test_res_rtp.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 2019, Sangoma, Inc.
 *
 * Ben Ford <bford@sangoma.com>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */
 
/*!
 * \file
 * \brief RTP/RTCP Unit Tests
 *
 * \author Ben Ford <bford@digium.com>
 *
 */
 
/*** MODULEINFO
    <depend>TEST_FRAMEWORK</depend>
    <support_level>core</support_level>
 ***/
 
#include "asterisk.h"
 
#include "asterisk/module.h"
#include "asterisk/test.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/data_buffer.h"
#include "asterisk/format_cache.h"
#include <assert.h>
#include <sched.h>
 
enum test_type {
    TEST_TYPE_NONE = 0, /* No special setup required */
    TEST_TYPE_NACK,     /* Enable NACK */
    TEST_TYPE_REMB,     /* Enable REMB */
    TEST_TYPE_STD_RTCP, /* Let the stack do RTCP */
};
 
static void ast_sched_context_destroy_wrapper(struct ast_sched_context *sched)
{
    if (sched) {
        ast_sched_context_destroy(sched);
    }
}
 
static int test_init_rtp_instances(struct ast_rtp_instance **instance1,
    struct ast_rtp_instance **instance2, struct ast_sched_context *test_sched,
    enum test_type type)
{
    struct ast_sockaddr addr1;
    struct ast_sockaddr addr2;
    enum ast_rtp_instance_rtcp rtcp_type = AST_RTP_INSTANCE_RTCP_MUX;
 
    ast_sockaddr_parse(&addr1, "127.0.0.1", 0);
    ast_sockaddr_parse(&addr2, "127.0.0.1", 0);
 
    *instance1 = ast_rtp_instance_new("asterisk", test_sched, &addr1, "instance1");
    *instance2 = ast_rtp_instance_new("asterisk", test_sched, &addr2, "instance2");
    if (!instance1 || !instance2) {
        return -1;
    }
 
    ast_rtp_instance_set_channel_id(*instance1, "instance1");
    ast_rtp_instance_set_channel_id(*instance2, "instance2");
 
    if (type == TEST_TYPE_STD_RTCP) {
        rtcp_type = AST_RTP_INSTANCE_RTCP_STANDARD;
    }
 
    ast_rtp_instance_set_prop(*instance1,
        AST_RTP_PROPERTY_RTCP, rtcp_type);
    ast_rtp_instance_set_prop(*instance2,
        AST_RTP_PROPERTY_RTCP, rtcp_type);
 
    if (type == TEST_TYPE_NACK) {
        ast_rtp_instance_set_prop(*instance1, AST_RTP_PROPERTY_RETRANS_RECV, 1);
        ast_rtp_instance_set_prop(*instance1, AST_RTP_PROPERTY_RETRANS_SEND, 1);
        ast_rtp_instance_set_prop(*instance2, AST_RTP_PROPERTY_RETRANS_RECV, 2);
        ast_rtp_instance_set_prop(*instance2, AST_RTP_PROPERTY_RETRANS_SEND, 2);
    } else if (type == TEST_TYPE_REMB) {
        ast_rtp_instance_set_prop(*instance1, AST_RTP_PROPERTY_REMB, 1);
        ast_rtp_instance_set_prop(*instance2, AST_RTP_PROPERTY_REMB, 1);
    }
 
    ast_rtp_instance_get_local_address(*instance1, &addr1);
    ast_rtp_instance_set_remote_address(*instance2, &addr1);
 
    ast_rtp_instance_get_local_address(*instance2, &addr2);
    ast_rtp_instance_set_remote_address(*instance1, &addr2);
 
    ast_rtp_instance_reset_test_engine(*instance1);
 
    ast_rtp_instance_activate(*instance1);
    ast_rtp_instance_activate(*instance2);
 
    return 0;
}
 
static void test_write_frames(struct ast_rtp_instance *instance, int seqno, int num)
{
    char data[320] = "";
    struct ast_frame frame_out = {
        .frametype = AST_FRAME_VOICE,
        .subclass.format = ast_format_ulaw,
        .data.ptr = data,
        .datalen = 160,
    };
    int index;
 
    ast_set_flag(&frame_out, AST_FRFLAG_HAS_SEQUENCE_NUMBER);
 
    for (index = 0; index < num; index++) {
        frame_out.seqno = seqno + index;
        ast_rtp_instance_write(instance, &frame_out);
    }
}
 
static void test_read_frames(struct ast_rtp_instance *instance, int num)
{
    struct ast_frame *frame_in;
    int index;
 
    for (index = 0; index < num; index++) {
        frame_in = ast_rtp_instance_read(instance, 0);
        if (frame_in) {
            ast_frfree(frame_in);
        }
    }
}
 
static void test_write_and_read_frames(struct ast_rtp_instance *instance1,
    struct ast_rtp_instance *instance2, int seqno, int num)
{
    test_write_frames(instance1, seqno, num);
    test_read_frames(instance2, num);
}
 
 
/*
 * Unfortunately, we can't use usleep() to create
 * packet spacing because there are signals in use
 * which cause usleep to immediately return.  Instead
 * we have to spin.  :(
 */
static void SLEEP_SPINNER(int ms)
{
    struct timeval a = ast_tvnow();
 
    while(1) {
        sched_yield();
        if (ast_remaining_ms(a, ms) <= 0) {
            break;
        }
    }
}
 
/*
 * This function is NOT really a reliable implementation.
 * Its purpose is only to aid in code development in res_rtp_asterisk.
 */
static void test_write_and_read_interleaved_frames(struct ast_rtp_instance *instance1,
    struct ast_rtp_instance *instance2, int howlong, int rtcp_interval)
{
    char data[320] = "";
    int pktinterval = 20;
 
    struct ast_frame frame_out1 = {
        .frametype = AST_FRAME_VOICE,
        .subclass.format = ast_format_ulaw,
        .seqno = 4556,
        .data.ptr = data,
        .datalen = 160,
        .samples = 1,
        .len = pktinterval,
        .ts = 4622295,
    };
    struct ast_frame frame_out2 = {
        .frametype = AST_FRAME_VOICE,
        .subclass.format = ast_format_ulaw,
        .seqno = 6554,
        .data.ptr = data,
        .datalen = 160,
        .samples = 1,
        .len = pktinterval,
        .ts = 8622295,
    };
    struct ast_frame *frame_in1;
    struct ast_frame *frame_in2;
    int index;
    int num;
    int rtcpnum;
    int reverse = 1;
    int send_rtcp = 0;
 
    num = howlong / pktinterval;
 
    rtcpnum = rtcp_interval / pktinterval;
 
    ast_set_flag(&frame_out1, AST_FRFLAG_HAS_SEQUENCE_NUMBER);
    ast_set_flag(&frame_out1, AST_FRFLAG_HAS_TIMING_INFO);
    ast_set_flag(&frame_out2, AST_FRFLAG_HAS_SEQUENCE_NUMBER);
    ast_set_flag(&frame_out2, AST_FRFLAG_HAS_TIMING_INFO);
 
    for (index = 0; index < num; index++) {
        struct timeval start = ast_tvnow();
        time_t ms;
 
        if (index == 1) {
            ast_clear_flag(&frame_out1, AST_FRFLAG_HAS_SEQUENCE_NUMBER);
            ast_clear_flag(&frame_out1, AST_FRFLAG_HAS_TIMING_INFO);
            ast_clear_flag(&frame_out2, AST_FRFLAG_HAS_SEQUENCE_NUMBER);
            ast_clear_flag(&frame_out2, AST_FRFLAG_HAS_TIMING_INFO);
        }
        frame_out1.seqno += index;
        frame_out1.delivery = start;
        frame_out1.ts += frame_out1.len;
        ast_rtp_instance_write(instance1, &frame_out1);
 
        if (send_rtcp && index && (index % rtcpnum == 0)) {
            ast_rtp_instance_queue_report(instance1);
        }
 
        frame_in2 = ast_rtp_instance_read(instance2, 0);
        ast_frfree(frame_in2);
        frame_in2 = ast_rtp_instance_read(instance2, 1);
        ast_frfree(frame_in2);
 
        if (reverse) {
            frame_out2.seqno += index;
            frame_out2.delivery = ast_tvnow();
            frame_out2.ts += frame_out2.len;
            ast_rtp_instance_write(instance2, &frame_out2);
 
            if (send_rtcp && index && (index % rtcpnum == 0)) {
                ast_rtp_instance_queue_report(instance2);
            }
 
            frame_in1 = ast_rtp_instance_read(instance1, 0);
            ast_frfree(frame_in1);
            frame_in1 = ast_rtp_instance_read(instance1, 1);
            ast_frfree(frame_in1);
 
        }
 
        ms = frame_out1.len - ast_tvdiff_ms(ast_tvnow(),start);
        ms += (index % 2 ? 5 : 12);
        ms += (index % 3 ? 2 : 30);
        SLEEP_SPINNER(ms);
    }
}
 
/* Verify failed RTP allocation without a scheduler is safe and releases its resources. */
AST_TEST_DEFINE(rtp_port_exhaustion)
{
    RAII_VAR(struct ast_rtp_instance *, occupied, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_rtp_instance *, instance, NULL, ast_rtp_instance_destroy);
    struct ast_sockaddr address;
    struct ast_rtp_instance_options options;
 
    switch (cmd) {
    case TEST_INIT:
        info->name = "rtp_port_exhaustion";
        info->category = "/res/res_rtp/";
        info->summary = "RTP port exhaustion without a scheduler";
        info->description =
            "Exhaust a per-instance RTP port range with no scheduler, as used by "
            "UnicastRTP. Verify allocation fails safely and succeeds after the "
            "occupied port is released.";
        return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
        break;
    }
 
    ast_sockaddr_parse(&address, "127.0.0.1", 0);
    occupied = ast_rtp_instance_new("asterisk", NULL, &address, NULL);
    if (!occupied) {
        ast_test_status_update(test, "Unable to allocate the initial RTP instance\n");
        return AST_TEST_FAIL;
    }
 
    ast_rtp_instance_get_local_address(occupied, &address);
    /* The occupied port is the only even port in this range. */
    options.port_start = ast_sockaddr_port(&address);
    options.port_end = options.port_start + 1;
    instance = ast_rtp_instance_new_with_options("asterisk", NULL, &address,
        NULL, &options);
    if (instance) {
        ast_test_status_update(test, "Allocation succeeded with an exhausted port range\n");
        return AST_TEST_FAIL;
    }
 
    ast_rtp_instance_destroy(occupied);
    occupied = NULL;
    instance = ast_rtp_instance_new_with_options("asterisk", NULL, &address,
        NULL, &options);
    if (!instance) {
        ast_test_status_update(test, "Unable to reuse the released RTP port\n");
        return AST_TEST_FAIL;
    }
    ast_rtp_instance_get_local_address(instance, &address);
    if (ast_sockaddr_port(&address) != options.port_start) {
        ast_test_status_update(test, "RTP instance did not reuse the released port\n");
        return AST_TEST_FAIL;
    }
 
    return AST_TEST_PASS;
}
 
AST_TEST_DEFINE(nack_no_packet_loss)
{
    RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_rtp_instance *, instance2, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_sched_context *, test_sched, NULL, ast_sched_context_destroy_wrapper);
 
    switch (cmd) {
    case TEST_INIT:
        info->name = "nack_no_packet_loss";
        info->category = "/res/res_rtp/";
        info->summary = "nack no packet loss unit test";
        info->description =
            "Tests sending packets with no packet loss and "
            "validates that the send buffer stores sent packets "
            "and the receive buffer is empty";
        return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
        break;
    }
 
    test_sched = ast_sched_context_create();
 
    if ((test_init_rtp_instances(&instance1, &instance2, test_sched, TEST_TYPE_NACK)) < 0) {
        ast_log(LOG_ERROR, "Failed to initialize test!\n");
        return AST_TEST_FAIL;
    }
 
    test_write_and_read_frames(instance1, instance2, 1000, 10);
 
    ast_test_validate(test, ast_rtp_instance_get_send_buffer_count(instance1) == 10,
        "Send buffer did not have the expected count of 10");
 
    ast_test_validate(test, ast_rtp_instance_get_recv_buffer_count(instance2) == 0,
        "Receive buffer did not have the expected count of 0");
 
    return AST_TEST_PASS;
}
 
AST_TEST_DEFINE(nack_nominal)
{
    RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_rtp_instance *, instance2, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_sched_context *, test_sched, NULL, ast_sched_context_destroy_wrapper);
 
    switch (cmd) {
    case TEST_INIT:
        info->name = "nack_nominal";
        info->category = "/res/res_rtp/";
        info->summary = "nack nominal unit test";
        info->description =
            "Tests sending packets with some packet loss and "
            "validates that a NACK request is sent on reaching "
            "the triggering amount of lost packets";
        return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
        break;
    }
 
    test_sched = ast_sched_context_create();
 
    if ((test_init_rtp_instances(&instance1, &instance2, test_sched, TEST_TYPE_NACK)) < 0) {
        ast_log(LOG_ERROR, "Failed to initialize test!\n");
        return AST_TEST_FAIL;
    }
 
    /* Start normally */
    test_write_and_read_frames(instance1, instance2, 1000, 10);
 
    /* Set the number of packets to drop when we send them next */
    ast_rtp_instance_drop_packets(instance2, 10);
    test_write_and_read_frames(instance1, instance2, 1010, 10);
 
    /* Send enough packets to reach the NACK trigger */
    test_write_and_read_frames(instance1, instance2, 1020, ast_rtp_instance_get_recv_buffer_max(instance2) / 2);
 
    /* This needs to be read as RTCP */
    test_read_frames(instance1, 1);
 
    /* We should have the missing packets to read now */
    test_read_frames(instance2, 10);
 
    ast_test_validate(test, ast_rtp_instance_get_recv_buffer_count(instance2) == 0,
        "Receive buffer did not have the expected count of 0");
 
    return AST_TEST_PASS;
}
 
AST_TEST_DEFINE(nack_overflow)
{
    RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_rtp_instance *, instance2, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_sched_context *, test_sched, NULL, ast_sched_context_destroy_wrapper);
    int max_packets;
 
    switch (cmd) {
    case TEST_INIT:
        info->name = "nack_overflow";
        info->category = "/res/res_rtp/";
        info->summary = "nack overflow unit test";
        info->description =
            "Tests that when the buffer hits its capacity, we "
            "queue all the packets we currently have stored";
        return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
        break;
    }
 
    test_sched = ast_sched_context_create();
 
    if ((test_init_rtp_instances(&instance1, &instance2, test_sched, TEST_TYPE_NACK)) < 0) {
        ast_log(LOG_ERROR, "Failed to initialize test!\n");
        return AST_TEST_FAIL;
    }
 
    /* Start normally */
    test_write_and_read_frames(instance1, instance2, 1000, 10);
 
    /* Send enough packets to fill the buffer */
    max_packets = ast_rtp_instance_get_recv_buffer_max(instance2);
    test_write_and_read_frames(instance1, instance2, 1020, max_packets);
 
    ast_test_validate(test, ast_rtp_instance_get_recv_buffer_count(instance2) == max_packets,
        "Receive buffer did not have the expected count of max buffer size");
 
    /* Send the packet that will overflow the buffer */
    test_write_and_read_frames(instance1, instance2, 1020 + max_packets, 1);
 
    ast_test_validate(test, ast_rtp_instance_get_recv_buffer_count(instance2) == 0,
        "Receive buffer did not have the expected count of 0");
 
    return AST_TEST_PASS;
}
 
AST_TEST_DEFINE(lost_packet_stats_nominal)
{
    RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_rtp_instance *, instance2, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_sched_context *, test_sched, NULL, ast_sched_context_destroy_wrapper);
    struct ast_rtp_instance_stats stats = { 0, };
    enum ast_rtp_instance_stat stat = AST_RTP_INSTANCE_STAT_ALL;
 
    switch (cmd) {
    case TEST_INIT:
        info->name = "lost_packet_stats_nominal";
        info->category = "/res/res_rtp/";
        info->summary = "lost packet stats nominal unit test";
        info->description =
            "Tests that when some packets are lost, we calculate that "
            "loss correctly when doing lost packet statistics";
        return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
        break;
    }
 
    test_sched = ast_sched_context_create();
 
    if ((test_init_rtp_instances(&instance1, &instance2, test_sched, TEST_TYPE_NONE)) < 0) {
        ast_log(LOG_ERROR, "Failed to initialize test!\n");
        return AST_TEST_FAIL;
    }
 
    /* Start normally */
    test_write_and_read_frames(instance1, instance2, 1000, 10);
 
    /* Send some more packets, but with a gap */
    test_write_and_read_frames(instance1, instance2, 1015, 5);
 
    /* Send a RR to calculate lost packet statistics. We should be missing 5 packets */
    ast_rtp_instance_queue_report(instance1);
    test_write_frames(instance2, 1000, 1);
 
    /* Check RTCP stats to see if we got the expected packet loss count */
    ast_rtp_instance_get_stats(instance2, &stats, stat);
    ast_test_validate(test, stats.rxploss == 5 && stats.local_minrxploss == 5 &&
        stats.local_maxrxploss == 5, "Condition of 5 lost packets was not met");
 
    /* Drop 3 before writing 5 more */
    test_write_and_read_frames(instance1, instance2, 1023, 5);
 
    ast_rtp_instance_queue_report(instance1);
    test_write_frames(instance2, 1001, 1);
    ast_rtp_instance_get_stats(instance2, &stats, stat);
 
    /* Should now be missing 8 total packets with a change in min */
    ast_test_validate(test, stats.rxploss == 8 && stats.local_minrxploss == 3 &&
        stats.local_maxrxploss == 5);
 
    /* Write 5 more with no gaps */
    test_write_and_read_frames(instance1, instance2, 1028, 5);
 
    ast_rtp_instance_queue_report(instance1);
    test_write_frames(instance2, 1002, 1);
    ast_rtp_instance_get_stats(instance2, &stats, stat);
 
    /* Should still only be missing 8 total packets */
    ast_test_validate(test, stats.rxploss == 8 && stats.local_minrxploss == 3 &&
        stats.local_maxrxploss == 5);
 
    /* Now drop 1, write another 5, drop 8, and then write 5 */
    test_write_and_read_frames(instance1, instance2, 1034, 5);
    test_write_and_read_frames(instance1, instance2, 1047, 5);
 
    ast_rtp_instance_queue_report(instance1);
    test_write_frames(instance2, 1003, 1);
    ast_rtp_instance_get_stats(instance2, &stats, stat);
 
    /* Now it should be missing 17 total packets, with a change in max */
    ast_test_validate(test, stats.rxploss == 17 && stats.local_minrxploss == 3 &&
        stats.local_maxrxploss == 9);
 
    return AST_TEST_PASS;
}
 
AST_TEST_DEFINE(remb_nominal)
{
    RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_rtp_instance *, instance2, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_sched_context *, test_sched, NULL, ast_sched_context_destroy_wrapper);
    RAII_VAR(struct ast_frame *, frame_in, NULL, ast_frfree);
    /* Use the structure softmix_remb_collector uses to store information for REMB */
    struct ast_rtp_rtcp_feedback feedback = {
        .fmt = AST_RTP_RTCP_FMT_REMB,
        .remb.br_exp = 0,
        .remb.br_mantissa = 1000,
    };
    struct ast_frame frame_out = {
        .frametype = AST_FRAME_RTCP,
        .subclass.integer = AST_RTP_RTCP_PSFB,
        .data.ptr = &feedback,
        .datalen = sizeof(feedback),
    };
    struct ast_rtp_rtcp_feedback *received_feedback;
 
    switch (cmd) {
    case TEST_INIT:
        info->name = "remb_nominal";
        info->category = "/res/res_rtp/";
        info->summary = "remb nominal unit test";
        info->description =
            "Tests sending and receiving a REMB packet";
        return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
        break;
    }
 
    test_sched = ast_sched_context_create();
 
    if ((test_init_rtp_instances(&instance1, &instance2, test_sched, TEST_TYPE_REMB)) < 0) {
        ast_log(LOG_ERROR, "Failed to initialize test!\n");
        return AST_TEST_FAIL;
    }
 
    /* The schedid must be 0 or greater, so let's do that now */
    ast_rtp_instance_set_schedid(instance1, 0);
 
    ast_rtp_instance_write(instance1, &frame_out);
 
    /* Verify the high level aspects of the frame */
    frame_in = ast_rtp_instance_read(instance2, 0);
    ast_test_validate(test, frame_in != NULL, "Did not receive a REMB frame");
    ast_test_validate(test, frame_in->frametype == AST_FRAME_RTCP,
        "REMB frame did not have the expected frametype");
    ast_test_validate(test, frame_in->subclass.integer == AST_RTP_RTCP_PSFB,
        "REMB frame did not have the expected subclass integer");
 
    /* Verify the actual REMB information itself */
    received_feedback = frame_in->data.ptr;
    ast_test_validate(test, received_feedback->fmt == AST_RTP_RTCP_FMT_REMB,
        "REMB frame did not have the expected feedback format");
    ast_test_validate(test, received_feedback->remb.br_exp == feedback.remb.br_exp,
        "REMB received exponent did not match sent exponent");
    ast_test_validate(test, received_feedback->remb.br_mantissa == feedback.remb.br_mantissa,
        "REMB received mantissa did not match sent mantissa");
 
    return AST_TEST_PASS;
}
 
AST_TEST_DEFINE(sr_rr_nominal)
{
    RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_rtp_instance *, instance2, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_sched_context *, test_sched, NULL, ast_sched_context_destroy_wrapper);
    RAII_VAR(struct ast_frame *, frame_in, NULL, ast_frfree);
 
    switch (cmd) {
    case TEST_INIT:
        info->name = "sr_rr_nominal";
        info->category = "/res/res_rtp/";
        info->summary = "SR/RR nominal unit test";
        info->description =
            "Tests sending SR/RR and receiving it; includes SDES";
        return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
        break;
    }
 
    test_sched = ast_sched_context_create();
 
    if ((test_init_rtp_instances(&instance1, &instance2, test_sched, TEST_TYPE_NONE)) < 0) {
        ast_log(LOG_ERROR, "Failed to initialize test!\n");
        return AST_TEST_FAIL;
    }
 
    test_write_and_read_frames(instance1, instance2, 1000, 10);
 
    /*
     * Set the send_report flag so we send a sender report instead of normal RTP. We
     * also need to ensure that SDES processed.
     */
    ast_rtp_instance_queue_report(instance1);
    test_write_frames(instance1, 1010, 1);
 
    frame_in = ast_rtp_instance_read(instance2, 0);
    ast_test_validate(test, frame_in->frametype == AST_FRAME_RTCP,
        "Sender report frame did not have the expected frametype");
    ast_test_validate(test, frame_in->subclass.integer == AST_RTP_RTCP_SR,
        "Sender report frame did not have the expected subclass integer");
    ast_test_validate(test, ast_rtp_instance_get_sdes_received(instance2) == 1,
        "SDES was never processed for sender report");
 
    ast_frfree(frame_in);
 
    /* Set the send_report flag so we send a receiver report instead of normal RTP */
    ast_rtp_instance_queue_report(instance1);
    test_write_frames(instance1, 1010, 1);
 
    frame_in = ast_rtp_instance_read(instance2, 0);
    ast_test_validate(test, frame_in->frametype == AST_FRAME_RTCP,
        "Receiver report frame did not have the expected frametype");
    ast_test_validate(test, frame_in->subclass.integer == AST_RTP_RTCP_RR,
        "Receiver report frame did not have the expected subclass integer");
 
    return AST_TEST_PASS;
}
 
AST_TEST_DEFINE(fir_nominal)
{
    RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_rtp_instance *, instance2, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_sched_context *, test_sched, NULL, ast_sched_context_destroy_wrapper);
    RAII_VAR(struct ast_frame *, frame_in, NULL, ast_frfree);
    struct ast_frame frame_out = {
        .frametype = AST_FRAME_CONTROL,
        .subclass.integer = AST_CONTROL_VIDUPDATE,
    };
 
    switch (cmd) {
    case TEST_INIT:
        info->name = "fir_nominal";
        info->category = "/res/res_rtp/";
        info->summary = "fir nominal unit test";
        info->description =
            "Tests sending and receiving a FIR packet";
        return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
        break;
    }
 
    test_sched = ast_sched_context_create();
 
    if ((test_init_rtp_instances(&instance1, &instance2, test_sched, TEST_TYPE_NONE)) < 0) {
        ast_log(LOG_ERROR, "Failed to initialize test!\n");
        return AST_TEST_FAIL;
    }
 
    /* Send some packets to learn SSRC */
    test_write_and_read_frames(instance2, instance1, 1000, 10);
 
    /* The schedid must be 0 or greater, so let's do that now */
    ast_rtp_instance_set_schedid(instance1, 0);
 
    /*
     * This will not directly write a frame out, but cause Asterisk to see it as a FIR
     * request, which will then trigger rtp_write_rtcp_fir, which will send out the
     * appropriate packet.
     */
    ast_rtp_instance_write(instance1, &frame_out);
 
    /*
     * We only receive one frame, the FIR request. It won't have a subclass integer of
     * 206 (PSFB) because ast_rtcp_interpret sets it to 18 (AST_CONTROL_VIDUPDATE), so
     * check for that.
     */
    frame_in = ast_rtp_instance_read(instance2, 0);
    ast_test_validate(test, frame_in != NULL, "Did not receive a FIR frame");
    ast_test_validate(test, frame_in->frametype == AST_FRAME_CONTROL,
        "FIR frame did not have the expected frametype");
    ast_test_validate(test, frame_in->subclass.integer == AST_CONTROL_VIDUPDATE,
        "FIR frame did not have the expected subclass integer");
 
    return AST_TEST_PASS;
}
 
/*
 * This test should not normally be run.  Its only purpose is to
 * aid in code development.
 */
AST_TEST_DEFINE(mes)
{
    RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_rtp_instance *, instance2, NULL, ast_rtp_instance_destroy);
    RAII_VAR(struct ast_sched_context *, test_sched, NULL, ast_sched_context_destroy_wrapper);
 
    switch (cmd) {
    case TEST_INIT:
        info->name = "mes";
        info->category = "/res/res_rtp/";
        info->summary = "Media Experience Score";
        info->description =
            "Tests calculation of Media Experience Score (only run by explicit request)";
        info->explicit_only = 1;
        return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
        break;
    }
 
    test_sched = ast_sched_context_create();
    ast_sched_start_thread(test_sched);
 
    if ((test_init_rtp_instances(&instance1, &instance2,
        test_sched, TEST_TYPE_NONE)) < 0) {
        ast_log(LOG_ERROR, "Failed to initialize test!\n");
        return AST_TEST_FAIL;
    }
 
    test_write_and_read_interleaved_frames(
        instance1, instance2, 1000, 5000);
 
    return AST_TEST_PASS;
}
 
AST_TEST_DEFINE(payload_merge_abandoned_offer)
{
    struct ast_rtp_codecs offered = AST_RTP_CODECS_NULL_INIT;
    struct ast_rtp_codecs active = AST_RTP_CODECS_NULL_INIT;
    struct ast_rtp_payload_type *payload;
    enum ast_test_result_state result = AST_TEST_PASS;
    int offered_initialized = 0;
    int active_initialized = 0;
 
    switch (cmd) {
    case TEST_INIT:
        info->name = "payload_merge_abandoned_offer";
        info->category = "/res/res_rtp/";
        info->summary = "Verify mappings retained after an uncommitted payload merge";
        info->description =
            "Characterizes the state left behind when an SDP offer's payloads "
            "are merged into an active RTP mapping but no later copy commits "
            "or rolls back that offer.";
        return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
        break;
    }
 
    if (ast_rtp_codecs_payloads_initialize(&offered)) {
        ast_test_status_update(test, "Unable to initialize RTP codecs\n");
        result = AST_TEST_FAIL;
        goto cleanup;
    }
    offered_initialized = 1;
    if (ast_rtp_codecs_payloads_initialize(&active)) {
        ast_test_status_update(test, "Unable to initialize RTP codecs\n");
        result = AST_TEST_FAIL;
        goto cleanup;
    }
    active_initialized = 1;
 
    /* Use dynamic PTs so lookup cannot fall back to the global static table. */
    if (ast_rtp_codecs_payloads_set_rtpmap_type(
            &active, NULL, 96, "audio", "PCMU", 0)
        || ast_rtp_codecs_payloads_set_rtpmap_type(
            &offered, NULL, 97, "audio", "PCMA", 0)) {
        ast_test_status_update(test, "Unable to create dynamic RTP mappings\n");
        result = AST_TEST_FAIL;
        goto cleanup;
    }
    ast_rtp_codecs_payloads_xover(&active, &active, NULL);
    ast_rtp_codecs_payloads_xover(&offered, &offered, NULL);
 
    ast_rtp_codecs_payloads_merge(&offered, &active, NULL);
 
    payload = ast_rtp_codecs_get_payload(&active, 96);
    if (!payload || !payload->asterisk_format
        || ast_format_cmp(payload->format, ast_format_ulaw) != AST_FORMAT_CMP_EQUAL) {
        ast_test_status_update(test, "Active PCMU payload 96 was not retained by merge\n");
        result = AST_TEST_FAIL;
    }
    ao2_cleanup(payload);
 
    payload = ast_rtp_codecs_get_payload(&active, 97);
    if (!payload || !payload->asterisk_format
        || ast_format_cmp(payload->format, ast_format_alaw) != AST_FORMAT_CMP_EQUAL) {
        ast_test_status_update(test, "Offered PCMA payload 97 was not retained after merge\n");
        result = AST_TEST_FAIL;
    }
    ao2_cleanup(payload);
 
cleanup:
    if (active_initialized) {
        ast_rtp_codecs_payloads_destroy(&active);
    }
    if (offered_initialized) {
        ast_rtp_codecs_payloads_destroy(&offered);
    }
    return result;
}
 
AST_TEST_DEFINE(payload_merge_preserves_preferences)
{
    struct ast_rtp_codecs offered = AST_RTP_CODECS_NULL_INIT;
    struct ast_rtp_codecs active = AST_RTP_CODECS_NULL_INIT;
    RAII_VAR(struct ast_format *, preferred, NULL, ao2_cleanup);
    enum ast_test_result_state result = AST_TEST_PASS;
    int offered_initialized = 0;
    int active_initialized = 0;
 
    switch (cmd) {
    case TEST_INIT:
        info->name = "payload_merge_preserves_preferences";
        info->category = "/res/res_rtp/";
        info->summary = "Verify a payload merge preserves active preferences";
        info->description =
            "Verifies that merging an SDP offer's provisional payload mappings "
            "updates framing needed to generate the SDP answer without changing "
            "the active preferred format or preferred DTMF metadata before the "
            "offer is committed.";
        return AST_TEST_NOT_RUN;
    case TEST_EXECUTE:
        break;
    }
 
    if (ast_rtp_codecs_payloads_initialize(&offered)) {
        ast_test_status_update(test, "Unable to initialize RTP codecs\n");
        result = AST_TEST_FAIL;
        goto cleanup;
    }
    offered_initialized = 1;
    if (ast_rtp_codecs_payloads_initialize(&active)) {
        ast_test_status_update(test, "Unable to initialize RTP codecs\n");
        result = AST_TEST_FAIL;
        goto cleanup;
    }
    active_initialized = 1;
 
    ast_rtp_codecs_set_preferred_format(&active, ast_format_ulaw);
    ast_rtp_codecs_set_preferred_dtmf_format(&active, 101, 8000);
    ast_rtp_codecs_set_framing(&active, 20);
    ast_rtp_codecs_set_preferred_format(&offered, ast_format_alaw);
    ast_rtp_codecs_set_preferred_dtmf_format(&offered, 110, 16000);
    ast_rtp_codecs_set_framing(&offered, 30);
 
    ast_rtp_codecs_payloads_merge(&offered, &active, NULL);
 
    preferred = ast_rtp_codecs_get_preferred_format(&active);
    if (!preferred
        || ast_format_cmp(preferred, ast_format_ulaw) != AST_FORMAT_CMP_EQUAL) {
        ast_test_status_update(test, "Payload merge changed the active preferred format\n");
        result = AST_TEST_FAIL;
    }
    if (ast_rtp_codecs_get_preferred_dtmf_format_pt(&active) != 101
        || ast_rtp_codecs_get_preferred_dtmf_format_rate(&active) != 8000) {
        ast_test_status_update(test,
            "Payload merge changed the active preferred DTMF metadata\n");
        result = AST_TEST_FAIL;
    }
    if (ast_rtp_codecs_get_framing(&active) != 30) {
        ast_test_status_update(test, "Payload merge did not update offered framing\n");
        result = AST_TEST_FAIL;
    }
 
cleanup:
    if (active_initialized) {
        ast_rtp_codecs_payloads_destroy(&active);
    }
    if (offered_initialized) {
        ast_rtp_codecs_payloads_destroy(&offered);
    }
    return result;
}
 
/* Unregister RTP regression tests when the module unloads. */
static int unload_module(void)
{
    AST_TEST_UNREGISTER(rtp_port_exhaustion);
    AST_TEST_UNREGISTER(payload_merge_preserves_preferences);
    AST_TEST_UNREGISTER(payload_merge_abandoned_offer);
    AST_TEST_UNREGISTER(mes);
    AST_TEST_UNREGISTER(nack_no_packet_loss);
    AST_TEST_UNREGISTER(nack_nominal);
    AST_TEST_UNREGISTER(nack_overflow);
    AST_TEST_UNREGISTER(lost_packet_stats_nominal);
    AST_TEST_UNREGISTER(remb_nominal);
    AST_TEST_UNREGISTER(sr_rr_nominal);
    AST_TEST_UNREGISTER(fir_nominal);
    return 0;
}
 
/* Register RTP regression tests with the unit test framework. */
static int load_module(void)
{
    AST_TEST_REGISTER(rtp_port_exhaustion);
    AST_TEST_REGISTER(nack_no_packet_loss);
    AST_TEST_REGISTER(nack_nominal);
    AST_TEST_REGISTER(nack_overflow);
    AST_TEST_REGISTER(lost_packet_stats_nominal);
    AST_TEST_REGISTER(remb_nominal);
    AST_TEST_REGISTER(sr_rr_nominal);
    AST_TEST_REGISTER(fir_nominal);
    AST_TEST_REGISTER(mes);
    AST_TEST_REGISTER(payload_merge_abandoned_offer);
    AST_TEST_REGISTER(payload_merge_preserves_preferences);
    return AST_MODULE_LOAD_SUCCESS;
}
 
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "RTP/RTCP test module");