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Fix remaining stream_index == 0 ? audio stream : video stream issue + test to check call with video mline before audio in sdp

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This commit is contained in:
Sylvain Berfini 2015-09-21 17:14:03 +02:00
parent 60807065ca
commit 9a99b97a40
4 changed files with 164 additions and 6 deletions
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1
.gitignore vendored
View file

@ -94,3 +94,4 @@ tester/ZIDCache*.xml
tester/stereo-record.wav
.dirstamp
git-clang-format.diff
tester/sipp/*.log

2
coreapi/linphonecall.c
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@ -2044,7 +2044,7 @@ static void port_config_set_random_choosed(LinphoneCall *call, int stream_index,
}
static void _linphone_call_prepare_ice_for_stream(LinphoneCall *call, int stream_index, bool_t create_checklist){
MediaStream *ms=stream_index == 0 ? (MediaStream*)call->audiostream : (MediaStream*)call->videostream;
MediaStream *ms = stream_index == call->main_audio_stream_index ? (MediaStream*)call->audiostream : stream_index == call->main_video_stream_index ? (MediaStream*)call->videostream : (MediaStream*)call->textstream;
if ((linphone_core_get_firewall_policy(call->core) == LinphonePolicyUseIce) && (call->ice_session != NULL)){
IceCheckList *cl;
rtp_session_set_pktinfo(ms->sessions.rtp_session, TRUE);

42
tester/complex_sip_call.c
View file

@ -33,7 +33,7 @@ static FILE *sip_start(const char *senario, const char* dest_username, LinphoneA
else
dest = ms_strdup_printf("%s",linphone_address_get_domain(dest_addres));
command = ms_strdup_printf("sipp -sf %s -s %s %s -trace_err -trace_msg -m 1 -d 1000 ",senario,dest_username,dest);
command = ms_strdup_printf("sipp -sf %s -s %s %s -trace_err -trace_msg -rtp_echo -m 1 -d 1000",senario,dest_username,dest);
ms_message("Starting sipp commad [%s]",command);
file = popen(command, "r");
@ -81,9 +81,7 @@ static void sip_update_within_icoming_reinvite_with_no_sdp(void) {
*/
scen = bc_tester_res("sipp/sip_update_within_icoming_reinvite_with_no_sdp.xml");
sipp_out = sip_start(scen
, linphone_address_get_username(mgr->identity)
, mgr->identity);
sipp_out = sip_start(scen, linphone_address_get_username(mgr->identity), mgr->identity);
if (sipp_out) {
BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallIncomingReceived, 1));
@ -96,8 +94,42 @@ static void sip_update_within_icoming_reinvite_with_no_sdp(void) {
}
static void call_with_video_mline_before_audio_in_sdp() {
LinphoneCoreManager *mgr;
char *identity_char;
char *scen;
FILE * sipp_out;
LinphoneCall *call = NULL;
/*currently we use direct connection because sipp do not properly set ACK request uri*/
mgr= linphone_core_manager_new2( "empty_rc", FALSE);
mgr->identity= linphone_core_get_primary_contact_parsed(mgr->lc);
linphone_address_set_username(mgr->identity,"marie");
identity_char=linphone_address_as_string(mgr->identity);
linphone_core_set_primary_contact(mgr->lc,identity_char);
linphone_core_iterate(mgr->lc);
scen = bc_tester_res("sipp/call_with_video_mline_before_audio_in_sdp.xml");
sipp_out = sip_start(scen, linphone_address_get_username(mgr->identity), mgr->identity);
if (sipp_out) {
BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallIncomingReceived, 1));
call = linphone_core_get_current_call(mgr->lc);
linphone_core_accept_call(mgr->lc, call);
BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallStreamsRunning, 1));
BC_ASSERT_EQUAL(call->main_audio_stream_index, 1, int, "%d");
BC_ASSERT_EQUAL(call->main_video_stream_index, 0, int, "%d");
BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallEnd, 1));
pclose(sipp_out);
}
linphone_core_manager_destroy(mgr);
}
static test_t tests[] = {
{ "SIP UPDATE within incoming reinvite witjout sdp", sip_update_within_icoming_reinvite_with_no_sdp},
{ "SIP UPDATE within incoming reinvite without sdp", sip_update_within_icoming_reinvite_with_no_sdp },
{ "Call with video mline before audio in sdp", call_with_video_mline_before_audio_in_sdp },
};
test_suite_t complex_sip_call_test_suite = {

125
tester/sipp/call_with_video_mline_before_audio_in_sdp.xml Normal file
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@ -0,0 +1,125 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uac' scenario. -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=video [media_port+2] RTP/AVP 96
a=rtpmap:96 VP8/90000
m=audio [media_port] RTP/AVP 96 97 0 8 101 98
a=rtpmap:96 speex/16000
a=fmtp:96 vbr=on
a=rtpmap:97 speex/8000
a=fmtp:97 vbr=on
a=rtpmap:101 telephone-event/16000
a=rtpmap:98 telephone-event/8000
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" optional="true">
</recv>
<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true">
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>