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April 22nd 2013 Pre-conference Workshop for the IMS World Forum Alan Quayle alan@alanquayle.com www.alanquayle.com/blog Jose de Castro jdecastro@voxeolabs.com www.voxeolabs.com
4/21/2013
Objectives
Bring together deep technical and deep business thought leadership on WebRTC with Jose de Castro, Alan Quayle, and many of the audience to
Provide attendees with a series of WebRTC demonstrations, to share their experiences on implementing WebRTC, and provide ample networking opportunities at the end of the workshop to discuss and consolidate what has been learned through the day.
4/21/2013
Structure (1 of 6)
Registration 09:30 - Introduction to WebRTC and Initial Market Review
o
o o o o o o
Standardization process Current status Battles and likely outcomes IETF and RTCWEB documents
Structure (2 of 6)
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Protocols
WebRTC triangle / trapezoid SIP, Jingle and the PSTN.
13:00-14:00 Lunch 14:00 What WebRTC means to Service Providers and IMS:
o o o o o
Extending enhanced communications services to web browsers Impact on OTT (Over The Top) and existing voice, messaging, video and VAS Impact of device compliance Customer experiences and behaviors Revenue, churn and relevance impacts
Impact on Unified Communication and the Contact Center Impact on company's website
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Demo Time will be divided into 2 sessions, its aim is to be informal and provide ample networking opportunities for attendees to consolidate their learning from the workshop:
Demo one-on-one: attendees can chat one-on-one with the demo presenters, notionally 30 minutes but can run on into discussions at the bar through the evening.
Embed a 'Call' button into the website. Visitors can click that button and the call is
forwarded to the website operator's preferred land-line or mobile phone. All that is
required is a website; all the visitors need is a browser and microphone.
Voxeo Labs (Ameche (new IMS/Web services), Tropo (leading call control API), Phono (Web comms innovation)). They will demo Phonos three types of
identity:
o
Anonymous Identity: user lands on web site and is able to call directly into the contact center Web Identity: use your web identity (twitter, foursquare, etc) to call each other.
Telco Identity: Phono sessions can attach to the telco network and assume the real
identity (phone number) of the subscriber, allowing calls to be routed to both the mobile and the browser simultaneously.
Telestax
o
Provides a complete stack from the client-side with Javascript JAIN SIP JS and WebRTC
as well as the server side with our SIP Over WebSockets. The demo will be a WebRTC video conferencing and IM.
Demonstration of RCS messaging and WebRTC to access to media components of devices to revamp the value of PSTN (and also mobile) lines. Shows how Unified Communications could be built just a mash-up of standards and APIs.
Quobis
o
Their approach to WebRTC is based on QoffeeSIP, a complete open source Javascript SIP stack that can be used in a website to exploit all the multimedia capabilities of WebRTC technology. Thanks to QoffeeSIP they have developed a corporate WebRTC webphone that can interop with different network devices; this webphone is going to be released at IMS World Forum event.
Allows providers of fixed, mobile and next generation VoIP services to deliver audio
conferencing as a direct, branded service. Hosted within your IP network on your servers, Drum audio conferencing is a standalone software solution with an integrated media server.
Video chat with fun video effects, take screenshots of calls, share them with friends or social networks. Bistri runs in the browser, so there's no need to install additional software or plugins.
apidaze.io
o
Is a cloud communications API for developers with tools for building web or mobile communication services, with a special focus on WebRTC. The demo will show how a web developer can easily use the regular WebRTC API to place calls to external numbers and audio conference rooms accessible from the PSTN too, using a simple raw WebSocket connection that carries JSON text.
Embedding Communications
Everywhere!
Opus, VP8
Codec Wars
Regardless IE Matters
Business Technology
Supporting Devices Ecosystem Support Customer Needs Interoperability Use Cases IPR
Business Technology
Latency Efficiency Resilience Performance Implementation Complexity
Given the ability to deliver a royalty-free platform with no compromises on quality, we see no reason to include mandatory royaltybearing codecs.
H.264 support is a requirement in some regulatory frameworks, such as emergency services. AMR narrow-band is playing a key role in mobile telephony and has a huge footprint.
G.711 is universal, unencumbered, and widely implemented. A mandate for Opus will limit initial RTCWeb clients to use software-based codecs
We would like to recommend AMR-WB and EVS, since we expect them to be available in mobile chipsets. 30
Codec Wars
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 latency and high quality for human voice and music. Specially designed for real time communications.
Cost
Delay
Third Parties
33
VP8 VP9
H.264 H.265
Million
3000 2500
2000 1500 1000 500
Source: Disruptive Analysis WebRTC Strategy Report, Feb 2013 Definitions & methodology in report - See disruptivewireless.blogspot.com for details
Copyright Disruptive Analysis Ltd 2013 Feb 2013
The WebRTC Train has left the station and it isnt going to wait for Telecom
WebRTC Triangle
Web Server
(Application)
The wheels!
Browser L
(Running HTML5 Application from Web Server)
Both browsers running the same web application from web server Peer Connection media session is established between them Signaling is not standardized, could be SIP, Jingle, proprietary. Uses HTTP or WebSockets for transport
Intro to WebRTC February 2013
40
The Beauty and Value of WebRTC is when we mash it up with other stuff
NetHead
CustHead
Impact of WebRTC?
Voice becomes just like all your other communications: organized into your preferred social or office tools.
For all the OTT (Over The Top) applications, they can now use their "directory service" i.e. your list of contacts also using their service to enable
need only break out to PSTN when the other person is not data connected,
or the call quality is too low due to their internet connection. PSTN becomes the communications path of last resort!
Impact of WebRTC?
The company's website now becomes its call center front end. A weblog becomes your personal communications assistant.
o
Communication service aggregators save customers running multiple clients on their phone, that would run in the cloud and be controlled from the
browser.
Click to call doesn't require an operator's voice network, just access to the internet.
Impact of WebRTC?
VAS (Value Added Services) leaves telco. Any web developer can create value and solve problems for customers, it the customer who will decide, and those developers
Advertising finally enters the communications space, opening up business model innovation. New CRM (Customer Relationship Management) methods: click from email, from webpage, from app, from TV. The ability to communicate becomes embedded in most transactions.
QoS (Quality of Service) remains an issue, but for the people using Vonage and Skype
Gaming becomes interesting as all the devices become controllers using gesture controls as well as the more traditional methods for network-based games.
Opportunities
o o o
Key Points
Voice traffic is going to be through the web
Telephone number is not important unless operators pull their finger out!
Lower communication costs Lower IT costs as fewer clients to maintain? Better home and mobile comms that are integrated with corporate systems New customer communications options How to integrate What needs to change
Clientless, plugin-less browser audio and video for realtime communications means were not dependent on browser software suppliers
Open-source codec ICE/STUN NAT transversal does not work all the time SRTP (Secure Real-time Transport Protocol) configuration and SSL certification issues
Understanding Old-IT
Hardware endpoints with hard to upgrade software
Telepresence Network
IT Generation Gap
RTP/RTCP Separated
NO STUN / ICE!!!!
H.26x
De-ICE
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Transcoding
Use Cases
Social Media and CRM Integration Video Conferencing to any device Inbound Click to call a New channel
In Summary
Enterprise is interested in WebRTC BUT Some Enterprises change even slower than Telcos. If Telcos dont help them, they will go to other service providers
Gateway as a Service
o o
http://webrtcbook.com
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DEMO TIME