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.




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