Network Working Group M. Westerlund
Request for Comments: 5117 Ericsson
Category: Informational S. Wenger
Nokia
January 2008
Network Working Group M. Westerlund
Request for Comments: 5117 Ericsson
Category: Informational S. Wenger
Nokia
January 2008
RTP Topologies
RTP拓扑
Status of This Memo
关于下段备忘
This memo provides information for the Internet community. It does not specify an Internet standard of any kind. Distribution of this memo is unlimited.
本备忘录为互联网社区提供信息。它没有规定任何类型的互联网标准。本备忘录的分发不受限制。
Abstract
摘要
This document discusses multi-endpoint topologies used in Real-time Transport Protocol (RTP)-based environments. In particular, centralized topologies commonly employed in the video conferencing industry are mapped to the RTP terminology.
本文档讨论基于实时传输协议（RTP）的环境中使用的多端点拓扑。特别是，视频会议行业中常用的集中式拓扑映射到RTP术语。
Table of Contents
目录
1. Introduction ....................................................2
2. Definitions .....................................................3
2.1. Glossary ...................................................3
2.2. Indicating Requirement Levels ..............................3
3. Topologies ......................................................3
3.1. Point to Point .............................................4
3.2. Point to Multipoint Using Multicast ........................5
3.3. Point to Multipoint Using the RFC 3550 Translator ..........6
3.4. Point to Multipoint Using the RFC 3550 Mixer Model .........9
3.5. Point to Multipoint Using Video Switching MCUs ............11
3.6. Point to Multipoint Using RTCP-Terminating MCU ............12
3.7. Non-Symmetric Mixer/Translators ...........................13
3.8. Combining Topologies ......................................14
4. Comparing Topologies ...........................................15
4.1. Topology Properties .......................................15
4.1.1. All to All Media Transmission ......................15
4.1.2. Transport or Media Interoperability ................16
4.1.3. Per Domain Bit-Rate Adaptation .....................16
4.1.4. Aggregation of Media ...............................16
4.1.5. View of All Session Participants ...................16
4.1.6. Loop Detection .....................................17
4.2. Comparison of Topologies ..................................17
5. Security Considerations ........................................17
6. Acknowledgements ...............................................19
7. References .....................................................19
7.1. Normative References ......................................19
7.2. Informative References ....................................20
1. Introduction ....................................................2
2. Definitions .....................................................3
2.1. Glossary ...................................................3
2.2. Indicating Requirement Levels ..............................3
3. Topologies ......................................................3
3.1. Point to Point .............................................4
3.2. Point to Multipoint Using Multicast ........................5
3.3. Point to Multipoint Using the RFC 3550 Translator ..........6
3.4. Point to Multipoint Using the RFC 3550 Mixer Model .........9
3.5. Point to Multipoint Using Video Switching MCUs ............11
3.6. Point to Multipoint Using RTCP-Terminating MCU ............12
3.7. Non-Symmetric Mixer/Translators ...........................13
3.8. Combining Topologies ......................................14
4. Comparing Topologies ...........................................15
4.1. Topology Properties .......................................15
4.1.1. All to All Media Transmission ......................15
4.1.2. Transport or Media Interoperability ................16
4.1.3. Per Domain Bit-Rate Adaptation .....................16
4.1.4. Aggregation of Media ...............................16
4.1.5. View of All Session Participants ...................16
4.1.6. Loop Detection .....................................17
4.2. Comparison of Topologies ..................................17
5. Security Considerations ........................................17
6. Acknowledgements ...............................................19
7. References .....................................................19
7.1. Normative References ......................................19
7.2. Informative References ....................................20
When working on the Codec Control Messages [CCM], considerable confusion was noticed in the community with respect to terms such as Multipoint Control Unit (MCU), Mixer, and Translator, and their usage in various topologies. This document tries to address this confusion by providing a common information basis for future discussion and specification work. It attempts to clarify and explain sections of the Real-time Transport Protocol (RTP) spec [RFC3550] in an informal way. It is not intended to update or change what is normatively specified within RFC 3550.
在处理编解码器控制消息[CCM]时，在社区中注意到关于多点控制单元（MCU）、混频器和转换器等术语及其在各种拓扑中的使用的相当混乱。本文档试图通过为将来的讨论和规范工作提供公共信息基础来解决这一困惑。它试图以非正式的方式澄清和解释实时传输协议（RTP）规范[RFC3550]的各个部分。其目的不是更新或更改RFC 3550中规范规定的内容。
When the Audio-Visual Profile with Feedback (AVPF) [RFC4585] was developed the main emphasis lay in the efficient support of point to point and small multipoint scenarios without centralized multipoint control. However, in practice, many small multipoint conferences operate utilizing devices known as Multipoint Control Units (MCUs). MCUs may implement Mixer or Translator (in RTP [RFC3550] terminology)
当开发带反馈的视听配置文件（AVPF）[RFC4585]时，主要重点在于在没有集中多点控制的情况下有效支持点对点和小型多点场景。然而，在实践中，许多小型多点会议使用称为多点控制单元（MCU）的设备进行操作。MCU可以实现混频器或转换器（用RTP[RFC3550]术语）
functionality and signalling support. They may also contain additional application functionality. This document focuses on the media transport aspects of the MCU that can be realized using RTP, as discussed below. Further considered are the properties of Mixers and Translators, and how some types of deployed MCUs deviate from these properties.
功能和信号支持。它们还可能包含其他应用程序功能。本文档重点介绍可以使用RTP实现的MCU的媒体传输方面，如下所述。进一步考虑混频器和转换器的属性，以及某些类型的已部署MCU如何偏离这些属性。
ASM - Any Source Multicast AVPF - The Extended RTP Profile for RTCP-based Feedback CSRC - Contributing Source Link - The data transport to the next IP hop MCU - Multipoint Control Unit Path - The concatenation of multiple links, resulting in an end-to-end data transfer. PtM - Point to Multipoint PtP - Point to Point SSM - Source-Specific Multicast SSRC - Synchronization Source
ASM-任意源多播AVPF-基于RTCP的反馈CSC的扩展RTP配置文件-贡献源链路-到下一个IP跃点MCU的数据传输-多点控制单元路径-多个链路的串联，导致端到端数据传输。PtM-点对多点PtP-点对点SSM-源特定多播SSRC-同步源
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [RFC2119].
本文件中的关键词“必须”、“不得”、“要求”、“应”、“不应”、“应”、“不应”、“建议”、“可”和“可选”应按照RFC 2119[RFC2119]中所述进行解释。
The RFC 2119 language is used in this document to highlight those important requirements and/or resulting solutions that are necessary to address the issues raised in this document.
本文件使用RFC 2119语言强调了解决本文件中提出的问题所需的重要要求和/或最终解决方案。
This subsection defines several basic topologies that are relevant for codec control. The first four relate to the RTP system model utilizing multicast and/or unicast, as envisioned in RFC 3550. The last two topologies, in contrast, describe the deployed system models as used in many H.323 [H323] video conferences, where both the media streams and the RTP Control Protocol (RTCP) control traffic terminate at the MCU. In these two cases, the media sender does not receive the (unmodified or Translator-modified) Receiver Reports from all sources (which it needs to interpret based on Synchronization Source (SSRC) values) and therefore has no full information about all the endpoint's situation as reported in RTCP Receiver Reports (RRs). More topologies can be constructed by combining any of the models; see Section 3.8.
本小节定义了与编解码器控制相关的几个基本拓扑。前四个与使用多播和/或单播的RTP系统模型有关，如RFC3550中所设想的。相比之下，后两种拓扑描述了许多H.323[H323]视频会议中使用的部署系统模型，其中媒体流和RTP控制协议（RTCP）控制流量都在MCU处终止。在这两种情况下，媒体发送方不会从所有来源接收（未修改或翻译器修改的）接收方报告（需要根据同步源（SSRC）值进行解释），因此没有RTCP接收方报告（RRs）中报告的所有端点情况的完整信息。通过组合任何一个模型，可以构建更多的拓扑结构；见第3.8节。
The topologies may be referenced in other documents by a shortcut name, indicated by the prefix "Topo-".
在其他文档中，拓扑可通过前缀“Topo-”指示的快捷方式名称引用。
For each of the RTP-defined topologies, we discuss how RTP, RTCP, and the carried media are handled. With respect to RTCP, we also introduce the handling of RTCP feedback messages as defined in [RFC4585] and [CCM]. Any important differences between the two will be illuminated in the discussion.
对于每个RTP定义的拓扑，我们将讨论如何处理RTP、RTCP和承载介质。关于RTCP，我们还介绍了[RFC4585]和[CCM]中定义的RTCP反馈消息的处理。这两者之间的任何重要区别都将在讨论中阐明。
Shortcut name: Topo-Point-to-Point
快捷方式名称：地形点到点
The Point to Point (PtP) topology (Figure 1) consists of two endpoints, communicating using unicast. Both RTP and RTCP traffic are conveyed endpoint-to-endpoint, using unicast traffic only (even if, in exotic cases, this unicast traffic happens to be conveyed over an IP-multicast address).
点到点（PtP）拓扑（图1）由两个端点组成，使用单播进行通信。RTP和RTCP通信量都是在端点之间传输的，仅使用单播通信量（即使在特殊情况下，该单播通信量恰好通过IP多播地址传输）。
+---+ +---+
| A |<------->| B |
+---+ +---+
+---+ +---+
| A |<------->| B |
+---+ +---+
Figure 1 - Point to Point
图1-点对点
The main property of this topology is that A sends to B, and only B, while B sends to A, and only A. This avoids all complexities of handling multiple endpoints and combining the requirements from them. Note that an endpoint can still use multiple RTP Synchronization Sources (SSRCs) in an RTP session.
此拓扑的主要特性是A发送到B，且仅发送到B，而B发送到A，且仅发送到A。这避免了处理多个端点和组合它们的需求的所有复杂性。请注意，端点仍可在RTP会话中使用多个RTP同步源（SSRC）。
RTCP feedback messages for the indicated SSRCs are communicated directly between the endpoints. Therefore, this topology poses minimal (if any) issues for any feedback messages.
指示SSRC的RTCP反馈消息在端点之间直接通信。因此，这种拓扑结构对任何反馈消息造成的问题最小（如果有的话）。
Shortcut name: Topo-Multicast
快捷方式名称：拓扑多播
+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ Cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
+-----+
+---+ / \ +---+
| A |----/ \---| B |
+---+ / Multi- \ +---+
+ Cast +
+---+ \ Network / +---+
| C |----\ /---| D |
+---+ \ / +---+
+-----+
Figure 2 - Point to Multipoint Using Multicast
图2-使用多播的点对多点
Point to Multipoint (PtM) is defined here as using a multicast topology as a transmission model, in which traffic from any participant reaches all the other participants, except for cases such as:
点对多点（PtM）在这里定义为使用多播拓扑作为传输模型，其中来自任何参与者的流量到达所有其他参与者，但以下情况除外：
o packet loss, or
o 数据包丢失，或
o when a participant does not wish to receive the traffic for a specific multicast group and therefore has not subscribed to the IP-multicast group in question. This is for the cases where a multi-media session is distributed using two or more multicast groups.
o 当参与者不希望接收特定多播组的通信量，因此没有订阅所讨论的IP多播组时。这适用于使用两个或多个多播组分发多媒体会话的情况。
In the above context, "traffic" encompasses both RTP and RTCP traffic. The number of participants can vary between one and many, as RTP and RTCP scale to very large multicast groups (the theoretical limit of the number of participants in a single RTP session is approximately two billion). The above can be realized using Any Source Multicast (ASM). Source-Specific Multicast (SSM) may be also be used with RTP. However, then only the designated source may reach all receivers. Please review [RTCP-SSM] for how RTCP can be made to work in combination with SSM.
在上述上下文中，“流量”包括RTP和RTCP流量。随着RTP和RTCP扩展到非常大的多播组，参与者的数量可以在一个或多个之间变化（单个RTP会话中参与者数量的理论限制约为20亿）。可以使用任何源多播（ASM）实现上述功能。源特定多播（SSM）也可以与RTP一起使用。然而，则只有指定的源可以到达所有接收机。请查看[RTCP-SSM]，了解如何使RTCP与SSM结合使用。
This document is primarily interested in that subset of multicast sessions wherein the number of participants in the multicast group is so low that it allows the participants to use early or immediate feedback, as defined in AVPF [RFC4585]. This document refers to those groups as "small multicast groups".
本文档主要关注多播会话的子集，其中多播组中的参与者数量非常少，从而允许参与者使用AVPF[RFC4585]中定义的早期或即时反馈。本文档将这些组称为“小型多播组”。
RTCP feedback messages in multicast will, like media, reach everyone (subject to packet losses and multicast group subscription). Therefore, the feedback suppression mechanism discussed in [RFC4585]
多播中的RTCP反馈消息将像媒体一样到达每个人（取决于数据包丢失和多播组订阅）。因此，[RFC4585]中讨论的反馈抑制机制
is required. Each individual node needs to process every feedback message it receives to determine if it is affected or if the feedback message applies only to some other participant.
是必需的。每个节点都需要处理它接收到的每个反馈消息，以确定它是否受到影响，或者反馈消息是否仅适用于某些其他参与者。
Shortcut name: Topo-Translator
快捷方式名称：拓扑转换器
Two main categories of Translators can be distinguished:
翻译人员可分为两大类：
Transport Translators (Topo-Trn-Translator) do not modify the media stream itself, but are concerned with transport parameters. Transport parameters, in the sense of this section, comprise the transport addresses (to bridge different domains) and the media packetization to allow other transport protocols to be interconnected to a session (in gateways). Of the transport Translators, this memo is primarily interested in those that use RTP on both sides, and this is assumed henceforth. Translators that bridge between different protocol worlds need to be concerned about the mapping of the SSRC/CSRC (Contributing Source) concept to the non-RTP protocol. When designing a Translator to a non-RTP-based media transport, one crucial factor lies in how to handle different sources and their identities. This problem space is not discussed henceforth.
传输转换器（Topo Trn Translator）不修改媒体流本身，但与传输参数有关。在本节中，传输参数包括传输地址（桥接不同域）和媒体分组，以允许其他传输协议互连到会话（在网关中）。对于运输翻译人员，本备忘录主要关注双方都使用RTP的翻译人员，并假设从现在开始。连接不同协议世界的翻译人员需要关注SSRC/CSC（贡献源）概念到非RTP协议的映射。在设计非RTP媒体传输的翻译器时，一个关键因素在于如何处理不同的源及其身份。以后不再讨论这个问题。
Media Translators (Topo-Media-Translator), in contrast, modify the media stream itself. This process is commonly known as transcoding. The modification of the media stream can be as small as removing parts of the stream, and it can go all the way to a full transcoding (down to the sample level or equivalent) utilizing a different media codec. Media Translators are commonly used to connect entities without a common interoperability point.
相反，媒体转换器（Topo Media Translator）会修改媒体流本身。这个过程通常被称为代码转换。对媒体流的修改可以小到删除部分流，并且可以一直使用不同的媒体编解码器进行完整的转码（直至样本级别或等效级别）。媒体转换器通常用于连接没有公共互操作点的实体。
Stand-alone Media Translators are rare. Most commonly, a combination of Transport and Media Translators are used to translate both the media stream and the transport aspects of a stream between two transport domains (or clouds).
独立的媒体翻译人员很少。最常见的是，传输和媒体转换器的组合用于在两个传输域（或云）之间转换媒体流和流的传输方面。
Both Translator types share common attributes that separate them from Mixers. For each media stream that the Translator receives, it generates an individual stream in the other domain. A Translator always keeps the SSRC for a stream across the translation, where a Mixer can select a media stream, or send them out mixed, always under its own SSRC, using the CSRC field to indicate the source(s) of the content.
这两种转换器类型共享将它们与混合器分开的公共属性。对于转换器接收到的每个媒体流，它在另一个域中生成一个单独的流。翻译人员始终在整个翻译过程中为流保留SSRC，混音器可以选择媒体流，或将其混合发送，始终在其自己的SSRC下，使用CSRC字段指示内容的来源。
The RTCP translation process can be trivial, for example, when Transport Translators just need to adjust IP addresses, or they can be quite complex as in the case of media Translators. See Section 7.2 of [RFC3550].
RTCP转换过程可能很简单，例如，当传输转换器只需要调整IP地址时，或者它们可能非常复杂，就像媒体转换器一样。见[RFC3550]第7.2节。
+-----+
+---+ / \ +------------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Translator |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +------------+ +---+
+-----+
+-----+
+---+ / \ +------------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Translator |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +------------+ +---+
+-----+
Figure 3 - Point to Multipoint Using a Translator
图3-使用转换器的点对多点
Figure 3 depicts an example of a Transport Translator performing at least IP address translation. It allows the (non-multicast-capable) participants B and D to take part in a multicast session by having the Translator forward their unicast traffic to the multicast addresses in use, and vice versa. It must also forward B's traffic to D, and vice versa, to provide each of B and D with a complete view of the session.
图3描述了至少执行IP地址转换的传输转换器的示例。它允许（不支持多播的）参与者B和D通过让转换器将其单播通信转发到正在使用的多播地址来参与多播会话，反之亦然。它还必须将B的流量转发给D，反之亦然，以便为B和D中的每一个提供会话的完整视图。
If B were behind a limited network path, the Translator may perform media transcoding to allow the traffic received from the other participants to reach B without overloading the path.
如果B在有限的网络路径后面，则翻译器可以执行媒体转码以允许从其他参与者接收的通信量到达B而不会使路径过载。
When, in the example depicted in Figure 3, the Translator acts only as a Transport Translator, then the RTCP traffic can simply be forwarded, similar to the media traffic. However, when media translation occurs, the Translator's task becomes substantially more complex, even with respect to the RTCP traffic. In this case, the Translator needs to rewrite B's RTCP Receiver Report before forwarding them to D and the multicast network. The rewriting is needed as the stream received by B is not the same stream as the other participants receive. For example, the number of packets transmitted to B may be lower than what D receives, due to the different media format. Therefore, if the Receiver Reports were forwarded without changes, the extended highest sequence number would indicate that B were substantially behind in reception, while it most likely it would not be. Therefore, the Translator must translate that number to a corresponding sequence number for the stream the Translator received. Similar arguments can be made for most other fields in the RTCP Receiver Reports.
在图3所示的示例中，当转换器仅充当传输转换器时，可以简单地转发RTCP通信量，类似于媒体通信量。然而，当发生媒体翻译时，翻译器的任务变得相当复杂，甚至与RTCP通信量有关。在这种情况下，转换器需要重写B的RTCP接收器报告，然后再将其转发给D和多播网络。由于B接收的流与其他参与者接收的流不同，因此需要重写。例如，由于不同的媒体格式，发送到B的分组数可能低于D接收的分组数。因此，如果接收器报告在没有改变的情况下被转发，则扩展的最高序列号将指示B在接收方面实质上落后，而它很可能不会落后。因此，转换器必须将该编号转换为转换器接收到的流的相应序列号。RTCP接收方报告中的大多数其他字段也可以使用类似的参数。
As specified in Section 7.1 of [RFC3550], the SSRC space is common for all participants in the session, independent of on which side they are of the Translator. Therefore, it is the responsibility of the participants to run SSRC collision detection, and the SSRC is a field the Translator should not change.
如[RFC3550]第7.1节所述，SSRC空间对于会话中的所有参与者来说都是通用的，与他们是翻译人员的哪一方无关。因此，运行SSRC冲突检测是参与者的责任，而SSRC是翻译人员不应更改的字段。
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Translator |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 4 - RTP Translator (Relay) with Only Unicast Paths
图4-仅具有单播路径的RTP转换器（中继）
Another Translator scenario is depicted in Figure 4. Herein, the Translator connects multiple users of a conference through unicast. This can be implemented using a very simple transport Translator, which in this document is called a relay. The relay forwards all traffic it receives, both RTP and RTCP, to all other participants. In doing so, a multicast network is emulated without relying on a multicast-capable network infrastructure.
另一个翻译器场景如图4所示。在此，翻译器通过单播连接会议的多个用户。这可以使用一个非常简单的传输转换器来实现，在本文中称之为中继。中继将接收到的所有流量（RTP和RTCP）转发给所有其他参与者。在这样做的过程中，模拟多播网络而不依赖于支持多播的网络基础设施。
A Translator normally does not use an SSRC of its own, and is not visible as an active participant in the session. One exception can be conceived when a Translator acts as a quality monitor that sends RTCP reports and therefore is required to have an SSRC. Another example is the case when a Translator is prepared to use RTCP feedback messages. This may, for example, occur when it suffers packet loss of important video packets and wants to trigger repair by the media sender, by sending feedback messages. To be able to do this it needs to have a unique SSRC.
翻译人员通常不使用自己的SSRC，也不作为会话的积极参与者。当翻译人员充当发送RTCP报告的质量监视器，因此需要SSRC时，可以设想一个例外情况。另一个例子是翻译人员准备使用RTCP反馈消息的情况。例如，当其遭受重要视频分组的分组丢失并且希望通过发送反馈消息触发媒体发送者的修复时，这可能发生。要做到这一点，它需要有一个独特的SSRC。
A media Translator may in some cases act on behalf of the "real" source and respond to RTCP feedback messages. This may occur, for example, when a receiver requests a bandwidth reduction, and the media Translator has not detected any congestion or other reasons for bandwidth reduction between the media source and itself. In that case, it is sensible that the media Translator reacts to the codec control messages itself, for example, by transcoding to a lower media rate. If it were not reacting, the media quality in the media sender's domain may suffer, as a result of the media sender adjusting its media rate (and quality) according to the needs of the slow past-Translator endpoint, at the expense of the rate and quality of all other session participants.
在某些情况下，媒体翻译人员可能代表“真实”来源行事，并对RTCP反馈信息作出回应。例如，当接收器请求带宽减少，并且媒体转换器没有检测到媒体源和其自身之间带宽减少的任何拥塞或其他原因时，这可能发生。在这种情况下，媒体翻译器对编解码器控制消息本身作出反应是明智的，例如，通过转码到较低的媒体速率。如果没有反应，媒体发送者的域中的媒体质量可能会受到影响，这是因为媒体发送者根据缓慢过去的翻译器端点的需要调整其媒体速率（和质量），以牺牲所有其他会话参与者的速率和质量为代价。
In general, a Translator implementation should consider which RTCP feedback messages or codec-control messages it needs to understand in relation to the functionality of the Translator itself. This is completely in line with the requirement to also translate RTCP messages between the domains.
一般来说，翻译器实现应该考虑哪一个RTCP反馈消息或编解码器控制消息，它需要理解与翻译本身的功能有关。这完全符合在域之间转换RTCP消息的要求。
Shortcut name: Topo-Mixer
快捷方式名称：Topo Mixer
A Mixer is a middlebox that aggregates multiple RTP streams, which are part of a session, by mixing the media data and generating a new RTP stream. One common application for a Mixer is to allow a participant to receive a session with a reduced amount of resources.
混合器是一种中间盒，通过混合媒体数据并生成新的RTP流来聚合作为会话一部分的多个RTP流。混合器的一个常见应用是允许参与者以较少的资源量接收会话。
+-----+
+---+ / \ +-----------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Mixer |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +-----------+ +---+
+-----+
+-----+
+---+ / \ +-----------+ +---+
| A |<---/ \ | |<---->| B |
+---+ / Multi- \ | | +---+
+ Cast +->| Mixer |
+---+ \ Network / | | +---+
| C |<---\ / | |<---->| D |
+---+ \ / +-----------+ +---+
+-----+
Figure 5 - Point to Multipoint Using the RFC 3550 Mixer Model
图5-使用RFC 3550混频器模型的点对多点
A Mixer can be viewed as a device terminating the media streams received from other session participants. Using the media data from the received media streams, a Mixer generates a media stream that is sent to the session participant.
混音器可以被视为终止从其他会话参与者接收的媒体流的设备。混合器使用来自所接收的媒体流的媒体数据生成发送给会话参与者的媒体流。
The content that the Mixer provides is the mixed aggregate of what the Mixer receives over the PtP or PtM paths, which are part of the same conference session.
混合器提供的内容是混合器通过PtP或PtM路径接收的内容的混合聚合，PtP或PtM路径是同一会议会话的一部分。
The Mixer is the content source, as it mixes the content (often in the uncompressed domain) and then encodes it for transmission to a participant. The CSRC Count (CC) and CSRC fields in the RTP header are used to indicate the contributors of to the newly generated stream. The SSRCs of the to-be-mixed streams on the Mixer input appear as the CSRCs at the Mixer output. That output stream uses a unique SSRC that identifies the Mixer's stream. The CSRC are forwarded between the two domains to allow for loop detection and identification of sources that are part of the global session. Note that Section 7.1 of RFC 3550 requires the SSRC space to be shared between domains for these reasons.
混合器是内容源，因为它混合内容（通常在未压缩域中），然后对其进行编码以传输给参与者。RTP头中的CSC计数（CC）和CSC字段用于指示新生成流的参与者。混合器输入端待混合流的SSRC在混合器输出端显示为CSRC。该输出流使用唯一的SSRC来标识混合器的流。CSC在两个域之间转发，以允许循环检测和识别作为全局会话一部分的源。请注意，由于这些原因，RFC 3550第7.1节要求在域之间共享SSRC空间。
The Mixer is responsible for generating RTCP packets in accordance with its role. It is a receiver and should therefore send reception reports for the media streams it receives. In its role as a media sender, it should also generate Sender Reports for those media streams sent. As specified in Section 7.3 of RFC 3550, a Mixer must not forward RTCP unaltered between the two domains.
混音器负责根据其角色生成RTCP数据包。它是一个接收器，因此应该发送接收媒体流的接收报告。作为媒体发送者，它还应该为发送的媒体流生成发送者报告。按照RFC 3550第7.3节的规定，混音器不得在两个域之间转发未更改的RTCP。
The Mixer depicted in Figure 5 is involved in three domains that need to be separated: the multicast network, participant B, and participant D. The Mixer produces different mixed streams to B and D, as the one to B may contain content received from D, and vice versa. However, the Mixer only needs one SSRC in each domain that is the receiving entity and transmitter of mixed content.
图5所示的混频器涉及三个需要分离的域：多播网络、参与者B和参与者D。混频器产生到B和D的不同混合流，因为到B的混频器可能包含从D接收的内容，反之亦然。然而，混音器在作为混合内容的接收实体和发送者的每个域中只需要一个SSRC。
In the multicast domain, a Mixer still needs to provide a mixed view of the other domains. This makes the Mixer simpler to implement and avoids any issues with advanced RTCP handling or loop detection, which would be problematic if the Mixer were providing non-symmetric behavior. Please see Section 3.7 for more discussion on this topic.
在多播域中，混合器仍然需要提供其他域的混合视图。这使得混频器的实现更加简单，并避免了高级RTCP处理或循环检测的任何问题，如果混频器提供非对称行为，则会出现问题。有关此主题的更多讨论，请参见第3.7节。
A Mixer is responsible for receiving RTCP feedback messages and handling them appropriately. The definition of "appropriate" depends on the message itself and the context. In some cases, the reception of a codec-control message may result in the generation and transmission of RTCP feedback messages by the Mixer to the participants in the other domain. In other cases, a message is handled by the Mixer itself and therefore not forwarded to any other domain.
混音器负责接收RTCP反馈消息并对其进行适当处理。“适当”的定义取决于信息本身和上下文。在某些情况下，编解码器控制消息的接收可能导致混合器生成RTCP反馈消息并将其发送给另一域中的参与者。在其他情况下，消息由混合器本身处理，因此不会转发到任何其他域。
When replacing the multicast network in Figure 5 (to the left of the Mixer) with individual unicast paths as depicted in Figure 6, the Mixer model is very similar to the one discussed in Section 3.6 below. Please see the discussion in Section 3.6 about the differences between these two models.
当将图5（混合器左侧）中的多播网络替换为图6所示的单个单播路径时，混合器模型与下面第3.6节中讨论的模型非常相似。请参见第3.6节中关于这两种模型之间差异的讨论。
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
+---+ +------------+ +---+
| A |<---->| |<---->| B |
+---+ | | +---+
| Mixer |
+---+ | | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 6 - RTP Mixer with Only Unicast Paths
图6-仅具有单播路径的RTP混频器
Shortcut name: Topo-Video-switch-MCU
快捷方式名称：Topo视频开关MCU
+---+ +------------+ +---+
| A |------| Multipoint |------| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |------| |------| D |
+---+ +------------+ +---+
+---+ +------------+ +---+
| A |------| Multipoint |------| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |------| |------| D |
+---+ +------------+ +---+
Figure 7 - Point to Multipoint Using a Video Switching MCU
图7-使用视频切换MCU的点对多点
This PtM topology is still deployed today, although the RTCP-terminating MCUs, as discussed in the next section, are perhaps more common. This topology, as well as the following one, reflect today's lack of wide availability of IP multicast technologies, as well as the simplicity of content switching when compared to content mixing. The technology is commonly implemented in what is known as "Video Switching MCUs".
尽管RTCP端接MCU（如下一节所述）可能更为常见，但这种PtM拓扑目前仍在部署中。这种拓扑以及下面的拓扑反映了当今IP多播技术缺乏广泛可用性，以及与内容混合相比内容切换的简单性。该技术通常在所谓的“视频交换MCU”中实现。
A video switching MCU forwards to a participant a single media stream, selected from the available streams. The criteria for selection are often based on voice activity in the audio-visual conference, but other conference management mechanisms (like presentation mode or explicit floor control) are known to exist as well.
视频交换MCU将从可用流中选择的单个媒体流转发给参与者。选择标准通常基于视听会议中的语音活动，但也存在其他会议管理机制（如演示模式或显式楼层控制）。
The video switching MCU may also perform media translation to modify the content in bit-rate, encoding, or resolution. However, it still may indicate the original sender of the content through the SSRC. In this case, the values of the CC and CSRC fields are retained.
视频切换MCU还可以执行媒体翻译以在比特率、编码或分辨率方面修改内容。但是，它仍然可以通过SSRC指示内容的原始发件人。在这种情况下，保留CC和CSC字段的值。
If not terminating RTP, the RTCP Sender Reports are forwarded for the currently selected sender. All RTCP Receiver Reports are freely forwarded between the participants. In addition, the MCU may also originate RTCP control traffic in order to control the session and/or report on status from its viewpoint.
如果未终止RTP，则为当前选定的发送方转发RTCP发送方报告。所有RTCP接收方报告在参与者之间自由转发。此外，MCU还可以发起RTCP控制通信量，以便从其角度控制会话和/或报告状态。
The video switching MCU has most of the attributes of a Translator. However, its stream selection is a mixing behavior. This behavior has some RTP and RTCP issues associated with it. The suppression of all but one media stream results in most participants seeing only a subset of the sent media streams at any given time, often a single stream per conference. Therefore, RTCP Receiver Reports only report on these streams. Consequently, the media senders that are not currently forwarded receive a view of the session that indicates
视频切换MCU具有翻译器的大部分属性。然而，其流选择是一种混合行为。此行为有一些与之相关的RTP和RTCP问题。除一个媒体流外，对所有媒体流的抑制导致大多数参与者在任何给定时间仅看到发送的媒体流的子集，通常每个会议只有一个流。因此，RTCP接收器只报告这些流。因此，当前未被转发的媒体发送方接收到一个会话视图，该视图指示
their media streams disappear somewhere en route. This makes the use of RTCP for congestion control, or any type of quality reporting, very problematic.
他们的媒体流在途中的某个地方消失了。这使得使用RTCP进行拥塞控制或任何类型的质量报告都非常困难。
To avoid the aforementioned issues, the MCU needs to implement two features. First, it needs to act as a Mixer (see Section 3.4) and forward the selected media stream under its own SSRC and with the appropriate CSRC values. Second, the MCU needs to modify the RTCP RRs it forwards between the domains. As a result, it is RECOMMENDED that one implement a centralized video switching conference using a Mixer according to RFC 3550, instead of the shortcut implementation described here.
为了避免上述问题，MCU需要实现两个功能。首先，它需要充当混合器（参见第3.4节），并在其自己的SSRC下转发选定的媒体流，并使用适当的CSRC值。其次，MCU需要修改它在域之间转发的RTCP RRs。因此，建议根据RFC 3550使用混音器实现集中式视频交换会议，而不是此处描述的快捷方式实现。
Shortcut name: Topo-RTCP-terminating-MCU
快捷方式名称：Topo RTCP终止MCU
+---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
+---+ +------------+ +---+
| A |<---->| Multipoint |<---->| B |
+---+ | Control | +---+
| Unit |
+---+ | (MCU) | +---+
| C |<---->| |<---->| D |
+---+ +------------+ +---+
Figure 8 - Point to Multipoint Using Content Modifying MCUs
图8-使用内容修改MCU的点对多点
In this PtM scenario, each participant runs an RTP point-to-point session between itself and the MCU. This is a very commonly deployed topology in multipoint video conferencing. The content that the MCU provides to each participant is either:
在这个PtM场景中，每个参与者在自己和MCU之间运行一个RTP点对点会话。这是多点视频会议中非常常见的部署拓扑。MCU向每个参与者提供的内容为：
a) a selection of the content received from the other participants, or
a) 从其他参与者处收到的内容选择，或
b) the mixed aggregate of what the MCU receives from the other PtP paths, which are part of the same conference session.
b) MCU从属于同一会议会话的其他PtP路径接收的内容的混合聚合。
In case a), the MCU may modify the content in bit-rate, encoding, or resolution. No explicit RTP mechanism is used to establish the relationship between the original media sender and the version the MCU sends. In other words, the outgoing sessions typically use a different SSRC, and may well use a different payload type (PT), even if this different PT happens to be mapped to the same media type. This is a result of the individually negotiated session for each participant.
在情况a）中，MCU可以在比特率、编码或分辨率方面修改内容。没有使用明确的RTP机制来建立原始媒体发送器和MCU发送的版本之间的关系。换句话说，传出会话通常使用不同的SSRC，并且很可能使用不同的有效负载类型（PT），即使该不同的PT恰好映射到相同的媒体类型。这是每个参与者单独协商会议的结果。
In case b), the MCU is the content source as it mixes the content and then encodes it for transmission to a participant. According to RTP [RFC3550], the SSRC of the contributors are to be signalled using the CSRC/CC mechanism. In practice, today, most deployed MCUs do not implement this feature. Instead, the identification of the participants whose content is included in the Mixer's output is not indicated through any explicit RTP mechanism. That is, most deployed MCUs set the CSRC Count (CC) field in the RTP header to zero, thereby indicating no available CSRC information, even if they could identify the content sources as suggested in RTP.
在情况b）中，MCU是内容源，因为它混合了内容，然后对其进行编码以传输给参与者。根据RTP[RFC3550]，投稿人的SSRC将使用CSC/CC机制发出信号。实际上，目前大多数部署的MCU都没有实现此功能。相反，其内容包含在混音器输出中的参与者的标识不是通过任何显式RTP机制指示的。也就是说，大多数部署的MCU将RTP头中的CSC计数（CC）字段设置为零，从而表明没有可用的CSC信息，即使它们可以按照RTP中的建议识别内容源。
The main feature that sets this topology apart from what RFC 3550 describes is the breaking of the common RTP session across the centralized device, such as the MCU. This results in the loss of explicit RTP-level indication of all participants. If one were using the mechanisms available in RTP and RTCP to signal this explicitly, the topology would follow the approach of an RTP Mixer. The lack of explicit indication has at least the following potential problems:
将此拓扑与RFC 3550描述的拓扑区别开来的主要功能是在集中式设备（如MCU）上中断公共RTP会话。这会导致所有参与者失去明确的RTP水平指示。如果使用RTP和RTCP中可用的机制来显式地发出信号，则拓扑结构将遵循RTP混频器的方法。缺乏明确指示至少有以下潜在问题：
1) Loop detection cannot be performed on the RTP level. When carelessly connecting two misconfigured MCUs, a loop could be generated.
1) 无法在RTP级别上执行循环检测。当不小心连接两个配置错误的MCU时，可能会生成循环。
2) There is no information about active media senders available in the RTP packet. As this information is missing, receivers cannot use it. It also deprives the client of information related to currently active senders in a machine-usable way, thus preventing clients from indicating currently active speakers in user interfaces, etc.
2) RTP数据包中没有关于活动媒体发送器的信息。由于缺少此信息，接收者无法使用它。它还以机器可用的方式剥夺客户端与当前活动发送者相关的信息，从而防止客户端在用户界面中指示当前活动的扬声器等。
Note that deployed MCUs (and endpoints) rely on signalling layer mechanisms for the identification of the contributing sources, for example, a SIP conferencing package [RFC4575]. This alleviates, to some extent, the aforementioned issues resulting from ignoring RTP's CSRC mechanism.
注意，部署的MCU（和端点）依赖于信令层机制来识别贡献源，例如，SIP会议包[RFC4575]。这在一定程度上缓解了由于忽视RTP的证监会机制而导致的上述问题。
As a result of the shortcomings of this topology, it is RECOMMENDED to instead implement the Mixer concept as specified by RFC 3550.
由于这种拓扑结构的缺点，建议改为实施RFC 3550规定的混频器概念。
Shortcut name: Topo-Asymmetric
快捷方式名称：拓扑不对称
It is theoretically possible to construct an MCU that is a Mixer in one direction and a Translator in another. The main reason to consider this would be to allow topologies similar to Figure 5, where the Mixer does not need to mix in the direction from B or D towards the multicast domains with A and C. Instead, the media streams from
从理论上讲，可以构造一个MCU，它在一个方向上是混频器，在另一个方向上是转换器。考虑这一点的主要原因是允许类似于图5的拓扑结构，其中混频器不需要用A和C从B或D向多播域的方向混合，而是从
B and D are forwarded without changes. Avoiding this mixing would save media processing resources that perform the mixing in cases where it isn't needed. However, there would still be a need to mix B's stream towards D. Only in the direction B -> multicast domain or D -> multicast domain would it be possible to work as a Translator. In all other directions, it would function as a Mixer.
B和D被转发而不作更改。避免这种混合将节省在不需要的情况下执行混合的媒体处理资源。但是，仍然需要将B的流混合到D。只有在方向B->multicast domain或D->multicast domain中，才有可能作为转换器工作。在所有其他方向上，它将起到混合器的作用。
The Mixer/Translator would still need to process and change the RTCP before forwarding it in the directions of B or D to the multicast domain. One issue is that A and C do not know about the mixed-media stream the Mixer sends to either B or D. Thus, any reports related to these streams must be removed. Also, receiver reports related to A and C's media stream would be missing. To avoid A and C thinking that B and D aren't receiving A and C at all, the Mixer needs to insert its Receiver Reports for the streams from A and C into B and D's Sender Reports. In the opposite direction, the Receiver Reports from A and C about B's and D's stream also need to be aggregated into the Mixer's Receiver Reports sent to B and D. Since B and D only have the Mixer as source for the stream, all RTCP from A and C must be suppressed by the Mixer.
混音器/转换器在将RTCP沿B或D方向转发到多播域之前，仍然需要处理和更改RTCP。一个问题是A和C不知道混合器发送给B或D的混合媒体流。因此，必须删除与这些流相关的任何报告。此外，与A和C的媒体流相关的接收器报告也将丢失。为了避免A和C认为B和D根本没有接收A和C，混合器需要将来自A和C的流的接收器报告插入B和D的发送者报告中。相反，来自A和C的关于B和D流的接收器报告也需要聚合到发送给B和D的混频器接收器报告中。由于B和D仅将混频器作为流的源，因此混频器必须抑制来自A和C的所有RTCP。
This topology is so problematic and it is so easy to get the RTCP processing wrong, that it is NOT RECOMMENDED to implement this topology.
此拓扑问题严重，很容易使RTCP处理出错，因此不建议实施此拓扑。
Topologies can be combined and linked to each other using Mixers or Translators. However, care must be taken in handling the SSRC/CSRC space. A Mixer will not forward RTCP from sources in other domains, but will instead generate its own RTCP packets for each domain it mixes into, including the necessary Source Description (SDES) information for both the CSRCs and the SSRCs. Thus, in a mixed domain, the only SSRCs seen will be the ones present in the domain, while there can be CSRCs from all the domains connected together with a combination of Mixers and Translators. The combined SSRC and CSRC space is common over any Translator or Mixer. This is important to facilitate loop detection, something that is likely to be even more important in combined topologies due to the mixed behavior between the domains. Any hybrid, like the Topo-Video-switch-MCU or Topo-Asymmetric, requires considerable thought on how RTCP is dealt with.
拓扑可以使用混合器或转换器进行组合并相互链接。但是，在处理SSRC/CSRC空间时必须小心。混频器不会从其他域中的源转发RTCP，而是为其混入的每个域生成自己的RTCP数据包，包括CSRC和SSRC的必要源描述（SDES）信息。因此，在混合域中，看到的唯一SSRC将是域中存在的SSRC，而所有域中的CSRC可以通过混频器和转换器的组合连接在一起。组合的SSRC和CSRC空间在任何翻译器或混音器上都是通用的。这对于促进循环检测非常重要，由于域之间的混合行为，这在组合拓扑中可能更为重要。任何混合，如Topo视频开关MCU或Topo不对称，都需要大量考虑如何处理RTCP。
The topologies discussed in Section 3 have different properties. This section first lists these properties and then maps the different topologies to them. Please note that even if a certain property is supported within a particular topology concept, the necessary functionality may, in many cases, be optional to implement.
第3节中讨论的拓扑具有不同的特性。本节首先列出这些特性，然后将不同的拓扑映射到它们。请注意，即使特定拓扑概念中支持某个属性，在许多情况下，必要的功能可能是可选的。
Multicast, at least Any Source Multicast (ASM), provides the functionality that everyone may send to, or receive from, everyone else within the session. MCUs, Mixers, and Translators may all provide that functionality at least on some basic level. However, there are some differences in which type of reachability they provide.
多播，至少是任何源多播（ASM），提供了每个人都可以向会话中的其他人发送或接收的功能。MCU、混音器和翻译器都可能至少在一些基本层面上提供该功能。然而，它们提供的可达性类型存在一些差异。
The transport Translator function called "relay", in Section 3.3, is the one that provides the emulation of ASM that is closest to true IP-multicast-based, all to all transmission. Media Translators, Mixers, and the MCU variants do not provide a fully meshed forwarding on the transport level; instead, they only allow limited forwarding of content from the other session participants.
在第3.3节中，称为“中继”的传输转换器功能提供了最接近真正基于IP多播的全对全传输的ASM仿真。媒体转换器、混音器和MCU变体在传输级别上不提供完全网状的转发；相反，它们只允许从其他会话参与者有限地转发内容。
The "all to all media transmission" requires that any media transmitting entity considers the path to the least capable receiver. Otherwise, the media transmissions may overload that path. Therefore, a media sender needs to monitor the path from itself to any of the participants, to detect the currently least capable receiver, and adapt its sending rate accordingly. As multiple participants may send simultaneously, the available resources may vary. RTCP's Receiver Reports help performing this monitoring, at least on a medium time scale.
“全对全媒体传输”要求任何媒体传输实体考虑到能力最低的接收器的路径。否则，媒体传输可能会使该路径过载。因此，媒体发送者需要监控从其自身到任何参与者的路径，以检测当前能力最低的接收器，并相应地调整其发送速率。由于多个参与者可能同时发送，可用资源可能会有所不同。RTCP的接收器报告有助于执行此监控，至少在中等时间范围内。
The transmission of RTCP automatically adapts to any changes in the number of participants due to the transmission algorithm, defined in the RTP specification [RFC3550], and the extensions in AVPF [RFC4585] (when applicable). That way, the resources utilized for RTCP stay within the bounds configured for the session.
由于RTP规范[RFC3550]中定义的传输算法和AVPF[RFC4585]中的扩展（如适用），RTCP的传输自动适应参与者数量的任何变化。这样，用于RTCP的资源将保持在为会话配置的边界内。
Translators, Mixers, and RTCP-terminating MCU all allow changing the media encoding or the transport to other properties of the other domain, thereby providing extended interoperability in cases where the participants lack a common set of media codecs and/or transport protocols.
转换器、混频器和RTCP端接MCU都允许更改媒体编码或传输到其他域的其他属性，从而在参与者缺少一组通用的媒体编解码器和/或传输协议的情况下提供扩展的互操作性。
Participants are most likely to be connected to each other with a heterogeneous set of paths. This makes congestion control in a Point to Multipoint set problematic. For the ASM and "relay" scenario, each individual sender has to adapt to the receiver with the least capable path. This is no longer necessary when Media Translators, Mixers, or MCUs are involved, as each participant only needs to adapt to the slowest path within its own domain. The Translator, Mixer, or MCU topologies all require their respective outgoing streams to adjust the bit-rate, packet-rate, etc., to adapt to the least capable path in each of the other domains. That way one can avoid lowering the quality to the least-capable participant in all the domains at the cost (complexity, delay, equipment) of the Mixer or Translator.
参与者最有可能通过一组异构路径相互连接。这使得点对多点集中的拥塞控制成为问题。对于ASM和“中继”场景，每个单独的发送方必须适应具有最低能力路径的接收方。当涉及媒体翻译器、混音器或MCU时，这不再是必要的，因为每个参与者只需要适应自己领域内最慢的路径。转换器、混频器或MCU拓扑都需要它们各自的输出流来调整比特率、分组速率等，以适应每个其他域中的最小能力路径。这样就可以避免以混音器或翻译器的成本（复杂性、延迟、设备）降低所有领域中能力最低的参与者的质量。
In the all to all media property mentioned above and provided by ASM, all simultaneous media transmissions share the available bit-rate. For participants with limited reception capabilities, this may result in a situation where even a minimal acceptable media quality cannot be accomplished. This is the result of multiple media streams needing to share the available resources. The solution to this problem is to provide for a Mixer or MCU to aggregate the multiple streams into a single one. This aggregation can be performed according to different methods. Mixing or selection are two common methods.
在上述由ASM提供的全对全媒体属性中，所有同步媒体传输共享可用比特率。对于接收能力有限的参与者，这可能导致即使是最低可接受的媒体质量也无法实现的情况。这是多个媒体流需要共享可用资源的结果。这个问题的解决方案是提供一个混合器或MCU来将多个流聚合为一个流。可以根据不同的方法执行此聚合。混合或选择是两种常见的方法。
The RTP protocol includes functionality to identify the session participants through the use of the SSRC and CSRC fields. In addition, it is capable of carrying some further identity information about these participants using the RTCP Source Descriptors (SDES). To maintain this functionality, it is necessary that RTCP is handled correctly in domain bridging function. This is specified for Translators and Mixers. The MCU described in Section 3.5 does not entirely fulfill this. The one described in Section 3.6 does not support this at all.
RTP协议包括通过使用SSRC和CSRC字段来识别会话参与者的功能。此外，它还能够使用RTCP源描述符（SDE）携带关于这些参与者的更多身份信息。要维护此功能，必须在域桥接功能中正确处理RTCP。这是为转换器和混音器指定的。第3.5节中描述的MCU不完全满足此要求。第3.6节中所述的完全不支持这一点。
In complex topologies with multiple interconnected domains, it is possible to form media loops. RTP and RTCP support detecting such loops, as long as the SSRC and CSRC identities are correctly set in forwarded packets. It is likely that loop detection works for the MCU, described in Section 3.5, at least as long as it forwards the RTCP between the participants. However, the MCU in Section 3.6 will definitely break the loop detection mechanism.
在具有多个互连域的复杂拓扑中，可能形成介质环路。RTP和RTCP支持检测此类循环，只要在转发的数据包中正确设置SSRC和CSC标识。循环检测可能适用于MCU，如第3.5节所述，至少只要它在参与者之间转发RTCP。但是，第3.6节中的MCU肯定会打破循环检测机制。
The table below attempts to summarize the properties of the different topologies. The legend to the topology abbreviations are: Topo-Point-to-Point (PtP), Topo-Multicast (Multic), Topo-Trns-Translator (TTrn), Topo-Media-Translator (including Transport Translator) (MTrn), Topo-Mixer (Mixer), Topo-Asymmetric (ASY), Topo-Video-switch-MCU (MCUs), and Topo-RTCP-terminating-MCU (MCUt). In the table below, Y indicates Yes or full support, N indicates No support, (Y) indicates partial support, and N/A indicates not applicable.
下表试图总结不同拓扑的特性。拓扑缩写的图例为：拓扑点对点（PtP）、拓扑多播（Multic）、拓扑Trns转换器（TTrn）、拓扑媒体转换器（包括传输转换器）（MTrn）、拓扑混音器（混音器）、拓扑非对称（ASY）、拓扑视频开关MCU（MCU）和拓扑RTCP终端MCU（MCUt）。在下表中，Y表示是或完全支持，N表示不支持，（Y）表示部分支持，N/A表示不适用。
Property PtP Multic TTrn MTrn Mixer ASY MCUs MCUt
------------------------------------------------------------------
All to All media N Y Y Y (Y) (Y) (Y) (Y)
Interoperability N/A N Y Y Y Y N Y
Per Domain Adaptation N/A N N Y Y Y N Y
Aggregation of media N N N N Y (Y) Y Y
Full Session View Y Y Y Y Y Y (Y) N
Loop Detection Y Y Y Y Y Y (Y) N
Property PtP Multic TTrn MTrn Mixer ASY MCUs MCUt
------------------------------------------------------------------
All to All media N Y Y Y (Y) (Y) (Y) (Y)
Interoperability N/A N Y Y Y Y N Y
Per Domain Adaptation N/A N N Y Y Y N Y
Aggregation of media N N N N Y (Y) Y Y
Full Session View Y Y Y Y Y Y (Y) N
Loop Detection Y Y Y Y Y Y (Y) N
Please note that the Media Translator also includes the transport Translator functionality.
请注意，媒体转换器还包括传输转换器功能。
The use of Mixers and Translators has impact on security and the security functions used. The primary issue is that both Mixers and Translators modify packets, thus preventing the use of integrity and source authentication, unless they are trusted devices that take part in the security context, e.g., the device can send Secure Realtime Transport Protocol (SRTP) and Secure Realtime Transport Control Protocol (SRTCP) [RFC3711] packets to session endpoints. If encryption is employed, the media Translator and Mixer need to be able to decrypt the media to perform its function. A transport Translator may be used without access to the encrypted payload in cases where it translates parts that are not included in the encryption and integrity protection, for example, IP address and UDP
混音器和翻译器的使用会影响安全性和使用的安全功能。主要问题是混频器和转换器都会修改数据包，从而阻止使用完整性和源身份验证，除非它们是参与安全上下文的受信任设备，例如，设备可以发送安全实时传输协议（SRTP）和安全实时传输控制协议（SRTCP）[RFC3711]将数据包发送到会话端点。如果采用加密，媒体转换器和混频器需要能够解密媒体以执行其功能。在传输转换器转换未包括在加密和完整性保护中的部分（例如，IP地址和UDP）的情况下，可以在不访问加密有效负载的情况下使用传输转换器
port numbers in a media stream using SRTP [RFC3711]. However, in general, the Translator or Mixer needs to be part of the signalling context and get the necessary security associations (e.g., SRTP crypto contexts) established with its RTP session participants.
使用SRTP[RFC3711]的媒体流中的端口号。然而，一般来说，转换器或混频器需要是信令上下文的一部分，并获得与其RTP会话参与者建立的必要安全关联（例如，SRTP加密上下文）。
Including the Mixer and Translator in the security context allows the entity, if subverted or misbehaving, to perform a number of very serious attacks as it has full access. It can perform all the attacks possible (see RFC 3550 and any applicable profiles) as if the media session were not protected at all, while giving the impression to the session participants that they are protected.
在安全上下文中包含混合器和转换器允许实体（如果被破坏或行为不当）执行大量非常严重的攻击，因为它具有完全访问权限。它可以执行所有可能的攻击（请参阅RFC 3550和任何适用的配置文件），就好像媒体会话根本没有受到保护一样，同时给会话参与者留下他们受到保护的印象。
Transport Translators have no interactions with cryptography that works above the transport layer, such as SRTP, since that sort of Translator leaves the RTP header and payload unaltered. Media Translators, on the other hand, have strong interactions with cryptography, since they alter the RTP payload. A media Translator in a session that uses cryptographic protection needs to perform cryptographic processing to both inbound and outbound packets.
传输转换器与传输层（如SRTP）之上的加密技术没有交互，因为这种转换器使RTP头和有效负载保持不变。另一方面，媒体翻译器与密码学有很强的交互作用，因为它们改变了RTP负载。使用加密保护的会话中的媒体转换器需要对入站和出站数据包执行加密处理。
A media Translator may need to use different cryptographic keys for the inbound and outbound processing. For SRTP, different keys are required, because an RFC 3550 media Translator leaves the SSRC unchanged during its packet processing, and SRTP key sharing is only allowed when distinct SSRCs can be used to protect distinct packet streams.
媒体转换器可能需要为入站和出站处理使用不同的加密密钥。对于SRTP，需要不同的密钥，因为RFC 3550媒体转换器在其数据包处理期间保持SSRC不变，并且只有当可以使用不同的SSRC来保护不同的数据包流时，才允许SRTP密钥共享。
When the media Translator uses different keys to process inbound and outbound packets, each session participant needs to be provided with the appropriate key, depending on whether they are listening to the Translator or the original source. (Note that there is an architectural difference between RTP media translation, in which participants can rely on the RTP Payload Type field of a packet to determine appropriate processing, and cryptographically protected media translation, in which participants must use information that is not carried in the packet.)
当媒体翻译器使用不同的密钥来处理入站和出站数据包时，需要向每个会话参与者提供适当的密钥，这取决于他们是在侦听翻译器还是原始源。（请注意，RTP媒体翻译（参与者可以依赖数据包的RTP有效负载类型字段来确定适当的处理）和受密码保护的媒体翻译（参与者必须使用数据包中未携带的信息）之间存在架构差异。）
When using security mechanisms with Translators and Mixers, it is possible that the Translator or Mixer could create different security associations for the different domains they are working in. Doing so has some implications:
当对转换器和混频器使用安全机制时，转换器或混频器可能会为其工作的不同域创建不同的安全关联。这样做会产生一些影响：
First, it might weaken security if the Mixer/Translator accepts a weaker algorithm or key in one domain than in another. Therefore, care should be taken that appropriately strong security parameters are negotiated in all domains. In many cases, "appropriate"
首先，如果混音器/转换器在一个域中接受的算法或密钥比在另一个域中接受的算法或密钥弱，则可能会削弱安全性。因此，应注意在所有域中协商适当的强安全参数。在许多情况下，“适当”
translates to "similar" strength. If a key management system does allow the negotiation of security parameters resulting in a different strength of the security, then this system SHOULD notify the participants in the other domains about this.
翻译成“相似”的力量。如果密钥管理系统允许协商安全参数，从而产生不同的安全强度，则该系统应将此情况通知其他域中的参与者。
Second, the number of crypto contexts (keys and security related state) needed (for example, in SRTP [RFC3711]) may vary between Mixers and Translators. A Mixer normally needs to represent only a single SSRC per domain and therefore needs to create only one security association (SRTP crypto context) per domain. In contrast, a Translator needs one security association per participant it translates towards, in the opposite domain. Considering Figure 3, the Translator needs two security associations towards the multicast domain, one for B and one for D. It may be forced to maintain a set of totally independent security associations between itself and B and D respectively, so as to avoid two-time pad occurrences. These contexts must also be capable of handling all the sources present in the other domains. Hence, using completely independent security associations (for certain keying mechanisms) may force a Translator to handle N*DM keys and related state; where N is the total number of SSRCs used over all domains and DM is the total number of domains.
其次，所需的加密上下文（密钥和安全相关状态）的数量（例如，在SRTP[RFC3711]中）在混频器和转换器之间可能有所不同。混合器通常只需要在每个域中表示一个SSRC，因此每个域只需要创建一个安全关联（SRTP加密上下文）。相反，翻译人员需要在相反的域中为其翻译的每个参与者建立一个安全关联。考虑到图3，翻译器需要两个多播域的安全关联，一个用于B，一个用于D。翻译器可能被迫在其自身与B和D之间分别维护一组完全独立的安全关联，以避免两次pad发生。这些上下文还必须能够处理其他域中存在的所有源。因此，使用完全独立的安全关联（对于某些键控机制）可能会迫使转换器处理N*DM密钥和相关状态；其中N是所有域上使用的SSRC总数，DM是域总数。
There exist a number of different mechanisms to provide keys to the different participants. One example is the choice between group keys and unique keys per SSRC. The appropriate keying model is impacted by the topologies one intends to use. The final security properties are dependent on both the topologies in use and the keying mechanisms' properties, and need to be considered by the application. Exactly which mechanisms are used is outside of the scope of this document.
有许多不同的机制为不同的参与者提供密钥。一个例子是每个SSRC在组密钥和唯一密钥之间进行选择。适当的键控模型受打算使用的拓扑的影响。最终的安全属性取决于使用的拓扑和密钥机制的属性，应用程序需要考虑这些属性。具体使用哪些机制不在本文档的范围内。
The authors would like to thank Bo Burman, Umesh Chandra, Roni Even, Keith Lantz, Ladan Gharai, Geoff Hunt, and Mark Baugher for their help in reviewing this document.
作者要感谢Bo Burman、Umesh Chandra、Roni Een、Keith Lantz、Ladan Gharai、Geoff Hunt和Mark Baugher在审查本文件时提供的帮助。
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2119]Bradner，S.，“RFC中用于表示需求水平的关键词”，BCP 14，RFC 2119，1997年3月。
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003.
[RFC3550]Schulzrinne，H.，Casner，S.，Frederick，R.，和V.Jacobson，“RTP：实时应用的传输协议”，STD 64，RFC 35502003年7月。
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004.
[RFC3711]Baugher，M.，McGrew，D.，Naslund，M.，Carrara，E.，和K.Norrman，“安全实时传输协议（SRTP）”，RFC 37112004年3月。
[RFC4575] Rosenberg, J., Schulzrinne, H., and O. Levin, Ed., "A Session Initiation Protocol (SIP) Event Package for Conference State", RFC 4575, August 2006.
[RFC4575]Rosenberg，J.，Schulzrinne，H.，和O.Levin，Ed.，“会议状态的会话启动协议（SIP）事件包”，RFC 45752006年8月。
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 2006.
[RFC4585]Ott，J.，Wenger，S.，Sato，N.，Burmeister，C.，和J.Rey，“基于实时传输控制协议（RTCP）的反馈（RTP/AVPF）的扩展RTP配置文件”，RFC 45852006年7月。
[CCM] Wenger, S., Chandra, U., Westerlund, M., Burman, B., "Codec Control Messages in the RTP Audio-Visual Profile with Feedback (AVPF)", Work in Progress, July 2007.
[CCM]Wenger，S.，Chandra，U.，Westerlund，M.，Burman，B.，“带反馈的RTP视听配置文件（AVPF）中的编解码器控制消息”，正在进行的工作，2007年7月。
[H323] ITU-T Recommendation H.323, "Packet-based multimedia communications systems", June 2006.
[H323]ITU-T建议H.323，“基于分组的多媒体通信系统”，2006年6月。
[RTCP-SSM] J. Ott, J. Chesterfield, E. Schooler, "RTCP Extensions for Single-Source Multicast Sessions with Unicast Feedback," Work in Progress, March 2007.
[RTCP-SSM]J.Ott，J.Chesterfield，E.Schooler，“具有单播反馈的单源多播会话的RTCP扩展”，正在进行的工作，2007年3月。
Authors' Addresses
作者地址
Magnus Westerlund Ericsson Research Ericsson AB SE-164 80 Stockholm, SWEDEN
Magnus Westerlund Ericsson Research Ericsson AB SE-164 80瑞典斯德哥尔摩
Phone: +46 8 7190000
EMail: magnus.westerlund@ericsson.com
Phone: +46 8 7190000
EMail: magnus.westerlund@ericsson.com
Stephan Wenger Nokia Corporation P.O. Box 100 FIN-33721 Tampere FINLAND
斯蒂芬·温格诺基亚公司芬兰坦佩雷100 FIN-33721邮政信箱
Phone: +358-50-486-0637
EMail: stewe@stewe.org
Phone: +358-50-486-0637
EMail: stewe@stewe.org
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本文件受BCP 78中包含的权利、许可和限制的约束，除其中规定外，作者保留其所有权利。
This document and the information contained herein are provided on an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY, THE IETF TRUST AND THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
本文件及其包含的信息以“原样”为基础提供，贡献者、他/她所代表或赞助的组织（如有）、互联网协会、IETF信托基金和互联网工程任务组不承担任何明示或暗示的担保，包括但不限于任何保证，即使用本文中的信息不会侵犯任何权利，或对适销性或特定用途适用性的任何默示保证。
Intellectual Property
知识产权
The IETF takes no position regarding the validity or scope of any Intellectual Property Rights or other rights that might be claimed to pertain to the implementation or use of the technology described in this document or the extent to which any license under such rights might or might not be available; nor does it represent that it has made any independent effort to identify any such rights. Information on the procedures with respect to rights in RFC documents can be found in BCP 78 and BCP 79.
IETF对可能声称与本文件所述技术的实施或使用有关的任何知识产权或其他权利的有效性或范围，或此类权利下的任何许可可能或可能不可用的程度，不采取任何立场；它也不表示它已作出任何独立努力来确定任何此类权利。有关RFC文件中权利的程序信息，请参见BCP 78和BCP 79。
Copies of IPR disclosures made to the IETF Secretariat and any assurances of licenses to be made available, or the result of an attempt made to obtain a general license or permission for the use of such proprietary rights by implementers or users of this specification can be obtained from the IETF on-line IPR repository at http://www.ietf.org/ipr.
向IETF秘书处披露的知识产权副本和任何许可证保证，或本规范实施者或用户试图获得使用此类专有权利的一般许可证或许可的结果，可从IETF在线知识产权存储库获取，网址为http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention any copyrights, patents or patent applications, or other proprietary rights that may cover technology that may be required to implement this standard. Please address the information to the IETF at ietf-ipr@ietf.org.
IETF邀请任何相关方提请其注意任何版权、专利或专利申请，或其他可能涵盖实施本标准所需技术的专有权利。请将信息发送至IETF的IETF-ipr@ietf.org.