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WebRTC and Asterisk
Overview and demos

Malaysian Asterisk User Group


sanjayws@gmail.com
• 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…
• 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)
• 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
•   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)
- Making calls with WebRTC on Chrome
- Receiving calls from Chrome
- Reviewing the codebases
• 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…
• 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
• 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…
- 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
WebRTC & Asterisk 11

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WebRTC & 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