Well....with my Asterisk test server, I was running the original licensed G729 CODEC on the WAN only (i.e. over my SIP trunk to the provider).
The phones that I was using for testing were Aastra 53i phones which also support G729 and most other standard CODECS. So there was no proprietary Asterisk CODECs involved, all standard, unless Asterisk, in this scenario, still had to internally transcode G711 (On the LAN)->internal Asterisk CODEC->G729 (On the WAN) before sending the call out the SIP trunk. If that was the case then only when the call load becomes slightly high the quality would suffer otherwise it should be all the same. -----Original Message----- From: Tony Graziano [mailto:[email protected]] Sent: Monday, 25 May 2009 12:54 a.m. To: [email protected]; [email protected] Cc: [email protected] Subject: Re: [sipx-users] VoIP quatlity difference between Asterisk and sipX It really depends on the codecs being used. sipXecs does not have any of its own codecs, and sipXbridge only passes through codecs. If your previous experience was by using an Asterisk proprietary codec, then it is possible "transcoding" was occurring (original codec was being rewritten as another codec and then sent), which usually happens at the server level in Asterisk. Rewriting (transcoding) would add a slght delay, CPU and RAM increase, and the quality your would get would be the lowest quality variable of all the codecs involved. sipXecs does neither transcoding and does not have any proprietary codecs, which allows "fewer things" required by sipXecs to establish or pass through a call. By way of "not" doing these types of actions, it requires less CPU, memory and allows for a possibly heavier call volume as compared to transcoding. >>> "Yakout Esmat" <[email protected]> 05/24/09 1:02 AM >>> 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 _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
