[Freeswitch-users] Changing codec during calls

Tue Feb 19 19:48:30 MSK 2013

pastebin the log. I'll bet Steven can diagnose it! ;)
-MC

On Tue, Feb 19, 2013 at 8:37 AM, Emrah <lists at kavun.ch> wrote:

> Excellent, that worked.
>
> However I am having a one way audio issue, where the peer that has its
> codecs changed can hear but cannot be heard.
> Even if both bridged peers are changed to a common codec it doesn't work.
>
> Any idea?
>
> Thanks a bunch for your help.
> On Feb 19, 2013, at 3:47 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
> > Try:
> > uuid_media_reneg 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca =PCMU
> >
> > (ie add = before PCMU)
> >
> > It appears to affect something in the code...
> >
> > -Steve
> >
> >
> > On 19 February 2013 05:54, Emrah <lists at kavun.ch> wrote:
> >> Hey,
> >>
> >> In short, nothing seems to happen. I tried several phones.
> >> I also tried enabling codec renegotiation on hold/unhold thinking that
> the change in SDP with uuid_media_reneg would do it.
> >>
> >> This is what happens:
> >> freeswitch at internal> show channels
> >>
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num
> >> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca,inbound,2013-02-19
> 00:44:35,1361252675,sofia/internal/10000 at sip.domain.net,CS_EXECUTE,Emrah
> - Macbook
> Pro,10000,100.100.100.100,conf-3111,conference,3111-sip.domain.net at ek-custom-16k
> ,XML,ek-conference,G722,16000,64000,G722,16000,64000,,
> cornflakes.domain.net,10000 at sip.domain.net,,ACTIVE,,,,,,
> >>
> >> 1 total.
> >>
> >> freeswitch at internal> uuid_media_reneg
> 49f428e5-1d72-42d8-b0e1-5b3f29f9a4ca PCMU
> >> +OK Success
> >>
> >> 2013-02-19 00:45:25.172659 [DEBUG] sofia_glue.c:2647 Local SDP:
> >> v=0
> >> o=FreeSWITCH 1361234787 1361234790 IN IP4 1.2.3.4
> >> s=FreeSWITCH
> >> c=IN IP4 1.2.3.4
> >> t=0 0
> >> m=audio 17888 RTP/AVP 9 0 101
> >> a=rtpmap:101 telephone-event/8000
> >> a=fmtp:101 0-16
> >> a=silenceSupp:off - - - -
> >> a=ptime:20
> >> a=sendrecv
> >>
> >> 2013-02-19 00:45:25.172659 [DEBUG] switch_core_session.c:975 Send
> signal sofia/internal/10000 at sip.domain.net [BREAK]
> >> 2013-02-19 00:45:25.182652 [DEBUG] sofia.c:5574 Channel sofia/internal/
> 10000 at sip.domain.net entering state [calling][0]
> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send
> signal sofia/internal/10000 at sip.domain.net [BREAK]
> >> 2013-02-19 00:45:25.292661 [DEBUG] switch_core_session.c:975 Send
> signal sofia/internal/10000 at sip.domain.net [BREAK]
> >> 2013-02-19 00:45:25.302683 [INFO] sofia.c:931 sofia/internal/
> 10000 at sip.domain.net Update Callee ID to "10000" <10000>
> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5574 Channel sofia/internal/
> 10000 at sip.domain.net entering state [completing][200]
> >> 2013-02-19 00:45:25.302683 [DEBUG] sofia.c:5585 Remote SDP:
> >> v=0
> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131
> >> s=pjmedia
> >> c=IN IP4 10.0.0.131
> >> t=0 0
> >> a=X-nat:0
> >> m=audio 4010 RTP/AVP 9 101
> >> a=rtcp:4011 IN IP4 10.0.0.131
> >> a=rtpmap:9 G722/8000
> >> a=sendrecv
> >> a=rtpmap:101 telephone-event/8000
> >> a=fmtp:101 0-15
> >>
> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send
> signal sofia/internal/10000 at sip.domain.net [BREAK]
> >> 2013-02-19 00:45:25.302683 [DEBUG] switch_core_session.c:975 Send
> signal sofia/internal/10000 at sip.domain.net [BREAK]
> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5574 Channel sofia/internal/
> 10000 at sip.domain.net entering state [ready][200]
> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia.c:5582 Duplicate SDP
> >> v=0
> >> o=- 3570241475 3570241476 IN IP4 10.0.0.131
> >> s=pjmedia
> >> c=IN IP4 10.0.0.131
> >> t=0 0
> >> a=X-nat:0
> >> m=audio 4010 RTP/AVP 9 101
> >> a=rtcp:4011 IN IP4 10.0.0.131
> >> a=rtpmap:9 G722/8000
> >> a=sendrecv
> >> a=rtpmap:101 telephone-event/8000
> >> a=fmtp:101 0-15
> >>
> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5137 Audio Codec
> Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000]
> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:3022 Already using G722
> >> 2013-02-19 00:45:25.322657 [DEBUG] sofia_glue.c:5266 Set 2833 dtmf
> send/recv payload to 101
> >>
> >>
> >> Thanks for any help,
> >> Emrah
> >> On Feb 18, 2013, at 1:38 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >>
> >>> What happened with the uuid_media_reneg command?
> >>> -MC
> >>>
> >>> On Sat, Feb 16, 2013 at 3:10 PM, Emrah <lists at kavun.ch> wrote:
> >>> Hi all,
> >>>
> >>> How do we go about renegotiating the codecs of a call that is already
> established?
> >>> I tried uuid_media_reneg to no avail.
> >>> E.g.: Upgrade someone who's entered a 16k conference from PCMU to G722.
> >>>
> >>> Cheers and thanks,
> >>> Emrah
> >>>
> _________________________________________________________________________
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> >>>
> >>> 
> >>> 
> >>>
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> >>>
> >>>
> >>> --
> >>> Michael S Collins
> >>> Twitter: @mercutioviz
> >>> http://www.FreeSWITCH.org
> >>> http://www.ClueCon.com
> >>> http://www.OSTAG.org
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
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> >>> http://www.cluecon.com
> >>>
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> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
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> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
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> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
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> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
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>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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