Hi all!

I have a fairly simple scenario which I use for automatic testing.

  ----------> INVITE
  <---------- 180
  <---------- 200
  ----------> ACK

  ----------> INVITE
  <---------- 200
  ----------> ACK

  ----------> BYE
  ----------> ACK
  <---------- 200
  [   4000ms] Pause

The sipp is run i.e. "sipp -rsa 127.0.0.1:5070 127.0.0.1 -m 1 -sf 
my_sipp_scenario.xml" for each separate test and sipp's return value is 
checked for zero.

The problem is, that sometimes (probably due UDP protocol?) sipp 
receives the second INVITE before ACK:

  ----------> INVITE
  <---------- 180
  <---------- 200

  ----------> INVITE    (WILL COMPLAIN AS UNEXPECTED SIP MESSAGE ERROR!)
  ----------> ACK

...


and marks the call as failed, but I think that this is still a valid 
call (since both INVITE and ACK are send). The question is, how can I 
solve this issue, so both sip flows will be valid?

The scenario (my_sipp_scenario.xml):

<scenario name="Test">
   <recv request="INVITE" crlf="true">
   </recv>

   <send>
     <![CDATA[

       SIP/2.0 180 Ringing
       [last_Via:]
       [last_From:]
       [last_To:];tag=[pid]SIPpTag01[call_number]
       [last_Call-ID:]
       [last_CSeq:]
       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
       Content-Length: 0

     ]]>
   </send>

   <send retrans="500">
     <![CDATA[

       SIP/2.0 200 OK
       [last_Via:]
       [last_From:]
       [last_To:];tag=[pid]SIPpTag01[call_number]
       [last_Call-ID:]
       [last_CSeq:]
       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
       Content-Type: application/sdp
       Content-Length: [len]

       v=0
       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
       s=-
       c=IN IP[media_ip_type] [media_ip]
       t=0 0
       m=audio [media_port] RTP/AVP 0
       a=rtpmap:0 PCMU/8000

     ]]>
   </send>

   <recv request="ACK" optional="true">
   </recv>

   <recv request="INVITE" crlf="true" />

   <send retrans="500">
     <![CDATA[

       SIP/2.0 200 OK
       [last_Via:]
       [last_From:]
       [last_To:];tag=[pid]SIPpTag01[call_number]
       [last_Call-ID:]
       [last_CSeq:]
       Contact: <sip:[local_ip]:[local_port];transport=[transport]>
       Content-Type: application/sdp
       Content-Length: [len]

       v=0
       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
       s=-
       c=IN IP[media_ip_type] [media_ip]
       t=0 0
       m=audio [media_port] RTP/AVP 0
       a=rtpmap:0 PCMU/8000

     ]]>
   </send>

....

Tnx in advance,
   Boris


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