[SIPForum-discussion] SIP Load testing tool

[Deb, Nihar Ranjan]

Hi All,
   I am using following xml codes to send DTMF to my SIP application and I am using the command line command "C:\Program Files\Sipp_3.1>sipp 10.8.2.39 -sf test.xml -i 10.8.2.144 -mi 10.8.2.14
4 -s 3990 -l 1" to start SIPp. From the ethereal trace I can see DTMF is sent to the destination IP address but my SIP application is not detecting the DTMF.Is there anything I am missing please suggest.Normally if I am calling from any phone dtmf is working fine
I am running SIPp in  windows platform and using RFC2833 DTMF mode.
 
 
<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="UAC with media">
<!--
 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:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Type: application/sdp
      Content-Length: [len]
 
      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[local_ip_type] [local_ip]
      t=0 0
      m=audio [auto_media_port] RTP/AVP 8
      a=rtpmap:8 PCMA/8000
      a=rtpmap:101 telephone-event/8000
      a=fmtp:101 0-11,16
 
 ]]>
</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" crlf="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:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Contact: sip:sipp@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0
 
]]> 
</send>
<pause milliseconds="8000"/>
<!--
 Play an out of band DTMF '1'                                     
-->
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_1.pcap"/>
</action>
</nop>
<nop>
<action>
<exec play_pcap_audio="pcap/dtmf_2833_pound.pcap"/>
</action>
</nop>
<pause milliseconds="9000"/>
<!--
 The 'crlf' option inserts a blank line in the statistics report. 
-->
<send retrans="500">
 
<![CDATA[
      BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 2 BYE
      Contact: sip:sipp@[local_ip]:[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>
 
 
 
Regards,
  Nihar Deb

-----Original Message-----
From: Allen, Toby Edward Gedis (Toby) [mailto:allen at avaya.com]
Sent: Wednesday, June 17, 2009 6:24 AM
To: Deb, Nihar Ranjan
Subject: RE: [SIPForum-discussion] SIP Load testing tool



Use SIPp

 

Toby Allen | AVTS - Software Engineer | Avaya | +612 9352 8699| allen at avaya.com
[ Please consider the environment before printing this email ]


  _____  


From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Deb, Nihar Ranjan
Sent: Wednesday, 17 June 2009 1:57 AM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] SIP Load testing tool

 

 

Hi All, 

   I need to traffic test a SIP based application. For that I need to configure the testing tool so that 

1. It can send INVITE packets at a regular interval 

2. It can send DTMF at a regular interval to the application as I will configure 

3. It will send BYE after completion of one cycle 

4. It should be able to run this kinds of more than 80 - 90 cycles same time. 

   Can anyone please suggest me any SIP load testing tool like this. 

Thanks in advance. 

 

Regards, 
   Nihar Deb 

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