asterisk users - Jan 2005 - Sound quality - commercial vs. Asterisk

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/a8bfa9b6/attachment.htm

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

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 (?).
A better jitterbuffer and Packet Loss Concealment is what you need.

It's coming to asterisk soon.

http://bugs.digium.com/bug_view_page.php?bug_id=0002532

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/613b553c/attachment.htm