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Report
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
PSTN
Voice over Internet
Protocol
VoIP Server
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
SIP Clients
SIP Clients
Asterisk
WebRTC Clients
PSTN
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.

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