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
 
 
<?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 sip:[EMAIL PROTECTED]:[remote_port] 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 sip:[EMAIL PROTECTED]:[remote_port] 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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