Hi Rob,
Please find the attached scenario file which we used(added the retrains)

My Apologize for the miscommunication. 
Till yesterday we had the scenario file where SIPp sends BYE.
In our load test call path we had 3 or 4 applications involved to handle the 
call flow.
Since we faced the failure call problem we thought of SIPp to treat if it 
receives a BYE within the timeout otherwise SIPp to send one.
So we changed the scenario file to handle the call flow; thats the actual 
reason which i forget to mention in the email.

Related to retrans from peer;--we will check why the peer had sent too many 
BYEs.
Thanks for the clarification;

The response time repartition table was also not present. shall we add them 
too..
<!-- 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"/>

can you please explain where & when these repartition values will be used,

Thank you for your time

Thanks & Regards
Mahu
-----Original Message-----
From: Robert Day [mailto:robertk...@gmail.com] On Behalf Of Rob Day
Sent: 06 June 2014 01:05
To: Mahudeswaran A
Cc: sipp-users@lists.sourceforge.net
Subject: Re: [Sipp-users] sipp not generating calls after a point;!!

On 05/06/14 19:13, Mahudeswaran A wrote:
> Hi Rob,
> In sequel to the above mail, we did used the attached scenario file, 
> [cid:image001.png@01CF8117.CDE96650]
> Would like to understand from the above snap shot
> 
> If you look at step5, the BYE is received from UAS, why the ‘Retrans’ keeps 
> increasing; how is ‘Retrans’ plays a role in receive end!!
> 
> Thanks & Regards
> Mahu
> From: Mahudeswaran A
> Sent: 05 June 2014 12:21
> To: 'Rob Day'
> Cc: sipp-users@lists.sourceforge.net
> Subject: RE: [Sipp-users] sipp not generating calls after a point;!!
> 
> Hi Rob,
> Please find the attached scenario file; We are trying for the below 
> call scenario in the load setup
> 
> 1.SIPp(uac) will initiate “1-call-per-second—call-duration-60-seconds” 
> to our sip server 2.Our sip application will answer the call and 
> sipp-uac will wait for 60 seconds 3.After 60secs sipp drop the call;

Hi Mahu,

In answer to your question about 'Retrans' - when SIPp is receiving a message, 
the Retrans column indicates how many times the peer has retransmitted that 
message. In your screenshot, the incoming BYE has been resent 15,836 times by 
the other end, and correspondingly, SIPp has resent the 200 OK (step 8) 15,836 
times. (Note that the screenshot attached to your last email doesn't match the 
scenario file you sent before - it expects to receive a BYE as well as send 
one).

I've also taken a look at your original screenshot again (attached for 
reference), and it appears that some of your INVITEs have just vanished.
Note that you have sent 14,970 INVITEs, and have received 11,305 200 OKs, 
received 2693 unexpected messages instead of the 100 Trying, and
872 unexpected messages instead of the 180 Ringing - a total of just
14,870 responses. This is why SIPp thinks 100 calls are still in progress and 
is failing to make any more calls. (You previously said that the BYE was not 
sent, but I think this is wrong - of the 11,305 calls that are correctly 
established, 1,401 fail due to an unexpected message during the pause, and the 
other 9,904 do have BYEs sent, so all
11,305 are accounted for).

From your screenshot, I see that you're using UDP - but you've removed the 
retrans="500" attribute from the INVITE in your scenario file, which is needed 
to make the unreliable UDP transport work reliably. That is, I think what's 
happening here is that occasionally a UDP packet containing an INVITE is lost 
due to network unreliability, and because you have no retransmission timer, 
that call just hangs. Eventually, 100 calls get into this state, and SIPp can't 
make any more. If you add this attribute back in, or switch to using TCP, I 
expect you'll stop seeing this problem.

Best,
Rob

Attachment: OB_sipp_BYE_opt_v3.xml
Description: OB_sipp_BYE_opt_v3.xml

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