Asterisk based web real time communication Advisor : Lian-Jou Tsai Student : Jhe-Yu Wu.

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Presentation transcript:

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

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.

Telephony Technology PSTN & VoIP

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

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

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

Application Asterisk & WebRTC

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

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

Asterisk

WebRTC Web Real Time Communication

WebRTC WebRTC is a open project that enables web browsers with Real-Time Communications capabilities via simple Javascript APIs.

WebRTC Supported Browsers

WebRTC CU-RTC-Web

WebRTC Customizable, Ubiquitous Real Time Communication over the Web

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.

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

System Design Asterisk PSTN WebRTC Clients SIP Clients

System Design

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.

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

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