A brief presentation on WebRTC and Standards delivered in Istanbul, at TAD Summit in a dedicated WebRTC Workshop. Topics include current status of WebRTC standard, a look at WebRTC supported browser, both on desktop and mobile devices

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WebRTC Workshop
What is (and isn't WebRTC)
Stefano Gioia
Master Principal Solution Specialist November 2014

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Stefano Gioia, Master Principal Solutions Architect, Oracle
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Expert in Service Delivery Platforms
Focused on WebRTC in EMEA
Customer advocate on Sales team
SDP Lead at Oracle – API, SIP, IMS, LTE
SDP Lead at BEA – APIs, SIP, VoIP,
Based in Rome – Covers all of Europe
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What is (and isn't WebRTC)
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History of WebRTC
The WebRTC specification
Browser support
Mobile and WebRTC
VoIP versus WebRTC
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Once upon a time...
•Internet was without form and void. And dark.
•Publishing and finding content on the Internet was an ancient and dark matter, only know by powerful Jedi Master (aka geek)
•Suddenly everything changed
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Pictures from http://evolutionofweb.appspot.com/

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Forever
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Pictures from http://evolutionofweb.appspot.com/

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There was a time...
•Where we use to see video by using plug-ins and external software
•Instant Messaging was delivered using the same concepts and Voice = Telephony
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There was a time...
•Where we use to see video by using plug-ins and external software
•Instant Messaging was delivered using the same concepts and Voice = Telephony
•Suddenly everything changed
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WebRTC @ Glance
WebRTC offers real time communication natively from a web browser
61%
WebRTC capable browser * share, including Mobile, Tablet and desktop
API
WebRTC is a “Media Engine” with javascript API
WebRTC is a technology, NOT a solution AND open source projects too
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So, what is WebRTC - Technically Speaking?
•Collaborative W3C and IETF standardization
•A built-in application program interface that enables browser-to- browser applications for:
–voice calling
–video chat
–peer to peer of any data
•Media engine in the browser, accessed by JavaScript, downloaded from web-server
•No defined signaling mechanism
•No MTI coded media
•WebRTC is a Standard Specification and an opensource project
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WEB SERVER (WEBRTC CONTROL
PEER CONNECTION
(AUDIO, VIDEO, DATA)
WEB SERVER (WEBRTC CONTROL
PEER CONNECTION
(AUDIO, VIDEO, DATA)
GATEWAY

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Where we are now?
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Source: Crossing the chasm, by Geoffrey A. Moore

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Program Agenda
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History of WebRTC
The WebRTC specification
Browser support
Mobile and WebRTC
VoIP versus WebRTC
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WebRTC Babel Standard & Engagement
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The WebRTC Tree
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IMS, LTE
VoIP
WWW

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Introduce webRTC & Standard
•Standardization by:
–W3C: Web Real-Time Communications Working Group, is to define client-side APIs to enable Real-Time Communications in Web browsers
–IETF: RTCWeb is taking care of the underlying protocols
•Browser side:
–WebRTC Initiative from Google (mozilla)
–CU-RTC-Web proposed and pushed by Microsoft
–Evolution to WebRTC 1.1 / ORTC promoted by Google, Microsoft and others
–OpenWebRTC release by Ericsson
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Pictures from http://webrtchacks.com/signalling-options-for-webrtc-applications/

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I’m confused... What are we talking about?
•WebRTC WG
–Mission: to define client-side APIs to enable Real-Time Communications in Web browsers
•WebRTC 1.0 Spec
–First Public Working Draft: 27 October 2011
–Last Call Working Draft: re- scheduled for Q4 2014
–Candidate Recommendation: initially scheduled for Q4 2012, but delayed probably until Q2 2015
•RTCWeb
–Mission: to specify a set of protocols that will allow an implementation to communicate with another implementation using audio, video and data sent along the most direct possible path between the participants.
•RTC Spec
–Define MTI AudioCodes OPUS and G.711
–Codes negotiated with SDP O/A
–MTI Security on DTSL/SRTP
–STUN/TURN/ICE
–Datachannel
•ORTC
–Mission: to define Object-centric APIs (client-side at first) to enable Real-Time Communications in Web browsers, Mobile endpoints and Servers.
–It is also a mandate that Object RTC be compatible with WebRTC.
–Original positioned as an alternative to WebRTC Spec
•ORTC Spec
–Community Group published the ORTC API Public Draft on August 20, 2014
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And the Telco ....
•3GPP Spec TS 23.228
–Specifies SIP over WebSockets as signalling between the UE to the network based
–SDP offer/answer exchange is the mechanism used for media plane feature negotiation.
–The architecture does not support media multiplexing that is defined
–Options other than SoWS, such as XMPP RESTful based interface, etc. are allowed but not described.
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P C E F N A T I P - C A N WWSF W1 W2 UE WIC I / S - CSCF eIMS - AGW Iq Mw eP - CSCF H / V - PCRF Gx Rx W3 IMS - ALG WAF W4 W5

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Signalling – The Two Approach
SIP over WebSockets
JSON/XMPP over WebSockets
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Why JSON, XMPP, * and not SIP?
•Much more easy
•There are more developers
•Language of the web
•And it’s natively supported by browser
•Drives more SIP into the network
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Great, ORTC as WebRTC 1.1
•The good
–ORTC positioned as successor of existing webRTC 1.0 API not as an alternative
–The first ORTC Public Draft Specification has been published, authored by Hookflash, Microsoft, and Google. This specification extends WebRTC 1.0
–No webRTC 1.0 implementation from MS and Apple
–OPUS, G7x codec will be implemented
•The bad
–At least 6-12 months to standard approvals and fruition
–H264 the supported codec, interoperability issue
–Adaption is required for existing application
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Program Agenda
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History of WebRTC
The WebRTC specification
Browser support
Mobile and WebRTC
VoIP versus WebRTC
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WebRTC Browser Support: Desktop
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Source from http://caniuse.com/

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WebRTC Regional Browser Support: Desktop
63,05
98,59 48,93
48,46 51,88
76,65 50,75
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Europe
Antarctica
Asia
North America
Oceania
South America
Africa
Support
Support
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Source from http://caniuse.com/

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A new Sheriff in town
•August 20, 2014. Microsoft published a blog post publicly supporting the ORCT draft.

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Audio & Video Codecs War
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G.711, Opus
VP8, VP9
G.711, Opus
VP8, H.264
G.711, G.722, Opus
H.264
G.711, G.722, Opus
H.264, H.265
Source from http://iswebrtcreadyyet.com/

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Program Agenda
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History of WebRTC
The WebRTC specification
Browser support
Mobile and WebRTC
VoIP versus WebRTC
Oracle Confidential – Internal/Restricted/Highly Restricted
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WebRTC Browser Support: Mobile
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Source from http://caniuse.com/

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Consideration
•Lack of WebRTC Support
–SDK/library for Windows Phone and IOS devices
•Hardware support for Mobile Phone
–Software encoding & decoding will kill your battery for VP8
•New device will benefits from power processor
–Snapdragon 800 4
–nVidia Tegra 4
•Operators want more audio codecs to support LTE & Legacy interoperability
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Program Agenda
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History of WebRTC
The WebRTC specification
Browser support
Mobile and WebRTC
VoIP versus WebRTC
Oracle Confidential – Internal/Restricted/Highly Restricted
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VoIP and WebRTC Similarities & difference
•Use RTP and SDP Offer/Answer model
•Audio codec G.711
•Run over IP networks
•Transmission of communication datat between users in real time
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So what I can do?
•Your SIP client or infrastructure actually...
–Use SIP or H.323 as signalling
–Use plain RTP rather than SRTP
–Don’t know anythign about STUN, ICE and ICE
–Use different codecs
–Use different identity mechanism
–Security SDES/DTLS
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ignore?
Built your own?

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Maybe use a simple Gateways
•SIP over WebSocket Gateway
•Web API to SIP Gateway
•Media gateway
•X-Coding gateway
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What we learned?
Different flavour of WebRTC
Media & Signalling are not compatible with existing operator’s network
WebRTC API is here, don’t wait to start building services
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