When I'm trying make a call the "INVITE" message arrives to the
server, it sends "180 Ringigng" and "200 OK" but the server couldn't
get the ACK message from the client.
How can I put the routing information to the ACK message?

I enclose the uac.xml

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

<!-- This program is free software; you can redistribute it and/or      -->
<!-- modify it under the terms of the GNU General Public License as     -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version.                    -->
<!--                                                                    -->
<!-- This program is distributed in the hope that it will be useful,    -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
<!-- GNU General Public License for more details.                       -->
<!--                                                                    -->
<!-- You should have received a copy of the GNU General Public License  -->
<!-- along with this program; if not, write to the                      -->
<!-- Free Software Foundation, Inc.,                                    -->
<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
<!--                                                                    -->
<!--                 Sipp default 'uac' scenario.                       -->
<!--                                                                    -->

<scenario name="UAC">
  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
  <!-- generated by sipp. To do so, use [call_id] keyword.                -->
  
<send retrans="500">
    <![CDATA[

      INVITE sip:[EMAIL PROTECTED] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[EMAIL PROTECTED]>;tag=[call_number]
      To: <sip:[EMAIL PROTECTED]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      Max-Forwards: 70
      Expires: 1800
      User-Agent: SIPp/Linux
      Subject: Performance Test
      Content-Type: application/sdp
      Content-Length: 0

      
    ]]>
 </send>
 <recv response="100"
        optional="true">
  </recv>
<recv response="407"
        auth="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:[EMAIL PROTECTED] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[EMAIL PROTECTED]>;tag=[call_number]
      To: <sip:[EMAIL PROTECTED]>
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      Max-Forwards: 70
      Expires: 1800
      User-Agent: SIPp/Linux
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

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

      INVITE sip:[EMAIL PROTECTED] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[EMAIL PROTECTED]>;tag=[call_number]
      To: <sip:[EMAIL PROTECTED]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      [field2]
      Max-Forwards: 70
      Expires: 1800
      User-Agent: SIPp/Linux
      Subject: Performance Test
      Content-Type: application/sdp
      Content-Length: 0

      
    ]]>
 </send>


  <!-- 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="100"
    optional="true">
</recv>    
    
<recv response="180"
    optional="true">
 </recv>	
 
<recv response="200"
	rtd="true">
</recv> 


  <!-- This delay can be customized by the -d command-line option       -->
  <!-- or by adding a 'milliseconds = "value"' option here.             -->
  

  <!-- The 'crlf' option inserts a blank line in the statistics report. -->

<send>
    <![CDATA[

      
      ACK sip:[EMAIL PROTECTED] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[EMAIL PROTECTED]>;tag=[call_number]
      To: <sip:[EMAIL PROTECTED]>
      Call-ID: [call_id]
      CSeq: 2 ACK
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      Max-Forwards: 70
      Expires: 1800
      User-Agent: SIPp/Linux
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

<pause milliseconds="10"/>

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

      
      BYE sip:[EMAIL PROTECTED] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: <sip:[EMAIL PROTECTED]>;tag=[call_number]
      To: <sip:[EMAIL PROTECTED]>
      Call-ID: [call_id]
      CSeq: 1 BYE
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      [field2]
      Max-Forwards: 70
      Expires: 1800
      User-Agent: SIPp/Linux
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

  <recv response="200">
  </recv>

  <!-- 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>

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