RE: [Sipp-users] uas question

[Peter H. <pet...@ne...>]

There is a command line option -p which you can use to set a fixed port at
the server.

 

Peter Higginson

Newport Networks Ltd,

Direct line 01494 470694

http://www.newport-networks.com/

  _____  

From: sip...@li...
[mailto:sip...@li...] On Behalf Of Deng Pan
Sent: 08 July 2005 04:56
To: sip...@li...
Subject: [Sipp-users] uas question

 

Hi,

I am using sipp as uac and uas to test a sip proxy server, and I have some
problems with uas.

What I want is to create 1000 uac and 1000 uas, which communicate through my
sip proxy server, to emulate 1000 concurrent calls. Since the local_port of
uas is randomly assigned, my proxy has no way to know its local_port (to
route packets) unless it registers to the proxy. However, if the uas
registers first in the scenario, it will not respond to the INVITE request.

My uas.xml is as follows:

<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="My UAS responder">
  <send retrans="500">
    <![CDATA[
      REGISTER sip:[field1]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port]
      From: [field1]
<sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
      To: [field1] <sip:[field1]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 REGISTER
      Contact: sip:[field1]@[local_ip]:[local_port]
      Max-Forwards: 70
      Expires: 6000
      Content-Length: 0
    ]]>
  </send>
  <recv response="200" rtd="true" optional="true">
  </recv>

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

  <send>
    <![CDATA[
      SIP/2.0 180 Ringing
      [last_Via:]
      [last_From:]
      [last_To:];tag=[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=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Type: application/sdp
      Content-Length: 136
      v=0
      o=user1 53655765 2353687637 IN IP4 127.0.0.1
      s=-
      c=IN IP4 [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000
    ]]>
  </send>
  <recv request="ACK"
        optional="true"
        rtd="true"
        crlf="true">
  </recv>
  <recv request="BYE">
  </recv>
  <send>
    <![CDATA[
      SIP/2.0 200 OK
      [last_Via:]
      [last_From:]
      [last_To:];tag=[call_number]
      [last_Call-ID:]
      [last_CSeq:]
      Contact: <sip:[local_ip]:[local_port];transport=[transport]>
      Content-Length: 0
    ]]>
  </send>
  <!-- Keep the call open for a while in case the 200 is lost to be     -->
  <!-- able to retransmit it if we receive the BYE again.               -->
  <pause milliseconds="4000"/>

  <!-- definition of the response time repartition table (unit is ms)   -->
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
  <!-- definition of the call length repartition table (unit is ms)     -->
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>