Asterisk RTMP channels

RTMP Channel driver


The Channel RTMP Asterisk module allows to place audio (and video) calls from a web browser using the FlashPlayer from Adobe(R). The license for chan_rtmp is GPL V2.

We offer a standard free FlashPhone to connect to the Asterisk using the Channel RTMP module.

- Main features
  • Asterisk 1.6 and 11 (help us to port it to Asterisk 13/14)
  • CLI commands
  • Text/Chat features
  • Audio and Video
  • Geo location (with IP)
  • Works with Vconference (Video / Switch module), Transcode (video transcoder)
  • And much more...

- Account provisioning
  • configuration file (rtmp.conf)
  • realtime configuration

- Codecs supported :
  • Audio Speex
  • Audio ulaw
  • Audio alaw
  • Audio PCM 16 bits
  • Video Sorenson (H263 frames, with a header mark)

Demonstration :
default : http://rtmp.ulex.fr/webphone
more looks : http://rtmp.ulex.fr/webphone/look.html
Call 700 (the "echo test"), 0001 or register a user rtmpXX, and call another rtmpYY already registred.

You can request support to :
http://www.ulex.fr
http://www.voximal.com

Sources, can be found at:
https://github.com/voximal/asterisk-rtmp
Created by: borja, Last modification: Wed 05 of Jul, 2017 (22:51 UTC)
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