[Freeswitch-users] Changing codec during calls

Tue Feb 19 19:37:44 MSK 2013

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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>>> 
>>> --
>>> Michael S Collins
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>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
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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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