End of the world presentation
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![Page 1: End of the world presentation](https://reader030.fdocuments.in/reader030/viewer/2022032700/55d4b2fdbb61eb05758b471d/html5/thumbnails/1.jpg)
WebRTCThe End Of The World (As We Know It)
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WelcomeTo The Beginning Of The Post-Telephony Era
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I’m SteveSteve Sokol, Entrepreneur In Residence /
Director of Strategic Programsat Digium
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What is WebRTC?
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Photo Credits: Tom Keating - TMC.net, Eric Hernaez - Netsapiens
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How does it work?
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WebRTC leverages existing VoIPtechnologies
WebRTC exposes communicationsdevelopment to the 20M web developers inthe world
WebRTC sets rules for media, leavessignaling up to the application developer
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Get media streams from camera, mic
Create an “offer” session description
Send the offer to the far-end party
Receive an “answer” session descriptionfrom the far-end party
Discover a path that works by testing allpaths
Send media to the far-end party
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WebRTC Call In A Nutshell
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Web Server
Web Browser Web Browser
SRTP Media Streams
Offe
r Sign
aling
(SD
P)O
ffer Signaling (SDP)
Answ
er S
ignali
ng (S
DP)
Answer Signaling (SDP)
HTTP
or W
ebSo
cket HTTP or W
ebSocket
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Web Server
Web Browser
SRTP Media Streams
Signa
ling
Signaling
Media ServerGateway
PBX
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New JavaScript APIs
Media Capture
Peer-To-Peer Networking
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Creating A Connection
Built-In NAT Traversal using ICE
STUN - Discover network details
TURN - Relay as last resort
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Encoding Media
Audio Codecs
Mandatory: Opus, G.711
Optional Codecs
Video Codecs
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Encrypting Media
Mandatory
Secure Realtime Protocol (SRTP)
SDES vs. DTLS-SRTP Key Brokering
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No mandatory protocol or mechanism
Can be done using SIP or Jingle usingJavaScript libraries
Can be done better using other methods:
WebSockets or XMLHttpRequest transport
Simple JSON signaling
Use a protocol that suits your use caseperfectly, not a protocol built to handle alluse cases adequately
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What About Signaling?
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No mandatory signaling protocol is aGOOD THING™
Gives developers absolute control overthe user experience
Avoids the tendency to rebuild the PSTN
Avoids the “federation” issue
Allows for identity to be more than anumber
It’s The Web (Stupid?)
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URL-Based Calling
http://www.digium.com/contact/sales
http://www.digium.com/contact/ssokol
Directory-Based Calling
Linked-In
Corporate LDAP
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“Inside” users will use a web-based ormobile client
“Outside” users will use portal pages torequest access to various resources
People
Departments
Expert Support
Identity can be from email, Facebook,
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You will still need a communicationssystem or a communications service
You (eventually) may not need a “phonecompany”
Prediction: wired and wireless carrierswill become glorified ISPs within the decade
WebRTC will make rich communicationsa 100% “OTT” business
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So, Is It Ready To Use?
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Yes and no...
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Implementations in Chrome, Mozilla
Not currently interoperable
Great for “controlled environments”
Not yet ready for use by “normal” users
Will be ready by the end of 2013
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Challenges
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Mobile Deployments
Large-Scale Multi-Party
Legacy Integration
Codec Selection
Fragmentation (Microsoft’s CU-RTC-Web)
Encryption Keys
Audio Quality / Echo Cancellation
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Future features andenhancements...
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Peer-To-Peer Data
Real-Time Text (Captions)
Media Recording
Screen / Desktop / Tab Sharing
Statistics / Monitoring
Possibly low-level APIs
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A few use cases:
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Social Media
Call Center Agent Interface
Conferencing & Collaboration
Enhanced Customer Care
Distance Learning
In-Game Communications
Broadcasting
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Big Changes(Welcome To The Post-Telephony Era)
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Telephony has been holding backcommunications for the past decade.
SIP was hijacked: what started out as apeer-to-peer system was twisted into“PSTN-Over-IP”
Improvements and price reductions inbandwidth, mobile, web make a real changepossible
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Fully Unified Communications
Integration of communications directlyinto business and social applications
Communications as a feature orfunction rather than as a service
Customized User Experience
Excellent Privacy / Security
Significant Cost Reduction
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Asterisk And WebRTC
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Asterisk 11 added ICE, STUN, TURNsupport, WebSocket transport for SIPchannel and other tweaks
You can now create web endpoints usingAsterisk and a JavaScript SIP library
SIPML5
JS-SIP
Asterisk can bridge between WebRTCand legacy communications technologies
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Demo Time!
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Future versions of Asterisk will do more:
Recording and playback of audio andvideo
Interfaces for additional / customsignaling protocols
Interactive voice and video applications
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