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Introduction to
WebRTC
November 20th, 2012
In a few words


 WebRTC (Real Time Communications) will be the next big thing in terms of
 unified communications during the next years. Web browser will be able to
 manage voice and video in a native way.

 During 60 minutes, Elías Pérez, Antón Román and Iago Soto will talk about
 their vision about this technology and how is the best way to implement
 WebRTC in the legacy and VOIP networks of corporations and telcos.




 Elías Pérez Carrera        Iago Soto Mata           Antón Román
            CEO                     CMO                     CTO
         @epcarrera              @iagosoto              @antonroman
  elias.perez@quobis.com   iago.soto@quobis.com   anton.roman@quobis.com
Agenda




   - Introduction to HTML5 and WebRTC
   - Network architecture
   - Tech challenges
   - Application cases
   - Demos
   - Identity Management
   - Questions and answers
HTML5 and WebRTC. Short intro


                  ... is the fifth version of the HTML language, that
                  offers new capabilities to create web pages,
                  specially in terms of dynamic elements.

                  Defines WebSockets as the new transport
                  protocol between browser and servers

                  It is promoted by W3C




                   ... is an opensource project that makes possible
                   to manage multimedia communications in the
                   web browsers, using simples API's in Javascript,
                   in a native way.

                   It is promoted by the team of Google Chrome
HTML5 and WebRTC. Standards


 Two main players:

  ●     IETF RTCWeb WG ( Internet world)
  ●     W3C WebRTC WG (web world)
 Drafts:

 WebRTC 1.0 Real-time Communication Between Browsers
 WebRTC MediaCapture
 The WebSocket API
 draft-ietf-rtcweb-data-channel
 draft-ietf-rtcweb-jsep
 draft-ietf-rtcweb-rtp-usage

 ... etc ...




 Just a few days ago !!!
HTML5 and WebRTC. Increasing interest
HTML5 and WebRTC. Advantages


 Opensystems, with no propietary implementations



 ¡No plugins!




  Multi-platform... and multi-device!
HTML5 and WebRTC. Application cases


Software is going to migrate
to WEbRTC like collab apps
HTML5 and WebRTC. Application cases




And Click2call will be a trend in
CRM or marketing websites
Architectures. Browser to browser



 Any browser will have
 capabilities to call any
 other with RTC.
Architectures. Multiconference
Architectures. SIP interconnection




   SIP to WebRTC gateway will be implemented for interconnection.
Architectures. PABX interconnection
Tech challeges. Codecs


  G711a/u (RFC 3551): supported by all the devices. Needs
  to use a lot of bandwidth.

  DTMF tones (RFC 4733, updates RFC 2833): needed for
  interactions with several systems (for instance IVRs).

  Opus (RFC 6716): bitrate variable, low latence and high
  quality for human voice and music. Specially designed for
  real time communications.
   In order to interact with VoIP systems, in several scenarios,
   it will be needed transcoding or interworking of DTMFs
   (RFC 4733-> INFO, RFC 4733-> in-band, etc).

   And this only for audio...
             ... battle in vídeo VP8 vs H264
Tech challeges. SRTP vs DTLS-SRTP


 VoIP devices implement
 normally RTP encryption
 using SRTP. They share
 the key in the SDP
 protocol.




   DTLS-SRTP implements a new method to manage the
   SRTP key. This is done using DTLS, a version of TLS
   based on datagrams.
Tech challeges. NAT. STUN and ICE

Browsers are allways behind NAT and Firewalls, so it's
difficult to exchange packets in real time.

The user want a seamless communication, with no problems
(similar to Skype).

It is needed to found a "way" for RTP/RTCP traffic,
independent to the architecture..

ICE, makes possible (using STUN and TURN, protocols) to
discover y to choose the addresses that are going to be used
to exchange packets.

ICE was a difficult protocol to implemnt and had to be defined
during some years.
Tech challeges. Media negotiation

Uses SDP for negotiating media.
Classic negotiation offer-answer

There is a draft IETF:
  SDP for the WebRTC / draft-nandakumar-rtcweb-sdp-00


Example SDP:
 | v=0
 | o=bob 16833 0 IN IP4 0.0.0.0
 | s=
 | t=0 0
 | a=ice-ufrag:c300d85b
 | a=ice-pwd:de4e99bd291c325921d5d47efbabd9
 | a2
 | a=fingerprint:sha-1
 | 99:41:49:83:4a:97:0e:1f:ef:6d:f7:c9:c7:7
 | 0:9d:1f:66:79:a8:07               |
 | m=audio 49203 RTP/AVP 109
Tech challeges. Signalling. Options

WebRTC does not define how to manage signalling

There are some possibilities
 ● XMPP / Jingle
 ● SIP
 ● Protocol ad-hoc that manages SDP's


Quobis chooses SIP:
 ● "Standard": NGN networks are supported by SIP
 ● "Interoperable": it's possible to connect to everything!!!
 ● "Powerful": allows to build apps over it


Exisiting SIP implementations:
 ● sipML5: first implementation available
 ● JsSIP: authors of the draft-ibc-sipcore-sip-websocket
Introduction to QoffeeSIP


 Complete implementation of SIP
 protocol using Javascript

 It runs directly in the browser

 Focused on developers,
 written using CoffeeScript

 Easy to extend.

 Light application (5 KB)

 Will be opensource in the nexts
 weeks.
Introduction to QoffeeSIP




                                     +
                                    SIP
                                     =
                            Interconnection of browsers
                              with SIP legacy devices.
Tech challeges. Adoption. Browsers

             Really involved in WebRTC

             First stable versions (for desktop)

             Implements PeerConnection, GetUserMedia, etc..



             Involved, working more slowly than Chrome

                              Implements part of the API

                        Iniciative Firefox OS for mobiles


              Involved in WebRTC

              First stable versions (for desktop)

              Implements PeerConnection, GetUserMedia, etc..
Tech challeges. Adoption. Browsers


           Unknown roadmap, patents involved

           Important for smartphones and tablets



                                         Involved in WebRTC

     Suggestion and proposal that do not fit with the standard

      There is a plugin developed by Google (Chrome Frame)



              Developed by Ericsson labs

              First browser with WebRTC in mobiles (without
              Websockets)

              Beta version for developers.
Tech challeges. Security.


Media access: must be allowed by the user

Models of allowance:

 ● Access to cam and mic in a unique session
 ● continuous access to cam and mic
 ● Accesses based on users:
    ○ Allow calls just to an user
    ○ Allow calls to known users

LED to show that camera o mic are been used

Automatic off in case of not attendance (change of window)

Defined in draft-ietf-rtcweb-security
Tech challeges. Monitoring


 It is possible to monitor all the
 traffic, similar to standard SIP.


 Similar to SIP over TLS, if WSS is
 used (secure Websockets)
 monitorization should be done at
 the edges (most usually in the
 server).


 We have modified the monitoring
 tools in order to see websocket
 traffic.
Case studies. Applications




               Telcos         Corporaciones




          SP / Social media    Fabricantes
Case studies. Applications




      Telcos




       ●   Webphone for existing customers
       ●   Integration in NGN and IMS
       ●   Inbound channel for corporate
           customers
       ●   Alternative to calls from mobiles
Case studies. Applications




                  Corporations




       ●   Inbound Click to call. New channel.
       ●   Can be deploy by ourtsourcers
       ●   Calls in internal directories
       ●   Multiconferences.
       ●   Integration with legacy systems
Case studies. Applications




                             SP / Social media




       ●   Social networks -> next MVNOs
       ●   New services for webs managing IDs
       ●   ISPs will offer UC modules/widgets
       ●   Entertainment: chats, etc...
Case studies. Applications




                                         Manufacturers




       ●   Online gaming
       ●   Set-top-boxes and web TV
       ●   Gateways WebRTC to IPBX
       ●   Mobile device manufacturers
Demo. TalkSetup




       Join a demo of WebRTC at
         http://TalkSetup.quobis.com
Demo. TalkSetup backend




     Easy to manage WebRTC calls and users.
Tech challenges. Identity management




  Makes possible to be sure of the
  identity using a third
  party




                         Adds a second factor of authentications because we
                         validate the device (smartphone or PC) and the
                         credentials are introduced ciphered in a SIP
                         signalling packet.
Tech challenges. Identity management
Tech challenges. Identity management
Tech challenges. Identity management




  Agents can be sure
  of the identity of
  the person who is
  calling.
Demo. IdentityCall




      Demo using national electronic ID card.
What we have learned



    ● Voice traffic is going to be thru web
    ● Browsers are the new endpoints
    ● A website of a company can be the call
      center
    ● Telephone number is not important
    ● Security, identity a privacy is very
      important
    ● New business opportunities
Any questions?


Elías Pérez Carrera       Iago Soto Mata           Antón Román
          CEO                     CMO                     CTO
       @epcarrera              @iagosoto              @antonroman
elias.perez@quobis.com   iago.soto@quobis.com   anton.roman@quobis.com

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Webinar WebRTC HTML5 (english)

  • 2. In a few words WebRTC (Real Time Communications) will be the next big thing in terms of unified communications during the next years. Web browser will be able to manage voice and video in a native way. During 60 minutes, Elías Pérez, Antón Román and Iago Soto will talk about their vision about this technology and how is the best way to implement WebRTC in the legacy and VOIP networks of corporations and telcos. Elías Pérez Carrera Iago Soto Mata Antón Román CEO CMO CTO @epcarrera @iagosoto @antonroman elias.perez@quobis.com iago.soto@quobis.com anton.roman@quobis.com
  • 3. Agenda - Introduction to HTML5 and WebRTC - Network architecture - Tech challenges - Application cases - Demos - Identity Management - Questions and answers
  • 4. HTML5 and WebRTC. Short intro ... is the fifth version of the HTML language, that offers new capabilities to create web pages, specially in terms of dynamic elements. Defines WebSockets as the new transport protocol between browser and servers It is promoted by W3C ... is an opensource project that makes possible to manage multimedia communications in the web browsers, using simples API's in Javascript, in a native way. It is promoted by the team of Google Chrome
  • 5. HTML5 and WebRTC. Standards Two main players: ● IETF RTCWeb WG ( Internet world) ● W3C WebRTC WG (web world) Drafts: WebRTC 1.0 Real-time Communication Between Browsers WebRTC MediaCapture The WebSocket API draft-ietf-rtcweb-data-channel draft-ietf-rtcweb-jsep draft-ietf-rtcweb-rtp-usage ... etc ... Just a few days ago !!!
  • 6. HTML5 and WebRTC. Increasing interest
  • 7. HTML5 and WebRTC. Advantages Opensystems, with no propietary implementations ¡No plugins! Multi-platform... and multi-device!
  • 8. HTML5 and WebRTC. Application cases Software is going to migrate to WEbRTC like collab apps
  • 9. HTML5 and WebRTC. Application cases And Click2call will be a trend in CRM or marketing websites
  • 10. Architectures. Browser to browser Any browser will have capabilities to call any other with RTC.
  • 12. Architectures. SIP interconnection SIP to WebRTC gateway will be implemented for interconnection.
  • 14. Tech challeges. Codecs G711a/u (RFC 3551): supported by all the devices. Needs to use a lot of bandwidth. DTMF tones (RFC 4733, updates RFC 2833): needed for interactions with several systems (for instance IVRs). Opus (RFC 6716): bitrate variable, low latence and high quality for human voice and music. Specially designed for real time communications. In order to interact with VoIP systems, in several scenarios, it will be needed transcoding or interworking of DTMFs (RFC 4733-> INFO, RFC 4733-> in-band, etc). And this only for audio... ... battle in vídeo VP8 vs H264
  • 15. Tech challeges. SRTP vs DTLS-SRTP VoIP devices implement normally RTP encryption using SRTP. They share the key in the SDP protocol. DTLS-SRTP implements a new method to manage the SRTP key. This is done using DTLS, a version of TLS based on datagrams.
  • 16. Tech challeges. NAT. STUN and ICE Browsers are allways behind NAT and Firewalls, so it's difficult to exchange packets in real time. The user want a seamless communication, with no problems (similar to Skype). It is needed to found a "way" for RTP/RTCP traffic, independent to the architecture.. ICE, makes possible (using STUN and TURN, protocols) to discover y to choose the addresses that are going to be used to exchange packets. ICE was a difficult protocol to implemnt and had to be defined during some years.
  • 17. Tech challeges. Media negotiation Uses SDP for negotiating media. Classic negotiation offer-answer There is a draft IETF: SDP for the WebRTC / draft-nandakumar-rtcweb-sdp-00 Example SDP: | v=0 | o=bob 16833 0 IN IP4 0.0.0.0 | s= | t=0 0 | a=ice-ufrag:c300d85b | a=ice-pwd:de4e99bd291c325921d5d47efbabd9 | a2 | a=fingerprint:sha-1 | 99:41:49:83:4a:97:0e:1f:ef:6d:f7:c9:c7:7 | 0:9d:1f:66:79:a8:07 | | m=audio 49203 RTP/AVP 109
  • 18. Tech challeges. Signalling. Options WebRTC does not define how to manage signalling There are some possibilities ● XMPP / Jingle ● SIP ● Protocol ad-hoc that manages SDP's Quobis chooses SIP: ● "Standard": NGN networks are supported by SIP ● "Interoperable": it's possible to connect to everything!!! ● "Powerful": allows to build apps over it Exisiting SIP implementations: ● sipML5: first implementation available ● JsSIP: authors of the draft-ibc-sipcore-sip-websocket
  • 19. Introduction to QoffeeSIP Complete implementation of SIP protocol using Javascript It runs directly in the browser Focused on developers, written using CoffeeScript Easy to extend. Light application (5 KB) Will be opensource in the nexts weeks.
  • 20. Introduction to QoffeeSIP + SIP = Interconnection of browsers with SIP legacy devices.
  • 21. Tech challeges. Adoption. Browsers Really involved in WebRTC First stable versions (for desktop) Implements PeerConnection, GetUserMedia, etc.. Involved, working more slowly than Chrome Implements part of the API Iniciative Firefox OS for mobiles Involved in WebRTC First stable versions (for desktop) Implements PeerConnection, GetUserMedia, etc..
  • 22. Tech challeges. Adoption. Browsers Unknown roadmap, patents involved Important for smartphones and tablets Involved in WebRTC Suggestion and proposal that do not fit with the standard There is a plugin developed by Google (Chrome Frame) Developed by Ericsson labs First browser with WebRTC in mobiles (without Websockets) Beta version for developers.
  • 23. Tech challeges. Security. Media access: must be allowed by the user Models of allowance: ● Access to cam and mic in a unique session ● continuous access to cam and mic ● Accesses based on users: ○ Allow calls just to an user ○ Allow calls to known users LED to show that camera o mic are been used Automatic off in case of not attendance (change of window) Defined in draft-ietf-rtcweb-security
  • 24. Tech challeges. Monitoring It is possible to monitor all the traffic, similar to standard SIP. Similar to SIP over TLS, if WSS is used (secure Websockets) monitorization should be done at the edges (most usually in the server). We have modified the monitoring tools in order to see websocket traffic.
  • 25. Case studies. Applications Telcos Corporaciones SP / Social media Fabricantes
  • 26. Case studies. Applications Telcos ● Webphone for existing customers ● Integration in NGN and IMS ● Inbound channel for corporate customers ● Alternative to calls from mobiles
  • 27. Case studies. Applications Corporations ● Inbound Click to call. New channel. ● Can be deploy by ourtsourcers ● Calls in internal directories ● Multiconferences. ● Integration with legacy systems
  • 28. Case studies. Applications SP / Social media ● Social networks -> next MVNOs ● New services for webs managing IDs ● ISPs will offer UC modules/widgets ● Entertainment: chats, etc...
  • 29. Case studies. Applications Manufacturers ● Online gaming ● Set-top-boxes and web TV ● Gateways WebRTC to IPBX ● Mobile device manufacturers
  • 30. Demo. TalkSetup Join a demo of WebRTC at http://TalkSetup.quobis.com
  • 31. Demo. TalkSetup backend Easy to manage WebRTC calls and users.
  • 32. Tech challenges. Identity management Makes possible to be sure of the identity using a third party Adds a second factor of authentications because we validate the device (smartphone or PC) and the credentials are introduced ciphered in a SIP signalling packet.
  • 35. Tech challenges. Identity management Agents can be sure of the identity of the person who is calling.
  • 36. Demo. IdentityCall Demo using national electronic ID card.
  • 37. What we have learned ● Voice traffic is going to be thru web ● Browsers are the new endpoints ● A website of a company can be the call center ● Telephone number is not important ● Security, identity a privacy is very important ● New business opportunities
  • 38. Any questions? Elías Pérez Carrera Iago Soto Mata Antón Román CEO CMO CTO @epcarrera @iagosoto @antonroman elias.perez@quobis.com iago.soto@quobis.com anton.roman@quobis.com