组播技术及基于组播技术的视频会议模型设计探究
|
摘要
随着世界经济格局在世纪之交的大调整和新技术革命所带来的更大的挑战，网络经济，电子商务，特别是和Internet相关的经济领域，已经成为知识经济时代的主要格局。Internet成为加快和改变人类信息和知识流通的巨大推动力。信息高速公路的出现消除了地域，时间和肤色的差异，成为各种经济活动的操作平台。同时，随着网络经济的不断深入发展，人们对网络互连设备的性能、网络安全性、稳定性以及与各种网络传输和服务相关的软件的可靠性、安全性和有效性提出了越来越高的要求。
在Internet上，多媒体业务诸如：流媒体、视频会议和视频点播等，正在成为信息传送的重要组成部分。点对点传输的单播方式不能适应此业务的传输特性(单点发送多点接收)，这使得服务器必须为每一个接收者提供一个相同内容的IP报文拷贝，并在网络上重复地传输相同内容的报文，这必然占用大量网络资源。虽然IP广播允许一个主机把一个1P报文发送给同一个网络的所有主机，但是由于不一定是所有的主机都需要这些报文，这又可能浪费大量网络资源。正是在这种情况下组播技术(multicast)应运而生，它的出现解决了一个主机如何向特定的多个接收者发送消息的问题。
本文分四个部分。
第一部分介绍组播相关概念。组播是基于UDP数据报方式的，非面向连接的一种网络数据发送方式，组播是RTP／RTCP数据传输的基础。
第二部分介绍实时传输协议(RTP)／实时控制协议(RTCP)协议。这两个协议的内容是非常重要的，实现原理都是采用组播方式，我们在互连网上传输视频音频将使用这两个协议，而对于更高层次的协议框架H．323，SIP，在其底层传输视频音频通常也是采用RTP／RTCP协议。RTP灵活地支持对众多视音频编解码的负载格式，这一部分的最后给出了就如何将视频音频数据进行RTP包封装的一般原则。
第三部分就如何实现一个视频会议系统进行阐述和研究，包括H．323和SIP协议框架，以及他们的比较。同时提出了一个基于H．323协议的简化视频会议原型。我们认为，基于RTP实时数据传送的视频会议，视频电话系统是未来若干年的应用系统的耀眼亮点，值得高度重视。
第四部分实现部分。给出了包括视频会议的各个模块的关键技术和值得关注的要点难点，以及相关需要改进之处。
文章的最后给出了本文应用程序实现的主要技术指标，以及需要进一步完善的地方。
On the Internet,multimedia services such as Streaming Media,Video/Audio Conference,Voice Over IP and Video On Demand,become more and more important, end-to-end data transmission or unicast can't be competent for the task which have the characteristic of one-to-many transmission.if we use unicast,the server has to offer the packet with same context for each connection,which will consume too much bandwidth.then the broadcast is alternative selection for us to deliver data in the LAN,as we known,when broadcast technology used,all the PCs in the LAN will receive the broadcast packets regardless that you would like to or not. So we have to find the other solution for these conditions,the appearance of multicast give us a chance to solve the problem.multicast servers send the packets only to those terminals which are interested in the packets.
Our document is made with four sections.
Section one,we illuminate the conception of multicast.IP multicasting is the transmission of an IP datagram to a "host group",the datagram is not guaranteed to arrive intact at all members of the destination group or in the same order relative to other datagrams.
Section two,This section describes RTP(real-time transport protocol) and RTCP(control protocol). RTF provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services.RTCP monitors the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality.H.323 and SIP use the real-time transport protocol to transmit real-time data.at the end of this section, we provide the guidelines for writting of RTF Payload Format Specifications.
Section three,ITU-T RECOMMENDATION H.323 and the Session Initiation Protocol(SIP) will been introduced. We give a outlook to the difference of the two protocols.Then we dwell on the H.323 protocol.at the end of the section we give the simplified video/audio conference model based on H.323.
Section four,the implementation of the video/audio conference, we discuss the key technology around each functions of the model.
In the remainder of this paper,the improvement about the implementation will been discussed.
在Internet上，多媒体业务诸如：流媒体、视频会议和视频点播等，正在成为信息传送的重要组成部分。点对点传输的单播方式不能适应此业务的传输特性(单点发送多点接收)，这使得服务器必须为每一个接收者提供一个相同内容的IP报文拷贝，并在网络上重复地传输相同内容的报文，这必然占用大量网络资源。虽然IP广播允许一个主机把一个1P报文发送给同一个网络的所有主机，但是由于不一定是所有的主机都需要这些报文，这又可能浪费大量网络资源。正是在这种情况下组播技术(multicast)应运而生，它的出现解决了一个主机如何向特定的多个接收者发送消息的问题。
本文分四个部分。
第一部分介绍组播相关概念。组播是基于UDP数据报方式的，非面向连接的一种网络数据发送方式，组播是RTP／RTCP数据传输的基础。
第二部分介绍实时传输协议(RTP)／实时控制协议(RTCP)协议。这两个协议的内容是非常重要的，实现原理都是采用组播方式，我们在互连网上传输视频音频将使用这两个协议，而对于更高层次的协议框架H．323，SIP，在其底层传输视频音频通常也是采用RTP／RTCP协议。RTP灵活地支持对众多视音频编解码的负载格式，这一部分的最后给出了就如何将视频音频数据进行RTP包封装的一般原则。
第三部分就如何实现一个视频会议系统进行阐述和研究，包括H．323和SIP协议框架，以及他们的比较。同时提出了一个基于H．323协议的简化视频会议原型。我们认为，基于RTP实时数据传送的视频会议，视频电话系统是未来若干年的应用系统的耀眼亮点，值得高度重视。
第四部分实现部分。给出了包括视频会议的各个模块的关键技术和值得关注的要点难点，以及相关需要改进之处。
文章的最后给出了本文应用程序实现的主要技术指标，以及需要进一步完善的地方。
On the Internet,multimedia services such as Streaming Media,Video/Audio Conference,Voice Over IP and Video On Demand,become more and more important, end-to-end data transmission or unicast can't be competent for the task which have the characteristic of one-to-many transmission.if we use unicast,the server has to offer the packet with same context for each connection,which will consume too much bandwidth.then the broadcast is alternative selection for us to deliver data in the LAN,as we known,when broadcast technology used,all the PCs in the LAN will receive the broadcast packets regardless that you would like to or not. So we have to find the other solution for these conditions,the appearance of multicast give us a chance to solve the problem.multicast servers send the packets only to those terminals which are interested in the packets.
Our document is made with four sections.
Section one,we illuminate the conception of multicast.IP multicasting is the transmission of an IP datagram to a "host group",the datagram is not guaranteed to arrive intact at all members of the destination group or in the same order relative to other datagrams.
Section two,This section describes RTP(real-time transport protocol) and RTCP(control protocol). RTF provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services.RTCP monitors the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality.H.323 and SIP use the real-time transport protocol to transmit real-time data.at the end of this section, we provide the guidelines for writting of RTF Payload Format Specifications.
Section three,ITU-T RECOMMENDATION H.323 and the Session Initiation Protocol(SIP) will been introduced. We give a outlook to the difference of the two protocols.Then we dwell on the H.323 protocol.at the end of the section we give the simplified video/audio conference model based on H.323.
Section four,the implementation of the video/audio conference, we discuss the key technology around each functions of the model.
In the remainder of this paper,the improvement about the implementation will been discussed.
引文
(1) Microsoft MSDN Library-July 2002
(2) Microsoft DirectX 8.1 (C++)
(3) Gary R. Wright, W. Richard Stenvens, TCP/IP详解 卷1：协议，机械工业出版社，2002
(4) Gary R. Wright, W. Richard Stenvens, TCP/IP详解 卷2：实现，机械工业出版社，2002
(5) Gary R. Wright, W. Richard Stenvens, TCP/IP详解 卷3：TCP事务协议，HTTP，NNTP和UNIX域协议，机械工业出版社，2002
(6) Uyless Black," Advanced Internet Technologies" ,电子工业出版社，2001
(7) Uyless Black," VOIP:IP语音技术”，机械工业出版社，2000
(8) 岩延，郭江涛等，组播路由协议设计以及应用，人民邮电出版社，2002
(9) 张益贞，刘滔，Visual C++实现MPEG/JPEG编解码技术，人民邮电出版社，2002
(10) 张新宇，Windows声音应用程序开发指南
(11) RFC1112, Host Extensions for IP Multicasting
(12) RFC1889 RTP: A Transport Protocol for Real-Time Applications
(13) RFC1890 RTP Profile for Audio and Video Conferences with Minimal Control
(14) RFC2032 TP Payload Format for H. 261 Video Streams
(15) RFC2038 RTP Payload Format for MPEG1/MPEG2 Video
(16) RFC219ORTP Payload Format for H. 263 Video Streams
(17) RFC2236, Internet Group Management Protocol, Version 2
(18) RFC2432, Terminology for IP Muiticast Benchmarking
(19) RFC2543, SIP: Session Initiation Protocol
(20) RFC2588, IP Multicast and Firewalls
(21) RFC2736, Guidelines for Writers of RTP Payload Format Specifications
(22) RFC3170, IP Multicast Applications: Challenges and Solutions
(23) RFC3171, ANA Guidelines for IPv4 Multicast Address Assignments
(24) Draft H. 323v4 (Including Editorial Corrections-February 2001)
(25) RECOMMENDATION H. 245-VERSION 5, June 3, 1999
(26) ITU-T Recommendation H. 225.0
(27) 软交换与下一代网络，陈建亚，余浩 北京邮电大学出版社 2003
(2) Microsoft DirectX 8.1 (C++)
(3) Gary R. Wright, W. Richard Stenvens, TCP/IP详解 卷1：协议，机械工业出版社，2002
(4) Gary R. Wright, W. Richard Stenvens, TCP/IP详解 卷2：实现，机械工业出版社，2002
(5) Gary R. Wright, W. Richard Stenvens, TCP/IP详解 卷3：TCP事务协议，HTTP，NNTP和UNIX域协议，机械工业出版社，2002
(6) Uyless Black," Advanced Internet Technologies" ,电子工业出版社，2001
(7) Uyless Black," VOIP:IP语音技术”，机械工业出版社，2000
(8) 岩延，郭江涛等，组播路由协议设计以及应用，人民邮电出版社，2002
(9) 张益贞，刘滔，Visual C++实现MPEG/JPEG编解码技术，人民邮电出版社，2002
(10) 张新宇，Windows声音应用程序开发指南
(11) RFC1112, Host Extensions for IP Multicasting
(12) RFC1889 RTP: A Transport Protocol for Real-Time Applications
(13) RFC1890 RTP Profile for Audio and Video Conferences with Minimal Control
(14) RFC2032 TP Payload Format for H. 261 Video Streams
(15) RFC2038 RTP Payload Format for MPEG1/MPEG2 Video
(16) RFC219ORTP Payload Format for H. 263 Video Streams
(17) RFC2236, Internet Group Management Protocol, Version 2
(18) RFC2432, Terminology for IP Muiticast Benchmarking
(19) RFC2543, SIP: Session Initiation Protocol
(20) RFC2588, IP Multicast and Firewalls
(21) RFC2736, Guidelines for Writers of RTP Payload Format Specifications
(22) RFC3170, IP Multicast Applications: Challenges and Solutions
(23) RFC3171, ANA Guidelines for IPv4 Multicast Address Assignments
(24) Draft H. 323v4 (Including Editorial Corrections-February 2001)
(25) RECOMMENDATION H. 245-VERSION 5, June 3, 1999
(26) ITU-T Recommendation H. 225.0
(27) 软交换与下一代网络，陈建亚，余浩 北京邮电大学出版社 2003