Apidaze WebRTC Workshop barcelona 21st april 2013
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Apidaze presentation at the WebRTC Workshop
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Apidaze WebRTC Workshop barcelona 21st april 2013
- 1. WebRTC on the telco side
- 2. Who are we? • Based in Paris, France • A developer oriented cloud communica>ons provider • We built our infrastructure with opensource so@ware • We provide a telecom API – apidaze.io : a REST/XML API for web developers
- 3. WebRTC and (opensource) telecom • How? – WebSocket + SIP – A JavaScript SIP+WebSocket API on the client side • JsSIP / Doubango / QoffeeSIP – A SIP+WebSocket server • Asterisk (possibly OverSIP or Kamailio as a frontend) • Turns the web browser into a SIP client
- 4. For the developer • Include a SIP+WebSocket library, and start coding <head> <script src="http://jssip.net/download/jssip-devel.js"> </head> <body> <script> var configuration = { 'outbound_proxy_set': 'ws://94.23.192.150:8088', 'uri': 'sip:alice@example.com', 'password': 'superpassword' }; var useAudio = true; var useVideo = true; var views = { 'selfView': document.getElementById('my-video'), 'remoteView': document.getElementById('peer-video')}; var eventHandlers = { 'connecting': function(e){ // Your code here }, 'progress': function(e){ // Your code here }, 'failed': function(e){ // Your code here }, 'started': function(e){ // Your code here }, 'ended': function(e){ // Your code here } }; var phone = new JsSIP.UA(configuration); phone.call('sip:bob@example.com', useAudio, useVideo, eventHandlers, views); … </script> </body>
- 5. Is SIP necessary? • A WebRTC developer is a web developer – SIP : What is this? I’m not a telco guy, I’m a web developer! – The WebRTC API is here, why not use it? • Solu>on : a WebRTC gateway, with no or liZle signalling – Use JSON, a well known format for web developers, to carry signalling messages
- 6. WebSocket gateway capabili>es • Must « speak » ICE • Must encrypt media traffic – SRTP/SDES now, SRTP/DTLS in the future
- 8. Show me the code! socket = new WebSocket('ws://94.23.192.150:8088/agaf6c31?secret=43fdre03fa1705a&action=joinroom&roomname=' + roomName, 'webrtc'); /* Set listeners on the socket for events 'open', 'close', 'message' */ socket.addEventListener("open", function(event) {console.log("S->C : WebSocket connection open")}); socket.addEventListener("close", function(event) {console.log("S->C : WebSocket connection closed")}); socket.addEventListener("message", onMessage, false); … function onMessage(evt) { console.log("S->C: "+evt.data); var obj = JSON.parse(evt.data); switch (obj.event.type) { case "confbridgejoin": break; case "confbridgetalking”: break; case “confbridgeleav”: break; } } … /* Send message as a JSON string */ function sendMessage(type, body) { var tmp = {}; tmp['type'] = type; tmp['sdp'] = body; message = JSON.stringify(tmp); console.log("C->S : " + message); socket.send(message); }
- 9. Interac>ng with the gateway • Enter key iden>fier/secret socket = new WebSocket('ws://94.23.192.150:8088/agaf6c31? secret=43fdre03fa1705a&action=joinroom&roomname=' + roomName, 'webrtc'); • Send commands through the WebSocket – JSON – Implement your own signalling protocol using our gateway
- 10. More to come