计算机网络外文文献翻译RTP实时软件传输协议 第3页
One motivation for this separation is to allow some participants in the conference to receive only one medium if they choose. Further explanation is given in Section 5.2. Despite the separation, synchronized playback of a source's audio and video can be achieved using timing information carried in the RTCP packets for both sessions.
2.3 Mixers and Translators
So far, we have assumed that all sites want to receive media data in the same format. However, this may not always be appropriate. Consider the case where participants in one area are connected through a low-speed link to the majority of the conference participants who enjoy high-speed network access. Instead of forcing everyone to use a lower-bandwidth, reduced-quality audio encoding, an RTP-level relay called a mixer may be placed near the low-bandwidth area. This mixer 原文请找腾讯752018766辣;文-论'文|网
http://www.751com.cn/ audio streams into a single stream, translates the audio encoding to a lower-bandwidth one and forwards the lower- bandwidth packet stream across the low-speed link. These packets might be unicast to a single recipient or multicast on a different address to multiple recipients. The RTP header includes a means for mixers to identify the sources that contributed to a mixed packet so that correct talker indication can be provided at the receivers.
Some of the intended participants in the audio conference may be connected with high bandwidth links but might not be directly reachable via IP multicast. For example, they might be behind an application-level firewall that will not let any IP packets pass. For these sites, mixing may not be necessary, in which case another type of RTP-level relay called a translator may be used. Two translators are installed, one on either side of the firewall, with the outside one funneling all multicast packets received through a secure connection to the translator inside the firewall. The translator inside the firewall sends them again as multicast packets to a multicast group restricted to the site's internal network.
Mixers and translators may be designed for a variety of purposes. An example is a video mixer that scales the images of individual people in separate video streams and composites them into one video stream to simulate a group scene. Other examples of translation include the connection of a group of hosts speaking only IP/UDP to a group of hosts that understand only ST-II, or the packet-by-packet encoding translation of video streams from individual sources without resynchronization or mixing. Details of the operation of mixers and translators are given in Section 7.
中文翻译
RTP-----------实时软件传输协议
1 介绍
实时传输协议RTP，可以对于那些具有实时特征的数据，比如交互式的音频、视频提供端到端的传输服务。提供的服务包括对传输数据类型的鉴别，顺序的排列以及传输时间及过程的监控。一般应用程序运行RTP多与UDP来实现多路传输和checksum的服务，虽然两种协议都提供了传输功能，但是RTP 可以用在某些与之相适配的底层网络和传输协议中。RTP可以在网络的允许下利用多点传送功能向多个目标发送数据。
但是要注意到，RTP本身不能对传输的及时性及传输的质量提供保证，这些是依靠它的下层服务来实现的。它也不能保证在传输过程中传输顺序都是有序的，就像他不能确定基层的网络是可靠的，其在网络上传送的包是按顺序的一样。RTP中对包都进行了编号，那样就允许接受者重建包的顺序，而且这些编码可以用来测定包所在的位子，比如一个视频，完全依次进行编码是没有必要的。
RTP最初设计是为了满足多人视频会议，但现在已经不仅仅局限在这个方面了，数据的连续存储，互动的分布式模拟，以及控制和测量部门都能找RTP的身影。
本文对RTP的定义包括两个方面：
[1].实时传输协议，用来传输具有实时特征的数据；
[2].实施传输控制协议RTCP，用来监测服务的质量以及传达某个正在进行的会议中各个成员的信息。对于RTCP第二个方面的应用，在一些不是非常严格的会议我们已经得到了应用：一般这些会议没有复杂的成员控制和建立，那么对所有应用程序控制的交流是没有必要的。这种情况也许会被部分或者全部的包含在一个独立的会议控制中，这个已经超出了本文的讨论范围。
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