WebRTC & Asterisk 11

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A brief presentation on WebRTC and Asterisk 11

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WebRTC & Asterisk 11

  1. 1. WebRTC and AsteriskOverview and demosMalaysian Asterisk User Groupsanjayws@gmail.com
  2. 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. 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. 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. 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. 6. - Making calls with WebRTC on Chrome- Receiving calls from Chrome- Reviewing the codebases
  7. 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. 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. 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. 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

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