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 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     

The Session Initiation Protocol: Internet-centric signaling

The Session Initiation Protocol (SIP) provides advanced signaling and control functionality for a wide variety of multimedia services. SIP can efficiently and scalably locate resources based on a location-independent name and then negotiate session characteristics. It can find use in applications ranging from Internet telephony and conferencing to instant messaging, event notification, and the control of networked devices. We summarize the main protocol features and describe a range of extensions currently being discussed within the Internet Engineering Task Force     

Structuring Internet media streams with cueing protocols

We propose a new, media-independent protocol for including program timing, structure and identity information in Internet media streams. The protocol uses signaling messages called cues to indicate events whose timing is significant to receivers, such as the start or stop time of a media program. We describe the implementation and operation of a prototype Internet radio station which transmits program cues in audio broadcasts using a real-time transport protocol (RTP). A collection of simple yet powerful stream processing applications we implemented demonstrate how application creation is greatly eased when media streams are enriched with program cues.     

Evaluation of transport protocols for the session initiation protocol

SCTP is a newly developed transport protocol tailored for signaling transport. Whereas in theory SCTP is supposed to achieve a much better performance than TCP and UDP, at present there are no experimental results showing SCTP's real benefits. This article analyzes SCTP's strengths and weaknesses and provides simulation results. We implemented SIP on top of UDP, TCP, and SCTP in the network simulator and compared the three transport protocols under different network conditions.     

Real-time communication in packet-switched networks

The dramatically increased bandwidths and processing capabilities of future high-speed networks make possible many distributed real-time applications, such as sensor-based applications and multimedia services. Since these applications will have traffic characteristics and performance requirements that differ dramatically from those of current data-oriented applications, new communication network architectures, and protocols will be required. In this paper we discuss the performance requirements and traffic characteristics of various real-time applications, survey recent developments in the areas of network architecture and protocols for supporting real-time services, and develop frameworks in which these, and future, research efforts can be considered     

Providing emergency services in Internet telephony

Our primary focus is on emergency calling and notification. We describe the components of the existing emergency calling and notification systems and our proposed IP-based architectures, each of which uses the session initiation protocol (SIP) as the signaling framework. The architecture could increase speed, scalability, and functionality in communication services     

The Dynamic Community of Interest and Its Realization in ZODIAC

The ZODIAC project has been exploring a security first approach to networking based on a new idea, the dynamic community of interest, based on groups of users with a demonstrable need to know. ZODIAC uses the most challenging network setting (the mobile ad hoc network) as a target, since each node must incorporate functions of both hosts and routers. The realization of the DCoI is a work in progress, but initial implementation results have shown that DCoI concepts can be translated into working systems. The current system applies virtual machine containers, extensive use of cryptography and digital signatures, dispersity routing, DHT-based naming, and explicit rate control among other advanced techniques. Putting security to the forefront in the design has led to interesting consequences for naming, authorization, and connection setup. In particular, it has demanded a hierarchical structure for DCoIs that may initially appear somewhat alien to Internet users. Nonetheless, our implementation has illustrated that a highly available network that provides confidentiality and integrity can be constructed and made usable.     

Integrating Internet telephony services

Cost savings and the ease of developing and adding new services have motivated great interest in Internet telephony, which integrates services provided by the Internet with the public switched telephone network (PSTN). Internet telephony relies on several protocols, including the real-time transport protocol (RTP) for multimedia data transport and the session initiation protocol (SIP) or H.323 for establishing and controlling sessions. SIP can integrate with other Internet services, such as email, the Web, voice mail, instant messaging, conference calling, and multimedia collaboration. We have implemented a SIP-based software suite called the Columbia Internet extensible multimedia architecture (Cinema), which we installed and integrated with the existing private branch exchange (PBX) infrastructure in the computer science department at Columbia University. The Cinema environment provides interoperability with the PSTN, programmable Internet telephony services, and IP-based voice mail. It also integrates Web access and e-mail for unified messaging and supports multiparty multimedia conferencing. The setup lets us extend our PBX capacity and will eventually let us replace it while keeping our existing phone numbers. It also provides an environment in which we can easily add new services and features, including interoperation with existing multimedia tools, e-mail access from standard. telephones, network appliance control, and instant messaging support