asterisk users - Sep 2009 - SIPP question

Hello

I would like to play file with sipp command.

I want to take value of RTPAUDIOQOS for every user.. I will make it hard
testing with 500 users.

But when all user leave from this conference I am unable to receive proper
value for highlighted in below line..

ssrc=877077954;*
themssrc=0;lp=0;rxjitter=0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000
*

However, when i made single call using SIP phone then i will receive all
value from RTPAUDIOQOs.

Any Idea.. how can I play or transfer/receive Audio packets while testing
with SIPP command [using below command]

I need specially value of receive streams .

*./sipp -sn uac -d 10800000 -s 8601 127.0.0.1 -l 50 -r 1 -rp 5000*
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20090922/b43fbde0/attachment.htm

You need to compile sipp with pcap support.  Here is an example  
scenario: http://sipp.sourceforge.net/doc3.0/reference.html#UAC+with+media
On Sep 22, 2009, at 5:13 AM, DHAVAL INDRODIYA wrote:
> Hello
>
> I would like to play file with sipp command.
>
> I want to take value of RTPAUDIOQOS for every user.. I will make it  
> hard testing with 500 users.
>
> But when all user leave from this conference I am unable to receive  
> proper value for highlighted in below line..
>
> ssrc 
> = 
> 877077954 
> ;themssrc 
> = 
> 0 
> ;lp 
> = 
> 0 
> ;rxjitter 
> =0.000000;rxcount=0;txjitter=0.000000;txcount=0;rlp=0;rtt=0.000000
>
> However, when i made single call using SIP phone then i will receive  
> all value from RTPAUDIOQOs.
>
> Any Idea.. how can I play or transfer/receive Audio packets while  
> testing with SIPP command [using below command]
>
> I need specially value of receive streams .
>
> ./sipp -sn uac -d 10800000 -s 8601 127.0.0.1 -l 50 -r 1 -rp 5000  
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2009 - October 13 - 15 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

Also, you can pass -sn uac_pcap to use the default pcap_play scenario  
which will play 8 seconds of audio and a dtmf digit.  You can also run  
sipp -sd uac_pcap > uac_pcap.xml to save the scenario and edit it.   
You can then play it back using sipp -sf uac_pcap.xml ...