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.
- 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 https://github.com/voximal/asterisk-rtmp chan_rtmp
cd chan_rtmp/src
make
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.
- default : https://rtmp.ulex.fr:44129/webphone/
- no https : http://rtmp.ulex.fr/webphone/
- more looks : http://rtmp.ulex.fr/webphone/look.html
Contact us with the Ulex web site : http://www.ulex.fr