mirror of
https://gitlab.linphone.org/BC/public/linphone-iphone.git
synced 2026-01-26 15:48:09 +00:00
Added test for multiple video streams in sdp
This commit is contained in:
Sylvain Berfini
parent
0e439d9196
commit
52a5ab76e7
2 changed files with 168 additions and 0 deletions
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@ -228,11 +228,50 @@ static void call_with_multiple_audio_mline_in_sdp() {
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linphone_core_manager_destroy(mgr);
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}
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static void call_with_multiple_video_mline_in_sdp() {
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LinphoneCoreManager *mgr;
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char *identity_char;
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char *scen;
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FILE * sipp_out;
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LinphoneCall *call = NULL;
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/*currently we use direct connection because sipp do not properly set ACK request uri*/
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mgr= linphone_core_manager_new2( "empty_rc", FALSE);
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mgr->identity = linphone_core_get_primary_contact_parsed(mgr->lc);
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linphone_address_set_username(mgr->identity,"marie");
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identity_char = linphone_address_as_string(mgr->identity);
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linphone_core_set_primary_contact(mgr->lc,identity_char);
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linphone_core_iterate(mgr->lc);
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scen = bc_tester_res("sipp/call_with_multiple_video_mline_in_sdp.xml");
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sipp_out = sip_start(scen, linphone_address_get_username(mgr->identity), mgr->identity);
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if (sipp_out) {
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BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallIncomingReceived, 1));
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call = linphone_core_get_current_call(mgr->lc);
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linphone_core_accept_call(mgr->lc, call);
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BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallStreamsRunning, 1));
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BC_ASSERT_EQUAL(call->main_audio_stream_index, 0, int, "%d");
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BC_ASSERT_EQUAL(call->main_video_stream_index, 1, int, "%d");
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BC_ASSERT_TRUE(call->main_text_stream_index > 3);
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BC_ASSERT_TRUE(linphone_call_log_video_enabled(linphone_call_get_call_log(call)));
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check_rtcp(call);
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BC_ASSERT_TRUE(wait_for(mgr->lc, mgr->lc, &mgr->stat.number_of_LinphoneCallEnd, 1));
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pclose(sipp_out);
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}
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linphone_core_manager_destroy(mgr);
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}
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static test_t tests[] = {
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{ "SIP UPDATE within incoming reinvite without sdp", sip_update_within_icoming_reinvite_with_no_sdp },
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{ "Call with audio mline before video in sdp", call_with_audio_mline_before_video_in_sdp },
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{ "Call with video mline before audio in sdp", call_with_video_mline_before_audio_in_sdp },
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{ "Call with multiple audio mline in sdp", call_with_multiple_audio_mline_in_sdp },
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{ "Call with multiple video mline in sdp", call_with_multiple_video_mline_in_sdp },
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};
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test_suite_t complex_sip_call_test_suite = {
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129
tester/sipp/call_with_multiple_video_mline_in_sdp.xml
Normal file
129
tester/sipp/call_with_multiple_video_mline_in_sdp.xml
Normal file
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@ -0,0 +1,129 @@
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<?xml version="1.0" encoding="ISO-8859-1" ?>
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<!DOCTYPE scenario SYSTEM "sipp.dtd">
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<!-- This program is free software; you can redistribute it and/or -->
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<!-- modify it under the terms of the GNU General Public License as -->
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<!-- published by the Free Software Foundation; either version 2 of the -->
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<!-- License, or (at your option) any later version. -->
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<!-- -->
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<!-- This program is distributed in the hope that it will be useful, -->
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<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
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<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
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<!-- GNU General Public License for more details. -->
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<!-- -->
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<!-- You should have received a copy of the GNU General Public License -->
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<!-- along with this program; if not, write to the -->
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<!-- Free Software Foundation, Inc., -->
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<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
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<!-- -->
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<!-- Sipp default 'uac' scenario. -->
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<!-- -->
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<scenario name="Basic Sipstone UAC">
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<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
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<!-- generated by sipp. To do so, use [call_id] keyword. -->
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<send retrans="500">
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<![CDATA[
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INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
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To: sut <sip:[service]@[remote_ip]:[remote_port]>
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Call-ID: [call_id]
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CSeq: 1 INVITE
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Contact: sip:sipp@[local_ip]:[local_port]
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Max-Forwards: 70
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Subject: Performance Test
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Content-Type: application/sdp
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Content-Length: [len]
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v=0
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o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
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s=-
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c=IN IP[media_ip_type] [media_ip]
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t=0 0
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m=audio [media_port] RTP/AVP 96 97 0 8 101 98
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a=rtpmap:96 speex/16000
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a=fmtp:96 vbr=on
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a=rtpmap:97 speex/8000
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a=fmtp:97 vbr=on
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a=rtpmap:101 telephone-event/16000
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a=rtpmap:98 telephone-event/8000
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m=video [media_port+2] RTP/AVP 96
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a=rtpmap:96 VP8/90000
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m=video [media_port+3] RTP/AVP 96
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a=rtpmap:96 VP8/90000
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m=video [media_port+4] RTP/AVP 96
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a=rtpmap:96 VP8/90000
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]]>
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</send>
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<recv response="100"
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optional="true">
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</recv>
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<recv response="180" optional="true">
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</recv>
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<recv response="183" optional="true">
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</recv>
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<!-- By adding rrs="true" (Record Route Sets), the route sets -->
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<!-- are saved and used for following messages sent. Useful to test -->
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<!-- against stateful SIP proxies/B2BUAs. -->
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<recv response="200" rtd="true">
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</recv>
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<!-- Packet lost can be simulated in any send/recv message by -->
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<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
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<send>
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<![CDATA[
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ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
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To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
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Call-ID: [call_id]
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CSeq: 1 ACK
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Contact: sip:sipp@[local_ip]:[local_port]
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Max-Forwards: 70
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Subject: Performance Test
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Content-Length: 0
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]]>
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</send>
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<!-- This delay can be customized by the -d command-line option -->
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<!-- or by adding a 'milliseconds = "value"' option here. -->
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<pause/>
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<!-- The 'crlf' option inserts a blank line in the statistics report. -->
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<send retrans="500">
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<![CDATA[
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BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
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Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
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From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
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To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
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Call-ID: [call_id]
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CSeq: 2 BYE
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Contact: sip:sipp@[local_ip]:[local_port]
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Max-Forwards: 70
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Subject: Performance Test
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Content-Length: 0
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]]>
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</send>
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<recv response="200" crlf="true">
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</recv>
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<!-- definition of the response time repartition table (unit is ms) -->
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<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
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<!-- definition of the call length repartition table (unit is ms) -->
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<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
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</scenario>
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