[asterisk-users] SIPP question

[Terry Wilson]

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  
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