WebRTC State of the Union

The leader in session delivery network solutions

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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 Intelligence Session Controls

SecurityInteroperability

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Proven widespread interoperabilityServices & applications

Endpoints (IP phones, IADs, MTAs, IP PBX, etc. )

Firewall/NATs

Call control & media gateways

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)

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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

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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

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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

w3c

IETF

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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)

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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.711Opus

Video Codecs

VP8H.264

Transport

DTLS-SRTPSDES-SRTP

STUNICE

TURNMultiplexing

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The signaling of WebRTCTraditional Role of Signaling is the information concerning the namespace, establishment, control, and billing of a communication session.

“…Hi Bob, this is Alice…”

Calling IdentityCalled Identity

Session DescriptionBilling Info

Calling IdentityCalled Identity

Session DescriptionBilling Info? ?

WebRTC has no traditional signaling channel! It relies on a Web, or Traditional Signaling Channel

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The Great Divide

TheWeb UC, IMS,

PSTN

SignalingMedia

SignalingMedia

BeepBeep

RESTHTTP

WebSockets

JSEP

XML

JSON

Opus

SRTP

ICESTUN

TURN

SIP-TLSDiameter

SIMPLESIP-UDPSIP

G.711 RTPG.729

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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

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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

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• 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 CenterUC

BankingIMS

Hosted UCEmergency SVC

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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

WebRTC Use Cases

SIP WebRTC

WebRTC vs. SIP: Bridging a Great Technology Divide

MediaREST

HTTP

WebSockets

JSEPXML

JSON

Opus SRTP

ICE STUNTURN

SIP-TLS

Diam

eter

SIMPL

E

SIP-UDP

SIP

G.711RTP

G.729

XMPPJingleH.323

AMREVRCVP8

H.264

IMS AKA

SIP NAT

iLBC

G.722

RTaudio

RTvideo

Speex

Silk H.263

iSACRTP

SRTP

SIGNALING

MEDIA

SIP-over-WebSocketsREST call control

SRTP en-/de-cryptTranscoding

ICE/STUN/TURN IWF 16

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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

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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

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

Enterprise UC & CC Use Cases

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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

How WebRTC helps contact centers

Customer AgentACD/PBX

CRM

WebRTC-enabled website page

Voice / video / browser view context shared between customer & agent

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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

How WebRTC helps enterprise UC

Enterprise UC

System

RTC apps on employeedevices

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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 + multi-vendor UC

WebRTCEndpoint

(third-partyUC app)

Acme PacketApplication 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)

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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 mobility example

PSTN

Existing Deskphone

UC client app(connected via

WebRTC)WebRTC

SIP

SIP Trunk

Acme PacketApplication

SessionDirector

Enterprise UC System

Employee Tablet

Acme Packet/Plivo Demos

Your Carrier of Choice

SIP

SIP

Origination, DIDs, Termination

TDM to SIP via gateway or direct via SIP

SIP

SIP

Dallas, DC, San Jose, Amsterdam, Singapore

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The following demos must be launched with Google Chrome:

Plivo Conference bridgeGo to nacr.plivo.comFollow directions on screen

Contact Mike Lauricello for the Softphone Call Center demo: Mike Lauricella, BD at Plivo Inccell: 415.857.2518 mdl@plivo.com

Plivo Demos

WebRTC 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

Summary

Questions &

Thank you