1. WebRTC State of the Union The leader in session delivery network solutions

2. 2 Acme Packet session delivery networks for trusted, first class services & applications IP transport network Session delivery network (SDN) Applications Services Over-the-top Quality Reliability Session IntelligenceSession Controls Security Interoperability

3. 3 Proven widespread interoperability Services & applications Endpoints (IP phones, IADs, MTAs, IP PBX, etc. ) Firewall/NATs Call control & media gateways

4.  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 What is WebRTC? (and why you should care) 4

5.  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 Level set 5

6.  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 WebRTC 6

7. What is WebRTC?  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 7

8.  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 WebRTC (technically speaking) 8

9.  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 The media engine of WebRTC Voice Codecs G.711 Opus Video Codecs VP8 H.264 Voice Codecs G.711 Opus Video Codecs VP8 H.264 Transport DTLS-SRTP SDES-SRTP STUN ICE TURN Multiplexing Transport DTLS-SRTP SDES-SRTP STUN ICE TURN Multiplexing 9

10. The signaling of WebRTC Traditional Role of Signaling is the information concerning the namespace, establishment, control, and billing of a communication session. “…Hi Bob, this is Alice…” Calling Identity Called Identity Session Description Billing Info Calling Identity Called Identity Session Description Billing Info Calling Identity Called Identity Session Description Billing Info Calling Identity Called Identity Session Description Billing Info ? ? ? ? WebRTC has no traditional signaling channel! It relies on a Web, or Traditional Signaling Channel 10

11. The Great Divide Signaling Media Signaling Media Beep 11

12. 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 12

13. 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 13

14. 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 Regulatory compliance Call Center UC Banking IMS Hosted UC Emergency SVC 14

15.  SP subscriber access via WebRTC 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 15 WebRTC Use Cases SIP WebRTC

16. WebRTC vs. SIP: Bridging a Great Technology Divide Media SIP-over-WebSockets REST call control SRTP en-/de-crypt Transcoding ICE/STUN /TURN IWF 16

17. 17 WebRTC: New Access Framework, Same Session Delivery Challenges Service provider, enterprise & contact center SIP WebRTC 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

18.  UC devices today use SIP or proprietary methods  WebRTC 1 & 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 18 Addressing BYOD & enterprise mobility challenges 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 UC devices today: SIP, SCCP, H.323, UNIStim, etc. Emerging devices & apps of tomorrow

19. Enterprise UC & CC Use Cases

20.  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 20 How WebRTC helps contact centers Customer Agent ACD/ PBX CRM WebRTC-enabled website page Voice / video / browser view context shared between customer & agent

21.  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 21 How WebRTC helps enterprise UC Enterprise UC System RTC apps on employee devices

22.  WebRTC endpoints must seamlessly interconnect to existing UC clients & servers  WebRTC Session Director ensures interop between WebRTC signaling & media and existing enterprise UC systems 22 WebRTC + multi-vendor UC WebRTC Endpoint (third-party UC app) Acme Packet Application Session Director Strong Security Flexible Interoperability High Availability WebRTC signaling methods (i.e. SIP over WebSockets or REST Enterprise UC System Existing Enterprise UC signaling methods (i.e. SIP) WebRTC-associated media methods (i.e. SRTP & STUN, TURN, ICE) SIP-associated media methods (i.e. RTP)

23.  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 23 Enterprise mobility example PSTN Existing Deskphone UC client app (connected via WebRTC) WebRTC SIP SIP Trunk Acme Packet Application Session Director Enterprise UC System Employee Tablet

24. Acme Packet/Plivo Demos

25. Your Carrier of Choice SIP Origination, DIDs, Termination TDM to SIP via gateway or direct via SIP SIP Dallas, DC, San Jose, Amsterdam, Singapore

27.  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 Plivo Inc cell: 415.857.2518 mdl@plivo.com 27 Plivo Demos

28. WebRTC Summary

29.  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 Summary

30. Questions & Thank you