> On 7 January 2013 15:09, Rajesh Bansal <bansal.rajes...@gmail.com> wrote:
>> Hi Rob,
>>
>> I am using sipp-3.3 at RHEL 5.3 (32 bit).
>>
>> I am getting following
>>
>> Invite ------------>
>> <----------- 100 Trying
>> <----------- 183 with SDP
>> Cancel ------------>
I tried to reproduce the error by using the attached file to send in a
183 message that matched the one in your PCAP file, and couldn't
reproduce it (i.e. I ran your crbt1.xml and it succeeded), so I think
I do need to see those logs.
Best,
Rob
<?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 separated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<send>
<![CDATA[
SIP/2.0 183 Session Progress
[last_Via:]
[last_From:]
[last_To:];tag=gFQtXFgZQmyUc
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:1001@10.168.1.64:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 245
Remote-Party-ID: "1001" <sip:1001@10.168.1.64>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1357538228 1357538229 IN IP4 [media_ip]
s=FreeSWITCH
c=IN IP4 [media_ip]
t=0 0
m=audio 22588 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
]]>
</send>
<pause />
<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
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
optional="true"
rtd="true"
crlf="true">
</recv>
<recv request="BYE">
</recv>
<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>
<!-- 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"/>
<!-- 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>
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