PJSIP Asteirks 13 – Audio Jitter In One Direction Only

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?We have upgraded a system from Asterisk 11 to Asterisk 13 with pjsip. We are experiencing random Jitter on outbound calls. This was not occurring when running asterisk 11.

We have two IP’s bound to pjsip one on the private vlan network the phones are on and the asterisk one on the asterisk wan vlan. We record the calls on the asterisk switch so we have the call legs. It appears that the audio is making it to the switch fine, but is being garbled before it leaves asterisk to the destination carrier. We have all media running through the server and this is happening when there is only 1 to 2 calls on the line. The cpu, and memory are not even being pushed. We are running G711 as the codec so there should be no real transcoding occurring..

What could be causing this. The users are very upset. This is a very transient issue so the breakup is can occur for two to four seconds and then goes away. It is like asterisk and pjsip are screwing with the audio. Please advise.


One thought on - PJSIP Asteirks 13 – Audio Jitter In One Direction Only

  • PJSIP doesn’t sit in the audio stream, so that’s unlikely to be the culprit. (You’ve also got a lot of variables in play going from 11 =>
    13 beyond just a chan_sip to chan_pjsip conversion).

    Asterisk sits in the audio stream, so it could obviously be causing an issue. Or not.

    How are you recording the calls? Are you using Monitor or MixMonitor?
    With what application arguments?

    If you look at a packet capture, does the packet capture reveal anything about the jitter, and on what call leg?

    Have you tried using a JITTERBUFFER?