[Freeswitch-users] Changing codec during calls

Tue Feb 19 19:46:53 MSK 2013

Great. The code seems to show that the default is to append it to the
current codec list, but = replaces the current codec list instead.
I'll add a comment to that effect to the Wiki.

As far the bridged peers goes, if you do a packet trace of the RTP do
you see media with the correct codec being sent to the client? That
might shed light on whether it's no media sent, wrong codec, or the
some problem/bug on the peer.

Steve

On 19 February 2013 16:37, 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:
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>>>> 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:
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>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>
>>>
>>> _________________________________________________________________________
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>>>
>>> 
>>> 
>>>
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>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>>
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>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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>
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