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Added new scenario file: UAS which answers with more than one
codec and starts sending RTP *not* with the first one
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<?xml version="1.0" encoding="ISO-8859-1" ?> | ||
<!DOCTYPE scenario SYSTEM "sipp.dtd"> | ||
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<!-- 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. --> | ||
<!-- --> | ||
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<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> | ||
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<!-- 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. --> | ||
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<send> | ||
<![CDATA[ | ||
SIP/2.0 180 Ringing | ||
[last_Via:] | ||
[last_From:] | ||
[last_To:];tag=[pid]SIPpTag01[call_number] | ||
[last_Call-ID:] | ||
[last_CSeq:] | ||
Contact: <sip:[local_ip]:[local_port];transport=[transport]> | ||
Content-Length: 0 | ||
]]> | ||
</send> | ||
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<send retrans="500"> | ||
<![CDATA[ | ||
SIP/2.0 200 OK | ||
[last_Via:] | ||
[last_From:] | ||
[last_To:];tag=[pid]SIPpTag01[call_number] | ||
[last_Call-ID:] | ||
[last_CSeq:] | ||
Contact: <sip:[local_ip]:[local_port];transport=[transport]> | ||
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 0 8 | ||
a=rtpmap:0 PCMU/8000 | ||
a=rtpmap:8 PCMA/8000 | ||
]]> | ||
</send> | ||
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<recv request="ACK" | ||
rtd="true" | ||
crlf="true"> | ||
</recv> | ||
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<!-- Play a pre-recorded PCAP file (RTP stream) --> | ||
<nop> | ||
<action> | ||
<exec play_pcap_audio="g711a.pcap"/> | ||
</action> | ||
</nop> | ||
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<!-- Pause 90 seconds, which is approximately the duration of the --> | ||
<!-- PCAP file --> | ||
<pause milliseconds="90000"/> | ||
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<recv request="BYE"> | ||
</recv> | ||
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<send> | ||
<![CDATA[ | ||
SIP/2.0 200 OK | ||
[last_Via:] | ||
[last_From:] | ||
[last_To:] | ||
[last_Call-ID:] | ||
[last_CSeq:] | ||
Contact: <sip:[local_ip]:[local_port];transport=[transport]> | ||
Content-Length: 0 | ||
]]> | ||
</send> | ||
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<!-- 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. --> | ||
<timewait milliseconds="4000"/> | ||
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<!-- definition of the response time repartition table (unit is ms) --> | ||
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/> | ||
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<!-- definition of the call length repartition table (unit is ms) --> | ||
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/> | ||
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</scenario> | ||
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