[asterisk-users] Force Jitter Buffer for SIP to SIP calls

[Thermal Wetland]

On Wed, Dec 30, 2009 at 8:27 AM, Matt Darnell <mattdarnell at gmail.com> wrote:

>
> #  Asterisk sip jbenable = yes|no : Enables the use of a jitterbuffer
> on the receiving side of a SIP channel. (Added in Version 1.4)
> # Asterisk sip jbforce = yes|no : Forces the use of a jitterbuffer on
> the receive side of a SIP channel. Defaults to "no". (Added in Version
> 1.4)
>
> It mentions the 'receiving side' which should be the incoming or
> upload form the clients.
> As I am sure you saw, it is not mentioned in the peers and clients section.
> Perhaps setting jbforce to no and jbimpl to adaptive.
>
> I am sure you read all that, anyone have any real world experience?
>
> Aloha,
> Matt
>

Thank you for confirming that I was reading it correctly.

I will be looking at the SPA-2102 to see if it can do anything in regards to
how it is transmitting voice.

-- 
-Thermal
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