Skip to content
Asterisk RTMP Channel
C JavaScript HTML Other
Branch: master
Clone or download
Fetching latest commit…
Cannot retrieve the latest commit at this time.
Type Name Latest commit message Commit time
Failed to load latest commit information.
doc Add specification and documentation files. Nov 24, 2016
src Add initial sources. Nov 24, 2016
.gitignore Add the FlashPhone client Dec 8, 2016
LICENSE Add license file. Nov 24, 2016 Update Jan 8, 2019


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)
  • This module supports realtime and static peers.
  • Text/Chat features
  • Audio and Video
  • Codecs supported : Speex, a/ulaw , PCM 16 bits, Video Sorenson
  • Geo localisation (with IP)
  • Works with Vconference (Video / Switch module), Transcode (video transcoder)
  • configuration file (rtmp.conf)
  • realtime configuration


export ASTERISKMACROSDIR=[Asterisk macros Git Voximal directory]
export ASTERISKDIR=[Asterisk sources directory]
export LINUX_BUILD=[x86-64 or i686 or armv6l]
export LIBGEOIPDIR=[GeoIP sources directory]/libGeoIP/

git clone chan_rtmp
cd chan_rtmp/src
make install


The client over FlashPlayer allows to set differents skins. (Chrome seems to need to host your HTML pages in a HTTS server)

An Android SDK for smartphone/webtv is available to create video call applications.



Contact us with the Ulex web site :

You can’t perform that action at this time.