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

Mon Oct 17 11:55:15 MSD 2011

Are you still gettint this Problem? I ran into similiar issues, but it's intermittent ... 

best 
Ray 

----- Original Message -----

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 > 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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>> > 
>> > FreeSWITCH-users mailing list 
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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 
>> 
>> FreeSWITCH-users mailing list 
>> FreeSWITCH-users at lists.freeswitch.org 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
>> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
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> 
> 
> _______________________________________________ 
> Join us at ClueCon 2011, Aug 9-11, Chicago 
> http://www.cluecon.com 877-7-4ACLUE 
> 
> FreeSWITCH-users mailing list 
> FreeSWITCH-users at lists.freeswitch.org 
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> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users 
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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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