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

[Danny Froberg]

Steve Kann wrote:

> 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
>
>------------------------------------------------------------------------
>
I would kill for this to be implemented!
It's sorely needed for us folks that use Transatlantic lines etc.
/Danny