[Freeswitch-users] members audio conference

[Michael Collins]

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 <steveayre at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *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>
> 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 <msc at freeswitch.org>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *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>
>> 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"/ <$1-$%7Bdomain_name%7D at wideband%22/>>
>>>                 </condition>
>>>         </extension>
>>>
>>> I tried to add
>>> <action application="export" data="absolute_codec_string=speex at 16000k"/<absolute_codec_string=speex at 16000k%22/>
>>> >
>>>
>>> but no success
>>>
>>> Thanks
>>>
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>>>
>>>
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