pjsip-pjproject/tests/pjsua/scripts-sipp/uas-reinv-with-less-media.xml

156 lines
4.6 KiB
XML

<?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 'uas' scenario. -->
<!-- -->
<scenario name="Sending OK and re-INVITE with less media (#16xx)">
<!-- 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 request="INVITE" crlf="true">
<action>
<ereg regexp=".*" search_in="hdr" header="From" assign_to="3"/>
<ereg regexp="sip:(.*)>" search_in="hdr" header="Contact" assign_to="4,5"/>
<assign assign_to="4" variable="5" />
</action>
</recv>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: sip:sipp@[local_ip]:[local_port]
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 3441953879 3441953879 IN IP4 192.168.0.15
s=pjmedia
c=IN IP4 192.168.0.15
t=0 0
m=audio 4000 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=sendrecv
]]>
</send>
<recv request="ACK" crlf="true">
</recv>
<pause milliseconds="2000"/>
<send retrans="500">
<![CDATA[
INVITE sip:[$5] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-same-branch
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To[$3]
Call-ID: [call_id]
Cseq: 2 INVITE
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 3441953879 3441953879 IN IP4 192.168.0.15
s=pjmedia
c=IN IP4 192.168.0.15
t=0 0
m=audio 4000 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=sendonly
]]>
</send>
<recv response="200" rtd="true">
</recv>
<send>
<![CDATA[
ACK sip:[$5] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=z9hG4bK-same-branch
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To[$3]
Call-ID: [call_id]
Cseq: 2 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Content-Length: 0
]]>
</send>
<recv request="INVITE" crlf="true">
</recv>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: sip:sipp@[local_ip]:[local_port]
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 3441953879 3441953879 IN IP4 192.168.0.15
s=pjmedia
c=IN IP4 192.168.0.15
t=0 0
m=audio 4000 RTP/AVP 0 96
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=inactive
]]>
</send>
<recv request="ACK" crlf="true">
</recv>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<pause milliseconds="4000"/>
<!-- 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>