Dissertations / Theses: 'Voice over IP'

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Derakhshanno, Homayoun. "Voice over IP over GPRS." Thesis, KTH, Kommunikationssystem, CoS, 2008. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-91856.

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The Voice over IP (VoIP) technology has become prevalent today due to its lower cost than traditional telephony and its ability to support new value-added services. Additionally, the increasing availability of wireless internet access has led to research studies examining the combination of wireless network access with voice over IP. With the widespread availability of advanced mobile phones and Pocket PCs, the need for VoIP applications on these mobile platforms is tangible. To enable this, we need to evaluate the current wireless access technologies to see if they can support the necessary traffic and implement software to offer these VoIP services to users. In order to easily implement an IP-based service on GSM technology, we should use the GPRS service provided by the GSM operators. In this thesis, we evaluate Voice over IP service over GPRS in terms of feasibility and quality. Following this we ported a locally developed VoIP program to a Pocket PC (with GSM SIM-card support) which runs Microsoft’s Windows Mobile in order to provide suitable software as needed to offer the service from such a portable device.
VoIP tekniken har blivit en rådande teknik numera på grund av dess lägre kostnader och mervärdestjänster som erbjuds jämfört med traditional telefoni. Samtidigt som tendensen mot mer tillgänglig trådlöst internet har underlättat och därmed driver mera studier inom dessa områden. Den allt mer utbredda användningen av avancerade mobiltelefoner och handdatorer numera har lett till ökat behov av att använda VoIP tekniken för dessa mobila utrustningar är alltmer kännbar. För att möjliggöra användadet av VoIP tekniken så behöver vi först och främst utvärdera dagens existerande teknologier för att stödja iden och för det andra måste vi kunna implementera en mjukvara vilket kan erbjuda olika typer av tjänster för slutanvändaren. För att kunna använda en IP-baserad tjänst på GSM teknologin så måste vi använda oss utan GPRS tjänster som tillhandahålls av GSM opratörer. I detta examens arbete kommer vi att utvärdera VoIP tjänster på GPRS när det gäller kvalitet och möjligheter. Därefter kommer vi att Portning en VoIP mjukvara till en handdator (utrustad med GSM sim-kort) vilket har windows Mobile operativsystemet som erbjuder en rad olika tjänster.

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Brännström, Nils. "Voice-over-IP over Enhanced Uplink." Thesis, Linköping University, Department of Electrical Engineering, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-8479.

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The traditional voice service in mobile networks is an important service that mobile users expect high quality from. With the convergence of mobile networks towards an all-IP network, an IP-based speech service becomes important which is referred to as Voice-over-IP (VoIP). The traditional voice service is highly optimized and a VoIP service must therefore fulfil strict quality requirements to provide the same speech service quality. The air interface technology, WCDMA, which is used in third generation communication systems in Europe is constantly developed. An improved concept for the mobile-to-network transmission, called the Enhanced Uplink (EUL) provides for higher uplink capacity for packet data services. It also includes features that may provide a sufficient VoIP service quality in mobile networks, when considering the uplink transmission.

The purpose of this thesis is to evaluate the VoIP capacity over EUL and identify crucial aspects of radio resource management in order to increase the capacity. This is done through dynamic system simulations, using a realistic VoIP traffic model. The VoIP capacity is also estimated by a derived theoretical framework.\newline

It is shown by simulation results and theoretical estimations, that power control is a vital mechanism in order to increase the capacity. Simulation results indicate that a VoIP over EUL capacity of 65\% of the traditional voice service capacity may be reached. The results also indicate that to improve the capacity for larger cells, the allowed VoIP packet delay must be increased.

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Fey, Marcus. "Voice over IP - Eine Einführung." Universitätsbibliothek Chemnitz, 2006. http://nbn-resolving.de/urn:nbn:de:swb:ch1-200600094.

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Eine kurze Einführung zu "Voice over IP" (dem Telefonieren über Datennetze). Es wird ein Überblick über technische Anforderungen und Lösungen geben. Behandelte Gebiete sind Audio-Codecs, das Transportprotokoll RTP sowie die Signalisierungsdienste SIP und H.323.

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Abad, Caballero Israel Manuel. "Secure Mobile Voice over IP." Thesis, KTH, Mikroelektronik och Informationsteknik, IMIT, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-93113.

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Voice over IP (VoIP) can be defined as the ability to make phone calls and to send faxes (i.e., to do everything we can do today with the Public Switched Telephone Network, PSTN) over IP−based data networks with a suitable quality of service and potentially a superior cost/benefit ratio. There is a desire to provide (VoIP) with the suitable security without effecting the performance of this technology. This becomes even more important when VoIP utilizes wireless technologies as the data networks (such as Wireless Local Area Networks, WLAN), given the bandwidth and other constraints of wireless environments, and the data processing costs of the security mechanisms. As for many other (secure) applications, we should consider the security in Mobile VoIP as a chain, where every link, from the secure establishment to the secure termination of a call, must be secure in order to maintain the security of the entire process. This document presents a solution to these issues, providing a secure model for Mobile VoIP that minimizes the processing costs and the bandwidth consumption. This is mainly achieved by making use of high− throughput, low packet expansion security protocols (such as the Secure Real−Time Protocol, SRTP); and high−speed encryption algorithms (such as the Advanced Encryption Standard, AES). In the thesis I describe in detail the problem and its alternative solutions. I also describe in detail the selected solution and the protocols and mechanisms this solution utilizes, such as the Transport Layer Security (TLS) for securing the Session Initiation Protocol (SIP), the Real−Time Protocol (RTP) profile Secure Real−Time Protocol (SRTP) for securing the media data transport , and the Multimedia Internet KEYing (MIKEY) as the key−management protocol. Moreover, an implementation of SRTP, called MINIsrtp, is also provided. The oral presentation will provide an overview of these topics, with an in depth examination of those parts which were the most significant or unexpectedly difficult. Regarding my implementation, evaluation, and testing of the model, this project in mainly focused on the security for the media stream (SRTP). However, thorough theoretical work has also been performed and will be presented, which includes other aspects, such as the establishment and termination of the call (using SIP) and the key−management protocol (MIKEY).
Voice over IP (VoIP) kan defineras som förmågan att göra ett telefonsamtal och att skicka fax (eller att göraallting som man idag kan göra över det publika telefonnätet) över ett IP−baserat nätverk med en passande kvalitet och till lägre kostnad, alternativt större nytta. VoIP måste tillhandahållas med nödvändiga säkerhetstjänster utan att teknikens prestanta påverkas. Detta blir allt viktigare när VoIP används över trådlösa länktekniker (såsom trådlösa lokala nätverk, WLAN), givet dessa länkars begränsade bandbredd och den bearbetningkraft som krävs för att exekvera säkerhetsmekanismerna. Vi måste tänka på VoIPs säkerhet likt en kedja där inte någon länk, från säker uppkoppling till säker nedkoppling, får fallera för att erhålla en säker process. I detta dokument presenteras en lösning på detta problem och innefattar en säker modell för Mobile VoIP som minimerar bearbetningskostnaderna och bandbreddsutnyttjandet. Detta erhålls huvudsakligen genom utnyttjande av säkerhetsprotokoll med hög genomströmning och låg paketexpansion, såsom "Secure Real− time Protocol" (SRTP), och av krypteringsprotokoll med hög hastighet, såsom "Advanced Encryption Standard" (AES). I detta dokument beskriver jag problemet och dess alternativa lösningar. Jag beskriver också den valda lösningen och dess protokoll och mekanismer mer detaljerat, till exempel "Transport Layer Security" (TLS) för att säkra "Session Initiation Protocol" (SIP), SRTP för att skydda transporten av data och "Multimedia Internet KEYing" (MIKEY) för nyckelhantering. En implementation av SRTP, kallad MINIsrtp, finns också beskriven. Beträffande praktiskt arbete och tester av lösningsmodellen har detta projekt fokuserats på skyddandet av datatransporten (SRTP), dess implementation och prestanda. Emellertid har en grundlig teoretisk undersökning genomförts, vilken innefattar andra aspekter såsom telefonsamtalets uppkoppling och nedkoppling (med hjälp av SIP) och valet av passande nyckelhanteringsprotokoll (MIKEY) för att stödja SRTP.

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Smith, Paxton J. "Voice conferencing over IP networks." Thesis, McGill University, 2002. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=29574.

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Traditional telephone conferencing has been accomplished by way of a centralized conference bridge. An Internet Protocol (IP)-based conference bridge is subject to speech distortions and substantial computational demands due to the tandem arrangement of high compression speech codecs. Decentralized architectures avoid the speech distortions and delay, but lack strong control and have a key dependence on silence suppression for endpoint scalability. One solution is to use centralized speaker selection and forwarding, and decentralized decoding and mixing. This approach eliminates the problem of tandem encodings and maintains tight control, thereby improving the speech quality and scalability of the conference. This thesis considers design options and solutions for this model, and evaluates performance through live conferences with real conferees. Conferees found the speaker selection of the new conference model to be transparent, and strongly preferred the resulting speech quality to that of a centralized IP-based conference bridge.

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Knappe, Sebastian. "Marktabgrenzung im Bereich Voice over IP." Hamburg Kovač, 2009. http://d-nb.info/1001547381/04.

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Sundstrom, Karin Heather. "Voice over IP, an engineering analysis." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1999. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape8/PQDD_0024/MQ51804.pdf.

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Franci, Alessandro. "Voice over IP e Web Conferencing." Master's thesis, Alma Mater Studiorum - Università di Bologna, 2012. http://amslaurea.unibo.it/4464/.

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Moodie, Myron L., Todd A. Newton, Thomas B. Grace, and William A. Malatesta. "Performance of Voice-over-IP over iNET Telemetric Networks." International Foundation for Telemetering, 2011. http://hdl.handle.net/10150/595774.

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ITC/USA 2011 Conference Proceedings / The Forty-Seventh Annual International Telemetering Conference and Technical Exhibition / October 24-27, 2011 / Bally's Las Vegas, Las Vegas, Nevada
Bidirectional networked radio frequency (RF) communications between the ground and test articles are quickly becoming a normal mode of operation. Not only can devices be remotely controlled, but other networking technologies are emerging into flight test. Voice over IP (VoIP) is ubiquitous in the workplace and in homes, but it presents unique challenges when used to communicate between test articles. This paper presents some issues to be considered and test results to help aid deployment of VoIP systems in network-based test systems such as iNET's Telemetry Network System (TmNS).

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Ram, Abhishek. "Assessment of Voice Over IP as a solution for Voice over ADSL." Thesis, Virginia Tech, 2002. http://hdl.handle.net/10919/33135.

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Voice over DSL (VoDSL) is a technology that enables the transport of data and multiple voice calls over a single copper-pair. VoDSL employs packet voice technology instead of the traditional circuit switched voice. Voice over ATM (VoATM) and Voice over IP (VoIP) are the two main alternatives for carrying voice packets over DSL. ATM is currently the preferred technology, since it offers the advantage of ATMâ s built-in Quality of Service (QoS) mechanisms. IP, on the other hand, cannot provide QoS guarantees in its traditional form. IP QoS mechanisms have been evolved only in the recent years. VoIP has gained popularity in the core networks. If it could replace VoATM in the access networks, it would open the door for end-to-end IP telephony that would result in major cost savings. In this thesis, we propose a VoIP-based VoDSL architecture that provides QoS guarantees comparable to those offered by ATM in the DSL access network. Our QoS architecture supports Premium and Regular service categories for voice traffic and the Best-Effort service category for data traffic. Voice and data packets are placed in separate output queues at the bottleneck link. The Weighted Fair Queuing algorithm in used to schedule voice and data packets for transmission over the bottleneck link. Fragmentation of large data packets reduces the waiting time for voice packets in the link. We also propose a new admission control mechanism called Admission Control by Implicit Signaling. This mechanism takes advantage of application layer signaling by mapping it to the IP header. The router can infer the resource requirements for the connection by looking at certain field in the IP header of the application layer signaling packets. This eliminates the need for an explicit signaling protocol. We evaluate the performance of our QoS architecture by means of a simulation study. Our primary metrics are the end-to-end delay of voice packets across the access network and the bandwidth consumed by a voice call. Our results show that the end-to-end delays of voice packets in our VoIP architecture are comparable to that in the VoATM architecture. ACIS limits the number of voice calls admitted into the premium service class and provides guaranteed service to those calls under all loads. It also provides acceptable service to regular calls under light loads. We also show that PPP is a better choice than ATM as a Layer 2 protocol for our VoIP architecture. PPP offers the advantages of low bandwidth requirement and interleaving of voice packets in between fragments of large data packets during transmission over the bottleneck link. We conclude that our VoIP architecture would be suitable for future VoDSL deployments.
Master of Science

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Yang, Xu. "On the development of Voice over IP." [College Station, Tex. : Texas A&M University, 2008. http://hdl.handle.net/1969.1/ETD-TAMU-2389.

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Yensen, Trevor. "Structures and interfaces for voice over IP." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 1998. http://www.collectionscanada.ca/obj/s4/f2/dsk2/tape17/PQDD_0010/MQ36903.pdf.

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Gong, Qi Peng. "Playout buffering for conversational voice over IP." Thesis, McGill University, 2013. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=114224.

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In Voice over IP, the quality of interactive conversation is important to users. Major factors affecting perceived quality are delay, delay jitter, and missing packets. For conversational VoIP, a conversational delay also plays an important role for perceived quality. Large conversational delay can result in double talk, echo or even the termination of the conversation. In practice, a playout buffer is introduced at the receiver's side to remove delay jitter, so that the voice information carried on packets can be available at regular intervals for decoding. A longer buffer reduces the possibility of late packet loss at the expense of increasing conversational delays. Since the time delay of a playout buffer is a major addition to conversational delay, to keep conversational interactivity, it is desirable to design a playout buffer to be short but capable of protecting packets against late packet loss.In this thesis, we will explore playout buffering algorithms with improved conversational quality. We propose a quality-based adaptive playout buffering algorithm with improved voice quality and reduced conversational delays. We use the E-Model R factor as the cost index to obtain playout delays which adapt for each talkspurt. Special steps are taken to reduce conversational delay : (1) immediately play out stretched speech carried on the first packet of a talkspurt when received (stretching provides additional buffer delay forfollowing packets) ; (2) compress the speech segment carried on the packets in the playout buffer at the end of a talkspurt (compression reduces the playout delay for the packets). As other quality-based algorithms, our scheme is subject to burst losses. To improve perceived quality further, we use sender-driven repair algorithms, in which a sender sends redundancy information, to mitigate the impact of the missing packets due to network (lost packets) and buffer underflow (late packets) without increasing buffer delays. In this thesis, we develop a new adaptive forward error correction (FEC) scheme to provide redundancy without additional delay and apply it to our adaptive playout buffering algorithm for improved perceived quality. As an alternative sender-based technique to send redundancy information, a path diversity scheme uses multiple paths (here we consider two paths). Redundant information is sent on a second path. We consider four different path diversity schemes (two of them are proposed based on E-model in this work), and design corresponding playout buffering algorithms based on conversational quality including both calling quality and interactivity.
Dans le domaine de la voix sur IP (VOIP), la qualité de conversation interactive est importante pour les utilisateurs. Les principaux facteurs qui affectent la qualité à la réception sont le retard, la désynchronisation et les pertes de paquets. Lors d'une conversation par voix sur IP, le retard joue un rôle important pour la qualité à la réception. Les délais prolongés de conversation peuvent provoquer de la double parole, de l'écho ou encore la fin de la conversation. En pratique, un jeu de buffers (ou mémoirer tampon) est introduit du coté du récepteur pour supprimer les délais indésirables. Ainsi, l'information de la voix contenue dans les paquets peut être disponible à des intervalles de temps réguliers pour le décodage. Un buffer plus grand réduit la possibilité de perdre des paquets en retard aux dépens de voir augmenter les délais de conversation. Comme la capacité du jeu de buffers est une somme de retards de conversations, pour garder l'interactivité de la conversation, il est préférable de concevoir un jeu qui soit court mais également capable de protéger les paquets contre les pertes dues aux retards.Dans cette thèse, nous explorons les algorithmes de jeux de buffers améliorant la qualité de conversation. Nous proposons un algorithme adaptatif en termes de qualité et qui a pour but d'augmenter la qualité de la voix et de réduire les retards dans la conversation. Nous utilisons le facteur R du modèle-E comme indice de coût pour obtenir les délais du jeu qui s'adaptent à chaque "talkspurt" (segment continu de parole inséré entre les temps de silence). Des étapes spécifiques sont entreprises pour réduire le retard de conversation : (1) traiter immédiatement la parole étirée qui est contenue dans le premier paquet lors de la réception d'un "talkspurt" (l'étirement fournit des délais de buffers supplémentaires pour suivre les paquets); (2) compresser le segment de parole contenu dans les paquets au niveau du jeu de buffers à la fin du "talkspurt" (la compression réduit les délais du jeu pour les paquets).Comme les autres algorithmes portant sur la qualité, notre montage est sujet aux pertes de signaux en rafale (burst loss). Pour améliorer encore plus la qualité à la réception, nous utilisons des algorithmes de réparation par émetteur. L'émetteur envoie de l'information redondante pour atténuer l'impact des paquets manquants causés par le réseau (pertes de paquets) et par le manque de capacité des buffers (paquets retardés), tout ceci sans augmenter les délais de buffer. Dans cette thèse, nous développons un nouveau schéma adaptatif de correction d'erreurs sans voie de retour (FEC) pour fournir de la redondance sans ajouter de délais. Nous appliquons ce dispositif de correction à notre algorithme adaptatif de jeu de buffers pour améliorer la qualité perçue. En tant que technique alternative d'envoie d'information de redondance, un schéma fournissant de la diversité de trajet utilise plusieurs chemins (nous en considérons deux ici). L'information de redondance est envoyée vers le deuxième chemin. Nous considérons quatre schémas de diversité de trajet (deux d'entre eux sont proposés par rapport au modèle-E utilisé dans ce travail). Nous concevons également des algorithmes de jeu de buffers avec comme critères de qualité de conversation : la qualité d'appel et l'interactivité.

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Bilien, Johan. "Key Agreement for Secure Voice over IP." Thesis, KTH, Mikroelektronik och Informationsteknik, IMIT, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-93069.

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This thesis reviews the usual properties and requirements for key agreement protocols. It then focuses on MIKEY, a work-in-progress protocol designed to conduct key agreements for secure multimedia exchanges. The protocol was implemented and incorporated in a SIP user agent - minisip. This implementation was used to measure the additional delay required for key exchange during call establishment. Finally, some schemes are proposed regarding the use of MIKEY in advanced VoIP scenarios, such as conferences and terminal mobility.

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Iranmanesh, Seyed Amir. "Security Enhancements in Voice Over Ip Networks." W&M ScholarWorks, 2017. https://scholarworks.wm.edu/etd/1530192357.

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Voice delivery over IP networks including VoIP (Voice over IP) and VoLTE (Voice over LTE) are emerging as the alternatives to the conventional public telephony networks. With the growing number of subscribers and the global integration of 4/5G by operations, VoIP/VoLTE as the only option for voice delivery becomes an attractive target to be abused and exploited by malicious attackers. This dissertation aims to address some of the security challenges in VoIP/VoLTE. When we examine the past events to identify trends and changes in attacking strategies, we find that spam calls, caller-ID spoofing, and DoS attacks are the most imminent threats to VoIP deployments. Compared to email spam, voice spam will be much more obnoxious and time consuming nuisance for human subscribers to filter out. Since the threat of voice spam could become as serious as email spam, we first focus on spam detection and propose a content-based approach to protect telephone subscribers' voice mailboxes from voice spam. Caller-ID has long been used to enable the callee parties know who is calling, verify his identity for authentication and his physical location for emergency services. VoIP and other packet switched networks such as all-IP Long Term Evolution (LTE) network provide flexibility that helps subscribers to use arbitrary caller-ID. Moreover, interconnecting between IP telephony and other Circuit-Switched (CS) legacy telephone networks has also weakened the security of caller-ID systems. We observe that the determination of true identity of a calling device helps us in preventing many VoIP attacks, such as caller-ID spoofing, spamming and call flooding attacks. This motivates us to take a very different approach to the VoIP problems and attempt to answer a fundamental question: is it possible to know the type of a device a subscriber uses to originate a call? By exploiting the impreciseness of the codec sampling rate in the caller's RTP streams, we propose a fuzzy rule-based system to remotely identify calling devices. Finally, we propose a caller-ID based public key infrastructure for VoIP and VoLTE that provides signature generation at the calling party side as well as signature verification at the callee party side. The proposed signature can be used as caller-ID trust to prevent caller-ID spoofing and unsolicited calls. Our approach is based on the identity-based cryptography, and it also leverages the Domain Name System (DNS) and proxy servers in the VoIP architecture, as well as the Home Subscriber Server (HSS) and Call Session Control Function (CSCF) in the IP Multimedia Subsystem (IMS) architecture. Using OPNET, we then develop a comprehensive simulation testbed for the evaluation of our proposed infrastructure. Our simulation results show that the average call setup delays induced by our infrastructure are hardly noticeable by telephony subscribers and the extra signaling overhead is negligible. Therefore, our proposed infrastructure can be adopted to widely verify caller-ID in telephony networks.

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Theander, Petter, and Thomas Hultgren. "Voice over IP for Sony Ericsson Cellular Phones." Thesis, Blekinge Tekniska Högskola, Avdelningen för programvarusystem, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-2623.

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This report presents an investigation of the possibilities to implement voice over IP (VoIP) in Sony Ericsson cellular phones. The results from this investigation show that it is partially possible to implement such a solution. The best option for doing so is to make use of the support for the Session Initiation Protocol and the Real-time Transport Protocol offered by the architecture. Another goal is to evaluate if Bluetooth is able to handle the requirements needed for the solution. The whole concept is proven by implementing a prototype. Measurements on this prototype show that Bluetooth will be able to handle the requirements of most IP-based voice communication, i.e., in respect to latency and bandwidth.

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Nesh-Nash, Ali. "Voice over IP in a resource constrained environment." Thesis, KTH, Kommunikationssystem, CoS, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-92257.

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Today, the telecommunication world is focused on mobility. This is popular because since the 1990s most people have integrated their mobile phones into their life. A new factor is the rise of the voice over IP(VoIP) technology, with VoIP over Wireless LANs (WLANs) as the clear next growth area for mobile communications. The purpose of this thesis was to understand how to save power based upon changing when some operations are performed in a VoIP client. In order to do this, we decided to port minisip to an HP iPAQ 5500 Personal Digital Assistant (PDA), in order to explore some of the issues of running such a client on a PDA - due to its constraints with regard to storage, processing power, and battery power. Minisip is a SIP open source user agent running on Linux and Windows. This thesis builds upon earlier theses which showed that minisip can offer a secure communications platform with the latest functions which are desired in a mobile personal VoIP system. However, most of these earlier theses utilized desktop, laptop, or server based system, i.e., with few resources constrains. The focus of this thesis was to examine the case of a highly constrained user platform such as an iPAQ.
Dagens telekommunikationssystem fokuserar på mobilitet. Detta har blivit populärt under 90-talet då mobilitet blev naturligt integrerad i människans vardagliga liv i form av exempelvis mobiltelefoner. Voice over IP (VoIP) har blivit en stor del av dagen teknik där trådlösa system Wireless LANs (WLANs) har blivit en större del av mobilkommunikation. Målet med denna rapport är att förstå hur strömförbrukningen kan minimeras genom att utföra vissa operationer med hjälp av en VoIP-klient. För att åstadkomma detta porterade vi minisip, en SIP agent som är baserad på öppen källkod och körs på Linux och Windows, till en HP iPAQ 5500, en så kallad Personal Digital Assistant (PDA). Vi valde PDAn för att kunna utforska de begränsningar den medför i form av lagringsutrymme, processorkapacitet, och batteri. Denna rapport bygger vidare på tidigare rapporter som visar att minisip kan erbjuda en säker kommunikationsplattform med de senaste funktionerna som önskas i mobila VoIPsystem. De flesta av dessa tidigare rapporter baseras på system med få begränsningar rörande resurser såsom stationära- eller bärbara datorer samt serverbaserade system. Denna rapports fokus är att utforska detta fall i en miljö med större begränsningar på resurser som till exempel en iPAQ.

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Elliott, Colm. "Stream synchronization for voice over IP conference bridges." Thesis, McGill University, 2004. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=82484.

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Potentially large network delay variability experienced by voice packets when travelling over IP networks complicates the design of a robust voice conference bridge. Intrastream synchronization is required at the conference bridge to smooth out network delay jitter on a given stream and provide a continuous stream of voice packets to the conference bridge core. Interstream synchronization is needed to provide time synchronization between packets in different streams, allowing for a mapping of selected voice streams to the conference bridge output and the creation of a periodic and synchronized output from the conference bridge.
This work presents a design and evaluation of a Synchronized conference bridge that maps N input voice streams to M output voice streams representing selected speakers. A conference simulator, designed for this thesis, is used to characterize the performance of this bridge in terms of delay and packet loss, speaker selection accuracy and conference audio quality.

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Zue, Cory L. "Modeling and assessing secure Voice over IP performance." Thesis, Massachusetts Institute of Technology, 2005. http://hdl.handle.net/1721.1/33377.

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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2005.
Includes bibliographical references (p. 102-106).
Voice over Internet Protocol (VoIP) systems enable efficient communications over data networks, but security of VoIP and the impact of that security on communications quality has not been quantitatively modeled. A conversational model is adapted for VoIP and a computational model of communication quality - the Z-Model - is developed. VolIP conversations are simulated for networks with a range of performance characteristics including differing bandwidth, latency and bit error rates to evaluate the impact of security on communication quality. Results show that improving conficlentiality via encryption of conversation data packets does not introduce significant delays, but does increase bandwidth. In certain restricted-bandwidth environments this results in dramatic reductions of perceived conversation quality.
by Cory L. Zue.
M.Eng.

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Meinberg, Ralf. "Voice over IP : IP-basierter Sprachdienst vor dem Hintergrund des novellierten TKG /." Münster : LIT-Verl, 2008. http://deposit.d-nb.de/cgi-bin/dokserv?id=3168567&prov=M&dok_var=1&dok_ext=htm.

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Ebstein, Sammy-Franklin. "Voice over IP Szenarien über die Diffusion von VoIP /." St. Gallen, 2005. http://www.biblio.unisg.ch/org/biblio/edoc.nsf/wwwDisplayIdentifier/00641852001/$FILE/00641852001.pdf.

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Lloyd, Patrick. "An exploration of covert channels within voice over IP /." Online version of thesis, 2010. http://hdl.handle.net/1850/12241.

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Nakkhongkham, Saiyoot. "Measuring the quality of service of voice over IP." Thesis, University of Hawaii at Manoa, 2003. http://hdl.handle.net/10125/7007.

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This thesis addresses and proposes the methods to measure and classify problems encountered when transmitting voice over the Internet. These problems are delay, jittering, and lost of data. Transmitting voice over the Internet is commonly known as Voice over Internet Protocol (VoIP). The fact that the current Internet infrastructure was not designed to transmit real-time data makes these problems become eminent. Solutions to improve the quality of real-time data transmissions over the Internet are currently under research. This thesis presents a VoIP software-based application tool to help VoIP engineers to better measure and classify these problems. This software tool is called Voice Over Internet Protocol Quality of Service Measuring Tool (VoIP QoS MT). This tool works in a live mode and in a test traffic mode. Live mode allows users to transmit actual voice conversations between two computers over the Internet, using a few digitization and compression schemes. Quality of service can be observed in real-time speech. The test traffic mode works without voice conversation. Its payloads are programmed to carry information for the purpose of testing delay, jittering, and lost statistics. Detailed functionalities and related background information for each components used in the application are presented accordingly in this paper
xii, 110 leaves

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Mahfuz, Ejaz. "Packet loss concealment for voice transmission over IP networks." Thesis, McGill University, 2001. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=32965.

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Voice-over-IP (VoIP) uses packetized transmission of speech over the Internet (IP network). However, at the receiving end, packets are missing due to network delay, network congestion (jitter) and network errors. This packet loss degrades the quality of speech at the receiving end of a voice transmission system in an IP network. Since the voice transmission is a real-time process, the receiver cannot request for retransmission of the missing packets. Concealment algorithms, either transmitter or receiver based, are used to replace these lost packets. The packet loss concealment (PLC) techniques described in the standards ANSI TI.521 (Annex B) and ITU-T Rec. G.711 (Appendix I), have good performance, but these algorithms do not use subsequent packets for reconstruction. Furthermore, there are discontinuities between the reconstructed and the subsequent packets, especially at the transitions from voiced to unvoiced and phoneme to phoneme.
The goal of this work is to develop an improved PLC algorithm, using the subsequent packet information when available. For this, we use the Time-Scale Modification (TSM) technique based on Waveform Similarity Over-Lap Add (WSOLA) to reconstruct the dropped or lost packets. The algorithm looks ahead for subsequent packets. If these packets are not available for reconstruction, algorithm uses information from past packets. Subjective tests show that the proposed method improves the reconstructed speech quality significantly.

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SANTOS, ALEXANDRE FERREIRA DOS. "ASSESSMENT OF QOS PARAMETERS IN VOICE OVER IP TRANSMISSION." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 2004. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=5243@1.

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Este trabalho apresenta um estudo visando a estabelecer uma metodologia para dimensionamento de um sistema VoIP, focalizando, em particular, o dimensionamento de um multiplexador estatístico. Procuramos aplicar modelos e resultados existentes para o problema geral do multiplexador estatístico ao caso específico de um sistema VoIP, levando em conta as características do tráfego, os requisitos de QoS e os princípios das arquiteturas Intserv e Diffserv. Para isto, apresentamos um resumo da tecnologia VoIP, incluindo seus requisitos de qualidade e os protocolos apropriados para transportar este tipo de mídia na Internet. Discorremos sobre os mecanismos de controle de tráfego usuais em redes de pacotes com QoS, assim como sobre as Arquiteturas de QoS definidas pelo IETF . É apresentada uma revisão de modelos de tráfego e modelos aplicáveis à análise de multiplexadores estatísticos, com destaque para o chamado modelo fluido aplicado à descrição do tráfego gerado por um agregado homogêneo de fontes de voz, além de um estudo comparativo entre respostas obtidas analiticamente com aquelas obtidas por meio de simulação. A influência do tipo de codificador e de parâmetros como tamanho de pacote é investigada, mostrando-se a dificuldade em se dispor de um modelo analítico capaz de levar em conta, de forma precisa, os diferentes formatos do sistema VoIP. Por fim, estabelece-se um cenário para aplicação dos modelos a um sistema VoIP.
This work presents a study aiming at to establish a methodology for sizing a VoIP system, focusing, in particular, the sizing of a statistical multiplexer. We apply existing models and results for the general problem of the statistical multiplexer to the specific case of a VoIP system, taking in account the characteristics of the traffic, the requirements of QoS and the principles of the architectures Intserv and Diffserv. For this, we present a summary of the VoIP technology, including its requirements of quality and the protocols appropriate to carry this type of media in the Internet. We discourse on the usual mechanisms of traffic control in packet networks with QoS, as well as on the Architectures of QoS defined by the IETF. A revision of traffic models and applicable models to the analysis of statistical multiplexers, with prominence for the fluid model applied to the description of the traffic generated for a homogeneous aggregate of voice sources, is presented. Besisdes, a comparative study of behavior gotten analytically with those gotten by means of simulation is made. The influence of the coder and parameters as so packet size is investigated, revealing the difficulty in finding an analytical model capable to take in account, with precision, the different formats of the VoIP system. Finally, we establish a scenario for application of the models to a VoIP system.

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Kannan, Steven (Steven K. ). "Secure Voice over IP conferencing with decentralized group encryption." Thesis, Massachusetts Institute of Technology, 2007. http://hdl.handle.net/1721.1/45979.

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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2007.
Includes bibliographical references (p. 107-109).
This thesis addresses the development of an end-to-end secure Voice over IP (VoIP) conference system. We are particularly interested in challenges associated with deploying such a system in ad-hoc networks containing low bandwidth and/or high latency data links. End-to-end security is handled by the decentralized Public Key Group Encryption library (PKGE) developed at Lincoln Laboratory; PKGE allows real-time keying of conference users without an on-line central keying authority.We present a system design and its prototype implementation in accordance with a set of appropriate design goals. The final product demonstrates the feasibility of using PKGE in the demanding conditions of VoIP conferencing. The system development sheds light on a number of issues and engineering challenges that ultimately affect call quality, functionality, security, and usability, motivating our recommendations for the next generation system.
by Steven Kannan.
M.Eng.

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Guerra, Carlos Humberto Martins Chagas. "Gateway SIP - Asterisk." Master's thesis, Universidade de Évora, 2012. http://hdl.handle.net/10174/15430.

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Esta tese consiste num estudo realizado sobre a tecnologia VoIP (Voice over IP) com aplicação dentro da área dos PBX(Private Branch eXchange) e Gateways VoIP Open Source. Em primeiro lugar foram abordados os conceitos, requisitos e elementos associados a esta tecnologia bem como a sua interligação com outras tecnologias de comunicação como a PSTN (Public Switched Telephone Network) , ISDN (Integrated Services Digital Network) ou GSM (Groupe Special Mobile).Al ém disso, foi efectuado um estudo teórico e prático sobre o software Open Source Asterisk, tendo com objectivo explorar o seu modo de funcionamento e funcionalidades disponíveis, passiveis de serem utilizadas em ambiente empresarial. Por fim, foi desenvolvida uma solução assente neste tipo de tecnologia na empresa Clidis - Laboratório de Análises Clinicas de Sines, onde ficou provado que a implementação de PBX/Gateways VoIP através de software Open Source e uma alternativa viável às reais necessidades de comunicação das empresas; ABSTRACT: This thesis is a study on VoIP (Voice over IP) application within the area of PBX (Private Branch eXchange) and VoIP Open Source Gateways. Firstly were addressed the concepts, requirements and elements associated with this technology and its interconnection with other communications technologies such as PSTN (Public Switched Telephone Network), ISDN (Integrated Services Digital Network) or GSM (Groupe Special Mobile). Furthermore, a study was carried out on the theoretical and practical Asterisk Open Source software, with the aim to explore its operation and features available, liable to be used in business environment. Finally, we developed in the company Clidis (Laboratory of Clinical Analyses in Sines) a solution based on this technology, where it was proved that the implementation of PBX / VoIP Gateways through Open Source software is a viable alternative to the real needs of business communication.

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Dely, Peter. "Adaptive Aggregation of Voice over IP in Wireless Mesh Networks." Thesis, Karlstad University, Faculty of Economic Sciences, Communication and IT, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:kau:diva-938.

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When using Voice over IP (VoIP) in Wireless Mesh Networks the overhead induced by the IEEE 802.11 PHY and MAC layer accounts for more than 80% of the channel utilization time, while the actual payload only uses 20% of the time. As a consequence, the Voice over IP capacity is very low. To increase the channel utilization efficiency and the capacity several IP packets can be aggregated in one large packet and transmitted at once. This paper presents a new hop-by-hop IP packet aggregation scheme for Wireless Mesh Networks.

The size of the aggregation packets is a very important performance factor. Too small packets yield poor aggregation efficiency; too large packets are likely to get dropped when the channel quality is poor. Two novel distributed protocols for calculation of the optimum respectively maximum packet size are described. The first protocol assesses network load by counting the arrival rate of routing protocol probe messages and constantly measuring the signal-to-noise ratio of the channel. Thereby the optimum packet size of the current channel condition can be calculated. The second protocol, which is a simplified version of the first one, measures the signal-to-noise ratio and calculates the maximum packet size.

The latter method is implemented in the ns-2 network simulator. Performance measurements with no aggregation, a fixed maximum packet size and an adaptive maximum packet size are conducted in two different topologies. Simulation results show that packet aggregation can more than double the number of supported VoIP calls in a Wireless Mesh Network. Adaptively determining the maximum packet size is especially useful when the nodes have different distances or the channel quality is very poor. In that case, adaptive aggregation supports twice as many VoIP calls as fixed maximum packet size aggregation.

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Morshed, Muhammad. "Voice over IP and Lawful Intercept : Good cop/Bad cop." Thesis, KTH, School of Information and Communication Technology (ICT), 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-24260.

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Lawful interception is a common practice for monitoring a telecommunication network by law enforcement agencies all over the world. It plays a vital role to ensure national security and to control crimes by providing authorized monitoring of communicating parties in a communication network. However, there are some important issues that need to be addressed, such as the privacy of individuals, malicious use of lawful interception by a “bad” cop, vulnerability of a lawful interception system to misuse by others, cost, legal liability, etc. These issues have lead to opposition to lawful interception. Many researchers have been looking for a secure and acceptable lawful interception system that would eliminate or minimize the undesirable aspects of lawful interception. One of the approaches that gained a lot of attention is a key escrow encryption system. For lawful interception a key recovery key is escrowed with a trusted third party. This key can subsequently be used for decryption by the law enforcement agency. The trusted third party might be a government agency or a private company. The process for recovering keys should be based on a predefined securitypolicy. The trusted third party’s responsibility is to store the key and to protect it from malicious use. This malicious use could be by a competitor, a telecommunication operator, Internet Service Provider (ISP), a law enforcement agency, or other party. If the trusted third party itself utilizes the key or improperly discloses the key to another party, then the data that was protected by encryption could be compromised Unfortunately, there is no easy means to detect if the data has been tampered with or not. This thesis focuses on therefore in the case of voice over IP, where there is a need for a means to determine if a recorded conversation is authentic or not. Hence the objective of the overall thesis project is to design, implement, and evaluate a security mechanism that can be used with a trusted third party -based key escrow encryption system that will prevent or reduce the risk of forgery by (a bad cop within) a lawenforcement agency using the escrowed key.

This thesis describes how a key escrow encryption system would be improved by the proposed mechanism – with a focus on the actions of a party that has access to the escrowed key. We do not examine how the party got access to this key, but for the purposes of this thesis we assumed that this party is either a good cop or a bad cop. We have defined the meaning of these terms and examine what operations a bad cop might attempt to perform – given the access to the master key. For example, this party could capture the data packets of a Voice over IP session, and then decrypt the packets using the key provided by the escrow agent. After decryption we examined the ability of a bad cop to modify or forge data packets, then encrypt these forged packets with the key – in order to fabricate evidence. We then examined how to detect such modifications or forgery. The proposed system is able to detect this forgery, based upon the inability of the forger to generate the correctly signed hashed message authentication coded. We also examine additional extensions to the user agent and the escrow agent to be able to identify which packets (or groups of packets) were not generated by the original participant in the conversation. The goal is to understand if the proposed mechanism could make lawful interception more secure, while increasing the protection of the communicating parties’ conversation from undetected manipulation and making the digital record of a conversion easier to authenticate.

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Zhang, Hongqi. "Performance of voice over IP in mobile ad hoc networks." Thesis, University of Ottawa (Canada), 2007. http://hdl.handle.net/10393/27943.

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The objectives of this thesis are to study the performance of voice over IP in ad-hoc networks, and to study the relationship among routing protocols, coding schemes and the quality of voice. The contribution is an adaptive rate control scheme which uses Mean Opinion Score (MOS) as criteria to improve the performance of the mobile ad hoc network. We first discuss and analyze the characteristics of wireless ad hoc network and Adaptive Multirate-Wideband (AMW) speech coding, and then provide the operation of the adaptive source-network rate control scheme consisting of the voice coding and transmission control, adaptive network rate control, the source rate control, and the frame combination. We implement the scheme in a test-bed and test the performance of speech in different scenarios using Multi-path Source Routing (MSR). Finally, we verify our testbed measurements with OPNET simulation.

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Montminy, Christian. "A study of speech compression algorithms for Voice over IP." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape4/PQDD_0017/MQ57147.pdf.

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Kamat, Narasinha. "A delay-efficient rerouting scheme for voice over ip traffic." [Gainesville, Fla.] : University of Florida, 2002. http://purl.fcla.edu/fcla/etd/UFE0000548.

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Nocito, Carlos Daniel. "A Network Conditions Estimator for Voice Over IP Objective Quality Assessment." Scholarly Repository, 2011. http://scholarlyrepository.miami.edu/oa_theses/292.

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Objective quality evaluation is a key element for the success of the emerging Voice over IP (VoIP) technologies. Although there are extensive economic incentives for the convergence of voice, data, and video networks, packet networks such as the Internet have inherent incompatibilities with the transport of real time services. Under this paradigm, network planners and administrators are interested in ongoing mechanisms to measure and ensure the quality of these real time services. Objective quality assessment algorithms can be broadly divided into a) intrusive (methods that require a reference signal), and b) non intrusive (methods that do not require a known reference signal). The latter group, typically requires knowledge of the network conditions (level of delay, jitter, packet loss, etc.), and that has been a very active area of research in the past decade. The state of the art methods for objective non-intrusive quality assessment provide high correlations with the subjective tests. Although good correlations have been achieved already for objective non-intrusive quality assessment, the current large voice transport networks are in a hybrid state, where the necessary network parameters cannot easily be observed from the packet traffic between nodes. This thesis proposes a new process, the Network Conditions Estimator (NCE), which can serve as bridge element to real-world hybrid networks. Two classifications systems, an artificial neural network and a C4.5 decision tree, were developed using speech from a database collected from experiments under controlled network conditions. The database was composed of a group of four female speakers and three male speakers, who conducted unscripted conversations without knowledge about the details of the experiment. Using mel frequency cepstral coefficients (MFCCs) as the feature-set, an accuracy of about 70% was achieved in detecting the presence of jitter or packet loss on the channel. This resulting classifier can be incorporated as an input to the E-Model, in order to properly estimate the QoS of a network in real time. Additionally, rather than just providing an estimation of subjective quality of service provided, the NCE provides an insight into the cause for low performance.

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Al-Najjar, Camelia. "An approach for improving performance of aggregate voice-over-IP traffic." Texas A&M University, 2005. http://hdl.handle.net/1969.1/4216.

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The emerging popularity and interest in Voice-over-IP (VoIP) has been accompanied by customer concerns about voice quality over these networks. The lack of an appropriate real-time capable infrastructure in packet networks along with the threats of denial-of service (DoS) attacks can deteriorate the service that these voice calls receive. And these conditions contribute to the decline in call quality in VoIP applications; therefore, error-correcting/concealing techniques remain the only alternative to provide a reasonable protection for VoIP calls against packet losses. Traditionally, each voice call employs its own end-to-end forward-error-correction (FEC) mechanisms. In this paper, we show that when VoIP calls are aggregated over a provider's link, with a suitable linear-time encoding for the aggregated voice traffic, considerable quality improvement can be achieved with little redundancy. We show that it is possible to achieve rates closer to channel capacity as more calls are combined with very small output loss rates even in the presence of significant packet loss rates in the network. The advantages of the proposed scheme far exceed similar or other coding techniques applied to individual voice calls.

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Dely, Peter. "Cross-Layer Optimization of Voice over IP in Wireless Mesh Networks." Licentiate thesis, Karlstads universitet, Avdelningen för datavetenskap, 2010. http://urn.kb.se/resolve?urn=urn:nbn:se:kau:diva-6280.

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Wireless Mesh Networks (WMNs) have emerged as a promising network technology, which combines the benefits of cellular networks and Wireless Local Area Networks (WLANs). In a WMN mesh routers wirelessly relay traffic on behalf of other mesh routers or clients and thereby provide coverage areas comparable to cellular networks, while having the low complexity and low costs of WLANs. As Voice over IP (VoIP) is a very important Internet service, it is critical for the success of WMNs to support high quality VoIP. However, currentWMNs are not adapted well for VoIP. The capacity and scalability of single-radio WMNs is low, especially for small packet transmissions of VoIP calls, because the MAC and PHY layer overhead for small packets is high. The scalability of multiradio/multi-channel WMNs is usually higher, since fewer nodes contend for a channel. However channel scheduling might be required, which can lead to excessive delay and jitter and result in poor VoIP quality. In this thesis we investigate how to deliver high quality VoIP in single radio and multi-radio networks by using cross-layer optimization. For single radio WMNs, we consider the use of IP packet aggregation and IEEE 802.11e transmission opportunities. We conclude that IP packet aggregation greatly improves the capacity and thereby the scalability of WMNs. We show that the key for providing good quality is to artificially delay packets prior to aggregation. We propose a distributed cross-layer optimization system, which, based on Fuzzy Logic Inference, derives an aggregation delay that enhances the capacity and quality. For multi-radio/multi-channel WMNs, we demonstrate the importance of qualityof- service-aware channel scheduling. We develop a quality-of-serviceaware channel scheduler that compared to a basic round-robin scheme significantly reduces jitter and in that way increases VoIP quality. Our analysis shows that there is a trade-off between the jitter of high priority VoIP traffic and the throughput of background TCP traffic. The proposed optimizations significantly increase the capacity of singleradio and multi-radio WMNs. This allows network operators to serve more users with an existing mesh infrastructure or provide better service delivery to existing users.

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Stenman, Peter, and Mikael Janson. "Talöverföring för trygghetslarm över internet : Voice over IP for personal alarms." Thesis, KTH, Skolan för informations- och kommunikationsteknik (ICT), 2016. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-205349.

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During the last couple of years there has been a shift in technology within the Swedish elderly care where the analog personal security alarm is being replaced with personal security alarm that uses internet to communicate. This transition happens due to the ever increasing access to the internet among the elderly and the decreasing availability of analog personal security alarms. This paper describes the work whose purpose is to develop a system that will act as a prototype of a personal security alarm that uses Voice Over IP and the protocol Session Initiation Protocol. The final system is to be comprised by a Raspberry Pi that uses the SIP protocol, a keypad and a soundcard that is built around PCM3060 chip.
De senaste åren pågår det ett teknikskifte inom den svenska äldreomsorgen där de analoga trygghetslarmen ersätts av larm som använder internet för att kommunicera. Denna övergång sker på grund av att tillgång till internet ökar hos äldre personer samt att hushåll med analoga anslutningarna blir allt färre. Denna rapport beskriver arbetet med att ta fram ett system som ska fungera som en prototyp för ett trygghetslarm. Detta system använder sig av Voice Over IP och protokollet Session Initiation Protocol. Det slutliga systemet består av en Raspberry Pi som använder sig av SIP protokollet, en knappsats samt ett ljudkort som är byggt runt ett PCM3060 chip.

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Parperis, Marios S. "Delay estimation and its QoS implications in voice over IP networks." Thesis, National Library of Canada = Bibliothèque nationale du Canada, 2000. http://www.collectionscanada.ca/obj/s4/f2/dsk1/tape4/PQDD_0024/MQ52394.pdf.

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Parperis, Marios S. (Marios Stavrou) Carleton University Dissertation Engineering Systems and Computer. "Delay estimation and its QoS implications in voice over IP networks." Ottawa, 2000.

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Araujo, Maria S., and Ben A. Abbott. "PCM vs. Networking: Spectral Efficiency Wars - A Pragmatic View." International Foundation for Telemetering, 2012. http://hdl.handle.net/10150/581747.

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ITC/USA 2012 Conference Proceedings / The Forty-Eighth Annual International Telemetering Conference and Technical Exhibition / October 22-25, 2012 / Town and Country Resort & Convention Center, San Diego, California
The expected efficiency of network-based telemetry systems vs. the tried and true PCM-based approaches is a debated topic. This paper chooses to use a lighthearted voice to pull the two sides of the "war" to a table of negotiation based on metrics. Ultimately, focusing on metrics that truly define efficiency is the key to understanding the varying points of view. A table of these metrics along with the "why and when" criteria for their use is presented based on historic mathematical information theory, true flight test data requirements, and lab analysis. With these metrics, the negotiation and reasonable compromises in the war may become clear. In other words, this paper attempts to provide a methodology that can be used by the community to aid in choosing the appropriate (or good enough) technologies for current and future telemetry testing demands.

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Johnsson, Sven. "Hardware and software development of a uClinux Voice over IP telephone platform." Thesis, Linköping University, Department of Science and Technology, 2007. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-9455.

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Voice over IP technology (VoIP) has recently gained popularity among consumers. Many popular VoIP services exist only as software for PCs. The need of taking such services out of the PC, into a stand-alone device has been discovered, and this thesis work deals with the development of such a device. The thesis work is done for Häger Scandinavia AB, a Swedish telephone manufacturer. This thesis work covers the design of a complete prototype of a table-top VoIP telephone running an embedded Linux Operating system. Design areas include product development, hardware design and software design.The result is a working prototype with hardware and corresponding Linux device drivers. The prototype can host a Linux application adapted to it. Conclusions are that the first hardware version has worked well and that using an open-source operating system is very useful. Further work consists of implementing a complete telephony software application in the system, evaluation of system requirements and adapting the prototype for a commercial design.

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Li, Hong. "Quality-of-service routing for Voice-over-IP in service overlay networks." Thesis, McGill University, 2010. http://digitool.Library.McGill.CA:80/R/?func=dbin-jump-full&object_id=86808.

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Voice-over-IP (VoIP) becomes more and more popular with the development of service convergence in the next generation network. This thesis focuses on improving VoIP quality with application-layer routing in service overlay networks.
Internet end-to-end delay is one of the most important impairments on VoIP quality. We therefore analyze, model and simulate it to better understand it and thus to discover potential advantages of routing in service overlay networks. Based on the investigation of the Internet end-to-end delay, we find that VoIP quality on a pair of diverse paths is better and more stable than that on a single path. We therefore propose a novel centralized data fusion approach to search for the best pair of diverse paths. This method jointly optimizes source routing with adaptive play-out scheduling at the receiver. It requires transmitting the delay distributions of all the overlay links for estimating the delay distributions of diverse paths. We propose to transmit only the model parameters of the link delay distributions to reduce the communication overhead. It is shown that the best pair of diverse paths can be estimated with a small error.
Nonetheless, the centralized approach is computationally expensive. We therefore propose an online diverse routing method, which uses distributed learning automata to actively probe path delays and to determine the best pair of diverse paths for VoIP based on the state of the learning automata. We have demonstrated the scalability and the optimality of the approach by simulations, and proven the optimality of the approach using Kushner's weak convergence method. VoIP quality has been shown to improve from unsatisfactory levels to satisfactory levels. In addition, we propose a method to detect and recover from link failures based on the state of the learning automata. Considerable improvement in link failure recovery time has been achieved.
In sum, this work demonstrates that the proposed centralized diverse routing approach is effective to improve VoIP quality in terms of R-factor for small overlay networks, and that the proposed distributive diverse routing approach together with the link failure detection scheme provides a scalable, effective and robust solution to VoIP routing for large overlay networks.
Voix sur IP (VoIP) est un service dont la popularité crôıt avec le développement de la convergence entre les services dans les réseaux dits de nouvelle génération. Dans cette thèse, nous nous appliquons à améliorer la qualité des services de VoIP grâce au routage au niveau la couche application en utilisant des réseaux dédiés. Dans cette thèse, nous procédons à l'étude des délais de bout en bout du réseau Internet, qui sont le facteur impactant le plus sur la qualité de la VoIP. Nous analysons, modelons et synthétisons des traces de délais de bout en bout, aﬁn de découvrir un potentiel intérêt relatif à leur utilisation dans le cadre du routage au niveau le la couche application utilisant des réseaux dédiés.
En nous appuyant sur l'étude des traces de délais de bout en bout, nous montrons que la qualité de la VoIP peut être améliorée et stable en utilisant un couple de routes diverses, au lieu d'une seule route. Nous donc proposon un centre de fusion de données qui utilise notre approche pour trouver le meilleur couple de routes diverses. Cette méthode optimise le routage source conjointement avec adaptation play-out au niveau du récepteur. Il faut transmettre des délais des-dites distributions de tous les liens au niveau le la couche application pour estimer des distributions de tous les couples de routes possibles. Nous proposons de transmettre uniquement les paramètres du modèle de la distribution des délais, aﬁn de réduire des côuts de communication. Nous prouvons que cette méthode peut trouver le meilleur couple de routes à une faible erreur.
Comme la technique centralisée requiert une grande puissance de calcul, nous proposons une solution de routage divers extensible en ligne, qui utilise l'apprentissage distribué au- tomata activement sonde des délais de bout en bout et détermine la meilleure paire de diverses voies de VoIP basé sur l'état de l'apprentissage d'automates. Nous avons dé- montré l'extensibilité et l'optimalité de cette approche par les simulations, et démontré l'optimalité de l'approche par l'utilisation de la méthode de convergence faible de Kushner. Nous montrons que la qualité de la VoIP est ameliorée, passant d'une qualité inacceptable à une qualité acceptable. De plus, nous proposons une méthode pour détecter les défaillances du lien et de sa récupération sur la base de paramètres de l'apprentissage d'automates, qui permettent une réduction considérable du temps de récupération à la suite de la défaillance d'un lien.
En somme, cette thèse démontre que la proposition de la diversité de routage centralisé approche est eﬃcace pour améliorer la qualité de la VoIP en termes de R-facteur pour les petits réseaux de la couche application, et que l'apprentissage de un couple de routes diverses avec des méthodes de détection de défaillance oﬀre un extensible, eﬃcace et robuste solution pour services de VoIP grâce au grands réseaux de la couche application.

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Hung, Shirley Kon-Jean. "Managing uncertainty : foresight and flexibility in cryptography and voice over IP policy." Thesis, Massachusetts Institute of Technology, 2008. http://hdl.handle.net/1721.1/49679.

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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Political Science, 2008.
"February 2008."
Includes bibliographical references (p. 235-248).
This main question in this dissertation is under what conditions government agencies show foresight in formulating strategies for managing emerging technologies. A secondary question is when they are capable of adaptation. Conventional wisdom and most organization theory literature suggest that organizations are reactive rather than proactive, reluctant to change, and responsive only to threats to their core mission or autonomy. The technological, economic, social, political, and sometimes security uncertainties that often accompany emerging technologies further complicate decision-making. More generally, organizations must often make decisions under conditions of limited information while guarding against lock-in effects that can constrain future choices. The two cases examined in this dissertation suggest that contrary to conventional wisdom, organizations can show foresight and flexibility in the management of emerging technologies. Key factors that promote foresight are: an organizational focus on technology, with the emerging technology in question being highly relevant to the organization's mission; technical expertise and a recognition of the limits of that knowledge; and experience dealing with other emerging technologies. The NSA recognized the inevitability of mass market encryption early on and adopted a sophisticated strategy of weakening the strength of, reducing the use of, and slowing down the deployment of mass market encryption in order to preserve its ability to easily monitor communications. The Agency showed considerable tactical adaptation in pursuit of this goal. The FCC adopted a rather unusual policy of forbearance toward VoIP. The Commission deliberately refrained from regulating VoIP in order to allow the technology to mature, innovation to occur, uncertainties to resolve, and to avoid potential market distortions due to too early or suboptimally formulated regulation. Eventually, however, pressure from outside interests such as law enforcement forced the Commission to act.
by Shirley K. Hung.
Ph.D.

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43

Majed, Najmeddine. "Measuring and improving the quality of experience of mobile voice over IP." Thesis, Ecole nationale supérieure Mines-Télécom Atlantique Bretagne Pays de la Loire, 2018. http://www.theses.fr/2018IMTA0099/document.

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Les réseaux mobiles 4G basés sur la norme LTE (Long Term Evolution), sont des réseaux tout IP. Les différents problèmes de transport IP comme le retard, la gigue et la perte despaquets peuvent fortement dégrader la qualité des communications temps réel telles que la téléphonie. Les opérateurs ont mis en oeuvre des mécanismes d’optimisation du transport de la voix dans le réseau afin d'améliorer la qualité perçue. Cependant, les algorithmes propriétaires de gestion de la qualité dans les terminaux ne sont pas spécifiés dans les standards. Dans ce contexte, nous nous intéressons aux mécanismes d'adaptation de média, intégrés dans les terminaux afin d'améliorer la qualité d’expérience (QoE). En particulier, nous évaluons de manière expérimentale des métriques QoE de la voix sur LTE (VoLTE) en utilisant une méthode de test standardisée. Nous proposons d’améliorer la méthode de test et discutons la manière dont cette méthode peut être étendue pour évaluer les performances du buffer de gigue. Nous évaluons également de manière expérimentale la qualité de WebRTC dans différentes conditions radios en utilisant un réseau réel. Nous évaluons l'impact du buffer de gigue et de la variation du débit sur la qualité mesurée. Pour améliorer la robustesse des codecs contre la perte de paquets, nous proposons d’utiliser une redondance simple au niveau applicatif. Nous implémentons cette redondance pour le codec EVS (Enhanced Voice Service) et nous évaluons ses performances. Enfin, nous proposons un protocole de signalisation qui permet d’envoyer des requêtes de redondance au cours d’une communication afin d’activer ou désactiver celle-ci dynamiquement
Fourth-generation mobile networks, based on the Long Term Evolution (LTE) standard, are all- IP networks. Thus, mobile telephony providers are facing new types of quality degradations related to the voice packet transport over IP network such as delay, jitter and packet loss. These factors can heavily degrade voice communications quality. The real-time constraint of such services makes them highly sensitive to delay and loss. Network providers have implemented several network optimizations for voice transport to enhance perceived quality. However, the proprietary quality management algorithms implemented in terminals are left unspecified in the standards. In this context, we are interested in media adaptation mechanisms integrated in terminals to enhance the overall Quality of Experience (QoE). In particular, we experimentally evaluate Voice over LTE (VoLTE) QoE metrics such as delay and Mean Opinion Score (MOS) sing a standardized test method. We propose some enhancements to the actual test method and discuss how this method can be extended to evaluate de-jitter buffer performance. We also experimentally evaluate WebRTC voice quality in different radio conditions using a realLTE test network. We evaluate the impact of jitter buffer and bit rate variations on the measured quality. To enhance voice codec robustness against packet loss, we propose a simple application layer redundancy. We implemented it for the Enhanced Voice Service (EVS) codec and evaluate it. Finally, we propose a signaling protocol that allows sending redundancy requests during a call to dynamically activate or deactivate the redundancy mechanism

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Bassil, Carole. "SVSP (Secure Voice over IP Simple Protocol) une solution pour la sécurisation de la voix sur IP." Phd thesis, Télécom ParisTech, 2005. http://pastel.archives-ouvertes.fr/pastel-00001577.

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Depuis l'invention du premier téléphone par Alexandre Graham Bell en 1869, la téléphonie n'a cessé d'évoluer : de la commutation de circuit à la commutation par paquet, d'un réseau fixe à un réseau mobile. Plusieurs architectures ont été crées où la voix est combinée aux données et à l'imagerie pour être transportées sur un seul réseau data. La nature de ces réseaux ouverts a un impact sur la voix en terme de sécurité. D'où le besoin imminent de sécuriser la voix tout en assurant une bonne qualité de service à la voix aussi bien dans un réseau fixe que dans un réseau mobile ou IP. Des solutions de sécurisation sont proposées pour les données. Mais des solutions partielles voire incomplètes sont proposées pour la voix. Dans un premier temps, nous définissons les besoins de la sécurisation de la téléphonie et les services de sécurité nécessaires à son déploiement. Ainsi, nous analysons la sécurité offerte par les différents réseaux offrant un service de téléphonie, à savoir la sécurité dans le réseau téléphonique traditionnel (RTC et RNIS), dans les réseaux mobiles (GSM et UMTS), et dans le réseau de données IP avec les deux architectures prépondérantes H.323 et SIP. Ceci va nous permettre de comparer les solutions de sécurité offertes par ces architectures de téléphonie et de pouvoir aborder d'une part, leur avantages et inconvénients et d'autre part, les exigences qu'ils ne peuvent pas satisfaire. Cette analyse nous conduit à un résultat éloquent qui est l'absence d'une solution de sécurité complète qui répond aux exigences de la téléphonie et qui permet d'effectuer un appel sécurisé de bout en bout.

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Bassil, Carole. "SVSP (Secure Voice over IP Simple Protocole) : une solution pour la sécurisation de la voix sur IP." Paris, ENST, 2005. http://www.theses.fr/2005ENST0045.

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Depuis l invention du premier téléphone par Alexandre Graham Bell en 1869, la téléphonie n a cessé d évoluer : de la commutation de circuit à la commutation par paquet, d un réseau fixe à un réseau mobile. Plusieurs architectures ont été crées où la voix est combinée aux données et à l imagerie pour être transportées sur un seul réseau data. La nature de ces réseaux ouverts a un impact sur la voix en terme de sécurité. D où le besoin imminent de sécuriser la voix tout en assurant une bonne qualité de service à la voix aussi bien dans un réseau fixe que dans un réseau mobile ou IP. Des solutions de sécurisation sont proposées pour les données. Mais des solutions partielles voire incomplètes sont proposées pour la voix. Dans un premier temps, nous définissons les besoins de la sécurisation de la téléphonie et les services de sécurité nécessaires à son déploiement. Ainsi, nous analysons la sécurité offerte par les différents réseaux offrant un service de téléphonie, à savoir la sécurité dans le réseau téléphonique traditionnel (RTC et RNIS), dans les réseaux mobiles (GSM et UMTS), et dans le réseau de données IP avec les deux architectures prépondérantes H. 323 et SIP. Ceci va nous permettre de comparer les solutions de sécurité offertes par ces architectures de téléphonie et de pouvoir aborder d une part, leur avantages et inconvénients et d autre part, les exigences qu ils ne peuvent pas satisfaire. Cette analyse nous conduit à un résultat éloquent qui est l absence d une solution de sécurité complète qui répond aux exigences de la téléphonie et qui permet d effectuer un appel sécurisé de bout en bout. Dans un second temps, nous proposons une architecture de sécurité pour une architecture de téléphonie abstraite. Cette architecture propose une nouvelle couche de service qui s'insère entre deux couches de niveaux N et N+1 du modèle de l'OSI. Ce choix permet d'assurer une transparence et une indépendance du réseau sous-jacent mais nécessite de revoir les interfaces et donc de définir une API qui s'occupe d'assurer cette transparence et l'interfaçage entre l'application de sécurité et le réseau sous-jacent. Cette architecture assure les services de sécurité et définit les politiques de sécurité nécessaires pour répondre aux exigences de la sécurisation des communications de voix. Notre proposition s'accompagne de la définition d'un protocole de sécurité que nous avons nommé SVSP pour Simple Voice Security Protocol. SVSP intègre les services de sécurité définis par cette architecture et permet la sécurisation d'un appel téléphonique de bout en bout. Des études ont été effectuées pour l'intégrer dans différentes infrastructures de téléphonie comme le réseau fixe, le réseau mobile GSM et une plate-forme de voix sur IP avec le standard H. 323. Un prototype de SVSP a été implémenté suivi d'une intégration avec SIP, le standard de voix sur IP de l'IETF
Since the invention of the first telephone by Alexander Graham Bell in 1869, network telephony technology did not stop evolving: from circuit switching to packet switching, from fixed network to wireless network. Several new architectures were created which combines the transport of voice, data and image in the same data network. The nature of these open networks has an impact on the voice in terms of security. This yields to the imminent need to secure voice communications while insuring a good quality of service to the voice as well in fixed, wireless and IP networks. Different security solutions are proposed for the data. But partial even incomplete solutions are proposed for the voice. First, we define the needs for securing the telephony and the security services required. Thus, we analyze the security offered by the different telephone networks, namely the security in the traditional telephone network (PSTN and ISDN), in the mobile networks (GSM and UMTS), and in the IP network based on the H. 323 and SIP architectures. This will allow us to compare the security solutions offered by these telephony architectures and to be able to present their advantages and limitations and the security requirements that they cannot satisfy. This analysis drives us to an eloquent result that is the absence of a complete end to end security solution that complies with the security requirements of telephony. Secondly, we propose security architecture for a unified telephony architecture. This security architecture proposes a service layer that is inserted between N and N + 1 layers of the OSI reference model. This choice provides a transparency and an independence of the underlying network but requires reviewing the interfaces and therefore the needs to define an API between the security application and the underlying network that insures transparency. This architecture provides the security services and defines necessary security policies to secure voice communications. Following the security architecture, we defined a security protocol that we named SVSP for Simple Voice Security Protocol. SVSP satisfies the security services defined by this architecture that provides a secure end-to-end phone call. Studies were carried out to integrate it in different telephony infrastructures, namely with the traditional telephone network, GSM the mobile network and with the H. 323 standard for voice over IP communications. A prototype of SVSP was implemented followed by integrating it with SIP the IETF voip standard

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46

Bassil, Carole. "SVSP, Secure voice over IP simple protocole, une solution pour la sécurisation de la voix sur IP /." Paris : École nationale supérieure des télécommunications, 2006. http://catalogue.bnf.fr/ark:/12148/cb40208342h.

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Junghänel, Jens. "VOCAL-Einsatz an der TU Chemnitz." Universitätsbibliothek Chemnitz, 2003. http://nbn-resolving.de/urn:nbn:de:swb:ch1-200301347.

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SCHEINER, LEONARDO NAHMIAS. "PERFOMANCE ANALISYS OF SIP PROTOCOL ON THE SIGNALING OF VOICE OVER IP CALLS." PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO, 2005. http://www.maxwell.vrac.puc-rio.br/Busca_etds.php?strSecao=resultado&nrSeq=7065@1.

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COORDENAÇÃO DE APERFEIÇOAMENTO DO PESSOAL DE ENSINO SUPERIOR
Impulsionada pelo grande crescimento da Internet, a telefonia IP conquistou a atenção do mercado e dos grandes fabricantes com promessas de redução de custo na operacão, gerência, provisionamento, manutenção e tarifação. Diversos protocolos foram desenvolvidos de modo a prover VoIP como o H.323, MGCP, Megaco e SIP. O SIP tem se destacado por ser um protocolo baseado em texto, estensível, independente do protocolo de transporte, e portanto mais flexível e simples que seu concorrente direto, o H.323. O SIP (Session Initiation Protocol) é um protocolo de sinalização utilizado para iniciar, modificar e terminar sessões, podendo ser usado para chamadas de voz sobre IP (VoIP) ou para troca de mensagens instantâneas, entre outras aplicações. Ele foi desenvolvido originalmente em 1996 e foi padronizado pela IETF em 1999. Neste trabalho, o desempenho do protocolo SIP para estabelecimento de chamadas VoIP será avaliado, já que há uma grande quantidade de trabalhos focando a qualidade da voz e poucos têm avaliado a sinalização [3]. Serão montados ambientes experimentais a fim de variar parâmetros como retardo, perda de pacotes, jitter, largura de banda e protocolo de transporte, permitindo verificar como esses parâmetros afetam isoladamente os tempos de post-dial delay, post-pickup delay e call release delay.
Pushed by the growth of the Internet, the IP Telephony conquered a great attention of the market and big suppliers, with promises of cost reductions on operation, management, provisioning, maintenance and billing. Different protocols were developed for providing VoIP such as H.323, MGCP, Megaco and SIP. SIP has been highlighted for being a text based protocol, extensible, independent of the transport protocol, therefore more flexible and simpler than your competitor, the H.323. SIP (Session Initiation Protocol) is a signaling protocol used for establish, modify and terminate sessions. It can be used for voice calls over IP (VoIP) or to exchange instant messaging, among other applications. It has been developed originally in 1996 and has been standardized by IETF in 1999. In this work, the performance of SIP protocol for establishing VoIP calls will be estimated, since there are a lot of papers focalizing in the voice quality and few treated the signaling [3]. Experimental environments will be used for varying parameters like delay, packet loss, jitter, bandwidth and transport protocol, allowing to verify how there parameters affect separately the post-dial delay, post-pickup delay and call release delay.

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Escobedo, Gonzalez Maiz Marco Antonio 1976. "Convivo communicator : an interface-adaptive voice over IP system for poor quality networks." Thesis, Massachusetts Institute of Technology, 2002. http://hdl.handle.net/1721.1/61126.

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Thesis (S.M.)--Massachusetts Institute of Technology, School of Architecture and Planning, Program in Media Arts and Sciences, 2002.
Includes bibliographical references (leaves 100-102).
This thesis presents Convivo, a VoIP system designed to provide reliable voice communication for poor quality networks, especially those found in rural areas of the developing world. Convivo introduces an original approach to maintain voice communication interaction in presence of poor network performance: an Interface-Adaptation mechanism that suggests adjusting the application user interface to conform to one of three voice communication modalities (full duplex, half duplex, and voice messaging). The thesis proposes that changes in communication modality are an option to sustain voice communication interaction despite poor network performance. The goals of the changes in communication modality are to reduce the impact of high latency and low bandwidth on voice communication interaction, to facilitate turn taking for a high latency connection, and to sustain voice communication for extremely low bandwidth or high error links. The system was tested via a user study in Bohechio, a small village in the Dominican Republic. The study found that Interface-Adaptation helped users to maintain voice communication interaction when network performance degrades. Transitions from full duplex to voice messaging were found particularly valuable. Initial results suggest that as users get more experience with the application they would like to manually control transitions based on feedback provided by the application and their own perceived voice quality.
by Marco Antonio Escobedo Gonzalez Maiz.
S.M.

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Shaikh, A. D. "Modelling data and voice traffic over IP networks using continuous-time Markov models." Thesis, Aston University, 2009. http://publications.aston.ac.uk/15385/.

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Common approaches to IP-traffic modelling have featured the use of stochastic models, based on the Markov property, which can be classified into black box and white box models based on the approach used for modelling traffic. White box models, are simple to understand, transparent and have a physical meaning attributed to each of the associated parameters. To exploit this key advantage, this thesis explores the use of simple classic continuous-time Markov models based on a white box approach, to model, not only the network traffic statistics but also the source behaviour with respect to the network and application. The thesis is divided into two parts: The first part focuses on the use of simple Markov and Semi-Markov traffic models, starting from the simplest two-state model moving upwards to n-state models with Poisson and non-Poisson statistics. The thesis then introduces the convenient to use, mathematically derived, Gaussian Markov models which are used to model the measured network IP traffic statistics. As one of the most significant contributions, the thesis establishes the significance of the second-order density statistics as it reveals that, in contrast to first-order density, they carry much more unique information on traffic sources and behaviour. The thesis then exploits the use of Gaussian Markov models to model these unique features and finally shows how the use of simple classic Markov models coupled with use of second-order density statistics provides an excellent tool for capturing maximum traffic detail, which in itself is the essence of good traffic modelling. The second part of the thesis, studies the ON-OFF characteristics of VoIP traffic with reference to accurate measurements of the ON and OFF periods, made from a large multi-lingual database of over 100 hours worth of VoIP call recordings. The impact of the language, prosodic structure and speech rate of the speaker on the statistics of the ON-OFF periods is analysed and relevant conclusions are presented. Finally, an ON-OFF VoIP source model with log-normal transitions is contributed as an ideal candidate to model VoIP traffic and the results of this model are compared with those of previously published work.

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Dissertations / Theses on the topic 'Voice over IP'

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