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WebRTC State of the Union. The leader in session delivery network solutions. Acme Packet session delivery networks for trusted, first class services & applications. Applications. Over-the-top. Services. Session delivery network ( SDN ). Session Controls. Session Intelligence.
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WebRTC State of the Union The leader in session delivery network solutions
Acme Packet session delivery networks for trusted, first class services & applications Applications Over-the-top Services Session delivery network (SDN) Session Controls Session Intelligence Interoperability Quality Reliability Security IP transport network
Proven widespread interoperability Services & applications Call control & media gateways Firewall/NATs Endpoints (IP phones, IADs, MTAs, IP PBX, etc. )
What is WebRTC? (and why you should care) • What is WebRTC (Real Time Communications)? • Emerging method of web-based RTC • Another type of access framework • Why the hype? • Web: most dynamic, innovative place on planet • RTC has largely been absent • WebRTC delivers RTC to those that create the Web • Why should you care? • WebRTC will be an important access method in the future for SPs, contact centers, and enterprises
Level set • What is The Web? • A distributed system overlaid on The Internet • Made up of interlinked text, images, videos, and other multimedia • “hyper”media • Over 640,000,000 websites on The Internet • Who makes the Web? • Web developers: the largest dev. community on Earth • What makes up the Web? • Made up of servers, databases, and browsers • Loosely linked using protocols and techniques • HTTP, PHP, JavaScript, MySQL, HTML5, CSS, Python, REST…and more! • The Web is very different from traditional communications networks such as IMS, UC, the PSTN
WebRTC • The World Wide Web gets eyes and ears • Real Time voice and video woven into The Web • Acme Packet SDN enables Enterprises, Service Providers, and OTTs to join the revolution
Is it WebRTC or RTCWeb? Both! (kind of) WebRTC is the name for W3C Standard efforts RTCWeb is the name for IETF Standard efforts What is the purpose? To enable Real Time Communications (RTC) in The Web environment What is all the hype about? Web environments are the most dynamic and innovative places in technology…but RTC has largely been absent WebRTC delivers the power of RTC directly into the hands of The Web What is WebRTC? w3c IETF
WebRTC (technically speaking) • A powerful set of audio/video communications tools • Best of breed echo canceler • Audio/video codecs • Video jitter buffer, image enhancer • SRTP stack • Peer-2-peer tools for NAT • STUN, TURN, ICE • …all built into your web browser! • These tools are utilized by Web developers • Accessed via JavaScript
The media engine of WebRTC • Still being finalized • VP8 vs. H.264 highly debated • G.711 and Opus are mandatory voice codecs • Video codecs not yet set • SRTP and STUN/ICE/TURN Required Voice Codecs G.711 Opus Video Codecs VP8 H.264 Transport DTLS-SRTP SDES-SRTP STUN ICE TURN Multiplexing
The signaling of WebRTC Traditional Role of Signaling is the information concerning the namespace, establishment, control, and billingof a communication session. WebRTC has no traditional signaling channel! It relies on a Web, or Traditional Signaling Channel ? Calling Identity Called Identity Session Description Billing Info ? Calling Identity Called Identity Session Description Billing Info “…Hi Bob, this is Alice…”
The Great Divide Beep Beep UC, IMS, PSTN The Web JSON JSEP SIP-TLS Diameter HTTP REST SIP SIMPLE SIP-UDP WebSockets XML Signaling Signaling Media Media ICE Opus G.711 RTP STUN TURN G.729 SRTP
Session delivery challenges • Security • Authentication and authorization • Confidentiality • DoS and overload protection • Identity management • Service reach • Interoperability and interworking • IPv4 <-> IPv6 • Firewalls • Service assurance • QoS • Reliability • Regulatory compliance • Lawful intercept • Call recording WebRTC NOT Exempt
Security • All the challenges/vulnerabilities of Web-based applications • Remote code execution • Format string vulnerabilities • XSS (Cross Site Scripting) • Username enumeration PLUS • Real time communications • Toll fraud/theft of service • Eavesdropping • Session overloads • (SPIT) SPAM over Internet Telephony • Session hijacking
Regulatory compliance Adding WebRTC to SDN means NO compromise on compliance for applications that require it Web-to-Call Center Web-to-Emergency Svc. Web-to-Financial Institute Web-to-Service Provider Lawful intercept Call recording Call Center UC Banking IMS Hosted UC Emergency SVC
WebRTC Use Cases • SP subscriber access viaWebRTC methods • Browser-based RTC to complement SIP offerings • Enterprise UC without thick or thin client soft phones • Easier to maintain & break single UC vendor lock • Contact centers embedding RTC into customer service web pages • Customer satisfaction & lower costs SIP WebRTC
WebRTC vs. SIP: Bridging a Great Technology Divide SIP HTTP SIP-over-WebSockets REST call control SIGNALING IMS AKA XMPP SIMPLE Jingle JSEP H.323 Diameter WebSockets SIP-UDP XML REST SIP-TLS JSON Media SRTP en-/de-crypt Transcoding ICE/STUN/TURN IWF AMR SRTP mEDIA H.263 VP8 Silk RTP EVRC iLBC Opus RTaudio SRTP H.264 G.711 RTvideo RTP G.729 G.722 ICE STUN TURN iSAC Speex SIP NAT
WebRTC: New Access Framework, Same Session Delivery Challenges Service provider, enterprise & contact center • Service reach • Interoperability and interworking • Firewall/NAT traversal • Peer-to-peer vs. core transit • Security • Authentication & authorization • Confidentiality • DoS and overload protection • Identity management • Service assurance • Availability & reliability • QoS, QoE • Regulatory compliance • Call recording • Lawful intercept SIP WebRTC
Addressing BYOD & enterprise mobility challenges • UC devices today use SIP or proprietary methods • WebRTC1 & HTML5 will play a role in BYOD/BYOA • But, new access methods must coexist with SIP-based UC infrastructure & device… • …and Acme Packet helps coexistence UC devices today: SIP, SCCP, H.323, UNIStim, etc. Emerging devices & apps of tomorrow Note 1: WebRTC is used here in a generic context to refer to any type of browser- or web-based real-time collaboration (RTC) app. The term “WebRTC” is a specific term used to describe an approach driven by Google but there are many other comparable app or browser driven RTC approaches
How WebRTC helps contact centers • WebRTC embeds RTC capabilities directly into web pages • No more click-to-call or call back required • Uses the browser’s media capabilities to directly establish customer-agent session (over-the-top, not via PSTN) • Browser context (page view) info can be shared with agent Voice / video / browser view context shared between customer & agent CRM ACD/ PBX Customer WebRTC-enabled website page Agent
How WebRTC helps enterprise UC • Provides UC on any device with a WebRTC-compatible browser • Allows for browser-based UC independent of UC client vendor • Doesn’t require a thick UC client to be installed & maintained • Seamless & secure remote worker access via secure WebRTC methods • Has potential to be compatible with existing enterprise UC system RTC appson employee devices Enterprise UC System
WebRTC + multi-vendor UC • WebRTC endpoints must seamlessly interconnect to existing UC clients & servers • WebRTC Session Director ensures interop between WebRTC signaling & media and existing enterprise UC systems WebRTC signaling methods (i.e. SIP over WebSockets or REST Existing Enterprise UC signaling methods (i.e. SIP) SIP-associated media methods (i.e. RTP) WebRTC-associated media methods (i.e. SRTP & STUN, TURN, ICE) WebRTC Endpoint (third-party UC app) • Acme Packet • Application Session Director • Strong Security • Flexible Interoperability • High Availability Enterprise UC System
Enterprise mobility example • WebRTC enables a new class of devices & applications that can be use for enterprise mobility • WebRTC Session Director enables WebRTC apps to act as an extension to existing UC system desk phones Enterprise UC System SIP Existing Deskphone SIP Trunk PSTN Acme Packet Application Session Director UC client app (connected via WebRTC) WebRTC Employee Tablet
Dallas, DC, San Jose, Amsterdam, Singapore Your Carrier of Choice SIP Origination, DIDs, Termination SIP SIP SIP TDM to SIP via gateway or direct via SIP
Plivo Demos • The following demos must be launched with Google Chrome: • Plivo Conference bridge Go to nacr.plivo.com Follow directions on screen • Contact Mike Lauricello for the Softphone Call Center demo: Mike Lauricella, BD at PlivoInc cell: 415.857.2518 mdl@plivo.com
Summary • WebRTC will revolutionize real-time communications • WebRTC is a new access framework (and poses the typical security & interop challenges) • Acme Packet’s Application Session Controller unifies emerging web-based RTC with SIP-based RTC