[asterisk-users] jitterbuffer

[Jeff LaCoursiere]

On Thu, 8 Apr 2010, Tim Nelson wrote:

> ----- "Jeff LaCoursiere" <jeff at jeff.net> wrote:
>> What is the consensus on using the 1.4 jitterbuffer?  Do most people
>> enable it?
>>
>> We have a "PSTN" server that has our RBS T1 trunks in a central
>> location,
>> then have clients that connect via SIP to us for access to those
>> trunks.
>> Most of them are just fine, but lately we have a handful that are
>> having
>> latency and jitter issues.  I am hesitant to just turn on the jitter
>> buffer in zapata.conf on the PSTN server for fear of impacting the
>> clients
>> that are "just fine".
>>
>> Should I be?
>
> I'm using the 1.4 jitterbuffer extensively as many of my customers have 
> poor connectivity (lossy wireless, satellite, etc). It functions well, 
> albeit keep in mind you'll likely need to do some fine tuning to get it 
> just right.
>

I guess that is part of my question - it would seem to me that "tuning" is 
basically sizing the buffer, correct?  And that the tuning would be 
different from client to client, as their latency/jitter needs will be 
different.  How did you handle that aspect?  Did you just keep playing 
until you found something that was a best fit for all clients?

I kind of understand that the dejitter must happen on the way "out" as the 
data gets placed onto a zap channel, and that the other direction should 
be dejittered at the customer's phone or adapter.  In our case this is 
mainly Polycom IP 501s.  I suppose some amount of tuning there will help 
what our client hears.

But the phones are on a 100mb LAN.  So would it be worthwhile to force a 
jitterbuffer on chan_sip on the asterisk server sitting at the client's 
location?

Sorry for trhe vague questions.  I think this would be a great topic for 
someone's BLOG - I haven't found too much in the way of advice via Google 
this morning.

j