<?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="Basic UAS responder"> | |
<!-- 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"> | |
</recv> | |
<!-- The '[last_*]' keyword is replaced automatically by the --> | |
<!-- specified header if it was present in the last message received --> | |
<!-- (except if it was a retransmission). If the header was not --> | |
<!-- present or if no message has been received, the '[last_*]' --> | |
<!-- keyword is discarded, and all bytes until the end of the line --> | |
<!-- are also discarded. --> | |
<!-- --> | |
<!-- If the specified header was present several times in the --> | |
<!-- message, all occurences are concatenated (CRLF seperated) --> | |
<!-- to be used in place of the '[last_*]' keyword. --> | |
<send retrans="500"> | |
<![CDATA[ | |
SIP/2.0 422 Session Timer too small | |
[last_Via:] | |
[last_From:] | |
[last_To:];tag=[call_number] | |
[last_Call-ID:] | |
[last_CSeq:] | |
Contact: <sip:[local_ip]:[local_port];transport=[transport]> | |
Min-SE: 5400 | |
Content-Length: 0 | |
]]> | |
</send> | |
<recv request="ACK" | |
optional="true" | |
rtd="true" | |
crlf="true"> | |
</recv> | |
<recv request="INVITE" crlf="true"> | |
</recv> | |
<send retrans="500"> | |
<![CDATA[ | |
SIP/2.0 200 OK | |
[last_Via:] | |
[last_From:] | |
[last_To:];tag=[call_number] | |
[last_Call-ID:] | |
[last_CSeq:] | |
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER | |
Allow-Events: telephone-event | |
Contact: <sip:[local_ip]:[local_port];transport=[transport]> | |
Supported: replaces | |
Session-Expires: 3600;refresher=uas | |
Require: timer | |
Content-Type: application/sdp | |
Content-Disposition: session;handling=required | |
Content-Length: [len] | |
v=0 | |
o=Some-UserAgent 68 210 IN IP4 [local_ip] | |
s=SIP Call | |
c=IN IP4 [local_ip] | |
t=0 0 | |
m=audio 17294 RTP/AVP 0 101 | |
c=IN IP4 [local_ip] | |
a=rtpmap:0 PCMU/8000 | |
a=rtpmap:101 telephone-event/8000 | |
a=fmtp:101 0-16 | |
a=ptime:20 | |
]]> | |
</send> | |
<recv request="ACK" | |
rtd="true" | |
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> | |