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      © mm1 Consulting & Management, Stuttgart/Aichwald


mm1 Consulting & Management
WebRTC: An introduction

Bonn, May 2012
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
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
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
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



                                                         5
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
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




                                                       7

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WebRTC: an introduction

  • 1. Hier bitte vollflächig Titelbild einfügen Bild immer bis zu den ODER Kanten führen Diesen Text und Begrenzungslinie unten mit einem weissen Kasten überdecken. Titel: Zweite Zeile Orange+ fett formatieren! © 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. 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.. 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 5
  • 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:  Integration of Call Center and CRM: coming soon  Social Communication: coming soon 7