SIP SDP media stream handling. More...

#include "asterisk.h"
#include <pjsip.h>
#include <pjsip_ua.h>
#include <pjmedia.h>
#include <pjlib.h>
#include "asterisk/utils.h"
#include "asterisk/module.h"
#include "asterisk/format.h"
#include "asterisk/format_cap.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/netsock2.h"
#include "asterisk/channel.h"
#include "asterisk/causes.h"
#include "asterisk/sched.h"
#include "asterisk/acl.h"
#include "asterisk/sdp_srtp.h"
#include "asterisk/dsp.h"
#include "asterisk/linkedlists.h"
#include "asterisk/stream.h"
#include "asterisk/logger_category.h"
#include "asterisk/format_cache.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/res_pjsip_session_caps.h"

Include dependency graph for res_pjsip_sdp_rtp.c:

Go to the source code of this file.

Functions
static void	__reg_module (void)

static void	__unreg_module (void)

static int	add_crypto_to_stream (struct ast_sip_session session, struct ast_sip_session_media session_media, pj_pool_t pool, pjmedia_sdp_media media)

static void	add_extmap_to_stream (struct ast_sip_session session, struct ast_sip_session_media session_media, pj_pool_t pool, pjmedia_sdp_media media)

static void	add_ice_to_stream (struct ast_sip_session session, struct ast_sip_session_media session_media, pj_pool_t pool, pjmedia_sdp_media media, unsigned int include_candidates)
	Function which adds ICE attributes to a media stream.

static void	add_msid_to_stream (struct ast_sip_session session, struct ast_sip_session_media session_media, pj_pool_t pool, pjmedia_sdp_media media, struct ast_stream *stream)

static void	add_rtcp_fb_to_stream (struct ast_sip_session session, struct ast_sip_session_media session_media, pj_pool_t pool, pjmedia_sdp_media media)

static void	add_ssrc_to_stream (struct ast_sip_session session, struct ast_sip_session_media session_media, pj_pool_t pool, pjmedia_sdp_media media)
	Function which adds ssrc attributes to a media stream.

static int	apply_cap_to_bundled (struct ast_sip_session_media session_media, struct ast_sip_session_media session_media_transport, struct ast_stream asterisk_stream, struct ast_format_cap joint)

static void	apply_dtls_attrib (struct ast_sip_session_media session_media, pjmedia_sdp_attr attr)

static int	apply_negotiated_sdp_stream (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_session local, const struct pjmedia_sdp_session remote, int index, struct ast_stream *asterisk_stream)

static int	assign_unique_bundle_payload (struct ast_sip_session session, struct ast_sip_session_media session_media, struct ast_sip_session_media session_media_transport, const pjmedia_sdp_session sdp, const pjmedia_sdp_media media, struct ast_format format)
	Assign a payload type that is unique across the generated BUNDLE group.

struct ast_module *	AST_MODULE_SELF_SYM (void)

static int	bundle_payload_type_used (struct ast_sip_session session, const pjmedia_sdp_session sdp, const pjmedia_sdp_media *media, int bundle_group, int payload)
	Determine whether a payload type is already advertised in a BUNDLE group.

static void	change_outgoing_sdp_stream_media_address (pjsip_tx_data tdata, struct pjmedia_sdp_media stream, struct ast_sip_transport *transport)
	Function which updates the media stream with external media address, if applicable.

static enum ast_sip_session_media_encryption	check_endpoint_media_transport (struct ast_sip_endpoint endpoint, const struct pjmedia_sdp_media stream)
	Checks whether the encryption offered in SDP is compatible with the endpoint's configuration.

static void	check_ice_support (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_media *remote_stream)
	Function which checks for ice attributes in an audio stream.

static int	create_outgoing_sdp_stream (struct ast_sip_session session, struct ast_sip_session_media session_media, struct pjmedia_sdp_session sdp, const struct pjmedia_sdp_session remote, struct ast_stream *stream)
	Function which creates an outgoing stream.

static int	create_rtp (struct ast_sip_session session, struct ast_sip_session_media session_media, const pjmedia_sdp_session *sdp)
	Internal function which creates an RTP instance.

static void	enable_rtcp (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_media *remote_media)
	Enable RTCP on an RTP session.

static void	enable_rtp_extension (struct ast_sip_session session, struct ast_sip_session_media session_media, enum ast_rtp_extension extension, enum ast_rtp_extension_direction direction, const pjmedia_sdp_session *sdp)
	Enable an RTP extension on an RTP session.

static int	ensure_payload_type_mapping (struct ast_rtp_codecs codecs, int payload, struct ast_format format)
	Ensure an RTP codecs structure can receive a format using a payload type.

static pjmedia_sdp_attr *	generate_fmtp_attr (pj_pool_t pool, struct ast_format format, int rtp_code)

static pjmedia_sdp_attr *	generate_rtpmap_attr (struct ast_sip_session session, pjmedia_sdp_media media, pj_pool_t pool, int rtp_code, int asterisk_format, struct ast_format format, int code)

static pjmedia_sdp_attr *	generate_rtpmap_attr2 (struct ast_sip_session session, pjmedia_sdp_media media, pj_pool_t pool, int rtp_code, int asterisk_format, struct ast_format format, int code, int sample_rate)

static int	generating_local_offer (const struct ast_sip_session *session)
	Determine whether the SDP currently being generated is a local offer.

static void	get_codecs (struct ast_sip_session session, const struct pjmedia_sdp_media stream, struct ast_rtp_codecs codecs, struct ast_sip_session_media session_media, struct ast_format_cap *astformats)

static enum ast_sip_session_media_encryption	get_media_encryption_type (pj_str_t transport, const struct pjmedia_sdp_media stream, unsigned int optimistic)
	figure out media transport encryption type from the media transport string

static int	load_module (void)
	Load the module.

static struct ast_frame *	media_session_rtcp_read_callback (struct ast_sip_session session, struct ast_sip_session_media session_media)

static struct ast_frame *	media_session_rtp_read_callback (struct ast_sip_session session, struct ast_sip_session_media session_media)

static int	media_session_rtp_write_callback (struct ast_sip_session session, struct ast_sip_session_media session_media, struct ast_frame *frame)

static int	media_stream_has_crypto (const struct pjmedia_sdp_media *stream)
	figure out if media stream has crypto lines for sdes

static int	negotiate_incoming_sdp_stream (struct ast_sip_session session, struct ast_sip_session_media session_media, const pjmedia_sdp_session sdp, int index, struct ast_stream asterisk_stream)
	Function which negotiates an incoming media stream.

static int	parse_dtls_attrib (struct ast_sip_session_media session_media, const struct pjmedia_sdp_session sdp, const struct pjmedia_sdp_media *stream)

static int	payload_type_compatible (struct ast_rtp_codecs codecs, int payload, struct ast_format format)
	Check whether a payload slot is either free or already mapped to the requested format.

static void	process_extmap_attributes (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_media *remote_stream)
	Function which processes extmap attributes in a stream.

static void	process_ice_attributes (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_session remote, const struct pjmedia_sdp_media remote_stream)
	Function which processes ICE attributes in an audio stream.

static void	process_ice_auth_attrb (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_session remote, const struct pjmedia_sdp_media remote_stream)

static void	process_ssrc_attributes (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_media *remote_stream)
	Function which processes ssrc attributes in a stream.

static int	rtp_check_timeout (const void *data)
	Check whether RTP is being received or not.

static int	send_keepalive (const void *data)

static int	set_caps (struct ast_sip_session session, struct ast_sip_session_media session_media, struct ast_sip_session_media session_media_transport, const struct pjmedia_sdp_media stream, int is_offer, struct ast_stream *asterisk_stream)

static void	set_ice_components (struct ast_sip_session session, struct ast_sip_session_media session_media)

static struct ast_format_cap *	set_incoming_call_offer_cap (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_media *stream)

static void	set_session_media_remotely_held (struct ast_sip_session_media session_media, const struct ast_sip_session session, const pjmedia_sdp_media media, const struct ast_stream stream, const struct ast_sockaddr *addrs)

static int	setup_dtls_srtp (struct ast_sip_session session, struct ast_sip_session_media session_media)

static int	setup_media_encryption (struct ast_sip_session session, struct ast_sip_session_media session_media, const struct pjmedia_sdp_session sdp, const struct pjmedia_sdp_media stream)

static int	setup_sdes_srtp (struct ast_sip_session_media session_media, const struct pjmedia_sdp_media stream)

static int	setup_srtp (struct ast_sip_session_media *session_media)

static void	stream_destroy (struct ast_sip_session_media *session_media)
	Function which destroys the RTP instance when session ends.

static void	stream_stop (struct ast_sip_session_media *session_media)
	Function which stops the RTP instance.

static int	unload_module (void)
	Unloads the sdp RTP/AVP module from Asterisk.

static int	video_info_incoming_request (struct ast_sip_session session, struct pjsip_rx_data rdata)

Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "PJSIP SDP RTP/AVP stream handler" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_pjsip,res_pjsip_session", }

static struct ast_sockaddr	address_rtp
	Address for RTP.

static const struct ast_module_info *	ast_module_info = &__mod_info

static struct ast_sip_session_sdp_handler	audio_sdp_handler
	SDP handler for 'audio' media stream.

static struct ast_sched_context *	sched
	Scheduler for RTCP purposes.

static const char	STR_AUDIO [] = "audio"

static const char	STR_VIDEO [] = "video"

static struct ast_sip_session_supplement	video_info_supplement

static struct ast_sip_session_sdp_handler	video_sdp_handler
	SDP handler for 'video' media stream.

Detailed Description

SIP SDP media stream handling.

Author: Joshua Colp jcolp.nosp@m.@dig.nosp@m.ium.c.nosp@m.om

Definition in file res_pjsip_sdp_rtp.c.

Function Documentation

◆ __reg_module()

static void __reg_module ( void )

static

Definition at line 2735 of file res_pjsip_sdp_rtp.c.

◆ __unreg_module()

static void __unreg_module ( void )

static

Definition at line 2735 of file res_pjsip_sdp_rtp.c.

◆ add_crypto_to_stream()

static int add_crypto_to_stream	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		pj_pool_t *	pool,
		pjmedia_sdp_media *	media
	)

static

Definition at line 1823 of file res_pjsip_sdp_rtp.c.

{
    pj_str_t stmp;
    pjmedia_sdp_attr *attr;
    enum ast_rtp_dtls_hash hash;
    const char *crypto_attribute;
    struct ast_rtp_engine_dtls *dtls;
    struct ast_sdp_srtp *tmp;
    static const pj_str_t STR_NEW = { "new", 3 };
    static const pj_str_t STR_EXISTING = { "existing", 8 };
    static const pj_str_t STR_ACTIVE = { "active", 6 };
    static const pj_str_t STR_PASSIVE = { "passive", 7 };
    static const pj_str_t STR_ACTPASS = { "actpass", 7 };
    static const pj_str_t STR_HOLDCONN = { "holdconn", 8 };
    static const pj_str_t STR_MEDSECREQ = { "requested", 9 };
    enum ast_rtp_dtls_setup setup;
 
    switch (session_media->encryption) {
    case AST_SIP_MEDIA_ENCRYPT_NONE:
    case AST_SIP_MEDIA_TRANSPORT_INVALID:
        break;
    case AST_SIP_MEDIA_ENCRYPT_SDES:
        if (!session_media->srtp) {
            session_media->srtp = ast_sdp_srtp_alloc();
            if (!session_media->srtp) {
                return -1;
            }
        }
 
        tmp = session_media->srtp;
 
        do {
            crypto_attribute = ast_sdp_srtp_get_attrib(tmp,
                0 /* DTLS running? No */,
                session->endpoint->media.rtp.srtp_tag_32 /* 32 byte tag length? */);
            if (!crypto_attribute) {
                /* No crypto attribute to add, bad news */
                return -1;
            }
 
            attr = pjmedia_sdp_attr_create(pool, "crypto",
                pj_cstr(&stmp, crypto_attribute));
            media->attr[media->attr_count++] = attr;
        } while ((tmp = AST_LIST_NEXT(tmp, sdp_srtp_list)));
 
        if (session->endpoint->security_negotiation == AST_SIP_SECURITY_NEG_MEDIASEC) {
            attr = pjmedia_sdp_attr_create(pool, "3ge2ae", &STR_MEDSECREQ);
            media->attr[media->attr_count++] = attr;
        }
 
        break;
    case AST_SIP_MEDIA_ENCRYPT_DTLS:
        if (setup_dtls_srtp(session, session_media)) {
            return -1;
        }
 
        dtls = ast_rtp_instance_get_dtls(session_media->rtp);
        if (!dtls) {
            return -1;
        }
 
        switch (dtls->get_connection(session_media->rtp)) {
        case AST_RTP_DTLS_CONNECTION_NEW:
            attr = pjmedia_sdp_attr_create(pool, "connection", &STR_NEW);
            media->attr[media->attr_count++] = attr;
            break;
        case AST_RTP_DTLS_CONNECTION_EXISTING:
            attr = pjmedia_sdp_attr_create(pool, "connection", &STR_EXISTING);
            media->attr[media->attr_count++] = attr;
            break;
        default:
            break;
        }
 
        /* If this is an answer we need to use our current state, if it's an offer we need to use
         * the configured value.
         */
        if (session->inv_session->neg
            && pjmedia_sdp_neg_get_state(session->inv_session->neg) != PJMEDIA_SDP_NEG_STATE_DONE) {
            setup = dtls->get_setup(session_media->rtp);
        } else {
            setup = session->endpoint->media.rtp.dtls_cfg.default_setup;
        }
 
        switch (setup) {
        case AST_RTP_DTLS_SETUP_ACTIVE:
            attr = pjmedia_sdp_attr_create(pool, "setup", &STR_ACTIVE);
            media->attr[media->attr_count++] = attr;
            break;
        case AST_RTP_DTLS_SETUP_PASSIVE:
            attr = pjmedia_sdp_attr_create(pool, "setup", &STR_PASSIVE);
            media->attr[media->attr_count++] = attr;
            break;
        case AST_RTP_DTLS_SETUP_ACTPASS:
            attr = pjmedia_sdp_attr_create(pool, "setup", &STR_ACTPASS);
            media->attr[media->attr_count++] = attr;
            break;
        case AST_RTP_DTLS_SETUP_HOLDCONN:
            attr = pjmedia_sdp_attr_create(pool, "setup", &STR_HOLDCONN);
            break;
        default:
            break;
        }
 
        hash = dtls->get_fingerprint_hash(session_media->rtp);
        crypto_attribute = dtls->get_fingerprint(session_media->rtp);
        if (crypto_attribute && (hash == AST_RTP_DTLS_HASH_SHA1 || hash == AST_RTP_DTLS_HASH_SHA256)) {
            RAII_VAR(struct ast_str *, fingerprint, ast_str_create(64), ast_free);
            if (!fingerprint) {
                return -1;
            }
 
            if (hash == AST_RTP_DTLS_HASH_SHA1) {
                ast_str_set(&fingerprint, 0, "SHA-1 %s", crypto_attribute);
            } else {
                ast_str_set(&fingerprint, 0, "SHA-256 %s", crypto_attribute);
            }
 
            attr = pjmedia_sdp_attr_create(pool, "fingerprint", pj_cstr(&stmp, ast_str_buffer(fingerprint)));
            media->attr[media->attr_count++] = attr;
        }
        break;
    }
 
    return 0;
}

References ast_free, AST_LIST_NEXT, AST_RTP_DTLS_CONNECTION_EXISTING, AST_RTP_DTLS_CONNECTION_NEW, AST_RTP_DTLS_HASH_SHA1, AST_RTP_DTLS_HASH_SHA256, AST_RTP_DTLS_SETUP_ACTIVE, AST_RTP_DTLS_SETUP_ACTPASS, AST_RTP_DTLS_SETUP_HOLDCONN, AST_RTP_DTLS_SETUP_PASSIVE, ast_rtp_instance_get_dtls(), ast_sdp_srtp_alloc(), ast_sdp_srtp_get_attrib(), AST_SIP_MEDIA_ENCRYPT_DTLS, AST_SIP_MEDIA_ENCRYPT_NONE, AST_SIP_MEDIA_ENCRYPT_SDES, AST_SIP_MEDIA_TRANSPORT_INVALID, AST_SIP_SECURITY_NEG_MEDIASEC, ast_str_buffer(), ast_str_create, ast_str_set(), ast_sip_session_media::encryption, ast_rtp_engine_dtls::get_connection, ast_rtp_engine_dtls::get_fingerprint, ast_rtp_engine_dtls::get_fingerprint_hash, ast_rtp_engine_dtls::get_setup, RAII_VAR, ast_sip_session_media::rtp, ast_sdp_srtp::sdp_srtp_list, session, setup_dtls_srtp(), and ast_sip_session_media::srtp.

Referenced by create_outgoing_sdp_stream().

◆ add_extmap_to_stream()

static void add_extmap_to_stream	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		pj_pool_t *	pool,
		pjmedia_sdp_media *	media
	)

static

Definition at line 1587 of file res_pjsip_sdp_rtp.c.

{
    int idx;
    char extmap_value[256];
 
    if (!session->endpoint->media.webrtc || session_media->type != AST_MEDIA_TYPE_VIDEO) {
        return;
    }
 
    /* RTP extension local unique identifiers start at '1' */
    for (idx = 1; idx <= ast_rtp_instance_extmap_count(session_media->rtp); ++idx) {
        enum ast_rtp_extension extension = ast_rtp_instance_extmap_get_extension(session_media->rtp, idx);
        const char *direction_str = "";
        pj_str_t stmp;
        pjmedia_sdp_attr *attr;
 
        /* If this is an unsupported RTP extension we can't place it into the SDP */
        if (extension == AST_RTP_EXTENSION_UNSUPPORTED) {
            continue;
        }
 
        switch (ast_rtp_instance_extmap_get_direction(session_media->rtp, idx)) {
        case AST_RTP_EXTENSION_DIRECTION_SENDRECV:
            /* Lack of a direction indicates sendrecv, so we leave it out */
            direction_str = "";
            break;
        case AST_RTP_EXTENSION_DIRECTION_SENDONLY:
            direction_str = "/sendonly";
            break;
        case AST_RTP_EXTENSION_DIRECTION_RECVONLY:
            direction_str = "/recvonly";
            break;
        case AST_RTP_EXTENSION_DIRECTION_NONE:
            /* It is impossible for a "none" direction extension to be negotiated but just in case
             * we treat it as inactive.
             */
        case AST_RTP_EXTENSION_DIRECTION_INACTIVE:
            direction_str = "/inactive";
            break;
        }
 
        snprintf(extmap_value, sizeof(extmap_value), "%d%s %s", idx, direction_str,
            ast_rtp_instance_extmap_get_uri(session_media->rtp, idx));
        attr = pjmedia_sdp_attr_create(pool, "extmap", pj_cstr(&stmp, extmap_value));
        pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
    }
}

References AST_MEDIA_TYPE_VIDEO, AST_RTP_EXTENSION_DIRECTION_INACTIVE, AST_RTP_EXTENSION_DIRECTION_NONE, AST_RTP_EXTENSION_DIRECTION_RECVONLY, AST_RTP_EXTENSION_DIRECTION_SENDONLY, AST_RTP_EXTENSION_DIRECTION_SENDRECV, AST_RTP_EXTENSION_UNSUPPORTED, ast_rtp_instance_extmap_count(), ast_rtp_instance_extmap_get_direction(), ast_rtp_instance_extmap_get_extension(), ast_rtp_instance_extmap_get_uri(), ast_sip_session_media::rtp, session, and ast_sip_session_media::type.

Referenced by create_outgoing_sdp_stream().

◆ add_ice_to_stream()

static void add_ice_to_stream	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		pj_pool_t *	pool,
		pjmedia_sdp_media *	media,
		unsigned int	include_candidates
	)

static

Function which adds ICE attributes to a media stream.

Definition at line 908 of file res_pjsip_sdp_rtp.c.

{
    struct ast_rtp_engine_ice *ice;
    struct ao2_container *candidates;
    const char *username, *password;
    pj_str_t stmp;
    pjmedia_sdp_attr *attr;
    struct ao2_iterator it_candidates;
    struct ast_rtp_engine_ice_candidate *candidate;
 
    if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp))) {
        return;
    }
 
    if (!session_media->remote_ice) {
        ice->stop(session_media->rtp);
        return;
    }
 
    if ((username = ice->get_ufrag(session_media->rtp))) {
        attr = pjmedia_sdp_attr_create(pool, "ice-ufrag", pj_cstr(&stmp, username));
        media->attr[media->attr_count++] = attr;
    }
 
    if ((password = ice->get_password(session_media->rtp))) {
        attr = pjmedia_sdp_attr_create(pool, "ice-pwd", pj_cstr(&stmp, password));
        media->attr[media->attr_count++] = attr;
    }
 
    if (!include_candidates) {
        return;
    }
 
    candidates = ice->get_local_candidates(session_media->rtp);
    if (!candidates) {
        return;
    }
 
    it_candidates = ao2_iterator_init(candidates, 0);
    for (; (candidate = ao2_iterator_next(&it_candidates)); ao2_ref(candidate, -1)) {
        struct ast_str *attr_candidate = ast_str_create(128);
 
        ast_str_set(&attr_candidate, -1, "%s %u %s %d %s ", candidate->foundation, candidate->id, candidate->transport,
                    candidate->priority, ast_sockaddr_stringify_addr_remote(&candidate->address));
        ast_str_append(&attr_candidate, -1, "%s typ ", ast_sockaddr_stringify_port(&candidate->address));
 
        switch (candidate->type) {
            case AST_RTP_ICE_CANDIDATE_TYPE_HOST:
                ast_str_append(&attr_candidate, -1, "host");
                break;
            case AST_RTP_ICE_CANDIDATE_TYPE_SRFLX:
                ast_str_append(&attr_candidate, -1, "srflx");
                break;
            case AST_RTP_ICE_CANDIDATE_TYPE_RELAYED:
                ast_str_append(&attr_candidate, -1, "relay");
                break;
        }
 
        if (!ast_sockaddr_isnull(&candidate->relay_address)) {
            ast_str_append(&attr_candidate, -1, " raddr %s rport", ast_sockaddr_stringify_addr_remote(&candidate->relay_address));
            ast_str_append(&attr_candidate, -1, " %s", ast_sockaddr_stringify_port(&candidate->relay_address));
        }
 
        attr = pjmedia_sdp_attr_create(pool, "candidate", pj_cstr(&stmp, ast_str_buffer(attr_candidate)));
        media->attr[media->attr_count++] = attr;
 
        ast_free(attr_candidate);
    }
 
    ao2_iterator_destroy(&it_candidates);
    ao2_ref(candidates, -1);
}

References ast_rtp_engine_ice_candidate::address, ao2_iterator_destroy(), ao2_iterator_init(), ao2_iterator_next, ao2_ref, ast_free, AST_RTP_ICE_CANDIDATE_TYPE_HOST, AST_RTP_ICE_CANDIDATE_TYPE_RELAYED, AST_RTP_ICE_CANDIDATE_TYPE_SRFLX, ast_rtp_instance_get_ice(), ast_sockaddr_isnull(), ast_sockaddr_stringify_addr_remote(), ast_sockaddr_stringify_port(), ast_str_append(), ast_str_buffer(), ast_str_create, ast_str_set(), ast_rtp_engine_ice_candidate::foundation, ast_rtp_engine_ice::get_local_candidates, ast_rtp_engine_ice::get_password, ast_rtp_engine_ice::get_ufrag, ast_rtp_engine_ice_candidate::id, ast_rtp_engine_ice_candidate::priority, ast_rtp_engine_ice_candidate::relay_address, ast_sip_session_media::remote_ice, ast_sip_session_media::rtp, session, ast_rtp_engine_ice::stop, ast_rtp_engine_ice_candidate::transport, and ast_rtp_engine_ice_candidate::type.

Referenced by create_outgoing_sdp_stream().

◆ add_msid_to_stream()

static void add_msid_to_stream	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		pj_pool_t *	pool,
		pjmedia_sdp_media *	media,
		struct ast_stream *	stream
	)

static

Definition at line 1505 of file res_pjsip_sdp_rtp.c.

{
    pj_str_t stmp;
    pjmedia_sdp_attr *attr;
    char msid[(AST_UUID_STR_LEN * 2) + 2];
    const char *stream_label = ast_stream_get_metadata(stream, "SDP:LABEL");
 
    if (!session->endpoint->media.webrtc) {
        return;
    }
 
    if (ast_strlen_zero(session_media->mslabel)) {
        /* If this stream is grouped with another then use its media stream label if possible */
        if (ast_stream_get_group(stream) != -1) {
            struct ast_sip_session_media *group_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, ast_stream_get_group(stream));
 
            ast_copy_string(session_media->mslabel, group_session_media->mslabel, sizeof(session_media->mslabel));
        }
 
        if (ast_strlen_zero(session_media->mslabel)) {
            ast_uuid_generate_str(session_media->mslabel, sizeof(session_media->mslabel));
        }
    }
 
    if (ast_strlen_zero(session_media->label)) {
        ast_uuid_generate_str(session_media->label, sizeof(session_media->label));
        /* add for stream identification to replace stream_name */
        ast_stream_set_metadata(stream, "MSID:LABEL", session_media->label);
    }
 
    snprintf(msid, sizeof(msid), "%s %s", session_media->mslabel, session_media->label);
    ast_debug(3, "Stream msid: %p %s %s\n", stream,
        ast_codec_media_type2str(ast_stream_get_type(stream)), msid);
    attr = pjmedia_sdp_attr_create(pool, "msid", pj_cstr(&stmp, msid));
    pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
 
    /* 'label' must come after 'msid' */
    if (!ast_strlen_zero(stream_label)) {
        ast_debug(3, "Stream Label: %p %s %s\n", stream,
            ast_codec_media_type2str(ast_stream_get_type(stream)), stream_label);
        attr = pjmedia_sdp_attr_create(pool, "label", pj_cstr(&stmp, stream_label));
        pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
    }
}

References ast_codec_media_type2str(), ast_copy_string(), ast_debug, ast_stream_get_group(), ast_stream_get_metadata(), ast_stream_get_type(), ast_stream_set_metadata(), ast_strlen_zero(), ast_uuid_generate_str(), AST_UUID_STR_LEN, AST_VECTOR_GET, ast_sip_session_media::label, ast_sip_session_media::mslabel, and session.

Referenced by create_outgoing_sdp_stream().

◆ add_rtcp_fb_to_stream()

static void add_rtcp_fb_to_stream	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		pj_pool_t *	pool,
		pjmedia_sdp_media *	media
	)

static

Definition at line 1552 of file res_pjsip_sdp_rtp.c.

{
    pj_str_t stmp;
    pjmedia_sdp_attr *attr;
 
    if (!session->endpoint->media.webrtc) {
        return;
    }
 
    /* transport-cc is supposed to be for the entire transport, and any media sources so
     * while the header does not appear in audio streams and isn't negotiated there, we still
     * place this attribute in as Chrome does.
     */
    attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* transport-cc"));
    pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
 
    if (session_media->type != AST_MEDIA_TYPE_VIDEO) {
        return;
    }
 
    /*
     * For now just automatically add it the stream even though it hasn't
     * necessarily been negotiated.
     */
    attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* ccm fir"));
    pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
 
    attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* goog-remb"));
    pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
 
    attr = pjmedia_sdp_attr_create(pool, "rtcp-fb", pj_cstr(&stmp, "* nack"));
    pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
}

References AST_MEDIA_TYPE_VIDEO, session, and ast_sip_session_media::type.

Referenced by create_outgoing_sdp_stream().

◆ add_ssrc_to_stream()

static void add_ssrc_to_stream	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		pj_pool_t *	pool,
		pjmedia_sdp_media *	media
	)

static

Function which adds ssrc attributes to a media stream.

Definition at line 1435 of file res_pjsip_sdp_rtp.c.

{
    pj_str_t stmp;
    pjmedia_sdp_attr *attr;
    char tmp[128];
 
    if (!session->endpoint->media.bundle || session_media->bundle_group == -1) {
        return;
    }
 
    snprintf(tmp, sizeof(tmp), "%u cname:%s", ast_rtp_instance_get_ssrc(session_media->rtp), ast_rtp_instance_get_cname(session_media->rtp));
    attr = pjmedia_sdp_attr_create(pool, "ssrc", pj_cstr(&stmp, tmp));
    media->attr[media->attr_count++] = attr;
}

References ast_rtp_instance_get_cname(), ast_rtp_instance_get_ssrc(), session, and ast_rtp_engine_ice::start.

Referenced by create_outgoing_sdp_stream().

◆ apply_cap_to_bundled()

static int apply_cap_to_bundled	(	struct ast_sip_session_media *	session_media,
		struct ast_sip_session_media *	session_media_transport,
		struct ast_stream *	asterisk_stream,
		struct ast_format_cap *	joint
	)

static

Definition at line 588 of file res_pjsip_sdp_rtp.c.

{
    if (!joint) {
        return -1;
    }
 
    ast_stream_set_formats(asterisk_stream, joint);
 
    /* If this is a bundled stream then apply the payloads to RTP instance acting as transport to prevent conflicts */
    if (session_media_transport != session_media && session_media->bundled) {
        int index;
 
        for (index = 0; index < ast_format_cap_count(joint); ++index) {
            struct ast_format *format = ast_format_cap_get_format(joint, index);
            int rtp_code;
 
            /* Ensure this payload is in the bundle group transport codecs, this purposely doesn't check the return value for
             * things as the format is guaranteed to have a payload already.
             */
            rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0);
            ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media_transport->rtp), rtp_code, format);
 
            ao2_ref(format, -1);
        }
    }
 
    return 0;
}

References ao2_ref, ast_format_cap_count(), ast_format_cap_get_format(), ast_rtp_codecs_payload_code(), ast_rtp_codecs_payload_set_rx(), ast_rtp_instance_get_codecs(), ast_stream_set_formats(), ast_sip_session_media::bundled, and ast_sip_session_media::rtp.

Referenced by negotiate_incoming_sdp_stream(), and set_caps().

◆ apply_dtls_attrib()

static void apply_dtls_attrib	(	struct ast_sip_session_media *	session_media,
		pjmedia_sdp_attr *	attr
	)

static

Definition at line 1283 of file res_pjsip_sdp_rtp.c.

{
    struct ast_rtp_engine_dtls *dtls = ast_rtp_instance_get_dtls(session_media->rtp);
    pj_str_t *value;
 
    if (!attr->value.ptr || !dtls) {
        return;
    }
 
    value = pj_strtrim(&attr->value);
 
    if (!pj_strcmp2(&attr->name, "setup")) {
        if (!pj_stricmp2(value, "active")) {
            dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_ACTIVE);
        } else if (!pj_stricmp2(value, "passive")) {
            dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_PASSIVE);
        } else if (!pj_stricmp2(value, "actpass")) {
            dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_ACTPASS);
        } else if (!pj_stricmp2(value, "holdconn")) {
            dtls->set_setup(session_media->rtp, AST_RTP_DTLS_SETUP_HOLDCONN);
        } else {
            ast_log(LOG_WARNING, "Unsupported setup attribute value '%*s'\n", (int)value->slen, value->ptr);
        }
    } else if (!pj_strcmp2(&attr->name, "connection")) {
        if (!pj_stricmp2(value, "new")) {
            dtls->reset(session_media->rtp);
        } else if (!pj_stricmp2(value, "existing")) {
            /* Do nothing */
        } else {
            ast_log(LOG_WARNING, "Unsupported connection attribute value '%*s'\n", (int)value->slen, value->ptr);
        }
    } else if (!pj_strcmp2(&attr->name, "fingerprint")) {
        char hash_value[256], hash[32];
        char fingerprint_text[value->slen + 1];
        ast_copy_pj_str(fingerprint_text, value, sizeof(fingerprint_text));
            if (sscanf(fingerprint_text, "%31s %255s", hash, hash_value) == 2) {
            if (!strcasecmp(hash, "sha-1")) {
                dtls->set_fingerprint(session_media->rtp, AST_RTP_DTLS_HASH_SHA1, hash_value);
            } else if (!strcasecmp(hash, "sha-256")) {
                dtls->set_fingerprint(session_media->rtp, AST_RTP_DTLS_HASH_SHA256, hash_value);
            } else {
                ast_log(LOG_WARNING, "Unsupported fingerprint hash type '%s'\n",
                hash);
            }
        }
    }
}

References ast_copy_pj_str(), ast_log, AST_RTP_DTLS_HASH_SHA1, AST_RTP_DTLS_HASH_SHA256, AST_RTP_DTLS_SETUP_ACTIVE, AST_RTP_DTLS_SETUP_ACTPASS, AST_RTP_DTLS_SETUP_HOLDCONN, AST_RTP_DTLS_SETUP_PASSIVE, ast_rtp_instance_get_dtls(), LOG_WARNING, ast_rtp_engine_dtls::reset, ast_sip_session_media::rtp, ast_rtp_engine_dtls::set_fingerprint, ast_rtp_engine_dtls::set_setup, and value.

Referenced by parse_dtls_attrib().

◆ apply_negotiated_sdp_stream()

static int apply_negotiated_sdp_stream	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_session *	local,
		const struct pjmedia_sdp_session *	remote,
		int	index,
		struct ast_stream *	asterisk_stream
	)

static

Definition at line 2402 of file res_pjsip_sdp_rtp.c.

{
    RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free);
    struct pjmedia_sdp_media *remote_stream = remote->media[index];
    enum ast_media_type media_type = session_media->type;
    char host[NI_MAXHOST];
    int res;
    int rtp_timeout;
    struct ast_sip_session_media *session_media_transport;
    SCOPE_ENTER(1, "%s Stream: %s\n", ast_sip_session_get_name(session),
        ast_str_tmp(128, ast_stream_to_str(asterisk_stream, &STR_TMP)));
 
    if (!session->channel) {
        SCOPE_EXIT_RTN_VALUE(1, "No channel\n");
    }
 
    /* Ensure incoming transport is compatible with the endpoint's configuration */
    if (!session->endpoint->media.rtp.use_received_transport &&
        check_endpoint_media_transport(session->endpoint, remote_stream) == AST_SIP_MEDIA_TRANSPORT_INVALID) {
        SCOPE_EXIT_RTN_VALUE(-1, "Incompatible transport\n");
    }
 
    /* Create an RTP instance if need be */
    if (!session_media->rtp && create_rtp(session, session_media, local)) {
        SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create rtp\n");
    }
 
    process_ssrc_attributes(session, session_media, remote_stream);
    process_extmap_attributes(session, session_media, remote_stream);
 
    session_media_transport = ast_sip_session_media_get_transport(session, session_media);
 
    if (session_media_transport == session_media || !session_media->bundled) {
        session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(remote_stream, "rtcp-mux", NULL) != NULL);
        set_ice_components(session, session_media);
 
        enable_rtcp(session, session_media, remote_stream);
 
        res = setup_media_encryption(session, session_media, remote, remote_stream);
        if (!session->endpoint->media.rtp.encryption_optimistic && res) {
            /* If optimistic encryption is disabled and crypto should have been enabled but was not
             * this session must fail.
             */
            SCOPE_EXIT_RTN_VALUE(-1, "Incompatible crypto\n");
        }
 
        if (!remote_stream->conn && !remote->conn) {
            SCOPE_EXIT_RTN_VALUE(1, "No connection info\n");
        }
 
        ast_copy_pj_str(host, remote_stream->conn ? &remote_stream->conn->addr : &remote->conn->addr, sizeof(host));
 
        /* Ensure that the address provided is valid */
        if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
            /* The provided host was actually invalid so we error out this negotiation */
            SCOPE_EXIT_RTN_VALUE(-1, "Host invalid\n");
        }
 
        /* Apply connection information to the RTP instance */
        ast_sockaddr_set_port(addrs, remote_stream->desc.port);
        ast_rtp_instance_set_remote_address(session_media->rtp, addrs);
 
        ast_sip_session_media_set_write_callback(session, session_media, media_session_rtp_write_callback);
        ast_sip_session_media_add_read_callback(session, session_media, ast_rtp_instance_fd(session_media->rtp, 0),
            media_session_rtp_read_callback);
        if (!session->endpoint->media.rtcp_mux || !session_media->remote_rtcp_mux) {
            ast_sip_session_media_add_read_callback(session, session_media, ast_rtp_instance_fd(session_media->rtp, 1),
                media_session_rtcp_read_callback);
        }
 
        /* If ICE support is enabled find all the needed attributes */
        process_ice_attributes(session, session_media, remote, remote_stream);
    } else {
        /* This is bundled with another session, so mark it as such */
        ast_rtp_instance_bundle(session_media->rtp, session_media_transport->rtp);
        ast_sip_session_media_set_write_callback(session, session_media, media_session_rtp_write_callback);
        enable_rtcp(session, session_media, remote_stream);
    }
 
    if (set_caps(session, session_media, session_media_transport, remote_stream, 0, asterisk_stream)) {
        SCOPE_EXIT_RTN_VALUE(-1, "set_caps failed\n");
    }
 
    /* Set the channel uniqueid on the RTP instance now that it is becoming active */
    ast_channel_lock(session->channel);
    ast_rtp_instance_set_channel_id(session_media->rtp, ast_channel_uniqueid(session->channel));
    ast_channel_unlock(session->channel);
 
    /* Ensure the RTP instance is active */
    ast_rtp_instance_set_stream_num(session_media->rtp, ast_stream_get_position(asterisk_stream));
    ast_rtp_instance_activate(session_media->rtp);
 
    /* audio stream handles music on hold */
    if (media_type != AST_MEDIA_TYPE_AUDIO && media_type != AST_MEDIA_TYPE_VIDEO) {
        if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE)
            && (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) {
            ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER);
        }
        SCOPE_EXIT_RTN_VALUE(1, "moh\n");
    }
 
    set_session_media_remotely_held(session_media, session, remote_stream, asterisk_stream, addrs);
 
    if (session_media->remotely_held_changed) {
        if (session_media->remotely_held) {
            /* The remote side has put us on hold */
            if (!session->endpoint->suppress_moh_on_sendonly) {
                ast_queue_hold(session->channel, session->endpoint->mohsuggest);
                ast_rtp_instance_stop(session_media->rtp);
                ast_queue_frame(session->channel, &ast_null_frame);
            }
            session_media->remotely_held_changed = 0;
        } else {
            /* The remote side has taken us off hold */
            if (!session->endpoint->suppress_moh_on_sendonly) {
                ast_queue_unhold(session->channel);
                ast_queue_frame(session->channel, &ast_null_frame);
            }
            session_media->remotely_held_changed = 0;
        }
    } else if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE)
        && (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) {
        ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER);
    }
 
    /* This purposely resets the encryption to the configured in case it gets added later */
    session_media->encryption = session->endpoint->media.rtp.encryption;
 
    if (session->endpoint->media.rtp.keepalive > 0 &&
        (session_media->type == AST_MEDIA_TYPE_AUDIO ||
            session_media->type == AST_MEDIA_TYPE_VIDEO)) {
        ast_rtp_instance_set_keepalive(session_media->rtp, session->endpoint->media.rtp.keepalive);
        /* Schedule the initial keepalive early in case this is being used to punch holes through
         * a NAT. This way there won't be an awkward delay before media starts flowing in some
         * scenarios.
         */
        AST_SCHED_DEL(sched, session_media->keepalive_sched_id);
        session_media->keepalive_sched_id = ast_sched_add_variable(sched, 500, send_keepalive,
            session_media, 1);
    }
 
    /* As the channel lock is not held during this process the scheduled item won't block if
     * it is hanging up the channel at the same point we are applying this negotiated SDP.
     */
    AST_SCHED_DEL(sched, session_media->timeout_sched_id);
 
    /* Due to the fact that we only ever have one scheduled timeout item for when we are both
     * off hold and on hold we don't need to store the two timeouts differently on the RTP
     * instance itself.
     */
    ast_rtp_instance_set_timeout(session_media->rtp, 0);
    if (session->endpoint->media.rtp.timeout && !session_media->remotely_held && !session_media->locally_held) {
        ast_rtp_instance_set_timeout(session_media->rtp, session->endpoint->media.rtp.timeout);
    } else if (session->endpoint->media.rtp.timeout_hold && (session_media->remotely_held || session_media->locally_held)) {
        ast_rtp_instance_set_timeout(session_media->rtp, session->endpoint->media.rtp.timeout_hold);
    }
 
    rtp_timeout = ast_rtp_instance_get_timeout(session_media->rtp);
 
    if (rtp_timeout) {
        session_media->timeout_sched_id = ast_sched_add_variable(sched, rtp_timeout*1000, rtp_check_timeout,
            session_media, 1);
    }
 
    SCOPE_EXIT_RTN_VALUE(1, "Handled\n");
}

◆ assign_unique_bundle_payload()

static int assign_unique_bundle_payload	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		struct ast_sip_session_media *	session_media_transport,
		const pjmedia_sdp_session *	sdp,
		const pjmedia_sdp_media *	media,
		struct ast_format *	format
	)

static

Assign a payload type that is unique across the generated BUNDLE group.

Definition at line 520 of file res_pjsip_sdp_rtp.c.

{
    struct ast_rtp_codecs *transport_codecs =
        ast_rtp_instance_get_codecs(session_media_transport->rtp);
    struct ast_rtp_codecs *stream_codecs =
        ast_rtp_instance_get_codecs(session_media->rtp);
    int preferred;
    int candidate;
    int offset;
 
    preferred = ast_rtp_codecs_payload_code(stream_codecs, 1, format, 0);
    if (preferred < 0) {
        return -1;
    }
 
    /* Preserve the stream's normal payload when it is not used by another m= section. */
    if (!bundle_payload_type_used(session, sdp, media, session_media->bundle_group, preferred)
        && payload_type_compatible(transport_codecs, preferred, format)
        && payload_type_compatible(stream_codecs, preferred, format)) {
        candidate = preferred;
        goto configure;
    }
 
    /* Remap duplicate static or dynamic payloads into the dynamic range. */
    candidate = preferred >= AST_RTP_PT_FIRST_DYNAMIC
        ? preferred + 1 : AST_RTP_PT_FIRST_DYNAMIC;
    for (offset = 0; offset < AST_RTP_MAX_PT - AST_RTP_PT_FIRST_DYNAMIC; ++offset) {
        if (candidate >= AST_RTP_MAX_PT) {
            candidate = AST_RTP_PT_FIRST_DYNAMIC;
        }
 
        if (!bundle_payload_type_used(session, sdp, media, session_media->bundle_group, candidate)
            && payload_type_compatible(transport_codecs, candidate, format)
            && payload_type_compatible(stream_codecs, candidate, format)) {
            goto configure;
        }
 
        ++candidate;
    }
 
    ast_log(LOG_WARNING, "No unique dynamic RTP payload type available for bundled stream %s format %s\n",
        S_OR(session_media->mid, "<unknown>"), ast_format_get_name(format));
    return -1;
 
configure:
    if (ensure_payload_type_mapping(transport_codecs, candidate, format)
        || ensure_payload_type_mapping(stream_codecs, candidate, format)) {
        ast_log(LOG_WARNING, "Unable to configure RTP payload type %d for bundled stream %s format %s\n",
            candidate, S_OR(session_media->mid, "<unknown>"), ast_format_get_name(format));
        return -1;
    }
 
    ast_debug(2, "Assigned RTP payload type %d to bundled stream %s format %s\n",
        candidate, S_OR(session_media->mid, "<unknown>"), ast_format_get_name(format));
    return candidate;
}

References ast_debug, ast_format_get_name(), ast_log, ast_rtp_codecs_payload_code(), ast_rtp_instance_get_codecs(), AST_RTP_MAX_PT, AST_RTP_PT_FIRST_DYNAMIC, ast_sip_session_media::bundle_group, bundle_payload_type_used(), ensure_payload_type_mapping(), LOG_WARNING, ast_sip_session_media::mid, payload_type_compatible(), ast_sip_session_media::rtp, S_OR, and session.

Referenced by create_outgoing_sdp_stream().

◆ AST_MODULE_SELF_SYM()

struct ast_module * AST_MODULE_SELF_SYM ( void )

Definition at line 2735 of file res_pjsip_sdp_rtp.c.

◆ bundle_payload_type_used()

static int bundle_payload_type_used	(	struct ast_sip_session *	session,
		const pjmedia_sdp_session *	sdp,
		const pjmedia_sdp_media *	media,
		int	bundle_group,
		int	payload
	)

static

Determine whether a payload type is already advertised in a BUNDLE group.

Definition at line 446 of file res_pjsip_sdp_rtp.c.

{
    int media_index;
 
    for (media_index = 0; media_index < sdp->media_count; ++media_index) {
        struct ast_sip_session_media *other_session_media;
        int format_index;
 
        if (media_index >= AST_VECTOR_SIZE(&session->pending_media_state->sessions)) {
            break;
        }
 
        other_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, media_index);
        if (!other_session_media || other_session_media->bundle_group != bundle_group) {
            continue;
        }
 
        for (format_index = 0; format_index < sdp->media[media_index]->desc.fmt_count; ++format_index) {
            if (pj_strtoul(&sdp->media[media_index]->desc.fmt[format_index]) == payload) {
                return 1;
            }
        }
    }
 
    /* The current media section is not in the SDP yet, so inspect it separately. */
    for (media_index = 0; media_index < media->desc.fmt_count; ++media_index) {
        if (pj_strtoul(&media->desc.fmt[media_index]) == payload) {
            return 1;
        }
    }
 
    return 0;
}

References AST_VECTOR_GET, AST_VECTOR_SIZE, ast_sip_session_media::bundle_group, and session.

Referenced by assign_unique_bundle_payload().

◆ change_outgoing_sdp_stream_media_address()

static void change_outgoing_sdp_stream_media_address	(	pjsip_tx_data *	tdata,
		struct pjmedia_sdp_media *	stream,
		struct ast_sip_transport *	transport
	)

static

Function which updates the media stream with external media address, if applicable.

Definition at line 2572 of file res_pjsip_sdp_rtp.c.

{
    RAII_VAR(struct ast_sip_transport_state *, transport_state, ast_sip_get_transport_state(ast_sorcery_object_get_id(transport)), ao2_cleanup);
    char host[NI_MAXHOST];
    struct ast_sockaddr our_sdp_addr = { { 0, } };
 
    /* If the stream has been rejected there will be no connection line */
    if (!stream->conn || !transport_state) {
        return;
    }
 
    ast_copy_pj_str(host, &stream->conn->addr, sizeof(host));
    ast_sockaddr_parse(&our_sdp_addr, host, PARSE_PORT_FORBID);
 
    /* Reversed check here. We don't check the remote endpoint being
     * in our local net, but whether our outgoing session IP is
     * local. If it is not, we won't do rewriting. No localnet
     * configured? Always rewrite. */
    if (ast_sip_transport_is_nonlocal(transport_state, &our_sdp_addr) && transport_state->localnet) {
        return;
    }
    ast_debug(5, "Setting media address to %s\n", ast_sockaddr_stringify_addr_remote(&transport_state->external_media_address));
    pj_strdup2(tdata->pool, &stream->conn->addr, ast_sockaddr_stringify_addr_remote(&transport_state->external_media_address));
}

References ao2_cleanup, ast_copy_pj_str(), ast_debug, ast_sip_get_transport_state(), ast_sip_transport_is_nonlocal, ast_sockaddr_parse(), ast_sockaddr_stringify_addr_remote(), ast_sorcery_object_get_id(), PARSE_PORT_FORBID, RAII_VAR, and ast_sip_session_media::transport.

◆ check_endpoint_media_transport()

static enum ast_sip_session_media_encryption check_endpoint_media_transport	(	struct ast_sip_endpoint *	endpoint,
		const struct pjmedia_sdp_media *	stream
	)

static

Checks whether the encryption offered in SDP is compatible with the endpoint's configuration.

Definition at line 1203 of file res_pjsip_sdp_rtp.c.

{
    enum ast_sip_session_media_encryption incoming_encryption;
    char transport_end = stream->desc.transport.ptr[stream->desc.transport.slen - 1];
    unsigned int optimistic;
 
    if ((transport_end == 'F' && !endpoint->media.rtp.use_avpf)
        || (transport_end != 'F' && endpoint->media.rtp.use_avpf)) {
        return AST_SIP_MEDIA_TRANSPORT_INVALID;
    }
 
    incoming_encryption = get_media_encryption_type(stream->desc.transport, stream, &optimistic);
 
    if (incoming_encryption == endpoint->media.rtp.encryption) {
        return incoming_encryption;
    }
 
    if (endpoint->media.rtp.force_avp ||
        endpoint->media.rtp.encryption_optimistic) {
        return incoming_encryption;
    }
 
    /* If an optimistic offer has been made but encryption is not enabled consider it as having
     * no offer of crypto at all instead of invalid so the session proceeds.
     */
    if (optimistic) {
        return AST_SIP_MEDIA_ENCRYPT_NONE;
    }
 
    return AST_SIP_MEDIA_TRANSPORT_INVALID;
}

References AST_SIP_MEDIA_ENCRYPT_NONE, AST_SIP_MEDIA_TRANSPORT_INVALID, ast_sip_media_rtp_configuration::encryption, ast_sip_media_rtp_configuration::encryption_optimistic, ast_sip_media_rtp_configuration::force_avp, get_media_encryption_type(), ast_sip_endpoint::media, ast_sip_endpoint_media_configuration::rtp, and ast_sip_media_rtp_configuration::use_avpf.

Referenced by apply_negotiated_sdp_stream(), and negotiate_incoming_sdp_stream().

◆ check_ice_support()

static void check_ice_support	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	remote_stream
	)

static

Function which checks for ice attributes in an audio stream.

Definition at line 983 of file res_pjsip_sdp_rtp.c.

{
    struct ast_rtp_engine_ice *ice;
    const pjmedia_sdp_attr *attr;
    unsigned int attr_i;
 
    if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp))) {
        session_media->remote_ice = 0;
        return;
    }
 
    /* Find all of the candidates */
    for (attr_i = 0; attr_i < remote_stream->attr_count; ++attr_i) {
        attr = remote_stream->attr[attr_i];
        if (!pj_strcmp2(&attr->name, "candidate")) {
            session_media->remote_ice = 1;
            break;
        }
    }
 
    if (attr_i == remote_stream->attr_count) {
        session_media->remote_ice = 0;
    }
}

References ast_rtp_instance_get_ice(), session, and ast_rtp_engine_ice::start.

Referenced by negotiate_incoming_sdp_stream().

◆ create_outgoing_sdp_stream()

static int create_outgoing_sdp_stream	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		struct pjmedia_sdp_session *	sdp,
		const struct pjmedia_sdp_session *	remote,
		struct ast_stream *	stream
	)

static

Function which creates an outgoing stream.

Definition at line 1953 of file res_pjsip_sdp_rtp.c.

{
    pj_pool_t *pool = session->inv_session->pool_prov;
    static const pj_str_t STR_RTP_AVP = { "RTP/AVP", 7 };
    static const pj_str_t STR_IN = { "IN", 2 };
    static const pj_str_t STR_IP4 = { "IP4", 3};
    static const pj_str_t STR_IP6 = { "IP6", 3};
    static const pj_str_t STR_SENDRECV = { "sendrecv", 8 };
    static const pj_str_t STR_SENDONLY = { "sendonly", 8 };
    static const pj_str_t STR_INACTIVE = { "inactive", 8 };
    static const pj_str_t STR_RECVONLY = { "recvonly", 8 };
    pjmedia_sdp_media *media;
    const char *hostip = NULL;
    struct ast_sockaddr addr;
    char tmp[512];
    pj_str_t stmp;
    pjmedia_sdp_attr *attr;
    int index = 0;
    int noncodec = (session->dtmf == AST_SIP_DTMF_RFC_4733 || session->dtmf == AST_SIP_DTMF_AUTO || session->dtmf == AST_SIP_DTMF_AUTO_INFO) ? AST_RTP_DTMF : 0;
    int min_packet_size = 0, max_packet_size = 0;
    int rtp_code;
    RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
    enum ast_media_type media_type = session_media->type;
    struct ast_sip_session_media *session_media_transport;
    pj_sockaddr ip;
    int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
        ast_format_cap_count(session->direct_media_cap);
 
    /* Keep track of the sample rates for offered codecs so we can build matching
       RFC 2833/4733 payload offers. */
    AST_VECTOR(, int) sample_rates;
    /* In case we can't init the sample rates, still try to do the rest. */
    int build_dtmf_sample_rates = 1;
 
    SCOPE_ENTER(1, "%s Type: %s %s\n", ast_sip_session_get_name(session),
        ast_codec_media_type2str(media_type), ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
 
    media = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_media));
    if (!media) {
        SCOPE_EXIT_RTN_VALUE(-1, "Pool alloc failure\n");
    }
    pj_strdup2(pool, &media->desc.media, ast_codec_media_type2str(session_media->type));
 
    /* If this is a removed (or declined) stream OR if no formats exist then construct a minimal stream in SDP */
    if (ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED || !ast_stream_get_formats(stream) ||
        !ast_format_cap_count(ast_stream_get_formats(stream))) {
        media->desc.port = 0;
        media->desc.port_count = 1;
 
        if (remote && remote->media[ast_stream_get_position(stream)]) {
            pjmedia_sdp_media *remote_media = remote->media[ast_stream_get_position(stream)];
            int index;
 
            media->desc.transport = remote_media->desc.transport;
 
            /* Preserve existing behavior by copying the formats provided from the offer */
            for (index = 0; index < remote_media->desc.fmt_count; ++index) {
                media->desc.fmt[index] = remote_media->desc.fmt[index];
            }
            media->desc.fmt_count = remote_media->desc.fmt_count;
        } else {
            /* This is actually an offer so put a dummy payload in that is ignored and sane transport */
            media->desc.transport = STR_RTP_AVP;
            pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], "32");
        }
 
        sdp->media[sdp->media_count++] = media;
        ast_stream_set_state(stream, AST_STREAM_STATE_REMOVED);
 
        SCOPE_EXIT_RTN_VALUE(1, "Stream removed or no formats\n");
    }
 
    if (!session_media->rtp && create_rtp(session, session_media, sdp)) {
        SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create rtp\n");
    }
 
    /* If this stream has not been bundled already it is new and we need to ensure there is no SSRC conflict */
    if (session_media->bundle_group != -1 && !session_media->bundled) {
        for (index = 0; index < sdp->media_count; ++index) {
            struct ast_sip_session_media *other_session_media;
 
            other_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
            if (!other_session_media->rtp || other_session_media->bundle_group != session_media->bundle_group) {
                continue;
            }
 
            if (ast_rtp_instance_get_ssrc(session_media->rtp) == ast_rtp_instance_get_ssrc(other_session_media->rtp)) {
                ast_rtp_instance_change_source(session_media->rtp);
                /* Start the conflict check over again */
                index = -1;
                continue;
            }
        }
    }
 
    session_media_transport = ast_sip_session_media_get_transport(session, session_media);
 
    if (session_media_transport == session_media || !session_media->bundled) {
        set_ice_components(session, session_media);
        enable_rtcp(session, session_media, NULL);
 
        /* Crypto has to be added before setting the media transport so that SRTP is properly
         * set up according to the configuration. This ends up changing the media transport.
         */
        if (add_crypto_to_stream(session, session_media, pool, media)) {
            SCOPE_EXIT_RTN_VALUE(-1, "Couldn't add crypto\n");
        }
 
        if (pj_strlen(&session_media->transport)) {
            /* If a transport has already been specified use it */
            media->desc.transport = session_media->transport;
        } else {
            media->desc.transport = pj_str(ast_sdp_get_rtp_profile(
                /* Optimistic encryption places crypto in the normal RTP/AVP profile */
                !session->endpoint->media.rtp.encryption_optimistic &&
                    (session_media->encryption == AST_SIP_MEDIA_ENCRYPT_SDES),
                session_media->rtp, session->endpoint->media.rtp.use_avpf,
                session->endpoint->media.rtp.force_avp));
        }
 
        media->conn = pj_pool_zalloc(pool, sizeof(struct pjmedia_sdp_conn));
        if (!media->conn) {
            SCOPE_EXIT_RTN_VALUE(-1, "Pool alloc failure\n");
        }
 
        /* Add connection level details */
        if (direct_media_enabled) {
            hostip = ast_sockaddr_stringify_fmt(&session_media->direct_media_addr, AST_SOCKADDR_STR_ADDR);
        } else if (ast_strlen_zero(session->endpoint->media.address)) {
            hostip = ast_sip_get_host_ip_string(session->endpoint->media.rtp.ipv6 ? pj_AF_INET6() : pj_AF_INET());
        } else {
            hostip = session->endpoint->media.address;
        }
 
        if (ast_strlen_zero(hostip)) {
            ast_log(LOG_ERROR, "No local host IP available for stream %s\n",
                ast_codec_media_type2str(session_media->type));
            SCOPE_EXIT_RTN_VALUE(-1, "No local host ip\n");
        }
 
        media->conn->net_type = STR_IN;
        /* Assume that the connection will use IPv4 until proven otherwise */
        media->conn->addr_type = STR_IP4;
        pj_strdup2(pool, &media->conn->addr, hostip);
 
        if ((pj_sockaddr_parse(pj_AF_UNSPEC(), 0, &media->conn->addr, &ip) == PJ_SUCCESS) &&
            (ip.addr.sa_family == pj_AF_INET6())) {
            media->conn->addr_type = STR_IP6;
        }
 
        /* Add ICE attributes and candidates */
        add_ice_to_stream(session, session_media, pool, media, 1);
 
        ast_rtp_instance_get_local_address(session_media->rtp, &addr);
        media->desc.port = direct_media_enabled ? ast_sockaddr_port(&session_media->direct_media_addr) : (pj_uint16_t) ast_sockaddr_port(&addr);
        media->desc.port_count = 1;
    } else {
        pjmedia_sdp_media *bundle_group_stream = sdp->media[session_media_transport->stream_num];
 
        /* As this is in a bundle group it shares the same details as the group instance */
        media->desc.transport = bundle_group_stream->desc.transport;
        media->conn = bundle_group_stream->conn;
        media->desc.port = bundle_group_stream->desc.port;
 
        if (add_crypto_to_stream(session, session_media_transport, pool, media)) {
            SCOPE_EXIT_RTN_VALUE(-1, "Couldn't add crypto\n");
        }
 
        add_ice_to_stream(session, session_media_transport, pool, media, 0);
 
        enable_rtcp(session, session_media, NULL);
    }
 
    if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
        ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n",
            ast_codec_media_type2str(session_media->type));
        SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create caps\n");
    }
 
    if (direct_media_enabled) {
        ast_format_cap_get_compatible(session->endpoint->media.codecs, session->direct_media_cap, caps);
    } else {
        ast_format_cap_append_from_cap(caps, ast_stream_get_formats(stream), media_type);
    }
 
    /* Init the sample rates before we start adding them. Assume we will have at least one. */
    if (AST_VECTOR_INIT(&sample_rates, 1)) {
        ast_log(LOG_ERROR, "Unable to add dtmf formats to SDP!\n");
        build_dtmf_sample_rates = 0;
    }
 
    for (index = 0; index < ast_format_cap_count(caps); ++index) {
        struct ast_format *format = ast_format_cap_get_format(caps, index);
 
        if (ast_format_get_type(format) != media_type) {
            ao2_ref(format, -1);
            continue;
        }
 
        /* It is possible for some formats not to have SDP information available for them
         * and if this is the case, skip over them so the SDP can still be created.
         */
        if (!ast_rtp_lookup_sample_rate2(1, format, 0)) {
            ast_log(LOG_WARNING, "Format '%s' can not be added to SDP, consider disallowing it on endpoint '%s'\n",
                ast_format_get_name(format), ast_sorcery_object_get_id(session->endpoint));
            ao2_ref(format, -1);
            continue;
        }
 
        /* If this stream is not a transport we need to use the transport codecs structure for payload management to prevent
         * conflicts.
         */
        if (session_media_transport != session_media) {
            if (generating_local_offer(session)) {
                rtp_code = assign_unique_bundle_payload(session, session_media,
                    session_media_transport, sdp, media, format);
            } else {
                rtp_code = ast_rtp_codecs_payload_code(
                    ast_rtp_instance_get_codecs(session_media_transport->rtp), 1, format, 0);
                if (rtp_code != -1) {
                    /* Our instance has to match the offered payload number. */
                    ast_rtp_codecs_payload_set_rx(
                        ast_rtp_instance_get_codecs(session_media->rtp), rtp_code, format);
                }
            }
 
            if (rtp_code == -1) {
                ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format));
                ao2_ref(format, -1);
                continue;
            }
        } else {
            if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0)) == -1) {
                ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format));
                ao2_ref(format, -1);
                continue;
            }
        }
 
        if ((attr = generate_rtpmap_attr(session, media, pool, rtp_code, 1, format, 0))) {
            int i, added = 0;
            int newrate = ast_rtp_lookup_sample_rate2(1, format, 0);
            if (build_dtmf_sample_rates) {
                for (i = 0; i < AST_VECTOR_SIZE(&sample_rates); i++) {
                    /* Only add if we haven't already processed this sample rate. For instance
                        A-law and u-law 'share' one 8K DTMF payload type. */
                    if (newrate == AST_VECTOR_GET(&sample_rates, i)) {
                        added = 1;
                        break;
                    }
                }
 
                if (!added) {
                    AST_VECTOR_APPEND(&sample_rates, newrate);
                }
            }
            media->attr[media->attr_count++] = attr;
        }
 
        if ((attr = generate_fmtp_attr(pool, format, rtp_code))) {
            media->attr[media->attr_count++] = attr;
        }
 
        if (ast_format_get_maximum_ms(format) &&
            ((ast_format_get_maximum_ms(format) < max_packet_size) || !max_packet_size)) {
            max_packet_size = ast_format_get_maximum_ms(format);
        }
        ao2_ref(format, -1);
 
        if (media->desc.fmt_count == PJMEDIA_MAX_SDP_FMT) {
            break;
        }
    }
 
    /* Add non-codec formats */
    if (ast_sip_session_is_pending_stream_default(session, stream) && media_type != AST_MEDIA_TYPE_VIDEO
        && media->desc.fmt_count < PJMEDIA_MAX_SDP_FMT) {
        for (index = 1LL; index <= AST_RTP_MAX; index <<= 1) {
            if (!(noncodec & index)) {
                continue;
            }
 
            if (index != AST_RTP_DTMF) {
                rtp_code = ast_rtp_codecs_payload_code(
                                ast_rtp_instance_get_codecs(session_media->rtp), 0, NULL, index);
                if (rtp_code == -1) {
                    continue;
                } else if ((attr = generate_rtpmap_attr(session, media, pool, rtp_code, 0, NULL, index))) {
                    media->attr[media->attr_count++] = attr;
                }
            } else if (build_dtmf_sample_rates) {
                /*
                 * Walk through the possible bitrates for the RFC 2833/4733 digits and generate the rtpmap
                 * attributes.
                 */
                int i, found_default_offer = 0;
                for (i = 0; i < AST_VECTOR_SIZE(&sample_rates); i++) {
                    rtp_code = ast_rtp_codecs_payload_code_sample_rate(
                                    ast_rtp_instance_get_codecs(session_media->rtp), 0, NULL, index, AST_VECTOR_GET(&sample_rates, i));
 
                    if (rtp_code == -1) {
                        continue;
                    }
 
                    if (AST_VECTOR_GET(&sample_rates, i) == DEFAULT_DTMF_SAMPLE_RATE_MS) {
                        /* we found and added a default offer, so no need to include a default one.*/
                        found_default_offer = 1;
                    }
 
                    if ((attr = generate_rtpmap_attr2(session, media, pool, rtp_code, 0, NULL, index, AST_VECTOR_GET(&sample_rates, i)))) {
                        media->attr[media->attr_count++] = attr;
                        snprintf(tmp, sizeof(tmp), "%d 0-16", (rtp_code));
                        attr = pjmedia_sdp_attr_create(pool, "fmtp", pj_cstr(&stmp, tmp));
                        media->attr[media->attr_count++] = attr;
                    }
                }
 
                /* If we weren't able to add any matching RFC 2833/4733, assume this endpoint is using a
                 * mismatched 8K offer and try to add one as a fall-back/default.
                 */
                if (!found_default_offer) {
                    rtp_code = ast_rtp_codecs_payload_code_sample_rate(
                                    ast_rtp_instance_get_codecs(session_media->rtp), 0, NULL, index, DEFAULT_DTMF_SAMPLE_RATE_MS);
 
                    if (rtp_code != -1 && (attr = generate_rtpmap_attr2(session, media, pool, rtp_code, 0, NULL, index, DEFAULT_DTMF_SAMPLE_RATE_MS))) {
                        media->attr[media->attr_count++] = attr;
                        snprintf(tmp, sizeof(tmp), "%d 0-16", (rtp_code));
                        attr = pjmedia_sdp_attr_create(pool, "fmtp", pj_cstr(&stmp, tmp));
                        media->attr[media->attr_count++] = attr;
                    }
                }
            }
 
            if (media->desc.fmt_count == PJMEDIA_MAX_SDP_FMT) {
                break;
            }
        }
    }
 
    /* we are done with the sample rates */
    AST_VECTOR_FREE(&sample_rates);
 
    /* If no formats were actually added to the media stream don't add it to the SDP */
    if (!media->desc.fmt_count) {
        SCOPE_EXIT_RTN_VALUE(1, "No formats added to stream\n");
    }
 
    /* If ptime is set add it as an attribute */
    min_packet_size = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(session_media->rtp));
    if (!min_packet_size) {
        min_packet_size = ast_format_cap_get_framing(caps);
    }
    if (min_packet_size) {
        snprintf(tmp, sizeof(tmp), "%d", min_packet_size);
        attr = pjmedia_sdp_attr_create(pool, "ptime", pj_cstr(&stmp, tmp));
        media->attr[media->attr_count++] = attr;
    }
 
    if (max_packet_size) {
        snprintf(tmp, sizeof(tmp), "%d", max_packet_size);
        attr = pjmedia_sdp_attr_create(pool, "maxptime", pj_cstr(&stmp, tmp));
        media->attr[media->attr_count++] = attr;
    }
 
    attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr);
    if (session_media->locally_held) {
        if (session_media->remotely_held) {
            attr->name = STR_INACTIVE; /* To place on hold a recvonly stream, send inactive */
        } else {
            attr->name = STR_SENDONLY; /* Send sendonly to initate a local hold */
        }
    } else {
        if (session_media->remotely_held) {
            attr->name = STR_RECVONLY; /* Remote has sent sendonly, reply recvonly */
        } else if (ast_stream_get_state(stream) == AST_STREAM_STATE_SENDONLY) {
            attr->name = STR_SENDONLY; /* Stream has requested sendonly */
        } else if (ast_stream_get_state(stream) == AST_STREAM_STATE_RECVONLY) {
            attr->name = STR_RECVONLY; /* Stream has requested recvonly */
        } else if (ast_stream_get_state(stream) == AST_STREAM_STATE_INACTIVE) {
            attr->name = STR_INACTIVE; /* Stream has requested inactive */
        } else {
            attr->name = STR_SENDRECV; /* No hold in either direction */
        }
    }
    media->attr[media->attr_count++] = attr;
 
    /* If we've got rtcp-mux enabled, add it unless we received an offer without it */
    if (session->endpoint->media.rtcp_mux && session_media->remote_rtcp_mux) {
        attr = pjmedia_sdp_attr_create(pool, "rtcp-mux", NULL);
        pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
    }
 
    add_ssrc_to_stream(session, session_media, pool, media);
    add_msid_to_stream(session, session_media, pool, media, stream);
    add_rtcp_fb_to_stream(session, session_media, pool, media);
    add_extmap_to_stream(session, session_media, pool, media);
 
    /* Add the media stream to the SDP */
    sdp->media[sdp->media_count++] = media;
 
    SCOPE_EXIT_RTN_VALUE(1, "RC: 1\n");
}

◆ create_rtp()

static int create_rtp	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const pjmedia_sdp_session *	sdp
	)

static

Internal function which creates an RTP instance.

Definition at line 225 of file res_pjsip_sdp_rtp.c.

{
    struct ast_rtp_engine_ice *ice;
    struct ast_sockaddr temp_media_address;
    struct ast_sockaddr *media_address =  &address_rtp;
 
    if (session->endpoint->media.bind_rtp_to_media_address && !ast_strlen_zero(session->endpoint->media.address)) {
        if (ast_sockaddr_parse(&temp_media_address, session->endpoint->media.address, 0)) {
            ast_debug_rtp(1, "Endpoint %s: Binding RTP media to %s\n",
                ast_sorcery_object_get_id(session->endpoint),
                session->endpoint->media.address);
            media_address = &temp_media_address;
        } else {
            ast_debug_rtp(1, "Endpoint %s: RTP media address invalid: %s\n",
                ast_sorcery_object_get_id(session->endpoint),
                session->endpoint->media.address);
        }
    } else {
        struct ast_sip_transport *transport;
 
        transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport",
            session->endpoint->transport);
        if (transport) {
            struct ast_sip_transport_state *trans_state;
 
            trans_state = ast_sip_get_transport_state(ast_sorcery_object_get_id(transport));
            if (trans_state) {
                char hoststr[PJ_INET6_ADDRSTRLEN];
 
                pj_sockaddr_print(&trans_state->host, hoststr, sizeof(hoststr), 0);
                if (ast_sockaddr_parse(&temp_media_address, hoststr, 0)) {
                    ast_debug_rtp(1, "Transport %s bound to %s: Using it for RTP media.\n",
                        session->endpoint->transport, hoststr);
                    media_address = &temp_media_address;
                } else {
                    ast_debug_rtp(1, "Transport %s bound to %s: Invalid for RTP media.\n",
                        session->endpoint->transport, hoststr);
                }
                ao2_ref(trans_state, -1);
            }
            ao2_ref(transport, -1);
        }
    }
 
    if (session->endpoint->media.rtp.port_start && session->endpoint->media.rtp.port_end) {
        struct ast_rtp_instance_options options = {
            .port_start = session->endpoint->media.rtp.port_start,
            .port_end = session->endpoint->media.rtp.port_end,
        };
        if (!(session_media->rtp = ast_rtp_instance_new_with_options(
                session->endpoint->media.rtp.engine, sched, media_address, NULL,
                &options))) {
            ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s' with port range %u-%u\n",
                session->endpoint->media.rtp.engine,
                session->endpoint->media.rtp.port_start,
                session->endpoint->media.rtp.port_end);
            return -1;
        }
    } else {
        if (!(session_media->rtp = ast_rtp_instance_new(session->endpoint->media.rtp.engine, sched, media_address, NULL))) {
            ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n", session->endpoint->media.rtp.engine);
            return -1;
        }
    }
 
    ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_NAT, session->endpoint->media.rtp.symmetric);
    ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_ASYMMETRIC_CODEC, session->endpoint->asymmetric_rtp_codec);
 
    if (!session->endpoint->media.rtp.ice_support && (ice = ast_rtp_instance_get_ice(session_media->rtp))) {
        ice->stop(session_media->rtp);
    }
 
    if (session->dtmf == AST_SIP_DTMF_RFC_4733 || session->dtmf == AST_SIP_DTMF_AUTO || session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
        ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
        ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 1);
    } else if (session->dtmf == AST_SIP_DTMF_INBAND) {
        ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
    }
 
    if (session_media->type == AST_MEDIA_TYPE_AUDIO &&
            (session->endpoint->media.tos_audio || session->endpoint->media.cos_audio)) {
        ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_audio,
                session->endpoint->media.cos_audio, "SIP RTP Audio");
    } else if (session_media->type == AST_MEDIA_TYPE_VIDEO) {
        ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_RECV, session->endpoint->media.webrtc);
        ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RETRANS_SEND, session->endpoint->media.webrtc);
        ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_REMB, session->endpoint->media.webrtc);
        if (session->endpoint->media.webrtc) {
            enable_rtp_extension(session, session_media, AST_RTP_EXTENSION_ABS_SEND_TIME, AST_RTP_EXTENSION_DIRECTION_SENDRECV, sdp);
            enable_rtp_extension(session, session_media, AST_RTP_EXTENSION_TRANSPORT_WIDE_CC, AST_RTP_EXTENSION_DIRECTION_SENDRECV, sdp);
        }
        if (session->endpoint->media.tos_video || session->endpoint->media.cos_video) {
            ast_rtp_instance_set_qos(session_media->rtp, session->endpoint->media.tos_video,
                    session->endpoint->media.cos_video, "SIP RTP Video");
        }
    }
 
    ast_rtp_instance_set_last_rx(session_media->rtp, time(NULL));
 
    return 0;
}

References address_rtp, ao2_ref, ast_debug_rtp, ast_log, AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_VIDEO, AST_RTP_DTMF_MODE_INBAND, AST_RTP_DTMF_MODE_RFC2833, AST_RTP_EXTENSION_ABS_SEND_TIME, AST_RTP_EXTENSION_DIRECTION_SENDRECV, AST_RTP_EXTENSION_TRANSPORT_WIDE_CC, ast_rtp_instance_dtmf_mode_set(), ast_rtp_instance_get_ice(), ast_rtp_instance_new(), ast_rtp_instance_new_with_options(), ast_rtp_instance_set_last_rx(), ast_rtp_instance_set_prop(), ast_rtp_instance_set_qos(), AST_RTP_PROPERTY_ASYMMETRIC_CODEC, AST_RTP_PROPERTY_DTMF, AST_RTP_PROPERTY_NAT, AST_RTP_PROPERTY_REMB, AST_RTP_PROPERTY_RETRANS_RECV, AST_RTP_PROPERTY_RETRANS_SEND, AST_SIP_DTMF_AUTO, AST_SIP_DTMF_AUTO_INFO, AST_SIP_DTMF_INBAND, AST_SIP_DTMF_RFC_4733, ast_sip_get_sorcery(), ast_sip_get_transport_state(), ast_sockaddr_parse(), ast_sorcery_object_get_id(), ast_sorcery_retrieve_by_id(), ast_strlen_zero(), enable_rtp_extension(), ast_sip_transport_state::host, LOG_ERROR, NULL, options, ast_sip_session_media::rtp, session, ast_rtp_engine_ice::stop, ast_sip_transport_state::transport, and ast_sip_session_media::type.

Referenced by apply_negotiated_sdp_stream(), create_outgoing_sdp_stream(), and negotiate_incoming_sdp_stream().

◆ enable_rtcp()

static void enable_rtcp	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	remote_media
	)

static

Enable RTCP on an RTP session.

Definition at line 162 of file res_pjsip_sdp_rtp.c.

{
    enum ast_rtp_instance_rtcp rtcp_type;
 
    if (session->endpoint->media.rtcp_mux && session_media->remote_rtcp_mux) {
        rtcp_type = AST_RTP_INSTANCE_RTCP_MUX;
    } else {
        rtcp_type = AST_RTP_INSTANCE_RTCP_STANDARD;
    }
 
    ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_RTCP, rtcp_type);
}

References AST_RTP_INSTANCE_RTCP_MUX, AST_RTP_INSTANCE_RTCP_STANDARD, ast_rtp_instance_set_prop(), AST_RTP_PROPERTY_RTCP, ast_sip_session_media::remote_rtcp_mux, ast_sip_session_media::rtp, and session.

Referenced by apply_negotiated_sdp_stream(), create_outgoing_sdp_stream(), and negotiate_incoming_sdp_stream().

◆ enable_rtp_extension()

static void enable_rtp_extension	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		enum ast_rtp_extension	extension,
		enum ast_rtp_extension_direction	direction,
		const pjmedia_sdp_session *	sdp
	)

static

Enable an RTP extension on an RTP session.

Definition at line 179 of file res_pjsip_sdp_rtp.c.

{
    int id = -1;
 
    /* For a bundle group the local unique identifier space is shared across all streams within
     * it.
     */
    if (session_media->bundle_group != -1) {
        int index;
 
        for (index = 0; index < sdp->media_count; ++index) {
            struct ast_sip_session_media *other_session_media;
            int other_id;
 
            if (index >= AST_VECTOR_SIZE(&session->pending_media_state->sessions)) {
                break;
            }
 
            other_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
            if (!other_session_media->rtp || other_session_media->bundle_group != session_media->bundle_group) {
                continue;
            }
 
            other_id = ast_rtp_instance_extmap_get_id(other_session_media->rtp, extension);
            if (other_id == -1) {
                /* Worst case we have to fall back to the highest available free local unique identifier
                 * for the bundle group.
                 */
                other_id = ast_rtp_instance_extmap_count(other_session_media->rtp) + 1;
                if (id < other_id) {
                    id = other_id;
                }
                continue;
            }
 
            id = other_id;
            break;
        }
    }
 
    ast_rtp_instance_extmap_enable(session_media->rtp, id, extension, direction);
}

References ast_rtp_instance_extmap_count(), ast_rtp_instance_extmap_enable(), ast_rtp_instance_extmap_get_id(), AST_VECTOR_GET, AST_VECTOR_SIZE, ast_sip_session_media::bundle_group, ast_sip_session_media::rtp, and session.

Referenced by create_rtp().

◆ ensure_payload_type_mapping()

static int ensure_payload_type_mapping	(	struct ast_rtp_codecs *	codecs,
		int	payload,
		struct ast_format *	format
	)

static

Ensure an RTP codecs structure can receive a format using a payload type.

Definition at line 502 of file res_pjsip_sdp_rtp.c.

{
    struct ast_rtp_payload_type *type;
    int matches;
 
    type = ast_rtp_codecs_get_payload(codecs, payload);
    if (type) {
        matches = type->asterisk_format
            && ast_format_cmp(type->format, format) == AST_FORMAT_CMP_EQUAL;
        ao2_ref(type, -1);
        return matches ? 0 : -1;
    }
 
    return ast_rtp_codecs_payload_set_rx(codecs, payload, format) < 0 ? -1 : 0;
}

References ao2_ref, ast_format_cmp(), AST_FORMAT_CMP_EQUAL, ast_rtp_codecs_get_payload(), ast_rtp_codecs_payload_set_rx(), codecs, ast_rtp_payload_type::format, ast_rtp_payload_type::payload, and type.

Referenced by assign_unique_bundle_payload().

◆ generate_fmtp_attr()

static pjmedia_sdp_attr * generate_fmtp_attr	(	pj_pool_t *	pool,
		struct ast_format *	format,
		int	rtp_code
	)

static

Definition at line 882 of file res_pjsip_sdp_rtp.c.

{
    struct ast_str *fmtp0 = ast_str_alloca(256);
    pj_str_t fmtp1;
    pjmedia_sdp_attr *attr = NULL;
    char *tmp;
 
    ast_format_generate_sdp_fmtp(format, rtp_code, &fmtp0);
    if (ast_str_strlen(fmtp0)) {
        tmp = ast_str_buffer(fmtp0) + ast_str_strlen(fmtp0) - 1;
        /* remove any carriage return line feeds */
        while (*tmp == '\r' || *tmp == '\n') --tmp;
        *++tmp = '\0';
        /* ast...generate gives us everything, just need value */
        tmp = strchr(ast_str_buffer(fmtp0), ':');
        if (tmp && tmp[1] != '\0') {
            fmtp1 = pj_str(tmp + 1);
        } else {
            fmtp1 = pj_str(ast_str_buffer(fmtp0));
        }
        attr = pjmedia_sdp_attr_create(pool, "fmtp", &fmtp1);
    }
    return attr;
}

References ast_format_generate_sdp_fmtp(), ast_str_alloca, ast_str_buffer(), ast_str_strlen(), and NULL.

Referenced by create_outgoing_sdp_stream().

◆ generate_rtpmap_attr()

static pjmedia_sdp_attr * generate_rtpmap_attr	(	struct ast_sip_session *	session,
		pjmedia_sdp_media *	media,
		pj_pool_t *	pool,
		int	rtp_code,
		int	asterisk_format,
		struct ast_format *	format,
		int	code
	)

static

Definition at line 811 of file res_pjsip_sdp_rtp.c.

{
#ifndef HAVE_PJSIP_ENDPOINT_COMPACT_FORM
    extern pj_bool_t pjsip_use_compact_form;
#else
    pj_bool_t pjsip_use_compact_form = pjsip_cfg()->endpt.use_compact_form;
#endif
    pjmedia_sdp_rtpmap rtpmap;
    pjmedia_sdp_attr *attr = NULL;
    char tmp[64];
    enum ast_rtp_options options = session->endpoint->media.g726_non_standard ?
        AST_RTP_OPT_G726_NONSTANDARD : 0;
 
    snprintf(tmp, sizeof(tmp), "%d", rtp_code);
    pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], tmp);
 
    if (rtp_code <= AST_RTP_PT_LAST_STATIC && pjsip_use_compact_form) {
        return NULL;
    }
 
    rtpmap.pt = media->desc.fmt[media->desc.fmt_count - 1];
    rtpmap.clock_rate = ast_rtp_lookup_sample_rate2(asterisk_format, format, code);
    pj_strdup2(pool, &rtpmap.enc_name, ast_rtp_lookup_mime_subtype2(asterisk_format, format, code, options));
    if (!pj_stricmp2(&rtpmap.enc_name, "opus")) {
        pj_cstr(&rtpmap.param, "2");
    } else {
        pj_cstr(&rtpmap.param, NULL);
    }
 
    pjmedia_sdp_rtpmap_to_attr(pool, &rtpmap, &attr);
 
    return attr;
}

References ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), AST_RTP_OPT_G726_NONSTANDARD, AST_RTP_PT_LAST_STATIC, NULL, options, and session.

Referenced by create_outgoing_sdp_stream().

◆ generate_rtpmap_attr2()

static pjmedia_sdp_attr * generate_rtpmap_attr2	(	struct ast_sip_session *	session,
		pjmedia_sdp_media *	media,
		pj_pool_t *	pool,
		int	rtp_code,
		int	asterisk_format,
		struct ast_format *	format,
		int	code,
		int	sample_rate
	)

static

Definition at line 847 of file res_pjsip_sdp_rtp.c.

{
#ifndef HAVE_PJSIP_ENDPOINT_COMPACT_FORM
    extern pj_bool_t pjsip_use_compact_form;
#else
    pj_bool_t pjsip_use_compact_form = pjsip_cfg()->endpt.use_compact_form;
#endif
    pjmedia_sdp_rtpmap rtpmap;
    pjmedia_sdp_attr *attr = NULL;
    char tmp[64];
    enum ast_rtp_options options = session->endpoint->media.g726_non_standard ?
        AST_RTP_OPT_G726_NONSTANDARD : 0;
 
    snprintf(tmp, sizeof(tmp), "%d", rtp_code);
    pj_strdup2(pool, &media->desc.fmt[media->desc.fmt_count++], tmp);
 
    if (rtp_code <= AST_RTP_PT_LAST_STATIC && pjsip_use_compact_form) {
        return NULL;
    }
 
    rtpmap.pt = media->desc.fmt[media->desc.fmt_count - 1];
    rtpmap.clock_rate = sample_rate;
    pj_strdup2(pool, &rtpmap.enc_name, ast_rtp_lookup_mime_subtype2(asterisk_format, format, code, options));
    if (!pj_stricmp2(&rtpmap.enc_name, "opus")) {
        pj_cstr(&rtpmap.param, "2");
    } else {
        pj_cstr(&rtpmap.param, NULL);
    }
 
    pjmedia_sdp_rtpmap_to_attr(pool, &rtpmap, &attr);
 
    return attr;
}

References ast_rtp_lookup_mime_subtype2(), AST_RTP_OPT_G726_NONSTANDARD, AST_RTP_PT_LAST_STATIC, NULL, options, and session.

Referenced by create_outgoing_sdp_stream().

◆ generating_local_offer()

static int generating_local_offer ( const struct ast_sip_session * session )

static

Determine whether the SDP currently being generated is a local offer.

Definition at line 582 of file res_pjsip_sdp_rtp.c.

{
    return !session->inv_session->neg
        || pjmedia_sdp_neg_get_state(session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE;
}

References session.

Referenced by create_outgoing_sdp_stream().

◆ get_codecs()

static void get_codecs	(	struct ast_sip_session *	session,
		const struct pjmedia_sdp_media *	stream,
		struct ast_rtp_codecs *	codecs,
		struct ast_sip_session_media *	session_media,
		struct ast_format_cap *	astformats
	)

static

Definition at line 328 of file res_pjsip_sdp_rtp.c.

{
    pjmedia_sdp_attr *attr;
    pjmedia_sdp_rtpmap *rtpmap;
    pjmedia_sdp_fmtp fmtp;
    struct ast_format *format;
    int i, num = 0, tel_event = 0;
    char name[256];
    char media[20];
    char fmt_param[256];
    enum ast_rtp_options options = session->endpoint->media.g726_non_standard ?
        AST_RTP_OPT_G726_NONSTANDARD : 0;
    SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
 
    ast_rtp_codecs_payloads_initialize(codecs);
 
    ast_format_cap_remove_by_type(astformats, AST_MEDIA_TYPE_UNKNOWN);
 
    /* Iterate through provided formats */
    for (i = 0; i < stream->desc.fmt_count; ++i) {
        /* The payload is kept as a string for things like t38 but for video it is always numerical */
        ast_rtp_codecs_payloads_set_m_type(codecs, NULL, pj_strtoul(&stream->desc.fmt[i]));
        /* Look for the optional rtpmap attribute */
        if (!(attr = pjmedia_sdp_media_find_attr2(stream, "rtpmap", &stream->desc.fmt[i]))) {
            continue;
        }
 
        /* Interpret the attribute as an rtpmap */
        if ((pjmedia_sdp_attr_to_rtpmap(session->inv_session->pool_prov, attr, &rtpmap)) != PJ_SUCCESS) {
            continue;
        }
 
        ast_copy_pj_str(name, &rtpmap->enc_name, sizeof(name));
        if (strcmp(name, "telephone-event") == 0) {
            if (tel_event == 0) {
                int dtmf_rate = 0, dtmf_code = 0;
                char dtmf_pt[8];
                ast_copy_pj_str(dtmf_pt, &rtpmap->pt, sizeof(dtmf_pt));
                dtmf_code = atoi(dtmf_pt);
                dtmf_rate = rtpmap->clock_rate;
                ast_rtp_codecs_set_preferred_dtmf_format(codecs, dtmf_code, dtmf_rate);
            }
            tel_event++;
        }
 
        ast_copy_pj_str(media, (pj_str_t*)&stream->desc.media, sizeof(media));
        ast_rtp_codecs_payloads_set_rtpmap_type_rate(codecs, NULL,
            pj_strtoul(&stream->desc.fmt[i]), media, name, options, rtpmap->clock_rate);
        /* Look for an optional associated fmtp attribute */
        if (!(attr = pjmedia_sdp_media_find_attr2(stream, "fmtp", &rtpmap->pt))) {
            continue;
        }
 
        if ((pjmedia_sdp_attr_get_fmtp(attr, &fmtp)) == PJ_SUCCESS) {
            ast_copy_pj_str(fmt_param, &fmtp.fmt, sizeof(fmt_param));
            if (sscanf(fmt_param, "%30d", &num) != 1) {
                continue;
            }
 
            if ((format = ast_rtp_codecs_get_payload_format(codecs, num))) {
                struct ast_format *format_parsed;
 
                ast_copy_pj_str(fmt_param, &fmtp.fmt_param, sizeof(fmt_param));
 
                format_parsed = ast_format_parse_sdp_fmtp(format, fmt_param);
                if (format_parsed) {
                    ast_rtp_codecs_payload_replace_format(codecs, num, format_parsed);
                    ao2_ref(format_parsed, -1);
                }
                ao2_ref(format, -1);
            }
        }
    }
 
    /* Parsing done, now fill the ast_format_cap struct in the correct order */
    for (i = 0; i < stream->desc.fmt_count; ++i) {
        if ((format = ast_rtp_codecs_get_payload_format(codecs, pj_strtoul(&stream->desc.fmt[i])))) {
            ast_format_cap_append(astformats, format, 0);
            ao2_ref(format, -1);
        }
    }
 
    if (session->dtmf == AST_SIP_DTMF_AUTO) {
        if  (tel_event) {
            ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
            ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 1);
        } else {
            ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
            ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 0);
        }
    }
 
    if (session->dtmf == AST_SIP_DTMF_AUTO_INFO) {
        if  (tel_event) {
            ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
            ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 1);
        } else {
            ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_NONE);
            ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 0);
        }
    }
 
 
    /* Get the packetization, if it exists */
    if ((attr = pjmedia_sdp_media_find_attr2(stream, "ptime", NULL))) {
        unsigned long framing = pj_strtoul(pj_strltrim(&attr->value));
        if (framing && session->endpoint->media.rtp.use_ptime) {
            ast_rtp_codecs_set_framing(codecs, framing);
            ast_format_cap_set_framing(astformats, framing);
        }
    }
 
    SCOPE_EXIT_RTN();
}

References ao2_ref, ast_copy_pj_str(), ast_format_cap_append, ast_format_cap_remove_by_type(), ast_format_cap_set_framing(), ast_format_parse_sdp_fmtp(), AST_MEDIA_TYPE_UNKNOWN, ast_rtp_codecs_get_payload_format(), ast_rtp_codecs_payload_replace_format(), ast_rtp_codecs_payloads_initialize(), ast_rtp_codecs_payloads_set_m_type(), ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_codecs_set_framing(), ast_rtp_codecs_set_preferred_dtmf_format(), AST_RTP_DTMF_MODE_INBAND, AST_RTP_DTMF_MODE_NONE, AST_RTP_DTMF_MODE_RFC2833, ast_rtp_instance_dtmf_mode_set(), ast_rtp_instance_set_prop(), AST_RTP_OPT_G726_NONSTANDARD, AST_RTP_PROPERTY_DTMF, AST_SIP_DTMF_AUTO, AST_SIP_DTMF_AUTO_INFO, ast_sip_session_get_name(), codecs, name, NULL, options, ast_sip_session_media::rtp, SCOPE_ENTER, SCOPE_EXIT_RTN, and session.

Referenced by set_caps(), and set_incoming_call_offer_cap().

◆ get_media_encryption_type()

static enum ast_sip_session_media_encryption get_media_encryption_type	(	pj_str_t	transport,
		const struct pjmedia_sdp_media *	stream,
		unsigned int *	optimistic
	)

static

figure out media transport encryption type from the media transport string

Definition at line 1171 of file res_pjsip_sdp_rtp.c.

{
    RAII_VAR(char *, transport_str, ast_strndup(transport.ptr, transport.slen), ast_free);
 
    *optimistic = 0;
 
    if (!transport_str) {
        return AST_SIP_MEDIA_TRANSPORT_INVALID;
    }
    if (strstr(transport_str, "UDP/TLS")) {
        return AST_SIP_MEDIA_ENCRYPT_DTLS;
    } else if (strstr(transport_str, "SAVP")) {
        return AST_SIP_MEDIA_ENCRYPT_SDES;
    } else if (media_stream_has_crypto(stream)) {
        *optimistic = 1;
        return AST_SIP_MEDIA_ENCRYPT_SDES;
    } else {
        return AST_SIP_MEDIA_ENCRYPT_NONE;
    }
}

References ast_free, AST_SIP_MEDIA_ENCRYPT_DTLS, AST_SIP_MEDIA_ENCRYPT_NONE, AST_SIP_MEDIA_ENCRYPT_SDES, AST_SIP_MEDIA_TRANSPORT_INVALID, ast_strndup, media_stream_has_crypto(), RAII_VAR, and ast_rtp_engine_ice_candidate::transport.

Referenced by check_endpoint_media_transport().

◆ load_module()

static int load_module ( void )

static

Load the module.

Module loading including tests for configuration or dependencies. This function can return AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_DECLINE, or AST_MODULE_LOAD_SUCCESS. If a dependency or environment variable fails tests return AST_MODULE_LOAD_FAILURE. If the module can not load the configuration file or other non-critical problem return AST_MODULE_LOAD_DECLINE. On success return AST_MODULE_LOAD_SUCCESS.

Definition at line 2692 of file res_pjsip_sdp_rtp.c.

{
    if (ast_check_ipv6()) {
        ast_sockaddr_parse(&address_rtp, "::", 0);
    } else {
        ast_sockaddr_parse(&address_rtp, "0.0.0.0", 0);
    }
 
    if (!(sched = ast_sched_context_create())) {
        ast_log(LOG_ERROR, "Unable to create scheduler context.\n");
        goto end;
    }
 
    if (ast_sched_start_thread(sched)) {
        ast_log(LOG_ERROR, "Unable to create scheduler context thread.\n");
        goto end;
    }
 
    if (ast_sip_session_register_sdp_handler(&audio_sdp_handler, STR_AUDIO)) {
        ast_log(LOG_ERROR, "Unable to register SDP handler for %s stream type\n", STR_AUDIO);
        goto end;
    }
 
    if (ast_sip_session_register_sdp_handler(&video_sdp_handler, STR_VIDEO)) {
        ast_log(LOG_ERROR, "Unable to register SDP handler for %s stream type\n", STR_VIDEO);
        goto end;
    }
 
    ast_sip_session_register_supplement(&video_info_supplement);
 
    return AST_MODULE_LOAD_SUCCESS;
end:
    unload_module();
 
    return AST_MODULE_LOAD_DECLINE;
}

References address_rtp, ast_check_ipv6(), ast_log, AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_SUCCESS, ast_sched_context_create(), ast_sched_start_thread(), ast_sip_session_register_sdp_handler(), ast_sip_session_register_supplement, ast_sockaddr_parse(), audio_sdp_handler, end, LOG_ERROR, STR_AUDIO, STR_VIDEO, unload_module(), video_info_supplement, and video_sdp_handler.

◆ media_session_rtcp_read_callback()

static struct ast_frame * media_session_rtcp_read_callback	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media
	)

static

Definition at line 2375 of file res_pjsip_sdp_rtp.c.

{
    struct ast_frame *f;
 
    if (!session_media->rtp) {
        return &ast_null_frame;
    }
 
    f = ast_rtp_instance_read(session_media->rtp, 1);
    if (!f) {
        return NULL;
    }
 
    ast_rtp_instance_set_last_rx(session_media->rtp, time(NULL));
 
    return f;
}

References ast_null_frame, ast_rtp_instance_read(), ast_rtp_instance_set_last_rx(), NULL, and ast_sip_session_media::rtp.

Referenced by apply_negotiated_sdp_stream().

◆ media_session_rtp_read_callback()

static struct ast_frame * media_session_rtp_read_callback	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media
	)

static

Definition at line 2357 of file res_pjsip_sdp_rtp.c.

{
    struct ast_frame *f;
 
    if (!session_media->rtp) {
        return &ast_null_frame;
    }
 
    f = ast_rtp_instance_read(session_media->rtp, 0);
    if (!f) {
        return NULL;
    }
 
    ast_rtp_instance_set_last_rx(session_media->rtp, time(NULL));
 
    return f;
}

References ast_null_frame, ast_rtp_instance_read(), ast_rtp_instance_set_last_rx(), NULL, and ast_sip_session_media::rtp.

Referenced by apply_negotiated_sdp_stream().

◆ media_session_rtp_write_callback()

static int media_session_rtp_write_callback	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		struct ast_frame *	frame
	)

static

Definition at line 2393 of file res_pjsip_sdp_rtp.c.

{
    if (!session_media->rtp) {
        return 0;
    }
 
    return ast_rtp_instance_write(session_media->rtp, frame);
}

References ast_rtp_instance_write(), and ast_sip_session_media::rtp.

Referenced by apply_negotiated_sdp_stream().

◆ media_stream_has_crypto()

static int media_stream_has_crypto ( const struct pjmedia_sdp_media * stream )

static

figure out if media stream has crypto lines for sdes

Definition at line 1151 of file res_pjsip_sdp_rtp.c.

{
    int i;
 
    for (i = 0; i < stream->attr_count; i++) {
        pjmedia_sdp_attr *attr;
 
        /* check the stream for the required crypto attribute */
        attr = stream->attr[i];
        if (pj_strcmp2(&attr->name, "crypto")) {
            continue;
        }
 
        return 1;
    }
 
    return 0;
}

Referenced by get_media_encryption_type().

◆ negotiate_incoming_sdp_stream()

static int negotiate_incoming_sdp_stream	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const pjmedia_sdp_session *	sdp,
		int	index,
		struct ast_stream *	asterisk_stream
	)

static

Function which negotiates an incoming media stream.

Definition at line 1718 of file res_pjsip_sdp_rtp.c.

{
    char host[NI_MAXHOST];
    RAII_VAR(struct ast_sockaddr *, addrs, NULL, ast_free);
    pjmedia_sdp_media *stream = sdp->media[index];
    struct ast_sip_session_media *session_media_transport;
    enum ast_media_type media_type = session_media->type;
    enum ast_sip_session_media_encryption encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
    struct ast_format_cap *joint;
    int res;
    SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
 
    /* If no type formats have been configured reject this stream */
    if (!ast_format_cap_has_type(session->endpoint->media.codecs, media_type)) {
        ast_debug(3, "Endpoint has no codecs for media type '%s', declining stream\n",
            ast_codec_media_type2str(session_media->type));
        SCOPE_EXIT_RTN_VALUE(0, "Endpoint has no codecs\n");
    }
 
    /* Ensure incoming transport is compatible with the endpoint's configuration */
    if (!session->endpoint->media.rtp.use_received_transport) {
        encryption = check_endpoint_media_transport(session->endpoint, stream);
 
        if (encryption == AST_SIP_MEDIA_TRANSPORT_INVALID) {
            SCOPE_EXIT_RTN_VALUE(-1, "Incompatible transport\n");
        }
    }
 
    ast_copy_pj_str(host, stream->conn ? &stream->conn->addr : &sdp->conn->addr, sizeof(host));
 
    /* Ensure that the address provided is valid */
    if (ast_sockaddr_resolve(&addrs, host, PARSE_PORT_FORBID, AST_AF_UNSPEC) <= 0) {
        /* The provided host was actually invalid so we error out this negotiation */
        SCOPE_EXIT_RTN_VALUE(-1, "Invalid host\n");
    }
 
    /* Using the connection information create an appropriate RTP instance */
    if (!session_media->rtp && create_rtp(session, session_media, sdp)) {
        SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create rtp\n");
    }
 
    process_ssrc_attributes(session, session_media, stream);
    process_extmap_attributes(session, session_media, stream);
    session_media_transport = ast_sip_session_media_get_transport(session, session_media);
 
    if (session_media_transport == session_media || !session_media->bundled) {
        /* If this media session is carrying actual traffic then set up those aspects */
        session_media->remote_rtcp_mux = (pjmedia_sdp_media_find_attr2(stream, "rtcp-mux", NULL) != NULL);
        set_ice_components(session, session_media);
 
        enable_rtcp(session, session_media, stream);
 
        res = setup_media_encryption(session, session_media, sdp, stream);
        if (res) {
            if (!session->endpoint->media.rtp.encryption_optimistic ||
                !pj_strncmp2(&stream->desc.transport, "RTP/SAVP", 8)) {
                /* If optimistic encryption is disabled and crypto should have been enabled
                 * but was not this session must fail. This must also fail if crypto was
                 * required in the offer but could not be set up.
                 */
                SCOPE_EXIT_RTN_VALUE(-1, "Incompatible crypto\n");
            }
            /* There is no encryption, sad. */
            session_media->encryption = AST_SIP_MEDIA_ENCRYPT_NONE;
        }
 
        /* If we've been explicitly configured to use the received transport OR if
         * encryption is on and crypto is present use the received transport.
         * This is done in case of optimistic because it may come in as RTP/AVP or RTP/SAVP depending
         * on the configuration of the remote endpoint (optimistic themselves or mandatory).
         */
        if ((session->endpoint->media.rtp.use_received_transport) ||
            ((encryption == AST_SIP_MEDIA_ENCRYPT_SDES) && !res)) {
            pj_strdup(session->inv_session->pool, &session_media->transport, &stream->desc.transport);
        }
    } else {
        /* This is bundled with another session, so mark it as such */
        ast_rtp_instance_bundle(session_media->rtp, session_media_transport->rtp);
 
        enable_rtcp(session, session_media, stream);
    }
 
    /* If ICE support is enabled find all the needed attributes */
    check_ice_support(session, session_media, stream);
 
    /* If ICE support is enabled then check remote ICE started? */
    if (session_media->remote_ice) {
        process_ice_auth_attrb(session, session_media, sdp, stream);
    }
 
    /* Check if incoming SDP is changing the remotely held state */
    set_session_media_remotely_held(session_media, session, stream, asterisk_stream, addrs);
 
    joint = set_incoming_call_offer_cap(session, session_media, stream);
    res = apply_cap_to_bundled(session_media, session_media_transport, asterisk_stream, joint);
    ao2_cleanup(joint);
    if (res != 0) {
        SCOPE_EXIT_RTN_VALUE(0, "Something failed\n");
    }
 
    SCOPE_EXIT_RTN_VALUE(1);
}

References ao2_cleanup, apply_cap_to_bundled(), AST_AF_UNSPEC, ast_codec_media_type2str(), ast_copy_pj_str(), ast_debug, ast_format_cap_has_type(), ast_free, ast_rtp_instance_bundle(), AST_SIP_MEDIA_ENCRYPT_NONE, AST_SIP_MEDIA_ENCRYPT_SDES, AST_SIP_MEDIA_TRANSPORT_INVALID, ast_sip_session_get_name(), ast_sip_session_media_get_transport(), ast_sockaddr_resolve(), ast_sip_session_media::bundled, check_endpoint_media_transport(), check_ice_support(), create_rtp(), enable_rtcp(), ast_sip_session_media::encryption, NULL, PARSE_PORT_FORBID, process_extmap_attributes(), process_ice_auth_attrb(), process_ssrc_attributes(), RAII_VAR, ast_sip_session_media::remote_ice, ast_sip_session_media::remote_rtcp_mux, ast_sip_session_media::rtp, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, session, set_ice_components(), set_incoming_call_offer_cap(), set_session_media_remotely_held(), setup_media_encryption(), ast_sip_session_media::transport, and ast_sip_session_media::type.

◆ parse_dtls_attrib()

static int parse_dtls_attrib	(	struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_session *	sdp,
		const struct pjmedia_sdp_media *	stream
	)

static

Definition at line 1332 of file res_pjsip_sdp_rtp.c.

{
    int i;
 
    for (i = 0; i < sdp->attr_count; i++) {
        apply_dtls_attrib(session_media, sdp->attr[i]);
    }
 
    for (i = 0; i < stream->attr_count; i++) {
        apply_dtls_attrib(session_media, stream->attr[i]);
    }
 
    ast_set_flag(session_media->srtp, AST_SRTP_CRYPTO_OFFER_OK);
 
    return 0;
}

References apply_dtls_attrib(), ast_set_flag, AST_SRTP_CRYPTO_OFFER_OK, and ast_sip_session_media::srtp.

Referenced by setup_media_encryption().

◆ payload_type_compatible()

static int payload_type_compatible	(	struct ast_rtp_codecs *	codecs,
		int	payload,
		struct ast_format *	format
	)

static

Check whether a payload slot is either free or already mapped to the requested format.

Definition at line 483 of file res_pjsip_sdp_rtp.c.

{
    struct ast_rtp_payload_type *type;
    int compatible;
 
    type = ast_rtp_codecs_get_payload(codecs, payload);
    if (!type) {
        return 1;
    }
 
    compatible = type->asterisk_format
        && ast_format_cmp(type->format, format) == AST_FORMAT_CMP_EQUAL;
    ao2_ref(type, -1);
 
    return compatible;
}

References ao2_ref, ast_format_cmp(), AST_FORMAT_CMP_EQUAL, ast_rtp_codecs_get_payload(), codecs, ast_rtp_payload_type::format, ast_rtp_payload_type::payload, and type.

Referenced by assign_unique_bundle_payload().

◆ process_extmap_attributes()

static void process_extmap_attributes	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	remote_stream
	)

static

Function which processes extmap attributes in a stream.

Definition at line 1637 of file res_pjsip_sdp_rtp.c.

{
    int index;
 
    if (!session->endpoint->media.webrtc || session_media->type != AST_MEDIA_TYPE_VIDEO) {
        return;
    }
 
    ast_rtp_instance_extmap_clear(session_media->rtp);
 
    for (index = 0; index < remote_stream->attr_count; ++index) {
        pjmedia_sdp_attr *attr = remote_stream->attr[index];
        char attr_value[pj_strlen(&attr->value) + 1];
        char *uri;
        int id;
        char direction_str[10] = "";
        char *attributes;
        enum ast_rtp_extension_direction direction = AST_RTP_EXTENSION_DIRECTION_SENDRECV;
 
        /* We only care about extmap attributes */
        if (pj_strcmp2(&attr->name, "extmap")) {
            continue;
        }
 
        ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
 
        /* Split the combined unique identifier and direction away from the URI and attributes for easier parsing */
        uri = strchr(attr_value, ' ');
        if (ast_strlen_zero(uri)) {
            continue;
        }
        *uri++ = '\0';
 
        if ((sscanf(attr_value, "%30d%9s", &id, direction_str) < 1) || (id < 1)) {
            /* We require at a minimum the unique identifier */
            continue;
        }
 
        /* Convert from the string to the internal representation */
        if (!strcasecmp(direction_str, "/sendonly")) {
            direction = AST_RTP_EXTENSION_DIRECTION_SENDONLY;
        } else if (!strcasecmp(direction_str, "/recvonly")) {
            direction = AST_RTP_EXTENSION_DIRECTION_RECVONLY;
        } else if (!strcasecmp(direction_str, "/inactive")) {
            direction = AST_RTP_EXTENSION_DIRECTION_INACTIVE;
        }
 
        attributes = strchr(uri, ' ');
        if (!ast_strlen_zero(attributes)) {
            *attributes++ = '\0';
        }
 
        ast_rtp_instance_extmap_negotiate(session_media->rtp, id, direction, uri, attributes);
    }
}

References ast_copy_pj_str(), AST_MEDIA_TYPE_VIDEO, AST_RTP_EXTENSION_DIRECTION_INACTIVE, AST_RTP_EXTENSION_DIRECTION_RECVONLY, AST_RTP_EXTENSION_DIRECTION_SENDONLY, AST_RTP_EXTENSION_DIRECTION_SENDRECV, ast_rtp_instance_extmap_clear(), ast_rtp_instance_extmap_negotiate(), ast_strlen_zero(), id, ast_sip_session_media::rtp, session, and ast_sip_session_media::type.

Referenced by apply_negotiated_sdp_stream(), and negotiate_incoming_sdp_stream().

◆ process_ice_attributes()

static void process_ice_attributes	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_session *	remote,
		const struct pjmedia_sdp_media *	remote_stream
	)

static

Function which processes ICE attributes in an audio stream.

Definition at line 1047 of file res_pjsip_sdp_rtp.c.

{
    struct ast_rtp_engine_ice *ice;
    const pjmedia_sdp_attr *attr;
    char attr_value[256];
    unsigned int attr_i;
 
    /* If ICE support is not enabled or available exit early */
    if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp))) {
        return;
    }
 
    ast_debug_ice(2, "(%p) ICE process attributes\n", session_media->rtp);
 
    attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-ufrag", NULL);
    if (!attr) {
        attr = pjmedia_sdp_attr_find2(remote->attr_count, remote->attr, "ice-ufrag", NULL);
    }
    if (attr) {
        ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
        ice->set_authentication(session_media->rtp, attr_value, NULL);
    } else {
        ast_debug_ice(2, "(%p) ICE no, or invalid ice-ufrag\n", session_media->rtp);
        return;
    }
 
    attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-pwd", NULL);
    if (!attr) {
        attr = pjmedia_sdp_attr_find2(remote->attr_count, remote->attr, "ice-pwd", NULL);
    }
    if (attr) {
        ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
        ice->set_authentication(session_media->rtp, NULL, attr_value);
    } else {
        ast_debug_ice(2, "(%p) ICE no, or invalid ice-pwd\n", session_media->rtp);
        return;
    }
 
    if (pjmedia_sdp_media_find_attr2(remote_stream, "ice-lite", NULL)) {
        ice->ice_lite(session_media->rtp);
    }
 
    /* Find all of the candidates */
    for (attr_i = 0; attr_i < remote_stream->attr_count; ++attr_i) {
        char foundation[33], transport[32], address[PJ_INET6_ADDRSTRLEN + 1], cand_type[6], relay_address[PJ_INET6_ADDRSTRLEN + 1] = "";
        unsigned int port, relay_port = 0;
        struct ast_rtp_engine_ice_candidate candidate = { 0, };
 
        attr = remote_stream->attr[attr_i];
 
        /* If this is not a candidate line skip it */
        if (pj_strcmp2(&attr->name, "candidate")) {
            continue;
        }
 
        ast_copy_pj_str(attr_value, (pj_str_t*)&attr->value, sizeof(attr_value));
 
        if (sscanf(attr_value, "%32s %30u %31s %30u %46s %30u typ %5s %*s %23s %*s %30u", foundation, &candidate.id, transport,
            (unsigned *)&candidate.priority, address, &port, cand_type, relay_address, &relay_port) < 7) {
            /* Candidate did not parse properly */
            continue;
        }
 
        if (session->endpoint->media.rtcp_mux && session_media->remote_rtcp_mux && candidate.id > 1) {
            /* Remote side may have offered RTP and RTCP candidates. However, if we're using RTCP MUX,
             * then we should ignore RTCP candidates.
             */
            continue;
        }
 
        candidate.foundation = foundation;
        candidate.transport = transport;
 
        ast_sockaddr_parse(&candidate.address, address, PARSE_PORT_FORBID);
        ast_sockaddr_set_port(&candidate.address, port);
 
        if (!strcasecmp(cand_type, "host")) {
            candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_HOST;
        } else if (!strcasecmp(cand_type, "srflx")) {
            candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_SRFLX;
        } else if (!strcasecmp(cand_type, "relay")) {
            candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_RELAYED;
        } else {
            continue;
        }
 
        if (!ast_strlen_zero(relay_address)) {
            ast_sockaddr_parse(&candidate.relay_address, relay_address, PARSE_PORT_FORBID);
        }
 
        if (relay_port) {
            ast_sockaddr_set_port(&candidate.relay_address, relay_port);
        }
 
        ice->add_remote_candidate(session_media->rtp, &candidate);
    }
 
    ice->set_role(session_media->rtp, pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_TRUE ?
        AST_RTP_ICE_ROLE_CONTROLLING : AST_RTP_ICE_ROLE_CONTROLLED);
    ice->start(session_media->rtp);
}

References ast_rtp_engine_ice::add_remote_candidate, ast_rtp_engine_ice_candidate::address, ast_copy_pj_str(), ast_debug_ice, AST_RTP_ICE_CANDIDATE_TYPE_HOST, AST_RTP_ICE_CANDIDATE_TYPE_RELAYED, AST_RTP_ICE_CANDIDATE_TYPE_SRFLX, AST_RTP_ICE_ROLE_CONTROLLED, AST_RTP_ICE_ROLE_CONTROLLING, ast_rtp_instance_get_ice(), ast_sockaddr_parse(), ast_sockaddr_set_port, ast_strlen_zero(), ast_rtp_engine_ice_candidate::foundation, ast_rtp_engine_ice::ice_lite, ast_rtp_engine_ice_candidate::id, NULL, PARSE_PORT_FORBID, ast_rtp_engine_ice_candidate::priority, ast_rtp_engine_ice_candidate::relay_address, ast_sip_session_media::remote_rtcp_mux, ast_sip_session_media::rtp, session, ast_rtp_engine_ice::set_authentication, ast_rtp_engine_ice::set_role, ast_rtp_engine_ice::start, ast_rtp_engine_ice_candidate::transport, and ast_rtp_engine_ice_candidate::type.

Referenced by apply_negotiated_sdp_stream().

◆ process_ice_auth_attrb()

static void process_ice_auth_attrb	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_session *	remote,
		const struct pjmedia_sdp_media *	remote_stream
	)

static

Definition at line 1009 of file res_pjsip_sdp_rtp.c.

{
    struct ast_rtp_engine_ice *ice;
    const pjmedia_sdp_attr *ufrag_attr, *passwd_attr;
    char ufrag_attr_value[256];
    char passwd_attr_value[256];
 
    /* If ICE support is not enabled or available exit early */
    if (!session->endpoint->media.rtp.ice_support || !(ice = ast_rtp_instance_get_ice(session_media->rtp))) {
        return;
    }
 
    ufrag_attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-ufrag", NULL);
    if (!ufrag_attr) {
        ufrag_attr = pjmedia_sdp_attr_find2(remote->attr_count, remote->attr, "ice-ufrag", NULL);
    }
    if (ufrag_attr) {
        ast_copy_pj_str(ufrag_attr_value, (pj_str_t*)&ufrag_attr->value, sizeof(ufrag_attr_value));
    } else {
        return;
    }
        passwd_attr = pjmedia_sdp_media_find_attr2(remote_stream, "ice-pwd", NULL);
    if (!passwd_attr) {
        passwd_attr = pjmedia_sdp_attr_find2(remote->attr_count, remote->attr, "ice-pwd", NULL);
    }
    if (passwd_attr) {
        ast_copy_pj_str(passwd_attr_value, (pj_str_t*)&passwd_attr->value, sizeof(passwd_attr_value));
    } else {
        return;
    }
 
    if (ufrag_attr && passwd_attr) {
        ice->set_authentication(session_media->rtp, ufrag_attr_value, passwd_attr_value);
    }
}

References ast_copy_pj_str(), ast_rtp_instance_get_ice(), NULL, session, ast_rtp_engine_ice::set_authentication, and ast_rtp_engine_ice::start.

Referenced by negotiate_incoming_sdp_stream().

◆ process_ssrc_attributes()

static void process_ssrc_attributes	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	remote_stream
	)

static

Function which processes ssrc attributes in a stream.

Definition at line 1451 of file res_pjsip_sdp_rtp.c.

{
    int index;
 
    if (!session->endpoint->media.bundle) {
        return;
    }
 
    for (index = 0; index < remote_stream->attr_count; ++index) {
        pjmedia_sdp_attr *attr = remote_stream->attr[index];
        char attr_value[pj_strlen(&attr->value) + 1];
        char *ssrc_attribute_name, *ssrc_attribute_value = NULL;
        unsigned int ssrc;
 
        /* We only care about ssrc attributes */
        if (pj_strcmp2(&attr->name, "ssrc")) {
            continue;
        }
 
        ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
 
        if ((ssrc_attribute_name = strchr(attr_value, ' '))) {
            /* This has an actual attribute */
            *ssrc_attribute_name++ = '\0';
            ssrc_attribute_value = strchr(ssrc_attribute_name, ':');
            if (ssrc_attribute_value) {
                /* Values are actually optional according to the spec */
                *ssrc_attribute_value++ = '\0';
            }
        }
 
        if (sscanf(attr_value, "%30u", &ssrc) < 1) {
            continue;
        }
 
        /* If we are currently negotiating as a result of the remote side renegotiating then
         * determine if the source for this stream has changed.
         */
        if (pjmedia_sdp_neg_get_state(session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_REMOTE_OFFER &&
            session->active_media_state) {
            struct ast_rtp_instance_stats stats = { 0, };
 
            if (!ast_rtp_instance_get_stats(session_media->rtp, &stats, AST_RTP_INSTANCE_STAT_REMOTE_SSRC) &&
                stats.remote_ssrc != ssrc) {
                session_media->changed = 1;
            }
        }
 
        ast_rtp_instance_set_remote_ssrc(session_media->rtp, ssrc);
        break;
    }
}

References ast_copy_pj_str(), ast_rtp_instance_get_stats(), ast_rtp_instance_set_remote_ssrc(), AST_RTP_INSTANCE_STAT_REMOTE_SSRC, ast_sip_session_media::changed, NULL, ast_rtp_instance_stats::remote_ssrc, ast_sip_session_media::rtp, session, and ast_rtp_engine_ice::start.

Referenced by apply_negotiated_sdp_stream(), and negotiate_incoming_sdp_stream().

◆ rtp_check_timeout()

static int rtp_check_timeout ( const void * data )

static

Check whether RTP is being received or not.

Definition at line 105 of file res_pjsip_sdp_rtp.c.

{
    struct ast_sip_session_media *session_media = (struct ast_sip_session_media *)data;
    struct ast_rtp_instance *rtp = session_media->rtp;
    struct ast_channel *chan;
    int elapsed;
    int now;
    int timeout;
 
    if (!rtp) {
        return 0;
    }
 
    chan = ast_channel_get_by_name(ast_rtp_instance_get_channel_id(rtp));
    if (!chan) {
        return 0;
    }
 
    /* Store these values locally to avoid multiple function calls */
    now = time(NULL);
    timeout = ast_rtp_instance_get_timeout(rtp);
 
    /* If the channel is not in UP state or call is redirected
     * outside Asterisk return for later check.
     */
    if (ast_channel_state(chan) != AST_STATE_UP || !ast_sockaddr_isnull(&session_media->direct_media_addr)) {
        /* Avoiding immediately disconnect after channel up or direct media has been stopped */
        ast_rtp_instance_set_last_rx(rtp, now);
        ast_channel_unref(chan);
        /* Recheck after half timeout for avoiding possible races
        * and faster reacting to cases while there is no an RTP at all.
        */
        return timeout * 500;
    }
 
    elapsed = now - ast_rtp_instance_get_last_rx(rtp);
    if (elapsed < timeout) {
        ast_channel_unref(chan);
        return (timeout - elapsed) * 1000;
    }
 
    ast_log(LOG_NOTICE, "Disconnecting channel '%s' for lack of %s RTP activity in %d seconds\n",
        ast_channel_name(chan), ast_codec_media_type2str(session_media->type), elapsed);
 
    ast_channel_lock(chan);
    ast_channel_hangupcause_set(chan, AST_CAUSE_REQUESTED_CHAN_UNAVAIL);
    ast_channel_unlock(chan);
 
    ast_softhangup(chan, AST_SOFTHANGUP_DEV);
    ast_channel_unref(chan);
 
    return 0;
}

References AST_CAUSE_REQUESTED_CHAN_UNAVAIL, ast_channel_get_by_name(), ast_channel_hangupcause_set(), ast_channel_lock, ast_channel_name(), ast_channel_unlock, ast_channel_unref, ast_codec_media_type2str(), ast_log, ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_last_rx(), ast_rtp_instance_get_timeout(), ast_rtp_instance_set_last_rx(), ast_sockaddr_isnull(), ast_softhangup(), AST_SOFTHANGUP_DEV, AST_STATE_UP, ast_sip_session_media::direct_media_addr, LOG_NOTICE, NULL, ast_sip_session_media::rtp, and ast_sip_session_media::type.

Referenced by apply_negotiated_sdp_stream().

◆ send_keepalive()

static int send_keepalive ( const void * data )

static

Definition at line 71 of file res_pjsip_sdp_rtp.c.

{
    struct ast_sip_session_media *session_media = (struct ast_sip_session_media *) data;
    struct ast_rtp_instance *rtp = session_media->rtp;
    int keepalive;
    time_t interval;
    int send_keepalive;
 
    if (!rtp) {
        return 0;
    }
 
    keepalive = ast_rtp_instance_get_keepalive(rtp);
 
    if (!ast_sockaddr_isnull(&session_media->direct_media_addr)) {
        ast_debug_rtp(3, "(%p) RTP not sending keepalive since direct media is in use\n", rtp);
        return keepalive * 1000;
    }
 
    interval = time(NULL) - ast_rtp_instance_get_last_tx(rtp);
    send_keepalive = interval >= keepalive;
 
    ast_debug_rtp(3, "(%p) RTP it has been %d seconds since RTP was last sent. %sending keepalive\n",
        rtp, (int) interval, send_keepalive ? "S" : "Not s");
 
    if (send_keepalive) {
        ast_rtp_instance_sendcng(rtp, 0);
        return keepalive * 1000;
    }
 
    return (keepalive - interval) * 1000;
}

References ast_debug_rtp, ast_rtp_instance_get_keepalive(), ast_rtp_instance_get_last_tx(), ast_rtp_instance_sendcng(), ast_sockaddr_isnull(), ast_sip_session_media::direct_media_addr, ast_rtp_instance::keepalive, NULL, ast_sip_session_media::rtp, and send_keepalive().

Referenced by apply_negotiated_sdp_stream(), and send_keepalive().

◆ set_caps()

static int set_caps	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		struct ast_sip_session_media *	session_media_transport,
		const struct pjmedia_sdp_media *	stream,
		int	is_offer,
		struct ast_stream *	asterisk_stream
	)

static

Definition at line 673 of file res_pjsip_sdp_rtp.c.

{
    RAII_VAR(struct ast_format_cap *, caps, NULL, ao2_cleanup);
    RAII_VAR(struct ast_format_cap *, peer, NULL, ao2_cleanup);
    RAII_VAR(struct ast_format_cap *, joint, NULL, ao2_cleanup);
    enum ast_media_type media_type = session_media->type;
    struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;
    int direct_media_enabled = !ast_sockaddr_isnull(&session_media->direct_media_addr) &&
        ast_format_cap_count(session->direct_media_cap);
    int dsp_features = 0;
    SCOPE_ENTER(1, "%s %s\n", ast_sip_session_get_name(session), is_offer ? "OFFER" : "ANSWER");
 
    if (!(caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
        !(peer = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT)) ||
        !(joint = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
        ast_log(LOG_ERROR, "Failed to allocate %s capabilities\n",
            ast_codec_media_type2str(session_media->type));
        SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create %s capabilities\n",
            ast_codec_media_type2str(session_media->type));
    }
 
    /* get the endpoint capabilities */
    if (direct_media_enabled) {
        ast_format_cap_get_compatible(session->endpoint->media.codecs, session->direct_media_cap, caps);
    } else {
        ast_format_cap_append_from_cap(caps, session->endpoint->media.codecs, media_type);
    }
 
    /* get the capabilities on the peer */
    get_codecs(session, stream, &codecs, session_media, peer);
 
    /* get the joint capabilities between peer and endpoint */
    ast_format_cap_get_compatible(caps, peer, joint);
    if (!ast_format_cap_count(joint)) {
        struct ast_str *usbuf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
        struct ast_str *thembuf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
 
        ast_rtp_codecs_payloads_destroy(&codecs);
        ast_log(LOG_NOTICE, "No joint capabilities for '%s' media stream between our configuration(%s) and incoming SDP(%s)\n",
            ast_codec_media_type2str(session_media->type),
            ast_format_cap_get_names(caps, &usbuf),
            ast_format_cap_get_names(peer, &thembuf));
        SCOPE_EXIT_RTN_VALUE(-1, "No joint capabilities for '%s' media stream between our configuration(%s) and incoming SDP(%s)\n",
            ast_codec_media_type2str(session_media->type),
            ast_format_cap_get_names(caps, &usbuf),
            ast_format_cap_get_names(peer, &thembuf));
    } else {
        struct ast_format *preferred_fmt = ast_format_cap_get_format(joint, 0);
 
        ast_rtp_codecs_set_preferred_format(&codecs, preferred_fmt);
        ao2_ref(preferred_fmt, -1);
    }
 
    if (is_offer) {
        /*
         * Setup rx payload type mapping to prefer the mapping
         * from the peer that the RFC says we SHOULD use.
         */
        ast_rtp_codecs_payloads_xover(&codecs, &codecs, NULL);
    }
 
    if (session->channel) {
        ast_channel_lock(session->channel);
    }
 
    ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
        session_media->rtp);
 
    apply_cap_to_bundled(session_media, session_media_transport, asterisk_stream, joint);
 
    if (session->channel && ast_sip_session_is_pending_stream_default(session, asterisk_stream)) {
        ast_format_cap_remove_by_type(caps, AST_MEDIA_TYPE_UNKNOWN);
        ast_format_cap_append_from_cap(caps, ast_channel_nativeformats(session->channel),
            AST_MEDIA_TYPE_UNKNOWN);
        ast_format_cap_remove_by_type(caps, media_type);
 
        if (session->endpoint->preferred_codec_only) {
            struct ast_format *preferred_fmt = ast_format_cap_get_format(joint, 0);
            ast_format_cap_append(caps, preferred_fmt, 0);
            ao2_ref(preferred_fmt, -1);
        } else if (!session->endpoint->asymmetric_rtp_codec) {
            struct ast_format *best;
            /*
             * If we don't allow the sending codec to be changed on our side
             * then get the best codec from the joint capabilities of the media
             * type and use only that. This ensures the core won't start sending
             * out a format that we aren't currently sending.
             */
 
            best = ast_format_cap_get_best_by_type(joint, media_type);
            if (best) {
                ast_format_cap_append(caps, best, ast_format_cap_get_framing(joint));
                ao2_ref(best, -1);
            }
        } else {
            ast_format_cap_append_from_cap(caps, joint, media_type);
        }
 
        /*
         * Apply the new formats to the channel, potentially changing
         * raw read/write formats and translation path while doing so.
         */
        ast_channel_nativeformats_set(session->channel, caps);
        if (media_type == AST_MEDIA_TYPE_AUDIO) {
            ast_set_read_format(session->channel, ast_channel_readformat(session->channel));
            ast_set_write_format(session->channel, ast_channel_writeformat(session->channel));
        }
 
        if ( ((session->dtmf == AST_SIP_DTMF_AUTO) || (session->dtmf == AST_SIP_DTMF_AUTO_INFO) )
            && (ast_rtp_instance_dtmf_mode_get(session_media->rtp) == AST_RTP_DTMF_MODE_RFC2833)
            && (session->dsp)) {
            dsp_features = ast_dsp_get_features(session->dsp);
            dsp_features &= ~DSP_FEATURE_DIGIT_DETECT;
            if (dsp_features) {
                ast_dsp_set_features(session->dsp, dsp_features);
            } else {
                ast_dsp_free(session->dsp);
                session->dsp = NULL;
            }
        }
 
        if (ast_channel_is_bridged(session->channel)) {
            ast_channel_set_unbridged_nolock(session->channel, 1);
        }
    }
 
    if (session->channel) {
        ast_channel_unlock(session->channel);
    }
 
    ast_rtp_codecs_payloads_destroy(&codecs);
    SCOPE_EXIT_RTN_VALUE(0);
}

References ao2_cleanup, ao2_ref, apply_cap_to_bundled(), ast_channel_is_bridged(), ast_channel_lock, ast_channel_nativeformats(), ast_channel_nativeformats_set(), ast_channel_readformat(), ast_channel_set_unbridged_nolock(), ast_channel_unlock, ast_channel_writeformat(), ast_codec_media_type2str(), ast_dsp_free(), ast_dsp_get_features(), ast_dsp_set_features(), ast_format_cap_alloc, ast_format_cap_append, ast_format_cap_append_from_cap(), ast_format_cap_count(), AST_FORMAT_CAP_FLAG_DEFAULT, ast_format_cap_get_best_by_type(), ast_format_cap_get_compatible(), ast_format_cap_get_format(), ast_format_cap_get_framing(), ast_format_cap_get_names(), AST_FORMAT_CAP_NAMES_LEN, ast_format_cap_remove_by_type(), ast_log, AST_MEDIA_TYPE_AUDIO, AST_MEDIA_TYPE_UNKNOWN, AST_RTP_CODECS_NULL_INIT, ast_rtp_codecs_payloads_copy(), ast_rtp_codecs_payloads_destroy(), ast_rtp_codecs_payloads_xover(), ast_rtp_codecs_set_preferred_format(), AST_RTP_DTMF_MODE_RFC2833, ast_rtp_instance_dtmf_mode_get(), ast_rtp_instance_get_codecs(), ast_set_read_format(), ast_set_write_format(), AST_SIP_DTMF_AUTO, AST_SIP_DTMF_AUTO_INFO, ast_sip_session_get_name(), ast_sip_session_is_pending_stream_default(), ast_sockaddr_isnull(), ast_str_alloca, codecs, ast_sip_session_media::direct_media_addr, get_codecs(), LOG_ERROR, LOG_NOTICE, NULL, RAII_VAR, ast_sip_session_media::rtp, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, session, and ast_sip_session_media::type.

Referenced by apply_negotiated_sdp_stream().

◆ set_ice_components()

static void set_ice_components	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media
	)

static

Definition at line 1414 of file res_pjsip_sdp_rtp.c.

{
    struct ast_rtp_engine_ice *ice;
 
    ast_assert(session_media->rtp != NULL);
 
    ice = ast_rtp_instance_get_ice(session_media->rtp);
    if (!session->endpoint->media.rtp.ice_support || !ice) {
        return;
    }
 
    if (session->endpoint->media.rtcp_mux && session_media->remote_rtcp_mux) {
        /* We both support RTCP mux. Only one ICE component necessary */
        ice->change_components(session_media->rtp, 1);
    } else {
        /* They either don't support RTCP mux or we don't know if they do yet. */
        ice->change_components(session_media->rtp, 2);
    }
}

References ast_assert, ast_rtp_instance_get_ice(), ast_rtp_engine_ice::change_components, NULL, session, and ast_rtp_engine_ice::start.

Referenced by apply_negotiated_sdp_stream(), create_outgoing_sdp_stream(), and negotiate_incoming_sdp_stream().

◆ set_incoming_call_offer_cap()

static struct ast_format_cap * set_incoming_call_offer_cap	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	stream
	)

static

Definition at line 619 of file res_pjsip_sdp_rtp.c.

{
    struct ast_format_cap *incoming_call_offer_cap;
    struct ast_format_cap *remote;
    struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;
    SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
 
 
    remote = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
    if (!remote) {
        ast_log(LOG_ERROR, "Failed to allocate %s incoming remote capabilities\n",
                ast_codec_media_type2str(session_media->type));
        SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't allocate caps\n");
    }
 
    /* Get the peer's capabilities*/
    get_codecs(session, stream, &codecs, session_media, remote);
 
    incoming_call_offer_cap = ast_sip_session_create_joint_call_cap(
        session, session_media->type, remote);
 
    ao2_ref(remote, -1);
 
    if (!incoming_call_offer_cap || ast_format_cap_empty(incoming_call_offer_cap)) {
        ao2_cleanup(incoming_call_offer_cap);
        ast_rtp_codecs_payloads_destroy(&codecs);
        SCOPE_EXIT_RTN_VALUE(NULL, "No incoming call offer caps\n");
    }
 
    /*
     * Setup rx payload type mapping to prefer the mapping
     * from the peer that the RFC says we SHOULD use.
     */
    ast_rtp_codecs_payloads_xover(&codecs, &codecs, NULL);
 
    /*
     * This only merges the newly offered payloads into the live RTP instance
     * rather than doing a destructive replace, so any format the channel is
     * still actively sending in (e.g. while a reINVITE is in flight) remains
     * usable right up until set_caps() commits the channel to the new
     * negotiated format. A destructive replace here, before the channel's
     * own read/write format has caught up, is what let a concurrent bridge
     * write briefly land on a payload mapping that no longer had its format.
     */
    ast_rtp_codecs_payloads_merge(&codecs,
        ast_rtp_instance_get_codecs(session_media->rtp), session_media->rtp);
 
    ast_rtp_codecs_payloads_destroy(&codecs);
 
    SCOPE_EXIT_RTN_VALUE(incoming_call_offer_cap);
}

References ao2_cleanup, ao2_ref, ast_codec_media_type2str(), ast_format_cap_alloc, ast_format_cap_empty(), AST_FORMAT_CAP_FLAG_DEFAULT, ast_log, AST_RTP_CODECS_NULL_INIT, ast_rtp_codecs_payloads_destroy(), ast_rtp_codecs_payloads_merge(), ast_rtp_codecs_payloads_xover(), ast_rtp_instance_get_codecs(), ast_sip_session_create_joint_call_cap(), ast_sip_session_get_name(), codecs, get_codecs(), LOG_ERROR, NULL, ast_sip_session_media::rtp, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, session, and ast_sip_session_media::type.

Referenced by negotiate_incoming_sdp_stream().

◆ set_session_media_remotely_held()

static void set_session_media_remotely_held	(	struct ast_sip_session_media *	session_media,
		const struct ast_sip_session *	session,
		const pjmedia_sdp_media *	media,
		const struct ast_stream *	stream,
		const struct ast_sockaddr *	addrs
	)

static

Definition at line 1694 of file res_pjsip_sdp_rtp.c.

{
    if (ast_sip_session_is_pending_stream_default(session, stream) &&
        (session_media->type == AST_MEDIA_TYPE_AUDIO)) {
        if (((addrs != NULL) && ast_sockaddr_isnull(addrs)) ||
            ((addrs != NULL) && ast_sockaddr_is_any(addrs)) ||
            pjmedia_sdp_media_find_attr2(media, "sendonly", NULL) ||
            pjmedia_sdp_media_find_attr2(media, "inactive", NULL)) {
            if (!session_media->remotely_held) {
                session_media->remotely_held = 1;
                session_media->remotely_held_changed = 1;
            }
        } else if (session_media->remotely_held) {
            session_media->remotely_held = 0;
            session_media->remotely_held_changed = 1;
        }
    }
}

References AST_MEDIA_TYPE_AUDIO, ast_sip_session_is_pending_stream_default(), ast_sockaddr_is_any(), ast_sockaddr_isnull(), NULL, ast_sip_session_media::remotely_held, ast_sip_session_media::remotely_held_changed, session, and ast_sip_session_media::type.

Referenced by apply_negotiated_sdp_stream(), and negotiate_incoming_sdp_stream().

◆ setup_dtls_srtp()

static int setup_dtls_srtp	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media
	)

static

Definition at line 1256 of file res_pjsip_sdp_rtp.c.

{
    struct ast_rtp_engine_dtls *dtls;
 
    if (!session->endpoint->media.rtp.dtls_cfg.enabled || !session_media->rtp) {
        return -1;
    }
 
    dtls = ast_rtp_instance_get_dtls(session_media->rtp);
    if (!dtls) {
        return -1;
    }
 
    session->endpoint->media.rtp.dtls_cfg.suite = ((session->endpoint->media.rtp.srtp_tag_32) ? AST_AES_CM_128_HMAC_SHA1_32 : AST_AES_CM_128_HMAC_SHA1_80);
    if (dtls->set_configuration(session_media->rtp, &session->endpoint->media.rtp.dtls_cfg)) {
        ast_log(LOG_ERROR, "Attempted to set an invalid DTLS-SRTP configuration on RTP instance '%p'\n",
            session_media->rtp);
        return -1;
    }
 
    if (setup_srtp(session_media)) {
        return -1;
    }
    return 0;
}

References AST_AES_CM_128_HMAC_SHA1_32, AST_AES_CM_128_HMAC_SHA1_80, ast_log, ast_rtp_instance_get_dtls(), LOG_ERROR, ast_sip_session_media::rtp, session, ast_rtp_engine_dtls::set_configuration, and setup_srtp().

Referenced by add_crypto_to_stream(), and setup_media_encryption().

◆ setup_media_encryption()

static int setup_media_encryption	(	struct ast_sip_session *	session,
		struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_session *	sdp,
		const struct pjmedia_sdp_media *	stream
	)

static

Definition at line 1387 of file res_pjsip_sdp_rtp.c.

{
    switch (session_media->encryption) {
    case AST_SIP_MEDIA_ENCRYPT_SDES:
        if (setup_sdes_srtp(session_media, stream)) {
            return -1;
        }
        break;
    case AST_SIP_MEDIA_ENCRYPT_DTLS:
        if (setup_dtls_srtp(session, session_media)) {
            return -1;
        }
        if (parse_dtls_attrib(session_media, sdp, stream)) {
            return -1;
        }
        break;
    case AST_SIP_MEDIA_TRANSPORT_INVALID:
    case AST_SIP_MEDIA_ENCRYPT_NONE:
        break;
    }
 
    return 0;
}

References AST_SIP_MEDIA_ENCRYPT_DTLS, AST_SIP_MEDIA_ENCRYPT_NONE, AST_SIP_MEDIA_ENCRYPT_SDES, AST_SIP_MEDIA_TRANSPORT_INVALID, ast_sip_session_media::encryption, parse_dtls_attrib(), session, setup_dtls_srtp(), and setup_sdes_srtp().

Referenced by apply_negotiated_sdp_stream(), and negotiate_incoming_sdp_stream().

◆ setup_sdes_srtp()

static int setup_sdes_srtp	(	struct ast_sip_session_media *	session_media,
		const struct pjmedia_sdp_media *	stream
	)

static

Definition at line 1351 of file res_pjsip_sdp_rtp.c.

{
    int i;
 
    for (i = 0; i < stream->attr_count; i++) {
        pjmedia_sdp_attr *attr;
        RAII_VAR(char *, crypto_str, NULL, ast_free);
 
        /* check the stream for the required crypto attribute */
        attr = stream->attr[i];
        if (pj_strcmp2(&attr->name, "crypto")) {
            continue;
        }
 
        crypto_str = ast_strndup(attr->value.ptr, attr->value.slen);
        if (!crypto_str) {
            return -1;
        }
 
        if (setup_srtp(session_media)) {
            return -1;
        }
 
        if (!ast_sdp_crypto_process(session_media->rtp, session_media->srtp, crypto_str)) {
            /* found a valid crypto attribute */
            return 0;
        }
 
        ast_debug(1, "Ignoring crypto offer with unsupported parameters: %s\n", crypto_str);
    }
 
    /* no usable crypto attributes found */
    return -1;
}

References ast_debug, ast_free, ast_sdp_crypto_process(), ast_strndup, NULL, RAII_VAR, ast_sip_session_media::rtp, setup_srtp(), and ast_sip_session_media::srtp.

Referenced by setup_media_encryption().

◆ setup_srtp()

static int setup_srtp ( struct ast_sip_session_media * session_media )

static

Definition at line 1237 of file res_pjsip_sdp_rtp.c.

{
    if (!session_media->srtp) {
        session_media->srtp = ast_sdp_srtp_alloc();
        if (!session_media->srtp) {
            return -1;
        }
    }
 
    if (!session_media->srtp->crypto) {
        session_media->srtp->crypto = ast_sdp_crypto_alloc();
        if (!session_media->srtp->crypto) {
            return -1;
        }
    }
 
    return 0;
}

References ast_sdp_crypto_alloc(), ast_sdp_srtp_alloc(), ast_sdp_srtp::crypto, and ast_sip_session_media::srtp.

Referenced by setup_dtls_srtp(), and setup_sdes_srtp().

◆ stream_destroy()

static void stream_destroy ( struct ast_sip_session_media * session_media )

static

Function which destroys the RTP instance when session ends.

Definition at line 2610 of file res_pjsip_sdp_rtp.c.

{
    if (session_media->rtp) {
        stream_stop(session_media);
        ast_rtp_instance_destroy(session_media->rtp);
    }
    session_media->rtp = NULL;
}

References ast_rtp_instance_destroy(), NULL, ast_sip_session_media::rtp, and stream_stop().

◆ stream_stop()

static void stream_stop ( struct ast_sip_session_media * session_media )

static

Function which stops the RTP instance.

Definition at line 2598 of file res_pjsip_sdp_rtp.c.

{
    if (!session_media->rtp) {
        return;
    }
 
    AST_SCHED_DEL(sched, session_media->keepalive_sched_id);
    AST_SCHED_DEL(sched, session_media->timeout_sched_id);
    ast_rtp_instance_stop(session_media->rtp);
}

References ast_rtp_instance_stop(), AST_SCHED_DEL, ast_sip_session_media::keepalive_sched_id, ast_sip_session_media::rtp, and ast_sip_session_media::timeout_sched_id.

Referenced by stream_destroy().

◆ unload_module()

static int unload_module ( void )

static

Unloads the sdp RTP/AVP module from Asterisk.

Definition at line 2669 of file res_pjsip_sdp_rtp.c.

{
    ast_sip_session_unregister_supplement(&video_info_supplement);
    ast_sip_session_unregister_sdp_handler(&video_sdp_handler, STR_VIDEO);
    ast_sip_session_unregister_sdp_handler(&audio_sdp_handler, STR_AUDIO);
 
    if (sched) {
        ast_sched_context_destroy(sched);
    }
 
    return 0;
}

References ast_sched_context_destroy(), ast_sip_session_unregister_sdp_handler(), ast_sip_session_unregister_supplement(), audio_sdp_handler, STR_AUDIO, STR_VIDEO, video_info_supplement, and video_sdp_handler.

Referenced by load_module().

◆ video_info_incoming_request()

static int video_info_incoming_request	(	struct ast_sip_session *	session,
		struct pjsip_rx_data *	rdata
	)

static

Definition at line 2641 of file res_pjsip_sdp_rtp.c.

{
    struct pjsip_transaction *tsx;
    pjsip_tx_data *tdata;
 
    if (!session->channel
        || !ast_sip_are_media_types_equal(&rdata->msg_info.msg->body->content_type,
            &pjsip_media_type_application_media_control_xml)) {
        return 0;
    }
 
    tsx = pjsip_rdata_get_tsx(rdata);
 
    ast_queue_control(session->channel, AST_CONTROL_VIDUPDATE);
 
    if (pjsip_dlg_create_response(session->inv_session->dlg, rdata, 200, NULL, &tdata) == PJ_SUCCESS) {
        pjsip_dlg_send_response(session->inv_session->dlg, tsx, tdata);
    }
 
    return 0;
}

References AST_CONTROL_VIDUPDATE, ast_queue_control(), ast_sip_are_media_types_equal(), NULL, pjsip_media_type_application_media_control_xml, and session.

Variable Documentation

◆ __mod_info

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "PJSIP SDP RTP/AVP stream handler" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_pjsip,res_pjsip_session", }

static

Definition at line 2735 of file res_pjsip_sdp_rtp.c.

◆ address_rtp

struct ast_sockaddr address_rtp

static

Address for RTP.

Definition at line 66 of file res_pjsip_sdp_rtp.c.

Referenced by create_rtp(), and load_module().

◆ ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info

static

Definition at line 2735 of file res_pjsip_sdp_rtp.c.

◆ audio_sdp_handler

struct ast_sip_session_sdp_handler audio_sdp_handler

static

SDP handler for 'audio' media stream.

Definition at line 2620 of file res_pjsip_sdp_rtp.c.

                                                              {
    .id = STR_AUDIO,
    .negotiate_incoming_sdp_stream = negotiate_incoming_sdp_stream,
    .create_outgoing_sdp_stream = create_outgoing_sdp_stream,
    .apply_negotiated_sdp_stream = apply_negotiated_sdp_stream,
    .change_outgoing_sdp_stream_media_address = change_outgoing_sdp_stream_media_address,
    .stream_stop = stream_stop,
    .stream_destroy = stream_destroy,
};

Referenced by load_module(), and unload_module().

◆ sched

struct ast_sched_context* sched

static

Scheduler for RTCP purposes.

Definition at line 63 of file res_pjsip_sdp_rtp.c.

◆ STR_AUDIO

const char STR_AUDIO[] = "audio"

static

Definition at line 68 of file res_pjsip_sdp_rtp.c.

Referenced by load_module(), and unload_module().

◆ STR_VIDEO

const char STR_VIDEO[] = "video"

static

Definition at line 69 of file res_pjsip_sdp_rtp.c.

Referenced by load_module(), and unload_module().

◆ video_info_supplement

struct ast_sip_session_supplement video_info_supplement

static

Initial value:

= {
    .method = "INFO",
    .incoming_request = video_info_incoming_request,
}

Definition at line 2663 of file res_pjsip_sdp_rtp.c.

                                                                 {
    .method = "INFO",
    .incoming_request = video_info_incoming_request,
};

Referenced by load_module(), and unload_module().

◆ video_sdp_handler

struct ast_sip_session_sdp_handler video_sdp_handler

static

SDP handler for 'video' media stream.

Definition at line 2631 of file res_pjsip_sdp_rtp.c.

                                                              {
    .id = STR_VIDEO,
    .negotiate_incoming_sdp_stream = negotiate_incoming_sdp_stream,
    .create_outgoing_sdp_stream = create_outgoing_sdp_stream,
    .apply_negotiated_sdp_stream = apply_negotiated_sdp_stream,
    .change_outgoing_sdp_stream_media_address = change_outgoing_sdp_stream_media_address,
    .stream_stop = stream_stop,
    .stream_destroy = stream_destroy,
};

Referenced by load_module(), and unload_module().

Functions

Variables

Detailed Description

Function Documentation

◆ __reg_module()

◆ __unreg_module()

◆ add_crypto_to_stream()

◆ add_extmap_to_stream()

◆ add_ice_to_stream()

◆ add_msid_to_stream()

◆ add_rtcp_fb_to_stream()

◆ add_ssrc_to_stream()

◆ apply_cap_to_bundled()

◆ apply_dtls_attrib()

◆ apply_negotiated_sdp_stream()

◆ assign_unique_bundle_payload()

◆ AST_MODULE_SELF_SYM()

◆ bundle_payload_type_used()

◆ change_outgoing_sdp_stream_media_address()

◆ check_endpoint_media_transport()

◆ check_ice_support()

◆ create_outgoing_sdp_stream()

◆ create_rtp()

◆ enable_rtcp()

◆ enable_rtp_extension()

◆ ensure_payload_type_mapping()

◆ generate_fmtp_attr()

◆ generate_rtpmap_attr()

◆ generate_rtpmap_attr2()

◆ generating_local_offer()

◆ get_codecs()

◆ get_media_encryption_type()

◆ load_module()

◆ media_session_rtcp_read_callback()

◆ media_session_rtp_read_callback()

◆ media_session_rtp_write_callback()

◆ media_stream_has_crypto()

◆ negotiate_incoming_sdp_stream()

◆ parse_dtls_attrib()

◆ payload_type_compatible()

◆ process_extmap_attributes()

◆ process_ice_attributes()

◆ process_ice_auth_attrb()

◆ process_ssrc_attributes()

◆ rtp_check_timeout()

◆ send_keepalive()

◆ set_caps()

◆ set_ice_components()

◆ set_incoming_call_offer_cap()

◆ set_session_media_remotely_held()

◆ setup_dtls_srtp()

◆ setup_media_encryption()

◆ setup_sdes_srtp()

◆ setup_srtp()

◆ stream_destroy()

◆ stream_stop()

◆ unload_module()

◆ video_info_incoming_request()

Variable Documentation

◆ __mod_info

◆ address_rtp

◆ ast_module_info

◆ audio_sdp_handler

◆ sched

◆ STR_AUDIO

◆ STR_VIDEO

◆ video_info_supplement

◆ video_sdp_handler