Signaling for Internet Telephony

Internet telephony must offer the standard telephony services.However, the transition to Internet-based telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network(PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services

The Impact of SCTP on Server Scalability and Performance

The Stream Control Transmission Protocol (SCTP) is a newer transport protocol, having additional features to TCP. Although SCTP is an alternative transport protocol for the Session Initiation Protocol (SIP), we do not know how SCTP features influence SIP server scalability and performance. To estimate this, we measured the scalability and performance of two servers, an echo server and a simplified SIP server on Linux, comparing to TCP. Our measurements found that using SCTP does not significantly affect on data latency: approximately 0.3 ms longer for the handshake than that for TCP. However, server scalability in terms of the number of sustainable associations drops to 17-21%, or to 43% of TCP if we adjust the acceptable gap size of unordered data delivery

One Server Per City: Using TCP for Very Large SIP Servers

The transport protocol for SIP can be chosen based on the requirements of services and network conditions. How does the choice of TCP affect the scalability and performance compared to UDP? We experimentally analyze the impact of using TCP as a transport protocol for a SIP server. We first investigate scalability of a TCP echo server, then compare performance of a SIP server for three TCP connection lifetimes: transaction, dialog, and persistent. Our results show that a Linux machine can establish 450,000+ TCP connections and maintaining connections does not affect the transaction response time. Additionally, the transaction response times using the three TCP connection lifetimes and UDP show no significant difference at 2,500 registration requests/second and at 500 call requests/second. However, sustainable request rate is lower for TCP than for UDP, since using TCP requires more message processing. More message processing causes longer delays at the thread queue for the server implementing a thread-pool model. Finally, we suggest how to reduce the impact of TCP for a scalable SIP server especially under overload control. This is applicable to other servers with very large connection counts

Programming Internet Telephony Services

Internet telephony enables a wealth of new service possibilities. Traditional telephony services, such as call forwarding, transfer, and 800 number services, can be enhanced by interaction with email, web, and directory services. Additional media types, like video and interactive chat, can be added as well. One of the challenges in providing these services is how to effectively program them. Programming these services requires decisions regarding where the code executes, how it interfaces with the protocols that deliver the services, and what level of control the code has. In this paper, we consider this problem in detail. We develop requirements for programming Internet telephony services, and we show that at least two solutions are required --- one geared for service creation by trusted users (such as administrators), and one geared for service creation by untrusted users (such as consumers). We review existing techniques for service programmability in the Internet and in the telephone network,and extract the best components of both. The result is a Common Gateway Interface (CGI) that allows trusted users to develop services, and the Call Processing Language (CPL) that allows untrusted users to develop services

A comparative study of in-band and out-of-band VOIP protocols in layer 3 and layer 2.5 environments

For more than a century the classic circuit-switched telephony in the form of PSTN (Public Service Telephone Network) has dominated the world of phone communications (Varshney et al., 2002). The alternative solution of VoIP (Voice over Internet Protocol) or Internet telephony has increased dramatically its share over the years though. Originally started among computer enthusiasts, nowadays it has become a huge research area in both the academic community as well as the industry (Karapantazis and Pavlidou, 2009). Therefore, many VoIP technologies have emerged in order to offer telephony services. However, the performance of these VoIP technologies is a key issue for the sound quality that the end-users receive. When making reference to sound quality PSTN still stands as the benchmark.Against this background, the aim of this project is to evaluate different VoIP signalling protocols in terms of their key performance metrics and the impact of security and packet transport mechanisms on them. In order to reach this aim in-band and out-of-band VoIP signalling protocols are reviewed along with the existing security techniques which protect phone calls and network protocols that relay voice over packet-switched systems. In addition, the various methods and tools that are used in order to carry out performance measurements are examined together with the open source Asterisk VoIP platform. The findings of the literature review are then used in order to design and implement a novel experimental framework which is employed for the evaluation of the in-band and out-of-band VoIP signalling protocols in respect to their key performance networks. The major issue of this framework though is the lack of fine-grained clock synchronisation which is required in order to achieve ultra precise measurements. However, valid results are still extracted. These results show that in-band signalling protocols are highly optimised for VoIP telephony and outperform out-of-band signalling protocols in certain key areas. Furthermore, the use of VoIP specific security mechanisms introduces just a minor overhead whereas the use of Layer 2.5 protocols against the Layer 3 routing protocols does not improve the performance of the VoIP signalling protocols

Service provisioning in two open-source SIP implementation, cinema and vocal

The distribution of real-time multimedia streams is seen nowadays as the next step forward for the Internet. One of the most obvious uses of such streams is to support telephony over the Internet, replacing and improving traditional telephony. This thesis investigates the development and deployment of services in two Internet telephony environments, namely CINEMA (Columbia InterNet Extensible Multimedia Architecture) and VOCAL (Vovida Open Communication Application Library), both based on the Session Initiation Protocol (SIP) and open-sourced. A classification of services is proposed, which divides services into two large groups: basic and advanced services. Basic services are services such as making point-to-point calls, registering with the server and making calls via the server. Any other service is considered an advanced service. Advanced services are defined by four categories: Call Related, Interactive, Internetworking and Hybrid. New services were implemented for the Call Related, Interactive and Internetworking categories. First, features involving call blocking, call screening and missed calls were implemented in the two environments in order to investigate Call-related services. Next, a notification feature was implemented in both environments in order to investigate Interactive services. Finally, a translator between MGCP and SIP was developed to investigate an Internetworking service in the VOCAL environment. The practical implementation of the new features just described was used to answer questions about the location of the services, as well as the level of required expertise and the ease or difficulty experienced in creating services in each of the two environments.KMBT_363Adobe Acrobat 9.54 Paper Capture Plug-i

Service provisioning in two open-source SIP implementation, cinema and vocal

The distribution of real-time multimedia streams is seen nowadays as the next step forward for the Internet. One of the most obvious uses of such streams is to support telephony over the Internet, replacing and improving traditional telephony. This thesis investigates the development and deployment of services in two Internet telephony environments, namely CINEMA (Columbia InterNet Extensible Multimedia Architecture) and VOCAL (Vovida Open Communication Application Library), both based on the Session Initiation Protocol (SIP) and open-sourced. A classification of services is proposed, which divides services into two large groups: basic and advanced services. Basic services are services such as making point-to-point calls, registering with the server and making calls via the server. Any other service is considered an advanced service. Advanced services are defined by four categories: Call Related, Interactive, Internetworking and Hybrid. New services were implemented for the Call Related, Interactive and Internetworking categories. First, features involving call blocking, call screening and missed calls were implemented in the two environments in order to investigate Call-related services. Next, a notification feature was implemented in both environments in order to investigate Interactive services. Finally, a translator between MGCP and SIP was developed to investigate an Internetworking service in the VOCAL environment. The practical implementation of the new features just described was used to answer questions about the location of the services, as well as the level of required expertise and the ease or difficulty experienced in creating services in each of the two environments.KMBT_363Adobe Acrobat 9.54 Paper Capture Plug-i

Distributed algorithms for dynamic topology construction and their applications

Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2004.Includes bibliographical references (p. 155-162).(cont.) of piconets is close to optimal, and any device is a member of at most two piconets.We introduce new distributed algorithms that dynamically construct network topologies. These algorithms not only adapt to dynamic topologies where nodes join and leave, but also actively set up and remove links between the nodes, to achieve certain global graph properties. First, we present a novel distributed algorithm for constructing overlay networks that are composed of d Hamilton cycles. The protocol is decentralized as no globally-known server is required. With high probability, the constructed topologies are expanders with O(logd n) diameters and ... second largest eigenvalues. Our protocol exploits the properties of random walks on expanders. A new node can join the network in O(logd n) time with O(dlogd n) messages. A node can leave in O(1) time with O(d) messages. Second, we investigate a layered construction of the random expander networks that can implement a distributed hash table. Layered expanders can achieve degree-optimal routing at O(log n/log log n) time, where each node has O(log n) neighbors. We also analyze a self-balancing scheme for the layered networks. Third, we study the resource discovery problem, in which a network of machines discover one another by making network connections. We present two randomized algorithms to solve the resource discovery problem in O(log n) time. Fourth, we apply the insight gained from the resource discovery algorithms on general networks to ad hoc wireless networks. A Bluetooth ad hoc network can be formed by interconnecting piconets into scatternets. We present and analyze a new randomized distributed protocol for Bluetooth scatternet formation. We prove that our protocol achieves O(log n) time complexity and O(n) message complexity. In the scatternets formed by our protocol, the numberby Ching Law.Ph.D

Måling av talekvalitet over IP

Denne rapporten handler om subjektiv talekvalitetstesting av IP-telefonitale brukt i luftfartskommunikasjon (her: typisk mellom pilot og flygeleder). Testingen har vært webbasert og har foregått ved at flygeledere har lyttet til digitale lydopptak beskjemmet med kontrollerte feil som de lastet ned fra prosjektets hjemmeside. Disse lydopptakene var basert på replikkutveksling, hvor replikkene var hentet fra opplæringsmateriale i flygeledertale og illustrerte kommunikasjon mellom fly og tårn. Det ble det utviklet et socket-program, dgramServer, som ble benyttet til å innstille en viss mengde pakketap i prosent eller pakkeforsinkelse i millisekunder som IP-telefonitalen ble utsatt for. På denne måten ble det simulert at talen ble utsatt for varierte nettverksforhold. Slik ble talekvaliteten på lydopptakene også variert. Fra hjemmesiden lastet flygelederende også ned et evalueringsskjema som de fylte inn etter avspilling av lydopptakene og returnerte via e-post. Dette skjemaet inneholdt skalaer for bedømmelse av talekvalitet, hvor mye innsats som trengtes for å forstå talen og om talekvaliteten var tilstrekkelig for luftfartskommunikasjon. De viktigste funnene i testingene var: *Et flertall på over 50% av flygelederne bedømte at alle testparametrene for pakkeforsinkelse holdt en talekvalitet tilstrekkelig for ATC. *Et flertall på over 50% av flygelederne bedømte at alle testparametrene for pakketap ikke holdt en talekvalitet tilstrekkelig for ATC. *Pakkeforsinkelse har mindre negativ påvirkning på talekvaliteten enn pakketap. *Pakketap opptil 5% har potensiale til å bli godkjent for bruk i ATC

A Decentralized Session Management Framework for Heterogeneous Ad-Hoc and Fixed Networks

Wireless technologies are continuously evolving. Second generation cellular networks have gained worldwide acceptance. Wireless LANs are commonly deployed in corporations or university campuses, and their diffusion in public hotspots is growing. Third generation cellular systems are yet to affirm everywhere; still, there is an impressive amount of research ongoing for deploying beyond 3G systems. These new wireless technologies combine the characteristics of WLAN based and cellular networks to provide increased bandwidth. The common direction where all the efforts in wireless technologies are headed is towards an IP-based communication. Telephony services have been the killer application for cellular systems; their evolution to packet-switched networks is a natural path. Effective IP telephony signaling protocols, such as the Session Initiation Protocol (SIP) and the H 323 protocol are needed to establish IP-based telephony sessions. However, IP telephony is just one service example of IP-based communication. IP-based multimedia sessions are expected to become popular and offer a wider range of communication capabilities than pure telephony. In order to conjoin the advances of the future wireless technologies with the potential of IP-based multimedia communication, the next step would be to obtain ubiquitous communication capabilities. According to this vision, people must be able to communicate also when no support from an infrastructured network is available, needed or desired. In order to achieve ubiquitous communication, end devices must integrate all the capabilities necessary for IP-based distributed and decentralized communication. Such capabilities are currently missing. For example, it is not possible to utilize native IP telephony signaling protocols in a totally decentralized way. This dissertation presents a solution for deploying the SIP protocol in a decentralized fashion without support of infrastructure servers. The proposed solution is mainly designed to fit the needs of decentralized mobile environments, and can be applied to small scale ad-hoc networks or also bigger networks with hundreds of nodes. A framework allowing discovery of SIP users in ad-hoc networks and the establishment of SIP sessions among them, in a fully distributed and secure way, is described and evaluated. Security support allows ad-hoc users to authenticate the sender of a message, and to verify the integrity of a received message. The distributed session management framework has been extended in order to achieve interoperability with the Internet, and the native Internet applications. With limited extensions to the SIP protocol, we have designed and experimentally validated a SIP gateway allowing SIP signaling between ad-hoc networks with private addressing space and native SIP applications in the Internet. The design is completed by an application level relay that permits instant messaging sessions to be established in heterogeneous environments. The resulting framework constitutes a flexible and effective approach for the pervasive deployment of real time applications.The invention of the phone has radically changed the way people communicate, as it allowed persons to get in contact instantly no matter of their location. However, phone communication has been confined for decades to a fixed location, be it one's own house or a phone boot. The widespread affirmation of cellular technologies has had for fixed telephony a similar impact that the invention of the phone has had on communications years before. With mobile phones, people are enabled to talk with each other anytime and anywhere. Internet has also revolutionized the way people communicate. E-mails have soon become one of the Internet killer applications. Later on, instant messaging, popularly known as chatting, has gained huge consensus among net surfers. Only recently, the use of the Internet for voice communication is becoming mainstream, and the so called Voice over IP (VoIP) applications (Skype is probably the most famous for the masses) are becoming common use. Despite its popularity, Internet still suffers from the inherent limitations that affected early telephony: it is fixed. The usage of Internet on the move still does not constitute the easiest and most satisfactory user experience, due to capabilities and limitations of the access technology, terminals, services and applications. Efforts for mobilizing the Internet are ongoing both in the industrial and in the academic worlds, but several bricks are needed to build the wall of mobile Internet. This dissertation provides one of these bricks, describing a solution that allows the deployment of multimedia applications (chat, VoIP, gaming) in mobile environments. In other words, this dissertation gives solutions for facilitating ubiquitous Internet-based communication, anytime and anywhere. The vision that we want to become true is that Internet must become mobile in the same way as fixed telephony has become mobile thanks to the cellular technology. More than this, we do not want that users are limited by the presence of an infrastructure to communicate with each other. In order to achieve this, we present solutions to deploy Internet-based services and applications in environments where no support from servers is available. In other words, we enable direct device-to-device, user-to-user Internet communication. Our contribution is mainly focused on the steps needed to establish the communication, the so called session establishment or signaling phase. We have validated our signaling framework by building a chat application that utilizes its features and works in server-less environments. The custom server-less solution does not prohibit to connect at the same time with the Internet, so that one can engage in a chess game using direct communication with a person in the proximity while having a chat in progress with a friend using standard Internet services. The challenge that we had to face is that Internet services and applications are usually built implying support from a centralized server. In order to deploy direct user-to-user Internet services, while maintaining interoperability with mainstream services, we had to enhance native Internet services to work without infrastructure support, without sacrificing interoperability with standard Internet applications. To conclude, we have placed our brick on the still yet to be completed wall of mobile Internet. Our hope is that one day, thanks also to this brick, everybody will be able to enjoy Internet-based applications as easily as now it is possible to use mobile telephony services