Thanks for the response Rob. I have attached the xml file to the e-mail,
listed the version, and command to start SIPp below. Please let me know if
you need anything else or if you have any suggestions. Thanks again!
./sipp -sf ctx_onestringdial_refresh.xml 10.88.87.216 -t tn -m 1
-trace_logs -trace_msg -rtp_echo -trace_screen -trace_err
SIPp v3.3-PCAP, built Mar 7 2013, 11:51:40.
On Sat, Mar 23, 2013 at 5:49 AM, Rob Day <r...@rkd.me.uk> wrote:
> On 23 March 2013 04:21, Matt Chung <itsmemattch...@gmail.com> wrote:
> > My main question is SIPp hard coded to exit
> > after 2 minutes of not receiving a request, regardless of pause/timeout
> > being set to greater than 2 minutes (i.e pause milliseconds="840000" or
> > timeout="840000") ?
> >
> > Essentially, I'm waiting for the session to be refreshed by the UAS at
> the
> > 15 minute mark (half of session-expires: 1800); call setup is
> successfully
> > however when SIPp reaches the following element), SIPp will exit after 2
> > minutes:
>
> This surprises me, and doesn't match what I've seen in my use of SIPp.
> I certainly don't think there should be a hard-coded limit of 2
> minutes.
>
> I'd like to see, if possible, the command line you're using to start
> SIPp, the XML file, and the version of SIPp (i.e. the first line of
> `sipp -v`). That should let me try and reproduce it.
>
> --
> Robert K. Day
> robert....@merton.oxon.org
>
--
-Matt Chung
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="MeetMe UAC">
<!-- Meeting ID to TesT: 15335485 02023601 - SIPp Testing Case Meeting ID: 64532595-->
<send retrans="50000">
<![CDATA[
INVITE sip:300301302**76396205@10.88.87.216:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [call_number] <sip:[call_number]@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:300301302**76396205@10.88.87.216:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:[call_number]@[local_ip]:[local_port]
Allow: INVITE,OPTIONS,INFO,BYE,CANCEL,ACK,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Allow-Events: presence,kpml
Supported: timer
Session-Expires: 1800
Max-Forwards: 70
User-Agent: SIPp
Content-Length: 0
]]>
</send>
<recv response="100">
</recv>
<recv response="180" optional="true" timeout="3000" ontimeout="1">
</recv>
<label id="1" />
<recv response="200" rtd="true" crlf="true">
</recv>
<send>
<![CDATA[
ACK sip:[service]@[remote_ip];transport=tcp SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Call-ID: [call_id]
From: [call_number] <sip:[call_number]@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:300301302**76396205@10.88.87.216:[remote_port]>[peer_tag_param]
CSeq: 1 ACK
Contact: <sip:[call_number]@[local_ip]:[local_port];transport=tcp>
Content-Type: application/sdp
Max-Forwards: 68
Allow-Events: presence,kpml
Content-Length: [len]
v=0
o=CiscoSystemsCCM-SIP 122391215824152 122391215824152 IN IP4 [local_ip]
s=-
c=IN IP4 [local_ip]
t=0 0
m=audio [media_port] RTP/AVP 96 101
b=TIAS:64000
a=rtpmap:96 mpeg4-generic/48000
a=rtpmap:101 telephone-event/8000
a=fmtp:96 profile-level-id=16;streamtype=5;config=B98C00;mode=AAC-hbr;sizeLength=13;indexLength=3;indexDeltaLength=3;constantDuration=480
a=fmtp:101 0-15
m=video [media_port+2] RTP/AVP 97
b=TIAS:4000000
a=rtpmap:97 H264/90000
a=label:95
a=fmtp:97 profile-level-id=42001F;packetization-mode=0;max-mbps=108000;max-fs=3600;max-cpb=90;max-br=2250;max-rcmd-nalu-size=3200;max-fps=3000
a=content:main
a=rtcp-fb:* nack pli
m=application 0 UDP/BFCP *
m=application 0 RTP/AVP 96
a=rtpmap:96 H224/0
m=video 0 RTP/AVP 31 34 96 97
a=rtpmap:96 H263-1998/90000
a=rtpmap:97 H264/90000
a=content:slides
]]>
</send>
<recv request="INVITE|UPDATE" regexp_match="true" ontimeout="2">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
Call-ID: [call_id]
[last_From:]
[last_To:];
[last_CSeq:]
Contact: <sip:nexthop@[local_ip]:[local_port];transport=tcp>
Content-Type: application/sdp
Max-Forwards: 68
Allow-Events: presence,kpml
Content-Length: [len]
v=0
o=CiscoSystemsCCM-SIP 122391215824152 122391215824152 IN IP4 [local_ip]
s=-
c=IN IP4 [local_ip]
t=0 0
m=audio [media_port] RTP/AVP 96 101
b=TIAS:64000
a=rtpmap:96 mpeg4-generic/48000
a=rtpmap:101 telephone-event/8000
a=fmtp:96 profile-level-id=16;streamtype=5;config=B98C00;mode=AAC-hbr;sizeLength=13;indexLength=3;indexDeltaLength=3;constantDuration=480
a=fmtp:101 0-15
m=video [media_port+2] RTP/AVP 97
b=TIAS:4000000
a=rtpmap:97 H264/90000
a=label:95
a=fmtp:97 profile-level-id=42001F;packetization-mode=0;max-mbps=108000;max-fs=3600;max-cpb=90;max-br=2250;max-rcmd-nalu-size=3200;max-fps=3000
a=content:main
a=rtcp-fb:* nack pli
m=application 0 UDP/BFCP *
m=application 0 RTP/AVP 96
a=rtpmap:96 H224/0
m=video 0 RTP/AVP 31 34 96 97
a=rtpmap:96 H263-1998/90000
a=rtpmap:97 H264/90000
a=content:slides
]]>
</send>
<label id="2" />
<pause milliseconds="840000" />
<recv request="INVITE|UPDATE" timeout="120000" ontimeout="4" regexp_match="true">
</recv>
<send next="2">
<![CDATA[
SIP/2.0 200 OK
[last_From:]
[last_To:]
Call-ID: [call_id]
[last_CSeq:]
[last_Contact:]
[last_Record-Route:]
Allow: INVITE,OPTIONS,INFO,BYE,CANCEL,ACK,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Allow-Events: presence,kpml
Supported: timer
[last_Session-Expires:]
User-Agent: SIPp
Content-Length: 0
]]>
</send>
<label id="4" />
<send>
<![CDATA[
BYE sip:300301302**76396205@10.88.87.216:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: [call_number] <sip:[call_number]@[local_ip]:[local_port]>;tag=[call_number]
To: <sip:300301302**76396205@10.88.87.216:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: [cseq+1] BYE
Contact: sip:[call_number]@[local_ip]:[local_port]
Content-Length: 0
]]>
</send>
</scenario>
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