This document discusses bridging the gap between WebRTC and VoIP technologies. It begins by defining WebRTC as a browser-based real-time communications standard and explaining its advantages over traditional VoIP in areas like developer adoption and deployment. The document then outlines several potential use cases for WebRTC in services providers, enterprises, and contact centers. It acknowledges that WebRTC lacks a defined signaling protocol and discusses SIP over WebSocket as a popular option. The document concludes by introducing the concept of a WebRTC gateway to interconnect WebRTC and existing SIP networks by translating between their protocols and addressing differences in media capabilities.
When people think about WebRTC, they think about video calls inside a web browser. WebRTC is much more than that. WebRTC can be used to create fundamentally better experiences by embedding live, peer-to-peer communications in SaaS products, mobile apps, and websites. But what is the state of WebRTC today? What does it take for a business to really reap the benefits?
My slide deck from the session I gave at Twilio's Signal event May 2015.
Tsahi, is gonna make sure you've all got the basic fundamentals of WebRTC under your belt. It's a 101 tutorial, it's a baseline, may have heard it before but we want no one left behind. Already an expert? Then consider this a 20 minute nap time!
In this session, we cover the basics of what WebRTC is, what network components participate in a WebRTC service and where to find the right resources to learn more about WebRTC.
WebRTC Business Use Cases | WebRTC Conference & Expo IIILawrence Byrd
Presentation on WebRTC Business Use Cases from WebRTC Conference & Expo Nov 19-21 in Santa Clara CA. This was part of Tuesday’s Business Introduction to WebRTC morning session delivered alongside presentations from Phil Edholm, Chris Vitek, Tsahi Levent-Levi, Brent Kelly and John Burke.
Mid-level review of server infrastructure that is required and often used with WebRTC, including signaling servers, NAT traversal servers (STUN and TURN), media servers, and WebRTC Gateways.
Presented at the WebRTC Japan Conference in Tokyo.
When people think about WebRTC, they think about video calls inside a web browser. WebRTC is much more than that. WebRTC can be used to create fundamentally better experiences by embedding live, peer-to-peer communications in SaaS products, mobile apps, and websites. But what is the state of WebRTC today? What does it take for a business to really reap the benefits?
My slide deck from the session I gave at Twilio's Signal event May 2015.
Tsahi, is gonna make sure you've all got the basic fundamentals of WebRTC under your belt. It's a 101 tutorial, it's a baseline, may have heard it before but we want no one left behind. Already an expert? Then consider this a 20 minute nap time!
In this session, we cover the basics of what WebRTC is, what network components participate in a WebRTC service and where to find the right resources to learn more about WebRTC.
WebRTC Business Use Cases | WebRTC Conference & Expo IIILawrence Byrd
Presentation on WebRTC Business Use Cases from WebRTC Conference & Expo Nov 19-21 in Santa Clara CA. This was part of Tuesday’s Business Introduction to WebRTC morning session delivered alongside presentations from Phil Edholm, Chris Vitek, Tsahi Levent-Levi, Brent Kelly and John Burke.
Mid-level review of server infrastructure that is required and often used with WebRTC, including signaling servers, NAT traversal servers (STUN and TURN), media servers, and WebRTC Gateways.
Presented at the WebRTC Japan Conference in Tokyo.
My talk on webRTC from June 2013
Demo application using XMPP for signalling
open source webRTC using websockets is here: implenentationhttps://github.com/pizuricv/webRTC-over-websockets
WebRTC Tutorial by Dean Bubley of Disruptive Analysis & Tim Panton of Westhaw...Dean Bubley
Tutorial on WebRTC technologies, standards, use-cases and business models. First given at the ICIN conference in Venice, October 2013.
By Dean Bubley, analyst at Disruptive Analysis, and Tim Panton, WebRTC developer at Westhawk Ltd
WebRTC is an exciting new technology that lets you easily add realtime communication capabilities to your web and native apps. Learn more about WebRTC in this presentation from the real-life practitioners at Gruveo (www.gruveo.com).
My presentation for the Kranky Geek April 2015 London event.
Took the audience through a history lesson of WebRTC, showing the position of some of the companies and the market opportunities the various vendors are going after.
WebRTC gives us a way to do real-time, peer-to-peer communication on the web. In this talk, we'll go over the current state of WebRTC (both the awesome parts and the parts which need to be improved) as well as what could come in the future. Mostly though, we'll take a look at how to combine WebRTC with other web technologies to create great experiences on the front-end for real-time, p2p web apps.
A short intro and update on WebRTC presented at WebRTC Boston 6 covering:
- some recognizable WebRTC use case examples
- review of all the standardized API's that come with WebRTC
- Intro to some of the servers that may be needed with WebRTC
- what's next for WebRTC including Machine learning, lower-level API's, new options for customization, new codecs, and a new transport
See the presentation at https://youtu.be/ptnceQZ4fPg
Thanks to WebRTC Boston 6 sponsors:
Google
YouTube
callstats.io - WebRTC Analytics https://callstats.io
Kranky Geek - RTC Events and Videos https://krankygeek.com
To Build or Not to Build Your WebRTC InfrastructureTsahi Levent-levi
These are the slides for the Upperside Webinar I talked at.
The acquisition of AddLive by SnapChat created some discomfort among companies using a WebRTC API platform. It made the threat, well known to all those building their future on someone else’s service, apparent and real. If you are now planning your service the first decision to be made is whether to build your own infrastructure or use an API platform.
Should decisions be made only in light of current happenings in the market? Are there more options except for to build or use a service?
The selection of an API platform is an important one. There are over 20 such platforms available. But they are different. They don't offer the same set of capabilities, they aren't focused on the same types of use cases and customers. The selection process requires an understanding of the use case, the business values, the features and requirements you have. In this webinar, we will review the various KPIs and selection criteria, offering an instruction manual for vendor selection and self built infrastructure options.
Kranky Geek WebRTC 2015 - The future of ORTC with WebRTCKranky Geek
Trent Johnsen from Hookflash will review of Object RTC (ORTC) and how its improvements are making they making their way into WebRTC already. Bernard Aboba (Microsoft) will then discuss some ORTC-based WebRTC implementation examples, including Microsoft's new Edge browser.
NUBOMEDIA: an elastic Platform as a Service (PaaS) cloud for interactive soci...Luis Lopez
NUBOMEDIA is the first cloud platform specifically designed for hosting interactive multimedia services. Its architecture is based on media pipelines: chains of elements providing media capabilities such as encryption, transcoding, augmented reality or video content analysis. These chains allow building arbitrarily complex media processing for applications. As a unique feature, from the point of view of the pipelines, the NUBOMEDIA cloud infrastructure behaves as a single virtual super-computer encompassing all the available resources of the underlying physical network. Thanks to this, NUBOMEDIA applications can elastically scale and adapt to the required load preserving Quality of Service (QoS) and Service Level Agreement (SLA) guarantees.
NUBOMEDIA mission is to democratize interactive multimedia communication services by making their creation, deployment and mass-scale exploitation a cheap, rapid and effortless process. To achieve this, we use a strategy composed of two axes. First, NUBOMEDIA exposes its capabilities through a simple to use and intuitive API that can be used by non-expert developers on most popular client platforms such as smartphones and WWW browsers. Second, the NUBOMEDIA infrastructure is released using a flexible and attractive Free Open Source Software license guaranteeing openness and neutrality.
WebRTC enables context based, embedded communication in any app or website. Skylink makes using WebRTC as simple as using jQuery for web developers.
Here is the link to the JS Remote Conf talk this presentation was held first: https://www.youtube.com/watch?v=x2IHJBp2TTo
WebRTC for Telco: Informa's WebRTC Global Summit PreconferenceTsahi Levent-levi
The preconference workshop I did at Informa's WebRTC Global Summit in London, 31st of March 2014
It is targeted at bringing people up to speed with what WebRTC is, how people and vendors are using it today and placing it also in the context of the telecom world (which is the focus of this specific conference).
My talk on webRTC from June 2013
Demo application using XMPP for signalling
open source webRTC using websockets is here: implenentationhttps://github.com/pizuricv/webRTC-over-websockets
WebRTC Tutorial by Dean Bubley of Disruptive Analysis & Tim Panton of Westhaw...Dean Bubley
Tutorial on WebRTC technologies, standards, use-cases and business models. First given at the ICIN conference in Venice, October 2013.
By Dean Bubley, analyst at Disruptive Analysis, and Tim Panton, WebRTC developer at Westhawk Ltd
WebRTC is an exciting new technology that lets you easily add realtime communication capabilities to your web and native apps. Learn more about WebRTC in this presentation from the real-life practitioners at Gruveo (www.gruveo.com).
My presentation for the Kranky Geek April 2015 London event.
Took the audience through a history lesson of WebRTC, showing the position of some of the companies and the market opportunities the various vendors are going after.
WebRTC gives us a way to do real-time, peer-to-peer communication on the web. In this talk, we'll go over the current state of WebRTC (both the awesome parts and the parts which need to be improved) as well as what could come in the future. Mostly though, we'll take a look at how to combine WebRTC with other web technologies to create great experiences on the front-end for real-time, p2p web apps.
A short intro and update on WebRTC presented at WebRTC Boston 6 covering:
- some recognizable WebRTC use case examples
- review of all the standardized API's that come with WebRTC
- Intro to some of the servers that may be needed with WebRTC
- what's next for WebRTC including Machine learning, lower-level API's, new options for customization, new codecs, and a new transport
See the presentation at https://youtu.be/ptnceQZ4fPg
Thanks to WebRTC Boston 6 sponsors:
Google
YouTube
callstats.io - WebRTC Analytics https://callstats.io
Kranky Geek - RTC Events and Videos https://krankygeek.com
To Build or Not to Build Your WebRTC InfrastructureTsahi Levent-levi
These are the slides for the Upperside Webinar I talked at.
The acquisition of AddLive by SnapChat created some discomfort among companies using a WebRTC API platform. It made the threat, well known to all those building their future on someone else’s service, apparent and real. If you are now planning your service the first decision to be made is whether to build your own infrastructure or use an API platform.
Should decisions be made only in light of current happenings in the market? Are there more options except for to build or use a service?
The selection of an API platform is an important one. There are over 20 such platforms available. But they are different. They don't offer the same set of capabilities, they aren't focused on the same types of use cases and customers. The selection process requires an understanding of the use case, the business values, the features and requirements you have. In this webinar, we will review the various KPIs and selection criteria, offering an instruction manual for vendor selection and self built infrastructure options.
Kranky Geek WebRTC 2015 - The future of ORTC with WebRTCKranky Geek
Trent Johnsen from Hookflash will review of Object RTC (ORTC) and how its improvements are making they making their way into WebRTC already. Bernard Aboba (Microsoft) will then discuss some ORTC-based WebRTC implementation examples, including Microsoft's new Edge browser.
NUBOMEDIA: an elastic Platform as a Service (PaaS) cloud for interactive soci...Luis Lopez
NUBOMEDIA is the first cloud platform specifically designed for hosting interactive multimedia services. Its architecture is based on media pipelines: chains of elements providing media capabilities such as encryption, transcoding, augmented reality or video content analysis. These chains allow building arbitrarily complex media processing for applications. As a unique feature, from the point of view of the pipelines, the NUBOMEDIA cloud infrastructure behaves as a single virtual super-computer encompassing all the available resources of the underlying physical network. Thanks to this, NUBOMEDIA applications can elastically scale and adapt to the required load preserving Quality of Service (QoS) and Service Level Agreement (SLA) guarantees.
NUBOMEDIA mission is to democratize interactive multimedia communication services by making their creation, deployment and mass-scale exploitation a cheap, rapid and effortless process. To achieve this, we use a strategy composed of two axes. First, NUBOMEDIA exposes its capabilities through a simple to use and intuitive API that can be used by non-expert developers on most popular client platforms such as smartphones and WWW browsers. Second, the NUBOMEDIA infrastructure is released using a flexible and attractive Free Open Source Software license guaranteeing openness and neutrality.
WebRTC enables context based, embedded communication in any app or website. Skylink makes using WebRTC as simple as using jQuery for web developers.
Here is the link to the JS Remote Conf talk this presentation was held first: https://www.youtube.com/watch?v=x2IHJBp2TTo
WebRTC for Telco: Informa's WebRTC Global Summit PreconferenceTsahi Levent-levi
The preconference workshop I did at Informa's WebRTC Global Summit in London, 31st of March 2014
It is targeted at bringing people up to speed with what WebRTC is, how people and vendors are using it today and placing it also in the context of the telecom world (which is the focus of this specific conference).
Digium 'Demo & Eggs' Breakfast Presentation slides, as shown at WebRTC World III on November 21, 2013.
These slides we used in a presentation which also featured a live demo of a WebRTC-enabled Asterisk appliance (based on a Raspberry Pi just for fun) serving a web page that contained the JsSIP soft phone.
Audience members were able to connect to our WiFi network and use Chrome or Firefox to browse to this page, and them make a call to each other, to a Digium phone, to hear a message from Allison (THE Voice of Asterisk) or to go into a conference call with each other.
Upperside Webinar- WebRTC from the service provider prism-finalAmir Zmora
A Webinar I did with Victor Pascual Avila (Quobis) and Sebastian Schumann (Slovak Telekom) for Upperside Conferences. Webinar talks about the different approaches service providers can take with WebRTC, what developers need and some actual examples of things Slovak Telekom has done.
Recording of this Webinar can be found here: https://attendee.gotowebinar.com/register/5051075414841550849
WebRTC Webinar & Q&A - W3C WebRTC JS API Test Platform & Updates from W3C Lis...Amir Zmora
On September 19-23 there was the W3C TPAC meeting in Lisbon. Dan will cover some of the highlights of the recent Lisbon WebRTC meeting, including what items are the sticking points, where work is focusing, progress estimates, and thoughts on what might go into the next version of WebRTC after 1.0 is finished.
Alex will cover the W3C testing platform: "Test The Web Forward". W3C, unlike IETF, is developing and maintaining a complete test suite for all its JS APIs. No specification is actually accepted by W3C and final without the corresponding test suite. Topics that will be addressed include what this testing platform implements, its status with respect to WebRTC and now it is used by different browser vendors as an indication of their compliance with the standards.
As always, we encourage you to submit your general WebRTC related questions beforehand in the Questions & Topics section to make sure we answer them during the session.
Event sponsored by WebRTC.Ventures & Blacc Spot Media
A Webinar by Victor Pascual Avila and Amir Zmora about WebRTC standards. IETF and W3C work on WebRTC as well as interworking with other networks such as IMS. The Webinar also talks about WebRTC signaling options and video codecs.
WebSphere Liberty Rtcomm: WebRTC Middleware for the EnterpriseBrian Pulito
In order to provide the type of services their customers crave, your clients need to be able to provide blazing fast communication capabilities and access important information in the blink of an eye. WebRTC (Web Real-Time Communications) allows for the creation of next-generation communication applications without the need for browser plugins. WebSphere Application Server Liberty Profile is changing the way people communicate by making it easy to provide web page context as part of real-time conversations. This webinar will cover all of the real-time communications features recently released in WebSphere Liberty, including the new Rtcomm feature for rapid development of WebRTC based applications, and the open-source Rtcomm client-side libraries. (link to webinar replay: http://www.websphereusergroup.org/khatch/go/gallery/item/1543395?type=video)
Architecting your WebRTC application for scalability, Arin SimeAlan Quayle
TADSummit 2022 8/9 Nov Aveiro Portugal
Architecting your WebRTC application for scalability
Arin Sime, CEO/Founder at WebRTC.ventures and AgilityFeat, & Alberto González Trastoy, CTO at WebRTC.ventures | Software/Telecom Engineer.
There are many ways to architecture your live video application with WebRTC. Open Source and CPaaS media servers are one consideration, but far from the only decision you’ll need to make.
In this session we will give an update on the most popular media servers to consider as well as go deeper into scalability with topics such as deployment using kubernetes/docker, persistence when using multiple SFU/MCU servers, and optimizations available with WebRTC for better performance.
WebRTC enables real-time communication through the web, while SIP is a protocol commonly used for initiating and maintaining real-time communication sessions, particularly in telephony networks.
Bridging WebRTC with SIP is essential in many industries, such as remote healthcare, education, and customer support, where current modern video solutions must communicate with telephony infrastructure at scale. The integration of WebRTC-based video conferencing with legacy SIP-based systems enables seamless communication across platforms and devices. In this presentation, we will talk about lessons learned and explore different approaches to bridging WebRTC and SIP, discussing their advantages and disadvantages.
Similar to WebRTC and VoIP: bridging the gap (Kamailio world conference 2013) (20)
WebRTC and VoIP: bridging the gap (Kamailio world conference 2013)
1. WebRTC
and
VoIP:
bridging
the
gap
@victorpascual
victor.pascual.avila@gmail.com
h>p://es.linkedin.com/in/victorpascualavila
Images
Source:
Google
Images
2. Intro
• What
is
WebRTC
(Real
Time
CommunicaDons)?
– A
browser-‐embedded
media
engine
(sDll
being
finalized)
– Emerging
and
standard
method
of
web-‐based
RTC
(W3C/IETF/3GPP)
– Another
type
of
access
framework
• Why
the
hype?
– Web:
most
dynamic,
innovaDve
place
on
planet
– RTC
has
largely
been
absent
– WebRTC
delivers
RTC
to
those
that
create
the
Web
• WebRTC
is
posed
to
grow
rapidly
and
WebRTC
will
be
an
important
access
method
in
the
future
for
Service
Providers,
contact
centers,
and
enterprises
– Number
of
developers:
1000s
of
SIP
programmers
vs.
100,000
of
JavaScript
programmers
– Deployment:
IP
phones
must
be
physically
deployed
or
soWware
installed
(soWware
updates
may
not
be
automated)
vs.
Browsers
automaDcally
updated
to
add
support;
client
updates
automaDcally
pushed
to
browser
– Several
dozens
of
vendors
with
unique
SIP
implementaDons
vs.
Handful
of
browser
vendors
with
unique
WebRTC
implementaDons
Opensource
code
available
• Moving
really
fast,
lots
of
early
implementaDons,
customer
trials…
BUT
sDll
a
number
of
open
issues
– MTI
video
codec
discussion,
SDP
O/A,
strict-‐firewall/HTTP-‐middlebox
traversal,
MicrosoW’s
CU-‐RTC-‐Web,
Apple,
etc
3. Use
Cases
• WebRTC
enables
innovaDve
use
cases
on
the
Web
– WebRTC
It’s
not
meant
to
be
the
new
Web
Telephony
• But
interworking
towards
exisDng
legacy
is
required
– Extend
exisDng
SIP
environments
into
the
exciDng
new
web
domain
• Service
Provider
subscriber
access
via
WebRTC
methods
– Browser-‐based
RTC
to
complement
SIP
offerings
or
simply
extend
exisDng
SIP-‐services
over
web
• Voice
service
extension
with
web-‐phone
• New
Telco-‐OTT
services
(over-‐the-‐top)
• RCS-‐e
(rich
communicaDons
services
–
enhanced)
• Conferencing
• Web-‐based
comms
provider
PSTN
break-‐out
• Enterprise
UC
without
thick
or
thin
client
soW
phones
– Easier
to
maintain
&
break
single
UC
vendor
lock
• Contact
centers
embedding
RTC
into
customer
service
web
pages
– Customer
saDsfacDon
&
lower
costs
• Not
only
Web
browsers
(e.g.
Chrome,
Firefox)
but
also
naDve
support
via
apps
or
OS
(e.g.
set-‐top
boxes,
FirefoxOS)
4. WebRTC
has
no
defined
signaling
method
Don’t
panic,
it’s
not
a
bad
thing!
5. Signaling
Plane
• WebRTC
has
no
defined
signaling
method.
JavaScript
app
downloaded
from
web
server.
Popular
choices
are:
• SIP
over
Websockets
– Standard
mechanism
(draW-‐ief-‐sipcore-‐sip-‐websocket)
–
soon
to
be
RFC
– Extend
SIP
directly
into
the
browser
by
embedding
a
SIP
stack
directly
into
the
webpage
–
typically
based
on
JavaScript
– WebSocket
create
a
full-‐duplex
channel
right
from
the
web
browser
– Popular
examples
are
jsSIP,
sip-‐js,
QoffeeSIP,
or
sipML5
• Call
Control
API
• propietary
signaling
scheme
based
on
more
tradiDonal
web
tools
and
techniques
• GSMA/OMA
extending
RCS-‐e
“standard”
API
to
include
WebRTC
support
• Other
alternaDves
based
on
XMPP,
JSON
or
foobar
6. Media
Plane
• A
browser-‐embedded
media
engine
– Audio
codecs
–
G.711,
Opus
are
MTI
– Video
codecs
–
H.264
vs.
VP8
(MTI
TBD
-‐
IPR
discussion)
– Media
codecs
are
negoDated
with
SDP
(for
now
at
least)
– Best-‐of-‐breed
echo
canceler
– Video
ji>er
buffer,
image
enhancer
– Requires
Secure
RTP
(SRTP)
–
DTLS
– Requires
Peer-‐2-‐peer
NAT
traversal
tools
(STUN,
TURN,
ICE)
–
trickle
ICE
– MulDplexing:
RTPs
&
RTP+RTCP
• Yes,
your
favorite
SIP
client
implementaDon
is
compaDble
with
most
of
this.
But,
the
vast
majority
of
deployments
– Use
plain
RTP
– Do
not
support
STUN/TURN/ICE
– Do
not
support
mulDplexing
(ok,
not
really
an
issue)
– Use
different
codecs
that
might
not
be
supported
on
the
WebRTC
side
7. OpDon
1:
Just
ignore
the
gap
And
expect
all
deployed
equipment
to
be
upgraded
soon
8. OpDon
2:
Try
to
fix
it
When/While
necessary
(hopefully
with
the
right
soluDon!)
9. OK,
might
need
to
do
a
couple
of
fixes
to
make
things
work,
but…
HOW?
11. WebRTC
Gateway
Taxonomy
(great
blog
post
by
Chad
Hart)
• SIP-‐over-‐WebSockets
gateway
– act
as
a
SIP-‐server
and
terminate
the
WebSocket
from
the
browser
and
convert
that
to
UDP/TCP
transport
into
the
SIP
network
• Web
API
to
SIP
gateway
– act
as
a
web-‐server
and
convert
the
API
calls
to
SIP
• Media
controller
– ICE/STUN/TURN
– SRTP/RTP
interworking
– Mux/demux
• Transcoding
gateway
– convert
from
one
codec
to
another
(incl.
video!)
• Flash
RTMP
gateway
– majority
of
browsers
on
the
web
today
support
flash
and
not
WebRTC
Several
implementaDon
opDons:
– one
or
more
of
the
opDons
above
– WebRTC
as
a
feature/product/service
– Other
features
(?):
idenDty,
charging,
session
recording,
LI,
middlebox-‐
traversal,
etc.