Hi

I have made changes in the script , so u can use the script and it may run
this time.

On Tue, May 27, 2008 at 5:56 PM, <[EMAIL PROTECTED]> wrote:

>  Hi All,
>
> I am trying to simulate simple call with SIPp. I had created a script which
> can send Invite to the softswitch, and i can able to receive 1xx and 200 Ok
> from the farend,
> but once i receive 200OK my script is not able to send ACK ..I am attaching
> the call secnario script. If anybody tells what's wrong(not able to send
> ACK) in my script??
> please help me.
>
> Regards
> Srinivas
>
>
>
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<?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="Basic Sipstone 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];user=phone SIP/2.0
From: <sip:[EMAIL PROTECTED];user=phone>;tag=8dde3b2c31;epid=B8DDA65423
To: <sip:[EMAIL PROTECTED];user=phone>
Call-ID: a96c6a5a-c5d2-4d93-a56d-4454d7e7f063
CSeq: 1 INVITE
MAX-FORWARDS: 70
Supported: 100rel
USER-AGENT: RTCC/3.0.0.0 MediationServer
Via: SIP/2.0/TCP 192.168.48.199:3899;branch=z9hG4bKfed0a31a
Contact: 
<sip:192.168.48.199:5060;maddr=192.168.48.199;transport=Tcp;ms-opaque=4e1752980aafe5b7>
Allow: UPDATE,ACK,CANCEL,BYE,INVITE
Content-Type: application/sdp;charset=utf-8
Content-Length: [len]

v=0 
o=- 99 1 IN IP4 192.168.48.199
s=- 
t=0 0 
m=audio 5404 RTP/AVP 0 8 101 111 
c=IN IP4 192.168.48.199
a=rtpmap:101 telephone-event/8000 
a=fmtp:101 0-15 
a=rtpmap:111 X-nt-inforeq/8000 
a=ptime:20 
a=maxptime:20 
a=sendrecv

    ]]>
  </send>

  <recv response="100"
        optional="true">
  </recv>

  <recv response="180" optional="true">
  </recv>

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

    ]]>
  </send>

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


  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
  <send retrans="500">
    <![CDATA[

      BYE [next_url] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
      To: sut <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 4 BYE
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0

    ]]>
  </send>

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