On Sat, May 14, 2011 at 12:28 AM, rad bogdan <bogdan_rad2...@yahoo.com>wrote:
> Hi everyone,
>
> I have a server running OpenSIPS and MediaProxy and I want to test several
> scenarios using SIPp.
> In the first phase I want to emulate a simple dialog between a caller and a
> callee with -rtp_echo.
> I was able to create a scenario for the caller which works as expected when
> it calls an SIP client (Linphone),
> but when it contacts the sipp instance running the callee script, the
> callee doesn't behave as I want:
>
> - although in callee's logs I can see the INVITE messages received from the
> server, the <recv request="INVITE" rrs="true" crlf="true" /> instruction
> doesn't seem to catch them.
>
Hi,
this is a known issue (but it is not a bug or problem with sipp).
SIPp identifies test calls based on the Call-ID that is sent (or comes) in
the first SIP request.
When the callee scenario (which is actually a UAC scenario) sends the
REGISTER request, it is establishes the Call-ID to be used for the rest of
the scenario. So when the INVITE arrives witn a different Call-ID (generated
by the caller), SIPp discards it and that call fails.
Read more about it here:
http://lists.sip-router.org/pipermail/sr-users/2011-January/066962.html
And a possible solution would be to use IMS Bench (but I mylself never used
it):
http://www.mail-archive.com/sipp-users@lists.sourceforge.net/msg03336.html
http://sipp.sourceforge.net/ims_bench/reference.html
regards,
takeshi
> - the caller doesn't get the 180 Ringing response.
>
> Attached are the two files that I tested with.
> OpenSIPS and SIPp run on the same machine.
>
> Thanks,
> Bogdan
>
>
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