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

[Matt Darnell]

On Wed, Dec 30, 2009 at 8:11 AM, Thermal Wetland
<thermalwetland at gmail.com> wrote:
> We have a customer on a wireless connection that has very bad jitter. They
> can hear people fine, but people have a very hard time hearing them. They
> are connected via a SPA-2102.
>
> It is a SIP client going to a SIP trunk.
>
> Something like this in sip.conf [general] would be in effect for all SIP
> clients:
> jbenable = yes
> jbmaxsize = 150
> jbresyncthreshold = 1000
> jbimpl = fixed
> jblog = yes
>
> I only want to enable the jitter buffer for the end points having the
> trouble.
>
> Reading the docs, it seems that the jitter buffer is only used when the end
> point is connected to an app like voicemail.
>
> --
> -Thermal
>
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This is from voip-info.org -
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
It is in the [general] section

#  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