Hi Rob,

I am using sipp-3.3 at RHEL 5.3 (32 bit).

I want to test following SIP scenario with SIPP. This is produced by making
a successful SIP call from Xlite softphone client to freeswitch
(Application Hosted).

CLIENT                                      Freeswtich

Invite           ------------>
                   <----------                  100 Trying
                   <----------                   183 with SDP
                  <-- Media Play -->
Bye             ------------>
                   <---------------              200 OK


I am getting following

Invite            ------------>
                   <-----------                  100 Trying
                   <-----------                  183 with SDP
Cancel         ------------>
                   <-----------                  200 OK

Following is my scenario file.


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

<scenario name="UAC with media">
  <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: [field0]
<sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:field0@[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="183">
  </recv>

  <pause milliseconds="10000"/>
  <nop>
    <action>
      <exec play_pcap_audio="pcap/mod_01012013_240to21.pcap"/>
    </action>
  </nop>

  <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: field0 <sip:field0@[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:field0@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0
    ]]>
  </send>

  <timewait milliseconds="1000"/>
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

Attached are Files.

1. Successful SIP Call PCAP        sip_succ.pcap
2. Failed SIPP call                        failed_sipp.pcap
3. Scenario Used in SIPP              crbt1.xml

Best Regards,
Rajesh Bansal


On Mon, Jan 7, 2013 at 8:11 PM, Rob Day <r...@rkd.me.uk> wrote:

> On 7 January 2013 13:31, Rajesh Bansal <bansal.rajes...@gmail.com> wrote:
> > I need urgent help. I want to test ring back tone application in which we
> > are playing media after 183(with SDP). But i am not able to make XML
> (UAC)
> > scenario for the same. Kindly help me in this regard,
>
> Well - what problem are you having? What have you tried so far, what
> are you expecting it to do, and what is actually happening?
>
> (You may find that http://www.catb.org/esr/faqs/smart-questions.html
> helps you to make the best use of mailing lists such as these,
> particularly http://www.catb.org/esr/faqs/smart-questions.html#beprecise).
>
>
> Best,
> Rob
>
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">

<scenario name="UAC with media">
  <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: [field0] <sip:[field0]@[local_ip]:[local_port]>;tag=[call_number]
      To: sut <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:field0@[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="183">
  </recv>

  <pause milliseconds="10000"/>
  <nop>
    <action>
      <exec play_pcap_audio="pcap/mod_01012013_240to21.pcap"/>
    </action>
  </nop>

  <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: field0 <sip:field0@[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:field0@[local_ip]:[local_port]
      Max-Forwards: 70
      Subject: Performance Test
      Content-Length: 0
    ]]>
  </send>

  <timewait milliseconds="1000"/>
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

Attachment: failes_sipp.pcap
Description: Binary data

Attachment: sip_succ.pcap
Description: Binary data

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