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The following information came from Anthony Minessale in a freeswitch-users thread:

Info

If the jitterbuffer is already on, calling it again will just resize it, so setting it to the same is redundant but harmless.

If you are surprised by why the jitterbuffer is paused during bridge:

If both sides of a bridge are RTP and both sides have a jb, its fairly useless. In fact if anything, it can worsen call quality.

You should only run jitterbuffers at points of termination change of protocol. Examples, if FS was hosting a conference or IVR, if you are bridging the call to a phone for instance, you want to not use a jitterbuffer because you want to preserve the original timestamps so your phone can use its own jitterbuffer.

For special examples where you are using FS jitterbuffer in front of something else that may not have one or some other special circumstance you can use the setting chris mentioned to leave it running.

Activation instructions

The jitter buffer can be activated via channel variable, dialplan app, or sofia param.

The jitter buffer has three params that control its behavior: length, max length, and max drift. Length is the initial size of the jitter buffer in milliseconds. Max length is the upper bound for how big the jitter buffer can grow. Max drift controls how much delay the jitter buffer will tolerate before dropping frames to make up ground.

Dialplan app

Enable 60 ms jitter buffer:

Code Block
<action application="jitterbuffer" data="60"/>

Enable 60 ms jitter buffer with 200ms max length and 20 ms max drift:

Code Block
<action application="jitterbuffer" data="60:200:20"/>

Sofia profile param

This param is the initial size of the jitter buffer in milliseconds. The max length and max drift values can't be set with this param.

Code Block
<param name="auto-jitterbuffer-msec" value="60"/>


Info

The parameter uses -hyphens- while the variable uses _underscore_

Channel Variable

jitterbuffer_msec

Excerpt Include
jitterbuffer_msec
jitterbuffer_msec
nopaneltrue

Other usage

rtp_jitter_buffer_plc

Enables/disables packet loss concealment (PLC) when using the jitter buffer. PLC is enabled by default when the jitter buffer is enabled. Set this variable before enabling the jitter buffer for it to have an effect.

Usage:

Code Block
<action application="set" data="rtp_jitter_buffer_plc=true"/>

or to disable PLC:

Code Block
<action application="set" data="rtp_jitter_buffer_plc=false"/>

rtp_jitter_buffer_during_bridge

Enables/disables the jitter buffer during bridge.

Usage:

Code Block
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/>

or,

Code Block
<action application="set" data="rtp_jitter_buffer_during_bridge=false"/>

To enable jitter buffer on only the B-leg of the call, issue commands based on these examples in this sequence:

<action application="export" data="rtp_jitter_buffer_during_bridge=true"/>
<action application="export" data="nolocal:jitterbuffer_msec=60:120"/>


Pause

The jitter buffer can be paused mid-call

Code Block
<action application="jitterbuffer" data="pause"/>

Resume

The jitter buffer can be resumed mid-call

Code Block
<action application="jitterbuffer" data="resume"/>

Debug

Jitter buffer debugging can be enabled/disabled.

Code Block
<action application="jitterbuffer" data="debug:${uuid}"/>
<action application="jitterbuffer" data="debug:off"/>


Lookahead

If using the Opus codec (popular with WebRTC) and the far end is sending F.E.C. (Forward Error Correction) information, you can enable a look-ahead jitter buffer in the codec configuration:

Code Block
titleautoload_configs/opus.conf.xml
<param name="use-jb-lookahead" value="1"/> 

On FreeSWITCH version 1.6 and later this greatly improves the audio performance even with heavy packet loss.