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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 :