Web RTC Gateways Introduction Turn the browser into

Web. RTC 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 : web. RTC uses SRTP • Make and receive calls to/from traditional PSTN, or H 323/ SIP network end points • Work with any html-sip clients ( sipml 5, jssip. . ) • May provide call control, management, accounting and security NICTA Copyright 2010 From imagination to impact 1

Web. RTC Gateways Architecture Source : http: //code. google. com/p/webrtc 2 sip/ Source :

JS SIP Clients Sip. ML 5 • Javascript SIP client libraries – Sip. ML 5 (http: //sipml 5. org) – Js. Sip (http: //www. jssip. net) • First open source HTML 5 SIP client in JS • Audio/video calls (Web. RTC), instant messaging and presence • SIP/SDP signaling over Web. Socket • The media stack (SRTP) rely on Web. RTC. • Can connect to any SIP/IMS network NICTA Copyright 2010 From imagination to impact 3

Sip. ML 5 http: //sipml 5. 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 Tele. Presence (Video Group chat) 3 GPP IMS standards NICTA Copyright 2010 From imagination to impact 4

Sip. ML 5 http: //sipml 5. org Sip. ML 5 API sip. ML 5

Asterisk http: //www. asterisk. org/ • An open source framework for communication applications • Can be an IP-PBX, Vo. IP Gateway, Conference server and other custom solutions • A* project started in 1999, today maintained by Digium and the Asterisk community. • Free, Open. Source • Flexible, Reliable • Scalable, Modular • Supports many signaling(& transport) SIP, H 323, ISDN etc. . • Transcoding of most of the audio codecs in telephony NICTA Copyright 2010 From imagination to impact 6

Asterisk Web. RTC Interface • • • https: //wiki. asterisk. org/wiki/display/AST/Asterisk+Web. RTC+Support From 11. * version, web. RTC 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 web. RTC : libsrtp and headers must be available • Configuration for web. RTC For ws call, sip. conf : - transport=ws, udp, tcp - avpf=yes - nat=yes, force_rport - encryption=yes NICTA Copyright 2010 From imagination to impact 7

Asterisk Web. RTC 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 tlscertfile=/etc/asterisk/keys/asterisk. pem tlsprivatekey=/etc/asterisk/keys/asterisk. ke y tlscipher=ALL tlsclientmethod=tlsv 1 NICTA Copyright 2010 From imagination to impact sip. conf transport=tls, udp, wss avpf=yes encryption=yes tlsenable=yes 8