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:
Integration of Call Center and CRM: coming soon
Social Communication: coming soon
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