国际分组交换网络上的多媒体电话会议

J. Crowcroft, P. Kirstein, D. Timm
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引用次数: 3

摘要

介绍了世界上第一个多媒体电话会议系统的当前配置和预计的未来工作。通信体系结构(作者指的是协议和支持网络管理系统)是基于DARPA互联网协议套件的,但有一定的增强以支持实时流量。编解码器运行视频的标准比特流编码,并由PC进行打包。然而,这不是一个足够灵活的协议来遍历Internet。然后,流协议(ST)网关运行两个进一步的协议来传输视频和语音:(1)分组视频协议(PVP)和(2)语音流量协议(VT)。这些都是针对视频和语音的特殊要求进行调整的。为了以合理一致的方式携带这些,网关然后运行st。系统的性能主要受到路径前几跳的带宽的限制。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
Multimedia teleconferencing over international packet switched networks
A description is presented of the current configuration and projected future work on the world's first multimedia teleconferencing system. The communications architecture (by which the authors refer to the protocols and supporting network management system) is based around the DARPA Internet Protocol suite, but with certain enhancements to support real time traffic. The CODEC runs a standard bitstream encoding of video, and this is packetized by the PC. However, this is not sufficiently flexible a protocol to traverse the Internet. The stream protocol (ST) gateway then runs two further protocols to carry the video and voice: (1) packet video protocol (PVP) and (2) voice traffic protocol (VT). Each of these is tuned to the special requirements of video and voice. To carry these in a reasonably consistent fashion, the gateways then run the ST. The performance of the system is limited mainly by the bandwidth of the first few hops of the path.<>
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