[Freeswitch-users] Changing codec during calls

[Emrah]

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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