NICTA Copyright 2010 From imagination to impact 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

NICTA Copyright 2010 From imagination to impact 2

WebRTC Gateways Architecture

Source : http://code.google.com/p/webrtc2sip/

Source : http://code.google.com/p/sipml5/wiki/Asterisk/

NICTA Copyright 2010 From imagination to impact 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

NICTA Copyright 2010 From imagination to impact 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

NICTA Copyright 2010 From imagination to impact 5

SipML5 http://sipml5.org SipML5 API

sipML5 API : How easy ?

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Asterisk http://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

NICTA Copyright 2010 From imagination to impact 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

NICTA Copyright 2010 From imagination to impact 8

Asterisk WebRTC Interface

• http.conf : - enabled=yes

- bindport=8088

For WSS call,

(https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial)

sip.conftransport=tls,udp,ws,wssavpf=yesencryption=yestlsenable=yes

http.conftlsenable=yestlsbindport=8089tlsbindaddr=0.0.0.0tlscertfile=/etc/asterisk/keys/asterisk.pemtlsprivatekey=/etc/asterisk/keys/asterisk.keytlscipher=ALLtlsclientmethod=tlsv1

NICTA Copyright 2010 From imagination to impact

SIP Signaling

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