1. What’s Next For SIP Trunking? WebRTC in the Enterprise
Carriers Enabling and Bringing WebRTC Features With Their Trunks
Prepared for: Ingate SIP Trunking, UC and WebRTC Seminars
Combination of the two presentations:
What’s Next For SIP Trunking? & WebRTC in the Enterprise
ITEXPO January 2015 Miami
By: Karl Ståhl
CEO Ingate Systems AB (and Intertex Data AB, now merged)
© 2015 Ingate Systems

2. What Can WebRTC Bring to the Enterprise
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 , asking people to click-to-call you or something.
MPLS
SIP System
LAN
Company
Web Server
SIP
But:
No Numbers!?
Passing links…
Browsers as Softclients!
HD Multimedia Telepresence
SIParator®
Data & VoIP LAN

3. Technically – What is it?
Voice
Video
Data
“For free!”
From the first WebRTC Conference November 2012

4. BASICS 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.
BASICS
What 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).
Voice
Video
Data
“For free!”

5. 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-05
WebRTC is reall...lly coming
There are plug-ins bringing WebRTC (including VP8) into Microsoft’s IE and Apple’s Safari today (Ingate’s test site installs those)
Apps (Applications - not browsers) implementing the WebRTC protocols are being built – especially for iPhone (iOS) and Android – Those are needed for real quality telephony and are expected mid 2015.
Enterprise usage will be a driver – Many immediate benefits

6. So Far SIP Trunking Has Been About Replacing Telephony Wires With IP Wires What’s Next?
MPLS
SIP System
Data & VoIP LAN
SIParator®
It’s about how we connect beyond the LAN. SIP Trunking has brought:
Lower cost, connecting to POTS, improved business continuity, disaster recovery etc. That isn’t the end.
Good E-SBCs connecting remote users to the PBX/UC is a popular step.
But what’s next to come?
More federation?
Unified Communication (UC) beyond the LAN
UC/multimedia global connectivity?
IMS (RCS, VoLTE) – Will those even peer?
WebRTC extensions - What’s that?

7. Today Carriers Offer Voice Services Over Their SIP Trunks WebRTC Opens For Offering More And Better
Typically delivered via a CPE (e.g. an E-SBC) as a demarcation point
SIP Trunks for POTS Connectivity
Hosted with desktop phones on LAN
Enterprise PBX with desktop phones
WebRTC is a game changer: More and Richer. The CPE of the “SIP Trunk” is where to do it:

8. Problems to solve when using cloud based WebRTC services:
(1!) 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
Carriers deploy SIP trunks via SBCs (to reach the PBX/UC).
For WebRTC carriers can provide quality pipes for more valuable communication.
(Remember - No one can be without WebRTC!)
Demo: Browser to Browser to myself

9. WebRTC - Like All Real-Time Communication Protocols - Has a NAT/Firewall Traversal Problem
LAN
Company
Web 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. However WebRTC prescribes, ICE/STUN/TURN to fool the firewall to let the real- time traffic through (similar to Skype.)
Websockets, WS/WSS, is often used to hold the signaling channel open
There are media issues…
Getting through
Quality
WS/WSS
ICE
media
STUN
TURN SERVER
signaling
media
Company
Web Server
LAN

10. 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 Trunking 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)

11. 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 PARALELL 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 (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
draft-schwartz-rtcweb-return-05
WebRTC browsers will then use the network provided TURN server crossing the enterprise firewall.
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)

12. (2!) Can The Carrier Also Offer The WebRTC Features to the Enterprise PBX / UC?
The Enterprise PBX / UC environment will benefit from:
Click-to-call buttons on the company website (context sensitive) New!
= The Contact Center killer WebRTC application!
High quality video conferencing clients
The browser is the most superior remote client – always available and anywhere (Try it out at:
Send http-links as invitations: To be called, or call into a conference bridge etc. (Try it out at:
Demo: Browser to Browser to myself

13. But Remember: Enterprises Want The WebRTC Calls Into the Contact Center
Voice / Video / Telepresence, from passed links and click- to-call buttons etc.
WebRTC by itself bypasses the enterprise SIP PBX/UC infrastructure.
Carriers can provide 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 provide WebRTC softclients

14. Demonstration of the call center Click-to-Call killer application, using Ingate’s local test site here and the 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.
DEMO: Call yourself with the link at webrtc.ingate.com
(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 , which is SIP trunked to registered at this web site.

15. Offering Web Click-to-Call Into the Enterprise Call Center Using the Carrier Supplied Ingate 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 Ingate CPE edge device with the WebRTC gateway.
Demo: Browser to Browser to myself

16. The public test site is Ingate’s WebRTC–SIP gateway combined with Ingates E-SBC. Let’s see WebRTC’s “social calling without numbers”
When the receiver of this link (e.g. via IM or ) clicks it, a window pops-up and sets up a video conference between the WebRTC browsers. No numbers, no SIP, no PSTN involved.
DEMO: Call yourself with the link at webrtc.ingate.com
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.

17. 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
DEMO: Call all of these ad hoc using the group button at
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 smartphone Galaxy S5 using the Chrome browser and Ms. Time telling time in Sweden at telephony number

18. With Good WebRTC Soft Clients For Smartphones Wireline Carriers Can be an OTT Virtual Mobile Operators
We hardly use desktop phones (no matter how good) anymore
Mobile smartphones are used everywhere
WebRTC’s superior remote connectivity: Talk everywhere you can surf!
WebRTC smartphone clients - OTT over LTE and WiFi - is all you need and WebRTC is far superior to the mobile voice we know.
With smartphone WebRTC clients connected to the enterprise SIP trunked PBX and UC (Unified Communications) environment, the SIP trunk provider becomes a virtual mobile operator!
WebRTC soft clients – both as Web clients and as apps are coming, even for iPhone. Be aware!
DEMO: Browser to Browser to myself

19. From 3.5 kHz Voice to HiFi HD Telepresence Quality!
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, , 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 only half the time that the pipe is crowded. 19

20. 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
Ingate’s Q-TURN technology is the long term remedy
Ring from nexus7 using camera2, Put it so the audience can see themself.