[Freeswitch-users] One to many w/o mod_conference

[Tom Hartnett]

Thanks for the clarification Michael. I was working off this old post
http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-June/002414.html
Which I guess is wrong or outdated.

On Wed, Oct 11, 2017 at 3:46 PM, Michael Jerris <mike at jerris.com> wrote:

> eavesdrop is going to transcode.  its copying the raw already decoded
> audio from one channel to the other then re-encoding it.
>
>
> On Oct 11, 2017, at 3:36 PM, Tom Hartnett <hartnett.tom at gmail.com> wrote:
>
> Sorry to beat a dead horse, but so you mind if I ask why? The info I could
> discern (which may or may not be accurate) was that eavesdropping uses a
> simple RTP copy mechanism, while conferencing would involve converting all
> audio back and forth to PCM for each user (as well as jitter buffers and
> other overhead for each participant). Is there something I'm missing?
>
> On Wed, Oct 11, 2017 at 2:56 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> mod_conference would be at least just as efficient as that, probably
>> more.  The feature you are looking for doesn’t exist today.
>>
>> On Oct 11, 2017, at 2:42 PM, Tom Hartnett <hartnett.tom at gmail.com> wrote:
>>
>> Thanks for your comment Michael,
>> Would it be reasonable to have a bunch of callers (listeners) spy or
>> eavesdrop on a single call to a dummy endpoint (sender)?
>>
>> On Wed, Oct 11, 2017 at 2:39 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> we currently do not have a media distribution method to other endpoint
>>> modules other than mod_conference.  You can use one of the stream recording
>>> methods to distribute to cdn as long as the delay isn’t an issue.
>>>
>>> > On Oct 11, 2017, at 12:38 PM, Tom Hartnett <hartnett.tom at gmail.com>
>>> wrote:
>>> >
>>> > Greetings all,
>>> > I have an application where I could use some expert advice. I'd like
>>> to set up an audio broadcast bridge, where the audio from one call would be
>>> distributed to multiple listeners. Listeners have no send audio. Think a
>>> corporate conference call with only one host. All listener calls are
>>> incoming.
>>> > I could of course set this up with mod_conference and just mute all
>>> the listeners. But this requires a bit of overhead to convert everything to
>>> PCM then back to the listener's codecs individually. I'm working on a
>>> resource-strained embedded system, so I'd like to find a way to simply
>>> bridge the speaker audio to multiple endpoints simultaneously. I have very
>>> tight control over the codecs used, and can make sure they are all the
>>> same.  Sort of like live MOH (I'd be using mod_portaudio for the sending
>>> endpoint in fact). Delay must be very low, so converting to a shoutcast
>>> stream and using the MOH function isn't an option either.
>>> > Can anyone offer any advice? Maybe FS is just not suited to this?
>>>
>>>
>
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