Transcript Intertex Data AB, Sweden

WebRTC in the Enterprise
Presentation, Status, Demo
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
ITEXPO January 2015 Miami
By:
Karl Erik Ståhl
CEO Ingate Systems AB
(and Intertex Data AB, now merged)
[email protected]
© 2015 Ingate Systems AB
1
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
2
Technically –
What is it?
Voice
Video
Data
“For free!”
From the first WebRTC Conference November 2012
3
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!”
4
WebRTC Today
 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 has (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
Much other still missing
Network provided TURN-servers are needed (will talk more about), awaited standards
• ietf-tram-turn-server-discovery
• draft-schwartz-rtcweb-return-04
 WebRTC is reall...lly coming
• There are plugins getting WebRTC (including VP8) into IE and Safari today (our test site
https://webrtc.ingate.com will hunt for those)
• Apps (not browsers, but using web view and more) implementing the WebRTC protocols
are being built – especially for iPhone (iOS) and Android – needed
 Enterprise usage may be a driver – many immediate benefits
5
Can The Carrier Also Offer
The WebRTC Features to the Enterprise?
The Enterprise PBX / UC environment will benefit from:
 Click-to-call buttons on the company website (context sensitive) New!
•
= The Call Center killer WebRTC application!
 High quality video conferencing clients
 The browser is the most superior remote client – always available and
anywhere
 Send http-links as invitations: to be called, or call into a conference
bridge etc.
6
Enabling WebRTC Usage in the Enterprise
(WebRTC may be blocked or give bad quality)
Problems to solve when using cloud based WebRTC services:
 Restrictive enterprise firewalls block WebRTC
• For WebRTC demonstration/evaluation, Carrier’s today have to use their
guest WiFi instead of their own LAN…
 Data-crowded enterprise firewalls means bad quality, QoS
SBCs are used to connect the PBX/UC (Unified Communication
solution) on the LAN to ITSP SIP Trunks on a WAN side.
Similarly a network provided turn server between the LAN and the
Internet WAN can provide a quality pipe for bandwidth demanding
WebRTC media.
7
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
8
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
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)
9
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
Q-TURN
(a to
Network
Provided
TURN server) will be
PARALELL
the firewall
and setup
Q-TURN Enables QoS and More:
added
in future
of the
Ingate SIParator®.
the media
flowsreleases
there under
control.
• Prioritization and traffic-shaping
• Security is back in the right place –
• Diffserv or RVSP QoS over the
Awaiting
standards
be used by browsers:
Where you
have thetofirewall.
Net
ietf-tram-turn-server-discovery
• draft-schwartz-rtcweb-return-04
The enterprise firewall in itself can still
• Authentication (in STUN and
be restrictive.
TURN)
• Accounting
browsers
then use the network
provided (usage of this pipe)
• WebRTC
The Carrier
provideswill
a “WebRTCTURN
crossing
SBC inserver
the Trunk
CPE”the enterprise firewall.
10
But Remember: Enterprises Want The WebRTC Calls
Into the Contact Center
WebRTC by
itself bypasses
the enterprise
SIP UC
infrastructure.
Voice/Video/Telepresen
ce, from passed links
and click-to-call
buttons etc.
Carriers can provide a
“WebRTC-SIP gateway in the
trunk CPE”, so WebRTC calls
goes into the existing auto
attendant, queues, forwards,
transfers, conference bridges
and PBX phones.
The same gateway can
integrate WebRTC softclients
11
Ingate’s public test site is on a WebRTC–SIP gateway combined with
an E-SBC. Let’s see WebRTC’s “social calling without numbers”
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.
12
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 Miami, from a Swedish mobile phone dial +46812205614, which is SIP trunked
to [email protected] registered at this web site.
13
Offering Web Click-to-Call Into the Enterprise Call Center
Using the Carrier Supplied CPE With WebRTC Gateway
Adding WebRTC click-to-call
buttons to the enterprise website
is simply to copy some JS-code
into the enterprise website.
Deployment and installation will
be the same as for SIP trunking –
with the trunk CPE already at the
demarcation point (with WAN and
LAN PBX connection) the
interface is the same as for carrier
SIP trunking using an CPE edge
device with the WebRTC gateway.
14
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 WebRTC-SIP gateway
is required
Ingate’s Companion gateway has
most of the softclient and an SDK
built-in, allowing customized
clients to be easily built.
15
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 before 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.
16
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.
17
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 bad video and
hacking sound, when data traffic is heavy.
4G/LTE can be excellent, but disturbed when data-crowded and weak signal
WiFi can be perfect – or unusable if data-crowded
18