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WebRTC in the Enterprise

© 2015 Ingate Systems AB

Prepared for: Ingate SIP Trunking, UC and WebRTC SeminarsWebRTC in the EnterpriseITEXPO October 2015 Anaheim

By: Karl Ståhl CEO Ingate Systems AB (and Intertex Data AB, now merged)

karl.stahl@ingate.com

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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!

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MPLS

What Can WebRTC Bring to the Enterprise?Something Beyond Just Using Cloud Services?

There Will 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/dialin.html?call=321@pbx.com

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VoiceVideoData

“For free!”

From the first WebRTC Conference November 2012

Technically – What is it?

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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 are already in contact.

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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…

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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 a Webex invitation, the conference is held without the need for 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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To add WebRTC click-to-call buttons to enterprise websites, simply copy some JS-code 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).

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

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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 karl@intertex.se by registered at this web site.

We just saw this demo…

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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 is Required

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

Ingate provides a WebRTC-SIP 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

WebRTC by itself bypasses the SIP PBX/UC infrastructure.

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

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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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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

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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 halve the time that the pipe is crowded.

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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 media issues…a) Getting throughb) Quality

media

ICE

mediaSTUNTURN

SERVER

signaling

WS/WSS

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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 “WebRTC-SBC” in the trunk CPE

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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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 “WebRTC-SBC” in the Trunk CPE

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)

Q-TURN (a network-provided TURN server) will be added in future releases of the Ingate SIParator®.

Awaiting standards to be used by browsers:ietf-tram-turn-server-discovery-04draft-ietf-rtcweb-return

WebRTC browsers will then use the network-provided TURN server crossing the enterprise firewall.

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© 2015 Ingate Systems AB

Thank You!

WebRTC in the Enterprise