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 .

1. Altanai Bisht
Telecom engineer and Software Developer
Email : tara181989@gmail.com
integration fundamentals

2. WebRTC revision
07/25/14
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 Standardized by W3C and IETF
IETF 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

3. How does it work ?
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 Signal
Web Server/service based signaling brokering
Offer/Answer JSEP ( Java Session Establishment Protocol ) model with SDP
( Session Description Protocol )
protocol NOT defined
 Media
Peer to peer media flow
SRTP/RTCP
 Firewall and NAT/NAPT
Uses ICE for STUN / TURN mechanism

4. 07/25/14
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WebRTC
Browser Browser
Media Path
4
Signal Path Signal Path
Signalling server

5. WebRTC Peer to Peer flow
07/25/14
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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 Offer
SDP Answer
GUI : JS/HTML/CSS
WebRTC supported
Browser
JavaScript API for
WebRTC signalling

6. 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)

7. 07/25/14
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altanaitelecom.wordpress.com7
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
WebRTC
Signalling
I-SBC
( Session Border
Controller )
3rd
Party
WebRTC client and
Web Domain
Telecom
Application
Server
UE
IMS Network Operator
IMS
Media
GUI : JS/HTML/CSS
WebRTC supported
Browser
JavaScript API for
WebRTC signalling
IMS /NGN
core

8. 07/25/14
Altanai Bisht . website:http://
altanaitelecom.wordpress.com8
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

9. 07/25/14
Altanai Bisht . website:http://
altanaitelecom.wordpress.com9
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
WebRTC
Signalling
GUI : JS/HTML/CSS
WebRTC supported
Browser
JavaScript API for
WebRTC signalling
GUI : JS/HTML/CSS
WebRTC supported
Browser
JavaScript API for
WebRTC signalling

10. Concerns
07/25/14
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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 .

11. Legal Issues ?
07/25/14
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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 service
Subjected 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 Interception
Tough 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

12. Thank you
07/25/14
Altanai Bisht . website:http://
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