[Freeswitch-users] Changing codec during calls

Wed Feb 20 16:04:24 MSK 2013

Wow! This great feature is really work. I have the same issue - one way
audio. And  another: "show channels" always return first negotiated codec.
When switching codec back, audio working fine. Tested with 3 different
softphones.
If someone interested, please confirm.
Thanks.

2013/2/19 Michael Collins <msc at freeswitch.org>

> 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
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> 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
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> 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
>> >> http://www.freeswitch.org
>> >> 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
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
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>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
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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
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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