chan_rtmp

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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

Installation

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

Client

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.

Demo

• default : https://rtmp.ulex.fr:44129/webphone/
• no https : http://rtmp.ulex.fr/webphone/
• more looks : http://rtmp.ulex.fr/webphone/look.html

Contact

Contact us with the Ulex web site : http://www.ulex.fr