作者: Jonathan David Rosenberg , Henning Schulzrinne
DOI:
关键词: Session Initiation Protocol 、 Internet layer 、 Service discovery 、 Voice over IP 、 Telephony 、 Multicast 、 SIP trunking 、 Computer network 、 The Internet 、 Computer science
摘要: Internet telephony service is defined as the provision of real-time, interactive, multimedia telecommunications services between human users, using public Internet. The most difficult problem in providing to overcome increased jitter, delay, and loss (as compared circuit-switched networks) suffered by voice. Past work has separately investigated Forward Error Correction (FEC) playout buffer adaptation mechanisms resolve these problems. We show that must be considered jointly. propose simulate a number algorithms for integrating FEC into schemes, they are superior non-integrated algorithms. Receiving feedback about network transport quality essential supporting adaptive applications. examine issues surrounding scalability large scale multicast groups. present, analyze, class termed reconsideration, which support congestion controlled highly dynamic groups, then how memory requirements our can reduced. We consider signaling protocols call establishment, management, features, After an analysis existing protocols, we new protocol, Session Initiation Protocol (SIP), overcomes limitations protocols. describe implementation this protocol software, discuss applications have built with it. interconnection telephone network, focus on discovery gateways. subset broader wide area problem. reviewing resource (and finding them lacking applications), present scalable discovery, ideal gateways, amongst other resources. Finally, architecture telephony, provides features complex users. review architectures been presented literature. architecture, application component combines best aspects work. used provide several