Hi Giuseppe!
As a solution/workaround for this situation I would suggest you after
INVITE request received by UAS, to assign first Via header to variable 1
and second Via header to variable 2. While sending responses from UAS
put the first Via with value of variable 1 and second Via will have
value of variable 2.
As far as I understand it should look like:
<recv request="INVITE">
<action>
<ereg regexp=".*" search_in="hdr" header="Via:" check_it="true"
assign_to="1,2" />
</action>
</recv>
At sending response procedure simply put the following lines:
Via: [$1]
Via: [$2]
I hope it will help.
Regards,
Evgeny Miloslavsky
Systest Engineer
Juniper Networks Solutions Israel LTD.
Office: 972-9-9712355 / 7320
-----Original Message-----
From: Giuseppe Roberti [mailto:[email protected]]
Sent: Wednesday, December 17, 2008 6:04 PM
To: [email protected]
Subject: [Sipp-users] Problem about 2nd Via
Hi,
i am testing a proxy using sipp but i have problem with Via headers.
10.0.0.1 is the proxy.
10.0.0.2 is the sipp server (-sn uas)
10.0.0.3 is the sipp client (-sn uac)
I have noticed that the 2nd via added by sipp uas is threat incorrectly
by the proxy but i don't know if it is my fault.
Here the sip flow.
> 1) The sipp uac send to the proxy the INVITE:
> INVITE sip:[email protected]:5060 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK
> From: sipp <sip:[email protected]:5060>;tag=23603SIPpTag091
> To: sut <sip:[email protected]:5060>
> Call-ID: [email protected]
> CSeq: 1 INVITE
> Contact: sip:[email protected]:5060
> Max-Forwards: 70
> Subject: Performance Test
> Content-Type: application/sdp
> Content-Length: 186
>
> v=0
> o=user1 53655765 2353687637 IN IP4 10.0.0.3
> s=-
> c=IN IP4 10.0.0.3
> t=0 0
> m=audio 6000 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11,16
>
>
> 2) The proxy send the INVITE to sipp uas, adding the second Via header
> INVITE sip:[email protected]:5060 SIP/2.0
> Record-Route:
<sip:10.0.0.1;lr=on;ftag=23603SIPpTag091;did=c04.37fe035>
> Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK21dd.7d5758a6.0;rport
> Via: SIP/2.0/UDP
10.0.0.3:5060;rport=5060;received=10.0.0.3;branch=z9hG4bK
> From: sipp <sip:[email protected]:5060>;tag=23603SIPpTag091
> To: sut <sip:[email protected]:5060>
> Call-ID: [email protected]
> CSeq: 1 INVITE
> Contact: sip:[email protected]:5060
> Max-Forwards: 69
> Subject: Performance Test
> Content-Type: application/sdp
> Content-Length: 187
>
> v=0
> o=user1 53655765 2353687637 IN IP4 10.0.0.3
> s=-
> c=IN IP4 10.0.0.1
> t=0 0
> m=audio 50110 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11,16
>
>
> 3) The sipp uas does not properly recognize (maybe) the 2nd Via and
send back this (please note the line after the first Via)
> SIP/2.0 180 Ringing
> Record-Route:
<sip:10.0.0.1;lr=on;ftag=23603SIPpTag091;did=c04.37fe035>
> Via: SIP/2.0/UDP 10.0.0.1;branch=z9hG4bK21dd.7d5758a6.0;rport,
SIP/2.0/UDP 10.0.0.3:5060;rport=5060;received=10.0.0.3;branch=z9hG4bK
> From: sipp <sip:[email protected]:5060>;tag=23603SIPpTag091
> To: sut <sip:[email protected]:5060>;tag=15130SIPpTag011
> Call-ID: [email protected]
> CSeq: 1 INVITE
> Contact: <sip:10.0.0.2:5060;transport=UDP>
> Content-Length: 0
Is it my fault ?
I'm using sipp from svn.
--
Giuseppe Roberti
<[email protected]>
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