WebRTC - Bridging Web and SIP Worlds

Shambhu Dayal Rai

srai@sonusnet.com

Distinguished Solution Architect,

Sonus Networks

2

WebRTC: Transforming Communications

Free, open Project to

enable browser with

RTC capabilities via JS API

ITS EASY for developers,

users & businesses

RTC Media engine build

in the browser,

voice/video codecs, ICE, DTLS-SRTP

Enable RTC on web apps No client or

plug-in required, JS on Webpage defines the

client behavior

What? Why? How? Value?

3

Real time communication System

• User registry, presence, location

• User profiles, preferences, Services

User management

• Establish, update and terminate sessions and services,

• Multimodal sessions, IM, voice, video, file share etc.

Session Management

• Interconnecting various communication islands

• Interwork Signaling and media

• NAT traversal

Inter-connection

• High Availability

• Session Rehydration for failures

• mobility

QoS & Reliability

• DOS, policing,

• Authorization & Call admission controls,

• Routing and Media policy

Policy & Security

An use case..

5

myBank has call center & WebSite

Employee

Customers

Partners

Authorization DB

IMS Core / PSTN

Mybank Call center

myBank.com

However, Website and call

center are not connected,

leading to disjointed user

experience !!

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WebRTC at myBank.com

SBC

Employee

Customers

Partners

Identity Provider

IMS Core / PSTN

Call center

WebRTC GW

myBank.com

• WebRTC Cloud enables

RTC on mybank.com.

• Agent added to the

multimodal Web Session.

• Rich user experience

Connecting WebRTC with the World..

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WebRTC - A World of Triangles

Non-real Time Traffic

e.g. Web Pages, Pictures,

Preformatted Video

Web World Web RTC

RT Voice RT Video RT Data

• With WebRTC any

Website can become a

“stand alone”

communication provider

for its community

• Each Service is an Island

• PSTN is NOT needed if

endpoints are both on the

same application

RT Media

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Communicating Beyond the Web Domain

Web RTC Web World Rest of World (Telco World)

Non-standardized (potentially custom) signaling needs to map to SIP

Media profile for WebRTC is “incompatible” with general telco VoIP and Video

Signalling

Media

Incompatible Domains

From Triangles to Trapezoids

10

Tech Divide : WebRTC vs Telco

IPv4 / IPv6 Network

Media

Transport

Media

Description

Signaling

Transport

Signaling

Protocol

Codecs

Identity

Media Plane

Control Plane

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WebRTC at myBank.com

TURN Server

SBC

Policy DB

Employee

Customers

Partners

Identity Provider

IMS Core / PSTN

TURN Server

Call center

WebRTC GW

myBank.com

User loads mybank.com

and loads webRTC client

Mybank authorize users

provides WebRTC client

WebSocket connection

established,

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WebRTC at myBank.com

TURN Server

SBC

Policy DB

Employee

Customers

Partners

Identity Provider

IMS Core / PSTN

TURN Server

Call center

WebRTC GW

myBank.com

User

decides to

make call

• Interwork Web to SIP signaling

• Inter work Web to Telco IDs

• Enforce enterprise policy for

media, applications, Auth etc

• STUN/ICE for NAT

• Interwork DTLS-SRTP to RTP

• Transcoding

• DOS, Policing, LI, recording.

13

WebRTC – Media Paths

TURN Server

SBC

Policy DB

Enterprise - A

Enterprise - B

Enterprise - C

Identity Provider

IMS Core / PSTN

TURN Server

Call center

WebRTC GW

WebRTC Access to UC System.

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User Registration based on Web Authentication

UE

1. HTTPS to web server,

returning auth grant

assertion

2. Open secure WebSocket (CORS)

3. LOGIN request with ID assertion

5. OK response OK

4. SIP

REGISTER

SBC PBXWeb

ServerWebRTC

GW

Validate Web Server assigned

ID is used by subscriber.

4. SIP

REGISTER

OK

AD

LDAP (Authentication)

Web Server Allocates or fetch the User-ID for User .

SBC registers the User to PBX, At this point user is registered and ready to make receive sessions.

Reliability and Quality over Internet..

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Overall latency

Length of path (speed of light)

Number of nodes in path

Available bandwidth

Type of access

End-to-end bandwidth

“Quality of the stream”

Packet delays

Packet drops

WebRTC over Internet

VoIP Engineered

Network

Deterministic

Engineered

Predictable

Internet

Variable

Contended

Variable

WebRTC use general purpose Internet access n/w on PC/phone for RTC

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Reliability

• Reestablish WebSocket and sessions

• Browser refresh

• Loss of IP connectivity

• HTTP server crash

Session

rehydration

• HA for WebRTC GW failure

• HA for SBC failure

• HA for browser, proxy failures,

High Availability

• Least Cost /latency routing

• Route to Transcoder / MCUs

• Failure detection and Routing fallback

Intelligent session routing

Browser refresh may result in loss of sessions

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QoS on WebRTC sessions

Monitor QoS

Analyze Policy

Actions

• WebRTC Endpoint reports

• Media relay (i.e.SBC) reports

• CAC - reject new session

• Adapt sessions - to lower bit rate

• Fallback to audio only call

• Trigger Policy Action once

threshold breached

Network conditions are unpredictable and dynamic

Security and Policy..

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Fine grained Policy Controls

• Partitioning of access, policy and user data,

• Customization of policy data as per enterprise needs

Multi-tenancy

• Social networks like Google, Facebook - Oauth,

• Enterprise Active directory - LDAP,

• Operators subscriber DB,

• SIP registrars

User Authentication

• Enterprise Session limits

• User session limits

• Subscribed Applications

Admission Control

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Securing the communication Signaling

• Secure WebSocket

• Hardened Connex-IP platform (same OS as SBC)

• Vend out ephemeral TURN credentials

User Authentication

• User Authentication with multiple ID providers.

• Oauth 2.0 - Google, Face book

• LDAP - Enterprise Active directory

• SIP - SIP PBX

SBC - Media

• Encrypted media stream (DTLS-SRTP),

• Wire rate policing of Media streams

• DOS, BW theft protection

• Lawful Intercept, blacklisting of misbehaving endpoints

TURN Server – Media

• Validate Users ephemeral TURN credentials Allocated by WebRTC GW

Concluding Remarks

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Concluding Remarks

You should pick a carrier grade, secure, reliable WebRTC solution for you business needs.

WebRTC is browser technology, It has similar challenges as the earlier real-time communications

like VoIP

You should consider WebRTC for you communication needs.

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Interconnecting WebRTC and the World

Rest Of World (PBX, PSTN, Video Room)

Any Interworking

INTERWORKING

Media

Server (Data)

Web

Server (Signaling)

INTERWORKING

Media

Server

Incompatible Domains

Sonus Confidential

Web

Server

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Interconnecting WebRTC and the World

Rest Of World (PBX, PSTN, Video Room)

Carrier Grade Interworking

INTERWORKING

Media

Server (Data)

Web

Server (Signaling)

Carrier-Grade Web

INTERWORKING

POLICY

SECURITY

SBC Media Server

Incompatible Domains

Sonus Confidential