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Fix and test for multiple mlines of same type in sdp

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This commit is contained in:
Sylvain Berfini 2015-09-22 10:47:43 +02:00
parent db6cb81f91
commit 0e439d9196
3 changed files with 208 additions and 15 deletions
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52
coreapi/linphonecall.c
View file

@ -1088,16 +1088,23 @@ void linphone_call_set_compatible_incoming_call_parameters(LinphoneCall *call, c
static void linphone_call_compute_streams_indexes(LinphoneCall *call, SalMediaDescription *md) {
int i, j;
bool_t audio_found = FALSE, video_found = FALSE, text_found = FALSE;
for (i = 0; i < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; i++) {
if (!sal_stream_description_active(&md->streams[i])) continue;
if (md->streams[i].type == SalAudio && i != call->main_audio_stream_index) {
call->main_audio_stream_index = i;
ms_message("audio stream index updated: %i", i);
if (md->streams[i].type == SalAudio) {
if (!audio_found) {
call->main_audio_stream_index = i;
audio_found = TRUE;
ms_message("audio stream index found: %i, updating main audio stream index", i);
} else {
ms_message("audio stream index found: %i, but main audio stream already set to %i", i, call->main_audio_stream_index);
}
// Check that the default value of a another stream doesn't match the new one
if (i == call->main_video_stream_index) {
for (j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
if (sal_stream_description_active(&md->streams[j])) continue;
if (j != call->main_video_stream_index && j != call->main_text_stream_index) {
ms_message("%i was used for video stream ; now using %i", i, j);
call->main_video_stream_index = j;
@ -1107,6 +1114,7 @@ static void linphone_call_compute_streams_indexes(LinphoneCall *call, SalMediaDe
}
if (i == call->main_text_stream_index) {
for (j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
if (sal_stream_description_active(&md->streams[j])) continue;
if (j != call->main_video_stream_index && j != call->main_text_stream_index) {
ms_message("%i was used for text stream ; now using %i", i, j);
call->main_text_stream_index = j;
@ -1114,13 +1122,19 @@ static void linphone_call_compute_streams_indexes(LinphoneCall *call, SalMediaDe
}
}
}
} else if (md->streams[i].type == SalVideo && i != call->main_video_stream_index) {
call->main_video_stream_index = i;
ms_message("video stream index updated: %i", i);
} else if (md->streams[i].type == SalVideo) {
if (!video_found) {
call->main_video_stream_index = i;
video_found = TRUE;
ms_message("video stream index found: %i, updating main video stream index", i);
} else {
ms_message("video stream index found: %i, but main video stream already set to %i", i, call->main_video_stream_index);
}
// Check that the default value of a another stream doesn't match the new one
if (i == call->main_audio_stream_index) {
for (j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
if (sal_stream_description_active(&md->streams[j])) continue;
if (j != call->main_audio_stream_index && j != call->main_text_stream_index) {
ms_message("%i was used for audio stream ; now using %i", i, j);
call->main_audio_stream_index = j;
@ -1130,6 +1144,7 @@ static void linphone_call_compute_streams_indexes(LinphoneCall *call, SalMediaDe
}
if (i == call->main_text_stream_index) {
for (j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
if (sal_stream_description_active(&md->streams[j])) continue;
if (j != call->main_audio_stream_index && j != call->main_text_stream_index) {
ms_message("%i was used for text stream ; now using %i", i, j);
call->main_text_stream_index = j;
@ -1137,25 +1152,32 @@ static void linphone_call_compute_streams_indexes(LinphoneCall *call, SalMediaDe
}
}
}
} else if (md->streams[i].type == SalText && i != call->main_text_stream_index) {
call->main_text_stream_index = i;
ms_message("text stream index updated: %i", i);
} else if (md->streams[i].type == SalText) {
if (!text_found) {
call->main_text_stream_index = i;
text_found = TRUE;
ms_message("text stream index found: %i, updating main text stream index", i);
} else {
ms_message("text stream index found: %i, but main text stream already set to %i", i, call->main_text_stream_index);
}
// Check that the default value of a another stream doesn't match the new one
if (i == call->main_video_stream_index) {
if (i == call->main_audio_stream_index) {
for (j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
if (sal_stream_description_active(&md->streams[j])) continue;
if (j != call->main_video_stream_index && j != call->main_audio_stream_index) {
ms_message("%i was used for video stream ; now using %i", i, j);
call->main_video_stream_index = j;
ms_message("%i was used for audio stream ; now using %i", i, j);
call->main_audio_stream_index = j;
break;
}
}
}
if (i == call->main_audio_stream_index) {
if (i == call->main_video_stream_index) {
for (j = 0; j < SAL_MEDIA_DESCRIPTION_MAX_STREAMS; j++) {
if (sal_stream_description_active(&md->streams[j])) continue;
if (j != call->main_video_stream_index && j != call->main_audio_stream_index) {
ms_message("%i was used for audio stream ; now using %i", i, j);
call->main_audio_stream_index = j;
ms_message("%i was used for video stream ; now using %i", i, j);
call->main_video_stream_index = j;
break;
}
}

39
tester/complex_sip_call.c
View file

@ -190,10 +190,49 @@ static void call_with_video_mline_before_audio_in_sdp() {
linphone_core_manager_destroy(mgr);
}
static void call_with_multiple_audio_mline_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_multiple_audio_mline_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, 0, int, "%d");
BC_ASSERT_EQUAL(call->main_video_stream_index, 2, int, "%d");
BC_ASSERT_TRUE(call->main_text_stream_index > 2);
BC_ASSERT_TRUE(linphone_call_log_video_enabled(linphone_call_get_call_log(call)));
check_rtcp(call);
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 without sdp", sip_update_within_icoming_reinvite_with_no_sdp },
{ "Call with audio mline before video in sdp", call_with_audio_mline_before_video_in_sdp },
{ "Call with video mline before audio in sdp", call_with_video_mline_before_audio_in_sdp },
{ "Call with multiple audio mline in sdp", call_with_multiple_audio_mline_in_sdp },
};
test_suite_t complex_sip_call_test_suite = {

132
tester/sipp/call_with_multiple_audio_mline_in_sdp.xml Normal file
View file

@ -0,0 +1,132 @@
<?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=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
m=audio [media_port+1] 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
m=video [media_port+2] RTP/AVP 96
a=rtpmap:96 VP8/90000
]]>
</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>