[Asterisk-Users] Sound quality - commercial vs. Asterisk

[Sean Kennedy]

Paul Fielding wrote:

> So far in my playing with Asterisk I've messed with soft phones 
> (x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters 
> (Grandstream 286, Digium IAXy).
>  
> I've also got a Vonage line, using a Linksys ATA.
>  
> None of the devices I've connected to my Asterisk server have been 
> able to maintain the same consistent sound quality over a long 
> distance as the Vonage line.    Don't get me wrong, the Grandstreams 
> are actually not too bad, but there is still some breakups that can be 
> annoying.
>  
> Meanwhile the Vonage ATA maintains an almost flawless connection, all 
> the time.
>  
> I'm assuming (perhaps wrongly?) that the Linksys ATA that Vonage uses 
> is still using SIP with some standardized codec.  If that assumption 
> is correct, then how the heck to they manage to get the consistent 
> connection quality?  Is it just a matter of the right setting tweaks 
> within Asterisk and/or the SIP devices?
>  
> I don't think it's a question of Asterisk hardware, since if I connect 
> via local network to the Asterisk server with a SIP device the quality 
> is pretty consistent.   It's generally when remotely connecting that I 
> have the inconsistent sound quality.  This would lead me to believe 
> that it's a matter of tweaking something to deal with latency or 
> packet dropping issues (?).
>  
> What has Vonage got figured out that I still need to?  Any comments 
> would be appreciated...
>  
> regards,
>  
> Paul

Likely, you are running into packet queue problems.  As I recall, the 
vonage device goes on the line before anything else, so it can shape the 
stream to put it's bits first, ensuring it's packets get out in a timely 
matter ( #1 important thing in voip ).  If you were to shape your stream 
and put your voip bits first, then I think you'd see an improvement in 
the qualty of service.

Granted, I don't know your particular situation, so this could all be 
guess work.

Sean