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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. - PowerPoint PPT Presentation
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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
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