Slide 1

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 2

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 3

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 4

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 5

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 6

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 7

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 8

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 9

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 10

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 11

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 12

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 13

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 14

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 15

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 16

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 17

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 18

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19


Slide 19

WebRTC in the Enterprise
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
WebRTC in the Enterprise
ITEXPO October 2015 Anaheim
By:

Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB

1

Where is WebRTC and What’s for the Enterprise?
 Standards (IETF and W3C WGs started 2011) progressing slowly
•
•
•
•
•
•

IETF has finally roughly hummed for VP8 AND H.264 as mandatory video codecs
Apple and Microsoft have (almost, maybe) committed, but will probably only do H.264
Google will ship Chrome with VP8, VP9 and H.264 built-in (no download)
Still only in some WebRTC browsers: Google’s Chrome, Mozilla’s Firefox and Opera
Many others still missing
Network-provided TURN-servers are needed (will talk more about), awaited standards
• draft-ietf-tram-turn-server-discovery-04
• draft-ietf-rtcweb-return

 Click-to-call is held up, even though…
• There are plugins getting WebRTC (including VP8) into IE (Microsoft’s) and
Safari(Apple’s) today (our test site https://webrtc.ingate.com will prompt for those)
• Apps (not browsers) implementing the WebRTC protocols are being built – especially for
iPhone (iOS) and Android – Needed!

 But is there more for the enterprise than click-to-call on the website
and the cloud services that we are starting to see?
 Yes! Enterprise usage may actually be a driver!
2

What Can WebRTC Bring to the Enterprise?
Something Beyond Just Using Cloud Services?
There Will be an Enhanced “Enterprise Social Network”
Pass a WebRTC link over IM or an email,
asking people to click-to-call you or something.
http://companion.smartcomp.com/[email protected]
SIP System

MPLS

SIParator®

Company
Web
Server

But:
No Numbers!?
Passing links…

Data & VoIP LAN

LAN

SIP

Browsers as
Softclients!
HD Multimedia
Telepresence

3

Technically –
What is it?

Voice
Video
Data
“For free!”

From the first WebRTC Conference November 2012
4

What WebRTC Does NOT Do:

BASICS

“No Numbers”
No rendezvous – “no
addressing” at all.
Not like SIP

What WebRTC Does:
• Sets up media directly between
browsers (SDP/RTP like SIP) –
typically using a common web
application.

-----------More communication islands?
Yes, but it is adding high quality
real-time communication when
we are already in contact.

• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).

Voice
Video
Data
“For free!”

5

What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You are already in contact:
Chatting, emailing. Just pass a link
(URL) to click!
• Or join a scheduled meeting
• No rendezvous protocol like SIP
required
• “Integrating into Facebook chat
takes about half an hour”, Google
said…

This is Internet/OTT and does not
enter VoIP, IMS networks or the
enterprise PBX, unless…
6

Demonstration of social calling without numbers using
Ingate’s public test site in Sweden

When the receiver (e.g.
via IM or email) of this link
clicks it, a window pops-up
and sets up a video
conference between our
WebRTC browsers. No
numbers, no SIP, no
PSTN involved.

Whoever clicks this link will be connected to a
conference bridge in the SIP PBX/UC solution
(a WebRTC-SIP gateway is required). Passed
together with a Webex invitation, the
conference is held without the need for phones.

7

And a Click-to-Call Website is Great
Don’t Dial, Just click!

Calling by Clicking at a Web Page
A great application
Company
Web Server

Do we need more than the company
website and the always available browser?
You are on the Web – Wanna talk?
– Don’t pick up your phone. Just
click! Communicate with voice,
video and data and screen.

This is the Call Center Killer App!
We want the call into the call center UC
solution! The click may be context sensitive, containing caller’s information.
Avaya showed at the WebRTC conference.
8

With a “WebRTC & SIP Companion” Gateway,
adding Click-to-Call to a Website is Simple

To add WebRTC click-to-call buttons to
enterprise websites, simply copy some JScode from the SIParator® Companion into
the enterprise website.
Deployment and installation will be the
same as for SIP trunking
– The SIParator is already at the
demarcation point (between the Internet
and the enterprise LAN) and interfaces
with the PBX/UC/contact center solution
– Just like when SIP trunking using a
SIParator, WebRTC goes into the
PBX/UC/Call center (and/or directly to a
browser anywhere).

9

Demonstration of the call center click-to-call killer application,
using Ingate’s local test site here and public test site in Sweden.

(1) Click-to-call buttons on
a website can open a
WebRTC voice or video
window connecting to the
right call agent also
forwarding context and
user information. A
WebRTC-to-SIP gateway
connects the WebRTC to
the SIP-based call center
solution.

We just saw this demo…

(2) To prove that we are really using SIP trunking hooked to good old telephony let’s
here in LV, a Swedish mobile phone dial +46812205614 which is SIP trunked to
[email protected] by registered at this web site.

10

The WebRTC Browser as a Softphone
Having the PBX/UC softphone available
everywhere, on every device that has a
browser, without any plug-in and not just
for plain voice phone calls, but potentially
also for HiFi HD telepresence-quality
videoconferencing, is of course a dream.

This is an obvious WebRTC
application for the enterprise PBX
or UC solution.
It will especially ease remote
PBX/UC usage, since WebRTC
includes the NAT/Firewall
traversal method
(ICE/STUN/TURN) in itself.

A Gateway WebRTCSIP Gateway is
Required

11

It’s not only the web:
We Need The WebRTC Calls Into the Contact Center

WebRTC by
itself bypasses
the SIP PBX/UC
infrastructure.

Voice/Video and
more, from click-tocall buttons and
passed links etc.

Ingate provides a WebRTCSIP gateway in the trunk
CPE, so WebRTC calls go
into the existing auto
attendant, queues, forwards,
transfers, conference
bridges and PBX phones.
The same gateway can
integrate WebRTC clients
12

Demonstration of HD Telepresence Quality Video Conferencing,
using Ingate’s public test site in Sweden.
This has only been available with 100 kUSD equipment in special
rooms before
Soon at everyone’s desktop and pocket.
Save flight tickets and other travel for quality meetings

The WebRTC browser gives a quality only seen in
expensive telepresence systems before. Here a
conference between a SIP-connected browser client,
two laptop WebRTC browsers, a mobile Galaxy S5
using Chrome browser and ”ms. Time” telling time in
Sweden at telephony number 90510.

13

Part 2: General WebRTC Usage in the Enterprise
Here it is about using cloud based WebRTC services – web
server applications that WebRTC was/is intended for (not the
previous where “WebRTC & SIP Companion” gateway is the web
application)
Today we see real-time communication applications like
UberConference, click-to-call usage, Google Hangouts and other
early usage

Enterprise usage – from the protected enterprise LAN – is of course
highly important. But there are problems to solve – same as with
any real-time communication, whether H.323, SIP or now WebRTC
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, carriers today have to use their guest
Wi-Fi instead of their own LAN…

 Data-crowded enterprise firewalls means bad quality, QoS
14

WebRTC and UC Require Better QoS Than Voice
* QoS discussion and details in footnote

From 3.5 kHz Voice to HiFi HD Telepresence Quality!
Audio HiFi Codec Opus & Video HD Codec VP8 and/or H.264
* The confusion around Quality of Service (QoS) requirements for real-time traffic:
While telcos mostly regard QoS as highly important and often do level 2 or 2.5 separated networks or reservation-type of QoS
for voice, even where level 3 IP QoS (e.g. diffserv) could achieve the same, others (within IETF and WebRTC enthusiasts)
often ignore QoS, assuming such problems will go away and believe “it is all about bandwidth”. That is true but only if the pipe
is not filled! However, TCP traffic (surf, email, file transfer) intermittently fills the pipe in its attempts to transfer the data as fast
as possible. Doubling the bandwidth when sharing real-time traffic with intense data traffic on the same pipe, will not make half
of the bandwidth usable for quality traffic - it will rather halve the time that the pipe is crowded.

15

WebRTC - Like All Real-Time Communication Protocols has a NAT/Firewall Traversal Problem
 Firewalls do not allow
unknown incoming
signaling and media is a
“surprise” (just like SIP)
signaling
Company
Web
Server

LAN

media

 SBCs are Firewalls that
know SIP and take it into
the LAN, but WebRTC
Company
prescribes
Web
ICE/STUN/TURN to fool Server
the firewall to let the real- WS/WSS
time traffic through (similar
ICE
to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open

media
STUN
TURN
SERVER

LAN

 There are media issues…
a) Getting through
b) Quality

16

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for networkprovided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
the media flows there under control.
• Security is back in the right place –
Where you have the firewall.
• The enterprise firewall in itself can still
be restrictive.
• The carrier provides a “WebRTCSBC” in the trunk CPE

Q-TURN Enables QoS and More:
• Prioritization and traffic-shaping
• Diffserv or RVSP QoS over the
Net
• Authentication (in STUN and
TURN)
• Accounting (usage of this pipe)
17

Ingate Has Been Driving the Idea of a TURN Server
PARALLEL to the Firewall (Q-TURN)
Upcoming standards for network
provided TURN servers will allow:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall.

• Have the TURN server functionality
PARALLEL to the firewall and setup
Q-TURN (a network-provided TURN server)
will beEnables QoS and More:
Q-TURN
the media flows there under control.
added in future releases of the Ingate SIParator®.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Where you
have thetofirewall.
Awaiting
standards
be used by browsers:Net
• ietf-tram-turn-server-discovery-04
The enterprise firewall in itself can still
• Authentication (in STUN and
draft-ietf-rtcweb-return
be restrictive.
TURN)
• Accounting (usage of this pipe)
• The Carrier provides a “WebRTCWebRTC browsers will then use the network-provided
SBC” in the Trunk CPE
TURN server crossing the enterprise firewall.
18

WebRTC in the Enterprise

Thank You!
© 2015 Ingate Systems AB

19