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. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
