[Asterisk-Users] Asterisk on a media stream vs. direct RTP communication between endpoints

[Irakli Natsvlishvili]

Hello everybody,

I'd like to know was there any load tasting done with *? Let's imagine 500
SIP clients on a server, 80 simultaneous calls. No transcoding, G711 or G729
codecs are used between endpoints.

How asterisk performs with 80 simultaneous calls when it sits on a media
stream? Is there any recommendation for hardware? Is there any graphs
available showing degradation of performance or adding latency on a same
hardware when number of simultaneous calls increases?

Anybody?

Thanks,
Irakli

P.S. The reason for this question is that I try in my VoIP designs to
eliminate central point for RTP streams. And so far I'm convinced that a
correct resign requires direct RTP communication between endpoints.