Transcript Intertex Data AB, Sweden

WebRTC in the Enterprise
Presentation, Status, Demo
Prepared for: WebRTC Pavilion
ITEXPO August 2014 Las Vegas
By:
Karl Erik Ståhl
CEO Ingate Systems AB
(and Intertex Data AB, now merged)
[email protected]
© 2014 Ingate Systems AB
1
What Can WebRTC Bring to the Enterprise?
Will There 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
Company
Web
Server
SIParator®
But:
No Numbers!?
Passing links?
Data & VoIP LAN
LAN
SIP
Browsers as
Softclients!
HD Multimedia
Telepresence
2
From the first WebRTC Conference November 2012
- Where are we now?
- Is it for the enterprise?
- What is it all about?
3
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 already are in contact.
• “Handles” NAT/FW traversal
(ICE, STUN, TURN) – fooling
firewalls (like Skype).
Voice
Video
Data
“For free!”
5
WebRTC Today
 Standards (IETF and W3C WGs started 2011) progressing slowly
• Mandatory video codec (VP8, VP9, H.264, H.265) not agreed upon
• IETF war will reopen in September
• Complex and advanced, but still closing in
 In some browsers: Google’s Chrome, Mozilla’s Firefox and Opera
• Impressive in many aspects, but not complete, not standard-compliant (of course) – but
close to, flaws and bugs still hindering some usage
• Not yet in Microsoft’s Internet Explorer and Apple’s Safari (expect when H.264 is
mandatory and standards set)
• Plug-ins, WebRTC browser components and libraries appearing to support more
platforms and building apps
 Still few real applications and services
 Enterprise usage may be a driver – many immediate benefits
6
What are the WebRTC applications? Social Calling…
Calling Without Phone Numbers
• You already are 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…
Demo:
1.
2.
Video conference between browsers
Inviting to Webex conference
7
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 an Webex invitation, the
conference is held without needing any phones.
8
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.
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.
(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 Required
11
The IMS view: Finally a softclient for the
IMS+RCS
multimedia telephone network!
An always-available quality IMS-RCS client that hopefully resolves the NAT/ FW issue.
A WebRTC –
SIP gateway
is required
But will carriers
ever peer the IMS
way instead of just
POTS peering?
12
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 half the time that the pipe is crowded.
13
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.
14
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 prescribes
ICE/STUN/TURN to fool
the firewall to let the
real-time traffic through
(similar to Skype.)
 Websockets, WS/WSS,
often used to hold the
signaling channel open
media
Company
Web
Server
STUN
TURN
SERVER
WS/WSS
ICE
LAN
 There are issues…
a) Getting through
b) Quality
15
The TURN Server IN the Firewall Fixes Traversal, Quality and can
Measure Usage: Q-TURN in the Firewall is “like an E-SBC”
A novel Ingate view:
Knock-knock; Give my media a Quality
Pipe
• Regard ICE as a request for real-time
traffic through the firewall. Interpret
the STUN & TURN signals in the
firewall
• Have the STUN/TURN server
functionality IN the firewall and setup
the media flows under control
• Security is back in the right place The firewall is in charge of what is
traversing
• The enterprise firewall can still be
restrictive
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)
16
That was Getting WebRTC in Itself Into the LAN…
But, Where did the Enterprise PBX/UC Infrastructure go?
media
Company
Web
Server
LAN
Enterprises have their
own “Social Network” –
their PBX/UC solution.
The E-SBC is already
hooked to the PBX SIP
Trunking interface and
often facing the Internet.
A good place to put the
“Gateway” in.
The E-SBC could
include:
 A WebRTC SIP
Gateway bringing
the PBX/UC
infrastructure back
into WebRTC calls
LAN
17
We Want the Calls Into the Contact Center!
media
Company
Web
Server
LAN
Such Gateway into the
enterprise PBX/UCsolution can
reintroduce the
PBX/UC’s Auto
Attendant, Queues,
Forwards, Transfers, Company
Web
Conference Bridges, Server
PBX Phones…
It’s Required!
LAN
media
WS
SIP
18
From POTS to Telepresence – A Gigantic Step
Pre-AM radio 3.5 kHz voice to
20 kHz audio and 3.5 Mbps HD video
• WebRTC has the potential of telepresence quality: Opus HiFi
audio and VP8 / H.264 HD video
• While taking the real-time traffic to the Internet/OTT…
• Internet has the largest bandwidth
• But it is NOT “Just About Bandwidth”
• Data crowded networks
• Surf, email, file transfer fill the pipes
• Layer 4 QoS: UDP favored over TCP is not sufficient
• We need to prioritize - Level 3 QoS
19
Quality Experiences
 WebRTC does have telepresence quality capacity and that is important:
Reactions after an employment interview overseas : “Twice as valuable
as a phone interview”, “No need to travel to interview in person”
 Observations without prioritization (QoS):
Fixed access (100 Mbps in a 20 person enterprise, 2/10 Mbps for residential):
Excellent when non-intensive data usage.
3G mobile (2-2.5G is unusable): Often usable, but periods of shrinking video
screen and hacking sound, when data traffic is heavy. There are (still) carriers
making unusable on purpose.
4G/LTE can be excellent , but disturbed when data-crowded and weak signal
WiFi can be perfect – or unusable if data-crowded
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Locally, Carriers Have Since Long Provided Quality Traffic
Over the Broadband Connection (but Wasted it at the Delivery)
But we need the real-time
traffic into the LAN
Internet
IP-TV
VoD
IMS
VoIP
– Not on an RJ11 = POTS
TR-069
RJ11
VLANs or ADSL
Virtual Circuits
WiFi
The Multimedia LAN
Telepresence
And today’s SIP trunking
sends the media into the
POTSoIP structure –
Thus becoming a PSTN
gateway. (SIP devices
could instead route to
the other endpoint!)
Prioritizing real-time
traffic over best-effort
traffic will be valuable to
both carriers and users!
21
Quality Traffic on the Internet: The Internet+ Model
There are (disabled) quality mechanisms on the Internet – Enable and
provide that quality to the users!
We need a “toll
to enter the
highway” or
everyone will
chose priority to
surf faster – and
we will be back to
the same priority.
SIP Connect 1.1
Internet+
Real-time traffic is
more valuable.
WebRTC is end-to-end. ICE/STUN/TURN is used through NAT/firewalls
There is no WebRTC proxy like in SIP that can classify, prioritize and measure
calls. A TURN server at the delivery point can fill those needs: Q-TURN.
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