Jitterbuffer is not working

Asterisk Asterisk Support
1

Hello,

I’m trying to use jitterbuffer, but it doesn’t seem to work as expected:

Flow: SIP (incoming channel 1) < - > Asterisk < - > SIP (outgoing channel 2)

And added the following to sip.conf:

jbenable=yes
jbforce=yes
jbmaxsize=200
jbresyncthreshold=1000
jbimpl=adaptive
jbtargetextra=40
jblog=yes

context:

[default]
exten => _[0-9a-z].,1,NoOp()
 same => n,Stasis(application,incoming)
 same => n,Hangup()

But I can’t see jitterbuffer logs or any difference during the calls.

2

chan_sip is no longer supported, so if it doesn’t work, it will never work.

Why do you think it is necessary? It is unusual to use this, especially to force it. Normally jitter buffering is done end to end within the VoIP part of a call.

3

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