Jitterbuffer

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There is a jitter buffer implemented in stfu.c.

Usage instructions

<action application="set" data="jitterbuffer_msec=180"/>
<action application="answer"/>

Or to set it on the subsequent outbound call: export sets a variable on both the current channel and on any channels it creates, the 'nolocal:' disables setting it on the current channel and only sets it on the subsequent outbound channels

<action application="export" data="nolocal:jitterbuffer_msec=180"/>
<action application="bridge" data="sofia/default/888@conference.freeswitch.org"/>

http://www.freeswitch.org/node/57


you can also activate the Jitter Buffer in the bridge as follows:

<action application="bridge" data="{jitterbuffer_msec=180}sofia/gateway/$1@gateway.com"/>

This will add a jitter buffer to packets flowing from a remote gateway towards a local freeswitch user. The network would look like this:

(local sip user)---->freeswitch------->(remote gateway)

Where the link between the freeswitch and the remote gateway has a bad, jitter causing connection, and say the local sip user has no jitter buffering on their IP-phone. This will help the voice quality for the incomming audio.

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