[Freeswitch-users] Changing codec during calls

Tue Feb 19 11:47:46 MSK 2013

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
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> 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
>> http://www.freeswitch.org
>>
>>
>>
>> --
>> 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
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.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
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org