[Asterisk-Users] Anyone using SipP to produce RTP load?

[Dinesh Nair]

On 06/29/05 11:51 Matthew Boehm said the following:
> Hey gang,
>  I've been able to use sipp to produce some call volume on our asterisk
> server. The server has no problems handling 50 simul calls. But then again,
> no RTP is being done. I tried to use the rtp echo ability of sipp but that

i've used the following sipp command line,

sipp -d 30000 -r 5 -t un -sn uac -l 50 -m 100 -s 20 -mp 10000 <asterisk ip>

which will generate 100 calls of 30 seconds each, limiting it to 50 
simultaneous calls at a time to extension 20 on asterisk. extensions 20 was 
a simple

exten => 20,1,Answer()
exten => 20,2,Playback(demo-instruct)
exten => 20,3,Goto(1)

this had asterisk send back the Playback output on RTP port 10000 to sipp.

if you wanted to test ulaw<-->g729 conversions between two asterisk 
servers, have the above exten lines in the second asterisk server with the 
exten lines in the first just being

exten => 20, 1, Dial(IAX2/asterisk2/20)

-- 
Regards,                           /\_/\   "All dogs go to heaven."
dinesh at alphaque.com                (0 0)    http://www.alphaque.com/
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