My personal experience, in a lab setup, is that sipX quality is better than Asterisk when NAT is involved. But as you mentioned there are too many variables to be able to say for sure.
Anyone else can shed some light here? -----Original Message----- From: Scott Lawrence [mailto:[email protected]] Sent: Saturday, 23 May 2009 1:32 a.m. To: Yakout Esmat Cc: [email protected] Subject: Re: [sipx-users] VoIP quatlity difference between Asterisk and sipX On Fri, 2009-05-22 at 23:32 +1200, Yakout Esmat wrote: > > If we had 2 endpoints separated by NAT, hence media has to travel > through the Media Relay agent (the sipX server) would the call quality > be better than if the same endpoints were using an Asterisk server? The real answer is probably that there are too many variables in a real installation to give a general answer that's useful. The sipXecs RTP relay function is designed to be as 'thin' as possible, and in real use (my everyday phone goes through it because I work from my home office behind a NAT), I find the call quality to be excellent. Mine is not a heavily loaded server. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
