[Freeswitch-users] Help with choppy audio after attended transfer
John
freeswitch at earthspike.net
Thu Nov 17 03:54:18 MSK 2011
I also get the problem intermittently, on the outgoing leg only. I have
a Sangoma B700 dual BRI card talking to the PSTN and Yealink T20Ps on
the inside. Everything is G.711 so there is no transcoding. The ISDN
incoming calls are answered by a Lua-based IVR, but the choppiness
starts after MOH and the call is picked up, or when the extension
voicemail greeting is played. Outgoing calls also have the outbound leg
experiencing chopping. I estimated the chopping to be at around 5Hz or
so, which matches the 160octet size being fed into the ISDN G.711. A
reboot cleared all the symptoms (drastic, I know, but I was on a dialup
connection on another job and didn't have time to debug).
I will go through and check that all the ptimes match (eg all at 20ms),
but it appears that this bug still lives, at least on this git:
> version
FreeSWITCH Version 1.0.head (git-6fe6d8d 2011-11-01 11-52-24 -0500)
John
On 17/10/11 08:55, Raimund Sacherer wrote:
> Are you still gettint this Problem? I ran into similiar issues, but
> it's intermittent ...
>
> best
> Ray
>
> ------------------------------------------------------------------------
> *From: *"Chris Cureau" <cmcureau at gmail.com>
> *To: *"FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> *Sent: *Wednesday, August 3, 2011 1:28:22 AM
> *Subject: *Re: [Freeswitch-users] Help with choppy audio after
> attended transfer
>
> 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 <mailto: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
> <mailto: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
> <http://sip:1003@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
> <http://sip:1003@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
> <mailto: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 <mailto: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
> >>
> >> FreeSWITCH http://www.freeswitch.org/
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> >>
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> >>
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> >
> >
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>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900 <tel:%2B19193869900>
>
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