res_rtp: Add unit tests for RTCP stats.

Added unit tests for RTCP video stats. These tests include NACK, REMB,
FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR
tests are currently disabled due to a bug. We expect to receive a
compound packet, but the code sends this out as a single packet, which
the browser accepts, but makes Asterisk upset.

While writing these tests, I noticed an issue with NACK as well. Where
it is handling a received NACK request, it was reading in only the first
8 bits of following packets that were also lost. This has been changed
to the correct value of 16 bits.

Also made a minor fix to the data buffer unit test.

Change-Id: I56107c7411003a247589bbb6086d25c54719901b
This commit is contained in:
Ben Ford 2019-08-28 14:25:57 -05:00 committed by Benjamin Keith Ford
parent 2ae1a22e0e
commit 0e56643d9f
5 changed files with 831 additions and 6 deletions

View File

@ -589,6 +589,26 @@ struct ast_rtp_engine_dtls {
const char *(*get_fingerprint)(struct ast_rtp_instance *instance);
};
#ifdef TEST_FRAMEWORK
/*! \brief Structure that represents the test functionality for res_rtp_asterisk unit tests */
struct ast_rtp_engine_test {
/*! Drops RTP packets while this has a value greater than 0 */
int packets_to_drop;
/*! Sends a SR/RR instead of RTP the next time RTP would be sent */
int send_report;
/*! Set to 1 whenever SDES is received */
int sdes_received;
/*! Get the number of packets in the receive buffer for a RTP instance */
size_t (*recv_buffer_count)(struct ast_rtp_instance *instance);
/*! Get the maximum number of packets the receive buffer can hold for a RTP instance */
size_t (*recv_buffer_max)(struct ast_rtp_instance *instance);
/*! Get the number of packets in the send buffer for a RTP instance */
size_t (*send_buffer_count)(struct ast_rtp_instance *instance);
/*! Set the schedid for RTCP */
void (*set_schedid)(struct ast_rtp_instance *instance, int id);
};
#endif
/*! Structure that represents an RTP stack (engine) */
struct ast_rtp_engine {
/*! Name of the RTP engine, used when explicitly requested */
@ -670,6 +690,10 @@ struct ast_rtp_engine {
struct ast_rtp_engine_ice *ice;
/*! Callback to pointer for optional DTLS SRTP support */
struct ast_rtp_engine_dtls *dtls;
#ifdef TEST_FRAMEWORK
/*! Callback to pointer for test callbacks for RTP/RTCP unit tests */
struct ast_rtp_engine_test *test;
#endif
/*! Callback to enable an RTP extension (returns non-zero if supported) */
int (*extension_enable)(struct ast_rtp_instance *instance, enum ast_rtp_extension extension);
/*! Linked list information */
@ -2513,6 +2537,18 @@ int ast_rtp_engine_unload_format(struct ast_format *format);
*/
struct ast_rtp_engine_ice *ast_rtp_instance_get_ice(struct ast_rtp_instance *instance);
#ifdef TEST_FRAMEWORK
/*!
* \brief Obtain a pointer to the test callbacks on an RTP instance
*
* \param instance the RTP instance
*
* \retval test callbacks if present
* \retval NULL if not present
*/
struct ast_rtp_engine_test *ast_rtp_instance_get_test(struct ast_rtp_instance *instance);
#endif
/*!
* \brief Obtain a pointer to the DTLS support present on an RTP instance
*
@ -2686,6 +2722,70 @@ struct stasis_message_type *ast_rtp_rtcp_sent_type(void);
*/
struct stasis_message_type *ast_rtp_rtcp_received_type(void);
#ifdef TEST_FRAMEWORK
/*!
* \brief Get the maximum size of the receive buffer
*
* \param instance The RTP instance
* \retval The recv_buffer max size if it exists, else 0
*/
size_t ast_rtp_instance_get_recv_buffer_max(struct ast_rtp_instance *instance);
/*!
* \brief Get the current size of the receive buffer
*
* \param instance The RTP instance
* \retval The recv_buffer size if it exists, else 0
*/
size_t ast_rtp_instance_get_recv_buffer_count(struct ast_rtp_instance *instance);
/*!
* \brief Get the current size of the send buffer
*
* \param instance The RTP instance
* \retval The send_buffer size if it exists, else 0
*/
size_t ast_rtp_instance_get_send_buffer_count(struct ast_rtp_instance *instance);
/*!
* \brief Set the schedid for RTCP
*
* \param instance The RTP instance
* \param id The number to set schedid to
*/
void ast_rtp_instance_set_schedid(struct ast_rtp_instance *instance, int id);
/*!
* \brief Set the number of packets to drop on RTP read
*
* \param instance The RTP instance
* \param num The number of packets to drop
*/
void ast_rtp_instance_drop_packets(struct ast_rtp_instance *instance, int num);
/*!
* \brief Sends a SR/RR report the next time RTP would be sent
*
* \param instance The RTP instance
*/
void ast_rtp_instance_queue_report(struct ast_rtp_instance *instance);
/*!
* \brief Get the value of sdes_received on the test engine
*
* \param instance The RTP instance
* \retval 1 if sdes_received, else 0
*/
int ast_rtp_instance_get_sdes_received(struct ast_rtp_instance *instance);
/*!
* \brief Resets all the fields to default values for the test engine
*
* \param instance The RTP instance
*/
void ast_rtp_instance_reset_test_engine(struct ast_rtp_instance *instance);
#endif
/*!
* \brief Convert given stat instance into json format
* \param stats

View File

@ -457,7 +457,7 @@ static void instance_destructor(void *obj)
int ast_rtp_instance_destroy(struct ast_rtp_instance *instance)
{
ao2_ref(instance, -1);
ao2_cleanup(instance);
return 0;
}
@ -2897,6 +2897,13 @@ struct ast_rtp_engine_ice *ast_rtp_instance_get_ice(struct ast_rtp_instance *ins
return NULL;
}
#ifdef TEST_FRAMEWORK
struct ast_rtp_engine_test *ast_rtp_instance_get_test(struct ast_rtp_instance *instance)
{
return instance->engine->test;
}
#endif
static int rtp_dtls_wrap_set_configuration(struct ast_rtp_instance *instance,
const struct ast_rtp_dtls_cfg *dtls_cfg)
{
@ -3759,6 +3766,123 @@ void ast_rtp_instance_set_stream_num(struct ast_rtp_instance *rtp, int stream_nu
ao2_unlock(rtp);
}
#ifdef TEST_FRAMEWORK
size_t ast_rtp_instance_get_recv_buffer_max(struct ast_rtp_instance *instance)
{
size_t res;
struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
if (!test) {
ast_log(LOG_ERROR, "There is no test engine set up!\n");
return 0;
}
ao2_lock(instance);
res = test->recv_buffer_max(instance);
ao2_unlock(instance);
return res;
}
size_t ast_rtp_instance_get_recv_buffer_count(struct ast_rtp_instance *instance)
{
size_t res;
struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
if (!test) {
ast_log(LOG_ERROR, "There is no test engine set up!\n");
return 0;
}
ao2_lock(instance);
res = test->recv_buffer_count(instance);
ao2_unlock(instance);
return res;
}
size_t ast_rtp_instance_get_send_buffer_count(struct ast_rtp_instance *instance)
{
size_t res;
struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
if (!test) {
ast_log(LOG_ERROR, "There is no test engine set up!\n");
return 0;
}
ao2_lock(instance);
res = test->send_buffer_count(instance);
ao2_unlock(instance);
return res;
}
void ast_rtp_instance_set_schedid(struct ast_rtp_instance *instance, int id)
{
struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
if (!test) {
ast_log(LOG_ERROR, "There is no test engine set up!\n");
return;
}
ao2_lock(instance);
test->set_schedid(instance, id);
ao2_unlock(instance);
}
void ast_rtp_instance_drop_packets(struct ast_rtp_instance *instance, int num)
{
struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
if (!test) {
ast_log(LOG_ERROR, "There is no test engine set up!\n");
return;
}
test->packets_to_drop = num;
}
void ast_rtp_instance_queue_report(struct ast_rtp_instance *instance)
{
struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
if (!test) {
ast_log(LOG_ERROR, "There is no test engine set up!\n");
return;
}
test->send_report = 1;
}
int ast_rtp_instance_get_sdes_received(struct ast_rtp_instance *instance)
{
struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
if (!test) {
ast_log(LOG_ERROR, "There is no test engine set up!\n");
return 0;
}
return test->sdes_received;
}
void ast_rtp_instance_reset_test_engine(struct ast_rtp_instance *instance)
{
struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
if (!test) {
ast_log(LOG_ERROR, "There is no test engine set up!\n");
return;
}
test->packets_to_drop = 0;
test->send_report = 0;
test->sdes_received = 0;
}
#endif
struct ast_json *ast_rtp_convert_stats_json(const struct ast_rtp_instance_stats *stats)
{
struct ast_json *j_res;

View File

@ -2371,6 +2371,60 @@ static struct ast_rtp_engine_dtls ast_rtp_dtls = {
#endif
#ifdef TEST_FRAMEWORK
static size_t get_recv_buffer_count(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
if (rtp && rtp->recv_buffer) {
return ast_data_buffer_count(rtp->recv_buffer);
}
return 0;
}
static size_t get_recv_buffer_max(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
if (rtp && rtp->recv_buffer) {
return ast_data_buffer_max(rtp->recv_buffer);
}
return 0;
}
static size_t get_send_buffer_count(struct ast_rtp_instance *instance)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
if (rtp && rtp->send_buffer) {
return ast_data_buffer_count(rtp->send_buffer);
}
return 0;
}
static void set_rtp_rtcp_schedid(struct ast_rtp_instance *instance, int id)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
if (rtp && rtp->rtcp) {
rtp->rtcp->schedid = id;
}
}
static struct ast_rtp_engine_test ast_rtp_test = {
.packets_to_drop = 0,
.send_report = 0,
.sdes_received = 0,
.recv_buffer_count = get_recv_buffer_count,
.recv_buffer_max = get_recv_buffer_max,
.send_buffer_count = get_send_buffer_count,
.set_schedid = set_rtp_rtcp_schedid,
};
#endif
/* RTP Engine Declaration */
static struct ast_rtp_engine asterisk_rtp_engine = {
.name = "asterisk",
@ -2410,6 +2464,9 @@ static struct ast_rtp_engine asterisk_rtp_engine = {
.set_stream_num = ast_rtp_set_stream_num,
.extension_enable = ast_rtp_extension_enable,
.bundle = ast_rtp_bundle,
#ifdef TEST_FRAMEWORK
.test = &ast_rtp_test,
#endif
};
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
@ -2923,11 +2980,21 @@ static int __rtp_recvfrom(struct ast_rtp_instance *instance, void *buf, size_t s
#ifdef HAVE_PJPROJECT
struct ast_sockaddr *loop = rtcp ? &rtp->rtcp_loop : &rtp->rtp_loop;
#endif
#ifdef TEST_FRAMEWORK
struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
#endif
if ((len = ast_recvfrom(rtcp ? rtp->rtcp->s : rtp->s, buf, size, flags, sa)) < 0) {
return len;
return len;
}
#ifdef TEST_FRAMEWORK
if (test && test->packets_to_drop > 0) {
test->packets_to_drop--;
return 0;
}
#endif
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
/* If this is an SSL packet pass it to OpenSSL for processing. RFC section for first byte value:
* https://tools.ietf.org/html/rfc5764#section-5.1.2 */
@ -4595,6 +4662,9 @@ static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *fr
struct ast_sockaddr remote_address = { {0,} };
int rate = rtp_get_rate(frame->subclass.format) / 1000;
unsigned int seqno;
#ifdef TEST_FRAMEWORK
struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
#endif
if (ast_format_cmp(frame->subclass.format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) {
frame->samples /= 2;
@ -4604,6 +4674,14 @@ static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *fr
return 0;
}
#ifdef TEST_FRAMEWORK
if (test && test->send_report) {
test->send_report = 0;
ast_rtcp_write(instance);
return 0;
}
#endif
if (frame->frametype == AST_FRAME_VOICE) {
pred = rtp->lastts + frame->samples;
@ -5641,7 +5719,7 @@ static int ast_rtp_rtcp_handle_nack(struct ast_rtp_instance *instance, unsigned
* packet (pid+i)(modulo 2^16). Otherwise, it is set to 0. We cannot assume bits set
* to 0 after a bit set to 1 have actually been received.
*/
blp = current_word & 0xFF;
blp = current_word & 0xffff;
blp_index = 1;
while (blp) {
if (blp & 1) {
@ -5721,6 +5799,9 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, s
unsigned int ssrc_seen;
struct ast_rtp_rtcp_report_block *report_block;
struct ast_frame *f = &ast_null_frame;
#ifdef TEST_FRAMEWORK
struct ast_rtp_engine_test *test_engine;
#endif
/* If this is encrypted then decrypt the payload */
if ((*rtcpheader & 0xC0) && res_srtp && srtp && res_srtp->unprotect(
@ -6161,6 +6242,11 @@ static struct ast_frame *ast_rtcp_interpret(struct ast_rtp_instance *instance, s
ast_verbose("Received an SDES from %s\n",
ast_sockaddr_stringify(addr));
}
#ifdef TEST_FRAMEWORK
if ((test_engine = ast_rtp_instance_get_test(instance))) {
test_engine->sdes_received = 1;
}
#endif
break;
case RTCP_PT_BYE:
if (rtcp_debug_test_addr(addr)) {
@ -7656,11 +7742,10 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
if (res < 0) {
ast_debug(1, "Failed to send NACK request out\n");
} else {
ast_debug(2, "Sending a NACK request on RTP instance '%p' to get missing packets\n", instance);
/* Update RTCP SR/RR statistics */
ast_rtcp_calculate_sr_rr_statistics(instance, rtcp_report, remote_address, ice, sr);
}
ast_debug(2, "Sending a NACK request on RTP instance '%p' to get missing packets\n", instance);
}
return &ast_null_frame;

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@ -18,7 +18,7 @@
/*!
* \file
* \brief Media Stream API Unit Tests
* \brief Data Buffer API Unit Tests
*
* \author Ben Ford <bford@digium.com>
*

516
tests/test_res_rtp.c Normal file
View File

@ -0,0 +1,516 @@
/*
* 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"
enum test_type {
TEST_TYPE_NONE = 0, /* No special setup required */
TEST_TYPE_NACK, /* Enable NACK */
TEST_TYPE_REMB, /* Enable REMB */
};
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 addr;
ast_sockaddr_parse(&addr, "127.0.0.1", 0);
*instance1 = ast_rtp_instance_new("asterisk", test_sched, &addr, NULL);
*instance2 = ast_rtp_instance_new("asterisk", test_sched, &addr, NULL);
if (!instance1 || !instance2) {
return -1;
}
ast_rtp_instance_set_prop(*instance1, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_MUX);
ast_rtp_instance_set_prop(*instance2, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_MUX);
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, &addr);
ast_rtp_instance_set_remote_address(*instance2, &addr);
ast_rtp_instance_get_local_address(*instance2, &addr);
ast_rtp_instance_set_remote_address(*instance1, &addr);
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);
}
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_RXPLOSS;
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,
"Condition of 5 lost packets was not met");
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),
};
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:
/* Disable for now - there's a bug! */
return AST_TEST_NOT_RUN;
}
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);
/*
* There may be some additional work that needs to be done here, depending on how
* Asterisk handles the reading in of compound packets. We might get an ast_null_frame
* here instead of the REMB frame. We'll need to check the frametype to distinguish
* between them (AST_FRAME_NULL for ast_null_frame, AST_FRAME_RTCP for REMB).
*/
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");
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:
/* Disable for now - there's a bug! */
return AST_TEST_NOT_RUN;
}
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.
*
* NOTE - similar to REMB, there may be more that needs to be done here when the
* packet is sent as a compound packet!
*/
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;
}
static int unload_module(void)
{
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;
}
static int load_module(void)
{
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);
return AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "RTP/RTCP test module");