Asterisk based web real time communication

Advisor : Lian-Jou TsaiStudent : 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

PSTNPublic Switched

Telephone Network

Figure 1. The PSTN architecture.

VoIPVoice over Internet

Protocol

• H.323• SIP• RTP• SDP• IAX• SRTP• Skype• And a lot more…

VoIPVoice over Internet

ProtocolVoIP 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 Communicationscapabilities

via simple Javascript APIs.

WebRTC

Supported Browsers

WebRTC

CU-RTC-Web

WebRTC

Customizable, UbiquitousReal Time Communicationover 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

SIP Clients

System Design

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. 2006.

• [2] Goode, Bur. "Voice over internet protocol (VoIP)." Proceedings of the IEEE90.9 (2002): 1495-1517.

Thanks for your patient.