Re: [Sipp-users] UAS Scenario - Calling SIPp from my VOIP phone.

Devdutt Mon, 30 Oct 2006 04:54:23 -0800

Title: Message

Hi Jeff, Makarand,

The scenario you provided works well for me. Thanks !

As the next step, I need to generate ( send back from SIPp, running as UAS ) the various possible responses such as 486 Busy Here, 301 Moved Permanently, 302 Moved Temporarily etc. How do generate these messages ? Is there some place where I can get information as to what should go into the corresponding header fields for these messages or is the RFC my best bet ?

I tried 486 Busy by replacing the 180 Ringing message in the default UAS scenario. That worked.

Thanks,

Devdutt.

From: Jeff Ollier [mailto:[EMAIL PROTECTED]
Sent: Friday, October 27, 2006 8:02 PM
To: [EMAIL PROTECTED]
Subject: RE: [Sipp-users] UAS Scenario - Calling SIPp from my VOIP phone.

I have the same setup. This scenario works for me.

<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="register_answer">
<send>
<![CDATA[
       REGISTER sip:[EMAIL PROTECTED] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: 4501 <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
      To: 4501 <sip:[EMAIL PROTECTED]:[local_port]>
      Call-ID: [call_id]
      CSeq: [cseq] REGISTER
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      Content-Length: 0
      Expires: 3600
    ]]>
</send>

<recv response="200">

</recv>

</scenario>

Invite

<?xml version="1.0" encoding="ISO-8859-1" ?>
<scenario name="register_answer">
<label id="5" />
<send>
<![CDATA[

      INVITE sip:[EMAIL PROTECTED] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      From: sip:[EMAIL PROTECTED]:[local_port];tag=[call_number]
      To: sip:[EMAIL PROTECTED]:[remote_port]
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      Max-Forwards: 70
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=test
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000

]]>
</send>

<recv response="200">
</recv>
<send>
<![CDATA[

      ACK sip:[EMAIL PROTECTED] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip];branch=[branch]
      From: "4501" <sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
      To: <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      Cseq: 1 ACK
      Contact: sip:[EMAIL PROTECTED]:[local_port]
      Max-Forwards: 70
      Content-Length: 0

]]>
</send>
<pause milliseconds="8000"/>
<send>
<![CDATA[

BYE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "4501"<sip:[EMAIL PROTECTED]:[local_port]>;tag=[call_number]
To: <sip:[EMAIL PROTECTED]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:[EMAIL PROTECTED]:[local_port]
Max-Forwards: 70
Content-Length: 0

]]>
</send>

</scenario>

Jeff Ollier│Network Quality Control Engineer

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-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of makarand bhagwat
Sent: Friday, October 27, 2006 8:41 AM
To: Devdutt
Cc: [email protected]
Subject: Re: [Sipp-users] UAS Scenario - Calling SIPp from my VOIP phone.

Yes Dedutt,

The scenario that you described should work.

Regards,
Makarand

On 10/26/06, Devdutt < [EMAIL PROTECTED]> wrote:

Hi All,

I am testing a VoIP phone and want to test scenarios where SIPp will send various responses back to the phone in response to an INVITE.

Now, my phone requires that I must REGISTER all endpoints that it can call with a proxy ( SER in my case ) using a numeric extension. This means that if I am to call SIPp ( running in UAS mode ), I will need to REGISTER an extension from SIPp with the proxy.

As I understand, I will need to :

1 > Write a scenario that will send a REGISTER from SIPp, with a numeric extension.

2 > Step 1 will enable me to call SIPp ( send an INVITE )

3 > Eventually I can create further scenarios with SIPp sending back various responses to the INVITE ( based on what responses I want to test for ).

Just needed a confirmation as I am a first time user of SIPp and am new to SIP as well. Please point out any important steps that I may have missed out.

Thanks,

Devdutt.

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Using Tomcat but need to do more? Need to support web services, security?
Get stuff done quickly with pre-integrated technology to make your job easier
Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo
http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642

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Re: [Sipp-users] UAS Scenario - Calling SIPp from my VOIP phone.

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