval > rtp->rtcp->maxrxlost) {
        rtp->rtcp->maxrxlost = rtp->rtcp->rxlost;
    }
 
    calc_mean_and_standard_deviation(rtp->rtcp->rxlost, &rtp->rtcp->normdev_rxlost,
        &rtp->rtcp->stdev_rxlost, &rtp->rtcp->rxlost_count);
}

References calc_mean_and_standard_deviation(), and ast_srtp::rtp.

Referenced by ast_rtcp_generate_report().

compare_by_value()

static int compare_by_value ( int  elem, int  value  )

static

Helper function to compare an elem in a vector by value.

Definition at line 3192 of file res_rtp_asterisk.c.

{
    return elem - value;
}

References value.

Referenced by ast_rtp_read().

create_dtmf_frame()

static struct ast_frame * create_dtmf_frame ( struct ast_rtp_instance *  instance, enum ast_frame_type  type, int  compensate  )

static

Definition at line 6014 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_sockaddr remote_address = { {0,} };
 
    ast_rtp_instance_get_remote_address(instance, &remote_address);
 
    if (((compensate && type == AST_FRAME_DTMF_END) || (type == AST_FRAME_DTMF_BEGIN)) && ast_tvcmp(ast_tvnow(), rtp->dtmfmute) < 0) {
        ast_debug_rtp(1, "(%p) RTP ignore potential DTMF echo from '%s'\n",
            instance, ast_sockaddr_stringify(&remote_address));
        rtp->resp = 0;
        rtp->dtmfsamples = 0;
        return &ast_null_frame;
    } else if (type == AST_FRAME_DTMF_BEGIN && rtp->resp == 'X') {
        ast_debug_rtp(1, "(%p) RTP ignore flash begin from '%s'\n",
            instance, ast_sockaddr_stringify(&remote_address));
        rtp->resp = 0;
        rtp->dtmfsamples = 0;
        return &ast_null_frame;
    }
 
    if (rtp->resp == 'X') {
        ast_debug_rtp(1, "(%p) RTP creating flash Frame at %s\n",
            instance, ast_sockaddr_stringify(&remote_address));
        rtp->f.frametype = AST_FRAME_CONTROL;
        rtp->f.subclass.integer = AST_CONTROL_FLASH;
    } else {
        ast_debug_rtp(1, "(%p) RTP creating %s DTMF Frame: %d (%c), at %s\n",
            instance, type == AST_FRAME_DTMF_END ? "END" : "BEGIN",
            rtp->resp, rtp->resp,
            ast_sockaddr_stringify(&remote_address));
        rtp->f.frametype = type;
        rtp->f.subclass.integer = rtp->resp;
    }
    rtp->f.datalen = 0;
    rtp->f.samples = 0;
    rtp->f.mallocd = 0;
    rtp->f.src = "RTP";
    AST_LIST_NEXT(&rtp->f, frame_list) = NULL;
 
    return &rtp->f;
}

References AST_CONTROL_FLASH, ast_debug_rtp, AST_FRAME_CONTROL, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_LIST_NEXT, ast_null_frame, ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_sockaddr_stringify(), ast_tvcmp(), ast_tvnow(), ast_frame::datalen, ast_rtp::dtmfmute, ast_rtp::dtmfsamples, ast_rtp::f, ast_frame::frametype, ast_frame_subclass::integer, ast_frame::mallocd, NULL, ast_rtp::resp, ast_frame::samples, ast_frame::src, ast_frame::subclass, and type.

Referenced by ast_rtp_interpret(), process_dtmf_cisco(), and process_dtmf_rfc2833().

create_new_socket()

static int create_new_socket ( const char *  type, struct ast_sockaddr *  bind_addr  )

static

Definition at line 3596 of file res_rtp_asterisk.c.

{
    int af, sock;
 
    af = ast_sockaddr_is_ipv4(bind_addr) ? AF_INET  :
         ast_sockaddr_is_ipv6(bind_addr) ? AF_INET6 : -1;
    sock = ast_socket_nonblock(af, SOCK_DGRAM, 0);
 
    if (sock < 0) {
        ast_log(LOG_WARNING, "Unable to allocate %s socket: %s\n", type, strerror(errno));
        return sock;
    }
 
#ifdef SO_NO_CHECK
    if (nochecksums) {
        setsockopt(sock, SOL_SOCKET, SO_NO_CHECK, &nochecksums, sizeof(nochecksums));
    }
#endif
 
#ifdef HAVE_SOCK_IPV6_V6ONLY
    if (AF_INET6 == af && ast_sockaddr_is_any(bind_addr)) {
        /* ICE relies on dual-stack behavior. Ensure it is enabled. */
        if (setsockopt(sock, IPPROTO_IPV6, IPV6_V6ONLY, &(int){0}, sizeof(int)) != 0) {
            ast_log(LOG_WARNING, "setsockopt IPV6_V6ONLY=0 failed: %s\n", strerror(errno));
        }
    }
#endif
 
    return sock;
}

References ast_log, ast_sockaddr_is_any(), ast_sockaddr_is_ipv4(), ast_sockaddr_is_ipv6(), ast_socket_nonblock, errno, LOG_WARNING, and type.

Referenced by ast_rtp_prop_set(), and rtp_allocate_transport().

find_by_value()

static int find_by_value ( int  elem, int  value  )

static

Helper function to find an elem in a vector by value.

Definition at line 3198 of file res_rtp_asterisk.c.

{
    return elem == value;
}

References value.

Referenced by ast_rtcp_generate_nack(), and ast_rtp_read().

handle_cli_rtcp_set_debug()

static char * handle_cli_rtcp_set_debug ( struct ast_cli_entry *  e, int  cmd, struct ast_cli_args *  a  )

static

Definition at line 10090 of file res_rtp_asterisk.c.

{
    switch (cmd) {
    case CLI_INIT:
        e->command = "rtcp set debug {on|off|ip}";
        e->usage =
            "Usage: rtcp set debug {on|off|ip host[:port]}\n"
            "       Enable/Disable dumping of all RTCP packets. If 'ip' is\n"
            "       specified, limit the dumped packets to those to and from\n"
            "       the specified 'host' with optional port.\n";
        return NULL;
    case CLI_GENERATE:
        return NULL;
    }
 
    if (a->argc == e->args) { /* set on or off */
        if (!strncasecmp(a->argv[e->args-1], "on", 2)) {
            ast_debug_category_set_sublevel(AST_LOG_CATEGORY_RTCP_PACKET, AST_LOG_CATEGORY_ENABLED);
            memset(&rtcpdebugaddr, 0, sizeof(rtcpdebugaddr));
            ast_cli(a->fd, "RTCP Packet Debugging Enabled\n");
            return CLI_SUCCESS;
        } else if (!strncasecmp(a->argv[e->args-1], "off", 3)) {
            ast_debug_category_set_sublevel(AST_LOG_CATEGORY_RTCP_PACKET, AST_LOG_CATEGORY_DISABLED);
            ast_cli(a->fd, "RTCP Packet Debugging Disabled\n");
            return CLI_SUCCESS;
        }
    } else if (a->argc == e->args +1) { /* ip */
        return rtcp_do_debug_ip(a);
    }
 
    return CLI_SHOWUSAGE;   /* default, failure */
}

References a, ast_cli_entry::args, ast_cli(), ast_debug_category_set_sublevel(), AST_LOG_CATEGORY_DISABLED, AST_LOG_CATEGORY_ENABLED, AST_LOG_CATEGORY_RTCP_PACKET, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, NULL, rtcp_do_debug_ip(), rtcpdebugaddr, and ast_cli_entry::usage.

handle_cli_rtcp_set_stats()

static char * handle_cli_rtcp_set_stats ( struct ast_cli_entry *  e, int  cmd, struct ast_cli_args *  a  )

static

Definition at line 10123 of file res_rtp_asterisk.c.

{
    switch (cmd) {
    case CLI_INIT:
        e->command = "rtcp set stats {on|off}";
        e->usage =
            "Usage: rtcp set stats {on|off}\n"
            "       Enable/Disable dumping of RTCP stats.\n";
        return NULL;
    case CLI_GENERATE:
        return NULL;
    }
 
    if (a->argc != e->args)
        return CLI_SHOWUSAGE;
 
    if (!strncasecmp(a->argv[e->args-1], "on", 2))
        rtcpstats = 1;
    else if (!strncasecmp(a->argv[e->args-1], "off", 3))
        rtcpstats = 0;
    else
        return CLI_SHOWUSAGE;
 
    ast_cli(a->fd, "RTCP Stats %s\n", rtcpstats ? "Enabled" : "Disabled");
    return CLI_SUCCESS;
}

References a, ast_cli_entry::args, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, NULL, rtcpstats, and ast_cli_entry::usage.

handle_cli_rtp_set_debug()

static char * handle_cli_rtp_set_debug ( struct ast_cli_entry *  e, int  cmd, struct ast_cli_args *  a  )

static

Definition at line 9989 of file res_rtp_asterisk.c.

{
    switch (cmd) {
    case CLI_INIT:
        e->command = "rtp set debug {on|off|ip}";
        e->usage =
            "Usage: rtp set debug {on|off|ip host[:port]}\n"
            "       Enable/Disable dumping of all RTP packets. If 'ip' is\n"
            "       specified, limit the dumped packets to those to and from\n"
            "       the specified 'host' with optional port.\n";
        return NULL;
    case CLI_GENERATE:
        return NULL;
    }
 
    if (a->argc == e->args) { /* set on or off */
        if (!strncasecmp(a->argv[e->args-1], "on", 2)) {
            ast_debug_category_set_sublevel(AST_LOG_CATEGORY_RTP_PACKET, AST_LOG_CATEGORY_ENABLED);
            memset(&rtpdebugaddr, 0, sizeof(rtpdebugaddr));
            ast_cli(a->fd, "RTP Packet Debugging Enabled\n");
            return CLI_SUCCESS;
        } else if (!strncasecmp(a->argv[e->args-1], "off", 3)) {
            ast_debug_category_set_sublevel(AST_LOG_CATEGORY_RTP_PACKET, AST_LOG_CATEGORY_DISABLED);
            ast_cli(a->fd, "RTP Packet Debugging Disabled\n");
            return CLI_SUCCESS;
        }
    } else if (a->argc == e->args +1) { /* ip */
        return rtp_do_debug_ip(a);
    }
 
    return CLI_SHOWUSAGE;   /* default, failure */
}

References a, ast_cli_entry::args, ast_cli(), ast_debug_category_set_sublevel(), AST_LOG_CATEGORY_DISABLED, AST_LOG_CATEGORY_ENABLED, AST_LOG_CATEGORY_RTP_PACKET, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, NULL, rtp_do_debug_ip(), rtpdebugaddr, and ast_cli_entry::usage.

handle_cli_rtp_settings()

static char * handle_cli_rtp_settings ( struct ast_cli_entry *  e, int  cmd, struct ast_cli_args *  a  )

static

Definition at line 10023 of file res_rtp_asterisk.c.

{
#ifdef HAVE_PJPROJECT
    struct sockaddr_in stunaddr_copy;
    const char *stun_hostname_copy = NULL;
    int stunaddr_ttl_copy = 0;
#endif
    switch (cmd) {
    case CLI_INIT:
        e->command = "rtp show settings";
        e->usage =
            "Usage: rtp show settings\n"
            "       Display RTP configuration settings\n";
        return NULL;
    case CLI_GENERATE:
        return NULL;
    }
 
    if (a->argc != 3) {
        return CLI_SHOWUSAGE;
    }
 
    ast_cli(a->fd, "\n\nGeneral Settings:\n");
    ast_cli(a->fd, "----------------\n");
    ast_cli(a->fd, "  Port start:      %d\n", rtpstart);
    ast_cli(a->fd, "  Port end:        %d\n", rtpend);
#ifdef SO_NO_CHECK
    ast_cli(a->fd, "  Checksums:       %s\n", AST_CLI_YESNO(nochecksums == 0));
#endif
    ast_cli(a->fd, "  DTMF Timeout:    %d\n", dtmftimeout);
    ast_cli(a->fd, "  Strict RTP:      %s\n", AST_CLI_YESNO(strictrtp));
 
    if (strictrtp) {
        ast_cli(a->fd, "  Probation:       %d frames\n", learning_min_sequential);
    }
 
    ast_cli(a->fd, "  Replay Protect:  %s\n", AST_CLI_YESNO(srtp_replay_protection));
#ifdef HAVE_PJPROJECT
    ast_cli(a->fd, "  ICE support:     %s\n", AST_CLI_YESNO(icesupport));
 
    ast_rwlock_rdlock(&stunaddr_lock);
    memcpy(&stunaddr_copy, &stunaddr, sizeof(stunaddr));
    stun_hostname_copy = ast_strdupa(S_OR(stun_hostname, ""));
    stunaddr_ttl_copy = stunaddr_ttl;
    ast_rwlock_unlock(&stunaddr_lock);
 
    ast_cli(a->fd, "  STUN:            %s\n", stunaddr_copy.sin_addr.s_addr ? "enbabled" : "disabled");
    if (ast_strlen_zero(stun_hostname_copy)) {
        ast_cli(a->fd, "   Address:        %s:%d\n", ast_inet_ntoa(stunaddr_copy.sin_addr),
            htons(stunaddr_copy.sin_port));
    } else {
        ast_cli(a->fd, "   Hostname:       %s:%d\n", stun_hostname_copy, htons(stunaddr_copy.sin_port));
        ast_cli(a->fd, "   Resolved Addr:  %s:%d%s\n", ast_inet_ntoa(stunaddr_copy.sin_addr),
            htons(stunaddr_copy.sin_port),
            stunaddr_copy.sin_addr.s_addr ? stunaddr_resolver ? "" : " (possibly stale)" : " (lookup failed)");
        ast_cli(a->fd, "   Last TTL:       %d  (periodic resolution %s)\n", stunaddr_ttl_copy,
            stunaddr_resolver ? "enabled" : "disabled");
        ast_cli(a->fd, "   Reresove TTL 0: %s\n", AST_CLI_YESNO(stunaddr_reresolve_ttl_0));
    }
    if (stun_acl) {
        ast_acl_output(a->fd, stun_acl, "   ");
    }
#endif
    return CLI_SUCCESS;
}

References a, ast_acl_output(), ast_cli(), AST_CLI_YESNO, ast_inet_ntoa(), ast_rwlock_rdlock, ast_rwlock_unlock, ast_strdupa, ast_strlen_zero(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, dtmftimeout, learning_min_sequential, NULL, rtpend, rtpstart, S_OR, srtp_replay_protection, strictrtp, and ast_cli_entry::usage.

load_module()

static int load_module ( void  )

static

Definition at line 10679 of file res_rtp_asterisk.c.

{
#ifdef HAVE_PJPROJECT
    pj_lock_t *lock;
 
    ast_sockaddr_parse(&lo6, "::1", PARSE_PORT_IGNORE);
 
    AST_PJPROJECT_INIT_LOG_LEVEL();
    if (pj_init() != PJ_SUCCESS) {
        return AST_MODULE_LOAD_DECLINE;
    }
 
    if (pjlib_util_init() != PJ_SUCCESS) {
        rtp_terminate_pjproject();
        return AST_MODULE_LOAD_DECLINE;
    }
 
    if (pjnath_init() != PJ_SUCCESS) {
        rtp_terminate_pjproject();
        return AST_MODULE_LOAD_DECLINE;
    }
 
    ast_pjproject_caching_pool_init(&cachingpool, &pj_pool_factory_default_policy, 0);
 
    pool = pj_pool_create(&cachingpool.factory, "timer", 512, 512, NULL);
 
    if (pj_timer_heap_create(pool, 100, &timer_heap) != PJ_SUCCESS) {
        rtp_terminate_pjproject();
        return AST_MODULE_LOAD_DECLINE;
    }
 
    if (pj_lock_create_recursive_mutex(pool, "rtp%p", &lock) != PJ_SUCCESS) {
        rtp_terminate_pjproject();
        return AST_MODULE_LOAD_DECLINE;
    }
 
    pj_timer_heap_set_lock(timer_heap, lock, PJ_TRUE);
 
    if (pj_thread_create(pool, "timer", &timer_worker_thread, NULL, 0, 0, &timer_thread) != PJ_SUCCESS) {
        rtp_terminate_pjproject();
        return AST_MODULE_LOAD_DECLINE;
    }
 
#endif
 
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP) && defined(HAVE_OPENSSL_BIO_METHOD)
    dtls_bio_methods = BIO_meth_new(BIO_TYPE_BIO, "rtp write");
    if (!dtls_bio_methods) {
#ifdef HAVE_PJPROJECT
        rtp_terminate_pjproject();
#endif
        return AST_MODULE_LOAD_DECLINE;
    }
    BIO_meth_set_write(dtls_bio_methods, dtls_bio_write);
    BIO_meth_set_ctrl(dtls_bio_methods, dtls_bio_ctrl);
    BIO_meth_set_create(dtls_bio_methods, dtls_bio_new);
    BIO_meth_set_destroy(dtls_bio_methods, dtls_bio_free);
#endif
 
    if (ast_rtp_engine_register(&asterisk_rtp_engine)) {
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP) && defined(HAVE_OPENSSL_BIO_METHOD)
        BIO_meth_free(dtls_bio_methods);
#endif
#ifdef HAVE_PJPROJECT
        rtp_terminate_pjproject();
#endif
        return AST_MODULE_LOAD_DECLINE;
    }
 
    if (ast_cli_register_multiple(cli_rtp, ARRAY_LEN(cli_rtp))) {
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP) && defined(HAVE_OPENSSL_BIO_METHOD)
        BIO_meth_free(dtls_bio_methods);
#endif
#ifdef HAVE_PJPROJECT
        ast_rtp_engine_unregister(&asterisk_rtp_engine);
        rtp_terminate_pjproject();
#endif
        return AST_MODULE_LOAD_DECLINE;
    }
 
    rtp_reload(0, 0);
 
    return AST_MODULE_LOAD_SUCCESS;
}

References ARRAY_LEN, ast_cli_register_multiple, AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_SUCCESS, ast_pjproject_caching_pool_init(), AST_PJPROJECT_INIT_LOG_LEVEL, ast_rtp_engine_register, ast_rtp_engine_unregister(), ast_sockaddr_parse(), asterisk_rtp_engine, cachingpool, cli_rtp, lock, NULL, PARSE_PORT_IGNORE, and rtp_reload().

ntp2timeval()

static void ntp2timeval ( unsigned int  msw, unsigned int  lsw, struct timeval *  tv  )

static

Definition at line 4806 of file res_rtp_asterisk.c.

{
    tv->tv_sec = msw - 2208988800u;
    /* Reverse the sequence in timeval2ntp() */
    tv->tv_usec = ((((lsw >> 7) * 125) >> 7) * 125) >> 12;
}

Referenced by ast_rtcp_interpret().

process_cn_rfc3389()

static struct ast_frame * process_cn_rfc3389 ( struct ast_rtp_instance *  instance, unsigned char *  data, int  len, unsigned int  seqno, unsigned int  timestamp, int  payloadtype, int  mark  )

static

Definition at line 6275 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    /* Convert comfort noise into audio with various codecs.  Unfortunately this doesn't
       totally help us out because we don't have an engine to keep it going and we are not
       guaranteed to have it every 20ms or anything */
    if (ast_debug_rtp_packet_is_allowed) {
        ast_debug(0, "- RTP 3389 Comfort noise event: Format %s (len = %d)\n",
            ast_format_get_name(rtp->lastrxformat), len);
    }
 
    if (!ast_test_flag(rtp, FLAG_3389_WARNING)) {
        struct ast_sockaddr remote_address = { {0,} };
 
        ast_rtp_instance_get_remote_address(instance, &remote_address);
 
        ast_log(LOG_NOTICE, "Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client address: %s\n",
            ast_sockaddr_stringify(&remote_address));
        ast_set_flag(rtp, FLAG_3389_WARNING);
    }
 
    /* Must have at least one byte */
    if (!len) {
        return NULL;
    }
    if (len < 24) {
        rtp->f.data.ptr = rtp->rawdata + AST_FRIENDLY_OFFSET;
        rtp->f.datalen = len - 1;
        rtp->f.offset = AST_FRIENDLY_OFFSET;
        memcpy(rtp->f.data.ptr, data + 1, len - 1);
    } else {
        rtp->f.data.ptr = NULL;
        rtp->f.offset = 0;
        rtp->f.datalen = 0;
    }
    rtp->f.frametype = AST_FRAME_CNG;
    rtp->f.subclass.integer = data[0] & 0x7f;
    rtp->f.samples = 0;
    rtp->f.delivery.tv_usec = rtp->f.delivery.tv_sec = 0;
 
    return &rtp->f;
}

References ast_debug, ast_debug_rtp_packet_is_allowed, ast_format_get_name(), AST_FRAME_CNG, AST_FRIENDLY_OFFSET, ast_log, ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_set_flag, ast_sockaddr_stringify(), ast_test_flag, ast_frame::data, ast_frame::datalen, ast_frame::delivery, ast_rtp::f, FLAG_3389_WARNING, ast_frame::frametype, ast_frame_subclass::integer, ast_rtp::lastrxformat, len(), LOG_NOTICE, NULL, ast_frame::offset, ast_frame::ptr, ast_rtp::rawdata, ast_frame::samples, and ast_frame::subclass.

Referenced by ast_rtp_interpret().

process_dtmf_cisco()

static struct ast_frame * process_dtmf_cisco ( struct ast_rtp_instance *  instance, unsigned char *  data, int  len, unsigned int  seqno, unsigned int  timestamp, int  payloadtype, int  mark  )

static

Definition at line 6195 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    unsigned int event, flags, power;
    char resp = 0;
    unsigned char seq;
    struct ast_frame *f = NULL;
 
    if (len < 4) {
        return NULL;
    }
 
    /*      The format of Cisco RTP DTMF packet looks like next:
        +0                              - sequence number of DTMF RTP packet (begins from 1,
                          wrapped to 0)
        +1                              - set of flags
        +1 (bit 0)              - flaps by different DTMF digits delimited by audio
                          or repeated digit without audio???
        +2 (+4,+6,...)  - power level? (rises from 0 to 32 at begin of tone
                          then falls to 0 at its end)
        +3 (+5,+7,...)  - detected DTMF digit (0..9,*,#,A-D,...)
        Repeated DTMF information (bytes 4/5, 6/7) is history shifted right
        by each new packet and thus provides some redundancy.
 
        Sample of Cisco RTP DTMF packet is (all data in hex):
            19 07 00 02 12 02 20 02
        showing end of DTMF digit '2'.
 
        The packets
            27 07 00 02 0A 02 20 02
            28 06 20 02 00 02 0A 02
        shows begin of new digit '2' with very short pause (20 ms) after
        previous digit '2'. Bit +1.0 flips at begin of new digit.
 
        Cisco RTP DTMF packets comes as replacement of audio RTP packets
        so its uses the same sequencing and timestamping rules as replaced
        audio packets. Repeat interval of DTMF packets is 20 ms and not rely
        on audio framing parameters. Marker bit isn't used within stream of
        DTMFs nor audio stream coming immediately after DTMF stream. Timestamps
        are not sequential at borders between DTMF and audio streams,
    */
 
    seq = data[0];
    flags = data[1];
    power = data[2];
    event = data[3] & 0x1f;
 
    if (ast_debug_rtp_packet_is_allowed)
        ast_debug(0, "Cisco DTMF Digit: %02x (len=%d, seq=%d, flags=%02x, power=%u, history count=%d)\n", event, len, seq, flags, power, (len - 4) / 2);
    if (event < 10) {
        resp = '0' + event;
    } else if (event < 11) {
        resp = '*';
    } else if (event < 12) {
        resp = '#';
    } else if (event < 16) {
        resp = 'A' + (event - 12);
    } else if (event < 17) {
        resp = 'X';
    }
    if ((!rtp->resp && power) || (rtp->resp && (rtp->resp != resp))) {
        rtp->resp = resp;
        /* Why we should care on DTMF compensation at reception? */
        if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE)) {
            f = create_dtmf_frame(instance, AST_FRAME_DTMF_BEGIN, 0);
            rtp->dtmfsamples = 0;
        }
    } else if ((rtp->resp == resp) && !power) {
        f = create_dtmf_frame(instance, AST_FRAME_DTMF_END, ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE));
        f->samples = rtp->dtmfsamples * (ast_rtp_get_rate(rtp->lastrxformat) / 1000);
        rtp->resp = 0;
    } else if (rtp->resp == resp) {
        rtp->dtmfsamples += 20 * (ast_rtp_get_rate(rtp->lastrxformat) / 1000);
    }
 
    rtp->dtmf_timeout = 0;
 
    return f;
}

References ast_debug, ast_debug_rtp_packet_is_allowed, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, ast_rtp_get_rate(), ast_rtp_instance_get_data(), ast_rtp_instance_get_prop(), AST_RTP_PROPERTY_DTMF_COMPENSATE, create_dtmf_frame(), ast_frame::data, ast_rtp::dtmf_timeout, ast_rtp::dtmfsamples, ast_frame::flags, ast_rtp::flags, ast_rtp::lastrxformat, len(), NULL, ast_rtp::resp, ast_frame::samples, and seq.

Referenced by ast_rtp_interpret().

process_dtmf_rfc2833()

static void process_dtmf_rfc2833 ( struct ast_rtp_instance *  instance, unsigned char *  data, int  len, unsigned int  seqno, unsigned int  timestamp, int  payloadtype, int  mark, struct frame_list *  frames  )

static

Definition at line 6057 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct ast_sockaddr remote_address = { {0,} };
    unsigned int event, event_end, samples;
    char resp = 0;
    struct ast_frame *f = NULL;
 
    ast_rtp_instance_get_remote_address(instance, &remote_address);
 
    /* Figure out event, event end, and samples */
    event = ntohl(*((unsigned int *)(data)));
    event >>= 24;
    event_end = ntohl(*((unsigned int *)(data)));
    event_end <<= 8;
    event_end >>= 24;
    samples = ntohl(*((unsigned int *)(data)));
    samples &= 0xFFFF;
 
    if (rtp_debug_test_addr(&remote_address)) {
        ast_verbose("Got  RTP RFC2833 from   %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6d, mark %d, event %08x, end %d, duration %-5.5u) \n",
                ast_sockaddr_stringify(&remote_address),
                payloadtype, seqno, timestamp, len, (mark?1:0), event, ((event_end & 0x80)?1:0), samples);
    }
 
    /* Print out debug if turned on */
    if (ast_debug_rtp_packet_is_allowed)
        ast_debug(0, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
 
    /* Figure out what digit was pressed */
    if (event < 10) {
        resp = '0' + event;
    } else if (event < 11) {
        resp = '*';
    } else if (event < 12) {
        resp = '#';
    } else if (event < 16) {
        resp = 'A' + (event - 12);
    } else if (event < 17) {        /* Event 16: Hook flash */
        resp = 'X';
    } else {
        /* Not a supported event */
        ast_debug_rtp(1, "(%p) RTP ignoring RTP 2833 Event: %08x. Not a DTMF Digit.\n", instance, event);
        return;
    }
 
    if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE)) {
        if (!rtp->last_end_timestamp.is_set || rtp->last_end_timestamp.ts != timestamp || (rtp->resp && rtp->resp != resp)) {
            rtp->resp = resp;
            rtp->dtmf_timeout = 0;
            f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE)));
            f->len = 0;
            rtp->last_end_timestamp.ts = timestamp;
            rtp->last_end_timestamp.is_set = 1;
            AST_LIST_INSERT_TAIL(frames, f, frame_list);
        }
    } else {
        /*  The duration parameter measures the complete
            duration of the event (from the beginning) - RFC2833.
            Account for the fact that duration is only 16 bits long
            (about 8 seconds at 8000 Hz) and can wrap is digit
            is hold for too long. */
        unsigned int new_duration = rtp->dtmf_duration;
        unsigned int last_duration = new_duration & 0xFFFF;
 
        if (last_duration > 64000 && samples < last_duration) {
            new_duration += 0xFFFF + 1;
        }
        new_duration = (new_duration & ~0xFFFF) | samples;
 
        if (event_end & 0x80) {
            /* End event */
            if (rtp->last_seqno != seqno && (!rtp->last_end_timestamp.is_set || timestamp > rtp->last_end_timestamp.ts)) {
                rtp->last_end_timestamp.ts = timestamp;
                rtp->last_end_timestamp.is_set = 1;
                rtp->dtmf_duration = new_duration;
                rtp->resp = resp;
                f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0));
                f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, ast_rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
                rtp->resp = 0;
                rtp->dtmf_duration = rtp->dtmf_timeout = 0;
                AST_LIST_INSERT_TAIL(frames, f, frame_list);
            } else if (ast_debug_rtp_packet_is_allowed) {
                ast_debug_rtp(1, "(%p) RTP dropping duplicate or out of order DTMF END frame (seqno: %u, ts %u, digit %c)\n",
                    instance, seqno, timestamp, resp);
            }
        } else {
            /* Begin/continuation */
 
            /* The second portion of the seqno check is to not mistakenly
             * stop accepting DTMF if the seqno rolls over beyond
             * 65535.
             */
            if ((rtp->last_seqno > seqno && rtp->last_seqno - seqno < 50)
               || (rtp->last_end_timestamp.is_set
                  && timestamp <= rtp->last_end_timestamp.ts)) {
                /* Out of order frame. Processing this can cause us to
                 * improperly duplicate incoming DTMF, so just drop
                 * this.
                 */
                if (ast_debug_rtp_packet_is_allowed) {
                    ast_debug(0, "Dropping out of order DTMF frame (seqno %u, ts %u, digit %c)\n",
                        seqno, timestamp, resp);
                }
                return;
            }
 
            if (rtp->resp && rtp->resp != resp) {
                /* Another digit already began. End it */
                f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0));
                f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, ast_rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
                rtp->resp = 0;
                rtp->dtmf_duration = rtp->dtmf_timeout = 0;
                AST_LIST_INSERT_TAIL(frames, f, frame_list);
            }
 
            if (rtp->resp) {
                /* Digit continues */
                rtp->dtmf_duration = new_duration;
            } else {
                /* New digit began */
                rtp->resp = resp;
                f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_BEGIN, 0));
                rtp->dtmf_duration = samples;
                AST_LIST_INSERT_TAIL(frames, f, frame_list);
            }
 
            rtp->dtmf_timeout = timestamp + rtp->dtmf_duration + dtmftimeout;
        }
 
        rtp->last_seqno = seqno;
    }
 
    rtp->dtmfsamples = samples;
 
    return;
}

References ast_debug, ast_debug_rtp, ast_debug_rtp_packet_is_allowed, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, ast_frdup, AST_LIST_INSERT_TAIL, ast_rtp_get_rate(), ast_rtp_instance_get_data(), ast_rtp_instance_get_prop(), ast_rtp_instance_get_remote_address, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_samp2tv(), ast_sockaddr_stringify(), ast_tv(), ast_tvdiff_ms(), ast_verbose, create_dtmf_frame(), ast_frame::data, ast_rtp::dtmf_duration, ast_rtp::dtmf_timeout, ast_rtp::dtmfsamples, dtmftimeout, ast_frame_subclass::format, frames, optional_ts::is_set, ast_rtp::last_end_timestamp, ast_rtp::last_seqno, len(), ast_frame::len, NULL, ast_rtp::resp, rtp_debug_test_addr(), ast_frame::samples, ast_frame::seqno, ast_frame::subclass, and optional_ts::ts.

Referenced by ast_rtp_interpret().

put_unaligned_time24()

static void put_unaligned_time24 ( void *  p, uint32_t  time_msw, uint32_t  time_lsw  )

static

Definition at line 5268 of file res_rtp_asterisk.c.

{
    unsigned char *cp = p;
    uint32_t datum;
 
    /* Convert the time to 6.18 format */
    datum = (time_msw << 18) & 0x00fc0000;
    datum |= (time_lsw >> 14) & 0x0003ffff;
 
    cp[0] = datum >> 16;
    cp[1] = datum >> 8;
    cp[2] = datum;
}

Referenced by ast_rtp_rtcp_handle_nack(), rtp_raw_write(), and rtp_transport_wide_cc_feedback_produce().

red_t140_to_red()

static struct ast_frame * red_t140_to_red ( struct rtp_red *  red )

static

Definition at line 5516 of file res_rtp_asterisk.c.

{
    unsigned char *data = red->t140red.data.ptr;
    int len = 0;
    int i;
 
    /* replace most aged generation */
    if (red->len[0]) {
        for (i = 1; i < red->num_gen+1; i++)
            len += red->len[i];
 
        memmove(&data[red->hdrlen], &data[red->hdrlen+red->len[0]], len);
    }
 
    /* Store length of each generation and primary data length*/
    for (i = 0; i < red->num_gen; i++)
        red->len[i] = red->len[i+1];
 
    /*
     * RED generation payload sizes are limited to 255 (UCHAR_MAX) bytes by virtue of
     * red->len being an array of usigned chars.  If the new primary payload exceeds that,
     * we're going to truncate it to 255.
     */
    if (red->t140.datalen > UCHAR_MAX) {
        ast_log(LOG_WARNING, "New T.140 frame of %d bytes exceeds max of %u. Discarding %d bytes.\n",
            red->t140.datalen, UCHAR_MAX, red->t140.datalen - UCHAR_MAX);
        red->t140.datalen = UCHAR_MAX;
    }
    red->len[i] = red->t140.datalen;
 
    /* write each generation length in red header */
    len = red->hdrlen;
    for (i = 0; i < red->num_gen; i++) {
        len += data[i*4+3] = red->len[i];
    }
 
    /* add primary data to buffer */
    memcpy(&data[len], red->t140.data.ptr, red->t140.datalen);
    red->t140red.datalen = len + red->t140.datalen;
 
    /* no primary data and no generations to send */
    if (len == red->hdrlen && !red->t140.datalen) {
        return NULL;
    }
 
    /* reset t.140 buffer */
    red->t140.datalen = 0;
 
    return &red->t140red;
}

References ast_log, ast_frame::data, ast_frame::datalen, rtp_red::hdrlen, len(), rtp_red::len, LOG_WARNING, NULL, rtp_red::num_gen, ast_frame::ptr, rtp_red::t140, and rtp_red::t140red.

Referenced by ast_rtp_write().

red_write()

static int red_write ( const void *  data )

static

Write t140 redundancy frame.

Parameters

data	primary data to be buffered

Scheduler callback

Definition at line 9342 of file res_rtp_asterisk.c.

{
    struct ast_rtp_instance *instance = (struct ast_rtp_instance*) data;
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 
    ao2_lock(instance);
    if (rtp->red->t140.datalen > 0) {
        ast_rtp_write(instance, &rtp->red->t140);
    }
    ao2_unlock(instance);
 
    return 1;
}

References ao2_lock, ao2_unlock, ast_rtp_instance_get_data(), ast_rtp_write(), ast_rtp_instance::data, ast_frame::datalen, ast_rtp::red, and rtp_red::t140.

Referenced by rtp_red_init().

reload_module()

static int reload_module ( void  )

static

Definition at line 10647 of file res_rtp_asterisk.c.

{
    rtp_reload(1, 0);
    return 0;
}

References rtp_reload().

rtcp_debug_test_addr()

static int rtcp_debug_test_addr ( struct ast_sockaddr *  addr )

inlinestatic

Definition at line 2854 of file res_rtp_asterisk.c.

{
    if (!ast_debug_rtcp_packet_is_allowed) {
        return 0;
    }
    if (!ast_sockaddr_isnull(&rtcpdebugaddr)) {
        if (rtcpdebugport) {
            return (ast_sockaddr_cmp(&rtcpdebugaddr, addr) == 0); /* look for RTCP packets from IP+Port */
        } else {
            return (ast_sockaddr_cmp_addr(&rtcpdebugaddr, addr) == 0); /* only look for RTCP packets from IP */
        }
    }
 
    return 1;
}

References ast_debug_rtcp_packet_is_allowed, ast_sockaddr_cmp(), ast_sockaddr_cmp_addr(), ast_sockaddr_isnull(), rtcpdebugaddr, and rtcpdebugport.

Referenced by ast_rtcp_calculate_sr_rr_statistics(), and ast_rtcp_interpret().

rtcp_do_debug_ip()

static char * rtcp_do_debug_ip ( struct ast_cli_args *  a )

static

Definition at line 9972 of file res_rtp_asterisk.c.

{
    char *arg = ast_strdupa(a->argv[4]);
    char *debughost = NULL;
    char *debugport = NULL;
 
    if (!ast_sockaddr_parse(&rtcpdebugaddr, arg, 0) || !ast_sockaddr_split_hostport(arg, &debughost, &debugport, 0)) {
        ast_cli(a->fd, "Lookup failed for '%s'\n", arg);
        return CLI_FAILURE;
    }
    rtcpdebugport = (!ast_strlen_zero(debugport) && debugport[0] != '0');
    ast_cli(a->fd, "RTCP Packet Debugging Enabled for address: %s\n",
        ast_sockaddr_stringify(&rtcpdebugaddr));
    ast_debug_category_set_sublevel(AST_LOG_CATEGORY_RTCP_PACKET, AST_LOG_CATEGORY_ENABLED);
    return CLI_SUCCESS;
}

References a, ast_cli(), ast_debug_category_set_sublevel(), AST_LOG_CATEGORY_ENABLED, AST_LOG_CATEGORY_RTCP_PACKET, ast_sockaddr_parse(), ast_sockaddr_split_hostport(), ast_sockaddr_stringify(), ast_strdupa, ast_strlen_zero(), CLI_FAILURE, CLI_SUCCESS, NULL, rtcpdebugaddr, and rtcpdebugport.

Referenced by handle_cli_rtcp_set_debug().

rtcp_mux()

static int rtcp_mux ( struct ast_rtp *  rtp, const unsigned char *  packet  )

static

Definition at line 3203 of file res_rtp_asterisk.c.

{
    uint8_t version;
    uint8_t pt;
    uint8_t m;
 
    if (!rtp->rtcp || rtp->rtcp->type != AST_RTP_INSTANCE_RTCP_MUX) {
        return 0;
    }
 
    version = (packet[0] & 0XC0) >> 6;
    if (version == 0) {
        /* version 0 indicates this is a STUN packet and shouldn't
         * be interpreted as a possible RTCP packet
         */
        return 0;
    }
 
    /* The second octet of a packet will be one of the following:
     * For RTP: The marker bit (1 bit) and the RTP payload type (7 bits)
     * For RTCP: The payload type (8)
     *
     * RTP has a forbidden range of payload types (64-95) since these
     * will conflict with RTCP payload numbers if the marker bit is set.
     */
    m = packet[1] & 0x80;
    pt = packet[1] & 0x7F;
    if (m && pt >= 64 && pt <= 95) {
        return 1;
    }
    return 0;
}

References AST_RTP_INSTANCE_RTCP_MUX, ast_rtp::rtcp, ast_rtcp::type, and version.

Referenced by ast_rtp_read().

rtcp_payload_subtype2str()

static const char * rtcp_payload_subtype2str ( unsigned int  pt, unsigned int  subtype  )

static

Definition at line 6668 of file res_rtp_asterisk.c.

{
    switch (pt) {
    case AST_RTP_RTCP_RTPFB:
        if (subtype == AST_RTP_RTCP_FMT_NACK) {
            return "NACK";
        }
        break;
    case RTCP_PT_PSFB:
        if (subtype == AST_RTP_RTCP_FMT_REMB) {
            return "REMB";
        }
        break;
    default:
        break;
    }
 
    return NULL;
}

References AST_RTP_RTCP_FMT_NACK, AST_RTP_RTCP_FMT_REMB, AST_RTP_RTCP_RTPFB, NULL, and RTCP_PT_PSFB.

Referenced by ast_rtcp_interpret().

rtcp_payload_type2str()

static const char * rtcp_payload_type2str ( unsigned int  pt )

static

Definition at line 6636 of file res_rtp_asterisk.c.

{
    const char *str;
 
    switch (pt) {
    case RTCP_PT_SR:
        str = "Sender Report";
        break;
    case RTCP_PT_RR:
        str = "Receiver Report";
        break;
    case RTCP_PT_FUR:
        /* Full INTRA-frame Request / Fast Update Request */
        str = "H.261 FUR";
        break;
    case RTCP_PT_PSFB:
        /* Payload Specific Feed Back */
        str = "PSFB";
        break;
    case RTCP_PT_SDES:
        str = "Source Description";
        break;
    case RTCP_PT_BYE:
        str = "BYE";
        break;
    default:
        str = "Unknown";
        break;
    }
    return str;
}

References RTCP_PT_BYE, RTCP_PT_FUR, RTCP_PT_PSFB, RTCP_PT_RR, RTCP_PT_SDES, RTCP_PT_SR, and str.

Referenced by ast_rtcp_interpret().

rtcp_recvfrom()

static int rtcp_recvfrom ( struct ast_rtp_instance *  instance, void *  buf, size_t  size, int  flags, struct ast_sockaddr *  sa  )

static

Precondition

instance is locked

Definition at line 3453 of file res_rtp_asterisk.c.

{
    return __rtp_recvfrom(instance, buf, size, flags, sa, 1);
}

References __rtp_recvfrom(), and buf.

Referenced by ast_rtcp_read().

rtcp_sendto()

static int rtcp_sendto ( struct ast_rtp_instance *  instance, void *  buf, size_t  size, int  flags, struct ast_sockaddr *  sa, int *  ice  )

static

Precondition

instance is locked

Definition at line 3539 of file res_rtp_asterisk.c.

{
    return __rtp_sendto(instance, buf, size, flags, sa, 1, ice, 1);
}

References __rtp_sendto(), and buf.

Referenced by ast_rtcp_write(), ast_rtp_read(), rtp_transport_wide_cc_feedback_produce(), rtp_write_rtcp_fir(), and rtp_write_rtcp_psfb().

rtp_allocate_transport()

static int rtp_allocate_transport ( struct ast_rtp_instance *  instance, struct ast_rtp *  rtp  )

static

Definition at line 4151 of file res_rtp_asterisk.c.

{
    int x, startplace, i, maxloops;
    unsigned int port_start, port_end;
 
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
    rtp->rekeyid = -1;
    rtp->dtls.timeout_timer = -1;
#endif
 
    rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_CLOSED : STRICT_RTP_OPEN);
 
    /* Determine the port range to use: per-instance override or global */
    port_start = ast_rtp_instance_get_port_start(instance);
    port_end = ast_rtp_instance_get_port_end(instance);
    if (port_start > 0 && port_end > 0 && port_end > port_start) {
        ast_debug_rtp(1, "(%p) RTP using per-instance port range %d-%d\n",
            instance, port_start, port_end);
    } else {
        port_start = rtpstart;
        port_end = rtpend;
    }
 
    /* Create a new socket for us to listen on and use */
    if ((rtp->s = create_new_socket("RTP", &rtp->bind_address)) < 0) {
        ast_log(LOG_WARNING, "Failed to create a new socket for RTP instance '%p'\n", instance);
        return -1;
    }
 
    /* Now actually find a free RTP port to use */
    x = (ast_random() % (port_end - port_start)) + port_start;
    x = x & ~1;
    startplace = x;
 
    /* Protection against infinite loops in the case there is a potential case where the loop is not broken such as an odd
       start port sneaking in (even though this condition is checked at load.) */
    maxloops = port_end - port_start;
    for (i = 0; i <= maxloops; i++) {
        ast_sockaddr_set_port(&rtp->bind_address, x);
        /* Try to bind, this will tell us whether the port is available or not */
        if (!ast_bind(rtp->s, &rtp->bind_address)) {
            ast_debug_rtp(1, "(%p) RTP allocated port %d\n", instance, x);
            ast_rtp_instance_set_local_address(instance, &rtp->bind_address);
            ast_test_suite_event_notify("RTP_PORT_ALLOCATED", "Port: %d", x);
            break;
        }
 
        x += 2;
        if (x > port_end) {
            x = (port_start + 1) & ~1;
        }
 
        /* See if we ran out of ports or if the bind actually failed because of something other than the address being in use */
        if (x == startplace || (errno != EADDRINUSE && errno != EACCES)) {
            ast_log(LOG_ERROR, "Oh dear... we couldn't allocate a port for RTP instance '%p'\n", instance);
            close(rtp->s);
            rtp->s = -1;
            return -1;
        }
    }
 
#ifdef HAVE_PJPROJECT
    /* Initialize synchronization aspects */
    ast_cond_init(&rtp->cond, NULL);
 
    generate_random_string(rtp->local_ufrag, sizeof(rtp->local_ufrag));
    generate_random_string(rtp->local_passwd, sizeof(rtp->local_passwd));
 
    /* Create an ICE session for ICE negotiation */
    if (icesupport) {
        rtp->ice_num_components = 2;
        ast_debug_ice(2, "(%p) ICE creating session %s (%d)\n", instance,
            ast_sockaddr_stringify(&rtp->bind_address), x);
        if (ice_create(instance, &rtp->bind_address, x, 0)) {
            ast_log(LOG_NOTICE, "(%p) ICE failed to create session\n", instance);
        } else {
            rtp->ice_port = x;
            ast_sockaddr_copy(&rtp->ice_original_rtp_addr, &rtp->bind_address);
        }
    }
#endif
 
    return 0;
}

References ast_bind(), ast_cond_init, ast_debug_ice, ast_debug_rtp, ast_log, ast_random(), ast_rtp_instance_get_port_end(), ast_rtp_instance_get_port_start(), ast_rtp_instance_set_local_address(), ast_sockaddr_copy(), ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_test_suite_event_notify, ast_rtp::bind_address, create_new_socket(), errno, generate_random_string(), LOG_ERROR, LOG_NOTICE, LOG_WARNING, NULL, rtpend, rtpstart, ast_rtp::s, STRICT_RTP_CLOSED, STRICT_RTP_OPEN, ast_rtp::strict_rtp_state, and strictrtp.

Referenced by ast_rtp_bundle(), and ast_rtp_new().

rtp_deallocate_transport()

static void rtp_deallocate_transport ( struct ast_rtp_instance *  instance, struct ast_rtp *  rtp  )

static

Definition at line 4236 of file res_rtp_asterisk.c.

{
    int saved_rtp_s = rtp->s;
#ifdef HAVE_PJPROJECT
    struct timeval wait = ast_tvadd(ast_tvnow(), ast_samp2tv(TURN_STATE_WAIT_TIME, 1000));
    struct timespec ts = { .tv_sec = wait.tv_sec, .tv_nsec = wait.tv_usec * 1000, };
#endif
 
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
    ast_rtp_dtls_stop(instance);
#endif
 
    /* Close our own socket so we no longer get packets */
    if (rtp->s > -1) {
        close(rtp->s);
        rtp->s = -1;
    }
 
    /* Destroy RTCP if it was being used */
    if (rtp->rtcp && rtp->rtcp->s > -1) {
        if (saved_rtp_s != rtp->rtcp->s) {
            close(rtp->rtcp->s);
        }
        rtp->rtcp->s = -1;
    }
 
#ifdef HAVE_PJPROJECT
    pj_thread_register_check();
 
    /*
     * The instance lock is already held.
     *
     * Destroy the RTP TURN relay if being used
     */
    if (rtp->turn_rtp) {
        rtp->turn_state = PJ_TURN_STATE_NULL;
 
        /* Release the instance lock to avoid deadlock with PJPROJECT group lock */
        ao2_unlock(instance);
        pj_turn_sock_destroy(rtp->turn_rtp);
        ao2_lock(instance);
        while (rtp->turn_state != PJ_TURN_STATE_DESTROYING) {
            ast_cond_timedwait(&rtp->cond, ao2_object_get_lockaddr(instance), &ts);
        }
        rtp->turn_rtp = NULL;
    }
 
    /* Destroy the RTCP TURN relay if being used */
    if (rtp->turn_rtcp) {
        rtp->turn_state = PJ_TURN_STATE_NULL;
 
        /* Release the instance lock to avoid deadlock with PJPROJECT group lock */
        ao2_unlock(instance);
        pj_turn_sock_destroy(rtp->turn_rtcp);
        ao2_lock(instance);
        while (rtp->turn_state != PJ_TURN_STATE_DESTROYING) {
            ast_cond_timedwait(&rtp->cond, ao2_object_get_lockaddr(instance), &ts);
        }
        rtp->turn_rtcp = NULL;
    }
 
    ast_debug_ice(2, "(%p) ICE RTP transport deallocating\n", instance);
    /* Destroy any ICE session */
    ast_rtp_ice_stop(instance);
 
    /* Destroy any candidates */
    if (rtp->ice_local_candidates) {
        ao2_ref(rtp->ice_local_candidates, -1);
        rtp->ice_local_candidates = NULL;
    }
 
    if (rtp->ice_active_remote_candidates) {
        ao2_ref(rtp->ice_active_remote_candidates, -1);
        rtp->ice_active_remote_candidates = NULL;
    }
 
    if (rtp->ice_proposed_remote_candidates) {
        ao2_ref(rtp->ice_proposed_remote_candidates, -1);
        rtp->ice_proposed_remote_candidates = NULL;
    }
 
    if (rtp->ioqueue) {
        /*
         * We cannot hold the instance lock because we could wait
         * for the ioqueue thread to die and we might deadlock as
         * a result.
         */
        ao2_unlock(instance);
        rtp_ioqueue_thread_remove(rtp->ioqueue);
        ao2_lock(instance);
        rtp->ioqueue = NULL;
    }
#endif
}

References ao2_lock, ao2_object_get_lockaddr(), ao2_ref, ao2_unlock, ast_cond_timedwait, ast_debug_ice, ast_samp2tv(), ast_tvadd(), ast_tvnow(), NULL, ast_rtp::rtcp, ast_rtp::s, ast_rtcp::s, and TURN_STATE_WAIT_TIME.

Referenced by ast_rtp_bundle(), and ast_rtp_destroy().

rtp_debug_test_addr()

static int rtp_debug_test_addr ( struct ast_sockaddr *  addr )

inlinestatic

Definition at line 2838 of file res_rtp_asterisk.c.

{
    if (!ast_debug_rtp_packet_is_allowed) {
        return 0;
    }
    if (!ast_sockaddr_isnull(&rtpdebugaddr)) {
        if (rtpdebugport) {
            return (ast_sockaddr_cmp(&rtpdebugaddr, addr) == 0); /* look for RTP packets from IP+Port */
        } else {
            return (ast_sockaddr_cmp_addr(&rtpdebugaddr, addr) == 0); /* only look for RTP packets from IP */
        }
    }
 
    return 1;
}

References ast_debug_rtp_packet_is_allowed, ast_sockaddr_cmp(), ast_sockaddr_cmp_addr(), ast_sockaddr_isnull(), rtpdebugaddr, and rtpdebugport.

Referenced by ast_rtp_dtmf_begin(), ast_rtp_dtmf_continuation(), ast_rtp_dtmf_end_with_duration(), ast_rtp_read(), ast_rtp_sendcng(), bridge_p2p_rtp_write(), process_dtmf_rfc2833(), and rtp_raw_write().

rtp_do_debug_ip()

static char * rtp_do_debug_ip ( struct ast_cli_args *  a )

static

Definition at line 9955 of file res_rtp_asterisk.c.

{
    char *arg = ast_strdupa(a->argv[4]);
    char *debughost = NULL;
    char *debugport = NULL;
 
    if (!ast_sockaddr_parse(&rtpdebugaddr, arg, 0) || !ast_sockaddr_split_hostport(arg, &debughost, &debugport, 0)) {
        ast_cli(a->fd, "Lookup failed for '%s'\n", arg);
        return CLI_FAILURE;
    }
    rtpdebugport = (!ast_strlen_zero(debugport) && debugport[0] != '0');
    ast_cli(a->fd, "RTP Packet Debugging Enabled for address: %s\n",
        ast_sockaddr_stringify(&rtpdebugaddr));
    ast_debug_category_set_sublevel(AST_LOG_CATEGORY_RTP_PACKET, AST_LOG_CATEGORY_ENABLED);
    return CLI_SUCCESS;
}

References a, ast_cli(), ast_debug_category_set_sublevel(), AST_LOG_CATEGORY_ENABLED, AST_LOG_CATEGORY_RTP_PACKET, ast_sockaddr_parse(), ast_sockaddr_split_hostport(), ast_sockaddr_stringify(), ast_strdupa, ast_strlen_zero(), CLI_FAILURE, CLI_SUCCESS, NULL, rtpdebugaddr, and rtpdebugport.

Referenced by handle_cli_rtp_set_debug().

rtp_find_instance_by_media_source_ssrc()

static struct ast_rtp_instance * rtp_find_instance_by_media_source_ssrc ( struct ast_rtp_instance *  instance, struct ast_rtp *  rtp, unsigned int  ssrc  )

static

Precondition

instance is locked

Definition at line 6630 of file res_rtp_asterisk.c.

{
    return __rtp_find_instance_by_ssrc(instance, rtp, ssrc, 1);
}

References __rtp_find_instance_by_ssrc().

Referenced by ast_rtcp_interpret().

rtp_find_instance_by_packet_source_ssrc()

static struct ast_rtp_instance * rtp_find_instance_by_packet_source_ssrc ( struct ast_rtp_instance *  instance, struct ast_rtp *  rtp, unsigned int  ssrc  )

static

Precondition

instance is locked

Definition at line 6623 of file res_rtp_asterisk.c.

{
    return __rtp_find_instance_by_ssrc(instance, rtp, ssrc, 0);
}

References __rtp_find_instance_by_ssrc().

Referenced by ast_rtcp_interpret(), and ast_rtp_read().

rtp_instance_parse_extmap_extensions()

static void rtp_instance_parse_extmap_extensions ( struct ast_rtp_instance *  instance, struct ast_rtp *  rtp, unsigned char *  extension, int  len  )

static

Definition at line 7923 of file res_rtp_asterisk.c.

{
    int transport_wide_cc_id = ast_rtp_instance_extmap_get_id(instance, AST_RTP_EXTENSION_TRANSPORT_WIDE_CC);
    int pos = 0;
 
    /* We currently only care about the transport-cc extension, so if that's not negotiated then do nothing */
    if (transport_wide_cc_id == -1) {
        return;
    }
 
    /* Only while we do not exceed available extension data do we continue */
    while (pos < len) {
        int id = extension[pos] >> 4;
        int extension_len = (extension[pos] & 0xF) + 1;
 
        /* We've handled the first byte as it contains the extension id and length, so always
         * skip ahead now
         */
        pos += 1;
 
        if (id == 0) {
            /* From the RFC:
             * In both forms, padding bytes have the value of 0 (zero).  They may be
             * placed between extension elements, if desired for alignment, or after
             * the last extension element, if needed for padding.  A padding byte
             * does not supply the ID of an element, nor the length field.  When a
             * padding byte is found, it is ignored and the parser moves on to
             * interpreting the next byte.
             */
            continue;
        } else if (id == 15) {
            /* From the RFC:
             * The local identifier value 15 is reserved for future extension and
             * MUST NOT be used as an identifier.  If the ID value 15 is
             * encountered, its length field should be ignored, processing of the
             * entire extension should terminate at that point, and only the
             * extension elements present prior to the element with ID 15
             * considered.
             */
            break;
        } else if ((pos + extension_len) > len) {
            /* The extension is corrupted and is stating that it contains more data than is
             * available in the extensions data.
             */
            break;
        }
 
        /* If this is transport-cc then we need to parse it further */
        if (id == transport_wide_cc_id) {
            rtp_instance_parse_transport_wide_cc(instance, rtp, extension + pos, extension_len);
        }
 
        /* Skip ahead to the next extension */
        pos += extension_len;
    }
}

References AST_RTP_EXTENSION_TRANSPORT_WIDE_CC, ast_rtp_instance_extmap_get_id(), len(), and rtp_instance_parse_transport_wide_cc().

Referenced by ast_rtp_interpret().

rtp_instance_parse_transport_wide_cc()

static void rtp_instance_parse_transport_wide_cc ( struct ast_rtp_instance *  instance, struct ast_rtp *  rtp, unsigned char *  data, int  len  )

static

Definition at line 7868 of file res_rtp_asterisk.c.

{
    uint16_t *seqno = (uint16_t *)data;
    struct rtp_transport_wide_cc_packet_statistics statistics;
    struct ast_rtp_instance *transport = rtp->bundled ? rtp->bundled : instance;
    struct ast_rtp *transport_rtp = ast_rtp_instance_get_data(transport);
 
    /* If the sequence number has cycled over then record it as such */
    if (((int)transport_rtp->transport_wide_cc.last_seqno - (int)ntohs(*seqno)) > 100) {
        transport_rtp->transport_wide_cc.cycles += RTP_SEQ_MOD;
    }
 
    /* Populate the statistics information for this packet */
    statistics.seqno = transport_rtp->transport_wide_cc.cycles + ntohs(*seqno);
    statistics.received = ast_tvnow();
 
    /* We allow at a maximum 1000 packet statistics in play at a time, if we hit the
     * limit we give up and start fresh.
     */
    if (AST_VECTOR_SIZE(&transport_rtp->transport_wide_cc.packet_statistics) > 1000) {
        AST_VECTOR_RESET(&rtp->transport_wide_cc.packet_statistics, AST_VECTOR_ELEM_CLEANUP_NOOP);
    }
 
    if (!AST_VECTOR_SIZE(&transport_rtp->transport_wide_cc.packet_statistics) ||
        statistics.seqno > transport_rtp->transport_wide_cc.last_extended_seqno) {
        /* This is the expected path */
        if (AST_VECTOR_APPEND(&transport_rtp->transport_wide_cc.packet_statistics, statistics)) {
            return;
        }
 
        transport_rtp->transport_wide_cc.last_extended_seqno = statistics.seqno;
        transport_rtp->transport_wide_cc.last_seqno = ntohs(*seqno);
    } else {
        /* This packet was out of order, so reorder it within the vector accordingly */
        if (AST_VECTOR_ADD_SORTED(&transport_rtp->transport_wide_cc.packet_statistics, statistics,
            rtp_transport_wide_cc_packet_statistics_cmp)) {
            return;
        }
    }
 
    /* If we have not yet scheduled the periodic sending of feedback for this transport then do so */
    if (transport_rtp->transport_wide_cc.schedid < 0 && transport_rtp->rtcp) {
        ast_debug_rtcp(1, "(%p) RTCP starting transport-cc feedback transmission on RTP instance '%p'\n", instance, transport);
        ao2_ref(transport, +1);
        transport_rtp->transport_wide_cc.schedid = ast_sched_add(rtp->sched, 1000,
            rtp_transport_wide_cc_feedback_produce, transport);
        if (transport_rtp->transport_wide_cc.schedid < 0) {
            ao2_ref(transport, -1);
            ast_log(LOG_WARNING, "Scheduling RTCP transport-cc feedback transmission failed on RTP instance '%p'\n",
                transport);
        }
    }
}

References ao2_ref, ast_debug_rtcp, ast_log, ast_rtp_instance_get_data(), ast_sched_add(), ast_tvnow(), AST_VECTOR_ADD_SORTED, AST_VECTOR_APPEND, AST_VECTOR_ELEM_CLEANUP_NOOP, AST_VECTOR_RESET, AST_VECTOR_SIZE, ast_rtp::bundled, rtp_transport_wide_cc_statistics::cycles, rtp_transport_wide_cc_statistics::last_extended_seqno, rtp_transport_wide_cc_statistics::last_seqno, LOG_WARNING, rtp_transport_wide_cc_statistics::packet_statistics, ast_rtp::rtcp, RTP_SEQ_MOD, rtp_transport_wide_cc_feedback_produce(), rtp_transport_wide_cc_packet_statistics_cmp(), ast_rtp::sched, rtp_transport_wide_cc_statistics::schedid, rtp_transport_wide_cc_packet_statistics::seqno, ast_rtp::seqno, statistics(), and ast_rtp::transport_wide_cc.

Referenced by rtp_instance_parse_extmap_extensions().

rtp_instance_unlock()

static void rtp_instance_unlock ( struct ast_rtp_instance *  instance )

static

Definition at line 7598 of file res_rtp_asterisk.c.

{
    if (instance) {
        ao2_unlock(instance);
    }
}

References ao2_unlock.

Referenced by ast_rtp_read().

rtp_learning_rtp_seq_update()

static int rtp_learning_rtp_seq_update ( struct rtp_learning_info *  info, uint16_t  seq  )

static

Definition at line 3652 of file res_rtp_asterisk.c.

{
    if (seq == (uint16_t) (info->max_seq + 1)) {
        /* packet is in sequence */
        info->packets--;
    } else {
        /* Sequence discontinuity; reset */
        info->packets = learning_min_sequential - 1;
        info->received = ast_tvnow();
    }
 
    /* Only check time if strictrtp is set to yes. Otherwise, we only needed to check seqno */
    if (strictrtp == STRICT_RTP_YES) {
        switch (info->stream_type) {
        case AST_MEDIA_TYPE_UNKNOWN:
        case AST_MEDIA_TYPE_AUDIO:
            /*
             * Protect against packet floods by checking that we
             * received the packet sequence in at least the minimum
             * allowed time.
             */
            if (ast_tvzero(info->received)) {
                info->received = ast_tvnow();
            } else if (!info->packets
                && ast_tvdiff_ms(ast_tvnow(), info->received) < learning_min_duration) {
                /* Packet flood; reset */
                info->packets = learning_min_sequential - 1;
                info->received = ast_tvnow();
            }
            break;
        case AST_MEDIA_TYPE_VIDEO:
        case AST_MEDIA_TYPE_IMAGE:
        case AST_MEDIA_TYPE_TEXT:
        case AST_MEDIA_TYPE_END:
            break;
        }
    }
 
    info->max_seq = seq;
 
    return info->packets;
}

References AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_END, AST_MEDIA_TYPE_IMAGE, AST_MEDIA_TYPE_TEXT, AST_MEDIA_TYPE_UNKNOWN, AST_MEDIA_TYPE_VIDEO, ast_tvdiff_ms(), ast_tvnow(), ast_tvzero(), learning_min_duration, learning_min_sequential, seq, STRICT_RTP_YES, and strictrtp.

Referenced by ast_rtp_read().

rtp_learning_seq_init()

static void rtp_learning_seq_init ( struct rtp_learning_info *  info, uint16_t  seq  )

static

Definition at line 3635 of file res_rtp_asterisk.c.

{
    info->max_seq = seq;
    info->packets = learning_min_sequential;
    memset(&info->received, 0, sizeof(info->received));
}

References learning_min_sequential, and seq.

Referenced by ast_rtp_read(), and rtp_learning_start().

rtp_learning_start()

static void rtp_learning_start ( struct ast_rtp *  rtp )

static

Start the strictrtp learning mode.

Parameters

rtp	RTP session description

Definition at line 3700 of file res_rtp_asterisk.c.

{
    rtp->strict_rtp_state = STRICT_RTP_LEARN;
    memset(&rtp->rtp_source_learn.proposed_address, 0,
        sizeof(rtp->rtp_source_learn.proposed_address));
    rtp->rtp_source_learn.start = ast_tvnow();
    rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t) rtp->lastrxseqno);
}

References ast_tvnow(), ast_rtp::lastrxseqno, rtp_learning_info::proposed_address, rtp_learning_seq_init(), ast_rtp::rtp_source_learn, rtp_learning_info::start, STRICT_RTP_LEARN, and ast_rtp::strict_rtp_state.

Referenced by ast_rtp_remote_address_set().

rtp_raw_write()

static int rtp_raw_write ( struct ast_rtp_instance *  instance, struct ast_frame *  frame, int  codec  )

static

Precondition

instance is locked

Definition at line 5283 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    int pred, mark = 0;
    unsigned int ms;
    struct ast_sockaddr remote_address = { {0,} };
    int rate = ast_rtp_get_rate(frame->subclass.format) / 1000;
    unsigned int seqno;
#ifdef TEST_FRAMEWORK
    struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
#endif
 
    if (ast_format_cmp(frame->subclass.format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) {
        frame->samples /= 2;
    }
 
    if (rtp->sending_digit) {
        return 0;
    }
 
    /* Suppressed audio must not consume elapsed transmit time. */
    ms = calc_txstamp(rtp, &frame->delivery);
 
#ifdef TEST_FRAMEWORK
    if (test && test->send_report) {
        test->send_report = 0;
        ast_rtcp_write(instance);
        return 0;
    }
#endif
 
    if (frame->frametype == AST_FRAME_VOICE) {
        pred = rtp->lastts + frame->samples;
 
        /* Re-calculate last TS */
        rtp->lastts = rtp->lastts + ms * rate;
        if (ast_tvzero(frame->delivery)) {
            /* If this isn't an absolute delivery time, Check if it is close to our prediction,
               and if so, go with our prediction */
            if (abs((int)rtp->lastts - pred) < MAX_TIMESTAMP_SKEW) {
                rtp->lastts = pred;
            } else {
                ast_debug_rtp(3, "(%p) RTP audio difference is %d, ms is %u\n",
                    instance, abs((int)rtp->lastts - pred), ms);
                mark = 1;
            }
        }
    } else if (frame->frametype == AST_FRAME_VIDEO) {
        mark = frame->subclass.frame_ending;
        pred = rtp->lastovidtimestamp + frame->samples;
        /* Re-calculate last TS */
        rtp->lastts = rtp->lastts + ms * 90;
        /* If it's close to our prediction, go for it */
        if (ast_tvzero(frame->delivery)) {
            if (abs((int)rtp->lastts - pred) < 7200) {
                rtp->lastts = pred;
                rtp->lastovidtimestamp += frame->samples;
            } else {
                ast_debug_rtp(3, "(%p) RTP video difference is %d, ms is %u (%u), pred/ts/samples %u/%d/%d\n",
                    instance, abs((int)rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, frame->samples);
                rtp->lastovidtimestamp = rtp->lastts;
            }
        }
    } else {
        pred = rtp->lastotexttimestamp + frame->samples;
        /* Re-calculate last TS */
        rtp->lastts = rtp->lastts + ms;
        /* If it's close to our prediction, go for it */
        if (ast_tvzero(frame->delivery)) {
            if (abs((int)rtp->lastts - pred) < 7200) {
                rtp->lastts = pred;
                rtp->lastotexttimestamp += frame->samples;
            } else {
                ast_debug_rtp(3, "(%p) RTP other difference is %d, ms is %u, pred/ts/samples %u/%d/%d\n",
                    instance, abs((int)rtp->lastts - pred), ms, rtp->lastts, pred, frame->samples);
                rtp->lastotexttimestamp = rtp->lastts;
            }
        }
    }
 
    /* If we have been explicitly told to set the marker bit then do so */
    if (ast_test_flag(rtp, FLAG_NEED_MARKER_BIT)) {
        mark = 1;
        ast_clear_flag(rtp, FLAG_NEED_MARKER_BIT);
    }
 
    /* If the timestamp for non-digt packets has moved beyond the timestamp for digits, update the digit timestamp */
    if (rtp->lastts > rtp->lastdigitts) {
        rtp->lastdigitts = rtp->lastts;
    }
 
    /* Assume that the sequence number we expect to use is what will be used until proven otherwise */
    seqno = rtp->seqno;
 
    /* If the frame contains sequence number information use it to influence our sequence number */
    if (ast_test_flag(frame, AST_FRFLAG_HAS_SEQUENCE_NUMBER)) {
        if (rtp->expectedseqno != -1) {
            /* Determine where the frame from the core is in relation to where we expected */
            int difference = frame->seqno - rtp->expectedseqno;
 
            /* If there is a substantial difference then we've either got packets really out
             * of order, or the source is RTP and it has cycled. If this happens we resync
             * the sequence number adjustments to this frame. If we also have packet loss
             * things won't be reflected correctly but it will sort itself out after a bit.
             */
            if (abs(difference) > 100) {
                difference = 0;
            }
 
            /* Adjust the sequence number being used for this packet accordingly */
            seqno += difference;
 
            if (difference >= 0) {
                /* This frame is on time or in the future */
                rtp->expectedseqno = frame->seqno + 1;
                rtp->seqno += difference;
            }
        } else {
            /* This is the first frame with sequence number we've seen, so start keeping track */
            rtp->expectedseqno = frame->seqno + 1;
        }
    } else {
        rtp->expectedseqno = -1;
    }
 
    if (ast_test_flag(frame, AST_FRFLAG_HAS_TIMING_INFO)) {
        rtp->lastts = frame->ts * rate;
    }
 
    ast_rtp_instance_get_remote_address(instance, &remote_address);
 
    /* If we know the remote address construct a packet and send it out */
    if (!ast_sockaddr_isnull(&remote_address)) {
        int hdrlen = 12;
        int res;
        int ice;
        int ext = 0;
        int abs_send_time_id;
        int packet_len;
        unsigned char *rtpheader;
 
        /* If the abs-send-time extension has been negotiated determine how much space we need */
        abs_send_time_id = ast_rtp_instance_extmap_get_id(instance, AST_RTP_EXTENSION_ABS_SEND_TIME);
        if (abs_send_time_id != -1) {
            /* 4 bytes for the shared information, 1 byte for identifier, 3 bytes for abs-send-time */
            hdrlen += 8;
            ext = 1;
        }
 
        packet_len = frame->datalen + hdrlen;
        rtpheader = (unsigned char *)(frame->data.ptr - hdrlen);
 
        put_unaligned_uint32(rtpheader, htonl((2 << 30) | (ext << 28) | (codec << 16) | (seqno) | (mark << 23)));
        put_unaligned_uint32(rtpheader + 4, htonl(rtp->lastts));
        put_unaligned_uint32(rtpheader + 8, htonl(rtp->ssrc));
 
        /* We assume right now that we will only ever have the abs-send-time extension in the packet
         * which simplifies things a bit.
         */
        if (abs_send_time_id != -1) {
            unsigned int now_msw;
            unsigned int now_lsw;
 
            /* This happens before being placed into the retransmission buffer so that when we
             * retransmit we only have to update the timestamp, not everything else.
             */
            put_unaligned_uint32(rtpheader + 12, htonl((0xBEDE << 16) | 1));
            rtpheader[16] = (abs_send_time_id << 4) | 2;
 
            timeval2ntp(ast_tvnow(), &now_msw, &now_lsw);
            put_unaligned_time24(rtpheader + 17, now_msw, now_lsw);
        }
 
        /* If retransmissions are enabled, we need to store this packet for future use */
        if (rtp->send_buffer) {
            struct ast_rtp_rtcp_nack_payload *payload;
 
            payload = ast_malloc(sizeof(*payload) + packet_len);
            if (payload) {
                payload->size = packet_len;
                memcpy(payload->buf, rtpheader, packet_len);
                if (ast_data_buffer_put(rtp->send_buffer, rtp->seqno, payload) == -1) {
                    ast_free(payload);
                }
            }
        }
 
        res = rtp_sendto(instance, (void *)rtpheader, packet_len, 0, &remote_address, &ice);
        if (res < 0) {
            if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT) || (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT) && (ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
                ast_debug_rtp(1, "(%p) RTP transmission error of packet %d to %s: %s\n",
                      instance, rtp->seqno,
                      ast_sockaddr_stringify(&remote_address),
                      strerror(errno));
            } else if (((ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || ast_debug_rtp_packet_is_allowed) && !ast_test_flag(rtp, FLAG_NAT_INACTIVE_NOWARN)) {
                /* Only give this error message once if we are not RTP debugging */
                if (ast_debug_rtp_packet_is_allowed)
                    ast_debug(0, "(%p) RTP NAT: Can't write RTP to private address %s, waiting for other end to send audio...\n",
                        instance, ast_sockaddr_stringify(&remote_address));
                ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN);
            }
        } else {
            if (rtp->rtcp && rtp->rtcp->schedid < 0) {
                ast_debug_rtcp(2, "(%s) RTCP starting transmission in %u ms\n",
                    ast_rtp_instance_get_channel_id(instance), ast_rtcp_calc_interval(rtp));
                ao2_ref(instance, +1);
                rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, instance);
                if (rtp->rtcp->schedid < 0) {
                    ao2_ref(instance, -1);
                    ast_log(LOG_WARNING, "scheduling RTCP transmission failed.\n");
                }
            }
        }
 
        if (rtp_debug_test_addr(&remote_address)) {
            ast_verbose("Sent RTP packet to      %s%s (type %-2.2d, seq %-6.6d, ts %-6.6u, len %-6.6d)\n",
                    ast_sockaddr_stringify(&remote_address),
                    ice ? " (via ICE)" : "",
                    codec, rtp->seqno, rtp->lastts, res - hdrlen);
        }
    }
 
    /* If the sequence number that has been used doesn't match what we expected then this is an out of
     * order late packet, so we don't need to increment as we haven't yet gotten the expected frame from
     * the core.
     */
    if (seqno == rtp->seqno) {
        rtp->seqno++;
    }
 
    return 0;
}

References abs, ao2_ref, ast_clear_flag, ast_data_buffer_put(), ast_debug, ast_debug_rtcp, ast_debug_rtp, ast_debug_rtp_packet_is_allowed, ast_format_cmp(), AST_FORMAT_CMP_EQUAL, ast_format_g722, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_free, AST_FRFLAG_HAS_SEQUENCE_NUMBER, AST_FRFLAG_HAS_TIMING_INFO, ast_log, ast_malloc, ast_rtcp_calc_interval(), ast_rtcp_write(), AST_RTP_EXTENSION_ABS_SEND_TIME, ast_rtp_get_rate(), ast_rtp_instance_extmap_get_id(), ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_data(), ast_rtp_instance_get_prop(), ast_rtp_instance_get_remote_address, AST_RTP_PROPERTY_NAT, ast_sched_add(), ast_set_flag, ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_test_flag, ast_tvnow(), ast_tvzero(), ast_verbose, ast_rtp_rtcp_nack_payload::buf, calc_txstamp(), ast_frame::data, ast_frame::datalen, ast_frame::delivery, errno, ast_rtp::expectedseqno, ext, FLAG_NAT_ACTIVE, FLAG_NAT_INACTIVE, FLAG_NAT_INACTIVE_NOWARN, FLAG_NEED_MARKER_BIT, ast_frame_subclass::format, ast_frame_subclass::frame_ending, ast_frame::frametype, ast_rtp::lastdigitts, ast_rtp::lastotexttimestamp, ast_rtp::lastovidtimestamp, ast_rtp::lastts, LOG_WARNING, MAX_TIMESTAMP_SKEW, ast_frame::ptr, put_unaligned_time24(), put_unaligned_uint32(), ast_rtp::rtcp, rtp_debug_test_addr(), rtp_sendto(), ast_frame::samples, ast_rtp::sched, ast_rtcp::schedid, ast_rtp::send_buffer, ast_rtp::sending_digit, ast_frame::seqno, ast_rtp::seqno, ast_rtp_rtcp_nack_payload::size, ast_rtp::ssrc, ast_frame::subclass, timeval2ntp(), and ast_frame::ts.

Referenced by ast_rtp_write().

rtp_recvfrom()

static int rtp_recvfrom ( struct ast_rtp_instance *  instance, void *  buf, size_t  size, int  flags, struct ast_sockaddr *  sa  )

static

Precondition

instance is locked

Definition at line 3459 of file res_rtp_asterisk.c.

{
    return __rtp_recvfrom(instance, buf, size, flags, sa, 0);
}

References __rtp_recvfrom(), and buf.

Referenced by ast_rtp_read().

rtp_red_buffer()

static int rtp_red_buffer ( struct ast_rtp_instance *  instance, struct ast_frame *  frame  )

static

Precondition

instance is locked

Warning

This code was written many years ago and it's unclear why we actually buffer OUTGOING T.140 text frames until a command is encountered but we do.

This may change in the future.

Definition at line 9395 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    struct rtp_red *red = rtp->red;
 
    if (!red) {
        return 0;
    }
 
    if (frame->datalen > 0) {
        int space_available = 0;
 
        if (red->t140.datalen > 0) {
            const unsigned char *primary = red->buf_data;
 
            /* There is something already in the T.140 buffer */
            if (primary[0] == 0x08 || primary[0] == 0x0a || primary[0] == 0x0d) {
                /* Flush the previous T.140 packet if it is a command */
                ast_rtp_write(instance, &rtp->red->t140);
            } else {
                primary = frame->data.ptr;
                if (primary[0] == 0x08 || primary[0] == 0x0a || primary[0] == 0x0d) {
                    /* Flush the previous T.140 packet if we are buffering a command now */
                    ast_rtp_write(instance, &rtp->red->t140);
                }
            }
        }
 
        /*
         * RED generation payload sizes are limited to 255 (UCHAR_MAX) bytes by virtue of
         * red->len being an array of usigned chars.  If adding the current frame will
         * exceed that, we're going to flush the saved frame then try again. If the new
         * frame fits, great otherwise we're going to toss it.  Without understanding
         * the purpose of the buffering, that's all we can do now.
         */
        space_available = UCHAR_MAX - red->t140.datalen;
 
        if (frame->datalen > space_available) {
            ast_rtp_write(instance, &rtp->red->t140);
            /*
             * ast_rtp_write() calls red_t140_to_red() which resets red->t140.datalen
             * back to 0 so we now have UCHAR_MAX space available.
             */
            space_available = UCHAR_MAX;
        }
 
        if (frame->datalen > space_available) {
            ast_log(LOG_WARNING, "%s: T.140 frame of %d bytes exceeds max of %u. Discarding.\n",
                ast_rtp_instance_get_channel_id(instance),
                frame->datalen, UCHAR_MAX);
            return -1;
        }
 
        memcpy(&red->buf_data[red->t140.datalen], frame->data.ptr, frame->datalen);
        red->t140.datalen += frame->datalen;
        red->t140.ts = frame->ts;
    }
 
    return 0;
}

References ast_log, ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_data(), ast_rtp_write(), rtp_red::buf_data, ast_frame::data, ast_frame::datalen, LOG_WARNING, ast_frame::ptr, ast_rtp::red, rtp_red::t140, and ast_frame::ts.

rtp_red_init()

static int rtp_red_init ( struct ast_rtp_instance *  instance, int  buffer_time, int *  payloads, int  generations  )

static

Precondition

instance is locked

Definition at line 9357 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    int x;
 
    rtp->red = ast_calloc(1, sizeof(*rtp->red));
    if (!rtp->red) {
        return -1;
    }
 
    rtp->red->t140.frametype = AST_FRAME_TEXT;
    rtp->red->t140.subclass.format = ast_format_t140_red;
    rtp->red->t140.data.ptr = &rtp->red->buf_data;
 
    rtp->red->t140red = rtp->red->t140;
    rtp->red->t140red.data.ptr = &rtp->red->t140red_data;
 
    rtp->red->num_gen = generations;
    rtp->red->hdrlen = generations * 4 + 1;
 
    for (x = 0; x < generations; x++) {
        rtp->red->pt[x] = payloads[x];
        rtp->red->pt[x] |= 1 << 7; /* mark redundant generations pt */
        rtp->red->t140red_data[x*4] = rtp->red->pt[x];
    }
    rtp->red->t140red_data[x*4] = rtp->red->pt[x] = payloads[x]; /* primary pt */
    rtp->red->schedid = ast_sched_add(rtp->sched, buffer_time, red_write, instance);
 
    return 0;
}

References ast_calloc, ast_format_t140_red, AST_FRAME_TEXT, ast_rtp_instance_get_data(), ast_sched_add(), rtp_red::buf_data, ast_frame::data, ast_frame_subclass::format, ast_frame::frametype, rtp_red::hdrlen, rtp_red::num_gen, rtp_red::pt, ast_frame::ptr, ast_rtp::red, red_write(), ast_rtp::sched, rtp_red::schedid, ast_frame::subclass, rtp_red::t140, rtp_red::t140red, and rtp_red::t140red_data.

rtp_reload()

static int rtp_reload ( int  reload, int  by_external_config  )

static

This resource is not "reloaded" so much as unloaded and loaded again. In the case of the TURN related variables, the memory referenced by a previously loaded instance should have been released when the corresponding pool was destroyed. If at some point in the future this resource were to support ACTUAL live reconfiguration and did NOT release the pool this will cause a small memory leak.

Definition at line 10343 of file res_rtp_asterisk.c.

{
    struct ast_config *cfg;
    const char *s;
    struct ast_flags config_flags = { (reload && !by_external_config) ? CONFIG_FLAG_FILEUNCHANGED : 0 };
 
#ifdef HAVE_PJPROJECT
    struct ast_variable *var;
    struct ast_ice_host_candidate *candidate;
    int acl_subscription_flag = 0;
#endif
 
    cfg = ast_config_load2("rtp.conf", "rtp", config_flags);
    if (!cfg || cfg == CONFIG_STATUS_FILEUNCHANGED || cfg == CONFIG_STATUS_FILEINVALID) {
        return 0;
    }
 
#ifdef SO_NO_CHECK
    nochecksums = 0;
#endif
 
    rtpstart = DEFAULT_RTP_START;
    rtpend = DEFAULT_RTP_END;
    rtcpinterval = RTCP_DEFAULT_INTERVALMS;
    dtmftimeout = DEFAULT_DTMF_TIMEOUT;
    strictrtp = DEFAULT_STRICT_RTP;
    learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL;
    learning_min_duration = DEFAULT_LEARNING_MIN_DURATION;
    srtp_replay_protection = DEFAULT_SRTP_REPLAY_PROTECTION;
 
    /** This resource is not "reloaded" so much as unloaded and loaded again.
     * In the case of the TURN related variables, the memory referenced by a
     * previously loaded instance  *should* have been released when the
     * corresponding pool was destroyed. If at some point in the future this
     * resource were to support ACTUAL live reconfiguration and did NOT release
     * the pool this will cause a small memory leak.
     */
 
#ifdef HAVE_PJPROJECT
    icesupport = DEFAULT_ICESUPPORT;
    stun_software_attribute = DEFAULT_STUN_SOFTWARE_ATTRIBUTE;
    turnport = DEFAULT_TURN_PORT;
    clean_stunaddr();
    turnaddr = pj_str(NULL);
    turnusername = pj_str(NULL);
    turnpassword = pj_str(NULL);
    host_candidate_overrides_clear();
#endif
 
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
    dtls_mtu = DEFAULT_DTLS_MTU;
#endif
 
    if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
        rtpstart = atoi(s);
        if (rtpstart < MINIMUM_RTP_PORT)
            rtpstart = MINIMUM_RTP_PORT;
        if (rtpstart > MAXIMUM_RTP_PORT)
            rtpstart = MAXIMUM_RTP_PORT;
    }
    if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
        rtpend = atoi(s);
        if (rtpend < MINIMUM_RTP_PORT)
            rtpend = MINIMUM_RTP_PORT;
        if (rtpend > MAXIMUM_RTP_PORT)
            rtpend = MAXIMUM_RTP_PORT;
    }
    if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
        rtcpinterval = atoi(s);
        if (rtcpinterval == 0)
            rtcpinterval = 0; /* Just so we're clear... it's zero */
        if (rtcpinterval < RTCP_MIN_INTERVALMS)
            rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
        if (rtcpinterval > RTCP_MAX_INTERVALMS)
            rtcpinterval = RTCP_MAX_INTERVALMS;
    }
    if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
#ifdef SO_NO_CHECK
        nochecksums = ast_false(s) ? 1 : 0;
#else
        if (ast_false(s))
            ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
#endif
    }
    if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
        dtmftimeout = atoi(s);
        if ((dtmftimeout < 0) || (dtmftimeout > 64000)) {
            ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
                dtmftimeout, DEFAULT_DTMF_TIMEOUT);
            dtmftimeout = DEFAULT_DTMF_TIMEOUT;
        };
    }
    if ((s = ast_variable_retrieve(cfg, "general", "strictrtp"))) {
        if (ast_true(s)) {
            strictrtp = STRICT_RTP_YES;
        } else if (!strcasecmp(s, "seqno")) {
            strictrtp = STRICT_RTP_SEQNO;
        } else {
            strictrtp = STRICT_RTP_NO;
        }
    }
    if ((s = ast_variable_retrieve(cfg, "general", "probation"))) {
        if ((sscanf(s, "%d", &learning_min_sequential) != 1) || learning_min_sequential <= 1) {
            ast_log(LOG_WARNING, "Value for 'probation' could not be read, using default of '%d' instead\n",
                DEFAULT_LEARNING_MIN_SEQUENTIAL);
            learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL;
        }
        learning_min_duration = CALC_LEARNING_MIN_DURATION(learning_min_sequential);
    }
    if ((s = ast_variable_retrieve(cfg, "general", "srtpreplayprotection"))) {
        srtp_replay_protection = ast_true(s);
    }
#ifdef HAVE_PJPROJECT
    if ((s = ast_variable_retrieve(cfg, "general", "icesupport"))) {
        icesupport = ast_true(s);
    }
    if ((s = ast_variable_retrieve(cfg, "general", "stun_software_attribute"))) {
        stun_software_attribute = ast_true(s);
    }
    if ((s = ast_variable_retrieve(cfg, "general", "stunaddr_reresolve_ttl_0"))) {
        stunaddr_reresolve_ttl_0 = ast_true(s);
    }
    if ((s = ast_variable_retrieve(cfg, "general", "stunaddr"))) {
        char *hostport, *host, *port;
        unsigned int port_parsed = STANDARD_STUN_PORT;
        struct ast_sockaddr stunaddr_parsed;
 
        hostport = ast_strdupa(s);
 
        if (!ast_parse_arg(hostport, PARSE_ADDR, &stunaddr_parsed)) {
            ast_debug_stun(3, "stunaddr = '%s' does not need name resolution\n",
                ast_sockaddr_stringify_host(&stunaddr_parsed));
            if (!ast_sockaddr_port(&stunaddr_parsed)) {
                ast_sockaddr_set_port(&stunaddr_parsed, STANDARD_STUN_PORT);
            }
            ast_rwlock_wrlock(&stunaddr_lock);
            ast_sockaddr_to_sin(&stunaddr_parsed, &stunaddr);
            /* Set stunaddr_ttl = -1 to indicate no resolution required in the future */
            stunaddr_ttl = -1;
            ast_rwlock_unlock(&stunaddr_lock);
        } else if (ast_sockaddr_split_hostport(hostport, &host, &port, 0)) {
            if (port) {
                ast_parse_arg(port, PARSE_UINT32|PARSE_IN_RANGE, &port_parsed, 1, 65535);
            }
 
            ast_rwlock_wrlock(&stunaddr_lock);
 
            stunaddr.sin_port = htons(port_parsed);
            ast_free(stun_hostname);
            stun_hostname = ast_strdup(host);
            if (!stun_hostname) {
                ast_log(LOG_ERROR, "Failed to set stun_hostname from '%s'", host);
            } else {
                stunaddr_resolver = ast_dns_resolve_recurring(host, T_A, C_IN,
                    &stunaddr_resolve_callback, NULL);
                if (!stunaddr_resolver) {
                    ast_log(LOG_ERROR, "Failed to setup recurring DNS resolution of stunaddr '%s'",
                        host);
                } else {
                    ast_debug_stun(2, "Attemping to start recurring stun hostname resolution for '%s'\n", stun_hostname);
                }
                /*
                 * Set stunaddr_ttl = 0 to indicate resolution is required.
                 * If a later query returns a positive ttl, great.  We'll use the results
                 * of the last query until it expires.  If it returns 0, we'll resolve
                 * every time we need it.
                 */
                stunaddr_ttl = 0;
            }
            ast_rwlock_unlock(&stunaddr_lock);
 
 
        } else {
            ast_log(LOG_ERROR, "Failed to parse stunaddr '%s'", hostport);
        }
    }
    if ((s = ast_variable_retrieve(cfg, "general", "turnaddr"))) {
        struct sockaddr_in addr;
        addr.sin_port = htons(DEFAULT_TURN_PORT);
        if (ast_parse_arg(s, PARSE_INADDR, &addr)) {
            ast_log(LOG_WARNING, "Invalid TURN server address: %s\n", s);
        } else {
            pj_strdup2_with_null(pool, &turnaddr, ast_inet_ntoa(addr.sin_addr));
            /* ntohs() is not a bug here. The port number is used in host byte order with
             * a pjnat API. */
            turnport = ntohs(addr.sin_port);
        }
    }
    if ((s = ast_variable_retrieve(cfg, "general", "turnusername"))) {
        pj_strdup2_with_null(pool, &turnusername, s);
    }
    if ((s = ast_variable_retrieve(cfg, "general", "turnpassword"))) {
        pj_strdup2_with_null(pool, &turnpassword, s);
    }
 
    AST_RWLIST_WRLOCK(&host_candidates);
    for (var = ast_variable_browse(cfg, "ice_host_candidates"); var; var = var->next) {
        struct ast_sockaddr local_addr, advertised_addr;
        unsigned int include_local_address = 0;
        char *sep;
 
        ast_sockaddr_setnull(&local_addr);
        ast_sockaddr_setnull(&advertised_addr);
 
        if (ast_parse_arg(var->name, PARSE_ADDR | PARSE_PORT_IGNORE, &local_addr)) {
            ast_log(LOG_WARNING, "Invalid local ICE host address: %s\n", var->name);
            continue;
        }
 
        sep = strchr((char *)var->value,',');
        if (sep) {
            *sep = '\0';
            sep++;
            sep = ast_skip_blanks(sep);
            include_local_address = strcmp(sep, "include_local_address") == 0;
        }
 
        if (ast_parse_arg(var->value, PARSE_ADDR | PARSE_PORT_IGNORE, &advertised_addr)) {
            ast_log(LOG_WARNING, "Invalid advertised ICE host address: %s\n", var->value);
            continue;
        }
 
        if (!(candidate = ast_calloc(1, sizeof(*candidate)))) {
            ast_log(LOG_ERROR, "Failed to allocate ICE host candidate mapping.\n");
            break;
        }
 
        candidate->include_local = include_local_address;
 
        ast_sockaddr_copy(&candidate->local, &local_addr);
        ast_sockaddr_copy(&candidate->advertised, &advertised_addr);
 
        AST_RWLIST_INSERT_TAIL(&host_candidates, candidate, next);
    }
    AST_RWLIST_UNLOCK(&host_candidates);
 
    ast_rwlock_wrlock(&ice_acl_lock);
    ast_rwlock_wrlock(&stun_acl_lock);
 
    ice_acl = ast_free_acl_list(ice_acl);
    stun_acl = ast_free_acl_list(stun_acl);
 
    for (var = ast_variable_browse(cfg, "general"); var; var = var->next) {
        const char* sense = NULL;
        struct ast_acl_list **acl = NULL;
        if (strncasecmp(var->name, "ice_", 4) == 0) {
            sense = var->name + 4;
            acl = &ice_acl;
        } else if (strncasecmp(var->name, "stun_", 5) == 0) {
            sense = var->name + 5;
            acl = &stun_acl;
        } else {
            continue;
        }
 
        if (strcasecmp(sense, "blacklist") == 0) {
            sense = "deny";
        }
 
        if (strcasecmp(sense, "acl") && strcasecmp(sense, "permit") && strcasecmp(sense, "deny")) {
            continue;
        }
 
        ast_append_acl(sense, var->value, acl, NULL, &acl_subscription_flag);
    }
    ast_rwlock_unlock(&ice_acl_lock);
    ast_rwlock_unlock(&stun_acl_lock);
 
    if (acl_subscription_flag && !acl_change_sub) {
        acl_change_sub = stasis_subscribe(ast_security_topic(), acl_change_stasis_cb, NULL);
        stasis_subscription_accept_message_type(acl_change_sub, ast_named_acl_change_type());
        stasis_subscription_set_filter(acl_change_sub, STASIS_SUBSCRIPTION_FILTER_SELECTIVE);
    } else if (!acl_subscription_flag && acl_change_sub) {
        acl_change_sub = stasis_unsubscribe_and_join(acl_change_sub);
    }
#endif
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
    if ((s = ast_variable_retrieve(cfg, "general", "dtls_mtu"))) {
        if ((sscanf(s, "%d", &dtls_mtu) != 1) || dtls_mtu < 256) {
            ast_log(LOG_WARNING, "Value for 'dtls_mtu' could not be read, using default of '%d' instead\n",
                DEFAULT_DTLS_MTU);
            dtls_mtu = DEFAULT_DTLS_MTU;
        }
    }
#endif
 
    ast_config_destroy(cfg);
 
    /* Choosing an odd start port casues issues (like a potential infinite loop) and as odd parts are not
       chosen anyway, we are going to round up and issue a warning */
    if (rtpstart & 1) {
        rtpstart++;
        ast_log(LOG_WARNING, "Odd start value for RTP port in rtp.conf, rounding up to %d\n", rtpstart);
    }
 
    if (rtpstart >= rtpend) {
        ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
        rtpstart = DEFAULT_RTP_START;
        rtpend = DEFAULT_RTP_END;
    }
    ast_verb(2, "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
    return 0;
}

References acl_change_stasis_cb(), acl_change_sub, ast_append_acl(), ast_calloc, ast_config_destroy(), ast_config_load2(), ast_debug_stun, ast_dns_resolve_recurring(), ast_false(), ast_free, ast_free_acl_list(), ast_inet_ntoa(), ast_log, ast_named_acl_change_type(), ast_parse_arg(), AST_RWLIST_INSERT_TAIL, AST_RWLIST_UNLOCK, AST_RWLIST_WRLOCK, ast_rwlock_unlock, ast_rwlock_wrlock, ast_security_topic(), ast_skip_blanks(), ast_sockaddr_copy(), ast_sockaddr_port, ast_sockaddr_set_port, ast_sockaddr_setnull(), ast_sockaddr_split_hostport(), ast_sockaddr_stringify_host(), ast_sockaddr_to_sin, ast_strdup, ast_strdupa, ast_true(), ast_variable_browse(), ast_variable_retrieve(), ast_verb, CALC_LEARNING_MIN_DURATION, CONFIG_FLAG_FILEUNCHANGED, CONFIG_STATUS_FILEINVALID, CONFIG_STATUS_FILEUNCHANGED, DEFAULT_DTLS_MTU, DEFAULT_DTMF_TIMEOUT, DEFAULT_ICESUPPORT, DEFAULT_LEARNING_MIN_DURATION, DEFAULT_LEARNING_MIN_SEQUENTIAL, DEFAULT_RTP_END, DEFAULT_RTP_START, DEFAULT_SRTP_REPLAY_PROTECTION, DEFAULT_STRICT_RTP, DEFAULT_STUN_SOFTWARE_ATTRIBUTE, DEFAULT_TURN_PORT, dtmftimeout, learning_min_duration, learning_min_sequential, LOG_ERROR, LOG_WARNING, MAXIMUM_RTP_PORT, MINIMUM_RTP_PORT, ast_variable::next, NULL, PARSE_ADDR, PARSE_IN_RANGE, PARSE_INADDR, PARSE_PORT_IGNORE, PARSE_UINT32, reload(), RTCP_DEFAULT_INTERVALMS, RTCP_MAX_INTERVALMS, RTCP_MIN_INTERVALMS, rtcpinterval, rtpend, rtpstart, srtp_replay_protection, STANDARD_STUN_PORT, stasis_subscribe, stasis_subscription_accept_message_type(), STASIS_SUBSCRIPTION_FILTER_SELECTIVE, stasis_subscription_set_filter(), stasis_unsubscribe_and_join(), STRICT_RTP_NO, STRICT_RTP_SEQNO, STRICT_RTP_YES, strictrtp, and var.

Referenced by load_module(), and reload_module().

rtp_sendto()

static int rtp_sendto ( struct ast_rtp_instance *  instance, void *  buf, size_t  size, int  flags, struct ast_sockaddr *  sa, int *  ice  )

static

Precondition

instance is locked

Definition at line 3545 of file res_rtp_asterisk.c.

{
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    int hdrlen = 12;
    int res;
 
    if ((res = __rtp_sendto(instance, buf, size, flags, sa, 0, ice, 1)) > 0) {
        rtp->txcount++;
        rtp->txoctetcount += (res - hdrlen);
    }
 
    return res;
}

References __rtp_sendto(), ast_rtp_instance_get_data(), buf, ast_rtp::flags, ast_rtp::txcount, and ast_rtp::txoctetcount.

Referenced by ast_rtp_dtmf_begin(), ast_rtp_dtmf_continuation(), ast_rtp_dtmf_end_with_duration(), ast_rtp_rtcp_handle_nack(), ast_rtp_sendcng(), bridge_p2p_rtp_write(), and rtp_raw_write().

rtp_transport_wide_cc_feedback_produce()

static int rtp_transport_wide_cc_feedback_produce ( const void *  data )

static

Definition at line 7681 of file res_rtp_asterisk.c.

{
    struct ast_rtp_instance *instance = (struct ast_rtp_instance *) data;
    struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
    unsigned char *rtcpheader;
    char bdata[1024];
    struct rtp_transport_wide_cc_packet_statistics *first_packet;
    struct rtp_transport_wide_cc_packet_statistics *previous_packet;
    int i;
    int status_vector_chunk_bits = 14;
    uint16_t status_vector_chunk = (1 << 15) | (1 << 14);
    int run_length_chunk_count = 0;
    int run_length_chunk_status = -1;
    int packet_len = 20;
    int delta_len = 0;
    int packet_count = 0;
    unsigned int received_msw;
    unsigned int received_lsw;
    struct ast_sockaddr remote_address = { { 0, } };
    int res;
    int ice;
    unsigned int large_delta_count = 0;
    unsigned int small_delta_count = 0;
    unsigned int lost_count = 0;
 
    if (!rtp || !rtp->rtcp || rtp->transport_wide_cc.schedid == -1) {
        ao2_ref(instance, -1);
        return 0;
    }
 
    ao2_lock(instance);
 
    /* If no packets have been received then do nothing */
    if (!AST_VECTOR_SIZE(&rtp->transport_wide_cc.packet_statistics)) {
        ao2_unlock(instance);
        return 1000;
    }
 
    rtcpheader = (unsigned char *)bdata;
 
    /* The first packet in the vector acts as our base sequence number and reference time */
    first_packet = AST_VECTOR_GET_ADDR(&rtp->transport_wide_cc.packet_statistics, 0);
    previous_packet = first_packet;
 
    /* We go through each packet that we have statistics for, adding it either to a status
     * vector chunk or a run length chunk. The code tries to be as efficient as possible to
     * reduce packet size and will favor run length chunks when it makes sense.
     */
    for (i = 0; i < AST_VECTOR_SIZE(&rtp->transport_wide_cc.packet_statistics); ++i) {
        struct rtp_transport_wide_cc_packet_statistics *statistics;
        int lost = 0;
        int res = 0;
 
        statistics = AST_VECTOR_GET_ADDR(&rtp->transport_wide_cc.packet_statistics, i);
 
        packet_count++;
 
        if (first_packet != statistics) {
            /* The vector stores statistics in a sorted fashion based on the sequence
             * number. This ensures we can detect any packets that have been lost/not
             * received by comparing the sequence numbers.
             */
            lost = statistics->seqno - (previous_packet->seqno + 1);
            lost_count += lost;
        }
 
        while (lost) {
            /* We append a not received status until all the lost packets have been accounted for */
            rtp_transport_wide_cc_feedback_status_append(rtcpheader, &packet_len, &status_vector_chunk_bits,
                &status_vector_chunk, &run_length_chunk_count, &run_length_chunk_status, 0);
            packet_count++;
 
            /* If there is no more room left for storing packets stop now, we leave 20
             * extra bits at the end just in case.
             */
            if (packet_len + delta_len + 20 > sizeof(bdata)) {
                res = -1;
                break;
            }
 
            lost--;
        }
 
        /* If the lost packet appending bailed out because we have no more space, then exit here too */
        if (res) {
            break;
        }
 
        /* Per the spec the delta is in increments of 250 */
        statistics->delta = ast_tvdiff_us(statistics->received, previous_packet->received) / 250;
 
        /* Based on the delta determine the status of this packet */
        if (statistics->delta < 0 || statistics->delta > 127) {
            /* Large or negative delta */
            rtp_transport_wide_cc_feedback_status_append(rtcpheader, &packet_len, &status_vector_chunk_bits,
                &status_vector_chunk, &run_length_chunk_count, &run_length_chunk_status, 2);
            delta_len += 2;
            large_delta_count++;
        } else {
            /* Small delta */
            rtp_transport_wide_cc_feedback_status_append(rtcpheader, &packet_len, &status_vector_chunk_bits,
                &status_vector_chunk, &run_length_chunk_count, &run_length_chunk_status, 1);
            delta_len += 1;
            small_delta_count++;
        }
 
        previous_packet = statistics;
 
        /* If there is no more room left in the packet stop handling of any subsequent packets */
        if (packet_len + delta_len + 20 > sizeof(bdata)) {
            break;
        }
    }
 
    if (status_vector_chunk_bits != 14) {
        /* If the status vector chunk has packets in it then place it in the RTCP packet */
        put_unaligned_uint16(rtcpheader + packet_len, htons(status_vector_chunk));
        packet_len += 2;
    } else if (run_length_chunk_count) {
        /* If there is a run length chunk in progress then place it in the RTCP packet */
        put_unaligned_uint16(rtcpheader + packet_len, htons((0 << 15) | (run_length_chunk_status << 13) | run_length_chunk_count));
        packet_len += 2;
    }
 
    /* We iterate again to build delta chunks */
    for (i = 0; i < AST_VECTOR_SIZE(&rtp->transport_wide_cc.packet_statistics); ++i) {
        struct rtp_transport_wide_cc_packet_statistics *statistics;
 
        statistics = AST_VECTOR_GET_ADDR(&rtp->transport_wide_cc.packet_statistics, i);
 
        if (statistics->delta < 0 || statistics->delta > 127) {
            /* We need 2 bytes to store this delta */
            put_unaligned_uint16(rtcpheader + packet_len, htons(statistics->delta));
            packet_len += 2;
        } else {
            /* We can store this delta in 1 byte */
            rtcpheader[packet_len] = statistics->delta;
            packet_len += 1;
        }
 
        /* If this is the last packet handled by the run length chunk or status vector chunk code
         * then we can go no further.
         */
        if (statistics == previous_packet) {
            break;
        }
    }
 
    /* Zero pad the end of the packet */
    while (packet_len % 4) {
        rtcpheader[packet_len++] = 0;
    }
 
    /* Add the general RTCP header information */
    put_unaligned_uint32(rtcpheader, htonl((2 << 30) | (AST_RTP_RTCP_FMT_TRANSPORT_WIDE_CC << 24)
        | (AST_RTP_RTCP_RTPFB << 16) | ((packet_len / 4) - 1)));
    put_unaligned_uint32(rtcpheader + 4, htonl(rtp->ssrc));
    put_unaligned_uint32(rtcpheader + 8, htonl(rtp->themssrc));
 
    /* Add the transport-cc specific header information */
    put_unaligned_uint32(rtcpheader + 12, htonl((first_packet->seqno << 16) | packet_count));
 
    timeval2ntp(first_packet->received, &received_msw, &received_lsw);
    put_unaligned_time24(rtcpheader + 16, received_msw, received_lsw);
    rtcpheader[19] = rtp->transport_wide_cc.feedback_count;
 
    /* The packet is now fully constructed so send it out */
    ast_sockaddr_copy(&remote_address, &rtp->rtcp->them);
 
    ast_debug_rtcp(2, "(%p) RTCP sending transport-cc feedback packet of size '%d' on '%s' with packet count of %d (small = %d, large = %d, lost = %d)\n",
        instance, packet_len, ast_rtp_instance_get_channel_id(instance), packet_count, small_delta_count, large_delta_count, lost_count);
 
    res = rtcp_sendto(instance, (unsigned int *)rtcpheader, packet_len, 0, &remote_address, &ice);
    if (res < 0) {
        ast_log(LOG_ERROR, "RTCP transport-cc feedback error to %s due to %s\n",
            ast_sockaddr_stringify(&remote_address), strerror(errno));
    }
 
    AST_VECTOR_RESET(&rtp->transport_wide_cc.packet_statistics, AST_VECTOR_ELEM_CLEANUP_NOOP);
 
    rtp->transport_wide_cc.feedback_count++;
 
    ao2_unlock(instance);
 
    return 1000;
}

References ao2_lock, ao2_ref, ao2_unlock, ast_debug_rtcp, ast_log, ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_data(), AST_RTP_RTCP_FMT_TRANSPORT_WIDE_CC, AST_RTP_RTCP_RTPFB, ast_sockaddr_copy(), ast_sockaddr_stringify(), ast_tvdiff_us(), AST_VECTOR_ELEM_CLEANUP_NOOP, AST_VECTOR_GET_ADDR, AST_VECTOR_RESET, AST_VECTOR_SIZE, ast_rtp_instance::data, errno, rtp_transport_wide_cc_statistics::feedback_count, LOG_ERROR, rtp_transport_wide_cc_statistics::packet_statistics, put_unaligned_time24(), put_unaligned_uint16(), put_unaligned_uint32(), rtp_transport_wide_cc_packet_statistics::received, ast_rtp::rtcp, rtcp_sendto(), rtp_transport_wide_cc_feedback_status_append(), rtp_transport_wide_cc_statistics::schedid, rtp_transport_wide_cc_packet_statistics::seqno, ast_rtp::ssrc, statistics(), ast_rtcp::them, ast_rtp::themssrc, timeval2ntp(), and ast_rtp::transport_wide_cc.

Referenced by rtp_instance_parse_transport_wide_cc().

rtp_transport_wide_cc_feedback_status_append()

static void rtp_transport_wide_cc_feedback_status_append ( unsigned char *  rtcpheader, int *  packet_len, int *  status_vector_chunk_bits, uint16_t *  status_vector_chunk, int *  run_length_chunk_count, int *  run_length_chunk_status, int  status  )

static

Definition at line 7640 of file res_rtp_asterisk.c.

{
    if (*run_length_chunk_status != status) {
        while (*run_length_chunk_count > 0 && *run_length_chunk_count < 8) {
            /* Realistically it only makes sense to use a run length chunk if there were 8 or more
             * consecutive packets of the same type, otherwise we could end up making the packet larger
             * if we have lots of small blocks of the same type. To help with this we backfill the status
             * vector (since it always represents 7 packets). Best case we end up with only that single
             * status vector and the rest are run length chunks.
             */
            rtp_transport_wide_cc_feedback_status_vector_append(rtcpheader, packet_len, status_vector_chunk_bits,
                status_vector_chunk, *run_length_chunk_status);
            *run_length_chunk_count -= 1;
        }
 
        if (*run_length_chunk_count) {
            /* There is a run length chunk which needs to be written out */
            put_unaligned_uint16(rtcpheader + *packet_len, htons((0 << 15) | (*run_length_chunk_status << 13) | *run_length_chunk_count));
            *packet_len += 2;
        }
 
        /* In all cases the run length chunk has to be reset */
        *run_length_chunk_count = 0;
        *run_length_chunk_status = -1;
 
        if (*status_vector_chunk_bits == 14) {
            /* We aren't in the middle of a status vector so we can try for a run length chunk */
            *run_length_chunk_status = status;
            *run_length_chunk_count = 1;
        } else {
            /* We're doing a status vector so populate it accordingly */
            rtp_transport_wide_cc_feedback_status_vector_append(rtcpheader, packet_len, status_vector_chunk_bits,
                status_vector_chunk, status);
        }
    } else {
        /* This is easy, the run length chunk count can just get bumped up */
        *run_length_chunk_count += 1;
    }
}

References put_unaligned_uint16(), rtp_transport_wide_cc_feedback_status_vector_append(), and status.

Referenced by rtp_transport_wide_cc_feedback_produce().

rtp_transport_wide_cc_feedback_status_vector_append()

static void rtp_transport_wide_cc_feedback_status_vector_append ( unsigned char *  rtcpheader, int *  packet_len, int *  status_vector_chunk_bits, uint16_t *  status_vector_chunk, int  status  )

static

Definition at line 7611 of file res_rtp_asterisk.c.

{
    /* Appending this status will use up 2 bits */
    *status_vector_chunk_bits -= 2;
 
    /* We calculate which bits we want to update the status of. Since a status vector
     * is 16 bits we take away 2 (for the header), and then we take away any that have
     * already been used.
     */
    *status_vector_chunk |= (status << (16 - 2 - (14 - *status_vector_chunk_bits)));
 
    /* If there are still bits available we can return early */
    if (*status_vector_chunk_bits) {
        return;
    }
 
    /* Otherwise we have to place this chunk into the packet */
    put_unaligned_uint16(rtcpheader + *packet_len, htons(*status_vector_chunk));
    *status_vector_chunk_bits = 14;
 
    /* The first bit being 1 indicates that this is a status vector chunk and the second
     * bit being 1 indicates that we are using 2 bits to represent each status for a
     * packet.
     */
    *status_vector_chunk = (1 << 15) | (1 << 14);
    *packet_len += 2;
}

References put_unaligned_uint16(), and status.

Referenced by rtp_transport_wide_cc_feedback_status_append().

rtp_transport_wide_cc_packet_statistics_cmp()

static int rtp_transport_wide_cc_packet_statistics_cmp ( struct rtp_transport_wide_cc_packet_statistics  a, struct rtp_transport_wide_cc_packet_statistics  b  )

static

Definition at line 7605 of file res_rtp_asterisk.c.

{
    return a.seqno - b.seqno;
}

References a, and b.

Referenced by rtp_instance_parse_transport_wide_cc().

rtp_write_rtcp_fir()

static void rtp_write_rtcp_fir ( struct ast_rtp_instance *  instance, struct ast_rtp *  rtp, struct ast_sockaddr *  remote_address  )

static

Definition at line 5567 of file res_rtp_asterisk.c.

{
    unsigned char *rtcpheader;
    unsigned char bdata[1024];
    int packet_len = 0;
    int fir_len = 20;
    int ice;
    int res;
    int sr;
    RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report, NULL, ao2_cleanup);
 
    if (!rtp || !rtp->rtcp) {
        return;
    }
 
    if (ast_sockaddr_isnull(&rtp->rtcp->them) || rtp->rtcp->schedid < 0) {
        /*
         * RTCP was stopped.
         */
        return;
    }
 
    if (!rtp->themssrc_valid) {
        /* We don't know their SSRC value so we don't know who to update. */
        return;
    }
 
    /* Prepare RTCP FIR (PT=206, FMT=4) */
    rtp->rtcp->firseq++;
    if(rtp->rtcp->firseq == 256) {
        rtp->rtcp->firseq = 0;
    }
 
    rtcpheader = bdata;
 
    ao2_lock(instance);
    rtcp_report = ast_rtp_rtcp_report_alloc(rtp->themssrc_valid ? 1 : 0);
    res = ast_rtcp_generate_compound_prefix(instance, rtcpheader, rtcp_report, &sr);
 
    if (res == 0 || res == 1) {
        ao2_unlock(instance);
        return;
    }
 
    packet_len += res;
 
    put_unaligned_uint32(rtcpheader + packet_len + 0, htonl((2 << 30) | (4 << 24) | (RTCP_PT_PSFB << 16) | ((fir_len/4)-1)));
    put_unaligned_uint32(rtcpheader + packet_len + 4, htonl(rtp->ssrc));
    put_unaligned_uint32(rtcpheader + packet_len + 8, htonl(rtp->themssrc));
    put_unaligned_uint32(rtcpheader + packet_len + 12, htonl(rtp->themssrc)); /* FCI: SSRC */
    put_unaligned_uint32(rtcpheader + packet_len + 16, htonl(rtp->rtcp->firseq << 24)); /* FCI: Sequence number */
    res = rtcp_sendto(instance, (unsigned int *)rtcpheader, packet_len + fir_len, 0, rtp->bundled ? remote_address : &rtp->rtcp->them, &ice);
    if (res < 0) {
        ast_log(LOG_ERROR, "RTCP FIR transmission error: %s\n", strerror(errno));
    } else {
        ast_rtcp_calculate_sr_rr_statistics(instance, rtcp_report, rtp->bundled ? *remote_address : rtp->rtcp->them, ice, sr);
    }
 
    ao2_unlock(instance);
}

References ao2_cleanup, ao2_lock, ao2_unlock, ast_log, ast_rtcp_calculate_sr_rr_statistics(), ast_rtcp_generate_compound_prefix(), ast_rtp_rtcp_report_alloc(), ast_sockaddr_isnull(), ast_rtp::bundled, errno, ast_rtcp::firseq, LOG_ERROR, NULL, put_unaligned_uint32(), RAII_VAR, ast_rtp::rtcp, RTCP_PT_PSFB, rtcp_sendto(), ast_rtcp::schedid, ast_rtp::ssrc, ast_rtcp::them, ast_rtp::themssrc, and ast_rtp::themssrc_valid.

Referenced by ast_rtp_read(), and ast_rtp_write().

rtp_write_rtcp_psfb()

static void rtp_write_rtcp_psfb ( struct ast_rtp_instance *  instance, struct ast_rtp *  rtp, struct ast_frame *  frame, struct ast_sockaddr *  remote_address  )

static

Definition at line 5628 of file res_rtp_asterisk.c.

{
    struct ast_rtp_rtcp_feedback *feedback = frame->data.ptr;
    unsigned char *rtcpheader;
    unsigned char bdata[1024];
    int remb_len = 24;
    int ice;
    int res;
    int sr = 0;
    int packet_len = 0;
    RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report, NULL, ao2_cleanup);
 
    if (feedback->fmt != AST_RTP_RTCP_FMT_REMB) {
        ast_debug_rtcp(1, "(%p) RTCP provided feedback frame of format %d to write, but only REMB is supported\n",
            instance, feedback->fmt);
        return;
    }
 
    if (!rtp || !rtp->rtcp) {
        return;
    }
 
    /* If REMB support is not enabled don't send this RTCP packet */
    if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_REMB)) {
        ast_debug_rtcp(1, "(%p) RTCP provided feedback REMB report to write, but REMB support not enabled\n",
            instance);
        return;
    }
 
    if (ast_sockaddr_isnull(&rtp->rtcp->them) || rtp->rtcp->schedid < 0) {
        /*
         * RTCP was stopped.
         */
        return;
    }
 
    rtcpheader = bdata;
 
    ao2_lock(instance);
    rtcp_report = ast_rtp_rtcp_report_alloc(rtp->themssrc_valid ? 1 : 0);
    res = ast_rtcp_generate_compound_prefix(instance, rtcpheader, rtcp_report, &sr);
 
    if (res == 0 || res == 1) {
        ao2_unlock(instance);
        return;
    }
 
    packet_len += res;
 
    put_unaligned_uint32(rtcpheader + packet_len + 0, htonl((2 << 30) | (AST_RTP_RTCP_FMT_REMB << 24) | (RTCP_PT_PSFB << 16) | ((remb_len/4)-1)));
    put_unaligned_uint32(rtcpheader + packet_len + 4, htonl(rtp->ssrc));
    put_unaligned_uint32(rtcpheader + packet_len + 8, htonl(0)); /* Per the draft, this should always be 0 */
    put_unaligned_uint32(rtcpheader + packet_len + 12, htonl(('R' << 24) | ('E' << 16) | ('M' << 8) | ('B'))); /* Unique identifier 'R' 'E' 'M' 'B' */
    put_unaligned_uint32(rtcpheader + packet_len + 16, htonl((1 << 24) | (feedback->remb.br_exp << 18) | (feedback->remb.br_mantissa))); /* Number of SSRCs / BR Exp / BR Mantissa */
    put_unaligned_uint32(rtcpheader + packet_len + 20, htonl(rtp->ssrc)); /* The SSRC this feedback message applies to */
    res = rtcp_sendto(instance, (unsigned int *)rtcpheader, packet_len + remb_len, 0, rtp->bundled ? remote_address : &rtp->rtcp->them, &ice);
    if (res < 0) {
        ast_log(LOG_ERROR, "RTCP PSFB transmission error: %s\n", strerror(errno));
    } else {
        ast_rtcp_calculate_sr_rr_statistics(instance, rtcp_report, rtp->bundled ? *remote_address : rtp->rtcp->them, ice, sr);
    }
 
    ao2_unlock(instance);
}

References ao2_cleanup, ao2_lock, ao2_unlock, ast_debug_rtcp, ast_log, ast_rtcp_calculate_sr_rr_statistics(), ast_rtcp_generate_compound_prefix(), ast_rtp_instance_get_prop(), AST_RTP_PROPERTY_REMB, AST_RTP_RTCP_FMT_REMB, ast_rtp_rtcp_report_alloc(), ast_sockaddr_isnull(), ast_rtp_rtcp_feedback_remb::br_exp, ast_rtp_rtcp_feedback_remb::br_mantissa, ast_rtp::bundled, ast_frame::data, errno, ast_rtp_rtcp_feedback::fmt, LOG_ERROR, NULL, ast_frame::ptr, put_unaligned_uint32(), RAII_VAR, ast_rtp_rtcp_feedback::remb, ast_rtp::rtcp, RTCP_PT_PSFB, rtcp_sendto(), ast_rtcp::schedid, ast_rtp::ssrc, ast_rtcp::them, and ast_rtp::themssrc_valid.

Referenced by ast_rtp_write().

timeval2ntp()

static void timeval2ntp ( struct timeval  tv, unsigned int *  msw, unsigned int *  lsw  )

static

Definition at line 4777 of file res_rtp_asterisk.c.

{
    unsigned int sec, usec, frac;
    sec = tv.tv_sec + 2208988800u; /* Sec between 1900 and 1970 */
    usec = tv.tv_usec;
    /*
     * Convert usec to 0.32 bit fixed point without overflow.
     *
     * = usec * 2^32 / 10^6
     * = usec * 2^32 / (2^6 * 5^6)
     * = usec * 2^26 / 5^6
     *
     * The usec value needs 20 bits to represent 999999 usec.  So
     * splitting the 2^26 to get the most precision using 32 bit
     * values gives:
     *
     * = ((usec * 2^12) / 5^6) * 2^14
     *
     * Splitting the division into two stages preserves all the
     * available significant bits of usec over doing the division
     * all at once.
     *
     * = ((((usec * 2^12) / 5^3) * 2^7) / 5^3) * 2^7
     */
    frac = ((((usec << 12) / 125) << 7) / 125) << 7;
    *msw = sec;
    *lsw = frac;
}

Referenced by ast_rtcp_generate_report(), ast_rtcp_interpret(), ast_rtp_rtcp_handle_nack(), rtp_raw_write(), rtp_transport_wide_cc_feedback_produce(), and update_rtt_stats().

unload_module()

static int unload_module ( void  )

static

Definition at line 10764 of file res_rtp_asterisk.c.

{
    ast_rtp_engine_unregister(&asterisk_rtp_engine);
    ast_cli_unregister_multiple(cli_rtp, ARRAY_LEN(cli_rtp));
 
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP) && defined(HAVE_OPENSSL_BIO_METHOD)
    if (dtls_bio_methods) {
        BIO_meth_free(dtls_bio_methods);
    }
#endif
 
#ifdef HAVE_PJPROJECT
    host_candidate_overrides_clear();
    pj_thread_register_check();
    rtp_terminate_pjproject();
 
    acl_change_sub = stasis_unsubscribe_and_join(acl_change_sub);
    rtp_unload_acl(&ice_acl_lock, &ice_acl);
    rtp_unload_acl(&stun_acl_lock, &stun_acl);
    clean_stunaddr();
#endif
 
    return 0;
}

References acl_change_sub, ARRAY_LEN, ast_cli_unregister_multiple(), ast_rtp_engine_unregister(), asterisk_rtp_engine, cli_rtp, and stasis_unsubscribe_and_join().

update_jitter_stats()

static void update_jitter_stats ( struct ast_rtp *  rtp, unsigned int  ia_jitter  )

static

Definition at line 6376 of file res_rtp_asterisk.c.

{
    int rate = ast_rtp_get_rate(rtp->f.subclass.format);
 
    rtp->rtcp->reported_jitter = ast_samp2sec(ia_jitter, rate);
 
    if (rtp->rtcp->reported_jitter_count == 0) {
        rtp->rtcp->reported_minjitter = rtp->rtcp->reported_jitter;
    }
    if (rtp->rtcp->reported_jitter < rtp->rtcp->reported_minjitter) {
        rtp->rtcp->reported_minjitter = rtp->rtcp->reported_jitter;
    }
    if (rtp->rtcp->reported_jitter > rtp->rtcp->reported_maxjitter) {
        rtp->rtcp->reported_maxjitter = rtp->rtcp->reported_jitter;
    }
 
    calc_mean_and_standard_deviation(rtp->rtcp->reported_jitter,
        &rtp->rtcp->reported_normdev_jitter, &rtp->rtcp->reported_stdev_jitter,
        &rtp->rtcp->reported_jitter_count);
}

References ast_rtp_get_rate(), ast_samp2sec(), calc_mean_and_standard_deviation(), ast_rtp::f, ast_frame_subclass::format, ast_rtcp::reported_jitter, ast_rtcp::reported_jitter_count, ast_rtcp::reported_maxjitter, ast_rtcp::reported_minjitter, ast_rtcp::reported_normdev_jitter, ast_rtcp::reported_stdev_jitter, ast_rtp::rtcp, and ast_frame::subclass.

Referenced by ast_rtcp_interpret().

update_local_mes_stats()

static void update_local_mes_stats ( struct ast_rtp *  rtp )

static

Definition at line 6565 of file res_rtp_asterisk.c.

{
    rtp->rxmes = calc_media_experience_score(rtp->owner,
        rtp->rtcp->normdevrtt,
        rtp->rtcp->normdev_rxjitter,
        rtp->rtcp->stdev_rxjitter,
        rtp->rtcp->normdev_rxlost);
 
    if (rtp->rtcp->rxmes_count == 0) {
        rtp->rtcp->minrxmes = rtp->rxmes;
    }
    if (rtp->rxmes < rtp->rtcp->minrxmes) {
        rtp->rtcp->minrxmes = rtp->rxmes;
    }
    if (rtp->rxmes > rtp->rtcp->maxrxmes) {
        rtp->rtcp->maxrxmes = rtp->rxmes;
    }
 
    calc_mean_and_standard_deviation(rtp->rxmes, &rtp->rtcp->normdev_rxmes,
        &rtp->rtcp->stdev_rxmes, &rtp->rtcp->rxmes_count);
 
    ast_debug_rtcp(2, "   %s: rtt: %.9f j: %.9f sjh: %.9f lost: %.9f mes: %4.1f\n",
        ast_rtp_instance_get_channel_id(rtp->owner),
        rtp->rtcp->normdevrtt,
                rtp->rtcp->normdev_rxjitter,
                rtp->rtcp->stdev_rxjitter,
                rtp->rtcp->normdev_rxlost, rtp->rxmes);
}

References ast_debug_rtcp, ast_rtp_instance_get_channel_id(), calc_mean_and_standard_deviation(), calc_media_experience_score(), ast_rtcp::maxrxmes, ast_rtcp::minrxmes, ast_rtcp::normdev_rxjitter, ast_rtcp::normdev_rxlost, ast_rtcp::normdev_rxmes, ast_rtcp::normdevrtt, ast_rtp::owner, ast_rtp::rtcp, ast_rtp::rxmes, ast_rtcp::rxmes_count, ast_rtcp::stdev_rxjitter, and ast_rtcp::stdev_rxmes.

Referenced by ast_rtcp_generate_report().

update_lost_stats()

static void update_lost_stats ( struct ast_rtp *  rtp, unsigned int  lost_packets  )

static

Definition at line 6401 of file res_rtp_asterisk.c.

{
    double reported_lost;
 
    rtp->rtcp->reported_lost = lost_packets;
 
    /*
     * lost_packets contains the cumulative number of lost packets as reported in
     * the peer's RTCP RR/SR report block (RFC 3550). Calculate the number of lost
     * packets in the current interval based on the difference from the previous
     * count.
     */
    reported_lost = (double)lost_packets - (double)rtp->rtcp->last_reported_lost;
    rtp->rtcp->last_reported_lost = lost_packets;
 
    if (reported_lost < 0) {
        reported_lost = 0;
    }
 
    if (rtp->rtcp->reported_lost_count == 0) {
        rtp->rtcp->reported_minlost = reported_lost;
    }
    if (reported_lost < rtp->rtcp->reported_minlost) {
        rtp->rtcp->reported_minlost = reported_lost;
    }
    if (reported_lost > rtp->rtcp->reported_maxlost) {
        rtp->rtcp->reported_maxlost = reported_lost;
    }
 
    calc_mean_and_standard_deviation(reported_lost, &rtp->rtcp->reported_normdev_lost,
        &rtp->rtcp->reported_stdev_lost, &rtp->rtcp->reported_lost_count);
}

References calc_mean_and_standard_deviation(), ast_rtcp::last_reported_lost, ast_rtcp::reported_lost, ast_rtcp::reported_lost_count, ast_rtcp::reported_maxlost, ast_rtcp::reported_minlost, ast_rtcp::reported_normdev_lost, ast_rtcp::reported_stdev_lost, and ast_rtp::rtcp.

Referenced by ast_rtcp_interpret().

update_reported_mes_stats()

static void update_reported_mes_stats ( struct ast_rtp *  rtp )

static

Definition at line 6530 of file res_rtp_asterisk.c.

{
    double mes = calc_media_experience_score(rtp->owner,
        rtp->rtcp->normdevrtt,
        rtp->rtcp->reported_normdev_jitter,
        rtp->rtcp->reported_stdev_jitter,
        rtp->rtcp->reported_normdev_lost);
 
    rtp->rtcp->reported_mes = mes;
    if (rtp->rtcp->reported_mes_count == 0) {
        rtp->rtcp->reported_minmes = mes;
    }
    if (mes < rtp->rtcp->reported_minmes) {
        rtp->rtcp->reported_minmes = mes;
    }
    if (mes > rtp->rtcp->reported_maxmes) {
        rtp->rtcp->reported_maxmes = mes;
    }
 
    calc_mean_and_standard_deviation(mes, &rtp->rtcp->reported_normdev_mes,
        &rtp->rtcp->reported_stdev_mes, &rtp->rtcp->reported_mes_count);
 
    ast_debug_rtcp(2, "%s: rtt: %.9f j: %.9f sjh: %.9f lost: %.9f mes: %4.1f\n",
        ast_rtp_instance_get_channel_id(rtp->owner),
        rtp->rtcp->normdevrtt,
                rtp->rtcp->reported_normdev_jitter,
                rtp->rtcp->reported_stdev_jitter,
                rtp->rtcp->reported_normdev_lost, mes);
}

References ast_debug_rtcp, ast_rtp_instance_get_channel_id(), calc_mean_and_standard_deviation(), calc_media_experience_score(), ast_rtcp::normdevrtt, ast_rtp::owner, ast_rtcp::reported_maxmes, ast_rtcp::reported_mes, ast_rtcp::reported_mes_count, ast_rtcp::reported_minmes, ast_rtcp::reported_normdev_jitter, ast_rtcp::reported_normdev_lost, ast_rtcp::reported_normdev_mes, ast_rtcp::reported_stdev_jitter, ast_rtcp::reported_stdev_mes, and ast_rtp::rtcp.

Referenced by ast_rtcp_interpret().

update_rtt_stats()

static int update_rtt_stats ( struct ast_rtp *  rtp, unsigned int  lsr, unsigned int  dlsr  )

static

Definition at line 6319 of file res_rtp_asterisk.c.

{
    struct timeval now;
    struct timeval rtt_tv;
    unsigned int msw;
    unsigned int lsw;
    unsigned int rtt_msw;
    unsigned int rtt_lsw;
    unsigned int lsr_a;
    unsigned int rtt;
 
    gettimeofday(&now, NULL);
    timeval2ntp(now, &msw, &lsw);
 
    lsr_a = ((msw & 0x0000ffff) << 16) | ((lsw & 0xffff0000) >> 16);
    if (lsr_a - dlsr < lsr) {
        return 1;
    }
 
    rtt = lsr_a - lsr - dlsr;
    rtt_msw = (rtt & 0xffff0000) >> 16;
    rtt_lsw = (rtt & 0x0000ffff);
    rtt_tv.tv_sec = rtt_msw;
    /*
     * Convert 16.16 fixed point rtt_lsw to usec without
     * overflow.
     *
     * = rtt_lsw * 10^6 / 2^16
     * = rtt_lsw * (2^6 * 5^6) / 2^16
     * = rtt_lsw * 5^6 / 2^10
     *
     * The rtt_lsw value is in 16.16 fixed point format and 5^6
     * requires 14 bits to represent.  We have enough space to
     * directly do the conversion because there is no integer
     * component in rtt_lsw.
     */
    rtt_tv.tv_usec = (rtt_lsw * 15625) >> 10;
    rtp->rtcp->rtt = (double)rtt_tv.tv_sec + ((double)rtt_tv.tv_usec / 1000000);
    rtp->rtcp->accumulated_transit += rtp->rtcp->rtt;
 
    if (rtp->rtcp->rtt_count == 0 || rtp->rtcp->minrtt > rtp->rtcp->rtt) {
        rtp->rtcp->minrtt = rtp->rtcp->rtt;
    }
    if (rtp->rtcp->maxrtt < rtp->rtcp->rtt) {
        rtp->rtcp->maxrtt = rtp->rtcp->rtt;
    }
 
    calc_mean_and_standard_deviation(rtp->rtcp->rtt, &rtp->rtcp->normdevrtt,
        &rtp->rtcp->stdevrtt, &rtp->rtcp->rtt_count);
 
    return 0;
}

References ast_rtcp::accumulated_transit, calc_mean_and_standard_deviation(), ast_rtcp::maxrtt, ast_rtcp::minrtt, ast_rtcp::normdevrtt, NULL, ast_rtp::rtcp, ast_rtcp::rtt, ast_rtcp::rtt_count, ast_rtcp::stdevrtt, and timeval2ntp().

Referenced by ast_rtcp_interpret().

Variable Documentation

__mod_info

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Asterisk RTP Stack" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .reload = reload_module, .load_pri = AST_MODPRI_CHANNEL_DEPEND, #ifdef HAVE_PJPROJECT .requires = "res_pjproject", #endif }

static

Definition at line 10798 of file res_rtp_asterisk.c.

ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info

static

Definition at line 10798 of file res_rtp_asterisk.c.

asterisk_rtp_engine

struct ast_rtp_engine asterisk_rtp_engine

static

Definition at line 2576 of file res_rtp_asterisk.c.

                                                   {
    .name = "asterisk",
    .new = ast_rtp_new,
    .destroy = ast_rtp_destroy,
    .dtmf_begin = ast_rtp_dtmf_begin,
    .dtmf_end = ast_rtp_dtmf_end,
    .dtmf_end_with_duration = ast_rtp_dtmf_end_with_duration,
    .dtmf_mode_set = ast_rtp_dtmf_mode_set,
    .dtmf_mode_get = ast_rtp_dtmf_mode_get,
    .update_source = ast_rtp_update_source,
    .change_source = ast_rtp_change_source,
    .write = ast_rtp_write,
    .read = ast_rtp_read,
    .prop_set = ast_rtp_prop_set,
    .fd = ast_rtp_fd,
    .remote_address_set = ast_rtp_remote_address_set,
    .red_init = rtp_red_init,
    .red_buffer = rtp_red_buffer,
    .local_bridge = ast_rtp_local_bridge,
    .get_stat = ast_rtp_get_stat,
    .dtmf_compatible = ast_rtp_dtmf_compatible,
    .stun_request = ast_rtp_stun_request,
    .stop = ast_rtp_stop,
    .qos = ast_rtp_qos_set,
    .sendcng = ast_rtp_sendcng,
#ifdef HAVE_PJPROJECT
    .ice = &ast_rtp_ice,
#endif
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
    .dtls = &ast_rtp_dtls,
    .activate = ast_rtp_activate,
#endif
    .ssrc_get = ast_rtp_get_ssrc,
    .cname_get = ast_rtp_get_cname,
    .set_remote_ssrc = ast_rtp_set_remote_ssrc,
    .set_stream_num = ast_rtp_set_stream_num,
    .extension_enable = ast_rtp_extension_enable,
    .bundle = ast_rtp_bundle,
#ifdef TEST_FRAMEWORK
    .test = &ast_rtp_test,
#endif
};

Referenced by load_module(), and unload_module().

cli_rtp

struct ast_cli_entry cli_rtp[]

static

Definition at line 10330 of file res_rtp_asterisk.c.

                                        {
    AST_CLI_DEFINE(handle_cli_rtp_set_debug,  "Enable/Disable RTP debugging"),
    AST_CLI_DEFINE(handle_cli_rtp_settings,   "Display RTP settings"),
    AST_CLI_DEFINE(handle_cli_rtcp_set_debug, "Enable/Disable RTCP debugging"),
    AST_CLI_DEFINE(handle_cli_rtcp_set_stats, "Enable/Disable RTCP stats"),
#ifdef AST_DEVMODE
    AST_CLI_DEFINE(handle_cli_rtp_drop_incoming_packets, "Drop RTP incoming packets"),
#endif
#ifdef HAVE_PJPROJECT
    AST_CLI_DEFINE(handle_cli_rtp_refresh_stun, "Force a resolution of the STUN hostname"),
#endif
};

Referenced by load_module(), and unload_module().

dtmftimeout

int dtmftimeout = DEFAULT_DTMF_TIMEOUT

static

Definition at line 208 of file res_rtp_asterisk.c.

Referenced by handle_cli_rtp_settings(), process_dtmf_rfc2833(), and rtp_reload().

learning_min_duration

int learning_min_duration = DEFAULT_LEARNING_MIN_DURATION

static

Lowest acceptable timeout between the first and the last sequential RTP frame.

Definition at line 223 of file res_rtp_asterisk.c.

Referenced by rtp_learning_rtp_seq_update(), and rtp_reload().

learning_min_sequential

int learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL

static

Number of sequential RTP frames needed from a single source during learning mode to accept new source.

Definition at line 222 of file res_rtp_asterisk.c.

Referenced by handle_cli_rtp_settings(), rtp_learning_rtp_seq_update(), rtp_learning_seq_init(), and rtp_reload().

res_srtp

struct ast_srtp_res* res_srtp

extern

Definition at line 182 of file rtp_engine.c.

Referenced by __rtp_sendto(), ast_rtcp_interpret(), ast_rtp_change_source(), ast_rtp_engine_register_srtp(), ast_rtp_engine_srtp_is_registered(), ast_rtp_engine_unregister_srtp(), ast_rtp_instance_add_srtp_policy(), ast_rtp_interpret(), and instance_destructor().

res_srtp_policy

struct ast_srtp_policy_res* res_srtp_policy

extern

Definition at line 183 of file rtp_engine.c.

Referenced by ast_rtp_engine_register_srtp(), ast_rtp_engine_srtp_is_registered(), and ast_rtp_engine_unregister_srtp().

rtcpdebugaddr

struct ast_sockaddr rtcpdebugaddr

static

Debug RTCP packets to/from this host

Definition at line 215 of file res_rtp_asterisk.c.

Referenced by handle_cli_rtcp_set_debug(), rtcp_debug_test_addr(), and rtcp_do_debug_ip().

rtcpdebugport

int rtcpdebugport

static

Debug only RTCP packets from IP or IP+Port if port is > 0

Definition at line 217 of file res_rtp_asterisk.c.

Referenced by rtcp_debug_test_addr(), and rtcp_do_debug_ip().

rtcpinterval

int rtcpinterval = RTCP_DEFAULT_INTERVALMS

static

Time between rtcp reports in millisecs

Definition at line 213 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_calc_interval(), and rtp_reload().

rtcpstats

int rtcpstats

static

Are we debugging RTCP?

Definition at line 212 of file res_rtp_asterisk.c.

Referenced by handle_cli_rtcp_set_stats().

rtpdebugaddr

struct ast_sockaddr rtpdebugaddr

static

Debug packets to/from this host

Definition at line 214 of file res_rtp_asterisk.c.

Referenced by handle_cli_rtp_set_debug(), rtp_debug_test_addr(), and rtp_do_debug_ip().

rtpdebugport

int rtpdebugport

static

Debug only RTP packets from IP or IP+Port if port is > 0

Definition at line 216 of file res_rtp_asterisk.c.

Referenced by rtp_debug_test_addr(), and rtp_do_debug_ip().

rtpend

int rtpend = DEFAULT_RTP_END

static

Last port for RTP sessions (set in rtp.conf)

Definition at line 211 of file res_rtp_asterisk.c.

Referenced by handle_cli_rtp_settings(), rtp_allocate_transport(), and rtp_reload().

rtpstart

int rtpstart = DEFAULT_RTP_START

static

First port for RTP sessions (set in rtp.conf)

Definition at line 210 of file res_rtp_asterisk.c.

Referenced by handle_cli_rtp_settings(), rtp_allocate_transport(), and rtp_reload().

srtp_replay_protection

int srtp_replay_protection = DEFAULT_SRTP_REPLAY_PROTECTION

static

Definition at line 224 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret(), ast_rtp_interpret(), handle_cli_rtp_settings(), and rtp_reload().

strictrtp

int strictrtp = DEFAULT_STRICT_RTP

static

Only accept RTP frames from a defined source. If we receive an indication of a changing source, enter learning mode.

Definition at line 221 of file res_rtp_asterisk.c.

Referenced by ast_rtp_remote_address_set(), handle_cli_rtp_settings(), rtp_allocate_transport(), rtp_learning_rtp_seq_update(), and rtp_reload().