SIP Troubleshooting - Enable Jitter Buffer for Asterisk

To solve broken audio issues, you can enable Jitter Buffer for Asterisk. Details below.

Change this in the sip_general_custom.conf 

;Enable a Jitter Buffer for Asterisk

jbenable=yes

jbforce=yes

jbimpl=adaptive

jbmaxsize=200

jbresyncthreshold=1000

jblog=yes

jbenable = yes|no

SIP channel. Defaults to "no". An enabled jitterbuffer willbe used only if the sending side can create and the receiving side can not accept jitter. The SIP channel can accept jitter, thus a jitterbuffer on the receive SIP side will be used only if it is forced and enabled.

jbforce = yes|no

Forces the use of a jitterbuffer on the receive side of a SIP channel. Defaults to "no".

jbmaxsize = #number

Max length of the jitterbuffer in milliseconds.

jbresyncthreshold = #number

Jump in the frame timestamps over which the jitterbuffer isresynchronized. Useful to improve the quality of the voice, with big jumps in/broken timestamps, usually sent from exotic devices and programs. Defaults to 1000.

jbimpl = fixed|adaptive

Jitterbuffer implementation, used on the receiving side of a SIP channel. Two implementations are currently available - "fixed" (with size always equals to jbmaxsize) and "adaptive" (withvariable size, actually the new jb of IAX2). Defaults to fixed.

jblog = no|yes

Enables jitterbuffer frame logging. Defaults to "no".


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