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Adding NTP and RTCP to a SIP User Agent
KTH, School of Information and Communication Technology (ICT), Communication Systems, CoS. (CCSlab)
2006 (English)Independent thesis Advanced level (degree of Master (Two Years)), 20 credits / 30 HE creditsStudent thesis
Abstract [en]

With its enormous potential Voice over Internet Protocol is one of the latest buzzwords in information technology. Despite the numerous advantages of Voice over IP, it is a major technical challenge to achieve a similar call quality as experienced in the ordinary Public Switched Telephone Network.

This thesis introduces standardized Internet protocols for Voice over IP, such as Session Initiation Protocol (SIP), Real-time Transport Protocol (RTP), in its background chapter. In order to provide better Quality of Service (QoS) Voice over IP applications should support a feedback mechanism, such as the Real-time Control Protocol (RTCP), and use accurate timing information, provided by the Network Time Protocol (NTP). Additionally this thesis considers synchronization issues in calls with two and more peers.

After a rather academic overview of Voice over IP, the open source real-time application “minisip”, a SIP user agent, and its operation and structure for handling audio streams will be introduced. Minisip was extended by an implementation of NTP and RTCP to provide a test platform for this thesis.

A clear conclusion is that the addition of global time helps facilitate synchronization of multiple streams from clients located any where in the network and in addition the ability to make one-way delay measurements helps SIP user agents to provide better quality audio to their users.

Abstract [sv]

Röst över IP, eller Internettelefoni baserad på “Voice over Internet Protocol” (VoIP), har med sin stora potential blivit ett av de senaste modeorden inom informationsteknologin. Vid sedan av ett antal fördelar med VoIP så innebär det en stor teknisk utmaning att uppnå en likadan samtalskvalitet som i det vanliga, fasta, telenätet.

I den här uppsatsen beskrivs hur tjänstevalitet för VoIP kan förbättras genom att noggrant tidssynkronisera de (två eller flera) klienter som deltar i ett telefonsamtal. För detta krävs dels en återkopplingsmekanism, såsom “Real-time Control Protocol” (RTCP), samt en gemensam tidsuppfattning i de inblandade klienterna, vilket kan uppnås med hjälp av “Network Time Protocol” (NTP). Dessa protokoll, liksom de övriga Internet-standarder som VoIP baseras på (såsom “Session Initiation Protocol” (SIP) och “Real-time Transport Protocol” (RTP), beskrivs inledningsvis i uppsatsen.

För studien har en SIP-klient baserad på öppen källkod använts (“Minisip”), och utökats med NTP och RCTP funktionalitet för att testa den föreslagna förbättringen av VoIP. En tydlig slutsats är att kännedom om en “global tid” möjliggör synkronisering av multipla ljudströmmar från klienter som befinner sig på olika nätverk. Möjligheten att mäta paketfördröjningen (envägs) bidrar också till en förbättrad ljudkvalitet.

Place, publisher, year, edition, pages
2006. , 47 p.
Series
Trita-ICT-COS, ISSN 1653-6347 ; COS/CCS 2006-14
Keyword [en]
RTCP, RTP, SIP, NTP, Synchronization, minisip, Voice over IP, VoIP, Quality of Service, QoS
National Category
Communication Systems
Identifiers
URN: urn:nbn:se:kth:diva-92200OAI: oai:DiVA.org:kth-92200DiVA: diva2:512647
Subject / course
Telecommunication Systems
Presentation
2006-06-19, Seminar room Grimeton, Isafjordsgatan 30B, Kista, 17:00 (English)
Uppsok
Technology
Supervisors
Examiners
Available from: 2012-03-29 Created: 2012-03-28 Last updated: 2013-09-09Bibliographically approved

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CiteExportLink to record
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Citation style
  • apa
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Output format
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