Internet telephony: architecture and protocols - an IETF perspective

被引：35
|
作者：
Schulzrinne, H [1 ]
Rosenberg, J
机构：
[1] Columbia Univ, Dept Comp Sci, New York, NY 10027 USA
[2] AT&T Bell Labs, Lucent Technol, Murray Hill, NJ 07974 USA
来源：
COMPUTER NETWORKS-THE INTERNATIONAL JOURNAL OF COMPUTER AND TELECOMMUNICATIONS NETWORKING | 1999年 / 31卷 / 03期
关键词：
Internet telephony; Internet multimedia; SIP; RTSP; RTP;
D O I：
10.1016/S0169-7552(98)00265-7
中图分类号：
TP3 [计算技术、计算机技术];
学科分类号：
0812 ;
摘要：
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. (C) 1999 Published by Elsevier Science B.V. All rights reserved.
引用
收藏
页码：237 / 255
页数：19
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