[SIPForum-discussion] FW: querry related to SIPp

[Stewart Bunce]

Looking at your 200 OK response I notice that the Record-Route header immediately follows a Via header that does not contain a "branch" tag. As far as I am aware it is mandatory to have a branch tag in a Via header, so perhaps SIPp is getting confused by its absence??. I would also note that the "branch" tag present in the Via header of your ACK request does not begin with the Magic Cookie "z9hG4bK", which is required for SIP 2.0 implementations.
 
Beyond that as far as I can see you are correct in your use of the rrs="true" and [routes] keyword as I have similiar scenarios that work fine for me.
 
Stew

-----Original Message-----
From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org]On Behalf Of dushyant
Sent: 16 June 2008 06:17
To: discussion at sipforum.org
Subject: [SIPForum-discussion] FW: querry related to SIPp



Hi all,

 

Can anybody help me in resolving a issue related to SIPp? In an IMS call scenario I need to preserve the record route received in 200OK and use the same to form Route header of ACK. I have made the following changes in the xml files but instead of using the values received in Record-Route of 200 OK, it's is using some default value. The received message (200OK) and xml portion are attached.

 

UDP message received [481] bytes :

 

SIP/2.0 200 OK^M

From: 9000<sip:9000 at 10.34.77.48:6666>;tag=1^M

To: 23400<sip:23400 at 10.34.77.48:5060>;tag=1^M

Call-ID: 1.9881.10.34.77.48 at sipp.call.id^M

CSeq: 1 INVITE^M

Via: SIP/2.0/UDP 10.34.77.48:6666^M

Record-Route: <sip:mo2 at 10.66.10.13:8055;lr>^M

Contact: <sip:10.34.77.48:5060;transport=UDP>^M

Content-Type: application/sdp^M

Content-Length: 136^M

^M

v=0^M

o=user1 53655765 2353687637 IN IP4 127.0.0.1^M

s=-^M

c=IN IP4       127.0.0.1^M

t=0 0^M

m=audio 10000 RTP/AVP 8^M

a=rtpmap:8 PCMA/8000^M

 

---------------------- XML------------------------------

 

<!-- 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 response="200" rtd="true" rrs="true">

  </recv>

 

  <!-- Packet lost can be simulated in any send/recv message by         -->

  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->

<send>

    <![CDATA[

 

ACK sip:[field1]@[remote_ip]:[remote_port] SIP/2.0

      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=dushyant

      From: [field0]<sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]

      To: [field1]<sip:[field1]@[remote_ip]:[remote_port]>[peer_tag_param]

      Call-ID: [call_id]

      CSeq: 1 ACK

      Contact: sip:[field0]@[local_ip]:[local_port]

      Max-Forwards: 70

      Subject: Performance Test

      Content-Length: 0

      [routes]

 

  ]]>

  </send>

 

Dushyant P S Dhalia

 



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