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© mm1 Consulting & Management, Stuttgart/Aichwald
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.
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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”
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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..
Classic VoIP WebRTC
SIP or H.323 in
Signaling Undefined
most cases
Media transport RTP/RTCP RTP/RTCP
SRTP in SIP,H.235
Security SRTP
in H.323
STUN/TURN/ICE in
STUN/TUR
NAT traversal SIP,H.450.x in
N/ICE
H.323
Video codecs H.263, H.264 VP8
G.7xx series of
G.711,
Voice codecs codecs, and then
iLBC, iSAC
some
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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..
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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
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