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Evaluating Voice over IP phone implementation on a freescale Cortex A9 processor running Linux using open source SIP and WebRTC
Umeå University, Faculty of Science and Technology, Department of Computing Science.
2016 (English)Independent thesis Advanced level (degree of Master (Two Years)), 20 credits / 30 HE creditsStudent thesis
Abstract [en]

Voice over IP (VoIP) is a methodology that refers to the delivery of multimedia and voice sessions over an Internet connection and it provides an alternative to regular voice calls using phone lines, usually referred to as the Public Switched Telephone Network (PSTN). Web Real-Time Communication (WebRTC) is an API denition for browser-to-browser VoIP applications; the denition acts as a foundation for applications using voice, video, chat, and P2P le sharing in a browser environment without the need of either internal or external plugins. To allow WebRTCto make calls to non-WebRTC VoIP applications, a initiation protocol (which is not included in the WebRTC implementation looked at here, i.e., the one released by Google) is needed. One such protocol is the Session Initiation Protocol (SIP), which is the standard protocol used for initialising, changing and terminating interactive sessions for multimedia today; it is particularly known for its use in VoIP applications.

In this thesis, we evaluate the possibility of the creation of a WebRTC implementation using SIP (this type of implementation is referred to as WebRTC-SIP) that runs on an ARM A9 processor architecture. The evaluation is split into two steps. The first step consists of analysingand performing tests of the Linux audio drivers on an ARM platform. The tests are used to determine how a WebRTC-SIP application could affect the audio drivers on such a platform. The second step involves implementation of a WebRTC VoIP application using SIP in a browse renvironment.

The measurements done on the audio drivers show that they can cope with the CPU load created by a WebRTC-SIP application. Based on this and the knowledge gained from implementing such an application for use in a browser, two theoretically possible implementation methods are presented. The first solution builds on the WebRTC-SIP application done in step two, which utilises the support of WebRTC that is built into many browsers to power the application. The second solution is a application which uses a WebRTC to SIP gateway to allow it to set up calls to non-WebRTC applications.

Place, publisher, year, edition, pages
2016. , 49 p.
Series
UMNAD, 1069
National Category
Engineering and Technology
Identifiers
URN: urn:nbn:se:umu:diva-128405OAI: oai:DiVA.org:umu-128405DiVA: diva2:1051971
External cooperation
Limes Audio
Educational program
Master of Science Programme in Computing Science and Engineering
Supervisors
Examiners
Available from: 2016-12-06 Created: 2016-12-05 Last updated: 2016-12-06Bibliographically approved

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CiteExportLink to record
Permanent link

Direct link
Cite
Citation style
  • apa
  • harvard1
  • ieee
  • modern-language-association-8th-edition
  • vancouver
  • Other style
More styles
Language
  • de-DE
  • en-GB
  • en-US
  • fi-FI
  • nn-NO
  • nn-NB
  • sv-SE
  • Other locale
More languages
Output format
  • html
  • text
  • asciidoc
  • rtf