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res_rtp: Add unit tests for RTCP stats.

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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 44068cc6bf
commit 5dad4f4ea1
5 changed files with 831 additions and 6 deletions
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93
res/res_rtp_asterisk.c
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@@ -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;