Webrt integration by altanai bisht

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Altanai Bisht Telecom engineer and Software Developer Email : [email protected] integration fundamentals

description

Explains WebRTC , 3 modes of WebRTC integration by telecom service provider , security concerns . Also throws light on untouched areas of WebRTC integration encountered in during actual project .

Transcript of Webrt integration by altanai bisht

Page 1: Webrt integration by altanai bisht

Altanai BishtTelecom engineer and Software

Developer Email : [email protected]

integration fundamentals

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WebRTC revision

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Standardized by W3C and IETFIETF RTCWeb WG ( Internet world, IP protocols)W3C WebRTC WG (web world, Browsers etc.)

Provides Multimedia communication through browser API’s alone No custom software , hardware No plugins , installations , extensions

Right now only supported by major browsers ( Mozilla , Chrome , opera ) . But support from all browser expected soon

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How does it work ?

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Signal Web Server/service based signaling brokeringOffer/Answer JSEP ( Java Session Establishment Protocol )

model with SDP ( Session Description Protocol )protocol NOT defined

Media Peer to peer media flowSRTP/RTCP

Firewall and NAT/NAPT Uses ICE for STUN / TURN mechanism

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WebRTC

Browser BrowserMedia Path

4

Signal Path Signal Path

Signalling server

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WebRTC Peer to Peer flow

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Web Server Web Server

GUI : JS/HTML/CSS

WebRTC supported Browser

JavaScript API for WebRTC signalling

Media Path

Peer to Peer - Transport framework based on SRTP

Signalling Path

Protocol not defined (possibilities include SIP, Jingle, XMPP)

Application defined interface (HTTPS / Websockets based)

Application defined interface (HTTPS / Websockets based)

SDP OfferSDP Answer

GUI : JS/HTML/CSS

WebRTC supported Browser

JavaScript API for WebRTC signalling

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WebRTC interworking

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Web Server

Media Path(SRTP)

Interconnect to IMS, NGN and PSTN networks(RTP)

Signalling Path

Interworking Function

JSEP ( Java Session Establishment protocol) offer/answer model signal and RTP ( Real Time Protocol ) based media provide the foundation for interworking to IMS/SIP networks

* ICE parameters are used for NAT traversal .

IMS /NGN core

GUI : JS/HTML/CSS

WebRTC supported Browser

JavaScript API for WebRTC signalling

WebRTC Signalling

server ( proxy)

WebRTC Signalling

server ( proxy)

ICE-Lite *ICE-Lite *

Media Server( Transcoding

codecs)

Media Server( Transcoding

codecs)

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One way for a Telecom Operator ( IMS based ) for integrating WebRTC is to get existing 3rd party WebRTC services and interconnect with them. Operator provides his backend network and the internetworking service ( SBC ) to IMS/ NGN world .

Web Server

WebRTC Signalling

server ( proxy)

WebRTC Signalling

server ( proxy)

ICE-LiteICE-Lite

Media Server( Transcoding

codecs)

Media Server( Transcoding

codecs)

Media

WebRTCSignalling

I-SBC( Session Border

Controller )

3rd PartyWebRTC client and Web Domain

Telecom Application

Server

UE

IMS Network Operator

IMS

MediaGUI : JS/HTML/CSS

WebRTC supported Browser

JavaScript API for WebRTC signalling

IMS /NGN core

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Web Server

P-CSCF

Media

IMS SIP

A-SBC ( Session Border Controller )

UE

IMS Network Operator

IMS

Telecom Operator run Web Service

Media

Second way for a Telecom Operator ( IMS based ) is to integrate WebRTC as a pseudo end point ie extend their network to host the WebRTC communication services and web Domain

GUI : JS/HTML/CSS

WebRTC supported Browser

JavaScript API for WebRTC signalling

Media Server( Transcoding

codecs)

Media Server( Transcoding

codecs)

WebRTC Signalling

server ( proxy)

WebRTC Signalling

server ( proxy)

Telecom Application

Server

IMS /NGN core

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The last ( recommended ) option for a Service provider is to enable native support of WebRTC.By not depending on any external entity he has the option to provide any signalling mechanism he likes which may or may not support IMS / NGN standards .

Web Server Web Server

Media

Service provider native WebRTC environment

WebRTCSignalling

GUI : JS/HTML/CSS

WebRTC supported Browser

JavaScript API for WebRTC signalling

GUI : JS/HTML/CSS

WebRTC supported Browser

JavaScript API for WebRTC signalling

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Concerns

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Identity resolution Can opt for oauth , token based authentication through 3rd parties (Facebook,

Twitter, Google circles, …) Can be automatic autonomous authentication for “Call Now” button on Personal &

Business Web pages Firewall / policies blockage

NAT issue is major ICE is complicated No proper documentation on how to setup own STUN or TURN server

Security SRTP / web breaker / DTLS too much and too ambiguous for a developer No one talking about - SPAM control / Unwanted call screening / risk mitigation

RTP stream multiplexing RTP + RTCP Conferencing involves Multiple RTP streams. No standard network topology to work

with Interworking

Simple to call between WebRTC bowers ( full audio and video ) Confusion on integration with OTT solutions (Skype, Viber, Google Jingle , XMPP etc.) SBC ( Media transponders and Proxy servers ) have demonstrated interworking with

NGN/IMS More clarity required on integration with legacy PSTN and PLMN . Even GSM / UMTS

as they have different number formats ( ENUM can be used ) ISUP and legacy codecs .

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Legal Issues ?

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User doesn’t know who is hosting the communication service behind the web page

User cannot see what ID used for his automatic registration via WebRTC webpage ?

Detecting the serviceSubjected to man-in-the-middle (“The Man”-in-the-middle ?)

attacks Network sniffing can tell a lot about users location , device and call

activity . Lawful interception required in many Organizations

Media InterceptionTough to track where is signal passing through in IP network?Media Encryption (DTLS) Who has the keys ?Media Server employed to force relays adding to complexity RTP multiplexing is hugely a black box

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Thank you

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