[Asterisk-code-review] res_rtp: Add unit tests for RTCP stats. (...asterisk[16])
George Joseph
asteriskteam at digium.com
Fri Sep 13 07:05:10 CDT 2019
George Joseph has submitted this change and it was merged. ( https://gerrit.asterisk.org/c/asterisk/+/12813 )
Change subject: res_rtp: Add unit tests for RTCP stats.
......................................................................
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
---
M include/asterisk/rtp_engine.h
M main/rtp_engine.c
M res/res_rtp_asterisk.c
M tests/test_data_buffer.c
A tests/test_res_rtp.c
5 files changed, 831 insertions(+), 6 deletions(-)
Approvals:
Kevin Harwell: Looks good to me, but someone else must approve
Joshua C. Colp: Looks good to me, but someone else must approve
George Joseph: Looks good to me, approved; Approved for Submit
diff --git a/include/asterisk/rtp_engine.h b/include/asterisk/rtp_engine.h
index 206ed63..57f29b4 100644
--- a/include/asterisk/rtp_engine.h
+++ b/include/asterisk/rtp_engine.h
@@ -589,6 +589,26 @@
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_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 @@
*/
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_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
diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index 39ad1b3..3403d70 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -457,7 +457,7 @@
int ast_rtp_instance_destroy(struct ast_rtp_instance *instance)
{
- ao2_ref(instance, -1);
+ ao2_cleanup(instance);
return 0;
}
@@ -2897,6 +2897,13 @@
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 @@
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;
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 08138cb..5da095e 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -2371,6 +2371,60 @@
#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 @@
.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 @@
#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 @@
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 @@
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 @@
* 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 @@
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 @@
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 @@
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;
diff --git a/tests/test_data_buffer.c b/tests/test_data_buffer.c
index 93c2c06..2fd56e1 100644
--- a/tests/test_data_buffer.c
+++ b/tests/test_data_buffer.c
@@ -18,7 +18,7 @@
/*!
* \file
- * \brief Media Stream API Unit Tests
+ * \brief Data Buffer API Unit Tests
*
* \author Ben Ford <bford at digium.com>
*
diff --git a/tests/test_res_rtp.c b/tests/test_res_rtp.c
new file mode 100644
index 0000000..ecedb4f
--- /dev/null
+++ b/tests/test_res_rtp.c
@@ -0,0 +1,516 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2019, Sangoma, Inc.
+ *
+ * Ben Ford <bford at 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 at 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");
--
To view, visit https://gerrit.asterisk.org/c/asterisk/+/12813
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Gerrit-Project: asterisk
Gerrit-Branch: 16
Gerrit-Change-Id: I56107c7411003a247589bbb6086d25c54719901b
Gerrit-Change-Number: 12813
Gerrit-PatchSet: 7
Gerrit-Owner: Benjamin Keith Ford <bford at digium.com>
Gerrit-Reviewer: Friendly Automation
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Joshua C. Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-MessageType: merged
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