Web rtc.intro

1. mm1 Consulting & Management WebRTC: An introduction Bonn, May 2012

2. WebRTC is an exciting new technology framework for rich media communications. WebRTC provides full video, voice and text communication capabilities to web browsers RTC: Real-Time Communications, i.e. text, voice and video It is an implementation of an IETF workgroup (RTCweb)  open standard Based on code Google acquired from Global IP Solutions (GIPS: the voice engine used by Skype) Released to the public domain:  open source WebRTC enables rich media applications such as voice calls, video chat, white-boarding , gaming etc . without any client or plug-in to run from a browser using simple HTML and JavaScript APIs. 2

3. WebRTC is a pure greenfield technology with heavy use of Web 2.0 and HTML 5 elements The vision is to establish the browser as the endpoint of all communication Media fully supported Pure JavaScript: no plugins or executable beyond the browser! open standard (IETF), open source (thx to Google) No signaling : switching and connecting is out of scope, i.e. bring your own network . No legacy integration provided: pure greenfield disruption Very high-level and user friendly: „ With the WebRTC spec, a great 1:1 video chat experience can be built with under 100 lines of JavaScript code” 3

4. Market penetration will be almost complete Chrome, Firefox and IE are working on it. Android browsers – tablets first – will follow Google Chrome : WebRTC in developer channel.   Google Talk and Google Hangouts (G+) will eventually be ported to WebRTC. Mozilla Firefox : Mozilla integrated WebRTC into its Firefox alpha in early 2012 which gave the browser the ability to perform audio mixing on a media stream.  In April 2012 Mozilla released a demo of WebRTC video calling that ran inside the Firefox browser Internet Explorer : Microsoft has also started work on implementation of the API Android browsers : rumours only...but think of the implications! mm1 bid management: The CCT approach. 4

5. WebRTC cs. SIP-based VoIP: technobabble.. 5 Classic VoIP WebRTC Signaling SIP or H.323 in most cases Undefined Media transport RTP/RTCP RTP/RTCP Security SRTP in SIP,H.235 in H.323 SRTP NAT traversal STUN/TURN/ICE in SIP,H.450.x in H.323 STUN/TURN/ICE Video codecs H.263, H.264 VP8 Voice codecs G.7xx series of codecs, and then some G.711, iLBC, iSAC

6. Opportunities for short term ventures WebRTC-SIP gateway: WebRTC does not specify signaling. The rest of the VoIP world and through it the can be reached by SIP. To bridge from PSTN and VoIP to the new world of WebRTC, there will be clear demand for SIP-to-WebRTC gateways. Examples: incoming: call centers, help desks outgoing: browser2phone Callcenter agent software: Throw out your softphones and hardphones in the Callcenter. Clients on a PC can video-talk. Phone clients just voice. Great savings in CAPEX! Incubation: now! is the time to incubate or invest. The technology is not hard to handle (for the savvy crowd) and the business opportunities are golden.. 6

7. Use Cases: by nature this is a pure OTT proposition Video conferencing: Video in browser Browser 2 phone Web-based help desk communication: http://www.junghanns.net/de/duoBRI2_produkt.html Integration of Call Center and CRM: coming soon Social Communication: coming soon 7