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Asterisk based web real time communication

Asterisk based web real time communication. Advisor : Lian - Jou Tsai Student : Jhe - Yu Wu. Outline. Motivation Abstract Telephony Technology PSTN VoIP Application Asterisk WebRTC System Design Conclusion Reference. Motivation.

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Asterisk based web real time communication

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  1. Asterisk based web real time communication Advisor : Lian-JouTsai Student : Jhe-Yu Wu

  2. Outline • Motivation • Abstract • Telephony Technology • PSTN • VoIP • Application • Asterisk • WebRTC • System Design • Conclusion • Reference

  3. Motivation • How to integrate brand new real time communication technology like WebRTC into SIP and PSTN?

  4. Abstract • This study is aimed to integrate new telephony technology like WebRTC with VoIP. • The following slides will introduce telephony technology including PSTN and VoIP. • The system design will show at the end of the presentation.

  5. Telephony Technology PSTN & VoIP

  6. PSTN Public Switched Telephone Network Figure 1. The PSTN architecture.

  7. H.323 • SIP • RTP • SDP • IAX • SRTP • Skype • And a lot more… VoIP Voice over Internet Protocol

  8. VoIP PSTN Voice over Internet Protocol VoIP Server Figure 2. The VoIP architecture.

  9. Application Asterisk & WebRTC

  10. Asterisk Asterisk is a flexible and extensible suite of integrated telecommunications software.

  11. Asterisk Asterisk designed to support many telephony technologies It powers IP PBX systems, VoIP gateways, conference servers The Asterisk application runs under the Linux operating system

  12. Asterisk

  13. WebRTC Web Real Time Communication

  14. WebRTC WebRTC is a open project  that enables web browsers withReal-TimeCommunicationscapabilities via simple Javascript APIs.

  15. WebRTC Supported Browsers

  16. WebRTC CU-RTC-Web

  17. WebRTC Customizable, Ubiquitous Real Time Communication over the Web

  18. WebRTC • MediaStream: get access to data streams, such as from the user's camera and microphone. • RTCPeerConnection : audio or video calling, with facilities for encryption and bandwidth management. • RTCDataChannel: peer-to-peer communication of generic data.

  19. WebRTC The offer/answer architecture is called JSEP JavaScript Session Establishment Protocol Figure 3. The JSEP architecture.

  20. System Design SIP Clients Asterisk SIP Clients WebRTC Clients PSTN

  21. System Design

  22. System Design

  23. Conclusion • This study intend to build a system that merge two telephony technologies (WebRTC and SIP) into a complete one. • When the system online, we are able to communication with other SIP clients in real time.

  24. References • [1] Clayton, Bradley, Barry Irwin, and Alfredo Terzoli. "Integrating Secure RTP into the Open Source VoIP PBX Asterisk." ISSA. 2006. • [2] Goode, Bur. "Voice over internet protocol (VoIP)." Proceedings of the IEEE90.9 (2002): 1495-1517.

  25. Thanks for your patient.

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