2 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 , asking people to click-to-call you or something.
3 Voice Video Data “For free!” From the first WebRTC Conference November 2012 Technically – What is it?
4 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.
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-04 WebRTC is reall...lly coming There are plugins getting WebRTC (including VP8) into IE and Safari today (our test site https://webrtc.ingate.com will hunt for those) https://webrtc.ingate.com Apps (not browsers, but using web view and more) implementing the WebRTC protocols are being built – especially for iPhone (iOS) and Android – needed Enterprise usage may be a driver – many immediate benefits
6 Can The Carrier Also Offer The WebRTC Features to the Enterprise? The Enterprise PBX / UC environment will benefit from: Click-to-call buttons on the company website (context sensitive) New! = The Call Center killer WebRTC application! High quality video conferencing clients The browser is the most superior remote client – always available and anywhere Send http-links as invitations: to be called, or call into a conference bridge etc.
7 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 SBCs are used to connect the PBX/UC (Unified Communication solution) on the LAN to ITSP SIP Trunks on a WAN side. Similarly a network provided turn server between the LAN and the Internet WAN can provide a quality pipe for bandwidth demanding WebRTC media.
8 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
9 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)
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 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 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 draft-schwartz-rtcweb-return-04 WebRTC browsers will then use the network provided TURN server crossing the enterprise firewall.
11 But Remember: Enterprises Want The WebRTC Calls Into the Contact Center Carriers can provide a “WebRTC-SIP gateway in the trunk CPE”, so WebRTC calls goes into the existing auto attendant, queues, forwards, transfers, conference bridges and PBX phones. The same gateway can integrate WebRTC softclients WebRTC by itself bypasses the enterprise SIP UC infrastructure. Voice/Video/Telepresen ce, from passed links and click-to-call buttons etc.
12 Ingate’s public test site is on a WebRTC–SIP gateway combined with an E-SBC. Let’s see WebRTC’s “social calling without numbers” When the receiver (e.g. via IM or ) 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.
13 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 Miami, from a Swedish mobile phone dial , which is SIP trunked to registered at this web site.
14 Offering Web Click-to-Call Into the Enterprise Call Center Using the Carrier Supplied 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 CPE edge device with the WebRTC gateway.
15 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 WebRTC-SIP gateway is required Ingate’s Companion gateway has most of the softclient and an SDK built-in, allowing customized clients to be easily built.
16 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 before 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
17 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 half the time that the pipe is crowded.
18 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