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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 !!
6 
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..
8 
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
9 
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
11 
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,
12 
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.
15 
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 PBX 
Web 
Server 
WebRTC 
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..
17 
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
18 
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
19 
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..
21 
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
22 
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
24 
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.
25 
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
26 
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
WebRTC - Bridging Web and SIP Worlds

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WebRTC - Bridging Web and SIP Worlds

  • 1. WebRTC - Bridging Web and SIP Worlds Shambhu Dayal Rai srai@sonusnet.com Distinguished Solution Architect, Sonus Networks
  • 2. 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. 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
  • 5. 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 !!
  • 6. 6 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
  • 8. 8 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
  • 9. 9 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. 10 Tech Divide : WebRTC vs Telco IPv4 / IPv6 Network Media Transport Media Description Signaling Transport Signaling Protocol Codecs Identity Media Plane Control Plane
  • 11. 11 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,
  • 12. 12 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. 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
  • 14. WebRTC Access to UC System.
  • 15. 15 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 PBX Web Server WebRTC 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.
  • 16. Reliability and Quality over Internet..
  • 17. 17 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
  • 18. 18 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
  • 19. 19 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
  • 21. 21 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
  • 22. 22 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
  • 24. 24 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.
  • 25. 25 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
  • 26. 26 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