A brief presentation on WebRTC and Asterisk 11

1. WebRTC and Asterisk
Overview and demos
Malaysian Asterisk User Group
sanjayws@gmail.com

2. • WEB RealTime Communications
• It’s a project started by Google to
• Enable RealTime Communication straight off browsers
• Run rich realtime media without extra software
• Run on existing supported browsers
• Is now adopted by the internet task force IETF and the W3C consortium
• A HTML5 standard
• Called by javascripts
• Supporting different types of media such as jingle/SIP/XMPP
• Audio, video, telepresence, chat, etc…

3. • Creates lesser interface/software
• Build extremely integrated browser/internet
functions with real time communication
• Less coding huge functionality (built-in APIs)
• Works with almost ANY types of setup (network)
• Works across different browsers, OS platforms
and version
• Works over Web ports (443/80)

4. • WebRTC protocol is officially supported on
Asterisk 11
• Coupled with STUN, ICE and TURN for best
“connectivity”
• Easy configuration and setup
• Supports g711, g722, iLBC and iSAC audio
codecs and VP8 video codecs
• Supports RTP and RTPS over web

5. • Compile asterisk with srtp support
• Configure sip.conf settings
• Configure rtp.conf settings
• Configure the webserver http.conf
• Create SIP accounts to use ws/wss (websocket)
protocol as its signaling protocol (don’t use
FreePBX)

6. - Making calls with WebRTC on Chrome
- Receiving calls from Chrome
- Reviewing the codebases

7. • Build web embedded solutions
• Integrate with CRM, websites, blogs etc…
• Create dial on demand services
• Build a site users just need to click to dial
• Standardize softphone
• Disaster recovery sites
• Etc…

8. • You can use many projects out there including
• PJSIP
• sipML5 (recommended)
• Phono
• Many new exciting projects or simply build your
own! That’s the point of WebRTC

9. • WebRTC in itself is ever growing, evolving, keep
yourself updated
• It is the first release on Asterisk, expect bugs,
expect fixes, again, keep yourself updated
• Keep yourself updated with clients browsers,
etc…

10. - Digium WebRTC wiki:
https://wiki.asterisk.org/wiki/display/AST/Asterisk
+WebRTC+Support
- SipML5: http://sipml5.org/
- My blog (installation guide on Debian)
http://highsecurity.blogspot.com/2012/12/webrtc-
and-asterisk-11-using-sipml5.html