Kirwan, David (David | 1 Apr 16:25 2009

Re: random reinvite

Here is a call flow diagram to go along with my reinvite question
 
I have subst the ip's out
 
|Time     | a.b.c.d    | w.x.y.z     |
|13.045   |         Request: INVITE sip           |SIP/SDP: Request: INVITE sip:9000 <at> w.x.y.z, with session description
|         |(5061)   ------------------>  (5060)   |
|13.046   |         Status: 100 Trying            |SIP: Status: 100 Trying
|         |(5060)   <------------------  (5060)   |
|13.047   |         Status: 200 OK, wit           |SIP/SDP: Status: 200 OK, with session description
|         |(5060)   <------------------  (5060)   |
|13.052   |         Request: ACK sip:60           |SIP: Request: ACK sip:6000 <at> w.x.y.z
|         |(5061)   ------------------>  (5060)   |
|15.577   |         Request: INVITE sip           |SIP/SDP: Request: INVITE sip:user1 <at> a.b.c.d;transport=udp, with session description
|         |(5060)   <------------------  (5060)   |
|15.578   |         Status: 100 Trying            |SIP: Status: 100 Trying
|         |(5060)   ------------------>  (5060)   |
|15.728   |         Status: 200 OK, wit           |SIP/SDP: Status: 200 OK, with session description
|         |(5060)   ------------------>  (5060)   |
|15.730   |         Request: ACK sip:jb           |SIP: Request: ACK sip:user1 <at> a.b.c.d;transport=udp
|         |(5060)   <------------------  (5060)   |
|20.050   |         Request: BYE sip:60           |SIP: Request: BYE sip:6000 <at> w.x.y.z
|         |(5061)   ------------------>  (5060)   |
|20.052   |         Status: 200 OK                |SIP: Status: 200 OK
|         |(5060)   <------------------  (5060)   |


From: Kirwan, David (David)
Sent: 01 April 2009 15:11
To: sipp-users <at> lists.sourceforge.net
Subject: random reinvite

Hi all,
 
Does SIPp have a way of handling random reinvites?

Is it possibly to dynamically change the ports for RTP used by sip?

Best regards,
David
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Kirwan, David (David | 1 Apr 16:10 2009

random reinvite

Hi all,
 
Does SIPp have a way of handling random reinvites?

Is it possibly to dynamically change the ports for RTP used by sip?

Best regards,
David
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KATWALA, KALPESH, ATTLABS | 1 Apr 22:02 2009
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Error on Invite/REG load

Hi,

I'm running a call load to an S/BC-->IMS Server.  The S/BC times out the
Registry after 30 minutes.  I can run a load for 30 minutes only.  As a
workaround, I added the following logic and I'm running into other
issues.

    <REGISTER all UE's>  -> Separate SIPp instance
	
Call Processing:
Step1:     	INVITE              	-->
    		180Ringing      	<--
    		200OK(INV)    	<--
    		ACK   		-->
    		BYE             	-->
    		200OK(BYE)    	<--	
    		ACK                    	-->
<Small % of the UE's ReRegister; the rest goes to step 1>
   		REGISTER          	-->
   		401                     	<--
    		REG-Chlng       	-->
    		200OK(REG)    	<--

    Then go back to the First step (Sending out Invite).

The Data file contains 500 origs and 500 term UE's.
The problem I see is that after completing the first 500 calls
successfully , the first UE gets a 503 error.  From Call traces on SIPp,
I see that SIPp tries to send another INVITE after the call is
established.

Any ideas?

Kalpesh Katwala
SVT Testing Group, IMS, AT&T

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hui cheng | 2 Apr 10:12 2009
Picon

sipp can not reply BYE request as UAS mode

Hi Buddy

I am a newby here. I met some issue while test my SIP Client.
the scenario is :
                                 Messages  Retrans   Timeout   Unexpected-Msg
  ----------> INVITE             1         3         0         0       
  <---------- 180                1         0                           
  ----------> BYE                0         0         0         0       
  <---------- 200                0         0                           
  <---------- 487                0         0                           
  ----------> ACK                0         0         0         0       
------------------------------ Test Terminated -----------------------------

as you can see from the scenario ,SIPp 's role is a UAS . The 200 response cannot deliver to the SIP Clinet(between SIP Clinet and SIPp ,there is a SIP Proxy)

I guest the 200 is delivered to the Proxy, but not the SIP Clinet.

my sipp command is :
#./sipp 135.252.181.44:5060 -sf $PWD/uas_cancel.xml -in
f $PWD/uas1.csv -t un -max_socket 150 -r 10  -p 10050 -trace_msg

I guest SIPp does not set the response dest address right , as it send the response on the same socket of request . How cant I config sipp to let it sendaddress follow first Via header content?which is the standard solution.

THANKS!!




     

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sandip gangakhedkar | 1 Apr 12:58 2009
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SIPP installation blues

Hi all,


Ive installed the SIPP Windows executable in Win XP. I have installed cygwin with all the packages. Now when I run sipp I get an ugly error like this: 
 
     18 [main] ? (2064) C:\Program Files\Sipp_3.1\sipp.exe: *** fatal error - sy
tem shared memory version mismatch detected - 0x8A88009C/0x2D1E009C.
This problem is probably due to using incompatible versions of the cygwin DLL.
Search for cygwin1.dll using the Windows Start->Find/Search facility
and delete all but the most recent version.  The most recent version *should*
reside in x:\cygwin\bin, where 'x' is the drive on which you have
installed the cygwin distribution.  Rebooting is also suggested if you
are unable to find another cygwin DLL.

I had a previous installation of cygwin which was corrupt, hence I deleted it and then did a fresh install of cygwin with ALL packages. I found 2 cygwin1.dll files - one in cygwin's /bin folder and the other in C:\Program Files\Sipp_3.1
I have no idea where the issue lies.
Any help in this regard will be greatly appreciated.

Thanks,
Sandip

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SourceForge.net | 1 Apr 11:51 2009
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[ sipp-Bugs-2724798 ] Segmentation fault using <exec command with SIPp 3.1

Bugs item #2724798, was opened at 2009-04-01 11:51
Message generated for change (Tracker Item Submitted) made by ericdelas
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=637564&aid=2724798&group_id=104305

Please note that this message will contain a full copy of the comment thread,
including the initial issue submission, for this request,
not just the latest update.
Category: None
Group: None
Status: Open
Resolution: None
Priority: 5
Private: No
Submitted By: Eric DELAS (ericdelas)
Assigned to: Nobody/Anonymous (nobody)
Summary: Segmentation fault using <exec command with SIPp 3.1

Initial Comment:
Hi all,

You will get a segmentation fault with SIPp 3.1 when using <exec command whithin an xml script.
For example:
  <nop>
    <action>
      <exec command="echo This leads to a segmentation fault"/>
    </action>
  </nop>

In order to avoid the segmentation fault, you can change the SIPP code.
Edit the call.cpp file and change the following lines:
            if (currentAction->getCmdLine()) {
                char* x = createSendingMessage(currentAction->getMessage(), -2 /* do not add crlf*/);
With:
            if (currentAction->getCmdLine()) {
                char* x = createSendingMessage(currentAction->getCmdLine(), -2 /* do not add crlf*/);

Hope it will help.
BR,
Eric Delas.

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mayamatakeshi | 2 Apr 16:36 2009
Picon

Re: sipp can not reply BYE request as UAS mode

On Thu, Apr 2, 2009 at 5:12 PM, hui cheng <avantasia2008 <at> gmail.com> wrote:
> Hi Buddy
>
> I am a newby here. I met some issue while test my SIP Client.
> the scenario is :
>                                  Messages  Retrans   Timeout
> Unexpected-Msg
>   ----------> INVITE             1         3        
0         0
>   <---------- 180                1         0
>   ----------> BYE                0         0        
0         0
>   <---------- 200                0         0
>   <---------- 487                0         0
>   ----------> ACK                0         0        
0         0
> ------------------------------ Test Terminated -----------------------------
>
> as you can see from the scenario ,SIPp 's role is a UAS .

Hello,
it seems there is something missing in your scenario.
Where is the 200 OK for INVITE?

> The 200 response
> cannot deliver to the SIP Clinet(between SIP Clinet and SIPp ,there is a SIP
> Proxy)
>
> I guest the 200 is delivered to the Proxy, but not the SIP Clinet.

Do not guess. Get a packet capture.

regards,
takeshi
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Elad Kugman | 2 Apr 17:07 2009

Does sipp support RTSP?

Hi all,

 

                I know that sipp support TLS (signaling ) does it support RTSP too?

 

Thanks,

Elad

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Kirwan, David (David | 2 Apr 17:54 2009

Re: Does sipp support RTSP?

Do you mean SRTP?
 
David
 

From: Elad Kugman [mailto:eladk <at> radvision.com]
Sent: 02 April 2009 16:07
To: sipp-users <at> lists.sourceforge.net
Subject: [Sipp-users] Does sipp support RTSP?

Hi all,

 

                I know that sipp support TLS (signaling ) does it support RTSP too?

 

Thanks,

Elad

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Mike Ayers | 2 Apr 19:32 2009

Re: SIPP installation blues

> From: sandip gangakhedkar [mailto:sandipfloyd <at> gmail.com] 
> Sent: Wednesday, April 01, 2009 3:58 AM

> I had a previous installation of cygwin which was corrupt, 
> hence I deleted it and then did a fresh install of cygwin 
> with ALL packages. I found 2 cygwin1.dll files - one in 
> cygwin's /bin folder and the other in C:\Program Files\Sipp_3.1
> I have no idea where the issue lies.
> Any help in this regard will be greatly appreciated.

	The problem is most likely that whomever distributed the SIPP "Windows" executable compiled it with
cygwin, then bundled in the cygwin DLL, which is necessary to run it.  The problem is that you can have only
one cygwin DLL on a system, and different versions *will* corrupt each other.  Best to delete your SIPP,
download the sources, and follow these:

http://sipp.sourceforge.net/wiki/index.php/Compilation#Windows_XP

	HTH,

Mike

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Gmane