[Freeswitch-users] Asynchronous PTIME

[Serge S. Yuriev]

Добрый день,

On Thu, 23 Dec 2010 11:48:50 -0500 Chris Burns <chris at cloudtel.com>
wrote:

Thank you so much for answers!

> The switch is receiving audio frames larger than it should for the
> negotiated codec, which causes it to analyze the timing of received
> frames and correct the issue.

So it's not wrong signaling problem it's somewhere in RTP, right?

> To stop the switch from trying to fix
> the timing you should be able to uncomment the rtp-autofix-timing
> line in your related sofia profile, but I cant speak to whether that
> will fix your audio.

How I can analyse side "bad" RTP from? WireShark?
 
> There is a logfile.conf.xml to alter any of the settings of how the
> switch logs to file. For debugging SIP to log file you should do it
> outside of the switch console, for instance with ngrep.

Ok, got it.

> On Wed, Dec 22, 2010 at 5:09 PM, Serge S. Yuriev <me at nevian.org>
> wrote:
> 
> > Hello,
> >
> > 2010-12-20 16:51:13.960502 [WARNING] mod_sofia.c:1036 Asynchronous
> > PTIME not supported, changing our end from 20 to 60
> >
> > I'm getting this warning and client hears chopped sound :(
> > That is "Our end"?
> >
> > Call flow is Cisco (sip) FS (sip) YATE (h323) MeraTransitSoftSwitch
> > All but MVTS under my control.
> >
> > I doesn't see any clue in logs and can't reproduce this with my
> > testing via Blink/PortSIP/PhonerLite/Twinkle or Grandstream devices
> >
> > Which debug/logs I should take? Any ideas?
> >
> > Thanks a lot.
> >
> > btw how I can save debug into log not only console?
> > --
> > wbr,
> > Serge
> >
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> >



-- 
Serge S. Yuriev
Lead VoIP engineer