The Experts below are selected from a list of 237 Experts worldwide ranked by ideXlab platform

H. G. Schulzrinne - One of the best experts on this subject based on the ideXlab platform.

  • an analysis of the skype peer to peer Internet Telephony protocol
    IEEE International Conference Computer and Communications, 2006
    Co-Authors: Salma A Ase, H. G. Schulzrinne
    Abstract:

    Skype is a peer-to-peer VoIP client developed in 2003 by the organization that created Kazaa. Skype claims that it can work almost seamlessly across NATs and firewalls and has better voice quality than other VoIP clients. It encrypts calls end-to-end, and stores user information in a decentralized fashion. Skype also supports instant messaging and conferencing. This paper analyzes key Skype functions such as login, NAT and firewall traversal, call establishment, media transfer, codecs, and conferencing under three different network setups. Analysis is performed by careful study of the Skype network traffic and by intercepting the shared library and system calls of Skype. We draw a map of super nodes to which Skype establishes a TCP connection at login.

  • reliable scalable and interoperable Internet Telephony
    2006
    Co-Authors: H. G. Schulzrinne, Kunda Singh
    Abstract:

    The public switched telephone network (PSTN) provides ubiquitous availability and very high scalability of more than a million busy hour call attempts per switch. If large carriers are to adopt Internet Telephony, then Internet Telephony servers should offer at least similar quantifiable guarantees for scalability and reliability using metrics such as call setup latency, server call handling capacity, busy hour call arrivals, mean-time between failures and mean-time to recover. This thesis presents a reliable, scalable and interoperable Internet Telephony architecture for user registration, call routing, conferencing and unified messaging using commodity hardware. The results extend beyond Internet Telephony to encompass multimedia communication in general. The architecture presented in this thesis deals with two aspects: at least PSTN-grade reliability and scalability of the Internet Telephony servers, and interoperable Internet Telephony services such as conferencing and voice mail using existing protocols. We describe the architecture and implementation of our Session Initiation Protocol (SIP)-based enterprise Internet Telephony architecture known as Columbia Internet Extensible Multimedia Architecture (CINEMA). It consists of a SIP registration and proxy server, a multi-party conferencing server, a gateway for interworking SIP with ITU's H.323, an interactive voice response system and a multimedia mail server. CINEMA provides a distributed interoperable architecture for collaboration using synchronous communications like multimedia conferencing, instant messaging, shared web-browsing, and asynchronous communications like discussion forum, shared files, voice and video mails. It allows seamless integration with various communication means like telephone, IP phone, web and electronic mail. We present two techniques for providing scalability and reliability in SIP: server redundancy and a novel peer-to-peer architecture. For the former, we use DNS-based load sharing among multiple distributed servers that use backend SQL databases to maintain user records. Our two-stage architecture scales linearly with the number of servers. For the latter, we propose a peer-to-peer Internet Telephony architecture that supports basic user registration and call setup as well as advanced services such as offline message delivery, voice mail and multi-party conferencing using SIP. It interworks with server-based SIP infrastructures.

  • An Analysis of the Skype Peer-to-Peer Internet Telephony Protocol
    Proceedings IEEE INFOCOM 2006. 25TH IEEE International Conference on Computer Communications, 2006
    Co-Authors: Salman A. Baset, H. G. Schulzrinne
    Abstract:

    Skype is a peer-to-peer VoIP client developed by KaZaa in 2003. Skype claims that it can work almost seamlessly across NATs and firewalls and has better voice quality than the MSN and Yahoo IM applications. It encrypts calls end-to-end, and stores user information in a decentralized fashion. Skype also supports instant messaging and conferencing. This report analyzes key Skype functions such as login, NAT and firewall traversal, call establishment, media transfer, codecs, and conferencing under three different network setups. Analysis is performed by careful study of Skype network traffic.

  • location based services in Internet Telephony
    Consumer Communications and Networking Conference, 2005
    Co-Authors: H. G. Schulzrinne
    Abstract:

    Many applications used in the Internet today benefit from using location information. To better handle location information in Internet Telephony applications, we did a comprehensive application-layer analysis of location information and location-based communication services. We first summarize and categorize end-user-oriented location description and location detection approaches. We then summarize and categorize how to use location information to provide communication services and introduce several interesting location based communication services. Based on the analysis, we have incorporated location-based service handling in our session initiation protocol (SIP) based Internet Telephony infrastructure and our language for end system services (LESS).

  • Peer-to-peer Internet Telephony using SIP
    Proceedings of the international workshop on Network and operating systems support for digital audio and video - NOSSDAV '05, 2005
    Co-Authors: Kundan Singh, H. G. Schulzrinne
    Abstract:

    P2P systems inherently have high scalability, robustness and fault tolerance because there is no centralized server and the network self-organizes itself. This is achieved at the cost of higher latency for locating the resources of interest in the P2P overlay network. Internet Telephony can be viewed as an application of P2P architecture where the participants form a self-organizing P2P overlay network to locate and communicate with other participants. We propose a pure P2P architecture for the Session Initiation Protocol (SIP)-based IP Telephony systems. Our P2P-SIP architecture supports basic user registration and call setup as well as advanced services such as offline message delivery, voice/video mails and multi-party conferencing. Additionally, we give an overview of our implementation.

Jonatha Rosenberg - One of the best experts on this subject based on the ideXlab platform.

  • the ietf Internet Telephony architecture and protocols
    IEEE Network, 1999
    Co-Authors: H. G. Schulzrinne, Jonatha Rosenberg
    Abstract:

    Internet Telephony was first used as a simple way to provide point-to-point voice transport between two IP hosts. However, the growing interest in providing integrated voice, data, and video services has caused its scope to be extended. Internet Telephony now encompasses a range of services, including not only traditional conferencing, call control, multimedia, and mobility services, but also new ones that integrate Web, e-mail, presence, and instant messaging applications with Telephony. Internet Telephony and traditional circuit-switched Telephony will coexist for quite some time, requiring interworking between the two. In this article we present a suite of protocols, developed in the IETF, which provide a partial solution to this complex problem.

  • programming Internet Telephony services
    IEEE Network, 1999
    Co-Authors: Jonatha Rosenberg, Jonatha Lenno, H. G. Schulzrinne
    Abstract:

    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 e-mail, 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 article 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 that allows trusted users to develop services, and the call processing language that allows untrusted users to develop services.

  • programming Internet Telephony services
    IEEE Internet Computing, 1999
    Co-Authors: Jonatha Rosenberg, Jonatha Lenno, H. G. Schulzrinne
    Abstract:

    The Internet offers an opportunity to enhance traditional Telephony services, such as call forwarding, through interaction with e-mail, the Web, and directory services, as well as traditional media types. How do you effectively program these services? The authors propose a CGI (common gateway interface) solution for trusted user/developers (such as administrators) and the Call Processing Language (CPL)-a simple, robust, safe, call processing language-for untrusted user/developers.

  • Internet Telephony architecture and protocols an ietf perspective
    Computer Networks, 1999
    Co-Authors: H. G. Schulzrinne, Jonatha Rosenberg
    Abstract:

    Internet Telephony offers the opportunity to design a global multimedia communications system that may eventually replace the existing Telephony infrastructure. We describe the upper-layer protocol components that are specific to Internet Telephony services: the Real-Time Transport Protocol (RTP) to carry voice and video data, and the Session Initiation Protocol (SIP) for signaling. We also mention some complementary protocols, including the Real Time Streaming Protocol (RTSP) for control of streaming media, and the Wide Area Service Discovery Protocol (WASRV) for location of Telephony gateways.

  • signaling for Internet Telephony
    International Conference on Network Protocols, 1998
    Co-Authors: H. G. Schulzrinne, Jonatha Rosenberg
    Abstract:

    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). We describe SIP, and show how its basic primitives can be used to construct a wide range of Telephony services.

Roma Schlegel - One of the best experts on this subject based on the ideXlab platform.

  • on spam over Internet Telephony spit prevention
    IEEE Communications Magazine, 2008
    Co-Authors: J Quittek, Sandra Tartarelli, Saverio Niccolini, Roma Schlegel
    Abstract:

    Spam over IP Telephony (SPIT) is expected to become a serious problem in the near future. One of the main requirements for a SPIT prevention system is to avoid the involvement of the callee in the SPIT detection process, because this implies disturbing the callee each time an unclassified potential SPIT call arrives. Further requirements include adaptability to different deployment scenarios and flexibility to quickly react to new kinds of SPIT that bypass existing prevention systems. This article analyzes the requirements for SPIT prevention, provides a systematic classification of currently known SPIT prevention methods, and introduces a reference model for SPIT prevention systems. As an instance of the reference model, we designed and implemented an advanced SPIT prevention system, composed of methods that avoid unnecessary callee interaction and adaptive in order to be customized for different scenarios.

  • ise03 2 spam over Internet Telephony spit prevention framework
    Global Communications Conference, 2006
    Co-Authors: Roma Schlegel, Saverio Niccolini, Sandra Tartarelli, Marcus Unne
    Abstract:

    SPam over Internet Telephony (SPIT) is expected to become a threat inhibiting the delivery of voice services over the Internet in the near future both because of its technical and economical characteristics. Experiences with email SPAM and its analogies with SPIT suggest that SPIT will be difficult to detect with a single detection method. Moreover, personalized management of detection strategies will be needed to increase effectiveness and to adapt the methods to special environments. This paper presents a modular framework for SPIT prevention designed to be easily manageable and extensible. Additionally, the framework makes use of a two-stage architecture in order to exploit the knowledge coming both from the signaling and from the media flows while still allowing real-time delivery of the media content to the user. Following the experience gained with our prototype implementation we also present the most important design issues we encountered and suggest solutions to these issues.

  • spam over Internet Telephony spit prevention framework
    Global Communications Conference, 2006
    Co-Authors: Roma Schlegel, Saverio Niccolini, Sandra Tartarelli, Marcus Unne
    Abstract:

    SPam over Internet Telephony (SPIT) is expected to become a threat inhibiting the delivery of voice services over the Internet in the near future both because of its technical and economical characteristics. Experiences with email SPAM and its analogies with SPIT suggest that SPIT will be difficult to detect with a single detection method. Moreover, personalized management of detection strategies will be needed to increase effectiveness and to adapt the methods to special environments. This paper presents a modular framework for SPIT prevention designed to be easily manageable and extensible. Additionally, the framework makes use of a two-stage architecture in order to exploit the knowledge coming both from the signaling and from the media flows while still allowing real-time delivery of the media content to the user. Following the experience gained with our prototype implementation we also present the most important design issues we encountered and suggest solutions to these issues. I. INTRODUCTION Everybody knows unsolicited bulk messages or SPAM emails which have become a serious problem not only for companies but also for private users. Many users already receive more SPAM than normal messages and some estimates put the percentage of SPAM at up to 80% of all email traffic. With the increasing deployment of Voice over IP there are concerns that it will also become a target of unsolicited advertisement which for the case of VoIP has been named SPIT (SPAM over Internet Telephony). SPAM already exists in the Public Switched Telephone Network (PSTN) but because of the relatively high cost for the caller the problem is limited. VoIP on the other hand is orders of magnitude cheaper (1) and therefore a much more attractive target. Among all the Voice over IP protocols available today we look at SIP (Session Initiation Protocol, (2)) in this paper because SIP solutions dominate today's enterprise market and will soon dominate the telecom business. The most obvious form of SPIT are marketing calls placed by a computer. Typically a list of SIP addresses is called one by one and every time a call is answered some marketing content is replayed. Creating a software application which can place such calls is a trivial exercise and any commodity hardware can be used to run it. The only additional requirement is an Internet connection which is able to handle a certain number of simultaneous VoIP calls. Current standardization activities (1) suggest that strongly authenticated identities (3) together with white lists (to allow known users to call) and a consent framework (4) (to solve the introduction problem) are the solution to SPIT calls. But even if these methods provide a good starting point, their effectiveness cannot be fully exploited because strongly authenticated identities cannot be enforced universally. If a callee receives a call without an authenticated identity he does not know whether the caller did not support them or whether it was intentionally not authenticated. Furthermore, there is no standardized technique to establish networks of trust in order to support a consent framework. Some research effort in this direction has also been pre- sented in (5), where a reputation-based SIP social networks approach is discussed. However, this approach is still in an early phase and requires more detailed investigation.

Sandra Tartarelli - One of the best experts on this subject based on the ideXlab platform.

  • on spam over Internet Telephony spit prevention
    IEEE Communications Magazine, 2008
    Co-Authors: J Quittek, Sandra Tartarelli, Saverio Niccolini, Roma Schlegel
    Abstract:

    Spam over IP Telephony (SPIT) is expected to become a serious problem in the near future. One of the main requirements for a SPIT prevention system is to avoid the involvement of the callee in the SPIT detection process, because this implies disturbing the callee each time an unclassified potential SPIT call arrives. Further requirements include adaptability to different deployment scenarios and flexibility to quickly react to new kinds of SPIT that bypass existing prevention systems. This article analyzes the requirements for SPIT prevention, provides a systematic classification of currently known SPIT prevention methods, and introduces a reference model for SPIT prevention systems. As an instance of the reference model, we designed and implemented an advanced SPIT prevention system, composed of methods that avoid unnecessary callee interaction and adaptive in order to be customized for different scenarios.

  • ise03 2 spam over Internet Telephony spit prevention framework
    Global Communications Conference, 2006
    Co-Authors: Roma Schlegel, Saverio Niccolini, Sandra Tartarelli, Marcus Unne
    Abstract:

    SPam over Internet Telephony (SPIT) is expected to become a threat inhibiting the delivery of voice services over the Internet in the near future both because of its technical and economical characteristics. Experiences with email SPAM and its analogies with SPIT suggest that SPIT will be difficult to detect with a single detection method. Moreover, personalized management of detection strategies will be needed to increase effectiveness and to adapt the methods to special environments. This paper presents a modular framework for SPIT prevention designed to be easily manageable and extensible. Additionally, the framework makes use of a two-stage architecture in order to exploit the knowledge coming both from the signaling and from the media flows while still allowing real-time delivery of the media content to the user. Following the experience gained with our prototype implementation we also present the most important design issues we encountered and suggest solutions to these issues.

  • spam over Internet Telephony spit prevention framework
    Global Communications Conference, 2006
    Co-Authors: Roma Schlegel, Saverio Niccolini, Sandra Tartarelli, Marcus Unne
    Abstract:

    SPam over Internet Telephony (SPIT) is expected to become a threat inhibiting the delivery of voice services over the Internet in the near future both because of its technical and economical characteristics. Experiences with email SPAM and its analogies with SPIT suggest that SPIT will be difficult to detect with a single detection method. Moreover, personalized management of detection strategies will be needed to increase effectiveness and to adapt the methods to special environments. This paper presents a modular framework for SPIT prevention designed to be easily manageable and extensible. Additionally, the framework makes use of a two-stage architecture in order to exploit the knowledge coming both from the signaling and from the media flows while still allowing real-time delivery of the media content to the user. Following the experience gained with our prototype implementation we also present the most important design issues we encountered and suggest solutions to these issues. I. INTRODUCTION Everybody knows unsolicited bulk messages or SPAM emails which have become a serious problem not only for companies but also for private users. Many users already receive more SPAM than normal messages and some estimates put the percentage of SPAM at up to 80% of all email traffic. With the increasing deployment of Voice over IP there are concerns that it will also become a target of unsolicited advertisement which for the case of VoIP has been named SPIT (SPAM over Internet Telephony). SPAM already exists in the Public Switched Telephone Network (PSTN) but because of the relatively high cost for the caller the problem is limited. VoIP on the other hand is orders of magnitude cheaper (1) and therefore a much more attractive target. Among all the Voice over IP protocols available today we look at SIP (Session Initiation Protocol, (2)) in this paper because SIP solutions dominate today's enterprise market and will soon dominate the telecom business. The most obvious form of SPIT are marketing calls placed by a computer. Typically a list of SIP addresses is called one by one and every time a call is answered some marketing content is replayed. Creating a software application which can place such calls is a trivial exercise and any commodity hardware can be used to run it. The only additional requirement is an Internet connection which is able to handle a certain number of simultaneous VoIP calls. Current standardization activities (1) suggest that strongly authenticated identities (3) together with white lists (to allow known users to call) and a consent framework (4) (to solve the introduction problem) are the solution to SPIT calls. But even if these methods provide a good starting point, their effectiveness cannot be fully exploited because strongly authenticated identities cannot be enforced universally. If a callee receives a call without an authenticated identity he does not know whether the caller did not support them or whether it was intentionally not authenticated. Furthermore, there is no standardized technique to establish networks of trust in order to support a consent framework. Some research effort in this direction has also been pre- sented in (5), where a reputation-based SIP social networks approach is discussed. However, this approach is still in an early phase and requires more detailed investigation.

Saverio Niccolini - One of the best experts on this subject based on the ideXlab platform.

  • on spam over Internet Telephony spit prevention
    IEEE Communications Magazine, 2008
    Co-Authors: J Quittek, Sandra Tartarelli, Saverio Niccolini, Roma Schlegel
    Abstract:

    Spam over IP Telephony (SPIT) is expected to become a serious problem in the near future. One of the main requirements for a SPIT prevention system is to avoid the involvement of the callee in the SPIT detection process, because this implies disturbing the callee each time an unclassified potential SPIT call arrives. Further requirements include adaptability to different deployment scenarios and flexibility to quickly react to new kinds of SPIT that bypass existing prevention systems. This article analyzes the requirements for SPIT prevention, provides a systematic classification of currently known SPIT prevention methods, and introduces a reference model for SPIT prevention systems. As an instance of the reference model, we designed and implemented an advanced SPIT prevention system, composed of methods that avoid unnecessary callee interaction and adaptive in order to be customized for different scenarios.

  • ise03 2 spam over Internet Telephony spit prevention framework
    Global Communications Conference, 2006
    Co-Authors: Roma Schlegel, Saverio Niccolini, Sandra Tartarelli, Marcus Unne
    Abstract:

    SPam over Internet Telephony (SPIT) is expected to become a threat inhibiting the delivery of voice services over the Internet in the near future both because of its technical and economical characteristics. Experiences with email SPAM and its analogies with SPIT suggest that SPIT will be difficult to detect with a single detection method. Moreover, personalized management of detection strategies will be needed to increase effectiveness and to adapt the methods to special environments. This paper presents a modular framework for SPIT prevention designed to be easily manageable and extensible. Additionally, the framework makes use of a two-stage architecture in order to exploit the knowledge coming both from the signaling and from the media flows while still allowing real-time delivery of the media content to the user. Following the experience gained with our prototype implementation we also present the most important design issues we encountered and suggest solutions to these issues.

  • spam over Internet Telephony spit prevention framework
    Global Communications Conference, 2006
    Co-Authors: Roma Schlegel, Saverio Niccolini, Sandra Tartarelli, Marcus Unne
    Abstract:

    SPam over Internet Telephony (SPIT) is expected to become a threat inhibiting the delivery of voice services over the Internet in the near future both because of its technical and economical characteristics. Experiences with email SPAM and its analogies with SPIT suggest that SPIT will be difficult to detect with a single detection method. Moreover, personalized management of detection strategies will be needed to increase effectiveness and to adapt the methods to special environments. This paper presents a modular framework for SPIT prevention designed to be easily manageable and extensible. Additionally, the framework makes use of a two-stage architecture in order to exploit the knowledge coming both from the signaling and from the media flows while still allowing real-time delivery of the media content to the user. Following the experience gained with our prototype implementation we also present the most important design issues we encountered and suggest solutions to these issues. I. INTRODUCTION Everybody knows unsolicited bulk messages or SPAM emails which have become a serious problem not only for companies but also for private users. Many users already receive more SPAM than normal messages and some estimates put the percentage of SPAM at up to 80% of all email traffic. With the increasing deployment of Voice over IP there are concerns that it will also become a target of unsolicited advertisement which for the case of VoIP has been named SPIT (SPAM over Internet Telephony). SPAM already exists in the Public Switched Telephone Network (PSTN) but because of the relatively high cost for the caller the problem is limited. VoIP on the other hand is orders of magnitude cheaper (1) and therefore a much more attractive target. Among all the Voice over IP protocols available today we look at SIP (Session Initiation Protocol, (2)) in this paper because SIP solutions dominate today's enterprise market and will soon dominate the telecom business. The most obvious form of SPIT are marketing calls placed by a computer. Typically a list of SIP addresses is called one by one and every time a call is answered some marketing content is replayed. Creating a software application which can place such calls is a trivial exercise and any commodity hardware can be used to run it. The only additional requirement is an Internet connection which is able to handle a certain number of simultaneous VoIP calls. Current standardization activities (1) suggest that strongly authenticated identities (3) together with white lists (to allow known users to call) and a consent framework (4) (to solve the introduction problem) are the solution to SPIT calls. But even if these methods provide a good starting point, their effectiveness cannot be fully exploited because strongly authenticated identities cannot be enforced universally. If a callee receives a call without an authenticated identity he does not know whether the caller did not support them or whether it was intentionally not authenticated. Furthermore, there is no standardized technique to establish networks of trust in order to support a consent framework. Some research effort in this direction has also been pre- sented in (5), where a reputation-based SIP social networks approach is discussed. However, this approach is still in an early phase and requires more detailed investigation.