2 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 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://email@example.com
3 From the first WebRTC Conference November 2012 -Where are we now? -Is it for the enterprise? -What is it all about?
4 Voice Video Data “For free!” From the first WebRTC Conference November 2012
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.
6 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
7 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
8 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.
9 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.
10 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 firstname.lastname@example.org by registered at this web site.
11 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
12 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!
13 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.
14 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.
15 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 issues… a)Getting through b)Quality media ICE media STUN TURN SERVER signaling WS/WSS
16 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)
17 LAN Company Web 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
18 We Want the Calls Into the Contact Center! LAN Company Web 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 Company Web Server media
19 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
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 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
21 Locally, Carriers Have Since Long 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!
22 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.