Internet telephony service is defined as the provision of real-time, interactive, multimedia telecommunications services between human users, using the public Internet. The most difficult problem in providing Internet telephony is to overcome the increased jitter, delay, and loss (as compared to circuit-switched networks) suffered by voice. In the past, the method of Forward Error Correction (FEC) was researched with buffer adaptation mechanisms to resolve these problems. Now, in this thesis, these mechanisms are considered jointly. A number of algorithms for integrating FEC into playout buffer adaptation schemes are proposed and simulated, they are superior to non-integrated algorithms. Receiving feedback about network transport quality is essential for supporting adaptive applications. The thesis shows the examine result of the issues surrounding scalability of transport feedback in large scale multicast groups. It presents a class of algorithms termed reconsideration, which support congestion controlled feedback in highly dynamic groups, and then it shows how the memory requirements of the algorithms can be reduced. The thesis concerns in signaling protocols for providing call establishment, management, features, and applications. It proposes also a new protocol, the Session Initiation Protocol (SIP), which overcomes the limitations of existing protocols. It describes an implementation of this protocol in software, and applications. It involves interconnection with the telephone network, and focuses on the problem of discovery of telephony gateways. It shows that this is a subset of a broader wide area service discovery problem. Finally, the thesis considers the problem of a service architecture for Internet telephony, which provides features and complex applications to users. Then it shows how this architecture can be used to provide several complex applications. Tác giả: Jonathan Rosenberg Nhà xuất bản: COLUMBIA UNIVERSITY
