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http://forums.asterisk.org/viewtopic.php?t=87763
Apr 07, 2015 · Hi, I'm pretty new to Asterisk, so I thought I would ask here if this is possible. I want to set up a VOIP server where the music callers hear when they are put on hold is not a saved .mp3 or .wav file, but rather a live online stream of a radio station.
https://github.com/voximal/asterisk-rtmp
Nov 24, 2016 · The RTMP Asterisk module allows to place audio (and video) calls from a web browser using the FlashPlayer from Adobe(R). We offer a free FlashPhone to connect to the Asterisk using the RTMP module. Main features. Writen in C using asterisk-macros. Asterisk 1.6 to Asterisk 11.(help requested to port it to Asterisk 13/14)
https://www.wowza.com/community/questions/3908/flashphoner-rtmp-sip-gateway.html
Dec 21, 2012 · Flashphoner RTMP SIP Gateway is a solution based on Wowza Media Server and designed to integrate VOIP SIP environment with Adobe Flash Player or Adobe Air platforms. Using RSGW, you can develop SIP softphones based on Wowza Media Server that will look just like simple Flash/Javascript applications.
http://forums.asterisk.org/viewtopic.php?t=85483
May 29, 2013 · Hi, I have just started to use Asterisk and have a question about it. I am currently planning to replace a live video streaming system made with Flash + RTMP protocol. The system is composed of a client-server style, and the server relays live video published by …
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
Sep 22, 2016 · If you would like to test Asterisk with WebRTC you can now use the latest shipping Chrome. Audio should work great, but Asterisk 11 does not support the VP8 video codec used by Chrome at the time of this writing. Passthrough support for the video codec VP8 (and Opus for audio) was added in Asterisk 12.
https://en.wikipedia.org/wiki/Real_Time_Messaging_Protocol
Use of librtmp provides these projects with full support of RTMP in all its variants without any additional development effort. FLVstreamer. FLVstreamer is a fork of RTMPdump, without the code which Adobe claims violates the DMCA in the USA. This was developed as a response to Adobe's attempt in 2008 to suppress RTMPdump.
https://github.com/ichramm/chan_rtmp
A RTMP Channel for Asterisk. Writen in C using librtmp. Asterisk 1.4 only, trying to upgrade to Asterisk 11. This module supports realtime only, support for static peers is planed for the near future. Installation
https://marc.info/?l=freeswitch-users&m=128484407813027
[prev in list] [next in list] [prev in thread] [next in thread] List: freeswitch-users Subject: [Freeswitch-users] RTMP Support (Flash) From: anthony.minessale gmail ...
https://www.youtube.com/watch?v=3h6-PSpD-Oc
Jul 29, 2010 · This is a quick demo of the new patch branch for the Asterisk communications toolkit by community member Philippe Sultan, who has created an open-source RTMP conversion tool …
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