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Tsahi Levent-Levi
tsahil@bloggeek.me
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Table of Contents
Table of Contents................................................................................................................................................... 1
List of Figures ......................................................................................................................................................... 4
List of Tables .......................................................................................................................................................... 4
Introduction........................................................................................................................................................... 5
Who Am I Anyway? ...................................................................................................................................................5
Why WebRTC for Business People? ...........................................................................................................................5
Why is this Report Open? ..........................................................................................................................................6
What is WebRTC? .................................................................................................................................................. 7
What WebRTC? .........................................................................................................................................................8
Free ...........................................................................................................................................................................9
VoIP Developers ......................................................................................................................................................10
ORTC and WebRTC ..................................................................................................................................................11
WebRTCs Job to be Done .................................................................................................................................... 12
Barriers of Entry for New Vendors ..........................................................................................................................12
Reducing End User Friction .....................................................................................................................................13
The Innovators Dilemma and WebRTC ..................................................................................................................13
Browser Support .................................................................................................................................................. 15
Interoperability between Chrome and Firefox ........................................................................................................17
Handling Safari and IE.............................................................................................................................................18
Dealing with Mobile ................................................................................................................................................19
WebRTC on Devices.................................................................................................................................................21
WebRTC API Components .................................................................................................................................... 22
GetUserMedia .........................................................................................................................................................22
PeerConnection .......................................................................................................................................................22
DataChannel ...........................................................................................................................................................23
Networking .......................................................................................................................................................... 24
P2P ..........................................................................................................................................................................25
Signaling .................................................................................................................................................................27
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Communities .........................................................................................................................................................100
Appendix B: Choosing a WebRTC API Platform report ....................................................................................... 101
Appendix C: Finding out More ........................................................................................................................... 102
Appendix D: About Kandy .................................................................................................................................. 103
Appendix E: Kandy Usage Scenarios ................................................................................................................... 104
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List of Figures
Figure 1: The innovators dilemma and WebRTC ......................................................... 14
Figure 1: Major milestones of WebRTC support in browsers ........................................ 16
Figure 2: Browser downloads on Android...................................................................... 19
Figure 3: Chat session between Web browsers with and without WebRTC .................. 25
Figure 4: WebRTC media traffic with and without a TURN server ................................ 26
Figure 5: Anatomy of VoIP - the 4 core components of any VoIP solution .................... 34
Figure 6: Developers interest in WebRTC over time ..................................................... 44
Figure 7: Social network interest in WebRTC over time ................................................ 45
Figure 8: Types of vendors in the WebRTC ecosystem ................................................ 46
Figure 9: Growth pattern in web APIs............................................................................ 53
Figure 10: The WebRTC value chain and the 2nd market .............................................. 54
List of Tables
Table 1: Browsers support for WebRTC........................................................................ 16
Table 2: Comparison of WebRTC to other prominent VoIP protocols (SIP and H.323) 39
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Introduction
Who Am I Anyway?
My name is Tsahi Levent-Levi. I am a developer at heart. I have been working in the
telecom and VoIP/UC industries of the past 15 years (and counting) in various roles: from
a developer, to project manager, product manager and CTO. Most of that time, I was
dealing with signaling and media products that were licensed to other developers who
built their own products with them. This gives me a broad view of the market and an
understanding of the challenges and opportunities that exist in the domain of VoIP.
I came across WebRTC when it was first announced by Google and saw the potential in
it. Since then, I have been watching the WebRTC space closely, consulting about it and
writing about it on my blog: BlogGeek.me. From a hobby it became a "profession".
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What is WebRTC?
WebRTC stands for Web Real-Time Communication. In essence, it is the fusion of two
separate branches in technology: VoIP and the web.
VoIP, short for Voice over IP, is a set of technologies and techniques that enable sending
media (usually voice and video) over an internet connection. Up until the introduction of
WebRTC, VoIP lived within its own ecosystem silo, next to the booming Internet we are
accessing daily via web browsers.
WebRTC comes to connect VoIP into our browsers, and by that, into websites and mobile
apps. It does that by offering a thin layer of Javascript APIs that are implemented by
modern web browsers, and are part of the HTML5 specification.
This means that now every web developer can add real time communication
capabilities to his website or web application.
It is free
It changes the definition of VoIP developers
Together, they lower the barrier of entry for communication services to a level that enables
use cases that were always required but never fulfilled by previous technologies.
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What WebRTC?
The term WebRTC in itself is a bit misleading, as it refers to two separate things at the
same time:
1. WebRTC the specification
2. WebRTC the open source project
Both notions for WebRTC are used interchangeably most of the time and it is important
you know the distinction between the two.
What goes on the wire what is sent through the network, so that two WebRTCcapable applications can interact with each other
What are the APIs that utilize the functionality itself the defined Java Script API
that is located on top of WebRTC
The standardization of the first version of WebRTC hasnt finalized yet, which detracts
incumbents and telecom vendors from adopting it. That said, there are still hundreds of
vendors who are using WebRTC today, ignoring the transient state WebRTC is in today.
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Free
WebRTC is free in every possible aspect.
1. It is packaged as an open source project and licensed under the permissive BSD
license. Past implementations of similar technologies were either provided under
a proprietary license that had to be purchased or under more restrictive open
source licenses which wasn't palatable for a lot of vendors.
2. It supports and promotes free codecs which don't require any royalty payments
for patent licenses. Codecs usually require multiple payments to use, starting
from a license fee of a vendor and royalties based on quantities sold. This poses
a financial barrier for adoption as well as a hassle when each quarter licensing
fees needs to be recalculated and paid for.
This allows developers to repurpose WebRTC and its components in every possible
mean:
Later chapters in the area of use cases will show how different vendors and entrepreneurs
are wielding the power of WebRTC.
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VoIP Developers
Developing VoIP capabilities is far from easy. It requires engineers in specific disciplines.
VoIP projects are usually built by teams with 10 developers or more, dealing with aspects
of codec integration, media engine development and fine tuning, operating system porting
and interoperability testing. While some of these capabilities can always be outsourced
or licensed, there are costs and time involved.
WebRTC is not just a specification. It was created from company acquisitions that Google
has made recently. As such, it isn't just a "pet project" of a couple of developers but a
commercial grade package.
With that in mind, you can see how WebRTC enables more developers to use VoIP
related capabilities, but it doesn't stop there. The most important aspect of it is that
addition of Javascript APIs and integration of WebRTC into the browser.
This changes the game for communication developers considerably:
It changes the type of communication developers from C/C++ and Java developers
to web and Javascript developers. This opens up the field to a considerably larger
audience of developers, but also changes the mindset of these developers. You
can think of it as a transition from waterfall development to agile development
techniques
It makes the browser the endpoint or engine that gets used. This reduces the
hassles of issues such as security patches, upgrades and interoperability
nonfunctional aspects of development that takes time and money to deal with
properly
The change in developers also means a change in focus from the need to focus on how
to implement a solution with high media quality, to a focus on the service being developed
and the user experience it provides.
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While the WebRTC open source project may not offer a nicely packaged mobile SDK, the code itself is
almost fully ready for both Android and iOS use already, with noticeable improvements with each new
release.
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Adding and acting from context coupled with the request to communicate right from
the website
Enabling users to stay in the website without going for their phone to dial for service
Adding capabilities to social interactions from within the same service, not losing
the users to other services such as Skype
Empowering contact center agents to work from homes and caf shops without
the need for any additional installations or setup
This reduction in friction has the potential of increasing the value of the service employing
WebRTC.
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Performance
Time
Figure 1: The innovators dilemma and WebRTC
Incumbents faced with WebRTC, wont see the big difference between it and their current
technology. This has the danger of making them complacent.
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Browser Support
WebRTC is an evolving standard. It started in 2011 and since then, it has progressed
faster than any earlier VoIP protocol both in its specification as well as in availability of
implementations. That said, there are differences in which parts of WebRTC browsers
implement, if at all, and availability of WebRTC in mobile browsers is patchy at best.
The diagram below gives a short historic view into the progress of WebRTC
implementation across the main browsers supporting it: Chrome, Firefox and Opera
2012
2013
Feb 2013: Interoperability
Initial interoperability between Chrome
and Firefox browsers achieved. This is still
early on in the process, so things still don't
work as expected, but this is an indication of
things to come
Source:
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2014
Feb 2014: Opera for Android Beta with WebRTC
First Android release for Opera supports WebRTC
Source:
As you can see, browsers are updated frequently, filling in gaps in the support of certain
WebRTC aspects. The table below provides the current state of affair:
Chrome
Firefox
IE
Safari
PC
Android
PC
Android
Peer Connection
Data Channel
The Opera browser for desktop and Android also has WebRTC support
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In the next subchapters I will indicate what solutions, if any, are available for developers.
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On mobile, browsers are not the main channel for user consumption. Applications
are
Only Chrome and Firefox in their mobile versions support WebRTC, and that on
Android devices only no iOS devices are supported at this point
For most of the use cases on mobile, the requirement will be to wrap WebRTC
inside an application instead of using the browser as the point of access.
Downloads in millions
If you look at the download numbers of these two browsers and compare them to the
number of activations of Android devices and other downloadable mobile browsers, the
following picture emerges:
1200
Browsers
Other apps
1000
800
600
400
200
Maxthon
Dolphin
UC Browser
Firefox
Opera
Opera Mini
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While browser downloads on Android have significantly improved in the past year, there
is still a long way to go. The figure above brings with it several insights:
That in itself isn't bad news, depending on the service you have in mind. Current behavior
patterns on mobile show that consumption of online services comes from apps and not
directly through the web browser: Flurry Analytics estimates that 86% of the time spent
on iOS and Android is on Apps, while only 14% is spent in web browsers. Most amount
of the Apps time is spent on Gaming, Facebook and Social Messaging.
An additional advantage of creating an application for mobile as opposed to using the
browser directly, is the opportunity to address challenges posed by the mobile
environment. These include network changes (switch between WiFi and cellular and no
coverage, camera rotation, interruptions from incoming native calls, etc.
Using the controlled environment of an application can assist in overcoming these issues,
but require the care and attention of the developer or the 3td party SDK used for the task.
Check how your service gets consumed on mobile. If users are expected to "bump" into
it when browsing the internet or by specifically going to your service via an app. Going via
an app means you can use WebRTC in its ported form, wrapped inside an application
something that several vendors have already done. Assuming you want WebRTC to be
embedded in your app, there are three different options open for you:
1. Port and integrate WebRTC on your own for the mobile devices you plan to
support. Vonage, for example, has taken this route
2. License a ported WebRTC stack from vendors offering such a service. There are
multiple WebRTC outsourcing outfits today, and some of them offer porting
services
3. Use a 3rd party WebRTC API platform that already have their offering ported to a
mobile SDK.
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WebRTC on Devices
An additional opportunity of utilizing WebRTC is in other devices set-top boxes, video
room systems and Chromecast-like dongles are some examples.
In this case, a device or component that needs to be able to process and send/receive
media can achieve that by a porting WebRTC and embedding it within the device.
There are two main reasons to select WebRTC for these devices:
1. WebRTC has a very permissive open source license, making it easy to use as
the basis of a generic media processing component. It has also been proven to
be portable numerous times already.
2. By using WebRTC embedded in the device makes it easy to interact with these
devices from other mobile devices, laptops and browsers.
While some may complain that WebRTC isn't a stable standard yet, using it may prove
the best approach for those who want to limit vendor lock-in for their media processing
components and systems.
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GetUserMedia
The Get User Media APIs enable capturing the input devices: camera and microphone,
giving access to them via JavaScript. To get over privacy issues, when these APIs are
called in a website, the user is asked for permission to give such access.
This API can be viewed as external to WebRTC in a way, it doesn't deal with
communication at all. That said, Get User Media opens up possibilities for a wide variety
of services from photo-booth experiences to capturing a user's photo when onboarding
a new service (this is what MailChimp does when you register a new account or edit your
profile).
PeerConnection
At the heart of WebRTC lies the Peer Connection. This is where most of WebRTC's VoIP
implementation lies.
The PeerConnection API holds inside it everything: SDP negotiation, media sending and
receiving, handling network issues such as packet losses, codec implementations, NAT
traversal, etc. The API itself wraps it all in a simple package that enables adding media
streams into a peer connection, negotiating capabilities and NAT traversal.
Most vendors focus on adding value by using the PeerConnection APIs.
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DataChannel
The Data Channel can be considered the "jack in a box" of WebRTC. It is a generic
interface that can carry any type of data media or otherwise between 2 WebRTC
devices.
While doing that in applications is rather straightforward, WebRTC offers a solid
infrastructure that usually isn't available for developers, along with the fact that it offers it
inside a web browser in a real P2P fashion:
1. Ability to send data either reliably or unreliably, giving the flexibility for developers
to choose what fits their needs without adding additional applicative layers
2. Same security and NAT traversal capabilities used for sending media is extended
to the data channel as well
3. P2P support within the web browser for any generic data
4. Adheres to the network characteristics by limiting or extending bandwidth to what
is available dynamically
These aspects make the Data Channel an unknown factor in what developers can do.
Some of the early implementations that have adopted the data channel include:
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Networking
WebRTC deals with networking much in the same way that other VoIP solutions treat
them. The main difference you will find is that WebRTC leaves a lot less options for
developers in certain areas; and in the case of WebRTC, fewer options means better
solutions.
VoIP standardization has suffered for over a decade from over-complexity. Put simply, if
something can be done with a specific VoIP protocol, it can probably be done in multiple
ways, making development, testing and interoperability a real headache.
WebRTC took the route of as little options as possible when it comes to issues of how to
implement the basics. This chapter covers several areas where such decisions were
made, providing a glimpse into the mindset behind WebRTC.
While WebRTC is based on existing specifications, it did divert from the tried and true
of VoIP. In some cases, it selected to support specifications that werent popular or in
wide use in the industry. In other cases, it tweaked and improved the existing
specifications to fit its own needs.
The topics covered here are:
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P2P the type and nature of the peer to peer that WebRTC provides
Signaling what role does signaling take in WebRTC
Firewall and NAT Traversal the mechanisms of NAT traversal supported by
WebRTC
Security what is the security paradigm employed by WebRTC
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P2P
WebRTC is said to be a P2P protocol, by which it means that 2 peers connected via
WebRTC get connected directly to each other. While that may well be true, there are
several aspects of WebRTC's P2P nature that need clarification:
Web Server
Web Server
Signaling
Signaling
Chat
Web Browser
Web Browser
Web Browser
with WebRTC
Chat
Web Browser
with WebRTC
Source:
Figure 4: Chat session between Web browsers with and without WebRTC
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Web Server
Web Server
Signaling
Signaling
Media
WebRTC
Client
Media
WebRTC
Client
WebRTC
Client
WebRTC
Client
Source:
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Signaling
Signaling was specifically left out of WebRTC. There are those that see this as a positive
decision while others believe that signaling needs to be added under the fold of WebRTC.
The reasons for not having signaling as part of WebRTC are various:
1. Adding signaling to the WebRTC specification would drag the process through
standardization there are more existing options for VoIP signaling than there is
for media processing
2. WebRTC can be hooked up to use different signaling protocols today. Selecting
one mechanism, either existing or new, would mean reducing the use cases and
deployments where WebRTC can bring value
3. VoIP signaling is good for VoIP. WebRTC adds a web component that was always
missing in VoIP, where a lot of services already have some kind of signaling. Being
able to accommodate these services made sense
The only component within WebRTC that does deal with signaling is SDP. SDP stands
for "Session Description Protocol" and is used for describing multimedia session
capabilities and negotiating them. It is a critical first step towards connecting media
streams.
SDP is used in SIP, and as with other VoIP related issues, there are those that believe it
isn't suitable for WebRTC while others see it as sufficient.
SDP can be considered as a "necessary evil" which will be part of WebRTC for years to
come until a better/different solution is negotiated in the standardization bodies.
A more detailed discussion on the various signaling options can be found in the next
chapter.
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At the end of the day, STUN and TURN servers need to be deployed; which is a hassle.
Developers use multiple solutions for such a case:
1. Decide to deploy only with STUN support, and use publicly known STUN server
addresses (hoping they will not be moved or removed)
2. Deploy with STUN and TURN, installing and maintaining such servers as part of
the operation
3. Use third party SaaS provider for NAT traversal, who provide paid STUN and
TURN servers access
It should be noted, that there are cases where none of the techniques used by WebRTC
can successfully get a session to connect. This usually happens when the endpoints in
question are within enterprises with strict security rules. There are multiple drafts
proposed to the IETF to address this issue mainly by tunneling media traffic over web
sockets.
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Security
Security is part and parcel of WebRTC. Where other VoIP protocols have taken the route
of allowing security as an optional feature (usually as an add-on years after the original
standard was specified), WebRTC has no means of sending media without having it
encrypted.
This approach reduces the number of options and decisions developers have to make
about the way they use WebRTC and results in a robust solution suitable for modern
communication.
There are several important aspects to remember when dealing with WebRTC security:
1. WebRTC uses SRTP and AES encryption for the media being sent. These are
both widely used and mature specifications
2. While media communication is secure with WebRTC, the signaling part isn't. It is
up to the developer doing the integration with a specific signaling solution to take
care of protecting that path it can be as "easy" as placing that communication
over HTTPS/TLS
3. WebRTC in the browser have an additional factor of security any security holes
found within WebRTC are handled by the browser vendors. While this reduces the
control of implementers, browser vendors have been very responsive to date in
closing known security holes much faster than vendors in other domains
4. With the current awareness around security and privacy due to the Snowden Affair
and talks about the NSA practices with both carriers and web vendors, WebRTC
is sometimes touted as a possible solution due to its secured nature. That is usually
overstated as WebRTC is a technology that can be used in decentralized
scenarios but also in centralized ones.
Never take security for granted, but keep in mind that WebRTC brings with it top notch
capabilities in security that are not readily available elsewhere.
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Requested Features
WebRTC is a relatively new technology. It has been with us for 2 years only and has
already amassed a nice number of vendors around it. There are a lot of gaps in this
technology for those who wish to employ it for various use cases. These can be seen as
hurdles to some and as opportunities to others.
In this chapter, I will outline the main pain points vendors are expressing today with
WebRTC.
Multipoint
Support for multipoint video calling is an issue for developers. For those in need for such
a feature in their use case, make sure to find a solid architectural solution.
Multipoint support means the ability to have more than 2 participants within the same
session. The common practice today with WebRTC services is to build video conferences
of up to 4 participants by directly connecting each browser to all other browsers. This
practice requires processing power and bandwidth in the endpoints, which means it
doesn't scale well.
The other option is to use a central routing/switching/transcoding unit in the backend that
takes care of all media mixing for the participants. Such solutions vary in size and price,
so if you need to go that route, start by writing down the use case and flow you need to
support before searching for a solution that fits.
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Recording
Many communication related use case require the ability to record the session.
While recording can be done on the browser side to some extent, its real value and
potential can only be realized when done on the server side. Most vendors and services
today don't offer a recording capability, but have expressed interest in adding one.
Recording in WebRTC on the server side can be done in one of two ways:
1. Route the session through a server instead of P2P and have the server side handle
the recording. This requires more processing power and bandwidth on the server
side to manage, but reduces the upload bandwidth as well as processing power
required from the client side
2. Have the client's JavaScript code send the media to an additional participant which
is the recording server. This makes it easier to decouple the recording capability
from the signaling and media paths of the sessions, but requires more upload
bandwidth as well as processing power from the client side, so it might not fit all
use cases and all geographies
It is important to remember that recording doesn't come alone: it also requires the ability
to archive and store the media sessions as well as retrieving them. The retrieval part can
be rather tricky, as it will depend on the devices and methods in which you plan on offering
playback capabilities for the sessions enabling retrieval on Internet Explorer or direct
streaming to mobile devices will require transcoding the sessions either in advance or on
demand.
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Interoperability
WebRTC can live in a flux, where a specific use case can live in its own island. In such
cases, there is no external ecosystem you need to connect with. In some other use cases,
connectivity to other domains is mandatory things like integrating with an existing PBX
system or to the global PSTN are examples of this.
Such a requirement around connectivity is usually referred to as interoperability.
Interoperability in WebRTC happens in three separate domains, each with its own
challenges:
1. Signaling adding signaling on top of WebRTC isn't always possible and has its
own nuances. SIP and H.323, for example, are challenging as WebRTC signaling
protocols:
a. Accommodating SIP isn't supposed to be hard, but it does have its nuances:
the SDP is managed slightly different, and many of the RFCs and features
that are mandated in the media streams for WebRTC are considered new
and optional for SIP. Bridging the gap requires careful planning and
awareness to the problem
b. H.323 cannot be glued on top of WebRTC within the browser itself. This
may be true with other signaling protocols as well. Providing connectivity
between WebRTC clients to an H.323 deployment is not trivial
2. Networking WebRTC comes with mandated security as well as support for the
latest RFCs (ability to send multiplex media on the same network port, etc.). It
means that existing and legacy deployments can't directly interoperate with
WebRTC without some mediation in-between.
3. Codecs Some of The codecs selected for WebRTC are different than those used
in other VoIP deployments. This means that connectivity can occur only by
transcoding the media between WebRTC and external networks.
These challenges are usually solved by mediation mechanisms, where the most common
ones used are gateways and session border controllers (SBC). There are numerous
vendors offering this equipment in different shapes and sizes, so finding one that fits your
needs without the hassles of developing it from scratch is possible.
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Infrastructure
Signaling
Media Processing
Codecs
Source:
Figure 6: Anatomy of VoIP - the 4 core components of any VoIP solution
This section will review each of these components and how WebRTC handles them.
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Codecs
A full HD video requires 1920x1080 pixels per frame with 30 frames per second. If each
pixel requires 24 bits of color, this gets you to about 186MB of data per second an
impossibly large amount of data even for 1Gb networks.
To that end, codecs (code-decode) are used. Their sole purpose is to encode
(=compress) video and voice signals prior to sending them and to decode (=decompress)
these signals once received. For VoIP, these codecs are lossy they lose information
when they encode to save on space, usually by making assumptions around the human
eye and the human ear and also by external restrictions such as the quality required or
the bandwidth available.
WebRTC comes with 2 mandatory codecs:
1. G.711, the baseline voice codec. Offers narrowband voice quality, without any
compression or resiliency as part of the codec. Ubiquitous in its availability in
existing products.
2. Opus, a top of the line wideband codec that made its first appearance in
WebRTC and is now trickling to other VoIP products as well.
There is an ongoing debate over the mandatory video codec in WebRTC.
Google has taken the stance of supporting VP8 and is working on adding VP9 to
Chrome. These are "proprietary" video codec of Google
Microsoft and Apple are gravitating towards H.264, the standard video codec used
everywhere today
Cisco has released an open H.264 implementation, paid for in terms of royalties.
It has aligned with Mozilla who are supporting both VP8 and H.264
The main reason for VP8 and VP9 royalty free codecs (as opposed to H.264 which is a
royalty bearing codec).
Each VoIP product and protocol makes its own selection of mandatory and optional
codecs, usually having multiple codecs supported to enhance the potential of
interoperability.
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Media processing
While codecs do a decent job, they are not enough. There are other aspects that need to
be taken care of when it comes to sending media across networks. The two main ones
are:
1. Networks aren't created equal, and what is being sent might not really be
received on the other end, so someone needs to handle these things
2. There are local transient issues, such as echo between the microphone and the
speakers of a device that needs to be cancelled, varying volumes from changing
position of the speakers, etc.
These can all be wrapped into what I call "media processing". To that end, vendors will
either self-develop or license "media engines" to do the work. Part of this module is
straightforward implementations while other are based on best practices, heuristics and
"black magic" that developers have placed into their code.
In the case of WebRTC, this is what that Google has acquired from GIPS and ended up
open sourcing as parts of what WebRTC is. There are developers who simply rip media
processing algorithms from WebRTC and integrate it into their own products where they
see fit.
There are three main approaches to WebRTC when it comes to media processing, where
different approaches may be selected when dealing with the client side part and the server
side part of media processing:
1. Use WebRTC as is, relying on whatever the web browser (and in effect Google at
this point in time) to deliver the quality necessary and handle the network
impairments on its own
2. Replace the innards of the media processing components of WebRTC to
provide a better experience in the given use case. This is done either on the server
side or in closed applications on mobile or desktop where WebRTC can be
controlled and modified. The things that people replace in these cases are:
a. Rate control mechanism, which handles the decision of how much
bandwidth is available at any given point in time
b. The audio algorithms things like echo cancellation, noise suppression and
audio gain control
3. Use WebRTC's media processing components in a totally different setting
treating it just like pieces of code to be repurposed elsewhere. More on this in the
ecosystem chapter
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Signaling
Now that we are able to send media from point A to point B, the question then becomes
how A knows about B and how do they signal to each other their intents. That is done by
using signaling protocols.
In the domain of VoIP, there are essentially 4 options:
1. H.323, an ITU protocol that is used mainly in video conferencing deployments
with dwindling support for it
2. SIP, the IETF equivalent of H.323. Used everywhere these days
3. XMPP, another IETF protocol that started with presence and IM focus but has its
own extensions for voice and video chat (Jingle)
4. Proprietary, where each vendor to its own. A good example to it is Skype
WebRTC decided not to deal with signaling at all and left that to implementers. The
selection of the signaling protocol used will affect the capabilities you will be able to
achieve and will usually be linked to the types of devices and ecosystems you wish to
connect.
Infrastructure
Infrastructure is where VoIP service diverges from each other and where real
differentiation happens. It is how VoIP clients get connected to back ends, what additional
capabilities are provided just because there's infrastructure involved.
What WebRTC did was commoditize codecs, media processing and signaling making
them all a lot less relevant and enabling vendors to focus on their infrastructure and use
cases. This is why there are so many new use cases that WebRTC vendors are
implementing that were technically possible before WebRTC but never touched they
required too much effort for the return they provide.
38
Tsahi Levent-Levi
Sponsored by
WebRTC
SIP
H.323
Market
Websites and
web apps
Enterprise video
conferencing
Media
SRTP
RTP or SRTP
RTP or SRTP
H.460.x
SDP
SDP
H.245
Voice codecs
G.711, Opus3
G.711, other
G.711, other
Video codecs
VP84, H.264
H.26x
H.26x
API
Java Script
Unspecified
Unspecified
NAT traversal
Media negotiation
Table 2: Comparison of WebRTC to other prominent VoIP protocols (SIP and H.323)
H.323 controls a lot of the enterprise video conferencing market deployments, although
its market share is probably diminishing; and SIP exist practically everywhere else,
including in IMS5 deployments.
Both of these protocols use RTP to carry their media and optionally use SRTP SRTP is
mandatory in WebRTC. The media transport protocol of choice for WebRTC. On the NAT
traversal and media negotiation front, WebRTC bears great resemblance to SIP while
H.323 has its own protocols.
In the voice and video codecs, WebRTC differs greatly from SIP and H.323. While SIP
and H.323 offer a wide variety of choice, that choice causes a lot of headache in
interoperability of high quality codecs. For WebRTC, Google took the route of selecting
specific codecs with the main focus being latest available technology coupled with royalty
Some browsers offer additional optional voice codecs as part of their WebRTC implementation
All browsers supporting WebRTC today use VP8. Firefox is in the process of adding H.264 support.
Microsoft announced support for only H.264 at this point
5 IMS stands for IP Multimedia Subsystem and is the standardized solution for carriers. Its VoIP protocol of
choice is SIP and you can find it in VoLTE, RCS and other IMS related specifications.
4
39
Tsahi Levent-Levi
Sponsored by
free business model. There is an ongoing debate in the industry whether a popular royalty
bearing codec (H.264) should be used instead of VP8. While that debate isnt over, the
majority of WebRTC services are using VP8 today.
On the API front, WebRTC offers a specific API that is used by web developers while SIP
and H.323 are provided with different APIs in different languages from the various
vendors.
H.323 and SIP don't live within the browser and using them requires applications to be
developed, compiled, packaged and delivered as part of the service. That said, SIP is
moving towards the web browser by introducing SIP over WebSockets; making it a
suitable companion for WebRTC.
While WebRTC doesn't compete with these protocols, and can even be used and
interoperate with them, it does pose a threat to them, simply because it doesn't mandate
the use of any of them. Many developers today opt for using WebRTC with their own
proprietary signaling implementation to reduce the complexities associated with SIP and
H.323 signaling. By doing so, they shun the traditional VoIP world and its use cases to
implement their narrow use case. This approach doesnt force sessions in WebRTC to
adhere to the decade old telephony system and enables anyone, with or without VoIP
experience, to use WebRTC with ease.
40
Tsahi Levent-Levi
Sponsored by
Flash
Flash was the only mechanism in the web browser where real time communication could
occur prior to WebRTC. Flash brought with it several problems:
1. For a long time it supported only H.263 video codec, which was inferior. The end
result was poor video quality
2. Flash only worked with a server for all media transmission. This proves to be a
barrier for many use cases and for experimentation by developers
3. It wasn't HTML. The tools required for development weren't as common place and
as accessible as WebRTC is today
4. Its echo canceller was bad, which resulted poor voice quality in conversations
conducted over the PC speakers
5. To a large extent, it was non-existent in smartphones and tablets. While that is due
to Apple, its implementation on Android wasn't good enough
Many of the Flash based companies are now migrating towards WebRTC either doing
it purely by switching the technology or by providing both options: using WebRTC when
possible and downgrading to Flash when WebRTC can't be used.
"Proprietary"
There are and will always be proprietary implementations. The most notable one today is
Skype. For the past decade, Skype has provided VoIP solutions based on a proprietary
signaling protocol. Even their selection of voice codecs wasn't reliant on best practices in
the industry.
This approach has worked well for Skype, enabling them to overcome technical
challenges that were debated in the standardization bodies. This gave them a head start
on most other competitors who relied on standardized SIP or H.323.
Opting for the proprietary route comes with its own set of challenges mainly the inability
to tap to the external ecosystems to license components while focusing on the core
business.
41
Tsahi Levent-Levi
Sponsored by
Vidyo
Vidyo is mentioned here due to its recent announcement with Google around adding its
SVC technology into VP9. VP9 is the predecessor of VP8. Google have already
integrated VP9 decoder into Chrome for YouTube videos. They plan on introducing VP9
to WebRTC during 2015.
Vidyo is based on its own technology, where SVC takes a key part. SVC stands for
Scalable Video Coding and it is a technique in which a video stream is encoded in layers
each layer adding more quality on top of the previous one. This layered approach allows
sending different layers to different participants, based on the needs of the conference
and the capabilities of the network and the devices. The end result is the ability to build
more efficient multipoint video bridges.
This kind of technology makes sense for use cases where there are more than 2
participants. Managing the backend in such scenarios isn't trivial and this is why Vidyo
went for the deal it hopes to bank on the ability to use WebRTC in the browser directly
for its customers and deployments without the need to install an application. For Google,
this means a better video codec and less server computations for its Google Hangouts
service.
42
Tsahi Levent-Levi
Sponsored by
WebRTC Hype
There is no hype when it comes to WebRTC.
Pundits have been debating this for quite some time now, with varying opinions. I believe
there isn't and wasn't any hype around WebRTC whatsoever.
When I started blogging about WebRTC about a year ago, I was alone in the field. Very
little information and thought leadership could be found about WebRTC on the web. Since
then, that has changed with individuals, vendors and media outlets now covering this
space.
Any apparent hype being felt can be attributed to many reasons:
1. WebRTC in a new technology. As such, the process of understanding it that
takes place these days in public on blogs look like hype for some: you will find few
observations these days about where SIP belongs within the domain of VoIP, but
a lot of writing about WebRTC's place in VoIP
2. WebRTC deals with VoIP and web. While VoIP belongs to a small set of
developers within the communication market, WebRTC opens up that market to a
distinctly larger set of developers. This increase in availability of communication
technology seem to some like hype
3. WebRTC reduces the barrier of entry. Use cases that had no ROI up until
WebRTC are now being introduced, enhancing the ways in which VoIP is use
In the past 8 months, I have collected data points indicating the interest around WebRTC.
43
Tsahi Levent-Levi
Sponsored by
3000
2500
2000
1500
github projects
1000
500
0
Jan-13
May-13
Aug-13
Nov-13
Mar-14
Jun-14
Sep-14
Dec-14
Source:
Figure 7: Developers interest in WebRTC over time
The graph above illustrates three domains that are prime destinations for developers
interested in WebRTC:
1. The Google group called discuss-webrtc
2. The number of github projects related to WebRTC
3. The number of questions around WebRTC on StackOverflow
44
Tsahi Levent-Levi
Sponsored by
Jul-14
Aug-14
Sep-14
The graph above details the number of LinkedIn profiles that have WebRTC somewhere
in their description. While this number is rather low, it is easy to see an upwards trend.
The results of these graph show a linear increase in interest this is an indication that we
are either before the real hype of WebRTC or that no hype will happen with WebRTC
at all.
45
Tsahi Levent-Levi
Sponsored by
2nd Market
Vendors
Repurpose
Tooling
WebRTC
Core
Before I go into detail about the categories there are a few things to keep in mind:
46
Tsahi Levent-Levi
Sponsored by
WebRTC Core
The supporting role at the heart of WebRTC belongs to vendors who contribute to it
directly. Here you will find Google and Mozilla, who are embedding WebRTC into their
browsers, but also companies like Vidyo who are adding SVC technologies into future
WebRTC releases.
The amount of vendors in this area is small, as they require extensive knowledge into the
inner-workings of WebRTC. This role comes with little upside in the way of direct revenue
but gives a lot of power in ensuring technical superiority versus other vendors:
Mozilla and Google may use their support of WebRTC to attract additional users
from Apple and Microsoft, increasing their market share
Vidyo may use their assistance to WebRTC's technology stack in maintaining their
first mover advantage in multipoint video conferencing capabilities
Ericsson with their latest openWebRTC stack, rivaling Googles own WebRTC
implementation
Cisco who open sourced their H.264 video codec in an effort to sway the decision
of WebRTCs mandatory video codec towards H.264
These vendors usually don't intend on banking from WebRTC directly and have indirect
business models for WebRTC; at times the business model isn't easily apparent.
Tooling
Vendors in this position offer tools that make development of WebRTC-based services
much easier. These vendors have varying business models focused around B2B, where
the target market is other vendors who wish to deploy communication services.
These vendors can be split into several domains:
47
API vendors
Services vendors
Infrastructure and SDK vendors
Projects vendors
Open source vendors
Tsahi Levent-Levi
Sponsored by
Services Vendors
There are pieces in WebRTC deployments that can be "outsourced" to others. These
pieces are offered as service, where the SLA, global footprint and focus of the vendor on
the specific service is the main offering.
In this domain, you can find XirSys as a notable NAT traversal providers, offering TURN
server hosting as a service to those who don't want to deal with media relay in WebRTC
and focus on their own service logic.
Another interesting set of vendors here are PubNub, Firebase and Pusher who offer low
latency messaging infrastructure for developers. All have added WebRTC into their
arsenal of tools; where they specialize in adding signaling to it.
The customers of the services vendors are usually those who are comfortable in
developing directly on top of WebRTC but view the hosting of WebRTC components as
less important to their core offering.
48
Tsahi Levent-Levi
Sponsored by
49
Tsahi Levent-Levi
Sponsored by
Projects Vendors
As with any vibrant community, there are those who offer their services to assist
companies in building whatever it is they require. WebRTC include large and small
software houses of this type ones that take on development tasks related to WebRTC.
Some of these vendors offer a breadth of capabilities coming either from a broad
outsourcing and system integration heritage, others are more focused into the VoIP and
video processing space and the rest are starting off with WebRTC or can be considered
as "heads for hire", usually focusing on projects to local vendors.
WebRTC being a new domain, there is a real need for such vendors as well as a rise in
the number of vendors who are publishing their expertise in WebRTC and even investing
in showcasing it in.
50
Tsahi Levent-Levi
Sponsored by
"bumping" into them while Googling for things they need to do and decide in an ad-hoc
basis if and how to adopt them. This practice needs to be handled with care, as the
licenses attached to these open source packages may be limiting the business models in
which the adopting vendor will be able to operate in the future.
Vendors
Vendors are those who end up offering WebRTC-based communication services to the
end users. Their market may be B2B or B2C, depending on who they serve.
It is hard to cluster these vendors together, as they cater different verticals and use cases.
It is also where business models diverge greatly.
Vendors in this category may use a Tooling vendor to rely on, or just build everything from
scratch on their own. Reasons for selecting different routes here depend on a large variety
of parameters, unique to each case.
The use cases chapter will try to take a closer look as some of the markets to which these
vendors belong to.
51
Tsahi Levent-Levi
Sponsored by
Repurpose
Repurposing WebRTC can be viewed as the gray market of WebRTC. There are active
vendors in this space, where most of them aren't widely known their use of WebRTC is
such that is left behind curtains.
WebRTC, the open source project, as opposed to the specification, is a state of the art
media engine. It is the result of GIPS' commercial offering and On2's proprietary video
codec, coupled with Google's ongoing investment in its quality.
To put things in perspective, when GIPS got acquired by Google, many vendors who were
GIPS' customers immediately started searching for alternatives, which were hard to find
at that level of quality and capability. Among GIPS customers you could have found IBM,
Google, Yahoo, WebEx, Nortel, AOL, Citrix, Avaya, Samsung and QQ.
When Google open sources WebRTC under a very permissive BSD license, they in effect
provided anyone who needed a media engine free access to a high quality solution only
thing missing was an SLA.
Vendors have taken note, and some have pried bits and pieces of WebRTC that fits their
needs and embedded them into their own products and services. This requires little
publication of the fact, so most of these activities are left unknown in the market. The
most known and notable such move was the one made by Vonage. Their story is outlined
in the use cases chapter.
52
Tsahi Levent-Levi
Sponsored by
2nd Market
The 2nd Market of WebRTC is something quite rare but its existence is important to note.
WebRTC fuses between VoIP and the web. This brings two important traits together:
1. VoIP gains an open and modern delivery system the web browser
2. The web gains P2P capabilities and real time communication
The web is based today on loosely coupled web sites and technologies that are mashed
up together to build new solutions and services. The ProgrammableWeb is an important
directory of web APIs used for mashup6 purposes. It documents APIs, mashups of them
and stories around them.
At the moment, there are over 10,000 documented APIs and 7,000 mashups on the
ProgrammableWeb:
A mashup is a web page or web application that uses content/functionality from multiple sources across
the web. When one web service offers open APIs, other services can consume that API (as well as other
APIs) to generate a new service offering
53
Tsahi Levent-Levi
Sponsored by
This trend of mashup and open API publication is also something you see with WebRTC,
and not only with the API vendors: many of the WebRTC services players are starting to
offer APIs of their own for various purposes:
1.
2.
3.
4.
To some extent, this creates a 2nd market of companies who integrate with these services
vendors and still see themselves as part of the WebRTC ecosystem.
The most distinct example of this is Intuitive Solutions, who are running a managed
contact center service for customers. Intuitive Solutions in this case, have even publicized
this initiative in a press release and see themselves as a WebRTC player to some extent.
Figure 11: The WebRTC value chain and the 2nd market
Intuitive Solutions run their service on top of LiveOps, which provides contact center SaaS
systems for its customers. LiveOps have built their own solution on top of Twilio, a
WebRTC API vendor.
54
Tsahi Levent-Levi
Sponsored by
Things to Remember
The vendors listed in this chapter were selected not due to their technical merit or
the type of support they provide these were not part of the analysis done. The
vendors listed here were selected as examples of the different services and
business models that vendors are offering in the WebRTC ecosystem.
Many of the vendors in this section have been interviewed for BlogGeek.me in the
past. You can find more about them here: http://bloggeek.me/webrtc-interviews/
Tooling
Tooling vendors are at the heart of the WebRTC ecosystem. They are the enablers filling
the gaps that the WebRTC specification doesn't solve this essentially mean a lot of
backend capabilities.
These vendors come in different shapes and sizes. I have selected 3 such vendors:
55
Digium the vendor behind the popular Asterisk SIP communication platform,
who has added WebRTC to Asterisk's capabilities
GENBANDs Kandy a Telco API vendor, enabling anyone to build a use case
on top of their platform instead of directly on top of WebRTC
XirSys offering a NAT traversal service in SaaS form
Tsahi Levent-Levi
Sponsored by
Digium
Digium offers a popular and widely used
open source communication framework
called Asterisk. Founded in 1999, Digium
have been in the VoIP arena a lot longer
than WebRTC.
Asterisk added support for WebRTC,
which enables anyone who need a PBX or
other SIP server to be able to connect calls
to it over WebRTC from a browser. This
support is being used by multiple software
vendors already.
Solution Type
PBX
Target Audience
Developers
Business Model
Dual License
Country
US
@digium
Media Types
Voice, Video
Use of WebRTC
Direct integration
Asterisk isn't alone in this domain. There are other such open source VoIP platforms that
have added or are in the process of adding WebRTC support.
The advantages of adopting an existing VoIP platform that added WebRTC are:
Easier integration with existing VoIP deployments. Main domains that are already
using this approach are contact centers and unified communication solutions
Rich framework and architecture, with an existing ecosystem of developers and
outsourcing vendors
Recently, Digium have introduced a new WebRTC API Platform called Respoke.
Things to Remember
56
Tsahi Levent-Levi
Sponsored by
GENBANDs Kandy
GENBAND is a vendor specializing in
carrier grade communication products.
In 2014, it launched a new Platform as a
Service called Kandy, which delivers its set
of capabilities via API and SDK to
developers, enterprises and carriers.
Kandy offers the ability to create real time
communication based services with little
knowledge of WebRTC or communication
related technologies. It does so by offering
REST APIs, SDKs for Web and mobile and
a cloud-based communication platform.
Solution Type
Telco API
Target Audience
Developers
Business Model
Subscription
Country
US
@KANDY_io
Media Types
Voice, Video
Use of WebRTC
Service Providers, who wish to deploy the solution within their own controlled
environment and use it to build their own services, or white label it altogether
Enterprises, where an SLA and the ability to customize and offer a solution on top
of an API platform is necessary at times
The long tail, where any developer can on board and start developing immediately
with little friction
The things Kandys team is most proud of is their ability to simplify the development effort
required, taking out of the equation a lot of the challenges associated with developing
WebRTC and communication related services for web and mobile.
Things to Remember
57
Many vendors are currently competing in this space, each with his own
differentiating business or technology factors
Kandy provides an end-to-end Communication-as-a-Service solution
This space is dynamic, with several players getting acquired and taken off the
market. The selection of a vendor to work with is an important and non-trivial one
Tsahi Levent-Levi
Sponsored by
XirSys
XirSys offers a SaaS for WebRTC NAT
Traversal. In many cases, developers don't
wish to deal with the hassles of installing,
setting up; integrating and maintaining
STUN and TURN with their WebRTC
service.
In such cases, they search for vendors who
can provide this solution under an SLA.
The advantages of outsourcing this to an
external vendor includes:
Solution Type
SaaS
Target Audience
Developers
Business Model
Subscription
Country
US
@XirSys
Media Types
Voice, Video
Use of WebRTC
Direct integration
A NAT Traversal SaaS vendor is also capable of providing additional insights to its users
via reporting and analytics areas where most developers will not invest until later into
the service lifecycle.
Things to Remember
58
SaaS can include other types of services. NAT traversal is the most obvious one,
but signaling, media processing and recording are all areas where developers
have indicated the need for assistance
When building a service, SaaS solutions can greatly reduce time to market with
the initial minimum viable product. In the longer run, it can simplify your work
Tsahi Levent-Levi
Sponsored by
Video Conferencing
Video Conferencing is a market that exists for over a decade. Its focus over the years has
always been Enterprise Video Conferencing, sold to multinational companies with large
IT budgets.
In recent years, companies have moved to offer Cloud based video conferencing
offerings, in an attempt to shift IT spending from CAPEX to OPEX and to start selling
video services instead. WebRTC is now part of this trend and can be viewed as an
accelerator of it.
Many of the initial WebRTC vendors tried to penetrate this market, with the main value
being a reduction in the end cost to the customer due to their lower development and
operating costs.
The vendors provided in this subchapter as example use cases were selected due to their
business models and geographical variety:
59
Tsahi Levent-Levi
Sponsored by
Pexip
Pexip can be viewed as the next
generation
of
enterprise
video
conferencing solutions. It comes with the
same world view of unified communication
solutions of multiple video participants in a
single meeting, but diverges from common
practices in the following ways:
Solution Type
Target Audience
SMB, Enterprise
Business Model
Subscription
Country
Norway
As with other cloud video conferencing platforms for the enterprise, Pexip markets its
solution as connecting to any system be it H.323, SIP or WebRTC coming from any
device.
Pexip is a Norwegian company, based out of the same talent pool that started off
Tandberg a video conferencing vendor that got acquired by Cisco a few years ago.
Unlike the other vendors in this section, Pexip does not offer a hosted service, but rather
licenses its software to run on-premise in its customer's data centers.
Things to Remember
Video conferencing in the enterprise is hard. A lot of issues there revolve around
interoperability and transcoding. Only engineers with such domain expertise can
build a solution for that market
For Pexip, WebRTC is just another device connecting into their network it isnt
at the heart of their architecture
60
Tsahi Levent-Levi
Sponsored by
Sqwiggle
Sqwiggle is trying to solve the collaboration
challenges of remote workers in the same
team. It does so by redefining video
conferencing in enterprises and making
use of ambient video technologies.
Sqwiggle places the whole team in a single
web page, where still images of the team
members get updated several times a
minute. This creates a type of presence
that bridges the distance within the team.
Solution Type
Video
Conferencing,
Collaboration
Target Audience
SMB
Business Model
Subscription
Country
US
@sqwiggleinc
Media Types
Voice, Video
Use of WebRTC
Direct integration
Things to Remember
61
Video communications in the enterprise comes in different shapes and sizes. The
fact that legacy vendors have focused on a specific use case doesnt mean other
use cases dont exist
Presence can be a lot more fluid than online, offline, busy and away. Sqwiggle
redefined it by making taking a picture of its users and making them available to
the team
Sqwiggle doesnt aim to connect to any existing video conferencing equipment. It
lives as a standalone solution and works well in that context
Tsahi Levent-Levi
Sponsored by
Veeting Rooms
Veeting Rooms provides a web meeting
service enabling multiple participants to
attend a video conference.
While many provide similar capabilities,
Veeting Rooms differentiates itself from
the rest of the pack by putting emphasis on
data privacy.
Everything in Veeting Rooms from the
data center used to hosting their service, to
the third party SaaS providers for analytics
are located within Switzerland, where
strict data privacy rules exist.
Solution Type
Target Audience
SMB, Enterprise
Business Model
Country
Switzerland
@VeetingCom
Media Types
Voice, Video
Use of WebRTC
Direct integration
Veeting Rooms have built their system in a way that enables them to white label and even
replicate the solution to another geography if necessary.
Things to Remember
62
The Enterprise Video Conferencing market competed in the past decade mainly
on port density, resolutions and video quality. With good enough solutions, it is
becoming hard to differentiate
Small startups like Veeting Rooms are finding gaps and needs in the market that
are ignored by the incumbents and offer tailored solutions. Data privacy is such a
gap for the competing cloud providers of video conferencing
As with the case of Sqwiggle, Veeting Rooms refrains from addressing the existing
video products, instead making use of WebRTC in the browser
Tsahi Levent-Levi
Sponsored by
Telecommunications
Carriers have their own interest in WebRTC. This stems from the basic notion that
WebRTC provides communication capabilities that can be either wielded by OTT
vendors, competing on voice and SMS services, or by the carriers themselves.
The domain of carriers is larger than what can fit in such a paper, with multiple directions
that can be taken and are taken today. In this section, several non-obvious approaches
of carriers into WebRTC are described:
63
Tsahi Levent-Levi
Sponsored by
appear.in
appear.in define themselves as a startup
within the incubator environment of
Telenor Digital. They offer a simple, adhoc, multipoint video conferencing service.
The service started as an interns project.
It gained traction and popularity, which
caused Telenor Digital to invest more
resources into it.
Today, appear.in offers a decent video
conferencing service that can work with up
to 8 participants, along with an iOS app.
Solution Type
Video Conferencing
Target Audience
Consumers
Business Model
Exploration
Country
Norway
@appear_in
Media Types
Voice, Video
Use of WebRTC
Things to Remember
64
There are similar services out there. This is the only one developed and managed
by a Telecom vendor
It is an experimental service for Telenor, as there is no business model apparent
at the moment behind it
Vendors (especially incumbents) need to tinker and play with WebRTC to
understand its capabilities, and decide from there where it fits
There is more to do with WebRTC in Telecom than simply make it an access point
to the IMS network
Small teams in large companies can develop great services if you let them
Tsahi Levent-Levi
Sponsored by
Comcast
Comcast is the largest cable provider in the
US. Its XFINITY X1 Platform is used by
many of its customers as their Pay TV
platform.
Recently, Comcast announced its plans of
adding video streaming capabilities which
will work similar to how Chromecast does:
it will enable their customers to live stream
personal video content from their mobile
devices to the television via the X1 set top
box.
Solution Type
Video streaming
Target Audience
Consumers
Business Model
Subscription
Country
US
@comcast
Media Types
Voice, Video
Use of WebRTC
Things to Remember
65
While the obvious addition of WebRTC to a set top box would have been video
calling, Comcast has taken the route of enabling better video streaming user
experience with WebRTC
There are many ways to use WebRTC besides enabling a peer to peer video call
WebRTC use cases are not limited to the browser or even to a mobile device. The
X1 platform is an embedded device that has WebRTC support to it
Telecom Carriers are large companies with multiple services. There are many
ways in which they can leverage WebRTC. Comcast seem to be doing it both as
a video streaming capability on their X1 platform and as an unannounced
multipoint video conferencing solution (based on their Jitsi Videobridge
investment)
Tsahi Levent-Levi
Sponsored by
Vonage
Vonage is a voice communication provider
that used WebRTC to build a mobile client
and to add video calling.
Vonage has been operating since 2001,
offering its ~2.3 million subscribers with
unlimited voice calling to over 60 countries
over its VoIP infrastructure.
Vonage was running predominantly on IP
phones over wireline connections, and
wanted to expand their offering in two new
trajectories:
Solution Type
Consumer VoIP
Target Audience
Consumers
Business Model
Subscription
Country
US
@vonage
Media Types
Voice, Video
Use of WebRTC
Porting, repurposing
1. Mobile introducing a BYOD mobile application that can run on Android and iOS
devices, enabling Vonage customers to use their service anywhere
2. Video add video calling capabilities on top of their extensive voice service
To do that, Vonage took the following steps:
This approach enabled Vonage to meet their requirements with little change to their core
infrastructure. Today, the Vonage application has been downloaded over 1 million times
from the app stores.
Things to Remember
66
Purists will not see this as a WebRTC initiative, as it treats WebRTC as any other
generic media engine
Such a development project was rather expensive at the time Vonage started
(2012), but costs a lot less than it used to prior to WebRTC
The fact that WebRTC can be plagiarized is one of its unique values
Tsahi Levent-Levi
Sponsored by
Amazon Mayday Amazon's new video support service. While its use of
WebRTC isn't confirmed, it may change the way contact centers are going to be
structured
FreeCRM a CRM service that adopted WebRTC to enable calling contacts right
from within the CRM service, without the need to pick up a phone and dial
Intuitive Solutions a Texas managed contact center company that changed the
way they make decisions by using WebRTC
Vacasa Rentals a vacations provider using WebRTC internally to run their day
to day operation
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Tsahi Levent-Levi
Sponsored by
Amazon Mayday
Amazon Mayday is a service offered by
Amazon for free to its Kindle Fire tablet and
Fire Phone customers. With it, a customer
can press a Mayday button on his device
and get connected with a service agent
within 10 seconds.
A lot has been written about this service on
the internet. The most interesting aspects
of this service are:
Solution Type
Contact Center
Target Audience
Consumers
Business Model
Differentiation
Country
US
@amazon
Media Types
Voice, Video
Use of WebRTC
Porting, repurposing
While the technical details were never provided, it is safe to assume that such a solution
is based on WebRTC: Amazon's Kindle Fire tablets and Fire Phones are based on the
Android operating system. WebRTC is developed by Google with support for Android, so
having something readily available that can be used and integrated without paying for it
seems like a good bet.
Things to Remember
68
The deep integration done by Amazon with the ability to maintain a live call while
using the tablet and sharing the screen across applications isn't trivial. It was made
possible by the fact that Amazon controls the chipset, operating system and
services on their tablets
This move may have a large impact on contact centers, customer support and
mobile device vendors alike. Since Amazons announcements, many companies
(startups and established) have been introducing their own angle to the Amazon
Mayday button
Tsahi Levent-Levi
Sponsored by
FreeCRM
FreeCRM offers a CRM system in a
freemium model. One of its paid features is
the ability to call contacts from the CRM
directly.
The contacts don't need to be using
WebRTC in any way they receive their
call on the regular telephony service
(PSTN).
Solution Type
CRM
Target Audience
SMB, Enterprise
Business Model
Subscription
Country
US
@FreeCRM
Media Types
Voice
Things to Remember
69
Seamless integration of calling capabilities into the contact center makes it easier
to handle and provides a single vendor solution
In this domain, PSTN connectivity is important
Such a solution can be achieved by self-development or by utilizing API vendors
Tsahi Levent-Levi
Sponsored by
Intuitive Solutions
Intuitive Solutions is a managed contact
center service located in Texas,
specializing in the pizza restaurant
business.
Intuitive Solutions was looking to grow, but
had several challenges:
1. Office space
2. VoIP software client for agents that
was hard to manage and maintain,
especially with remote workforce
3. 4 different vendors handling their
contact center
Solution Type
Target Audience
SMB, Enterprise
Business Model
Subscription
Country
US
None
Media Types
Voice
Use of WebRTC
3rd party
Intuitive Solutions searched for a single vendor to offer all of their IT needs, be it CRM or
contact center related. They found such a third party platform that provided them CRM
capabilities along with the ability to receive incoming calls using WebRTC from within the
browser.
This solution offered Intuitive Solutions the following benefits:
Hassle free deployment and management of VoIP clients, as they are now a part
of the Chrome browser
Ability for contact center agents to work from home or a coffee shop, freeing them
from the brick and mortar office space, and allowing more flexibility in work hours
Better decision making due to a clear view of their day to day operations, as the
new vendor had integrated CRM and telephony reports
Things to Remember
70
Intuitive Solutions isnt a technology provider, and yet it sees itself as a company
directly benefiting from WebRTC
Fusing VoIP with the web enables tighter integration across software applications,
bringing with it better visibility and insights for the business
As companies mature and try to grow out of their niche, with CRM vendors win the
contact center market, or will VoIP PBX and contact center vendors win over the
CRM market?
Tsahi Levent-Levi
Sponsored by
Vacasa Rentals
Vacasa Rentals is a US based company
providing a vacation homes rental service.
They handle around 500 homes spread
across 6 different states in the US. They do
this by operating solely on the cloud using
SaaS services. Besides dealing with
physical laptops and smartphones, all the
rest runs off virtual services and that
includes their telephony system as well.
Solution Type
Contact Center
Target Audience
Internal consumption
Business Model
None
Country
US
@vacasarentals
Media Types
Voice
rd
3 party
Vacasa Rentals debated as to the Use of WebRTC
approach to use between VoIP and softphones to WebRTC, and ended up using
WebRTC. This provided them more flexibility in how calls are routed to the relevant
Vacasa Rentals agents, who at the end of the day received the calls via WebRTC on their
browser.
Vacasa Rentals planned on doing the whole development work on their own, but ended
up using Plivo, one of the WebRTC API Platform vendors.
Things to Remember
71
This is a contact distribution solution, where incoming calls from PSTN get routed
via WebRTC to the relevant agent
While not a VoIP company or a contact center vendor, such use cases exist out
there vendors who do their business mainly via the phone and need intelligent
routing. They may end up developing it on their own, using a third party WebRTC
API Platform vendor, using a contact center solution or using traditional VoIP
Tsahi Levent-Levi
Sponsored by
Expert Marketplace
There is a market forming about expert services people with knowledge in a specific
domain who wish to share that information. Think of this market as the personal gym
trainer, only online and yes, there is more than a single vendor who offers this kind of a
personal gym trainer service online via WebRTC.
But the expert marketplace is larger than your typical gym, with experts from a large
variety of domain selling their time online. If you are a domain expert, then why not reach
out to customers via the web? While this can be achieved via Skype on the technical
level, these platforms offer the means to discover experts, understand their skills, see
ratings and comments of their customers, have the ability to interact with them, schedule
the session online and pay for the service all from within the same
website/webpage/service, with a single identity and without the hassle of installing any
software that has real value.
It isn't surprising then that there are WebRTC vendors in this market already, and two of
them are:
Apart from the two vendors detailed here, there are other vendors who offer marketplaces
for smaller niches.
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Tsahi Levent-Levi
Sponsored by
LiveNinja
LiveNinja is one of the vendors vying into
the expert marketplace.
It started off running on top of TokBox
platform in its Flash days and shifting with
TokBox into WebRTC.
Solution Type
Target Audience
Soho
Business Model
Revenue sharing
Things to Remember
73
While expert advice over the internet has existed prior to WebRTC, WebRTC offers
a simpler user experience where the whole business process is handled by a
single vendor, under a single roof, from a single website
What enables LiveNinja to move fast and offer new capabilities such as webinars
and events is their reliance on an API vendor, who provides these capabilities as
part of the service
For its Katana service offering, LiveNinja have opted for building its infrastructure
on its own, due to limitations of the API vendor it used in the features necessary in
Katana
Tsahi Levent-Levi
Sponsored by
Popexpert
popexpert is similar to LiveNinja in many
ways. It too offers a platform for experts to
build their profile pages, schedule and sell
their time online with the same type of
business model revenue sharing.
Solution Type
Experts market
Target Audience
Soho
Revenue sharing
Unlike LiveNinja, popexpert have taken the Business Model
US
route of developing the solution from the Country
@popexpert
ground up on their own including all of the Twitter
WebRTC related parts. While popexpert Media Types
Voice, Video
did start out by using an API vendor, they Use of WebRTC
Direct integration
switched along the way to their own
implementation this gave them an independence from the feature set (and bugs) of an
API vendor.
Things to Remember
74
An expert marketplace solution is about marketing and service. To that end, the
ability to customize the experience end to end is important. As with LiveNinja, a lot
of the attention of popexpert goes to the user experience domain
In the same industry, two different vendors choose different directions with
WebRTC some will go for tooling vendors for assistance while another would
prefer independence and self-development. The fact that getting started with
WebRTC is simple, these two options for every use case can almost always be
found
Tsahi Levent-Levi
Sponsored by
Telehealth
Telehealth is an interesting market segment when it comes to WebRTC. The challenge
with healthcare services is dealing with regulation and the liabilities involved in the
interactions created. US based vendors, for example, need to adhere to HIPAA
compliance, which stands for Health Insurance Portability and Accountability Act. HIPAA
includes directives around privacy of patients amongst other issues areas requiring
special attention and care from the vendors offering Telehealth solutions.
WebRTC is uniquely positioned as a very suitable technology. The reasons for that
include:
WebRTC is secured and private by default, whereas other systems can optionally
be made secure
WebRTC is suitable for easy incorporation into business processes and
Telehealth is all about business processes (from a regulatory point of view)
Development of Telehealth requires a lot of knowledge and experience of
healthcare something that is hard to come by in the VoIP domain, so having
something like WebRTC that makes it easy to develop VoIP use cases, makes it
possible to develop Telehealth services
The vendors discussed in this section provide different Telehealth solutions that are built
differently and have very different business processes:
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Tsahi Levent-Levi
Sponsored by
Claris Healthcare
Clearis Healthcare develops and sells the
Claris Companion, an Android-based
tablet for seniors.
Their role is to enable seniors to increase
their connectivity with their loved ones.
This is achieved by offering a specialized
tablet device with a user interface and
applications capable of supporting specific
activities and needs of seniors.
Solution Type
Healthcare
Target Audience
Consumers
Business Model
Device selling,
subscription
Country
US
@clarishealth
Media Types
Voice, Video
Things to Remember
76
While not a healthcare service per se, the Claris Companion is targeted at an
audience with specific needs focused around healthcare
WebRTC and the rise of tablets enabled Claris Healthcare to offer this type of
service while focusing on the user experience
Claris Healthcare are combining a business model of selling a device with a
monthly subscription fee
Tsahi Levent-Levi
Sponsored by
Regroup Therapy
Regroup Therapy enables the creation of
personal therapy sessions or group
therapy sessions.
While the solution may seem similar to the
services provided by LiveNinja or
popexpert, it is actually quite different; the
reason being the need to handle the
healthcare regulation.
Solution Type
Healthcare
Target Audience
Vertical
Business Model
Revenue sharing
Country
US
@RGTherapy
Media Types
Voice, Video
Things to Remember
77
Developing such a solution require as much knowledge and experience (or even
more) in group therapy as it does in communication technologies this knowledge
spans down from legal aspects, through bureaucratic/political aspects and up to
behavioral aspects
Healthcare comes in different shapes and sizes. Group therapy is one such
domain. Many other care related services can be provided by utilizing WebRTC
Tsahi Levent-Levi
Sponsored by
TruClinic
TruClinic provides a healthcare portal,
where patients can virtually visit doctors
online.
In the case of TruClinic, WebRTC plays the
role of providing video calling stitched into
specific business processes ones that
need to be secure and private due to the
regulations
surrounding
healthcare
services.
Solution Type
Healthcare
Target Audience
Vertical
Business Model
Revenue sharing
Country
US
@truclinic
Media Types
Voice, Video
Use of WebRTC
3rd party
Things to Remember
78
Tsahi Levent-Levi
Sponsored by
Gaming
One of the most untapped markets when it comes to WebRTC is gaming. There is a lot
of chatter about the potential uses of WebRTC in this space, but very little action.
This is why this section covers more demos than actual games. It is here to provide an
overview of what capabilities of WebRTC are currently being researched and played
around with by developers, and where that may lead us looking forward.
The games and concepts in this section are:
It is interesting to note that the traditional VoIP people are more passionate about
WebRTC in gaming than the gaming vendors at this point.
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Tsahi Levent-Levi
Sponsored by
BananaBread
BananaBread is a traditional 3D first
person shooter game ported to run purely
on HTML5. This was done using Java
Script and WebGL, where most of the initial
code got automatic translation from C++.
WebRTC was introduced in 2013 to
BananaBread. The data channel was
integrated into BananaBread offering low
latency network support for the exchange
of information between gamers in a multiplayer gaming session.
Solution Type
Gaming
Status
Demo
Use of WebRTC
Data channel
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Tsahi Levent-Levi
Sponsored by
CubeSlam
CubeSlam is Google's attempt
showcasing two capabilities:
at
Solution Type
Gaming
Status
Demo
Use of WebRTC
The interaction and data passed between the two players of the game is sent via WebRTC
data channel. Added to the game is a video chat session that enables both players to see
each other in real time. The video display is tilted with the board game as necessary using
WebGL.
The video component of this game isn't used for the game itself, but rather as a way to
get the players closer to each other and enhance the experience of the game.
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Tsahi Levent-Levi
Sponsored by
Jocly
Jocly Games is a bootstrap company from
France, who are focused at delivering
strategy and board games to their
community of players.
The games are all HTML5 games, and
recently, WebRTC was added to the
experience offering the ability of people
to have a more immersive experience,
where they see each other during the
game itself.
While not all the gamers are using the
video chat option that WebRTC offers
there, it does seem to be gaining traction.
Solution Type
Gaming
Target Audience
Consumers
Business Model
None
Country
France
@JoclyGames
Media Types
Voice, Video
Use of WebRTC
Direct integration
At the moment, Jocly Games have no monetization going around, but this may change
as they grow in audience. The most obvious choice here would be ads.
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Tsahi Levent-Levi
Sponsored by
Snake
The game of snake can be considered as
an HTML5 demo written by Nicolas
Beauvais for no other reason than to
show it is possible to do.
It is the classing Snake game, where a
small snake needs to be navigated to eat
eggs and grow in the process. The game
ends when the Stake steps on his own
body.
Solution Type
Gaming
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Tsahi Levent-Levi
Sponsored by
It is interesting to note, that most social network solutions prefer the route of direct
integration with WebRTC over the use of a 3rd party API vendor.
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Tsahi Levent-Levi
Sponsored by
MaxiAmigos
MaxiAmigos was the first dating/social
network to embrace WebRTC. They offer
an experience similar to Facebook, but
operate locally in Brazil, and in it, mostly in
So Paulo.
Solution Type
Social Network
Target Audience
Consumers
Freemium
MaxiAmigos offers an offline messaging Business Model
Brazil
service,
where
voice
and
video Country
communication comes secondary. They Twitter
n/a
first relied on a Flash based service, but Media Types
Voice, Video
once that service switched to a paid model,
Use of WebRTC
Direct integration
they opted for WebRTC. WebRTC in this
sense offered them more control and freedom over the solution.
MaxiAmigos started off integrating voice into the service, and recently they have added
video calling as well.
In terms of monetization, MaxiAmigos started off with an ads strategy and is planning to
move towards a different freemium model.
Things to Remember
85
While not a distinct dating service, this does show the potential. There are a couple
of dating services who are planning to introduce WebRTC in the near future
Switching from Flash to WebRTC offered MaxiAmigos the freedom they needed
and at the same time improved the quality and experience of the service they
provided to their users
Tsahi Levent-Levi
Sponsored by
Solaborate
Solaborate define themselves as the
missing link between Facebook,
LinkedIn and Yammer dedicated to
technology
professionals.
This
positions them as a niche social
network that caters the need of a
specific target audience.
Solaborate uses WebRTC to enable
voice and video calling both 1:1
calls as well as multipoint
conference calls.
Solution Type
Social Business
Target Audience
Soho
Business Model
None
Country
Macedonia
@solaborate
Media Types
Voice, Video
Use of WebRTC
Direct integration
One of the interesting facts about Solaborate is that they opted to run in Microsoft Azure
cloud instead of other cloud providers. While this is different than how other vendors have
approached the backend challenge, it seems to work well for Solaborate.
Things to Remember
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Tsahi Levent-Levi
Sponsored by
87
Peer5 a vendor specializing in peer assisted data delivery, focused on CDN and
file transfer
PeerMesh a P2P GIS solution
Swarmify providing CDN augmentation via the data channel
Tsahi Levent-Levi
Sponsored by
Peer5
Peer5 is an Israeli startup that has focused
from the start on the data channel. The
entrepreneurs behind Peer5 came with
web experience and knowledge and no
background in VoIP.
What they found in WebRTC was the
ability to connect browsers directly, and
they have built on top of that basic function
the ability for multiple browsers to
cooperate directly amongst themselves
when trying to access similar content.
Solution Type
Distributed CDN
Target Audience
SMB, Enterprise
Business Model
Pay as you go
Country
Israel
@peer5
Media Types
Data Channels
Use of WebRTC
Direct integration
That capability can be used to reduce the load from video streaming servers, which is
what Peer5 does; and it can be used to share files online, which is what Sharefest,
another service by Peer5 does.
The advantages of using such peer assisted technologies are:
88
Tsahi Levent-Levi
Sponsored by
PeerMesh
PeerMesh is
technology.
peer-to-peer
GIS
Solution Type
Distributed CDN
Target Audience
Vertical
Business Model
Unknown
Country
Turkey
None
Media Types
Data Channels
Use of WebRTC
Direct integration
The service has been created by a developer and a UX designer living in Turkey and
working for a vendor in the defense industry. While they have no solid commercial plans
for the service at the moment, it shows how WebRTC can be used to solve challenges in
a large range of domains.
Things to Remember
89
A relatively small team can make use of WebRTC to solve a challenging problem
WebRTC isnt limited to voice and video communications. There are many areas
in which it can be suitable via its data channel
The use of WebRTC is worldwide phenomena. Developers the world over are
tinkering with it at any given time
Tsahi Levent-Levi
Sponsored by
Swarmify
Swarmify is a service that provides similar
capabilities to those offered by Peer5: it
enables multiple web browsers accessing
the same content to share pieces of the
content directly between each other,
similar to how BitTorrent works.
Solution Type
Distributed CDN
Target Audience
SMB, Enterprise
Business Model
Subscription
US
@SwarmCDN
Data Channels
Direct integration
On the business model front, Swarmify offers just as simple tiered price plan, with
customized plans for large enterprises. The idea behind it is the reduction of the CDN
costs paid by websites.
The advantages of using this scheme:
90
Tsahi Levent-Levi
Sponsored by
Other
There are many more stories and use cases that didn't make it to this research paper.
Some because they bear similarities to other stories here, some because of breadth and
time while others still because of my knowledge on them.
In this section, I want to introduce a few more use cases ones that didn't fit elsewhere.
These are:
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Tsahi Levent-Levi
Sponsored by
Solution Type
Web development
Target Audience
Enterprise, SMB
Business Model
Project based
Country
US
@freshtilledsoil
Media Types
Voice, Video
Use of WebRTC
Direct integration
fresh tilled soil started dabbling into WebRTC without a specific customer. In their case,
it meant being able to understand the technology and showcase it to potential future
customers that wanted something special for their next web project.
They ended up releasing their own WebRTC widget to the world as part of this pet project.
Things to Remember
92
fresh tilled soil is at the forefront of such development outfits. Others will follow if
customer demand for such capabilities start piling up
The challenge for such vendors will be in the deployment of such a service, which
requires a bit more handholding and maintenance than the usual web project. They
will probably end up leaning towards one of the tooling vendors that offer SaaS
models to deliver projects to their customers
Tsahi Levent-Levi
Sponsored by
OpenVRI
VRI stands for Video Remote Interpreting.
VRI enables two people who are located in
the same room to communicate when they
don't share the same language or when
one of them is hard of hearing this is
achieved by opening a video call to another
person, who knows languages of both of
the people or knows sign language and
can interpret a deaf person.
VRI solutions are usually expensive, as
they are provided by the same devices and
services of enterprise video conferencing
solutions.
Solution Type
Distributed CDN
Target Audience
Vertical (VRI)
Business Model
None
Country
US
None
Media Types
Voice, Video
Use of WebRTC
Direct integration
OpenVRI is the result of a single person, hard of hearing himself, who decided to offer a
similar solution based on WeRTC and make it freely available to anyone both as a
service and in source code form.
OpenVRI is different in many aspects from both other VRI solutions as well as WebRTC
video calling solutions:
93
It has been built by a single person, hard of hearing himself this isn't an easy
feat, especially considering the fact that it includes a voice channel
It is free in every sense of the word. The service is hosted online and is accessible
and usable freely, while the source code of the service can be forked and modified
via github
It has been developed for the personal use of the developer he needed such a
service, so he just created it
The service is based around the concept of ad-hoc meetings. It generates a
hashed, one-time ink that can be emailed to the relevant parties that need to be
on the same session
There is a text chat window for each participant, as opposed to a single timeline
for all participants. This makes more sense to this type of a service, where people
would like to see the text as soon as it is typed, and not necessarily at its entirety
at times, this is the main channel of conversation
The developer of OpenVRI has since then joined a VRI company, where he
continues to develop WebRTC based solution for the hard of hearing in an
enterprise scale
Tsahi Levent-Levi
Sponsored by
WebRTC School
WebRTC School isn't a WebRTC vendor in
the "classic" sense. It doesn't develop
anything that is WebRTC-based, but rather
provides
training
services
around
WebRTC.
Solution Type
Learning
Target Audience
Developers
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Tsahi Levent-Levi
Sponsored by
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Tsahi Levent-Levi
Sponsored by
Recommendations
This chapter can be seen as a shortlist of recommendations of the various insights found
throughout this research paper. They are general guidelines to those who wish to build
services that use WebRTC.
1
2
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Tsahi Levent-Levi
97
Sponsored by
Tsahi Levent-Levi
Sponsored by
Blogs
There are many blogs who are now covering WebRTC; some of them are large news
outlets while others bring deep thinking as well as user stories.
Below is a hand-picked list of such blogs. You can consider them mandatory reading list
when it comes to understanding WebRTC.
BlogGeek.me
My very own blog, where I write a lot about WebRTC and try also to give room from
experts by way of guest posts.
WebRTC Weekly
A weekly newsletter providing roundup of posts and articles from around the web about
WebRTC. Curated by Chris Koehncke and me.
WebRTC Glossary
A glossary site for all terms related to WebRTC.
webrtcHacks
This is a blog site operated by Chad Hart, Reid Stidolph, Victor Pascual vila and Tsahi
Levent-Levi. While dealing more with the technical aspects of WebRTC, it does offer
business insights as well as providing an events directory for WebRTC.
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Tsahi Levent-Levi
Sponsored by
Chris Kranky
Chris Koehncke's personal blog is a great source of information about WebRTC. He is
opinionated, to the point and blunt. He also writes with a unique voice that is enjoyable to
read every time.
Alan Quayle
Alan Quayle, an independent consultant in the telecommunication space. His focus
around Telco APIs and enablers to these have brought him to look closely at WebRTC
as well.
Disruptive Analysis
Disruptive Analysis is Dean Bubley's blog. He is a consultant in the telecommunication
space that is looking at WebRTC as well.
He has his own research paper on WebRTC.
WebRTC World
WebRTC World is a dedicated portal for WebRTC created and maintained by TMCnet. It
is the place to go for up to date information about vendors and their plans and
announcements around WebRTC.
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Tsahi Levent-Levi
Sponsored by
Communities
There are several online communities where conversations around WebRTC happen.
The ones here are those that I find the most informative and engaging.
WebRTC on Google+
Google+ has two distinct groups for WebRTC. There is no difference in terms of the type
of content between them. People in this group are very responsive to questions asked.
Meetup Events
At the time of writing, there were 18 different WebRTC related meetup groups around the
world.
These are localized groups that conduct local events about WebRTC. The target
audience as well as the content of each meeting vary widely, but if there is a meetup
located close by to where you live, join it to gauge the local community that is hacking its
way around WebRTC.
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Tsahi Levent-Levi
Sponsored by
What is WebRTC?
What are the KPIs when selecting an API platform for WebRTC?
What API platforms are there and how do they fit your needs?
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Tsahi Levent-Levi
Sponsored by
BlogGeek.Me Blog
My blog is the place where the bulk of my writing exists. It is available freely and can be
subscribed via both RSS and email. See http://bloggeek.me
Reports
This is my second report about WebRTC. To find out about other paid reports I have
written, see http://bloggeek.me/reports/
Consulting
I provide consulting services to vendors, especially around VoIP, video conferencing
and WebRTC. See http://bloggeek.me/consulting/
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Tsahi Levent-Levi
Sponsored by
About GENBAND
GENBAND is a global leader in smart networking solutions for service providers and
enterprises in more than 80 countries. From the Core to the Edge to the Experience,
the companys technology seamlessly evolves IP networks to new levels in scalability,
security, profitability and efficiency. GENBANDs market-leading technology facilitates
multimedia voice, data and video sessions and anywhere and any device services that
scale on public and private networks. GENBAND is headquartered in Frisco, Texas, and
has R&D, sales and support resources in more than 50 countries.
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Sponsored by
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Tsahi Levent-Levi
Sponsored by
Business Assistant - Keep work and home life separate with your own business
number you can use for voice, texting, and voicemail.
See http://assistant.kandy.io/
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