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

Chris Cureau cmcureau at gmail.com
Wed Aug 3 03:28:22 MSD 2011


Latest git doesn't help unfortunately.

More research is beginning to confirm its a codec issue.  When I limit the
phone to codes that use 20ms packetization, I hear no incoming audio on
outgoing calls.  Incoming calls are okay.  When using All for codes (and the
30ms packetization) I get the choppy voice when leaving MOH.



On Mon, Aug 1, 2011 at 8:34 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> can you do make current and try latest git?
>
>
> On Mon, Aug 1, 2011 at 10:46 AM, Chris Cureau <cmcureau at gmail.com> wrote:
> > 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)
> >
> > 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]
> > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:2753 Already using PCMU
> > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:4819 Set 2833 dtmf send
> > payload to 101
> > 2011-08-01 09:12:37.332892 [DEBUG] sofia.c:5599 Processing updated SDP
> > 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.
> > 2011-08-01 09:12:37.332892 [DEBUG] sofia_glue.c:3052
> > sofia/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?
> >> >
> >> > _______________________________________________
> >> > Join us at ClueCon 2011, Aug 9-11, Chicago
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> >> >
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> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
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> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
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> >> googletalk:conf+888 at conference.freeswitch.org
> >> pstn:+19193869900
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
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> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
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> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
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