Looking for more info about mod_rtmp
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
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@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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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@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
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@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@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
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton
participants (4)
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Brian West
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Jarle Aase
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Richard Alam
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Steven Ayre