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.

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An Analysis of the Skype Peer-to-Peer Internet Telephony Protocol

An intelligent gateway device provided at a premise (home or business) for providing and managing application services associated with use and support of a plurality of digital endpoint devices associated with the premises. The device includes a communications and processing infrastructure integrated with a peer and presence messaging based communications protocol for enabling communications between the device and an external support network and between the device and connected digital endpoint devices. A services framework at the gateway device implements the communications and processing infrastructure for enabling service management, service configuration, and authentication of user of services at the intelligent gateway. The framework provides a storage and execution environment for supporting and executing received service logic modules relating to use, management, and support of the digital endpoint devices. Thus, the gateway device provides a network-based services point of presence for a plurality of digital endpoint devices at the premises.

Multi-services application gateway and system employing the same

A plurality of computer nodes communicate using seemingly random Internet Protocol source and destination addresses. Data packets matching criteria defined by a moving window of valid addresses are accepted for further processing, while those that do not meet the criteria are quickly rejected. Improvements to the basic design include (1) a load balancer that distributes packets across different transmission paths according to transmission path quality; (2) a DNS proxy server that transparently creates a virtual private network in response to a domain name inquiry; (3) a large-to-small link bandwidth management feature that prevents denial-of-service attacks at system chokepoints; (4) a traffic limiter that regulates incoming packets by limiting the rate at which a transmitter can be synchronized with a receiver; and (5) a signaling synchronizer that allows a large number of nodes to communicate with a central node by partitioning the communication function between two separate entities.

Agile network protocol for secure communications with assured system availability

Supporting mobile Internet multimedia applications requires more than just the ability to maintain connectivity across subnet changes. We describe how the Session Initiation Protocol (SIP) can help provide terminal, personal, session and service mobility to applications ranging from Internet telephony to presence and instant messaging. We also briefly discuss application-layer mobility for streaming multimedia applications initiated by RTSP.

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Application-layer mobility using SIP

Methods and apparatus for delivery of packetized content (e.g., video, audio, data, etc.) over a content delivery network. In one embodiment, the content is packetized using an Internet Protocol (IP), and delivered by a service provider over both managed and unmanaged networks to subscribers of the provider, so as to provide delivery at any time, at any location, and via any designated user device. The delivered content may originate from the service provider, third-party content sources (e.g., networks or studios), the subscriber(s) themselves, or other sources including the Internet. Use of a common control and service functions within the network afford the ability to integrate or blend services together, thereby affording the service provider and subscriber new service and economic opportunities. Content delivery sessions may also be migrated from one device to another. A network-based user interface infrastructure, and gateway-based client-side architecture, are also disclosed.

Methods and apparatus for packetized content delivery over a content delivery network

The invention provides a telecommunication system that allows communication between a circuit-based wireless telephony network and a packet-based Internet telephony network without requiring access to the Public Switched Telephone Network. The system includes a circuit-based wireless telephony network providing wireless access to the system, a packet-based Internet telephony network providing Internet telephony access to the system, and a base station gateway controller for providing an interface between the wireless telephony network and the Internet telephony network. The invention further provides methods for facilitating communication between a first device, the first device being a packet-based Internet telephony network based device, and a second device, the second device being a circuit-based wireless telephony network based device, by facilitating connection of a call from the first device to the second device, without requiring access to the public switched telephone network.

Internet protocol based wireless call processing

This document defines the Call Processing Language (CPL), a language
to describe and control Internet telephony services. It is designed to
be implementable on either network servers or user agents. It is meant
to be simple, extensible, easily edited by graphical clients, and
independent of operating system or signalling protocol. It is suitable
for running on a server where users may not be allowed to execute
arbitrary programs, as it has no variables, loops, or ability to run
external programs. [STANDARDS-TRACK]

Call Processing Language (CPL): A Language for User Control of Internet Telephony Services

We describe the architecture and implementation of our Internet telephony system CINEMA (Columbia InterNet Extensible Multimedia Architecture, intended to replace the departmental PBX (telephone switch). It interworks with the traditional telephone networks via PSTN/IP gateways. It also serves as a corporate or campus infrastructure for existing and future services like web, email, video and streaming media. Initially intended for a few users, it will eventually replace the plain old telephones from our offices, due to the cost benefits and new services it offers. We also discuss common inter-operability problems between the PBX and the gateway. This paper is intended as a design document of the overall system. keywords: Internet telephony deployment; VoIP test-bed; advanced IP telephony; PSTN/IP interoperability; SIP

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CINEMA: Columbia InterNet Extensible Multimedia Architecture

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.

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Programming Internet telephony services

A large number of the services we wish to make possible for Internet telephony require fairly elaborate combinations of signalling operations, often in network devices, to complete. We want a simple and standardized way to create such services to make them easier to implement and deploy. This document describes an architectural framework for such a mechanism, which we call a call processing language. It also outlines requirements for such a language.

Jonathan Lennox

Papers

Internet protocol based wireless call processing

Call Processing Language (CPL): A Language for User Control of Internet Telephony Services

CINEMA: Columbia InterNet Extensible Multimedia Architecture

Programming Internet telephony services

Call Processing Language Framework and Requirements