1. 1 WebRTC Introduction and Overview © 2015 Ingate Systems AB Prepared for:Ingate SIP Trunking, UC and WebRTC Seminars WebRTC Introduction and Overview ITEXPO October 2015 Anaheim By:Karl Ståhl CEO Ingate Systems AB (and Intertex Data AB, now merged) karl.stahl@ingate.com

2. 2 WebRTC - A Google Initiative  Webifying Real-Time Communications OR IS IT  Going After the $2,000 Billion Telephony Industry  For global real-time communication, we still use telephony. Even if phones have become mobile, the telephone service is pre-AM radio quality and 50 years overdue  Better services, e.g. Skype, are free, may be big/huge but still proprietary island (We use phone numbers to call people and ask them to sign into Skype…) And inside the enterprise is UC.  But nowadays we most often find each other on the web: Why don’t we just connect person-to-person real-time in the browser then?

3. 3 There is Power Behind – It Will Happen! Google acquired GIPS (known from the Skype voice engine etc.) for 80 MUSD just to implement WebRTC in Chrome. And another 130 MUSD for the VP8 licence free (H.264 like) video codec. “Google recently released nearly $70M worth of open source code to the world…” Intense standardization work (since 2011): IETF - the protocols W3C - the Web application API (JS) From the first WebRTC Conference November 2012

4. 4 Voice Video Data “For free!” From the first WebRTC Conference November 2012 Technically – What is it?

5. 5 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!” 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. From the first WebRTC Conference November 2012

6. 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…

7. 7 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? Company Web 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.

8. 8 Or is it bringing HD Telepresence Quality Video Conferencing, to everyone’s desktop. There will be demos in this hall later. 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.

9. 9 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!

10. 10 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 But: No Numbers!? Passing links… Browsers as Softclients! HD Multimedia Telepresence LAN Company Web 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

11. 11 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 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 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-00  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)https://webrtc.ingate.com 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!

12. 12 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

13. 13 LAN Company Web Server 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, 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 through b)Quality media ICE media STUN TURN SERVER signaling WS/WSS

14. 14 Locally, Carriers Have Long Since Provided Quality Traffic Over the Broadband Connection (but Wasted it at the Delivery) TR-069 Internet IP-TV VoD IP-TV VoD IMS VoIP IMS VoIP VLANs or ADSL Virtual Circuits The Multimedia LAN WiFiWiFi 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!

15. 15 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

16. 16 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 data firewall can still be restrictive. Carriers can 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 Offered with the network access Accounting (usage of this pipe)