208 lines
6.7 KiB
XML
208 lines
6.7 KiB
XML
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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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<!-- -->
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<!-- Note:
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For this test to work, PJSUA-LIB needs to add video line, with
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this patch:
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pjsua_media.c:1253, after call to pjmedia_endpt_create_sdp():
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if (1) {
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pjmedia_sdp_media *m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media);
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m->desc.media = pj_str("video");
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m->desc.port = 3000;
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m->desc.transport = pj_str("RTP/AVP");
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m->desc.fmt_count = 1;
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m->desc.fmt[0] = pj_str("0");
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sdp->media[sdp->media_count++] = m;
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}
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-->
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<scenario name="UAC with bad ACK">
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<!-- UAC with bad ACK causes assertion with pjsip 1.4 -->
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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=[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=Tester 234 123 IN IP4 89.208.145.194
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s=Tester
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c=IN IP4 89.208.145.194
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t=0 0
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m=audio 17424 RTP/AVP 111 0 18 101
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a=rtpmap:111 SPEEX/16000
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a=rtpmap:0 PCMU/8000
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a=rtpmap:18 G729/8000
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a=rtpmap:101 telephone-event/8000
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a=sendrecv
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a=rtcp:17425
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m=video 11128 RTP/AVP 34 103 104
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a=rtpmap:34 H263/90000
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a=rtpmap:103 H263-1998/90000
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a=rtpmap:104 H264/90000
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a=sendrecv
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a=rtcp:11129
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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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<!-- 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=[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 milliseconds="2000"/>
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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=[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 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=Tester 234 124 IN IP4 89.208.145.194
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s=Tester
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c=IN IP4 89.208.145.194
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t=0 0
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m=audio 17424 RTP/AVP 111 0 18 101
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a=rtpmap:111 SPEEX/16000
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a=rtpmap:0 PCMU/8000
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a=rtpmap:18 G729/8000
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a=rtpmap:101 telephone-event/8000
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a=sendrecv
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a=rtcp:17425
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m=video 0 RTP/AVP 34 103 104
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a=sendrecv
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]]>
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</send>
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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=[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 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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<pause milliseconds="2000"/>
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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=[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: 3 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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