Title: Message

Jeff,

 

You are not taking all of my advice. For example you have the same tag in the invite and you have the same user names. Also you are not actually setting up to receive a 180/183 or a 200 for the INVITE.

 

My guess is that your CA is rejecting the tag, but it could be rejecting the Call-ID, the media addresses or not know the destination – what is the return message or the error message? (What is the CA?)

 

Peter

 

Peter Higginson

Newport Networks Ltd,

Direct line 01494 470694

http://www.newport-networks.com/


From: Jeff Ollier [mailto:[EMAIL PROTECTED]
Sent: 14 October 2006 02:15
To: Peter Higginson
Subject: RE: [Sipp-users] Registration and calls from multiple users

 

I can register any number of phones now.  Thanks for the pid advice.  That made the difference.  However, I can't invite.  This is what I have, but the SIP Server refuses it:

 

<?xml version="1.0" encoding="ISO-8859-1" ?>
 <scenario name="register_answer">
 <send>
 <![CDATA[
        REGISTER sip:10.0.60.1 SIP/2.0
        Via: SIP/2.0/UDP 10.0.60.100:5060;branch=[branch]
        From: <sip:[EMAIL PROTECTED]:5060>;tag=[pid]tkhyexy[call_number]
        To: <sip:[EMAIL PROTECTED]:5060>
        Call-ID: [call_id]
        CSeq: [cseq] REGISTER
        Contact: <sip:[EMAIL PROTECTED]:5060>
        Content-Length: 0
        Expires: 3600
    ]]>
 </send>
 
 <nop><action><log message="sent REGISTER."/></action></nop>

 

<recv response="200" >
 </recv> 
<![CDATA[
INVITE sip:10.0.60.1 SIP/2.0
       Via: SIP/2.0/[transport] [local_ip]:[local_port]
        From: <sip:[EMAIL PROTECTED]:5060>;tag=[pid]tkhyexy[call_number]
       To: <sip:[EMAIL PROTECTED]:5060>
       Call-ID: [call_id]
       Cseq: [cseq] INVITE
       Contact: <sip:[EMAIL PROTECTED]:5060>
       Max-Forwards: 70
       Content-Type: application/sdp
       Content-Length: [len]
 
       v=0
       o=user1 53655765 2353687637 IN IP4 127.0.0.1
       s=-
       t=0 0
       c=IN IP4 [media_ip]
       m=audio [media_port] RTP/AVP 0
       a=rtpmap:0 PCMU/8000
 
     ]]>
   </send>
 
   <recv response="100"
         optional="true">
   </recv>

 

 <nop><action><log message="CALL OVER."/></action></nop>
 
 </scenario>

 

 

Jeff OllierNetwork Quality Control Engineer

 

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-----Original Message-----
From: Peter Higginson [mailto:[EMAIL PROTECTED]
Sent: Friday, October 13, 2006 2:42 PM
To: Jeff Ollier; [email protected]
Subject: RE: [Sipp-users] Registration and calls from multiple users

You will need a number of SIPP machines (or at least processes – I use Linux myself on which several processes are easy to do).

 

It depends quite what your other equipment is. Assuming it is some form of CA then what I would do for 100 calls is have three processes, and have “-m 50” in the calling parameters of (a) and (c):

 

a)       do 50 registers to register some users to receive the calls (first, before (b) and (c))

b)       start a server to receive the calls (a) registered for

c)       start a client to register and make a call for each of 50 users

 

If you are testing some form of proxy then you can write the server to do the registration and just use (b) and (c) but it does not sound like you are doing that.

 

In your register, you need  to vary the ua name in the To: From: and Contact:. I would use 410x[call_number] or similar where x is different between (a) and (c).

 

Also “tag=[call_number]” is a bit simplistic. Make sure that your invite and the response have different tags from each other and the registers. “tag=[pid]tkhyexy[call_number]” would be more realistic where you vary the bit in the middle whenever the protocols allows. (The [pid] means if you run the test again you get a different value.) If it is a good RFC 3261 implementation the tag is as important as the Call-ID.

 

Peter

 

Peter Higginson

Newport Networks Ltd,

Direct line 01494 470694

http://www.newport-networks.com/


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeff Ollier
Sent: 13 October 2006 15:47
To: [email protected]
Subject: [Sipp-users] Registration and calls from multiple users

 

Here is the current problem:

 

I have multiple UAs (in this scenario: 2) that need to register, then invite, then talk, then tear down.  My problem is that the SIP server is seeing these UAs as the same call.  The call_id will not increment on the second register, causing the system to believe it is the same UA registering.  Here is my scenario.  Is it possible to have multiple SIP stations (100+)register, then invite, then converse, then bye or will I need to run the SIPp program on multiple machines to accomplish this feat?

 

My command line:  sipp 10.0.60.1 -sf register.xml -cid_str [EMAIL PROTECTED] -m 1 -base_cseq 0

 

I am using Windows version 1.1r6

 

Scenario:

<?xml version="1.0" encoding="ISO-8859-1" ?>
 <scenario name="register_answer">
 <send>
  <![CDATA[
    REGISTER sip:10.0.60.1 SIP/2.0
        Via: SIP/2.0/UDP 10.0.60.100:5060;branch=[branch]
        From: <sip:[EMAIL PROTECTED]:5060>;tag=[call_number]
        To: <sip:[EMAIL PROTECTED]:5060>
        Call-ID: [call_id]
        CSeq: [cseq+1] REGISTER
        Contact: <sip:[EMAIL PROTECTED]:5060>
        Content-Length: 0
        Expires: 3600
    ]]>
 </send>
 
 <nop><action><log message="sent REGISTER."/></action></nop>

 

<recv response="200" >
 </recv>
<pause milliseconds="2000">
</pause>
 <send>
 <![CDATA[
        REGISTER sip:10.0.60.1 SIP/2.0
        Via: SIP/2.0/UDP 10.0.60.100:5060;branch=[branch]
        From: <sip:[EMAIL PROTECTED]:5060>;tag=[call_number]
        To: <sip:[EMAIL PROTECTED]:5060>
        Call-ID: [call_id]
        CSeq: [cseq] REGISTER
        Contact: <sip:[EMAIL PROTECTED]:5060>
        Content-Length: 0
        Expires: 3600
    ]]>
 </send>
 <nop><action><log message="sent 2nd."/></action></nop>
   <recv response="200" >
  </recv>
 
 <nop><action><log message="CALL OVER."/></action></nop>
 
 </scenario>

 

Jeff OllierNetwork Quality Control Engineer

 

 


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