[core] RTP: a media timeout fix + add pcap based-unit tests.

This commit is contained in:
Dragos Oancea 2022-03-20 17:25:37 +03:00 committed by Andrey Volk
parent 75e858407f
commit beffab1d68
8 changed files with 687 additions and 2 deletions

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@ -1970,6 +1970,22 @@ else
AC_MSG_WARN([python3 support disabled, building mod_python3 will fail!])
fi
# pcap lib for unit-testing
AC_MSG_CHECKING(libpcap)
AC_CHECK_PROG(HAVE_PCAP_CONFIG,pcap-config,[true],[false])
if test x"$HAVE_PCAP_CONFIG" = x"true"; then
AC_MSG_RESULT(yes)
PCAP_CONFIG=pcap-config
PCAP_LIBS="`$PCAP_CONFIG --libs`"
PCAP_CFLAGS="`$PCAP_CONFIG --cflags`"
AM_CONDITIONAL([HAVE_PCAP], [true])
else
AC_MSG_RESULT(no)
AM_CONDITIONAL([HAVE_PCAP], [false])
fi
AC_SUBST([PCAP_CFLAGS])
AC_SUBST([PCAP_LIBS])
#
# SNMP checks for mod_snmp
#

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@ -2882,8 +2882,18 @@ static void check_media_timeout_params(switch_core_session_t *session, switch_rt
if (switch_rtp_ready(engine->rtp_session) && engine->media_timeout) {
switch_rtp_set_media_timeout(engine->rtp_session, engine->media_timeout);
}
if (engine->type == SWITCH_MEDIA_TYPE_AUDIO) {
/* the values are in milliseconds, not in seconds as the deprecated rtp_timeout_sec */
engine->max_missed_packets = (engine->read_impl.samples_per_second * engine->media_timeout / 1000) / engine->read_impl.samples_per_packet;
switch_rtp_set_max_missed_packets(engine->rtp_session, engine->max_missed_packets);
if (!engine->media_hold_timeout) {
engine->media_hold_timeout = engine->media_timeout * 10;
}
engine->max_missed_hold_packets = (engine->read_impl.samples_per_second * engine->media_hold_timeout / 1000) / engine->read_impl.samples_per_packet;
}
}
}
SWITCH_DECLARE(switch_status_t) switch_core_media_read_frame(switch_core_session_t *session, switch_frame_t **frame,

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@ -7350,7 +7350,7 @@ static void check_timeout(switch_rtp_t *rtp_session)
}
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG10,
"%s MEDIA TIMEOUT %s %d/%d", switch_core_session_get_name(rtp_session->session), rtp_type(rtp_session),
"%s MEDIA TIMEOUT %s %d/%d\n", switch_core_session_get_name(rtp_session->session), rtp_type(rtp_session),
elapsed, rtp_session->media_timeout);
if (elapsed > rtp_session->media_timeout) {

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@ -5,6 +5,12 @@ noinst_PROGRAMS = switch_event switch_hash switch_ivr_originate switch_utils swi
noinst_PROGRAMS += switch_core_video switch_core_db switch_vad switch_packetizer switch_core_session test_sofia switch_ivr_async switch_core_asr switch_log
noinst_PROGRAMS+= switch_hold switch_sip
if HAVE_PCAP
noinst_PROGRAMS += switch_rtp_pcap
AM_LDFLAGS += $(PCAP_LIBS)
endif
AM_LDFLAGS += -avoid-version -no-undefined $(SWITCH_AM_LDFLAGS) $(openssl_LIBS)
AM_LDFLAGS += $(FREESWITCH_LIBS) $(switch_builddir)/libfreeswitch.la $(CORE_LIBS) $(APR_LIBS)

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@ -0,0 +1,73 @@
<?xml version="1.0"?>
<document type="freeswitch/xml">
<X-PRE-PROCESS cmd="exec-set" data="test=echo 1234"/>
<X-PRE-PROCESS cmd="set" data="default_password=$${test}"/>
<X-PRE-PROCESS cmd="set" data="core_video_blank_image=$${conf_dir}/freeswitch-logo.png"/>
<section name="configuration" description="Various Configuration">
<configuration name="modules.conf" description="Modules">
<modules>
<load module="mod_console"/>
<load module="mod_loopback"/>
<load module="mod_commands"/>
<load module="mod_sndfile"/>
<load module="mod_dptools"/>
<load module="mod_tone_stream"/>
<load module="mod_test"/>
</modules>
</configuration>
<configuration name="console.conf" description="Console Logger">
<mappings>
<map name="all" value="console,debug,info,notice,warning,err,crit,alert"/>
</mappings>
<settings>
<param name="colorize" value="true"/>
<param name="loglevel" value="debug"/>
</settings>
</configuration>
<configuration name="switch.conf" description="Core Configuration">
<default-ptimes>
</default-ptimes>
<settings>
<param name="colorize-console" value="false"/>
<param name="dialplan-timestamps" value="false"/>
<param name="loglevel" value="debug"/>
<param name="rtp-start-port" value="1234"/>
<param name="rtp-end-port" value="1234"/>
<param name="rtp-enable-zrtp" value="false"/>
</settings>
</configuration>
<configuration name="timezones.conf" description="Timezones">
<timezones>
<zone name="GMT" value="GMT0" />
</timezones>
</configuration>
</section>
<section name="dialplan" description="Regex/XML Dialplan">
<context name="test">
<extension>
<condition field="${sip_h_X-COUNTDOWN}" expression="^0$" break="on-true">
<action application="answer"/>
<action application="playback" data="tone_stream://%(251,0,1004);loops=-1"/>
</condition>
<condition field="${sip_h_X-COUNTDOWN}" expression="^(\d+)$" break="never">
<action application="export" data="_nolocal_sip_h_X-COUNTDOWN=${expr($1 - 1)}"/>
<anti-action application="export" data="_nolocal_sip_h_X-COUNTDOWN=10"/>
</condition>
<condition>
<action application="bridge" data="sofia/test/1234@127.0.0.1"/>
</condition>
</extension>
</context>
</section>
</document>

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@ -0,0 +1,580 @@
/*
* FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application
* Copyright (C) 2005-2021, Anthony Minessale II <anthm@freeswitch.org>
*
* Version: MPL 1.1
*
* The contents of this file are subject to the Mozilla Public License Version
* 1.1 (the "License"); you may not use this file except in compliance with
* the License. You may obtain a copy of the License at
* http://www.mozilla.org/MPL/
*
* Software distributed under the License is distributed on an "AS IS" basis,
* WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License
* for the specific language governing rights and limitations under the
* License.
*
* The Original Code is FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application
*
* The Initial Developer of the Original Code is
* Anthony Minessale II <anthm@freeswitch.org>
* Portions created by the Initial Developer are Copyright (C)
* the Initial Developer. All Rights Reserved.
*
* Contributor(s):
* Dragos Oancea <dragos@signalwire.com>
*
* switch_rtp_pcap.c -- tests RTP stack using PCAP.
*/
#include <switch.h>
#include <test/switch_test.h>
/* before adding a pcap file: tcprewrite --dstipmap=X.X.X.X/32:192.168.0.1/32 --srcipmap=X.X.X.X/32:192.168.0.2/32 -i in.pcap -o out.pcap */
#include <pcap.h>
#ifndef MSG_CONFIRM
#define MSG_CONFIRM 0
#endif
static const char *rx_host = "127.0.0.1";
static const char *tx_host = "127.0.0.1";
static switch_rtp_t *rtp_session = NULL;
const char *err = NULL;
switch_rtp_packet_t rtp_packet;
switch_frame_flag_t *frame_flags;
switch_io_flag_t io_flags;
switch_payload_t read_pt;
static switch_port_t audio_rx_port = 1234;
static int got_media_timeout = 0;
//#define USE_RTCP_PCAP
#define NTP_TIME_OFFSET 2208988800UL
/* https://www.tcpdump.org/pcap.html */
/* IP header */
struct sniff_ip {
u_char ip_vhl; /* version << 4 | header length >> 2 */
u_char ip_tos; /* type of service */
u_short ip_len; /* total length */
u_short ip_id; /* identification */
u_short ip_off; /* fragment offset field */
#define IP_RF 0x8000 /* reserved fragment flag */
#define IP_DF 0x4000 /* dont fragment flag */
#define IP_MF 0x2000 /* more fragments flag */
#define IP_OFFMASK 0x1fff /* mask for fragmenting bits */
u_char ip_ttl; /* time to live */
u_char ip_p; /* protocol */
u_short ip_sum; /* checksum */
struct in_addr ip_src,ip_dst; /* source and dest address */
};
#define IP_HL(ip) (((ip)->ip_vhl) & 0x0f)
/* switch_rtp.c - calc_local_lsr_now() */
static inline uint32_t test_calc_local_lsr_now(switch_time_t now, uint32_t past /*milliseconds*/)
{
// switch_time_t now;
uint32_t ntp_sec, ntp_usec, lsr_now, sec;
// now = switch_micro_time_now() - (past * 1000);
now = now - (past * 1000);
sec = (uint32_t)(now/1000000); /* convert to seconds */
ntp_sec = sec+NTP_TIME_OFFSET; /* convert to NTP seconds */
ntp_usec = (uint32_t)(now - ((switch_time_t) sec*1000000)); /* remove seconds to keep only the microseconds */
lsr_now = (uint32_t)(ntp_usec*0.065536) | (ntp_sec&0x0000ffff)<<16; /* 0.065536 is used for convertion from useconds to fraction of 65536 (x65536/1000000) */
return lsr_now;
}
#if 0
static void test_prepare_rtcp(void *rtcp_packet, float est_last, uint32_t rtt, uint8_t loss)
{
/* taken from switch_rtp.c, rtcp_generate_sender_info() */
/* === */
char *rtcp_sr_trigger = rtcp_packet;
switch_time_t now;
uint32_t sec, ntp_sec, ntp_usec;
uint32_t ntp_msw;
uint32_t ntp_lsw;
uint32_t *ptr_msw;
uint32_t *ptr_lsw;
uint32_t lsr;
uint32_t *ptr_lsr;
uint32_t dlsr = 0;
uint32_t *ptr_dlsr;
uint8_t *ptr_loss;
now = switch_micro_time_now();
sec = (uint32_t)(now/1000000); /* convert to seconds */
ntp_sec = sec+NTP_TIME_OFFSET; /* convert to NTP seconds */
ntp_msw = htonl(ntp_sec); /* store result in "most significant word" */
ntp_usec = (uint32_t)(now - (sec*1000000)); /* remove seconds to keep only the microseconds */
ntp_lsw = htonl((u_long)(ntp_usec*(double)(((uint64_t)1)<<32)*1.0e-6));
/* === */
/*patch the RTCP payload to set the RTT we want */
ptr_msw = (uint32_t *)rtcp_sr_trigger + 2;
*ptr_msw = ntp_msw;
ptr_lsw = (uint32_t *)rtcp_sr_trigger + 3;
*ptr_lsw = ntp_lsw;
lsr = test_calc_local_lsr_now(now, est_last * 1000 + rtt /*ms*/);
ptr_lsr = (uint32_t *)rtcp_sr_trigger + 11;
*ptr_lsr = htonl(lsr);
ptr_dlsr = (uint32_t *)rtcp_sr_trigger + 12;
*ptr_dlsr = htonl(dlsr);
ptr_loss = (uint8_t *)rtcp_sr_trigger + 32;
*ptr_loss = loss;
}
#endif
static switch_status_t rtp_test_start_call(switch_core_session_t **psession)
{
char *r_sdp;
uint8_t match = 0, p = 0;
switch_core_session_t *session;
switch_channel_t *channel = NULL;
switch_status_t status;
switch_media_handle_t *media_handle;
switch_core_media_params_t *mparams;
switch_stream_handle_t stream = { 0 };
switch_call_cause_t cause;
/*tone stream extension*/
status = switch_ivr_originate(NULL, psession, &cause, "null/+1234", 2, NULL, NULL, NULL, NULL, NULL, SOF_NONE, NULL, NULL);
session = *psession;
if (!(session)) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "no session\n");
return SWITCH_STATUS_FALSE;
}
if (status != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "switch_ivr_originate() failed\n");
return SWITCH_STATUS_FALSE;
}
channel = switch_core_session_get_channel(session);
if (!channel) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "switch_core_session_get_channel() failed\n");
return SWITCH_STATUS_FALSE;
}
mparams = switch_core_session_alloc(session, sizeof(switch_core_media_params_t));
mparams->inbound_codec_string = switch_core_session_strdup(session, "PCMU");
mparams->outbound_codec_string = switch_core_session_strdup(session, "PCMU");
mparams->rtpip = switch_core_session_strdup(session, (char *)rx_host);
status = switch_media_handle_create(&media_handle, session, mparams);
if (status != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "switch_media_handle_create() failed\n");
return SWITCH_STATUS_FALSE;
}
switch_channel_set_variable(channel, "absolute_codec_string", "PCMU");
switch_channel_set_variable(channel, "send_silence_when_idle", "-1");
switch_channel_set_variable(channel, "rtp_timer_name", "soft");
switch_channel_set_variable(channel, "media_timeout", "1000");
switch_channel_set_variable(channel, SWITCH_LOCAL_MEDIA_IP_VARIABLE, rx_host);
switch_channel_set_variable_printf(channel, SWITCH_LOCAL_MEDIA_PORT_VARIABLE, "%d", audio_rx_port);
r_sdp = switch_core_session_sprintf(session,
"v=0\n"
"o=FreeSWITCH 1632033305 1632033306 IN IP4 %s\n"
"s=-\n"
"c=IN IP4 %s\n"
"t=0 0\n"
"m=audio 11114 RTP/AVP 0 101\n"
"a=rtpmap:0 PCMU/8000\n"
"a=rtpmap:101 telephone-event/8000\n"
"a=rtcp-mux\n",
tx_host, tx_host);
status = switch_core_media_prepare_codecs(session, SWITCH_FALSE);
if (status != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "switch_core_media_prepare_codecs() failed\n");
return SWITCH_STATUS_FALSE;
}
match = switch_core_media_negotiate_sdp(session, r_sdp, &p, SDP_TYPE_REQUEST);
if (match != 1) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "switch_core_media_negotiate_sdp() failed\n");
return SWITCH_STATUS_FALSE;
}
status = switch_core_media_choose_ports(session, SWITCH_TRUE, SWITCH_FALSE);
if (status != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "switch_core_media_choose_ports() failed\n");
return SWITCH_STATUS_FALSE;
}
status = switch_core_media_activate_rtp(session);
if (status != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_ERROR, "switch_core_media_activate_rtp() failed\n");
return SWITCH_STATUS_FALSE;
}
switch_core_media_set_rtp_flag(session, SWITCH_MEDIA_TYPE_AUDIO, SWITCH_RTP_FLAG_DEBUG_RTP_READ);
switch_core_media_set_rtp_flag(session, SWITCH_MEDIA_TYPE_AUDIO, SWITCH_RTP_FLAG_DEBUG_RTP_WRITE);
SWITCH_STANDARD_STREAM(stream);
switch_api_execute("fsctl", "debug_level 10", session, &stream);
switch_safe_free(stream.data);
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t rtp_test_end_call(switch_core_session_t **psession)
{
switch_channel_t *channel = NULL;
switch_core_session_t *session = *psession;
channel = switch_core_session_get_channel(session);
if (!channel) {
return SWITCH_STATUS_FALSE;
}
switch_channel_hangup(channel, SWITCH_CAUSE_NORMAL_CLEARING);
switch_media_handle_destroy(session);
switch_core_session_rwunlock(session);
return SWITCH_STATUS_SUCCESS;
}
static void rtp_test_init_frame(switch_frame_t **pwrite_frame, switch_core_session_t **psession)
{
const unsigned char hdr_packet[]="\x80\x00\xcd\x15\xfd\x86\x00\x00\x61\x5a\xe1\x37";
switch_frame_alloc(pwrite_frame, SWITCH_RECOMMENDED_BUFFER_SIZE);
(*pwrite_frame)->codec = switch_core_session_get_write_codec(*psession);
(*pwrite_frame)->datalen = SWITCH_RTP_HEADER_LEN; /*init with dummy RTP header*/
memcpy((*pwrite_frame)->data, &hdr_packet, SWITCH_RTP_HEADER_LEN);
}
static void show_event(switch_event_t *event) {
char *str;
/*print the event*/
switch_event_serialize_json(event, &str);
if (str) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "%s\n", str);
switch_safe_free(str);
}
}
static void event_handler(switch_event_t *event)
{
const char *new_ev = switch_event_get_header(event, "Event-Name");
if (new_ev && !strcmp(new_ev, "CHANNEL_HANGUP")) {
if (!strcmp(switch_event_get_header(event, "Hangup-Cause"), "MEDIA_TIMEOUT")) {
got_media_timeout = 1;
}
}
show_event(event);
}
FST_CORE_DB_BEGIN("./conf_rtp")
{
FST_SUITE_BEGIN(switch_rtp_pcap)
{
FST_SETUP_BEGIN()
{
fst_requires_module("mod_loopback");
}
FST_SETUP_END()
FST_TEARDOWN_BEGIN()
{
}
FST_TEARDOWN_END()
#if 0
FST_TEST_BEGIN(test_rtp_stall_with_rtcp_muxed_with_timer)
{
switch_core_session_t *session = NULL;
switch_status_t status;
uint32_t plen = SWITCH_RTP_HEADER_LEN;
char rpacket[SWITCH_RECOMMENDED_BUFFER_SIZE];
switch_payload_t pt = { 0 };
switch_frame_flag_t frameflags = { 0 };
int x = 0;
switch_frame_t *write_frame;
pcap_t *pcap;
const unsigned char *packet;
char errbuf[PCAP_ERRBUF_SIZE];
struct pcap_pkthdr pcap_header;
char rtcp_sr_trigger[] = "\x81\xc8\x00\x0c\x78\x9d\xac\x45\xe2\x67\xa5\x74\x30\x60\x56\x81\x00\x19"
"\xaa\x00\x00\x00\x06\xd7\x00\x01\x2c\x03\x5e\xbd\x2f\x0b\x00"
"\x00\x00\x00\x00\x00\x57\xc4\x00\x00\x00\x39\xa5\x73\xfe\x90\x00\x00\x2c\x87"
"\x81\xca\x00\x0c\x78\x9d\xac\x45\x01\x18\x73\x69\x70\x3a\x64\x72\x40\x31\x39\x32\x2e"
"\x31\x36\x38\x2e\x30\x2e\x31\x33\x3a\x37\x30\x36\x30\x06\x0e\x4c\x69\x6e\x70\x68\x6f"
"\x6e\x65\x2d\x33\x2e\x36\x2e\x31\x00\x00";
const struct sniff_ip *ip; /* The IP header */
int size_ip, jump_over;
struct timeval prev_ts = { 0 };
switch_time_t time_nowpacket = 0, time_prevpacket = 0;
switch_socket_t *sock_rtp = NULL;
switch_sockaddr_t *sock_addr = NULL;
const char *str_err;
switch_size_t rough_add = 0;
status = rtp_test_start_call(&session);
fst_requires(status == SWITCH_STATUS_SUCCESS);
fst_requires(session);
pcap = pcap_open_offline_with_tstamp_precision("pcap/milliwatt.long.pcmu.rtp.pcap", PCAP_TSTAMP_PRECISION_MICRO, errbuf);
fst_requires(pcap);
switch_core_media_set_rtp_flag(session, SWITCH_MEDIA_TYPE_AUDIO, SWITCH_RTP_FLAG_ENABLE_RTCP);
rtp_session = switch_core_media_get_rtp_session(session, SWITCH_MEDIA_TYPE_AUDIO);
rtp_test_init_frame(&write_frame, &session);
switch_rtp_clear_flag(rtp_session, SWITCH_RTP_FLAG_PAUSE);
if (switch_socket_create(&sock_rtp, AF_INET, SOCK_DGRAM, 0, switch_core_session_get_pool(session)) != SWITCH_STATUS_SUCCESS) {
fst_requires(0); /*exit*/
}
switch_sockaddr_new(&sock_addr, rx_host, audio_rx_port, switch_core_session_get_pool(session));
fst_requires(sock_addr);
switch_rtp_set_remote_address(rtp_session, tx_host, switch_sockaddr_get_port(sock_addr), 0, SWITCH_FALSE, &str_err);
switch_rtp_reset(rtp_session);
while ((packet = pcap_next(pcap, &pcap_header))) {
/*assume only UDP/RTP packets in the pcap*/
uint32_t rcvd_datalen = pcap_header.caplen;
size_t len;
switch_size_t tmp_len;
int diff_us = (pcap_header.ts.tv_sec-prev_ts.tv_sec)*1000000+(pcap_header.ts.tv_usec-prev_ts.tv_usec);
if (diff_us > 0) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "SENT pkt diff: %d us\n", diff_us);
usleep(diff_us);
}
prev_ts = pcap_header.ts;
len = pcap_header.caplen;
if (len <= 42) {
continue;
}
ip = (struct sniff_ip*)(packet + 14);
size_ip = IP_HL(ip) * 4;
jump_over = 14 /*SIZE_ETHERNET*/ + size_ip /*IP HDR size*/ + 8 /* UDP HDR SIZE */; /* jump 42 bytes over network layers/headers */
packet += jump_over;
x++;
if (!(x%10)) { /* send a RTCP SR packet every 10th RTP packet */
int add_rtt = 200;
test_prepare_rtcp(&rtcp_sr_trigger, 2, add_rtt, 0xa0);
tmp_len = sizeof(rtcp_sr_trigger);
/*RTCP muxed*/
if (switch_socket_sendto(sock_rtp, sock_addr, MSG_CONFIRM, (const char*)rtcp_sr_trigger, &tmp_len) != SWITCH_STATUS_SUCCESS) {
fst_requires(0);
}
plen = sizeof(rtcp_sr_trigger);
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_INFO, "Sent RTCP. Packet size = [%u]\n", plen);
status = switch_rtp_read(rtp_session, (void *)rpacket, &rcvd_datalen, &pt, &frameflags, io_flags);
if (pt == SWITCH_RTP_CNG_PAYLOAD /*timeout*/) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "read CNG/RTCP, skip\n");
while (1) {
status = switch_rtp_read(rtp_session, (void *)&rpacket, &rcvd_datalen, &pt, &frameflags, io_flags);
if (frameflags || SFF_RTCP) break;
}
}
fst_requires(status == SWITCH_STATUS_SUCCESS);
}
if (packet[0] == 0x80 && packet[1] == 0 /*PCMU*/) {
int16_t *seq = (int16_t *)packet + 1;
plen = len - jump_over;
tmp_len = plen;
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Sent RTP. Packet size = [%u] seq = [%d]\n", plen, htons(*seq));
if (switch_socket_sendto(sock_rtp, sock_addr, MSG_CONFIRM, (const char*)packet, &tmp_len) != SWITCH_STATUS_SUCCESS) {
fst_requires(0);
}
}
status = switch_rtp_read(rtp_session, (void *)&rpacket, &rcvd_datalen, &pt, &frameflags, io_flags);
if (pt == SWITCH_RTP_CNG_PAYLOAD /*timeout*/) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "read CNG, skip\n");
continue;
}
time_prevpacket = time_nowpacket;
time_nowpacket = switch_time_now();
if (time_prevpacket) { // skip init.
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "RECV pkt diff: %ld us\n", time_nowpacket - time_prevpacket);
fst_requires((time_nowpacket - time_prevpacket) < 80000);
rough_add += time_nowpacket - time_prevpacket; /* just add to var for visual comparison */
}
fst_requires(status == SWITCH_STATUS_SUCCESS);
if (pt == SWITCH_RTP_CNG_PAYLOAD /*timeout*/) continue;
fst_requires(rcvd_datalen == plen - SWITCH_RTP_HEADER_LEN);
}
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "RECV total delay: %lu\n", rough_add); /*around 17092408 us*/
switch_yield(1000 * 1000);
if (write_frame) switch_frame_free(&write_frame);
switch_rtp_destroy(&rtp_session);
rtp_test_end_call(&session);
switch_socket_close(sock_rtp);
pcap_close(pcap);
switch_yield(1000 * 1000);
}
FST_TEST_END()
#endif
FST_TEST_BEGIN(test_rtp_media_timeout)
{
switch_core_session_t *session = NULL;
switch_status_t status;
uint32_t plen = SWITCH_RTP_HEADER_LEN;
char rpacket[SWITCH_RECOMMENDED_BUFFER_SIZE];
switch_payload_t pt = { 0 };
switch_frame_flag_t frameflags = { 0 };
int x = 0;
switch_frame_t *write_frame;
pcap_t *pcap;
const unsigned char *packet;
char errbuf[PCAP_ERRBUF_SIZE];
struct pcap_pkthdr pcap_header;
const struct sniff_ip *ip; /* The IP header */
int size_ip, jump_over;
struct timeval prev_ts = { 0 };
switch_socket_t *sock_rtp = NULL;
switch_sockaddr_t *sock_addr = NULL;
const char *str_err;
status = rtp_test_start_call(&session);
fst_requires(status == SWITCH_STATUS_SUCCESS);
fst_requires(session);
switch_event_bind("", SWITCH_EVENT_ALL, SWITCH_EVENT_SUBCLASS_ANY, event_handler, NULL);
pcap = pcap_open_offline_with_tstamp_precision("pcap/milliwatt.pcmu.rtp.pcap", PCAP_TSTAMP_PRECISION_MICRO, errbuf);
fst_requires(pcap);
switch_core_media_set_rtp_flag(session, SWITCH_MEDIA_TYPE_AUDIO, SWITCH_RTP_FLAG_ENABLE_RTCP);
rtp_session = switch_core_media_get_rtp_session(session, SWITCH_MEDIA_TYPE_AUDIO);
fst_requires(rtp_session);
rtp_test_init_frame(&write_frame, &session);
switch_rtp_clear_flag(rtp_session, SWITCH_RTP_FLAG_PAUSE);
if (switch_socket_create(&sock_rtp, AF_INET, SOCK_DGRAM, 0, switch_core_session_get_pool(session)) != SWITCH_STATUS_SUCCESS) {
fst_requires(0); /*exit*/
}
switch_sockaddr_new(&sock_addr, rx_host, audio_rx_port, switch_core_session_get_pool(session));
fst_requires(sock_addr);
switch_rtp_set_remote_address(rtp_session, tx_host, switch_sockaddr_get_port(sock_addr), 0, SWITCH_FALSE, &str_err);
switch_rtp_reset(rtp_session);
/* send 3 packets then wait and expect RTP timeout */
while ((packet = pcap_next(pcap, &pcap_header)) && x < 3) {
/*assume only UDP/RTP packets in the pcap*/
uint32_t rcvd_datalen = pcap_header.caplen;
size_t len;
switch_size_t tmp_len;
int diff_us = (pcap_header.ts.tv_sec-prev_ts.tv_sec)*1000000+(pcap_header.ts.tv_usec-prev_ts.tv_usec);
if (diff_us > 0) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "SENT pkt diff: %d us\n", diff_us);
usleep(diff_us);
}
x++;
prev_ts = pcap_header.ts;
len = pcap_header.caplen;
if (len <= 42) {
continue;
}
ip = (struct sniff_ip*)(packet + 14);
size_ip = IP_HL(ip) * 4;
jump_over = 14 /*SIZE_ETHERNET*/ + size_ip /*IP HDR size*/ + 8 /* UDP HDR SIZE */; /* jump 42 bytes over network layers/headers */
packet += jump_over;
if (packet[0] == 0x80 && packet[1] == 0 /*PCMU*/) {
int16_t *seq = (int16_t *)packet + 1;
plen = len - jump_over;
tmp_len = plen;
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "Sent RTP. Packet size = [%u] seq = [%d]\n", plen, htons(*seq));
if (switch_socket_sendto(sock_rtp, sock_addr, MSG_CONFIRM, (const char*)packet, &tmp_len) != SWITCH_STATUS_SUCCESS) {
fst_requires(0);
}
}
status = switch_rtp_read(rtp_session, (void *)&rpacket, &rcvd_datalen, &pt, &frameflags, io_flags);
if (pt == SWITCH_RTP_CNG_PAYLOAD /*timeout*/) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "read CNG, skip\n");
continue;
}
fst_requires(status == SWITCH_STATUS_SUCCESS);
fst_requires(rcvd_datalen == plen - SWITCH_RTP_HEADER_LEN);
}
x = 150; /* 3 seconds max */
while (x || !got_media_timeout) {
uint32_t rcvd_datalen;
status = switch_rtp_read(rtp_session, (void *)&rpacket, &rcvd_datalen, &pt, &frameflags, io_flags);
if (pt == SWITCH_RTP_CNG_PAYLOAD /*timeout*/) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_DEBUG, "read CNG, skip\n");
}
switch_yield(20 * 1000);
fst_requires(status == SWITCH_STATUS_SUCCESS);
x--;
}
if (write_frame) switch_frame_free(&write_frame);
switch_rtp_destroy(&rtp_session);
rtp_test_end_call(&session);
switch_socket_close(sock_rtp);
pcap_close(pcap);
fst_check(got_media_timeout);
}
FST_TEST_END()
}
FST_SUITE_END()
}
FST_CORE_END()