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WebRTC Gateways Introduction

WebRTC Gateways Introduction. Turn the browser into a phone ( with audio, video and sms ..) Why do we need a gateway? - In the browser, signalling is via web-socket - Media : webRTC uses SRTP Make and receive calls to/from traditional PSTN, or H323/ SIP network end points

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WebRTC Gateways Introduction

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  1. WebRTC Gateways Introduction • Turn the browser into a phone ( with audio, video and sms..) • Why do we need a gateway? - In the browser, signalling is via web-socket - Media : webRTC uses SRTP • Make and receive calls to/from traditional PSTN, or H323/ SIP network end points • Work with any html-sip clients ( sipml5, jssip..) • May provide call control, management, accounting and security

  2. WebRTC Gateways Architecture Source : http://code.google.com/p/webrtc2sip/ Source : http://code.google.com/p/sipml5/wiki/Asterisk/

  3. JS SIP Clients SipML5 • Javascript SIP client libraries • SipML5 (http://sipml5.org) • JsSip (http://www.jssip.net) • First open source HTML5 SIP client in JS • Audio/video calls (WebRTC), instant messaging and presence • SIP/SDP signaling over WebSocket • The media stack (SRTP) rely on WebRTC. • Can connect to any SIP/IMS network

  4. SipML5 http://sipml5.org Features • Source code under BSD license. • Works on Chrome, Firefox, IE, Safari, Opera and Bowser • Audio / Video call • Screen/Desktop sharing from Chrome to any SIP client • Instant messaging, Presence • Call Hold / Resume • Explicit Call transfer • Multi-line and multi-account • Dual-tone multi-frequency signaling (DTMF) using SIP INFO • Click-to-Call • SIP TelePresence (Video Group chat) • 3GPP IMS standards

  5. SipML5 http://sipml5.org SipML5 API sipML5 API : How easy ?

  6. Asteriskhttp://www.asterisk.org/ • An open source framework for communication applications • Can be an IP-PBX, VoIP Gateway, Conference server and other custom solutions • A* project started in 1999, today maintained by Digium and the Asterisk community. • Free, OpenSource • Flexible, Reliable • Scalable, Modular • Supports many signaling(& transport) SIP,H323,ISDN etc.. • Transcoding of most of the audio codecs in telephony

  7. Asterisk WebRTC Interface • https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support • From 11.* version, webRTC support added to Asterisk • res_http_websocket module created (for SIP) • ICE, STUN, and TURN support added to res_rtp_asterisk • SRTP support was already added before webRTC : libsrtp and headers must be available • Configuration for webRTC For ws call, sip.conf : - transport=ws,udp,tcp - avpf=yes - nat=yes,force_rport - encryption=yes

  8. Asterisk WebRTC Interface • http.conf : - enabled=yes - bindport=8088 For WSS call, (https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial) • http.conf • tlsenable=yes • tlsbindport=8089 • tlsbindaddr=0.0.0.0 • tlscertfile=/etc/asterisk/keys/asterisk.pem • tlsprivatekey=/etc/asterisk/keys/asterisk.key • tlscipher=ALL • tlsclientmethod=tlsv1 • sip.conf • transport=tls,udp,ws,wss • avpf=yes • encryption=yes • tlsenable=yes

  9. SIP Signaling

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