[asterisk-users] PJSIP Asteirks 13 - Audio Jitter in one direction only

Matthew Jordan mjordan at digium.com
Mon Oct 23 13:09:55 CDT 2017

On Wed, Oct 18, 2017 at 9:52 AM, Bryant Zimmerman <BryantZ at zktech.com> wrote:
> ?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.
> zktech

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?

Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

More information about the asterisk-users mailing list