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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

Asterisk based web real time communication

Advisor : Lian-JouTsai

Student : Jhe-Yu Wu

outline
Outline
  • Motivation
  • Abstract
  • Telephony Technology
    • PSTN
    • VoIP
  • Application
    • Asterisk
    • WebRTC
  • System Design
  • Conclusion
  • Reference
motivation
Motivation
  • How to integrate brand new real time communication technology like WebRTC into SIP and PSTN?
abstract
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.
slide6
PSTN

Public Switched Telephone Network

Figure 1. The PSTN architecture.

slide7

H.323

  • SIP
  • RTP
  • SDP
  • IAX
  • SRTP
  • Skype
  • And a lot more…
VoIP

Voice over Internet Protocol

slide8
VoIP

PSTN

Voice over Internet Protocol

VoIP Server

Figure 2. The VoIP architecture.

application
Application

Asterisk & WebRTC

asterisk
Asterisk

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

asterisk1
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

webrtc
WebRTC

Web Real Time Communication

webrtc1
WebRTC

WebRTC is a open project 

that enables web browsers withReal-TimeCommunicationscapabilities

via simple Javascript APIs.

webrtc2
WebRTC

Supported Browsers

webrtc3
WebRTC

CU-RTC-Web

webrtc4
WebRTC

Customizable, Ubiquitous

Real Time Communication

over the Web

webrtc5
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.
webrtc6
WebRTC

The offer/answer architecture is called

JSEP

JavaScript Session Establishment Protocol

Figure 3. The JSEP architecture.

system design
System Design

SIP Clients

Asterisk

SIP Clients

WebRTC Clients

PSTN

conclusion
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
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.
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