[Freeswitch-users] members audio conference

Tue Dec 21 10:53:48 MSK 2010

Conferencing is one of the cases where every member will be doing
transcoding, because even if they all use the same codec it still
needs to uncompress every member's media stream so that it can mix and
recompress them.

If you're worried about CPU usage, the try to get all members to be
using one of the codecs that's a little nicer on the CPU.

-Steve

On 21 December 2010 00:48, Madovsky <infos at madovsky.org> wrote:
> I did a lot of silly things since I learn FS   :D
> but not to try to transcode on FS if I use a proxy media....
> I would just to minimize the transcoding....
>
> ----- Original Message -----
> From: Michael Collins
> To: FreeSWITCH Users Help
> Sent: Monday, December 20, 2010 4:29 PM
> Subject: Re: [Freeswitch-users] members audio conference
>
>
> On Mon, Dec 20, 2010 at 12:02 PM, Madovsky <infos at madovsky.org> wrote:
>>
>> > FS core converts every member from their codec to L16
>> ok I understand, so is it possible to change from L16 to another codec in
>> a conference dialplan ?
>
> I don't think you understood what Steven was saying. You don't need to worry
> about the transcoding. As long as there is a codec that FS supports then the
> transcoding happens without you having to do anything. Unless you've done
> something silly like proxy media then you shouldn't need to worry about
> codecs.
> -MC
>>
>> ----- Original Message -----
>> From: Steven Ayre
>> To: FreeSWITCH Users Help
>> Sent: Monday, December 20, 2010 2:48 PM
>> Subject: Re: [Freeswitch-users] members audio conference
>> More specifically FS core converts every member from their codec to L16
>> which is given to the conference. The conference combines all speaking
>> channels and the resulting L16 is given back to the core to send to the
>> members, the core converting to the correct codec for each member.
>>
>> Steve on iPhone
>> On 20 Dec 2010, at 17:59, Michael Collins <msc at freeswitch.org> wrote:
>>
>> The conference itself doesn't do any codec stuff - FreeSWITCH core does.
>> All I can say is compare the working versus non-working logs and look for
>> clues.
>> -MC
>>
>> On Fri, Dec 17, 2010 at 6:49 PM, Madovsky <infos at madovsky.org> wrote:
>>>
>>> nohgint strange on logs.
>>> but I guess it's a codec and rate problem.
>>> tried with different SIP phones and it works.
>>> how a conference manage the codecs  ?
>>> I know the rate can be set in profile,
>>> but how conference codec is managed
>>>  if all members have different codec and rate ?
>>>
>>>
>>>
>>>
>>> ----- Original Message -----
>>> From: Michael Collins
>>> To: FreeSWITCH Users Help
>>> Sent: Friday, December 17, 2010 7:54 PM
>>> Subject: Re: [Freeswitch-users] members audio conference
>>> What do you see in the debug logs? Did you compare the logs for a working
>>> vs. non-working call? Anything different?
>>> -MC
>>>
>>> On Fri, Dec 17, 2010 at 7:35 AM, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>> when the first member creates and enters in a new conference
>>>> everything is ok. but if a new memeber enters there is no audio
>>>> in the conference, unless the ivr.
>>>>
>>>> I have a very simple conference dialplan like this
>>>>
>>>>         <extension name="create_conference">
>>>>                 <condition field="${sip_to_uri}"
>>>> expression="^000(\d{10,15})@$${domain}$">
>>>>                         <action application="set"
>>>> data="instant_ringback=true"/>
>>>>                         <action application="ring_ready"/>
>>>>                         <action application="sleep" data="3000"/>
>>>>                         <action application="answer"/>
>>>>                         <action application="conference"
>>>> data="$1-${domain_name}@wideband"/>
>>>>                 </condition>
>>>>         </extension>
>>>>
>>>> I tried to add
>>>> <action application="export" data="absolute_codec_string=speex at 16000k"/>
>>>>
>>>> but no success
>>>>
>>>> Thanks
>>>> _______________________________________________
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>>>
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>>
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>
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