[Freeswitch-users] Looking for more info about mod_rtmp

Richard Alam ritzalam at gmail.com
Wed Nov 24 18:43:33 PST 2010


Hi,

Is there any work being done on this (mod_rtmp)? We are willing commit
a bounty of $2K.

We at bigbluebutton.org need to connect to a conference in FS directly
from a flash client.

We want to basically bypass our current voip (bbb-voice) app and have
the client connect directly to freeswitch.

The use case will be for to click on a button and connect to the voice
conference. It would be great if we can have the user connect in
listen only mode but then allow them to click on a button to be able
to talk (push to talk capability).

This saves bandwidth as the user will have a stream to FreeSWITCH when
she wants to talk. The only permanent stream is for listening.
However, the presenter may need both streams at the same time so it
would be great if the implementation is flexible to support both by
passing a parameter or whatever's is best way.

I sent an email to the person I thought was developing RTMP support
(http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-August/004155.html)
but haven't heard anything back.

Please let me know if anybody is interested to take this on or if
there are any questions/clarifications.

The resulting work will be open sourced. Great if it becomes part of
FS or just live in github.

There may be some enhancements later on (e.g. DTMF support) but the
above scenario will be good for the first iteration.

Would be great to have some working prototype by end of Dec or early Jan 2011.

Thanks so much.

Richard

PS. Should I send this also to the dev list?

On Tue, Sep 28, 2010 at 5:21 PM, Brian West <brian at freeswitch.org> wrote:
> See question number 1 on the FAQ!
>
> /b
>
> On Sep 28, 2010, at 3:59 PM, Steven Ayre wrote:
>
>> Not in either git trunk or contrib repositories so it hasn't been released yet.
>>
>> It's still under development.
>>
>> -Steve
>>
>>
>> On 28 September 2010 14:39, Jarle Aase <jgaa at jgaa.com> wrote:
>>> I'm looking for a way to bridge rtmp to sip / audio-conferences, and
>>> came across a reference to "mod_rtmp". Does anyone have any information
>>> about that project?
>>>
>>> Jarle
>>
>
>
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-- 
---
BigBlueButton
http://www.bigbluebutton.org
http://code.google.com/p/bigbluebutton



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