asterisk based web real time communication
DESCRIPTION
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. - PowerPoint PPT PresentationTRANSCRIPT
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Asterisk based web real time communication
Advisor : Lian-Jou TsaiStudent : Jhe-Yu Wu
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Outline• Motivation• Abstract• Telephony Technology
• PSTN• VoIP
• Application• Asterisk• WebRTC
• System Design• Conclusion• Reference
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Motivation
• How to integrate brand new real time communication technology like WebRTC into SIP and PSTN?
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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.
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Telephony Technology
PSTN & VoIP
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PSTNPublic Switched
Telephone Network
Figure 1. The PSTN architecture.
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VoIPVoice over Internet
Protocol
• H.323• SIP• RTP• SDP• IAX• SRTP• Skype• And a lot more…
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VoIPVoice over Internet
ProtocolVoIP Server
PSTN
Figure 2. The VoIP architecture.
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Application
Asterisk & WebRTC
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Asterisk
Asterisk is a flexible and extensible suite of integrated telecommunications software.
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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
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Asterisk
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WebRTC
Web Real Time Communication
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WebRTC
WebRTC is a open project
that enables web browsers with
Real-Time Communicationscapabilities
via simple Javascript APIs.
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WebRTC
Supported Browsers
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WebRTC
CU-RTC-Web
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WebRTC
Customizable, UbiquitousReal Time Communicationover the Web
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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.
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WebRTC
The offer/answer architecture is called
JSEP
JavaScript Session Establishment Protocol Figure 3. The JSEP architecture.
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System Design
Asterisk
PSTN
WebRTC Clients
SIP Clients
SIP Clients
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System Design
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System Design
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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.
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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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Thanks for your patient.