1 Enabling WebRTC in the Enterprise A) How Can WebRTC Enhance the Enterprise PBX/UC Solution? B)...

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Enabling WebRTC in the Enterprise

A) How Can WebRTC Enhance the Enterprise PBX/UC Solution?

B) Will SIP Trunking E-SBCs Include WebRTC Support?C) Can Carriers Provide a "WebRTC-Ready" Access?

© 2014 Ingate Systems AB

Prepared for: Ingate’s SIP Trunking, UC and WebRTC Seminars ITEXPO August 2014 Las Vegas

By: Karl Erik Ståhl CEO Ingate Systems AB

(and Intertex Data AB, now merged)[email protected]

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MPLS

What Can WebRTC Bring to the Enterprise? Will There be an Enhanced “Enterprise Social Network”?

SIP System

Data & VoIP LAN

SIParator®

But: No Numbers!?

Passing links?

Browsers as Softclients!

HD Multimedia Telepresence

LAN

CompanyWeb

Server

SIP

Pass a WebRTC link over IM or an email, asking people to click-to-call you or something. http://companion.smartcomp.com/[email protected]

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From the first WebRTC Conference November 2012

- Where are we now?- Is it for the enterprise?- What is it all about?

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VoiceVideoData

“For free!”

From the first WebRTC Conference November 2012

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BASICSWhat WebRTC Does:

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

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

VoiceVideoData

“For free!”

What WebRTC Does NOT Do:

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

------------

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

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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

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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. Video conference between browsers

2. Inviting to Webex conference

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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.

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And a Click-to-Call Website is Great

You are on the Web – Wanna talk?

– Don’t pick up your phone. Just click! Communicate with voice, video and data and screen.

Don’t Dial, Just click!

Calling by Clicking at a Web Page

A great application

Do we need more than the company website and the always available browser?

CompanyWeb Server

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.

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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.

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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 WebRTC-SIP Gateway Required

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An always-available quality IMS-RCS client that hopefully resolves the NAT/ FW issue.

But will carriers ever peer the IMS way instead of just POTS peering?

A WebRTC – SIP gateway is required

The IMS view: Finally a softclient for the IMS+RCS multimedia telephone network!

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B) Will SIP Trunking E-SBCs Include WebRTC Support?

There are two questions to address:1) WebRTC into the enterprise (as it is)2) WebRTC integrated with the PBX / UC Solution

Infrastructure





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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.

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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.

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LAN

CompanyWeb

Server

WebRTC - Like All Real-Time Communication Protocols - has a NAT/Firewall Traversal Problem

LAN

CompanyWeb

Server

Firewalls do not allow unknown incoming signaling and media is a “surprise” (just like SIP)

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

There are issues…a) Getting throughb) Quality

media

ICE

mediaSTUNTURN

SERVER

signaling

WS/WSS

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ICE/STUN/TURN Means There Are No SBCs

• ICE was developed and standardized for SIP (long after SIP), but not used much for SIP… It is supposed to work without the NAT/firewall being aware of what is traversed (like Skype).

• Sometimes a TURN-server is required

• With restrictive enterprise firewalls – ICE is not sufficient.

• Best: WebRTC is end-to-end and does not encourage application-specific networks

• Worst: The firewalls are unaware of what is being traversed – Quality: The firewall cannot prioritize RTC traffic.

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How Do We Get WebRTC Into The Enterprise?

LAN

CompanyWeb

Server

TURN SERVER

Will WebRTC work through the enterprise firewall?

What about Quality? (prioritization, traffic-shaping in the firewall. Diffserv or RSVP for the network?) The firewall is often the congestion point.

There are remedies

media

Q-TURN

LAN

CompanyWeb

Server

media

Q-TURN

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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

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)

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LAN

CompanyWeb

Server

media

That was Getting WebRTC in Itself Into the LAN…But, Where did the Enterprise PBX/UC Infrastructure go?

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

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We Want the Calls Into the Contact Center!

LAN

CompanyWeb

Server

SIP

WS

media

Such Gateway into the enterprise PBX/UC-solution can reintroduce the PBX/UC’s Auto Attendant, Queues, Forwards, Transfers, Conference Bridges, PBX Phones…

It’s Required! LAN

CompanyWeb

Server

media

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LAN

CompanyWeb

Server

media

Same When Passing a Link I Want to be Reached at my Current PBX Client!

Same

problem

Same

solution

An E-SBC could include:

A WebRTC SIP gateway bringing the PBX/UC infrastructure back into WebRTC calls

LAN

CompanyWeb

Server

SIP

WS

media

Pass WebRTC link over IM or email, inviting people to click-to-call you or something. http://companion.smartcomp.com/[email protected]

The call should reach the enterprise via the SIP PBX/UC infrastructure with all its features.

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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, is of course a dream.

The E-SBC is usually hooked up to the LAN and the Internet – A good place to put the softphone browser interface in.

An E-SBC could include:A WebRTC SIP Gateway allowing easy creation of browser-based softphones for the PBX/UC solution.

The E-SBC facing the Internet and the NAT/Firewall traversal method (ICE/STUN/TURN) of WebRTC itself, will allow remote users and make mobility solutions “automatic”.

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“Automatic Mobility” is a Major Feature

PSTN

Data & VoIP LAN

IP-PBX

SIParator®

GWSIP System

SIP Trunking Provider

Today, only the best E-SBCs support remote SIP clients and also do Far-End NAT Traversal (FENT). And mobile operators’ mobility solutions require a lot and gives few of the UC features.

Remote User

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The WebRTC Browser for Local and Remote Users

The browser becomes your softphone by surfing to a webpage. (An application build of WebRTC browser components can do the same, if preferred.)

The browser can run a full SIP Client in JS or just “a page of JS” where the SIP UA is in the gateway (often preferred).

Any Web server can be used.

The interface to the PBX/UC can be as a registered client or via the SIP trunking interface.

The registrar can be in the PBX/UC or in the gateway (if advanced and extended.)

Authenticated websockets (WS or WSS) interface between the browser and the gateway running either SIP over websockets or the simpler protocol when the UA is in the gateway.

Q-TURN in an firewall or E-SBC can also be the STUN/TURN server to traverse remote NAT/firewalls.

A WebRTC to SIP gateway will also benefit from the interoperability and security features of an E-SBC and the ability to work in parallel with the enterprise firewall.

LAN

(Any)Web

ServerWS

1

media

HTTP

SIP

SIP TRUNK(Opt) Web

Server

2 HTTP

WSmedia

media

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Answer to:Will SIP Trunking E-SBCs Include WebRTC Support?

They could - but much more is required!

Two new product classes are needed:1) The Q-TURN firewall, and2) The WebRTC-SIP gateway

These are complex products, but both are at the same location and use the same interfaces as the E-SBC device, at the enterprise edge, between the private enterprise LAN and the global network (the Internet). And just like the E-SBC, the Gateway has to interoperate with the PBX/UC SIP.


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C) Can Carriers Provide a "WebRTC-Ready" Access?





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From POTS to Telepresence – A Gigantic Step

• 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

Pre-AM radio 3.5 kHz voice to 20 kHz audio and 3.5 Mbps HD video

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Quality Experiences

WebRTC does have telepresence quality capacity and that is important:

Reactions after an employment interview oversea s: “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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VoIP in the Application-Specific Telephone Network has Not Helped – It isn’t Even Good for Faxing Anymore

The Telephony application is still only POTS, someday maybe RCS, but… Carriers are Peering their VoIP networks PSTN Style, degrading quality, interop…

It is even destructive for the 160-years-old Fax service!**

And carrier’s billing is by voice minutes – Far away from any UC! And where did the reliability, scalability and good performance of IP networks go?

** Mike Coffee, CEO of Commetrex: Work in progress by SIP Forum’s FoIP Task Group and the i3 Forum. T.38 works fine in one hop!

Computers, Internet and related applications follow Moore’s law… Telephony has over a 20 yearsperiod brought great mobility and popular text

messaging (SMS)*, but otherwise shown a NEGATIVE Moore’s law (below)… WebRTC is on the Internet, has to stay there, but needs quality!

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Locally, Carriers Have Long Since Provided Quality Traffic Over the Broadband Connection (but Wasted it at the Delivery)

TR-069TR-069InternetInternet

IP-TVVoD

IP-TVVoD

IMSVoIP

IMSVoIP

VLANs or ADSL Virtual Circuits

The Multimedia LAN

WiFi

Telepresence

But we need the real-time traffic into the LAN

– Not on an RJ11 = POTS

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!)

RJ11

Prioritizing real-time traffic over best-effort traffic will be valuable to both carriers and users!

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Quality Traffic on the Internet: The Internet+ Model

There are (disabled) quality mechanisms on the Internet – Enable and provide that quality to the users!

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.

SIP Connect 1.1

Internet+

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.

Real-time traffic is more valuable.

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The TURN Server IN the Firewall Fixes Traversal, Quality and can Measure: Q-TURN in the Firewall

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

• Enterprise firewall can still be restrictive

Q-TURN

Q-TURN Enables QoS and More:• Prioritization and Traffic Shaping• Diffserve or RVSP QoS over the

Net• Authentication (in STUN and

TURN)• Accounting (usage of this pipe)

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Q-TURN as the Carrier Broadband Delivery

Sell a “WebRTC-Ready” Access!

• Why only deliver Best-Effort Data?

• Quality Traffic - prioritized real-time traffic within the same pipe - is highly valuable, but cost no more bandwidth to produce!

• OTT can be more than data delivery. Telepresence in your pocket!

Q-TURN at the Carrier Demarcation Points• Mobile (replace the DPI behind the

Cell Tower)• Enterprise and SMB delivery• Residential delivery – Fits

embedded CPEs

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A Healthy Win-Win Economy for Users and Carriers

Telephony Income (highly charged)

Low-Charged Internet Bandwidth

Data

Limited Quality RTC

SIP, WebRTC = Telephony+

Skype etc.

Bandwidth Usage

Data

RTC

Quality Bandwidth New Value

Now I I I I I I I

E-SBCs with SIP proxies and TURN servers at the carrier demarcation point allow the already available bandwidth to be used for high quality real-time traffic delivery in addition to the best-effort data delivery.

The future loss of income from specific telephone networks, may be replaced by prioritized OTT and Internet traffic, counted separately from less-valuable data traffic and especially separate from movies and TV (that have very low value per gigabyte). The Internet+ model applies to fixed, Wi-Fi and mobile broadband delivery for both SIP and WebRTC traffic.

Decreasing telephony income being replaced by real-time traffic over data crowed OTT and Internet best effort traffic is a lose-lose situation for both carriers and users.

Delivering prioritized, separately counted high quality multimedia traffic over existing OTT and Internet bandwidth, is a win-win solution for both carriers and users

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