[Freeswitch-users] Help with choppy audio after attended transfer

[Chris Cureau]

Anthony,

Thanks for answering...and sorry for the delay.  I've already checked all of
the ptime settings I can, and all phones plus freeswitch are set to use 20ms
packetization.  I've even set "scrooge" in the codec negotiation, but I keep
running into this issue.  I've updated my post with "sofia global siptrace
on".

I am assuming that the ptime issue happens around line 2462 (
http://pastebin.freeswitch.org/16935)


   1. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4711 Audio Codec
   Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:30:64000]
   2. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using
   PCMU
   3. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf
   send payload to 101
   4. 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP
   5. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3042 Audio params are
   unchanged for sofia/internal/sip:1003 at 10.0.1.205:5060.
   6. 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052sofia/internal/sip:
   1003 at 10.0.1.205:5060 Setting audio receive payload in Re-INVITE to 0

Could this be an issue with the Aastra'a firmware?  Or maybe the MOH is
being processed at 30ms instead of 20ms, and the negotiation is not updated
somehow?

I don't mean to sound ignorant, but I'm really at a loss here...and thanks
again for any help!

Cheers,
Chris

On Fri, Jul 29, 2011 at 10:44 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> probably ptime related thing.
> you should have included the sip trace "sofia global siptrace on"
>
>
> On Fri, Jul 29, 2011 at 12:28 AM, Chris Cureau <cmcureau at gmail.com> wrote:
> > I'm having some issues with extremely choppy audio after a call has been
> > sent to another extension via an automated transfer.  The audio is great
> > when the call is answered.  Shortly after, the transfer button is pressed
> > and the incoming call hears music on hold.  The music on hold is sent to
> the
> > caller sounds fine as does the conversation between extensions.  When the
> > transfer is completed, the caller hears what sounds like someone speaking
> > through a fan (though slower) but incoming audio sounds fine.
> >
> > Since it's such a large log, I posted it to the FreeSWITCH pastebin:
> > http://pastebin.freeswitch.org/16911
> >
> > I'm thinking that it has something to do with the transition from MOH to
> the
> > internal extension, but I can't figure out what is happening.
> >
> > Any ideas?
> >
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>
>
>
> --
> Anthony Minessale II
>
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