[SIPForum-discussion] FW: querry related to SIPp

[Raj Jain]

What you've done looks correct. What are your sipp command line
arguments like? Below is a snippet of my UAC XML file which works fine
for me with respect to record-routing.

  <recv response="200" rtd="true" rrs="true">
  </recv>

  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: "[service]"<sip:[service]@[local_ip]:[local_port]>;tag=[call_number]
      To: "[service]"<sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: <sip:[service]@[local_ip]:[local_port]>
      [routes]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>


--
Raj Jain


On Mon, Jun 16, 2008 at 1:16 AM, dushyant
<dushyant.dhalia at rancoretech.com> wrote:
> 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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