Internet Engineering Task Force (IETF)                     M. Westerlund
Request for Comments: 8088                                      Ericsson
Updates: 2736                                                   May 2017
Category: Informational
ISSN: 2070-1721
Internet Engineering Task Force (IETF)                     M. Westerlund
Request for Comments: 8088                                      Ericsson
Updates: 2736                                                   May 2017
Category: Informational
ISSN: 2070-1721

How to Write an RTP Payload Format




This document contains information on how best to write an RTP payload format specification. It provides reading tips, design practices, and practical tips on how to produce an RTP payload format specification quickly and with good results. A template is also included with instructions.


Status of This Memo


This document is not an Internet Standards Track specification; it is published for informational purposes.


This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Not all documents approved by the IESG are a candidate for any level of Internet Standard; see Section 2 of RFC 7841.

本文件是互联网工程任务组(IETF)的产品。它代表了IETF社区的共识。它已经接受了公众审查,并已被互联网工程指导小组(IESG)批准出版。并非IESG批准的所有文件都适用于任何级别的互联网标准;见RFC 7841第2节。

Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at


Copyright Notice


Copyright (c) 2017 IETF Trust and the persons identified as the document authors. All rights reserved.

版权所有(c)2017 IETF信托基金和确定为文件作者的人员。版权所有。

This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents ( in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License.

本文件受BCP 78和IETF信托有关IETF文件的法律规定的约束(自本文件出版之日起生效。请仔细阅读这些文件,因为它们描述了您对本文件的权利和限制。从本文件中提取的代码组件必须包括信托法律条款第4.e节中所述的简化BSD许可证文本,并提供简化BSD许可证中所述的无担保。

Table of Contents


   1. Introduction ....................................................4
      1.1. Structure ..................................................4
   2. Terminology .....................................................5
      2.1. Definitions ................................................5
      2.2. Abbreviations ..............................................5
      2.3. Use of Normative Requirements Language .....................6
   3. Preparations ....................................................6
      3.1. Read and Understand the Media Coding Specification .........6
      3.2. Recommended Reading ........................................7
           3.2.1. IETF Process and Publication ........................7
           3.2.2. RTP .................................................9
      3.3. Important RTP Details .....................................13
           3.3.1. The RTP Session ....................................13
           3.3.2. RTP Header .........................................14
           3.3.3. RTP Multiplexing ...................................16
           3.3.4. RTP Synchronization ................................16
      3.4. Signaling Aspects .........................................18
           3.4.1. Media Types ........................................19
           3.4.2. Mapping to SDP .....................................20
      3.5. Transport Characteristics .................................23
           3.5.1. Path MTU ...........................................23
           3.5.2. Different Queuing Algorithms .......................23
           3.5.3. Quality of Service .................................24
   4. Standardization Process for an RTP Payload Format ..............24
      4.1. IETF ......................................................25
           4.1.1. Steps from Idea to Publication .....................25
           4.1.2. WG Meetings ........................................27
           4.1.3. Draft Naming .......................................27
           4.1.4. Writing Style ......................................28
           4.1.5. How to Speed Up the Process ........................29
      4.2. Other Standards Bodies ....................................29
      4.3. Proprietary and Vendor Specific ...........................30
      4.4. Joint Development of Media Coding Specification
           and RTP Payload Format ....................................31
   5. Designing Payload Formats ......................................31
      5.1. Features of RTP Payload Formats ...........................32
           5.1.1. Aggregation ........................................32
           5.1.2. Fragmentation ......................................33
           5.1.3. Interleaving and Transmission Rescheduling .........33
           5.1.4. Media Back Channels ................................34
           5.1.5. Media Scalability ..................................34
           5.1.6. High Packet Rates ..................................37
      5.2. Selecting Timestamp Definition ............................37
   1. Introduction ....................................................4
      1.1. Structure ..................................................4
   2. Terminology .....................................................5
      2.1. Definitions ................................................5
      2.2. Abbreviations ..............................................5
      2.3. Use of Normative Requirements Language .....................6
   3. Preparations ....................................................6
      3.1. Read and Understand the Media Coding Specification .........6
      3.2. Recommended Reading ........................................7
           3.2.1. IETF Process and Publication ........................7
           3.2.2. RTP .................................................9
      3.3. Important RTP Details .....................................13
           3.3.1. The RTP Session ....................................13
           3.3.2. RTP Header .........................................14
           3.3.3. RTP Multiplexing ...................................16
           3.3.4. RTP Synchronization ................................16
      3.4. Signaling Aspects .........................................18
           3.4.1. Media Types ........................................19
           3.4.2. Mapping to SDP .....................................20
      3.5. Transport Characteristics .................................23
           3.5.1. Path MTU ...........................................23
           3.5.2. Different Queuing Algorithms .......................23
           3.5.3. Quality of Service .................................24
   4. Standardization Process for an RTP Payload Format ..............24
      4.1. IETF ......................................................25
           4.1.1. Steps from Idea to Publication .....................25
           4.1.2. WG Meetings ........................................27
           4.1.3. Draft Naming .......................................27
           4.1.4. Writing Style ......................................28
           4.1.5. How to Speed Up the Process ........................29
      4.2. Other Standards Bodies ....................................29
      4.3. Proprietary and Vendor Specific ...........................30
      4.4. Joint Development of Media Coding Specification
           and RTP Payload Format ....................................31
   5. Designing Payload Formats ......................................31
      5.1. Features of RTP Payload Formats ...........................32
           5.1.1. Aggregation ........................................32
           5.1.2. Fragmentation ......................................33
           5.1.3. Interleaving and Transmission Rescheduling .........33
           5.1.4. Media Back Channels ................................34
           5.1.5. Media Scalability ..................................34
           5.1.6. High Packet Rates ..................................37
      5.2. Selecting Timestamp Definition ............................37
   6. Noteworthy Aspects in Payload Format Design ....................39
      6.1. Audio Payloads ............................................39
      6.2. Video .....................................................40
      6.3. Text ......................................................41
      6.4. Application ...............................................41
   7. Important Specification Sections ...............................42
      7.1. Media Format Description ..................................42
      7.2. Security Considerations ...................................43
      7.3. Congestion Control ........................................44
      7.4. IANA Considerations .......................................45
   8. Authoring Tools ................................................45
      8.1. Editing Tools .............................................46
      8.2. Verification Tools ........................................46
   9. Security Considerations ........................................47
   10. Informative References ........................................47
   Appendix A. RTP Payload Format Template ...........................58
     A.1.  Title .....................................................58
     A.2.  Front-Page Boilerplate ....................................58
     A.3.  Abstract ..................................................58
     A.4.  Table of Contents .........................................58
     A.5.  Introduction ..............................................59
     A.6.  Conventions, Definitions, and Abbreviations ...............59
     A.7.  Media Format Description ..................................59
     A.8.  Payload Format ............................................59
       A.8.1.  RTP Header Usage ......................................59
       A.8.2.  Payload Header ........................................59
       A.8.3.  Payload Data ..........................................60
     A.9.  Payload Examples ..........................................60
     A.10. Congestion Control Considerations .........................60
     A.11. Payload Format Parameters .................................60
       A.11.1.  Media Type Definition ................................60
       A.11.2.  Mapping to SDP .......................................62
     A.12. IANA Considerations .......................................63
     A.13. Security Considerations ...................................63
     A.14. RFC Editor Considerations .................................64
     A.15. References ................................................64
       A.15.1.  Normative References .................................64
       A.15.2.  Informative References ...............................64
     A.16. Authors' Addresses ........................................64
   Acknowledgements ..................................................64
   Contributors ......................................................65
   Author's Address ..................................................65
   6. Noteworthy Aspects in Payload Format Design ....................39
      6.1. Audio Payloads ............................................39
      6.2. Video .....................................................40
      6.3. Text ......................................................41
      6.4. Application ...............................................41
   7. Important Specification Sections ...............................42
      7.1. Media Format Description ..................................42
      7.2. Security Considerations ...................................43
      7.3. Congestion Control ........................................44
      7.4. IANA Considerations .......................................45
   8. Authoring Tools ................................................45
      8.1. Editing Tools .............................................46
      8.2. Verification Tools ........................................46
   9. Security Considerations ........................................47
   10. Informative References ........................................47
   Appendix A. RTP Payload Format Template ...........................58
     A.1.  Title .....................................................58
     A.2.  Front-Page Boilerplate ....................................58
     A.3.  Abstract ..................................................58
     A.4.  Table of Contents .........................................58
     A.5.  Introduction ..............................................59
     A.6.  Conventions, Definitions, and Abbreviations ...............59
     A.7.  Media Format Description ..................................59
     A.8.  Payload Format ............................................59
       A.8.1.  RTP Header Usage ......................................59
       A.8.2.  Payload Header ........................................59
       A.8.3.  Payload Data ..........................................60
     A.9.  Payload Examples ..........................................60
     A.10. Congestion Control Considerations .........................60
     A.11. Payload Format Parameters .................................60
       A.11.1.  Media Type Definition ................................60
       A.11.2.  Mapping to SDP .......................................62
     A.12. IANA Considerations .......................................63
     A.13. Security Considerations ...................................63
     A.14. RFC Editor Considerations .................................64
     A.15. References ................................................64
       A.15.1.  Normative References .................................64
       A.15.2.  Informative References ...............................64
     A.16. Authors' Addresses ........................................64
   Acknowledgements ..................................................64
   Contributors ......................................................65
   Author's Address ..................................................65
1. Introduction
1. 介绍

RTP [RFC3550] payload formats define how a specific real-time data format is structured in the payload of an RTP packet. A real-time data format without a payload format specification cannot be transported using RTP. This creates an interest in many individuals/ organizations with media encoders or other types of real-time data to define RTP payload formats. However, the specification of a well-designed RTP payload format is nontrivial and requires knowledge of both RTP and the real-time data format.


This document is intended to help any author of an RTP payload format specification make important design decisions, consider important features of RTP and RTP security, etc. The document is also intended to be a good starting point for any person with little experience in the IETF and/or RTP to learn the necessary steps.


This document extends and updates the information that is available in "Guidelines for Writers of RTP Payload Format Specifications" [RFC2736]. Since that RFC was written, further experience has been gained on the design and specification of RTP payload formats. Several new RTP profiles and robustness tools have been defined, and these need to be considered.


This document also discusses the possible venues for defining an RTP payload format: the IETF, other standards bodies, and proprietary ones.


Note, this document does discuss IETF, IANA, and RFC Editor processes and rules as they were when this document was published. This to make clear how the work to specify an RTP payload formats depends, uses, and interacts with these rules and processes. However, these rules and processes are subject to change and the formal rule and process specifications always takes precedence over what is written here.


1.1. Structure
1.1. 结构

This document has several different parts discussing different aspects of the creation of an RTP payload format specification. Section 3 discusses the preparations the author(s) should make before starting to write a specification. Section 4 discusses the different processes used when specifying and completing a payload format, with focus on working inside the IETF. Section 5 discusses the design of payload formats themselves in detail. Section 6 discusses current design trends and provides good examples of practices that should be followed when applicable. Following that, Section 7 provides a discussion on important sections in the RTP payload format


specification itself such as Security Considerations and IANA Considerations. This document ends with an appendix containing a template that can be used when writing RTP payload formats specifications.


2. Terminology
2. 术语
2.1. Definitions
2.1. 定义

RTP Stream: A sequence of RTP packets that together carry part or all of the content of a specific media (audio, video, text, or data whose form and meaning are defined by a specific real-time application) from a specific sender source within a given RTP session.


RTP Session: An association among a set of participants communicating with RTP. The distinguishing feature of an RTP session is that each session maintains a full, separate space of synchronization source (SSRC) identifiers. See also Section 3.3.1.


RTP Payload Format: The RTP payload format specifies how units of a specific encoded media are put into the RTP packet payloads and how the fields of the RTP packet header are used, thus enabling the format to be used in RTP applications.


A Taxonomy of Semantics and Mechanisms for Real-Time Transport Protocol (RTP) Sources [RFC7656] defines many useful terms.


2.2. Abbreviations
2.2. 缩写

ABNF: Augmented Backus-Naur Form [RFC5234]


ADU: Application Data Unit


ALF: Application Level Framing


ASM: Any-Source Multicast


BCP: Best Current Practice


I-D: Internet-Draft


IESG: Internet Engineering Steering Group


MTU: Maximum Transmission Unit


WG: Working Group


QoS: Quality of Service


RFC: Request For Comments


RTP: Real-time Transport Protocol


RTCP: RTP Control Protocol


RTT: Round-Trip Time


SSM: Source-Specific Multicast


2.3. Use of Normative Requirements Language
2.3. 规范性要求语言的使用

As this document is both Informational and instructional rather than a specification, this document does not use any RFC 2119 language and the use of "may", "should", "recommended", and "must" carries no special connotation.

由于本文件为信息性和指导性文件,而非规范性文件,因此本文件不使用任何RFC 2119语言,且“可能”、“应该”、“建议”和“必须”的使用没有特殊含义。

3. Preparations
3. 准备工作

RTP is a complex real-time media delivery framework, and it has a lot of details that need to be considered when writing an RTP payload format. It is also important to have a good understanding of the media codec / format so that all of its important features and properties are considered. Only when one has sufficient understanding of both parts can one produce an RTP payload format of high quality. On top of this, one needs to understand the process within the IETF and especially the Working Group responsible for standardizing payload formats (currently the PAYLOAD WG) to go quickly from the initial idea stage to a finished RFC. This and the next sections help an author prepare himself in those regards.


3.1. Read and Understand the Media Coding Specification
3.1. 阅读并理解媒体编码规范

It may be obvious, but it is necessary for an author of an RTP payload specification to have a solid understanding of the media to be transported. Important are not only the specifically spelled out transport aspects (if any) in the media coding specification, but also core concepts of the underlying technology. For example, an RTP payload format for video coded with inter-picture prediction will perform poorly if the payload designer does not take the use of inter-picture prediction into account. On the other hand, some (mostly older) media codecs offer error-resilience tools against bit errors, which, when misapplied over RTP, in almost all cases would only introduce overhead with no measurable return.


3.2. Recommended Reading
3.2. 推荐阅读

The following subsections list a number of documents. Not all need to be read in full detail. However, an author basically needs to be aware of everything listed below.


3.2.1. IETF Process and Publication
3.2.1. IETF过程和发布

Newcomers to the IETF are strongly recommended to read the "Tao of the IETF" [TAO] that goes through most things that one needs to know about the IETF: the history, organizational structure, how the WGs and meetings work, etc.


It is very important to note and understand the IETF Intellectual Property Rights (IPR) policy that requires early disclosures based on personal knowledge from anyone contributing in IETF. The IETF policies associated with IPR are documented in BCP 78 [BCP78] (related to copyright, including software copyright, for example, code) and BCP 79 [BCP79] (related to patent rights). These rules may be different from other standardization organizations. For example, a person that has a patent or a patent application that he or she reasonably and personally believes to cover a mechanism that gets added to the Internet-Draft they are contributing to (e.g., by submitting the draft, posting comments or suggestions on a mailing list, or speaking at a meeting) will need to make a timely IPR disclosure. Read the above documents for the authoritative rules. Failure to follow the IPR rules can have dire implications for the specification and the author(s) as discussed in [RFC6701].

注意并理解IETF知识产权(IPR)政策非常重要,该政策要求根据IETF贡献者的个人知识提前披露。与知识产权相关的IETF政策记录在BCP 78[BCP78](与版权相关,包括软件版权,例如代码)和BCP 79[BCP79](与专利权相关)中。这些规则可能不同于其他标准化组织。例如,一个人拥有一项专利或一项专利申请,他或她个人合理地认为该专利或专利申请涵盖了一种机制,该机制被添加到他们正在参与的互联网草案中(例如,通过提交草案、在邮件列表上发布评论或建议、或在会议上发言)需要及时披露知识产权信息。请阅读上述文件以了解权威规则。如[RFC6701]所述,不遵守IPR规则可能会对规范和作者造成严重影响。

Note: These IPR rules apply on what is specified in the RTP payload format Internet-Draft (and later RFC); an IPR that relates to a codec specification from an external body does not require IETF IPR disclosure. Informative text explaining the nature of the codec would not normally require an IETF IPR declaration. Appropriate IPR declarations for the codec itself would normally be found in files of the external body defining the codec, in accordance with that external body's own IPR rules.

注:这些知识产权规则适用于RTP有效载荷格式互联网草案(以及后来的RFC)中规定的内容;与外部主体的编解码器规范相关的IPR不需要披露IETF IPR。解释编解码器性质的信息性文本通常不需要IETF IPR声明。根据外部机构自身的IPR规则,通常可以在定义编解码器的外部机构的文件中找到编解码器本身的适当IPR声明。

The main part of the IETF process is formally defined in BCP 9 [BCP9]. BCP 25 [BCP25] describes the WG process, the relation between the IESG and the WG, and the responsibilities of WG Chairs and participants.

IETF过程的主要部分在BCP 9[BCP9]中正式定义。BCP 25[BCP25]描述了工作组流程、IESG和工作组之间的关系以及工作组主席和参与者的职责。

It is important to note that the RFC Series contains documents of several different publication streams as defined by The RFC Series and RFC Editor [RFC4844]. The most important stream for RTP payload formats authors is the IETF Stream. In this stream, the work of the IETF is published. The stream contains documents of several


different categories: Standards Track, Informational, Experimental, Best Current Practice, and Historic. "Standards Track" contains two maturity levels: Proposed Standard and Internet Standard [RFC6410]. A Standards Track document must start as a Proposed Standard; after successful deployment and operational experience with at least two implementations, it can be moved to an Internet Standard. The Independent Submission Stream could appear to be of interest as it provides a way of publishing documents of certain categories such as Experimental and Informational with a different review process. However, as long as IETF has a WG that is chartered to work on RTP payload formats, this stream should not be used.


As the content of a given RFC is not allowed to change once published, the only way to modify an RFC is to write and publish a new one that either updates or replaces the old one. Therefore, whether reading or referencing an RFC, it is important to consider both the Category field in the document header and to check if the RFC is the latest on the subject and still valid. One way of checking the current status of an RFC is to use the RFC Editor's RFC search page (, which displays the current status and which if any RFC has updated or obsoleted it. The RFC Editor search engine will also indicate if there exist any errata reports for the RFC. Any verified errata report contains issues of significant importance with the RFC; thus, they should be known prior to an update and replacement publication.


Before starting to write a draft, one should also read the Internet-Draft writing guidelines (, the I-D checklist (, and the RFC Style Guide [RFC7322]. Another document that can be useful is "Guide for Internet Standards Writers" [RFC2360].


There are also a number of documents to consider in the process of writing drafts intended to become RFCs. These are important when writing certain types of text.


RFC 2606: When writing examples using DNS names in Internet-Drafts, those names shall be chosen from the,, and domains.

RFC 2606:在互联网草稿中使用DNS名称编写示例时,这些名称应从、example.net和example.org域中选择。

RFC 3849: Defines the range of IPv6 unicast addresses (2001:DB8::/32) that should be used in any examples.

RFC 3849:定义应在任何示例中使用的IPv6单播地址范围(2001:DB8::/32)。

RFC 5737: Defines the ranges of IPv4 unicast addresses reserved for documentation and examples:,, and

RFC 5737:定义为文档和示例保留的IPv4单播地址范围:、和203.0.113.0/24。

RFC 5234: Augmented Backus-Naur Form (ABNF) is often used when writing text field specifications. Not commonly used in RTP payload formats, but may be useful when defining media type parameters of some complexity.

RFC 5234:在编写文本字段规范时,通常使用增广的巴科斯瑙格式(ABNF)。在RTP有效负载格式中不常用,但在定义某些复杂的媒体类型参数时可能有用。

3.2.2. RTP
3.2.2. RTP

The recommended reading for RTP consists of several different parts: design guidelines, the RTP protocol, profiles, robustness tools, and media-specific recommendations.


Any author of RTP payload formats should start by reading "Guidelines for Writers of RTP Payload Format Specifications" [RFC2736], which contains an introduction to the Application Level Framing (ALF) principle, the channel characteristics of IP channels, and design guidelines for RTP payload formats. The goal of ALF is to be able to transmit Application Data Units (ADUs) that are independently usable by the receiver in individual RTP packets, thus minimizing dependencies between RTP packets and the effects of packet loss.


Then, it is advisable to learn more about the RTP protocol, by studying the RTP specification "RTP: A Transport Protocol for Real-Time Applications" [RFC3550] and the existing profiles. As a complement to the Standards Track documents, there exists a book totally dedicated to RTP [CSP-RTP]. There exist several profiles for RTP today, but all are based on "RTP Profile for Audio and Video Conferences with Minimal Control" [RFC3551] (abbreviated as RTP/AVP). The other profiles that one should know about are "The Secure Real-time Transport Protocol (SRTP)" (RTP/SAVP) [RFC3711], "Extended RTP Profile for RTCP-based Feedback (RTP/AVPF)" [RFC4585], and "Extended Secure Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)" [RFC5124]. It is important to understand RTP and the RTP/AVP profile in detail. For the other profiles, it is sufficient to have an understanding of what functionality they provide and the limitations they create.


A number of robustness tools have been developed for RTP. The tools are for different use cases and real-time requirements.


RFC 2198: "RTP Payload for Redundant Audio Data" [RFC2198] provides functionalities to transmit redundant copies of audio or text payloads. These redundant copies are sent together with a primary format in the same RTP payload. This format relies on the RTP timestamp to determine where data belongs in a sequence; therefore, it is usually most suitable to be used with audio. However, the RTP Payload format for T.140 [RFC4103] text format also uses this format. The format's major property is that it only preserves the timestamp of the redundant payloads, not the

RFC 2198:“冗余音频数据的RTP有效载荷”[RFC2198]提供传输音频或文本有效载荷的冗余副本的功能。这些冗余副本与同一RTP有效负载中的主格式一起发送。该格式依赖于RTP时间戳来确定数据在序列中所属的位置;因此,它通常最适合与音频一起使用。但是,T.140[RFC4103]文本格式的RTP有效负载格式也使用此格式。该格式的主要特性是,它只保留冗余有效负载的时间戳,而不保留

original sequence number. This makes it unusable for most video formats. This format is also only suitable for media formats that produce relatively small RTP payloads.


RFC 6354: The "Forward-Shifted RTP Redundancy Payload Support" [RFC6354] is a variant of RFC 2198 that allows the redundant data to be transmitted prior to the original.

RFC 6354:“前移RTP冗余有效负载支持”[RFC6354]是RFC 2198的一个变体,它允许在原始数据之前传输冗余数据。

RFC 5109: The "RTP Payload Format for Generic Forward Error Correction" [RFC5109] provides an XOR-based Forward Error Correction (FEC) of the whole or parts of a number of RTP packets. This specification replaced the previous specification for XOR-based FEC [RFC2733]. These FEC packets are sent in a separate stream or as a redundant encoding using RFC 2198. This FEC scheme has certain restrictions in the number of packets it can protect. It is suitable for applications with low-to-medium delay tolerance with a limited amount of RTP packets.

RFC 5109:“通用前向纠错的RTP有效负载格式”[RFC5109]提供了一个基于异或的前向纠错(FEC),用于对多个RTP数据包的全部或部分进行纠错。本规范取代了先前基于XOR的FEC[RFC2733]规范。这些FEC分组以单独的流或使用RFC 2198作为冗余编码发送。该FEC方案在其可以保护的数据包数量上有一定的限制。它适用于RTP数据包数量有限的低到中等延迟容限的应用。

RFC 6015: "RTP Payload Format for 1-D Interleaved Parity Forward Error Correction (FEC)" [RFC6015] provides a variant of the XOR-based Generic protection defined in [RFC2733]. The main difference is to use interleaving scheme on which packets gets included as source packets for a particular protection packet. The interleaving is defined by using every L packets as source data and then producing protection data over D number of packets. Thus, each block of D x L source packets will result in L number of Repair packets, each capable of repairing one loss. The goal is to provide better burst-error robustness when the packet rate is higher.

RFC 6015:“用于1-D交错奇偶校验前向纠错(FEC)的RTP有效负载格式”[RFC6015]提供了[RFC2733]中定义的基于异或的通用保护的变体。主要区别在于使用交织方案,其中数据包作为特定保护数据包的源数据包被包括在内。交织是通过使用每L个数据包作为源数据,然后在D个数据包上生成保护数据来定义的。因此,dxl源分组的每个块将导致L个修复分组,每个能够修复一个丢失。目标是在包速率较高时提供更好的突发错误鲁棒性。

FEC Framework: "Forward Error Correction (FEC) Framework" [RFC6363] defines how to use FEC protection for arbitrary packet flows. This framework can be applied for RTP/RTCP packet flows, including using RTP for transmission of repair symbols, an example is in "RTP Payload Format for Raptor Forward Error Correction (FEC)" [RFC6682].


RTP Retransmission: The RTP retransmission scheme [RFC4588] is used for semi-reliability of the most important RTP packets in a RTP stream. The level of reliability between semi- and in-practice full reliability depends on the targeted properties and situation where parameters such as round-trip time (RTT) allowed additional overhead and allowable delay. It often requires the application to be quite delay tolerant as a minimum of one round-trip time plus processing delay is required to perform a retransmission. Thus, it is mostly suitable for streaming applications but may also be usable in certain other cases when operating in networks with short round-trip times.


RTP over TCP: RFC 4571 [RFC4571] defines how one sends RTP and RTCP packets over connection-oriented transports like TCP. If one uses TCP, one gets reliability for all packets but loses some of the real-time behavior that RTP was designed to provide. Issues with TCP transport of real-time media include head-of-line blocking and wasting resources on retransmission of data that is already late. TCP is also limited to point-to-point connections, which further restricts its applicability.

TCP上的RTP:RFC 4571[RFC4571]定义了如何通过面向连接的传输(如TCP)发送RTP和RTCP数据包。如果使用TCP,则可以获得所有数据包的可靠性,但会丢失RTP设计用于提供的一些实时行为。实时媒体的TCP传输存在的问题包括线首阻塞和在重新传输已经延迟的数据时浪费资源。TCP还限于点对点连接,这进一步限制了其适用性。

There have been both discussion and design of RTP payload formats, e.g., Adaptive Multi-Rate (AMR) and AMR Wideband (AMR-WB) [RFC4867], supporting the unequal error detection provided by UDP-Lite [RFC3828]. The idea is that by not having a checksum over part of the RTP payload one can allow bit errors from the lower layers. By allowing bit errors one can increase the efficiency of some link layers and also avoid unnecessary discarding of data when the payload and media codec can get at least some benefit from the data. The main issue is that one has no idea of the level of bit errors present in the unprotected part of the payload. This makes it hard or impossible to determine whether or not one can design something usable. Payload format designers are not recommended to consider features for unequal error detection using UDP-Lite unless very clear requirements exist.

已经讨论并设计了RTP有效负载格式,例如自适应多速率(AMR)和AMR宽带(AMR-WB)[RFC4867],支持UDP Lite[RFC3828]提供的不等错误检测。其思想是,通过在RTP有效负载的一部分上不具有校验和,可以允许来自较低层的位错误。通过允许比特错误,可以提高某些链路层的效率,并且当有效负载和媒体编解码器至少可以从数据中获得一些好处时,还可以避免不必要的数据丢弃。主要问题是,人们不知道有效载荷未受保护部分中存在的比特错误级别。这使得人们很难或不可能确定是否能设计出有用的东西。有效载荷格式设计者不推荐考虑使用UDP Lite的不相等错误检测的特征,除非存在非常明确的要求。

There also exist some management and monitoring extensions.


RFC 2959: The RTP protocol Management Information Database (MIB) [RFC2959] that is used with SNMP [RFC3410] to configure and retrieve information about RTP sessions.

RFC 2959:RTP协议管理信息数据库(MIB)[RFC2959],与SNMP[RFC3410]一起使用,以配置和检索有关RTP会话的信息。

RFC 3611: The RTCP Extended Reports (RTCP XR) [RFC3611] consists of a framework for reports sent within RTCP. It can easily be extended by defining new report formats, which has and is occurring. The XRBLOCK WG in the IETF is chartered (at the time of writing) with defining new report formats. The list of specified formats is available in IANA's RTCP XR Block Type registry ( The report formats that are defined in RFC 3611 provide report information on packet loss, packet duplication, packet reception times, RTCP statistics summary, and VoIP Quality. [RFC3611] also defines a mechanism that allows receivers to calculate the RTT to other session participants when used.

RFC 3611:RTCP扩展报告(RTCP XR)[RFC3611]由RTCP内发送的报告框架组成。通过定义新的报告格式,可以很容易地对其进行扩展,而新的报告格式已经出现并且正在出现。IETF中的XRBLOCK工作组(在编写本文时)被授权定义新的报告格式。IANA的RTCP XR块类型注册表中提供了指定格式的列表( RFC 3611中定义的报告格式提供关于数据包丢失、数据包重复、数据包接收时间、RTCP统计摘要和VoIP质量的报告信息。[RFC3611]还定义了一种机制,允许接收者在使用时向其他会话参与者计算RTT。

RMONMIB: The Remote Network Monitoring WG has defined a mechanism [RFC3577] based on usage of the MIB that can be an alternative to RTCP XR.

RMONMIB:远程网络监控工作组根据MIB的使用情况定义了一种机制[RFC3577],该机制可以作为RTCP XR的替代方案。

A number of transport optimizations have also been developed for use in certain environments. They are all intended to be transparent and do not require special consideration by the RTP payload format writer. Thus, they are primarily listed here for informational reasons.


RFC 2508: "Compressing IP/UDP/RTP Headers for Low-Speed Serial Links" (CRTP) [RFC2508] is the first IETF-developed RTP header compression mechanism. It provides quite good compression; however, it has clear performance problems when subject to packet loss or reordering between compressor and decompressor.

RFC 2508:“压缩低速串行链路的IP/UDP/RTP报头”(CRTP)[RFC2508]是IETF开发的第一个RTP报头压缩机制。它提供了相当好的压缩;然而,当在压缩器和解压缩器之间发生数据包丢失或重新排序时,它有明显的性能问题。

RFCs 3095 and 5795: These are the base specifications of the robust header compression (ROHC) protocol version 1 [RFC3095] and version 2 [RFC5795]. This solution was created as a result of CRTP's lack of performance when compressed packets are subject to loss.

RFCs 3095和5795:这些是健壮报头压缩(ROHC)协议版本1[RFC3095]和版本2[RFC5795]的基本规范。此解决方案是由于CRTP在压缩数据包丢失时缺乏性能而创建的。

RFC 3545: Enhanced compressed RTP (E-CRTP) [RFC3545] was developed to provide extensions to CRTP that allow for better performance over links with long RTTs, packet loss, and/or reordering.

RFC 3545:开发增强型压缩RTP(E-CRTP)[RFC3545]是为了提供CRTP的扩展,允许在长RTT、丢包和/或重新排序的链路上实现更好的性能。

RFC 4170: "Tunneling Multiplexed Compressed RTP (TCRTP)" [RFC4170] is a solution that allows header compression within a tunnel carrying multiple multiplexed RTP flows. This is primarily used in voice trunking.

RFC 4170:“隧道多路复用压缩RTP(TCRTP)”[RFC4170]是一种解决方案,允许在承载多个多路复用RTP流的隧道内进行报头压缩。这主要用于语音中继。

There exist a couple of different security mechanisms that may be used with RTP. By definition, generic mechanisms are transparent for the RTP payload format and do not need special consideration by the format designer. The main reason that different solutions exist is that different applications have different requirements; thus, different solutions have been developed. For more discussion on this, please see "Options for Securing RTP Sessions" [RFC7201] and "Securing the RTP Framework: Why RTP Does Not Mandate a Single Media Security Solution" [RFC7202]. The main properties for an RTP security mechanism are to provide confidentiality for the RTP payload, integrity protection to detect manipulation of payload and headers, and source authentication. Not all mechanisms provide all of these features, a point that will need to be considered when a specific mechanisms is chosen.


The profile for Secure RTP - SRTP (RTP/SAVP) [RFC3711] and the derived profile (RTP/SAVPF [RFC5124]) are a solution that enables confidentiality, integrity protection, replay protection, and partial source authentication. It is the solution most commonly used with RTP at the time of writing this document. There exist several key-management solutions for SRTP, as well other choices, affecting the


security properties. For a more in-depth review of the options and solutions other than SRTP consult "Options for Securing RTP Sessions" [RFC7201].


3.3. Important RTP Details
3.3. 重要RTP细节

This section reviews a number of RTP features and concepts that are available in RTP, independent of the payload format. The RTP payload format can make use of these when appropriate, and even affect the behavior (RTP timestamp and marker bit), but it is important to note that not all features and concepts are relevant to every payload format. This section does not remove the necessity to read up on RTP. However, it does point out a few important details to remember when designing a payload format.


3.3.1. The RTP Session
3.3.1. RTP会话

The definition of the RTP session from RFC 3550 is:

RFC 3550中RTP会话的定义为:

An association among a set of participants communicating with RTP. A participant may be involved in multiple RTP sessions at the same time. In a multimedia session, each medium is typically carried in a separate RTP session with its own RTCP packets unless the encoding itself multiplexes multiple media into a single data stream. A participant distinguishes multiple RTP sessions by reception of different sessions using different pairs of destination transport addresses, where a pair of transport addresses comprises one network address plus a pair of ports for RTP and RTCP. All participants in an RTP session may share a common destination transport address pair, as in the case of IP multicast, or the pairs may be different for each participant, as in the case of individual unicast network addresses and port pairs. In the unicast case, a participant may receive from all other participants in the session using the same pair of ports, or may use a distinct pair of ports for each.


The distinguishing feature of an RTP session is that each session maintains a full, separate space of SSRC identifiers (defined next). The set of participants included in one RTP session consists of those that can receive an SSRC identifier transmitted by any one of the participants either in RTP as the SSRC or a CSRC (also defined below) or in RTCP. For example, consider a three-party conference implemented using unicast UDP with each participant receiving from the other two on separate port pairs. If each participant sends RTCP feedback about data received from one other participant only back to that participant, then the conference is composed of three separate point-to-point RTP sessions. If each participant provides RTCP feedback about its


reception of one other participant to both of the other participants, then the conference is composed of one multi-party RTP session. The latter case simulates the behavior that would occur with IP multicast communication among the three participants.


The RTP framework allows the variations defined here, but a particular control protocol or application design will usually impose constraints on these variations.


3.3.2. RTP Header
3.3.2. RTP报头

The RTP header contains a number of fields. Two fields always require additional specification by the RTP payload format, namely the RTP timestamp and the marker bit. Certain RTP payload formats also use the RTP sequence number to realize certain functionalities, primarily related to the order of their application data units. The payload type is used to indicate the used payload format. The SSRC is used to distinguish RTP packets from multiple senders and media sources identifying the RTP stream. Finally, [RFC5285] specifies how to transport payload format independent metadata relating to the RTP packet or stream.


Marker Bit: A single bit normally used to provide important indications. In audio, it is normally used to indicate the start of a talk burst. This enables jitter buffer adaptation prior to the beginning of the burst with minimal audio quality impact. In video, the marker bit is normally used to indicate the last packet part of a frame. This enables a decoder to finish decoding the picture, where it otherwise may need to wait for the next packet to explicitly know that the frame is finished.


Timestamp: The RTP timestamp indicates the time instance the media sample belongs to. For discrete media like video, it normally indicates when the media (frame) was sampled. For continuous media, it normally indicates the first time instance the media present in the payload represents. For audio, this is the sampling time of the first sample. All RTP payload formats must specify the meaning of the timestamp value and the clock rates allowed. Selecting a timestamp rate is an active design choice and is further discussed in Section 5.2.


Discontinuous Transmission (DTX) that is common among speech codecs, typically results in gaps or jumps in the timestamp values due to that there is no media payload to transmit and the next used timestamp value represent the actual sampling time of the data transmitted.


Sequence Number: The sequence number is monotonically increasing and is set as the packet is sent. This property is used in many payload formats to recover the order of everything from the whole stream down to fragments of application data units (ADUs) and the order they need to be decoded. Discontinuous transmissions do not result in gaps in the sequence number, as it is monotonically increasing for each sent RTP packet.


Payload Type: The payload type is used to indicate, on a per-packet basis, which format is used. The binding between a payload type number and a payload format and its configuration are dynamically bound and RTP session specific. The configuration information can be bound to a payload type value by out-of-band signaling (Section 3.4). An example of this would be video decoder configuration information. Commonly, the same payload type is used for a media stream for the whole duration of a session. However, in some cases it may be necessary to change the payload format or its configuration during the session.


SSRC: The synchronization source (SSRC) identifier is normally not used by a payload format other than to identify the RTP timestamp and sequence number space a packet belongs to, allowing simultaneously reception of multiple media sources. However, some of the RTP mechanisms for improving resilience to packet loss uses multiple SSRCs to separate original data and repair or redundant data, as well as multi-stream transmission of scalable codecs.


Header Extensions: RTP payload formats often need to include metadata relating to the payload data being transported. Such metadata is sent as a payload header, at the start of the payload section of the RTP packet. The RTP packet also includes space for a header extension [RFC5285]; this can be used to transport payload format independent metadata, for example, an SMPTE time code for the packet [RFC5484]. The RTP header extensions are not intended to carry headers that relate to a particular payload format, and must not contain information needed in order to decode the payload.


The remaining fields do not commonly influence the RTP payload format. The padding bit is worth clarifying as it indicates that one or more bytes are appended after the RTP payload. This padding must be removed by a receiver before payload format processing can occur. Thus, it is completely separate from any padding that may occur within the payload format itself.


3.3.3. RTP Multiplexing
3.3.3. RTP多路复用

RTP has three multiplexing points that are used for different purposes. A proper understanding of this is important to correctly use them.


The first one is separation of RTP streams of different types or usages, which is accomplished using different RTP sessions. So, for example, in the common multimedia session with audio and video, RTP commonly multiplexes audio and video in different RTP sessions. To achieve this separation, transport-level functionalities are used, normally UDP port numbers. Different RTP sessions can also be used to realize layered scalability as it allows a receiver to select one or more layers for multicast RTP sessions simply by joining the multicast groups over which the desired layers are transported. This separation also allows different Quality of Service (QoS) to be applied to different media types. Use of multiple transport flows has potential issues due to NAT and firewall traversal. The choices how one applies RTP sessions as well as transport flows can affect the transport properties an RTP media stream experiences.


The next multiplexing point is separation of different RTP streams within an RTP session. Here, RTP uses the SSRC to identify individual sources of RTP streams. An example of individual media sources would be the capture of different microphones that are carried in an RTP session for audio, independently of whether they are connected to the same host or different hosts. There also exist cases where a single media source, is transmitted using multiple RTP streams. For each SSRC, a unique RTP sequence number and timestamp space is used.


The third multiplexing point is the RTP header payload type field. The payload type identifies what format the content in the RTP payload has. This includes different payload format configurations, different codecs, and also usage of robustness mechanisms like the one described in RFC 2198 [RFC2198].

第三个多路复用点是RTP报头有效负载类型字段。有效负载类型标识RTP有效负载中内容的格式。这包括不同的有效负载格式配置、不同的编解码器,以及使用RFC 2198[RFC2198]中描述的健壮性机制。

3.3.4. RTP Synchronization
3.3.4. RTP同步

There are several types of synchronization, and we will here describe how RTP handles the different types:


Intra media: The synchronization within a media stream from a synchronization source (SSRC) is accomplished using the RTP timestamp field. Each RTP packet carries the RTP timestamp, which specifies the position in time of the media payload contained in this packet relative to the content of other RTP packets in the same RTP stream (i.e., a given SSRC). This is especially useful


in cases of discontinuous transmissions. Discontinuities can be caused by network conditions; when extensive losses occur the RTP timestamp tells the receiver how much later than previously received media the present media should be played out.


Inter-media: Applications commonly have a desire to use several media sources, possibly of different media types, at the same time. Thus, there exists a need to synchronize different media from the same endpoint. This puts two requirements on RTP: the possibility to determine which media are from the same endpoint and if they should be synchronized with each other and the functionality to facilitate the synchronization itself.


The first step in inter-media synchronization is to determine which SSRCs in each session should be synchronized with each other. This is accomplished by comparing the CNAME fields in the RTCP source description (SDES) packets. SSRCs with the same CNAME sent in any of multiple RTP sessions can be synchronized.


The actual RTCP mechanism for inter-media synchronization is based on the idea that each RTP stream provides a position on the media specific time line (measured in RTP timestamp ticks) and a common reference time line. The common reference time line is expressed in RTCP as a wall-clock time in the Network Time Protocol (NTP) format. It is important to notice that the wall-clock time is not required to be synchronized between hosts, for example, by using NTP [RFC5905]. It can even have nothing at all to do with the actual time; for example, the host system's up-time can be used for this purpose. The important factor is that all media streams from a particular source that are being synchronized use the same reference clock to derive their relative RTP timestamp time scales. The type of reference clock and its timebase can be signaled using RTP Clock Source Signaling [RFC7273].


Figure 1 illustrates how if one receives RTCP Sender Report (SR) packet P1 for one RTP stream and RTCP SR packet P2 for the other RTP stream, then one can calculate the corresponding RTP timestamp values for any arbitrary point in time T. However, to be able to do that, it is also required to know the RTP timestamp rates for each RTP stream currently used in the sessions.

图1说明了如果接收到一个RTP流的RTCP发送者报告(SR)数据包P1和另一个RTP流的RTCP SR数据包P2,则可以计算任意时间点T的相应RTP时间戳值。然而,为了能够做到这一点,还需要知道会话中当前使用的每个RTP流的RTP时间戳速率。

   TS1   --+---------------+------->
           |               |
          P1               |
           |               |
   NTP  ---+-----+---------T------>
                 |         |
                P2         |
                 |         |
   TS2  ---------+---------+---X-->
   TS1   --+---------------+------->
           |               |
          P1               |
           |               |
   NTP  ---+-----+---------T------>
                 |         |
                P2         |
                 |         |
   TS2  ---------+---------+---X-->

Figure 1: RTCP Synchronization


Assume that medium 1 uses an RTP timestamp clock rate of 16 kHz, and medium 2 uses a clock rate of 90 kHz. Then, TS1 and TS2 for point T can be calculated in the following way: TS1(T) = TS1(P1) + 16000 * (NTP(T)-NTP(P1)) and TS2(T) = TS2(P2) + 90000 * (NTP(T)-NTP(P2)). This calculation is useful as it allows the implementation to generate a common synchronization point for which all time values are provided (TS1(T), TS2(T) and T). So, when one wishes to calculate the NTP time that the timestamp value present in packet X corresponds to, one can do that in the following way: NTP(X) = NTP(T) + (TS2(X) - TS2(T))/90000.

假设介质1使用16 kHz的RTP时间戳时钟速率,介质2使用90 kHz的时钟速率。然后,点T的TS1和TS2可按以下方式计算:TS1(T)=TS1(P1)+16000*(NTP(T)-NTP(P1))和TS2(T)=TS2(P2)+90000*(NTP(T)-NTP(P2))。此计算非常有用,因为它允许实现生成一个公共同步点,为该点提供所有时间值(TS1(T)、TS2(T)和T)。因此,当希望计算分组X中存在的时间戳值对应的NTP时间时,可以通过以下方式进行:NTP(X)=NTP(T)+(TS2(X)-TS2(T))/90000。

Improved signaling for layered codecs and fast tune-in have been specified in "Rapid Synchronization for RTP Flows" [RFC6051].


Leap seconds are extra seconds added or seconds removed to keep our clocks in sync with the earth's rotation. Adding or removing seconds can impact the reference clock as discussed in "RTP and Leap Seconds" [RFC7164]; also, in cases where the RTP timestamp values are derived using the wall clock during the leap second event, errors can occur. Implementations need to consider leap seconds and should consider the recommendations in [RFC7164].


3.4. Signaling Aspects
3.4. 信号方面

RTP payload formats are used in the context of application signaling protocols such as SIP [RFC3261] using the Session Description Protocol (SDP) [RFC4566] with Offer/Answer [RFC3264], RTSP [RFC7826], or the Session Announcement Protocol [RFC2974]. These examples all use out-of-band signaling to indicate which type of RTP streams are desired to be used in the session and how they are configured. To be able to declare or negotiate the media format and RTP payload packetization, the payload format must be given an identifier. In addition to the identifier, many payload formats also have the need to signal further configuration information out-of-band for the RTP payloads prior to the media transport session.


The above examples of session-establishing protocols all use SDP, but other session description formats may be used. For example, there was discussion of a new XML-based session description format within the IETF (SDP-NG). In the end, the proposal did not get beyond draft protocol specification because of the enormous installed base of SDP implementations. However, to avoid locking the usage of RTP to SDP based out-of-band signaling, the payload formats are identified using a separate definition format for the identifier and associated parameters. That format is the media type.


3.4.1. Media Types
3.4.1. 媒体类型

Media types [RFC6838] are identifiers originally created for identifying media formats included in email. In this usage, they were known as MIME types, where the expansion of the MIME acronym includes the word "mail". The term "media type" was introduced to reflect a broader usage, which includes HTTP [RFC7231], Message Session Relay Protocol (MSRP) [RFC4975], and many other protocols to identify arbitrary content carried within the protocols. Media types also provide a media hierarchy that fits RTP payload formats well. Media type names are of two parts and consist of content type and sub-type separated with a slash, e.g., 'audio/PCMA' or 'video/ h263-2000'. It is important to choose the correct content-type when creating the media type identifying an RTP payload format. However, in most cases, there is little doubt what content type the format belongs to. Guidelines for choosing the correct media type and registration rules for media type names are provided in "Media Type Specifications and Registration Procedures" [RFC6838]. The additional rules for media types for RTP payload formats are provided in "Media Type Registration of RTP Payload Formats" [RFC4855].


Registration of the RTP payload name is something that is required to avoid name collision in the future. Note that "x-" names are not suitable for any documented format as they have the same problem with name collision and can't be registered. The list of already-registered media types can be found at <>.


Media types are allowed any number of parameters, which may be required or optional for that media type. They are always specified on the form "name=value". There exist no restrictions on how the value is defined from the media type's perspective, except that parameters must have a value. However, the usage of media types in


SDP, etc., has resulted in the following restrictions that need to be followed to make media types usable for RTP-identifying payload formats:


1. Arbitrary binary content in the parameters is allowed, but it needs to be encoded so that it can be placed within text-based protocols. Base64 [RFC4648] is recommended, but for shorter content Base16 [RFC4648] may be more appropriate as it is simpler to interpret for humans. This needs to be explicitly stated when defining a media type parameter with binary values.

1. 允许参数中包含任意二进制内容,但需要对其进行编码,以便将其放入基于文本的协议中。建议使用Base64[RFC4648],但对于较短的内容,Base16[RFC4648]可能更合适,因为它更容易为人类解释。在使用二进制值定义媒体类型参数时,需要明确说明这一点。

2. The end of the value needs to be easily found when parsing a message. Thus, parameter values that are continuous and not interrupted by common text separators, such as space and semicolon characters, are recommended. If that is not possible, some type of escaping should be used. Usage of quote (") is recommended; do not forget to provide a method of encoding any character used for quoting inside the quoted element.

2. 解析消息时,需要轻松找到值的结尾。因此,建议使用连续且不被常用文本分隔符(如空格和分号字符)打断的参数值。如果不可能,则应使用某种类型的转义。建议使用引号(“);不要忘记提供一种在引号元素内对用于引号的任何字符进行编码的方法。

3. A common representation form for the media type and its parameters is on a single line. In that case, the media type is followed by a semicolon-separated list of the parameter value pairs, e.g.:

3. 媒体类型及其参数的通用表示形式在一行上。在这种情况下,媒体类型后面是以分号分隔的参数值对列表,例如:

       audio/amr octet-align=0; mode-set=0,2,5,7; mode-change-period=2
       audio/amr octet-align=0; mode-set=0,2,5,7; mode-change-period=2
3.4.2. Mapping to SDP
3.4.2. 映射到SDP

Since SDP [RFC4566] is so commonly used as an out-of-band signaling protocol, a mapping of the media type into SDP exists. The details on how to map the media type and its parameters into SDP are described in [RFC4855]. However, this is not sufficient to explain how certain parameters must be interpreted, for example, in the context of Offer/Answer negotiation [RFC3264].

由于SDP[RFC4566]通常用作带外信令协议,因此存在媒体类型到SDP的映射。有关如何将媒体类型及其参数映射到SDP的详细信息,请参见[RFC4855]。然而,这不足以解释某些参数必须如何解释,例如,在要约/应答协商[RFC3264]的上下文中。 The Offer/Answer Model 提供/应答模型

The Offer/Answer (O/A) model allows SIP to negotiate which media formats and payload formats are to be used in a session and how they are to be configured. However, O/A does not define a default behavior; instead, it points out the need to define how parameters behave. To make things even more complex, the direction of media within a session has an impact on these rules, so that some cases may require separate descriptions for RTP streams that are send-only, receive-only, or both sent and received as identified by the SDP attributes a=sendonly, a=recvonly, and a=sendrecv. In addition, the usage of multicast adds further limitations as the same RTP stream is


delivered to all participants. If those multicast-imposed restrictions are too limiting for unicast, then separate rules for unicast and multicast will be required.


The simplest and most common O/A interpretation is that a parameter is defined to be declarative; i.e., the SDP Offer/Answer sending agent can declare a value and that has no direct impact on the other agent's values. This declared value applies to all media that are going to be sent to the declaring entity. For example, most video codecs have a level parameter that tells the other participants the highest complexity the video decoder supports. The level parameter can be declared independently by two participants in a unicast session as it will be the media sender's responsibility to transmit a video stream that fulfills the limitation the other side has declared. However, in multicast, it will be necessary to send a stream that follows the limitation of the weakest receiver, i.e., the one that supports the lowest level. To simplify the negotiation in these cases, it is common to require any answerer to a multicast session to take a yes or no approach to parameters.

最简单也是最常见的O/A解释是将参数定义为声明性的;i、 例如,SDP提供/应答发送代理可以声明一个值,该值对其他代理的值没有直接影响。此声明的值适用于将要发送到声明实体的所有媒体。例如,大多数视频编解码器都有一个级别参数,告诉其他参与者视频解码器支持的最高复杂度。级别参数可以由单播会话中的两个参与者独立声明,因为媒体发送方负责传输满足另一方声明的限制的视频流。然而,在多播中,必须发送一个遵循最弱接收器限制的流,即支持最低级别的接收器。为了简化这些情况下的协商,通常要求多播会话的任何应答者对参数采取是或否的方法。

A "negotiated" parameter is a different case, for which both sides need to agree on its value. Such a parameter requires the answerer to either accept it as it is offered or remove the payload type the parameter belonged to from its answer. The removal of the payload type from the answer indicates to the offerer the lack of support for the parameter values presented. An unfortunate implication of the need to use complete payload types to indicate each possible configuration so as to maximize the chances of achieving interoperability, is that the number of necessary payload types can quickly grow large. This is one reason to limit the total number of sets of capabilities that may be implemented.


The most problematic type of parameters are those that relate to the media the entity sends. They do not really fit the O/A model, but can be shoehorned in. Examples of such parameters can be found in the H.264 video codec's payload format [RFC6184], where the name of all parameters with this property starts with "sprop-". The issue with these parameters is that they declare properties for a RTP stream that the other party may not accept. The best one can make of the situation is to explain the assumption that the other party will accept the same parameter value for the media it will receive as the offerer of the session has proposed. If the answerer needs to change any declarative parameter relating to streams it will receive, then the offerer may be required to make a new offer to update the parameter values for its outgoing RTP stream.


Another issue to consider is the send-only RTP streams in offers. Parameters that relate to what the answering entity accepts to receive have no meaning other than to provide a template for the answer. It is worth pointing out in the specification that these really provide a set of parameter values that the sender recommends. Note that send-only streams in answers will need to indicate the offerer's parameters to ensure that the offerer can match the answer to the offer.


A further issue with Offer/Answer that complicates things is that the answerer is allowed to renumber the payload types between offer and answer. This is not recommended, but allowed for support of gateways to the ITU conferencing suite. This means that it must be possible to bind answers for payload types to the payload types in the offer even when the payload type number has been changed, and some of the proposed payload types have been removed. This binding must normally be done by matching the configurations originally offered against those in the answer. This may require specification in the payload format of which parameters that constitute a configuration, for example, as done in Section 8.2.2 of the H.264 RTP Payload format [RFC6184], which states: "The parameters identifying a media format configuration for H.264 are profile-level-id and packetization-mode".

要约/应答的另一个使事情复杂化的问题是,应答者可以在要约和应答之间重新编号有效负载类型。不建议这样做,但允许支持到ITU会议套件的网关。这意味着,即使有效负载类型编号已更改,并且某些建议的有效负载类型已删除,也必须能够将有效负载类型的答案绑定到报价中的有效负载类型。这种绑定通常必须通过将最初提供的配置与应答中的配置进行匹配来完成。这可能需要有效载荷格式的规范,其中的参数构成配置,例如,如H.264 RTP有效载荷格式[RFC6184]第8.2.2节所述,其中规定:“识别H.264媒体格式配置的参数是配置文件级别id和打包模式”。 Declarative Usage in RTSP and SAP RTSP和SAP中的声明性用法

SAP (Session Announcement Protocol) [RFC2974] was experimentally used for announcing multicast sessions. Similar but better protocols are using SDP in a declarative style to configure multicast-based applications. Independently of the usage of Source-Specific Multicast (SSM) [RFC3569] or Any-Source Multicast (ASM), the SDP provided by these configuration delivery protocols applies to all participants. All media that is sent to the session must follow the RTP stream definition as specified by the SDP. This enables everyone to receive the session if they support the configuration. Here, SDP provides a one-way channel with no possibility to affect the configuration that the session creator has decided upon. Any RTP payload format that requires parameters for the send direction and that needs individual values per implementation or instance will fail in a SAP session for a multicast session allowing anyone to send.

SAP(Session Announcement Protocol)[RFC2974]被实验性地用于宣布多播会话。类似但更好的协议使用声明式SDP来配置基于多播的应用程序。独立于源特定多播(SSM)[RFC3569]或任何源多播(ASM)的使用,这些配置交付协议提供的SDP适用于所有参与者。发送到会话的所有媒体必须遵循SDP指定的RTP流定义。这使每个人都能够接收会话(如果他们支持该配置)。这里,SDP提供单向通道,不可能影响会话创建者决定的配置。对于允许任何人发送的多播会话,任何需要发送方向参数且每个实现或实例需要单个值的RTP有效负载格式都将在SAP会话中失败。

Real-Time Streaming Protocol (RTSP) [RFC7826] allows the negotiation of transport parameters for RTP streams that are part of a streaming session between a server and client. RTSP has divided the transport parameters from the media configuration. SDP is commonly used for media configuration in RTSP and is sent to the client prior to session establishment, either through use of the DESCRIBE method or


by means of an out-of-band channel like HTTP, email, etc. The SDP is used to determine which RTP streams and what formats are being used prior to session establishment.


Thus, both SAP and RTSP use SDP to configure receivers and senders with a predetermined configuration for a RTP stream including the payload format and any of its parameters. All parameters are used in a declarative fashion. This can result in different treatment of parameters between Offer/Answer and declarative usage in RTSP and SAP. Any such difference will need to be spelled out by the payload format specification.


3.5. Transport Characteristics
3.5. 传输特性

The general channel characteristics that RTP flows experience are documented in Section 3 of "Guidelines for Writers of RTP Payload Format Specifications" [RFC2736]. The discussion below provides additional information.


3.5.1. Path MTU
3.5.1. 路径MTU

At the time of writing, the most common IP Maximum Transmission Unit (MTU) in commonly deployed link layers is 1500 bytes (Ethernet data payload). However, there exist both links with smaller MTUs and links with much larger MTUs. An example for links with small MTU size is older generation cellular links. Certain parts of the Internet already support an IP MTU of 8000 bytes or more, but these are limited islands. The most likely places to find MTUs larger than 1500 bytes are within enterprise networks, university networks, data centers, storage networks, and over high capacity (10 Gbps or more) links. There is a slow, ongoing evolution towards larger MTU sizes. However, at the same time, it has become common to use tunneling protocols, often multiple ones, whose overhead when added together can shrink the MTU significantly. Thus, there exists a need both to consider limited MTUs as well as enable support of larger MTUs. This should be considered in the design, especially in regard to features such as aggregation of independently decodable data units.

在撰写本文时,通常部署的链路层中最常见的IP最大传输单元(MTU)为1500字节(以太网数据有效负载)。然而,存在具有较小MTU的链路和具有更大MTU的链路。具有较小MTU大小的链路的一个示例是老一代蜂窝链路。互联网的某些部分已经支持8000字节或更多的IP MTU,但这些都是有限的孤岛。在企业网络、大学网络、数据中心、存储网络和超大容量(10 Gbps或以上)链路中,最有可能找到大于1500字节的MTU。向更大的MTU尺寸发展是一个缓慢、持续的过程。然而,同时,使用隧道协议(通常是多个协议)已变得很普遍,当将这些协议的开销加在一起时,可以显著减少MTU。因此,既需要考虑有限的MTU,也允许支持更大的MTU。在设计中应考虑这一点,尤其是在诸如独立可解码数据单元的聚合等特性方面。

3.5.2. Different Queuing Algorithms
3.5.2. 不同的排队算法

Routers and switches on the network path between an IP sender and a particular receiver can exhibit different behaviors affecting the end-to-end characteristics. One of the more important aspects of this is queuing behavior. Routers and switches have some amount of queuing to handle temporary bursts of data that designated to leave the switch or router on the same egress link. A queue, when not empty, results in an increased path delay.


The implementation of the queuing affects the delay and also how congestion signals (Explicit Congestion Notification (ECN) [RFC6679] or packet drops) are provided to the flow. The other aspects are if the flow shares the queue with other flows and how the implementation affects the flow interaction. This becomes important, for example, when real-time flows interact with long-lived TCP flows. TCP has a built-in behavior in its congestion control that strives to fill the buffer; thus, all flows sharing the buffer experienced the delay build up.


A common, but quite poor, queue-handling mechanism is tail-drop, i.e., only drop packets when the incoming packet doesn't fit in the queue. If a bad queuing algorithm is combined with too much queue space, the queuing time can grow to be very significant and can even become multiple seconds. This is called "bufferbloat" [BLOAT]. Active Queue Management (AQM) is a term covering mechanisms that try to do something smarter by actively managing the queue, for example, sending congestion signals earlier by dropping packets earlier in the queue. The behavior also affects the flow interactions. For example, Random Early Detection (RED) [RED] selects which packet(s) to drop randomly. This gives flows that have more packets in the queue a higher probability to experience the packet loss (congestion signal). There is ongoing work in the IETF WG AQM to find suitable mechanisms to recommend for implementation and reduce the use of tail-drop.


3.5.3. Quality of Service
3.5.3. 服务质量

Using best-effort Internet has no guarantees for the path's properties. QoS mechanisms are intended to provide the possibility to bound the path properties. Where Diffserv [RFC2475] markings affect the queuing and forwarding behaviors of routers, the mechanism provides only statistical guarantees and care in how much marked packets of different types that are entering the network. Flow-based QoS, like IntServ [RFC1633], has the potential for stricter guarantees as the properties are agreed on by each hop on the path, at the cost of per-flow state in the network.


4. Standardization Process for an RTP Payload Format
4. RTP有效负载格式的标准化过程

This section discusses the recommended process to produce an RTP payload format in the described venues. This is to document the best current practice on how to get a well-designed and specified payload format as quickly as possible. For specifications that are defined by standards bodies other than the IETF, the primary milestone is the registration of the media type for the RTP payload format. For


proprietary media formats, the primary goal depends on whether interoperability is desired at the RTP level. However, there is also the issue of ensuring best possible quality of any specification.


4.1. IETF
4.1. IETF

For all standardized media formats, it is recommended that the payload format be specified in the IETF. The main reason is to provide an openly available RTP payload format specification that has been reviewed by people experienced with RTP payload formats. At the time of writing, this work is done in the PAYLOAD Working Group (WG), but that may change in the future.


4.1.1. Steps from Idea to Publication
4.1.1. 从构思到出版的步骤

There are a number of steps that an RTP payload format should go through from the initial idea until it is published. This also documents the process that the PAYLOAD WG applies when working with RTP payload formats.


Idea: Determine the need for an RTP payload format as an IETF specification.


Initial effort: Using this document as a guideline, one should be able to get started on the work. If one's media codec doesn't fit any of the common design patterns or one has problems understanding what the most suitable way forward is, then one should contact the PAYLOAD WG and/or the WG Chairs. The goal of this stage is to have an initial individual draft. This draft needs to focus on the introductory parts that describe the real-time media format and the basic idea on how to packetize it. Not all the details are required to be filled in. However, the security chapter is not something that one should skip, even initially. From the start, it is important to consider any serious security risks that need to be solved. The first step is completed when one has a draft that is sufficiently detailed for a first review by the WG. The less confident one is of the solution, the less work should be spent on details; instead, concentrate on the codec properties and what is required to make the packetization work.


Submission of the first version: When one has performed the above, one submits the draft as an individual draft ( This can be done at any time, except for a period prior to an IETF meeting (see important dates related to the next IETF meeting for draft submission cutoff date). When the Internet-Draft announcement has been sent out on

提交第一版:完成上述工作后,将草案作为单独草案提交( 这可以在任何时候进行,IETF会议之前的一段时间除外(草案提交截止日期见与下一次IETF会议相关的重要日期)。当互联网公告草案于

the draft announcement list (, forward it to the PAYLOAD WG ( and request that it be reviewed. In the email, outline any issues the authors currently have with the design.


Iterative improvements: Taking the feedback received into account, one updates the draft and tries resolve issues. New revisions of the draft can be submitted at any time (again except for a short period before meetings). It is recommended to submit a new version whenever one has made major updates or has new issues that are easiest to discuss in the context of a new draft version.


Becoming a WG document: Given that the definition of RTP payload formats is part of the PAYLOAD WG's charter, RTP payload formats that are going to be published as Standards Track RFCs need to become WG documents. Becoming a WG document means that the WG Chairs or an appointed document shepherd are responsible for administrative handling, for example, issuing publication requests. However, be aware that making a document into a WG document changes the formal ownership and responsibility from the individual authors to the WG. The initial authors normally continue being the document editors, unless unusual circumstances occur. The PAYLOAD WG accepts new RTP payload formats based on their suitability and document maturity. The document maturity is a requirement to ensure that there are dedicated document editors and that there exists a good solution.


Iterative improvements: The updates and review cycles continue until the draft has reached the level of maturity suitable for publication. The authors are responsible for judging when the document is ready for the next step, most likely WG Last Call, but they can ask the WG chairs or Shepherd.


WG Last Call: A WG Last Call of at least two weeks is always performed for payload formats in the PAYLOAD WG (see Section 7.4 of [RFC2418]). The authors request WG Last Call for a draft when they think it is mature enough for publication. The WG Chairs or shepherd perform a review to check if they agree with the authors' assessment. If the WG Chairs or shepherd agree on the maturity, the WG Last Call is announced on the WG mailing list. If there are issues raised, these need to be addressed with an updated draft version. For any more substantial changes to the draft, a new WG Last Call is announced for the updated version. Minor changes, like editorial fixes, can be progressed without an additional WG Last Call.

WG Last Call(工作组最后调用):对于有效负载工作组中的有效负载格式,始终执行至少两周的工作组最后调用(见[RFC2418]第7.4节)。当作者认为草稿已经足够成熟,可以出版时,他们要求工作组最后一次征求意见稿。工作组主席或shepherd进行审查,以检查他们是否同意作者的评估。如果工作组主席或shepherd同意到期日,工作组最后一次通话将在工作组邮件列表中公布。如果提出了问题,则需要通过更新的草案版本解决这些问题。对于草案的任何实质性更改,将宣布更新版本的新工作组最后通知。微小的更改,如编辑修复,无需额外的WG Last Call即可进行。

Publication requested: For WG documents, the WG Chairs or shepherd request publication of the draft after it has passed WG Last Call. After this, the approval and publication process described in BCP 9 [BCP9] is performed. The status after the publication has been requested can be tracked using the IETF Datatracker [TRACKER]. Documents do not expire as they normally do after publication has been requested, so authors do not have to issue keep-alive updates. In addition, any submission of document updates requires the approval of WG Chair(s). The authors are commonly asked to address comments or issues raised by the IESG. The authors also do one last review of the document immediately prior to its publication as an RFC to ensure that no errors or formatting problems have been introduced during the publication process.

要求公布:对于工作组文件,工作组主席或shepherd在通过工作组最后一次会议后要求公布草案。在此之后,执行BCP 9[BCP9]中描述的批准和发布流程。可以使用IETF数据跟踪器[TRACKER]跟踪请求发布后的状态。文档不会像通常在请求发布后那样过期,因此作者不必发布保持活动状态的更新。此外,任何文件更新的提交都需要工作组主席的批准。作者通常被要求回答IESG提出的评论或问题。作者还将在文件作为RFC发布之前对其进行最后一次审查,以确保在发布过程中不会出现错误或格式问题。

4.1.2. WG Meetings
4.1.2. 工作组会议

WG meetings are for discussing issues, not presentations. This means that most RTP payload formats should never need to be discussed in a WG meeting. RTP payload formats that would be discussed are either those with controversial issues that failed to be resolved on the mailing list or those including new design concepts worth a general discussion.


There exists no requirement to present or discuss a draft at a WG meeting before it becomes published as an RFC. Thus, even authors who lack the possibility to go to WG meetings should be able to successfully specify an RTP payload format in the IETF. WG meetings may become necessary only if the draft gets stuck in a serious debate that cannot easily be resolved.


4.1.3. Draft Naming
4.1.3. 草案命名

To simplify the work of the PAYLOAD WG Chairs and WG members, a specific Internet-Draft file-naming convention shall be used for RTP payload formats. Individual submissions shall be named using the template: draft-<lead author family name>-payload-rtp-<descriptive name>-<version>. The WG documents shall be named according to this template: draft-ietf-payload-rtp-<descriptive name>-<version>. The inclusion of "payload" in the draft file name ensures that the search for "payload-" will find all PAYLOAD-related drafts. Inclusion of "rtp" tells us that it is an RTP payload format draft. The descriptive name should be as short as possible while still describing what the payload format is for. It is recommended to use the media format or codec abbreviation. Please note that the version must start at 00 and is increased by one for each submission to the IETF secretary of the draft. No version numbers may be skipped. For more details on draft naming, please see Section 7 of [ID-GUIDE].


4.1.4. Writing Style
4.1.4. 写作风格

When writing an Internet-Draft for an RTP payload format, one should observe some few considerations (that may be somewhat divergent from the style of other IETF documents and/or the media coding spec's author group may use):


Include Motivations: In the IETF, it is common to include the motivation for why a particular design or technical path was chosen. These are not long statements: a sentence here and there explaining why suffice.


Use the Defined Terminology: There exists defined terminology both in RTP and in the media codec specification for which the RTP payload format is designed. A payload format specification needs to use both to make clear the relation of features and their functions. It is unwise to introduce or, worse, use without introduction, terminology that appears to be more accessible to average readers but may miss certain nuances that the defined terms imply. An RTP payload format author can assume the reader to be reasonably familiar with the terminology in the media coding specification.


Keeping It Simple: The IETF has a history of specifications that are focused on their main usage. Historically, some RTP payload formats have a lot of modes and features, while the actual deployments have only included the most basic features that had very clear requirements. Time and effort can be saved by focusing on only the most important use cases and keeping the solution simple. An extension mechanism should be provided to enable backward-compatible extensions, if that is an organic fit.


Normative Requirements: When writing specifications, there is commonly a need to make it clear when something is normative and at what level. In the IETF, the most common method is to use "Key words for use in RFCs to Indicate Requirement Levels" [RFC2119], which defines the meaning of "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL".


4.1.5. How to Speed Up the Process
4.1.5. 如何加快这一进程

There a number of ways to lose a lot of time in the above process. This section discusses what to do and what to avoid.


o Do not update the draft only for the meeting deadline. An update to each meeting automatically limits the draft to three updates per year. Instead, ignore the meeting schedule and publish new versions as soon as possible.

o 不要仅在会议截止日期前更新草案。每次会议的更新会自动将草稿限制为每年三次更新。相反,请忽略会议日程安排,并尽快发布新版本。

o Try to avoid requesting reviews when people are busy, like the few weeks before a meeting. It is actually more likely that people have time for them directly after a meeting.

o 当人们很忙时,尽量避免要求评论,比如开会前的几周。事实上,人们更有可能在会议结束后有时间与他们交谈。

o Perform draft updates quickly. A common mistake is that the authors let the draft slip. By performing updates to the draft text directly after getting resolution on an issue, things speed up. This minimizes the delay that the author has direct control over. The time taken for reviews, responses from Area Directors and WG Chairs, etc., can be much harder to speed up.

o 快速执行草稿更新。一个常见的错误是作者疏忽了草稿。通过在问题得到解决后直接对草案文本进行更新,事情会加快。这样可以最大限度地减少作者直接控制的延迟。审查、区域主管和工作组主席的回应等所需的时间可能更难加快。

o Do not fail to take human nature into account. It happens that people forget or need to be reminded about tasks. Send a kind reminder to the people you are waiting for if things take longer than expected. Ask people to estimate when they expect to fulfill the requested task.

o 不要不考虑人性。人们往往会忘记或需要提醒任务。如果事情比预期的时间长,给你正在等待的人发一个善意的提醒。让员工估计他们期望何时完成要求的任务。

o Ensure there is enough review. It is common that documents take a long time and many iterations because not enough review is performed in each iteration. To improve the amount of review you get on your own document, trade review time with other document authors. Make a deal with some other document author that you will review their draft if they review yours. Even inexperienced reviewers can help with language, editorial, or clarity issues. Also, try approaching the more experienced people in the WG and getting them to commit to a review. The WG Chairs cannot, even if desirable, be expected to review all versions. Due to workload, the Chairs may need to concentrate on key points in a draft evolution like checking on initial submissions, a draft's readiness to become a WG document, or its readiness for WG Last Call.

o 确保有足够的审查。文档花费很长时间和多次迭代是很常见的,因为在每次迭代中没有执行足够的评审。要提高您对自己文档的审阅量,请与其他文档作者交换审阅时间。与其他文档作者达成协议,如果他们审阅您的草稿,您将审阅他们的草稿。即使是没有经验的评论员也可以帮助解决语言、编辑或清晰度问题。此外,试着与工作组中更有经验的人接触,让他们承诺进行审查。即使可取,工作组主席也不能期望审查所有版本。由于工作量,主席们可能需要将注意力集中在草案演变过程中的关键点上,如检查初始提交的文件、草案是否准备好成为工作组文件或是否准备好工作组最后一次电话会议。

4.2. Other Standards Bodies
4.2. 其他标准机构

Other standards bodies may define RTP payloads in their own specifications. When they do this, they are strongly recommended to contact the PAYLOAD WG Chairs and request review of the work. It is recommended that at least two review steps are performed. The first


should be early in the process when more fundamental issues can be easily resolved without abandoning a lot of effort. Then, when nearing completion, but while it is still possible to update the specification, a second review should be scheduled. In that pass, the quality can be assessed; hopefully, no updates will be needed. Using this procedure can avoid both conflicting definitions and serious mistakes, like breaking certain aspects of the RTP model.


RTP payload media types may be registered in the standards tree by other standards bodies. The requirements on the organization are outlined in the media types registration documents [RFC4855] and [RFC6838]). This registration requires a request to the IESG, which ensures that the filled-in registration template is acceptable. To avoid last-minute problems with these registrations the registration template must be sent for review both to the PAYLOAD WG and the media types list ( and is something that should be included in the IETF reviews of the payload format specification.


4.3. Proprietary and Vendor Specific
4.3. 专有和供应商专用

Proprietary RTP payload formats are commonly specified when the real-time media format is proprietary and not intended to be part of any standardized system. However, there are reasons why also proprietary formats should be correctly documented and registered:


o Usage in a standardized signaling environment, such as SIP/SDP. RTP needs to be configured with the RTP profiles, payload formats, and their payload types being used. To accomplish this, it is desirable to have registered media type names to ensure that the names do not collide with those of other formats.

o 在标准化信令环境(如SIP/SDP)中的使用。RTP需要配置RTP配置文件、有效负载格式以及所使用的有效负载类型。为此,需要注册媒体类型名称,以确保名称不会与其他格式的名称冲突。

o Sharing with business partners. As RTP payload formats are used for communication, situations often arise where business partners would like to support a proprietary format. Having a well-written specification of the format will save time and money for both parties, as interoperability will be much easier to accomplish.

o 与业务伙伴共享。由于RTP有效负载格式用于通信,通常会出现业务合作伙伴希望支持专有格式的情况。有一个编写良好的格式规范将为双方节省时间和金钱,因为互操作性将更容易实现。

o To ensure interoperability between different implementations on different platforms.

o 确保不同平台上不同实现之间的互操作性。

To avoid name collisions, there is a central registry keeping track of the registered media type names used by different RTP payload formats. When it comes to proprietary formats, they should be registered in the vendor's own tree. All vendor-specific registrations use sub-type names that start with "vnd.<vendor-name>". Names in the vendor's own tree are not required to be registered with IANA. However, registration [RFC6838] is recommended if the media type is used at all in public environments.

为了避免名称冲突,有一个中央注册表,用于跟踪不同RTP有效负载格式使用的已注册媒体类型名称。当涉及到专有格式时,它们应该在供应商自己的树中注册。所有特定于供应商的注册都使用以“vnd.<vendor name>”开头的子类型名称。供应商自身树中的名称不需要向IANA注册。但是,如果媒体类型在公共环境中使用,则建议注册[RFC6838]。

If interoperability at the RTP level is desired, a payload type specification should be standardized in the IETF following the process described above. The IETF does not require full disclosure of the codec when defining an RTP payload format to carry that codec, but a description must be provided that is sufficient to allow the IETF to judge whether the payload format is well designed. The media type identifier assigned to a standardized payload format of this sort will lie in the standards tree rather than the vendor tree.


4.4. Joint Development of Media Coding Specification and RTP Payload Format

4.4. 媒体编码规范和RTP有效载荷格式的联合开发

In the last decade, there have been a few cases where the media codec and the associated RTP payload format have been developed concurrently and jointly. Developing the two specs not only concurrently but also jointly, in close cooperation with the group developing the media codec, allows one to leverage the benefits joint source/channel coding can provide. Doing so has historically resulted in well-performing payload formats and in success of both the media coding specification and associated RTP payload format. Insofar, whenever the opportunity presents it, it may be useful to closely keep the media coding group in the loop (through appropriate liaison means whatever those may be) and influence the media coding specification to be RTP friendly. One example for such a media coding specification is H.264, where the RTP payload header co-serves as the H.264 NAL unit header and vice versa, and is documented in both specifications.

在过去的十年中,有一些情况下,媒体编解码器和相关的RTP有效载荷格式被同时和联合开发。与开发媒体编解码器的团队密切合作,不仅同时而且联合开发这两个规范,可以利用联合信源/信道编码可以提供的好处。这样做在历史上已经产生了性能良好的有效负载格式,并成功地实现了媒体编码规范和相关的RTP有效负载格式。到目前为止,只要有机会,密切联系媒体编码组(通过适当的联络方式,无论这些联络方式是什么)并影响媒体编码规范,使其对RTP友好可能是有用的。这种媒体编码规范的一个示例是H.264,其中RTP有效负载报头共同用作H.264 NAL单元报头,反之亦然,并且在两个规范中都有记录。

5. Designing Payload Formats
5. 设计有效负载格式

The best summary of payload format design is KISS (Keep It Simple, Stupid). A simple payload format is easier to review for correctness, easier to implement, and has low complexity. Unfortunately, contradictory requirements sometimes make it hard to do things simply. Complexity issues and problems that occur for RTP payload formats are:


Too many configurations: Contradictory requirements lead to the result that one configuration is created for each conceivable case. Such contradictory requirements are often between functionality and bandwidth. This outcome has two big disadvantages; First all configurations need to be implemented. Second, the user application must select the most suitable configuration. Selecting the best configuration can be very difficult and, in negotiating applications, this can create interoperability problems. The recommendation is to try to select


a very limited set of configurations (preferably one) that perform well for the most common cases and are capable of handling the other cases, but maybe not that well.


Hard to implement: Certain payload formats may become difficult to implement both correctly and efficiently. This needs to be considered in the design.


Interaction with general mechanisms: Special solutions may create issues with deployed tools for RTP, such as tools for more robust transport of RTP. For example, a requirement for an unbroken sequence number space creates issues for mechanisms relying on payload type switching interleaving media-independent resilience within a stream.


5.1. Features of RTP Payload Formats
5.1. RTP有效负载格式的特点

There are a number of common features in RTP payload formats. There is no general requirement to support these features; instead, their applicability must be considered for each payload format. In fact, it may be that certain features are not even applicable.


5.1.1. Aggregation
5.1.1. 聚集

Aggregation allows for the inclusion of multiple Application Data Units (ADUs) within the same RTP payload. This is commonly supported for codecs that produce ADUs of sizes smaller than the IP MTU. One reason for the use of aggregation is the reduction of header overhead (IP/UDP/RTP headers). When setting into relation the ADU size and the MTU size, do remember that the MTU may be significantly larger than 1500 bytes. An MTU of 9000 bytes is available today and an MTU of 64k may be available in the future. Many speech codecs have the property of ADUs of a few fixed sizes. Video encoders may generally produce ADUs of quite flexible sizes. Thus, the need for aggregation may be less. But some codecs produce small ADUs mixed with large ones, for example, H.264 Supplemental Enhancement Information (SEI) messages. Sending individual SEI message in separate packets are not efficient compared to combing the with other ADUs. Also, some small ADUs are, within the media domain, semantically coupled to the larger ADUs (for example, in-band parameter sets in H.264 [RFC6184]). In such cases, aggregation is sensible, even if not required from a payload/header overhead viewpoint. There also exist cases when the ADUs are pre-produced and can't be adopted to a specific networks MTU. Instead, their packetization needs to be adopted to the network. All above factors should be taken into account when deciding on the inclusion of aggregation, and weighting its benefits

聚合允许在同一RTP负载中包含多个应用程序数据单元(ADU)。这通常适用于产生小于IP MTU的ADU的编解码器。使用聚合的一个原因是减少了报头开销(IP/UDP/RTP报头)。当设置ADU大小和MTU大小的关系时,请记住MTU可能明显大于1500字节。现在可以使用9000字节的MTU,将来可能可以使用64k的MTU。许多语音编解码器都具有ADU的特性,只有几个固定大小。视频编码器通常可以产生尺寸相当灵活的ADU。因此,对聚合的需求可能更少。但是一些编解码器会产生小ADU和大ADU的混合,例如,H.264补充增强信息(SEI)消息。与与其他ADU组合相比,在单独的数据包中发送单个SEI消息并不高效。此外,在媒体域内,一些小的adu在语义上耦合到较大的adu(例如,H.264[RFC6184]中的带内参数集)。在这种情况下,聚合是合理的,即使从负载/报头开销的角度来看不需要聚合。也存在ADU是预先生产的,无法用于特定网络MTU的情况。相反,需要将其打包到网络中。在决定是否包含聚合并对其效益进行加权时,应考虑上述所有因素

against the complexity of defining them (which can be significant especially when aggregation is performed over ADUs with different playback times).


The main disadvantage of aggregation, beyond implementation complexity, is the extra delay introduced (due to buffering until a sufficient number of ADUs have been collected at the sender) and reduced robustness against packet loss. Aggregation also introduces buffering requirements at the receiver.


5.1.2. Fragmentation
5.1.2. 碎裂

If the real-time media format has the property that it may produce ADUs that are larger than common MTU sizes, then fragmentation support should be considered. An RTP payload format may always fall back on IP fragmentation; however, as discussed in RFC 2736, this has some drawbacks. Perhaps the most important reason to avoid IP fragmentation is that IP fragmented packets commonly are discarded in the network, especially by NATs or firewalls. The usage of fragmentation at the RTP payload format level allows for more efficient usage of RTP packet loss recovery mechanisms. It may also in some cases also allow better usage of partial ADUs by doing media specific fragmentation at media-specific boundaries. In use cases where the ADUs are pre-produced and can't be adopted to the network's MTU size, support for fragmentation can be crucial.

如果实时媒体格式具有这样的特性,即它可能会产生大于普通MTU大小的ADU,则应考虑碎片支持。RTP有效负载格式可能总是依赖于IP碎片;然而,正如RFC 2736中所讨论的,这有一些缺点。避免IP碎片化的最重要原因可能是,IP碎片化的数据包通常会在网络中被丢弃,尤其是被NAT或防火墙丢弃。在RTP有效负载格式级别使用分段允许更有效地使用RTP数据包丢失恢复机制。在某些情况下,它还可以通过在特定于介质的边界上进行特定于介质的碎片化来更好地使用部分ADU。在ADU是预生产的且无法适应网络MTU大小的使用情况下,对碎片化的支持至关重要。

5.1.3. Interleaving and Transmission Rescheduling
5.1.3. 交织和传输重调度

Interleaving has been implemented in a number of payload formats to allow for less quality reduction when packet loss occurs. When losses are bursty and several consecutive packets are lost, the impact on quality can be quite severe. Interleaving is used to convert that burst loss to several spread-out individual packet losses. It can also be used when several ADUs are aggregated in the same packets. A loss of an RTP packet with several ADUs in the payload has the same effect as a burst loss if the ADUs would have been transmitted in individual packets. To reduce the burstiness of the loss, the data present in an aggregated payload may be interleaved, thus, spreading the loss over a longer time period.


A requirement for doing interleaving within an RTP payload format is the aggregation of multiple ADUs. For formats that do not use aggregation, there is still a possibility of implementing a transmission order rescheduling mechanism. That has the effect that the packets transmitted consecutively originate from different points in the RTP stream. This can be used to mitigate burst losses, which may be useful if one transmits packets at frequent intervals. However, it may also be used to transmit more significant data


earlier in combination with RTP retransmission to allow for more graceful degradation and increased possibility to receive the most important data, e.g., intra frames of video.


The drawback of interleaving is the significantly increased transmission buffering delay, making it less useful for low-delay applications. It may also create significant buffering requirements on the receiver. That buffering is also problematic, as it is usually difficult to indicate when a receiver may start consume data and still avoid buffer under run caused by the interleaving mechanism itself. Transmission rescheduling is only useful in a few specific cases, as in streaming with retransmissions. The potential gains must be weighed against the complexity of these schemes.


5.1.4. Media Back Channels
5.1.4. 媒体后台频道

A few RTP payload formats have implemented back channels within the media format. Those have been for specific features, like the AMR [RFC4867] codec mode request (CMR) field. The CMR field is used in the operation of gateways to circuit-switched voice to allow an IP terminal to react to the circuit-switched network's need for a specific encoder mode. A common motivation for media back channels is the need to have signaling in direct relation to the media or the media path.


If back channels are considered for an RTP payload format they should be for a specific requirements which cannot be easily satisfied by more generic mechanisms within RTP or RTCP.


5.1.5. Media Scalability
5.1.5. 媒体可扩展性

Some codecs support various types of media scalability, i.e. some data of a RTP stream may be removed to adapt the media's properties, such as bitrate and quality. The adaptation may be applied in the following dimensions of the media:


Temporal: For most video codecs it is possible to adapt the frame rate without any specific definition of a temporal scalability mode, e.g., for H.264 [RFC6184]. In these cases, the sender changes which frames it delivers and the RTP timestamp makes it clear the frame interval and each frames relative capture time. H.264 Scalable Video Coding (SVC) [RFC6190] has more explicit support for temporal scalability.


Spatial: Video codecs supporting scalability may adapt the resolution, e.g., in SVC [RFC6190].


Quality: The quality of the encoded stream may be scaled by adapting the accuracy of the coding process, as, e.g. possible with Signal to Noise Ratio (SNR) fidelity scalability of SVC [RFC6190].


At the time of writing this document, codecs that support scalability have a bit of a revival. It has been realized that getting the required functionality for supporting the features of the media stream into the RTP framework is quite challenging. One of the recent examples for layered and scalable codecs is SVC [RFC6190].


SVC is a good example for a payload format supporting media scalability features, which have been in its basic form already included in RTP. A layered codec supports the dropping of data parts of a RTP stream, i.e., RTP packets may not be transmitted or forwarded to a client in order to adapt the RTP streams bitrate as well as the received encoded stream's quality, while still providing a decodable subset of the encoded stream to a client. One example for using the scalability feature may be an RTP Mixer (Multipoint Control Unit) [RFC7667], which controls the rate and quality sent out to participants in a communication based on dropping RTP packets or removing part of the payload. Another example may be a transport channel, which allows for differentiation in Quality of Service (QoS) parameters based on RTP sessions in a multicast session. In such a case, the more important packets of the scalable encoded stream (base layer) may get better QoS parameters than the less important packets (enhancement layer) in order to provide some kind of graceful degradation. The scalability features required for allowing an adaptive transport, as described in the two examples above, are based on RTP multiplexing in order to identify the packets to be dropped or transmitted/forwarded. The multiplexing features defined for Scalable Video Coding [RFC6190] are:


Single Session Transmission (SST), where all media layers of the media are transported as a single synchronization source (SSRC) in a single RTP session; as well as


Multi-Session Transmission (MST), which should more accurately be called multi-stream transmission, where different media layers or a set of media layers are transported in different RTP streams, i.e., using multiple sources (SSRCs).


In the first case (SST), additional in-band as well as out-of-band signaling is required in order to allow identification of packets belonging to a specific media layer. Furthermore, an adaptation of the encoded stream requires dropping of specific packets in order to provide the client with a compliant encoded stream. In case of using encryption, it is typically required for an adapting network device


to be in the security context to allow packet dropping and providing an intact RTP session to the client. This typically requires the network device to be an RTP mixer.


In general, having a media-unaware network device dropping excessive packets will be more problematic than having a Media-Aware Network Entity (MANE). First is the need to understand the media format and know which ADUs or payloads belong to the layers, that no other layer will be dependent on after the dropping. Second, if the MANE can work as an RTP mixer or translator, it can rewrite the RTP and RTCP in such a way that the receiver will not suspect unintentional RTP packet losses needing repair actions. This as the receiver can't determine if a lost packet was an important base layer packet or one of the less important extension layers.


In the second case (MST), the RTP packet streams can be sent using a single or multiple RTP session, and thus transport flows, e.g., on different multicast groups. Transmitting the streams in different RTP sessions, then the out-of-band signaling typically provides enough information to identify the media layers and its properties. The decision on dropping packets is based on the Network Address that identifies the RTP session to be dropped. In order to allow correct data provisioning to a decoder after reception from different sessions, data realignment mechanisms are required. In some cases, existing generic tools, as described below, can be employed to enable such realignment; when those generic mechanisms are sufficient, they should be used. For example, "Rapid Synchronisation for RTP Flows" [RFC6051], uses existing RTP mechanisms, i.e. the NTP timestamp, to ensure timely inter-session synchronization. Another is the signaling feature for indicating dependencies of RTP sessions in SDP, as defined in the Media Decoding Dependency Grouping in SDP [RFC5583].


Using MST within a single RTP session is also possible and allows stream level handling instead of looking deeper into the packets by a MANE. However, transport flow-level properties will be the same unless packet based mechanisms like Diffserv is used.


When QoS settings, e.g., Diffserv markings, are used to ensure that the extension layers are dropped prior the base layer the receiving endpoint has the benefit in MST to know which layer or set of layers the missing packets belong to as it will be bound to different RTP sessions or RTP packet streams (SSRCs), thus, explicitly indicating the importance of the loss.


5.1.6. High Packet Rates
5.1.6. 高分组速率

Some media codecs require high packet rates; in these cases, the RTP sequence number wraps too quickly. As a rule of thumb, it must not be possible to wrap the sequence number space within at least three RTCP reporting intervals. As the reporting interval can vary widely due to configuration and session properties, and also must take into account the randomization of the interval, one can use the TCP maximum segment lifetime (MSL), i.e., 2 minutes, in ones consideration. If earlier wrapping may occur, then the payload format should specify an extended sequence number field to allow the receiver to determine where a specific payload belongs in the sequence, even in the face of extensive reordering. The RTP payload format for uncompressed video [RFC4175] can be used as an example for such a field.


RTCP is also affected by high packet rates. For RTCP mechanisms that do not use extended counters, there is significant risk that they wrap multiple times between RTCP reporting or feedback; thus, producing uncertainty about which packet(s) are referenced. The payload designer can't effect the RTCP packet formats used and their design, but can note this considerations when configuring RTCP bandwidth and reporting intervals to avoid to wrapping issues.


5.2. Selecting Timestamp Definition
5.2. 选择时间戳定义

The RTP timestamp is an important part and has two design choices associated with it. The first is the definition that determines what the timestamp value in a particular RTP packet will be, the second is which timestamp rate should be used.


The timestamp definition needs to explicitly define what the timestamp value in the RTP packet represent for a particular payload format. Two common definitions are used; for discretely sampled media, like video frames, the sampling time of the earliest included video frame which the data represent (fully or partially) is used; for continuous media like audio, the sampling time of the earliest sample which the payload data represent. There exist cases where more elaborate or other definitions are used.


RTP payload formats with a timestamp definition that results in no or little correlation between the media time instance and its transmission time cause the RTCP jitter calculation to become unusable due to the errors introduced on the sender side. A common example is a payload format for a video codec where the RTP timestamp represents the capture time of the video frame, but frames are large


enough that multiple RTP packets need to be sent for each frame spread across the framing interval. It should be noted whether or not the payload format has this property.


An RTP payload format also needs to define what timestamp rates, or clock rates (as it is also called), may be used. Depending on the RTP payload format, this may be a single rate or multiple ones or theoretically any rate. So what needs to be considered when selecting a rate?


The rate needs be selected so that one can determine where in the time line of the media a particular sample (e.g., individual audio sample, or video frame) or set of samples (e.g., audio frames) belong. To enable correct synchronization of this data with previous frames, including over periods of discontinuous transmission or irregularities.


For audio, it is common to require audio sample accuracy. Thus, one commonly selects the input sampling rate as the timestamp rate. This can, however, be challenging for audio codecs that support multiple different sampling frequencies, either as codec input or being used internally but effecting output, for example, frame duration. Depending on how one expects to use these different sampling rates one can allow multiple timestamp rates, each matching a particular codec input or sampling rate. However, due to the issues with using multiple different RTP timestamp rates for the same source (SSRC) [RFC7160], this should be avoided if one expects to need to switch between modes.


Then, an alternative is to find a common denominator frequency between the different modes, e.g., OPUS [RFC7587] that uses 48 kHz. If the different modes uses or can use a common input/output frequency, then selecting this also needs to be considered. However, it is important to consider all aspects as the case of AMR-WB+ [RFC4352] illustrates. AMR-WB+'s RTP timestamp rate has the very unusual value of 72 kHz, despite the fact that output normally is at a sample rate of 48kHz. The design is motivated by the media codec's production of a large range of different frame lengths in time perspective. The 72 kHz timestamp rate is the smallest found value that would make all of the frames the codec could produce result in an integer frame length in RTP timestamp ticks. This way, a receiver can always correctly place the frames in relation to any other frame, even when the frame length changes. The downside is that the decoder outputs for certain frame lengths are, in fact, partial samples. The result is that the output in samples from the codec will vary from frame to frame, potentially making implementation more difficult.

然后,另一种方法是在不同模式之间找到共同分母频率,例如,使用48 kHz的OPUS[RFC7587]。如果不同模式使用或可以使用公共输入/输出频率,则也需要考虑选择该频率。然而,重要的是考虑AMR WB+[RCF4352]所示的所有方面。AMR-WB+的RTP时间戳速率具有非常不寻常的72 kHz值,尽管事实上输出通常为48 kHz的采样率。该设计的动机是媒体编解码器在时间角度上产生大量不同的帧长度。72 kHz时间戳速率是发现的最小值,该值将使编解码器可能产生的所有帧以RTP时间戳滴答为单位产生整数帧长度。这样,即使帧长度发生变化,接收机也可以始终相对于任何其他帧正确放置帧。缺点是,解码器对于某些帧长度的输出实际上是部分采样。结果是,来自编解码器的样本中的输出将因帧而异,这可能使实现更加困难。

Video codecs have commonly been using 90 kHz; the reason is this is a common denominator between the usually used frame rates such as 24, 25, 30, 50 and 60, and NTSC's odd 29.97 Hz. There does, however, exist at least one exception in the payload format for SMPTE 292M video [RFC3497] that uses a clock rate of 148.5 MHz. The reason here is that the timestamp then identify the exact start sample within a video frame.

视频编解码器通常使用90 kHz;原因是这是通常使用的帧速率(如24、25、30、50和60)与NTSC的奇数29.97 Hz之间的共同点。然而,SMPTE 292M视频[RFC3497]的有效负载格式中至少存在一个例外,该格式使用148.5 MHz的时钟频率。这里的原因是时间戳随后标识视频帧中的确切开始样本。

Timestamp rates below 1000 Hz are not appropriate, because this will cause a resolution too low in the RTCP measurements that are expressed in RTP timestamps. This is the main reason that the text RTP payload formats, like T.140 [RFC4103], use 1000 Hz.

低于1000 Hz的时间戳速率不合适,因为这将导致RTP测量中以RTP时间戳表示的分辨率过低。这是文本RTP有效负载格式(如T.140[RFC4103])使用1000 Hz的主要原因。

6. Noteworthy Aspects in Payload Format Design
6. 有效载荷格式设计中值得注意的方面

This section provides a few examples of payload formats that are worth noting for good or bad design in general or in specific details.


6.1. Audio Payloads
6.1. 音频有效载荷

The AMR [RFC4867], AMR-WB [RFC4867], EVRC [RFC3558], SMV [RFC3558] payload formats are all quite similar. They are all for frame-based audio codecs and use a table of contents structure. Each frame has a table of contents entry that indicates the type of the frame and if additional frames are present. This is quite flexible, but produces unnecessary overhead if the ADU is of fixed size and if, when aggregating multiple ADUs, they are commonly of the same type. In that case, a solution like the one in AMR-WB+ [RFC4352] may be more suitable.


The RTP payload format for MIDI [RFC6295] contains some interesting features. MIDI is an audio format sensitive to packet losses, as the loss of a "note off" command will result in a note being stuck in an "on" state. To counter this, a recovery journal is defined that provides a summarized state that allows the receiver to recover from packet losses quickly. It also uses RTCP and the reported highest sequence number to be able to prune the state the recovery journal needs to contain. These features appear limited in applicability to media formats that are highly stateful and primarily use symbolic media representations.

MIDI[RFC6295]的RTP有效负载格式包含一些有趣的特性。MIDI是一种对数据包丢失非常敏感的音频格式,因为丢失“note off”命令将导致音符卡在“on”状态。为了解决这个问题,定义了一个恢复日志,它提供了一个汇总状态,允许接收方快速从数据包丢失中恢复。它还使用RTCP和报告的最高序列号来删除恢复日志需要包含的状态。这些功能在适用于高度有状态且主要使用符号媒体表示的媒体格式方面受到限制。

There exists a security concern with variable bitrate audio and speech codecs that changes their payload length based on the input data. This can leak information, especially in structured communication like a speech recognition prompt service that asks people to enter information verbally. This issue also exists to some degree for discontinuous transmission as that allows the length of


phrases to be determined. The issue is further discussed in "Guidelines for the Use of Variable Bit Rate Audio with Secure RTP" [RFC6562], which needs to be read by anyone writing an RTP payload format for an audio or speech codec with these properties.


6.2. Video
6.2. 视频

The definition of RTP payload formats for video has seen an evolution from the early ones such as H.261 [RFC4587] towards the latest for VP8 [RFC7741] and H.265/HEVC [RFC7798].


The H.264 RTP payload format [RFC3984] can be seen as a smorgasbord of functionality: some of it, such as the interleaving, being pretty advanced. The reason for this was to ensure that the majority of applications considered by the ITU-T and MPEG that can be supported by RTP are indeed supported. This has created a payload format that rarely is fully implemented. Despite that, no major issues with interoperability has been reported with one exception namely the Offer/Answer and parameter signaling, which resulted in a revised specification [RFC6184]. However, complaints about its complexity are common.

H.264 RTP有效负载格式[RFC3984]可以看作是一个功能的大杂烩:其中一些,如交织,非常先进。这样做的原因是为了确保ITU-T和MPEG考虑的RTP支持的大多数应用程序确实得到支持。这创建了一种很少完全实现的有效负载格式。尽管如此,互操作性方面没有重大问题,只有一个例外,即提供/应答和参数信令,这导致修订了规范[RFC6184]。然而,对其复杂性的抱怨是常见的。

The RTP payload format for uncompressed video [RFC4175] must be mentioned in this context as it contains a special feature not commonly seen in RTP payload formats. Due to the high bitrate and thus packet rate of uncompressed video (gigabits rather than megabits per second) the payload format includes a field to extend the RTP sequence number since the normal 16-bit one can wrap in less than a second. [RFC4175] also specifies a registry of different color sub-samplings that can be reused in other video RTP payload formats.


Both the H.264 and the uncompressed video format enable the implementer to fulfill the goals of application-level framing, i.e., each individual RTP Packet's payload can be independently decoded and its content used to create a video frame (or part of) and that irrespective of whether preceding packets has been lost (see Section 4) [RFC2736]. For uncompressed, this is straightforward as each pixel is independently represented from others and its location in the video frame known. H.264 is more dependent on the actual implementation, configuration of the video encoder and usage of the RTP payload format.


The common challenge with video is that, in most cases, a single compressed video frame doesn't fit into a single IP packet. Thus, the compressed representation of a video frame needs to be split over multiple packets. This can be done unintelligently with a basic payload level fragmentation method or more integrated by interfacing with the encoder's possibilities to create ADUs that are independent


and fit the MTU for the RTP packet. The latter is more robust and commonly recommended unless strong packet loss mechanisms are used and sufficient delay budget for the repair exist. Commonly, both payload-level fragmentation as well as explaining how tailored ADUs can be created are needed in a video payload format. Also, the handling of crucial metadata, like H.264 Parameter Sets, needs to be considered as decoding is not possible without receiving the used parameter sets.


6.3. Text
6.3. 文本

Only a single format text format has been standardized in the IETF, namely T.140 [RFC4103]. The 3GPP Timed Text format [RFC4396] should be considered to be text, even though in the end was registered as a video format. It was registered in that part of the tree because it deals with decorated text, usable for subtitles and other embellishments of video. However, it has many of the properties that text formats generally have.


The RTP payload format for T.140 was designed with high reliability in mind as real-time text commonly is an extremely low bitrate application. Thus, it recommends the use of RFC 2198 with many generations of redundancy. However, the format failed to provide a text-block-specific sequence number and instead relies on the RTP one to detect loss. This makes detection of missing text blocks unnecessarily difficult and hinders deployment with other robustness mechanisms that would involve switching the payload type, as that may result in erroneous error marking in the T.140 text stream.

T.140的RTP有效负载格式设计时考虑到了高可靠性,因为实时文本通常是一种极低比特率的应用。因此,建议使用具有多代冗余的RFC 2198。但是,该格式无法提供文本块特定的序列号,而是依赖RTP序列号来检测丢失。这使得检测丢失的文本块变得不必要的困难,并妨碍使用其他健壮性机制进行部署,这些机制将涉及切换有效负载类型,因为这可能导致T.140文本流中错误的错误标记。

6.4. Application
6.4. 应用

At the time of writing, the application content type contains two media types that aren't RTP transport robustness tools such as FEC [RFC3009] [RFC5109] [RFC6015] [RFC6682] and RTP retransmission [RFC4588].


The first one is H.224 [RFC4573], which enables far-end camera control over RTP. This is not an IETF-defined RTP format, only an IETF-performed registration.


The second one is "RTP Payload Format for Society of Motion Picture and Television Engineers (SMPTE) ST 336 Encoded Data" [RFC6597], which carries generic key length value (KLV) triplets. These pairs may contain arbitrary binary metadata associated with video transmissions. It has a very basic fragmentation mechanism requiring reception without packet loss, not only of the triplet itself but also one packet before and after the sequence of fragmented KLV triplet, to ensure correct reception. Specific KLV triplets


themselves may have recommendations on how to handle incomplete ones allowing the use and repair of them. In general, the application using such a mechanism must be robust to errors and also use some combination of application-level repetition, RTP-level transport robustness tools, and network-level requirements to achieve low levels of packet loss rates and repair of KLV triplets.


An author should consider applying for a media subtype under the application media type (application/<foo>) when the payload format is of a generic nature or does not clearly match any of the media types described above (audio, video, or text). However, existing limitations in, for example, SDP, have resulted in generic mechanisms normally registered in all media types possibly having been associated with any existing media types in an RTP session.

当有效载荷格式具有通用性或与上面描述的任何媒体类型(音频、视频或文本)不匹配时,作者应考虑在应用媒体类型(应用程序/< Fo>)下应用媒体子类型。然而,例如SDP中的现有限制已导致通常在所有媒体类型中注册的通用机制可能已与RTP会话中的任何现有媒体类型关联。

7. Important Specification Sections
7. 重要规范章节

A number of sections in the payload format draft need special consideration. These include the Security Considerations and IANA Considerations sections that are required in all drafts. Payload formats are also strongly recommended to have the media format description and congestion control considerations. The included RTP payload format template (Appendix A) contains sample text for some of these sections.


7.1. Media Format Description
7.1. 媒体格式说明

The intention of this section is to enable reviewers and other readers to get an overview of the capabilities and major properties of the media format. It should be kept short and concise and is not a complete replacement for reading the media format specification.


The actual specification of the RTP payload format generally uses normative references to the codec format specification to define how codec data elements are included in the payload format. This normative reference can be to anything that have sufficient stability for a normative reference. There exist no formal requirement on the codec format specification being publicly available or free to access. However, it significantly helps in the review process if that specification is made available to any reviewer. There exist RTP payload format RFCs for open-source project specifications as well as an individual company's proprietary format, and a large variety of standards development organizations or industrial forums.


7.2. Security Considerations
7.2. 安全考虑

All Internet-Drafts require a Security Considerations section. The Security Considerations section in an RTP payload format needs to concentrate on the security properties this particular format has. Some payload formats have very few specific issues or properties and can fully fall back on the security considerations for RTP in general and those of the profile being used. Because those documents are always applicable, a reference to these is normally placed first in the Security Considerations section. There is suggested text in the template below.


The security issues of confidentiality, integrity protection, replay protection and source authentication are common issue for all payload formats. These should be solved by mechanisms external to the payload and do not need any special consideration in the payload format except for a reminder on these issues. There exist exceptions, such as payload formats that includes security functionality, like ISMAcrypt [ISMACrypt2]. Reasons for this division is further documented in "Securing the RTP Protocol Framework: Why RTP Does Not Mandate a Single Media Security Solution" [RFC7202]. For a survey of available mechanisms to meet these goals, review "Options for Securing RTP Sessions" [RFC7201]. This also includes key-exchange mechanisms for the security mechanisms, which can be both integrated or separate. The choice of key-management can have significant impact on the security properties of the RTP-based application. Suitable stock text to inform people about this is included in the template.


Potential security issues with an RTP payload format and the media encoding that need to be considered if they are applicable:


1. The decoding of the payload format or its media results in substantial non-uniformity, either in output or in complexity to perform the decoding operation. For example, a generic non-destructive compression algorithm may provide an output of almost an infinite size for a very limited input, thus consuming memory or storage space out of proportion with what the receiving application expected. Such inputs can cause some sort of disruption, i.e., a denial-of-service attack on the receiver side by preventing that host from performing usable work. Certain decoding operations may also vary in the amount of processing needed to perform those operations depending on the input. This may also be a security risk if it is possible to raise processing load significantly above nominal simply by designing a malicious input sequence. If such potential attacks exist, this must be

1. 有效载荷格式或其媒体的解码导致在输出或执行解码操作的复杂性方面的实质性不均匀性。例如,通用无损压缩算法可以为非常有限的输入提供几乎无限大小的输出,从而消耗与接收应用程序预期不相称的内存或存储空间。此类输入可能会导致某种中断,即通过阻止主机执行可用工作,在接收方发起拒绝服务攻击。根据输入,某些解码操作在执行这些操作所需的处理量上也可能有所不同。如果仅仅通过设计恶意输入序列就可以将处理负载显著提高到标称负载以上,那么这也可能是一种安全风险。如果存在此类潜在攻击,则必须

made clear in the Security Considerations section to make implementers aware of the need to take precautions against such behavior.


2. The inclusion of active content in the media format or its transport. "Active content" means scripts, etc., that allow an attacker to perform potentially arbitrary operations on the receiver. Most active contents has limited possibility to access the system or perform operations outside a protected sandbox. RFC 4855 [RFC4855] has a requirement that it be noted in the media types registration whether or not the payload format contains active content. If the payload format has active content, it is strongly recommended that references to any security model applicable for such content are provided. A boilerplate text for "no active content" is included in the template. This must be changed if the format actually carries active content.

2. 在媒体格式或其传输中包含活动内容。“活动内容”指允许攻击者在接收器上执行潜在任意操作的脚本等。大多数活动内容访问系统或在受保护的沙箱之外执行操作的可能性有限。RFC 4855[RFC4855]要求在媒体类型注册中注明有效负载格式是否包含活动内容。如果有效负载格式具有活动内容,强烈建议提供适用于此类内容的任何安全模型的参考。模板中包含“无活动内容”的样板文本。如果格式实际包含活动内容,则必须更改此设置。

3. Some media formats allow for the carrying of "user data", or types of data which are not known at the time of the specification of the payload format. Such data may be a security risk and should be mentioned.

3. 一些媒体格式允许携带“用户数据”,或在指定有效负载格式时未知的数据类型。此类数据可能存在安全风险,应予以提及。

4. Audio or Speech codecs supporting variable bitrate based on 'audio/speech' input or having discontinuous transmission support must consider the issues discussed in "Guidelines for the Use of Variable Bit Rate Audio with Secure RTP" [RFC6562].

4. 支持基于“音频/语音”输入或具有不连续传输支持的可变比特率的音频或语音编解码器必须考虑在“使用安全RTP的可变比特率音频的准则”中讨论的问题[RCF6562]。

Suitable stock text for the Security Considerations section is provided in the template in Appendix A. However, authors do need to actively consider any security issues from the start. Failure to address these issues may block approval and publication.


7.3. Congestion Control
7.3. 拥塞控制

RTP and its profiles do discuss congestion control. There is ongoing work in the IETF with both a basic circuit-breaker mechanism [RFC8083] using basic RTCP messages intended to prevent persistent congestion and also work on more capable congestion avoidance / bitrate adaptation mechanism in the RMCAT WG.

RTP及其配置文件确实讨论了拥塞控制。IETF中正在进行使用基本RTCP消息的基本断路器机制[RFC8083]的工作,以防止持续拥塞,并在RMCAT WG中进行更有效的拥塞避免/比特率自适应机制的工作。

Congestion control is an important issue in any usage in networks that are not dedicated. For that reason, it is recommended that all RTP payload format documents discuss the possibilities that exist to regulate the bitrate of the transmissions using the described RTP payload format. Some formats may have limited or step-wise regulation of bitrate. Such limiting factors should be discussed.


7.4. IANA Considerations
7.4. IANA考虑

Since all RTP payload formats contain a media type specification, they also need an IANA Considerations section. The media type name must be registered, and this is done by requesting that IANA register that media name. When that registration request is written, it shall also be requested that the media type is included under the "RTP Payload Format media types" subregistry of the RTP registry (


Parameters for the payload format need to be included in this registration and can be specified as required or optional ones. The format of these parameters should be such that they can be included in the SDP attribute "a=fmtp" string (see Section 6 [RFC4566]), which is the common mapping. Some parameters, such as "Channel" are normally mapped to the rtpmap attribute instead; see Section 3 of [RFC4855].


In addition to the above request for media type registration, some payload formats may have parameters where, in the future, new parameter values need to be added. In these cases, a registry for that parameter must be created. This is done by defining the registry in the IANA Considerations section. BCP 26 [BCP26] provides guidelines to specifying such registries. Care should be taken when defining the policy for new registrations.

除了上述媒体类型注册请求之外,一些有效负载格式可能具有参数,其中将来需要添加新的参数值。在这些情况下,必须为该参数创建注册表。这是通过在IANA注意事项部分中定义注册表来完成的。BCP 26[BCP26]提供了指定此类注册的指南。在定义新注册的策略时应小心。

Before specifying a new registry, it is worth checking the existing ones in the IANA "MIME Media Type Sub-Parameter Registries". For example, video formats that need a media parameter expressing color sub-sampling may be able to reuse those defined for 'video/raw' [RFC4175].


8. Authoring Tools
8. 创作工具

This section provides information about some tools that may be used. Don't feel pressured to follow these recommendations. There exist a number of alternatives, including the ones listed at <>. But these suggestions are worth checking out before deciding that the grass is greener somewhere else.

本节提供有关可能使用的某些工具的信息。不要觉得遵循这些建议有压力。存在许多备选方案,包括在<>. 但在决定其他地方的草是否更绿之前,这些建议值得一看。

Note that these options are related to the old text only RFC format, and do not cover tools for at the time of publication recently approved new RFC format, see [RFC7990].


8.1. Editing Tools
8.1. 编辑工具

There are many choices when it comes to tools to choose for authoring Internet-Drafts. However, in the end, they need to be able to produce a draft that conforms to the Internet-Draft requirements. If you don't have any previous experience with authoring Internet-Drafts, xml2rfc does have some advantages. It helps by creating a lot of the necessary boilerplate in accordance with the latest rules, thus reducing the effort. It also speeds up publication after approval as the RFC Editor can use the source XML document to produce the RFC more quickly.


Another common choice is to use Microsoft Word and a suitable template (see [RFC5385]) to produce the draft and print that to file using the generic text printer. It has some advantages when it comes to spell checking and change bars. However, Word may also produce some problems, like changing formatting, and inconsistent results between what one sees in the editor and in the generated text document, at least according to the author's personal experience.

另一种常见的选择是使用Microsoft Word和合适的模板(请参见[RFC5385])生成草稿,并使用通用文本打印机将其打印到文件中。当涉及到拼写检查和更改条时,它有一些优势。但是,Word也可能会产生一些问题,如更改格式,以及在编辑器中看到的内容与生成的文本文档中看到的结果不一致,至少根据作者的个人经验是这样。

8.2. Verification Tools
8.2. 验证工具

There are a few tools that are very good to know about when writing a draft. These help check and verify parts of one's work. These tools can be found at <>.


o I-D Nits checker ( It checks that the boilerplate and some other things that are easily verifiable by machine are okay in your draft. Always use it before submitting a draft to avoid direct refusal in the submission step.

o I-D Nits检测器( 它检查样板文件和其他一些可以通过机器轻松验证的东西在草稿中是否正确。在提交草稿之前务必使用它,以避免在提交步骤中直接拒绝。

o ABNF Parser and verification ( abnf.cgi). Checks that your ABNF parses correctly and warns about loose ends, like undefined symbols. However, the actual content can only be verified by humans knowing what it intends to describe.

o ABNF解析器与验证( abnf.cgi)。检查ABNF是否正确解析,并警告未定义的符号等松散端。然而,实际的内容只能由知道它打算描述什么的人来验证。

o RFC diff ( A diff tool that is optimized for drafts and RFCs. For example, it does not point out that the footer and header have moved in relation to the text on every page.

o RFC差异( 为草稿和RFC优化的差异工具。例如,它没有指出页脚和页眉相对于每页上的文本已经移动。

9. Security Considerations
9. 安全考虑

As this is an Informational RFC about writing drafts that are intended to become RFCs, there are no direct security considerations. However, the document does discuss the writing of Security Considerations sections and what should be particularly considered when specifying RTP payload formats.


10. Informative References
10. 资料性引用

[BCP9] Bradner, S., "The Internet Standards Process -- Revision 3", BCP 9, RFC 2026, October 1996.

[BCP9]Bradner,S.,“互联网标准过程——第3版”,BCP 9,RFC 2026,1996年10月。

Kolkman, O., Bradner, S., and S. Turner, "Characterization of Proposed Standards", BCP 9, RFC 7127, January 2014.

Kolkman,O.,Bradner,S.和S.Turner,“拟定标准的特征”,BCP 9,RFC 7127,2014年1月。

Dusseault, L. and R. Sparks, "Guidance on Interoperation and Implementation Reports for Advancement to Draft Standard", BCP 9, RFC 5657, September 2009.

Dusseault,L.和R.Sparks,“推进标准草案的互操作和实施报告指南”,BCP 9,RFC 5657,2009年9月。

Housley, R., Crocker, D., and E. Burger, "Reducing the Standards Track to Two Maturity Levels", BCP 9, RFC 6410, October 2011.

Housley,R.,Crocker,D.,和E.Burger,“将标准轨道降低到两个成熟度水平”,BCP 9,RFC 6410,2011年10月。

Resnick, P., "Retirement of the "Internet Official Protocol Standards" Summary Document", BCP 9, RFC 7100, December 2013.

Resnick,P.,“互联网官方协议标准”的退役摘要文件,BCP 9,RFC 7100,2013年12月。

Dawkins, S., "Increasing the Number of Area Directors in an IETF Area", BCP 9, RFC 7475, March 2015.

Dawkins,S.,“增加IETF区域的区域主管数量”,BCP 9,RFC 7475,2015年3月。


[BCP25] Wasserman, M., "Updates to RFC 2418 Regarding the Management of IETF Mailing Lists", BCP 25, RFC 3934, October 2004.

[BCP25]Wasserman,M.,“关于IETF邮件列表管理的RFC 2418更新”,BCP 25,RFC 3934,2004年10月。

Bradner, S., "IETF Working Group Guidelines and Procedures", BCP 25, RFC 2418, September 1998.

Bradner,S.,“IETF工作组指南和程序”,BCP 25,RFC 2418,1998年9月。

Resnick, P. and A. Farrel, "IETF Anti-Harassment Procedures", BCP 25, RFC 7776, March 2016.

Resnick,P.和A.Farrel,“IETF反骚扰程序”,BCP 25,RFC 77762016年3月。


[BCP26] Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 5226, May 2008, <>.

[BCP26]Narten,T.和H.Alvestrand,“在RFCs中编写IANA注意事项部分的指南”,BCP 26,RFC 5226,2008年5月<>.

[BCP78] Bradner, S., Ed. and J. Contreras, Ed., "Rights Contributors Provide to the IETF Trust", BCP 78, RFC 5378, November 2008, <>.

[BCP78]Bradner,S.,Ed.和J.Contreras,Ed.,“向IETF信托提供的权利出资人”,BCP 78,RFC 5378,2008年11月<>.

[BCP79] Bradner, S., Ed., "Intellectual Property Rights in IETF Technology", BCP 79, RFC 3979, March 2005.

[BCP79]Bradner,S.,Ed.,“IETF技术中的知识产权”,BCP 79,RFC 3979,2005年3月。

Narten, T., "Clarification of the Third Party Disclosure Procedure in RFC 3979", BCP 79, RFC 4879, April 2007.

Narten,T.,“RFC 3979中第三方披露程序的澄清”,BCP 79,RFC 4879,2007年4月。


[BLOAT] Nichols, K. and V. Jacobson, "Controlling Queue Delay", ACM Networks, Vol. 10, No. 5, DOI 10.1145/2208917.2209336, May 2012, <>.

[BLOAT]Nichols,K.和V.Jacobson,“控制队列延迟”,ACM网络,第10卷,第5期,DOI 10.1145/2208917.2209336,2012年5月<>.

[CSP-RTP] Perkins, C., "RTP: Audio and Video for the Internet", Addison-Wesley Professional, ISBN 0-672-32249-8, June 2003.

[CSP-RTP]Perkins,C.,“RTP:互联网音频和视频”,Addison-Wesley Professional,ISBN 0-672-32249-82003年6月。

[ID-GUIDE] Housley, R., "Guidelines to Authors of Internet-Drafts", December 2010, <>.


[ISMACrypt2] Internet Streaming Media Alliance (ISMA), "ISMA Encryption and Authentication, Version 2.0 release version", November 2007, <>.


[RED] Floyd, S. and V. Jacobson, "Random Early Detection (RED) gateways for Congestion Avoidance", IEEE/ACM Transactions on Networking 1(4) 397--413, August 1993, <>.


[RFC1633] Braden, R., Clark, D., and S. Shenker, "Integrated Services in the Internet Architecture: an Overview", RFC 1633, DOI 10.17487/RFC1633, June 1994, <>.

[RFC1633]Braden,R.,Clark,D.,和S.Shenker,“互联网体系结构中的综合服务:概述”,RFC 1633,DOI 10.17487/RFC1633,1994年6月<>.

[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/RFC2119, March 1997, <>.

[RFC2119]Bradner,S.,“RFC中用于表示需求水平的关键词”,BCP 14,RFC 2119,DOI 10.17487/RFC2119,1997年3月<>.

[RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, DOI 10.17487/RFC2198, September 1997, <>.

[RFC2198]Perkins,C.,Kouvelas,I.,Hodson,O.,Hardman,V.,Handley,M.,Bolot,J.,Vega Garcia,A.,和S.Fosse Parisis,“冗余音频数据的RTP有效载荷”,RFC 2198,DOI 10.17487/RFC2198,1997年9月<>.

[RFC2360] Scott, G., "Guide for Internet Standards Writers", BCP 22, RFC 2360, DOI 10.17487/RFC2360, June 1998, <>.

[RFC2360]Scott,G.“互联网标准编写者指南”,BCP 22,RFC 2360,DOI 10.17487/RFC2360,1998年6月<>.

[RFC2418] Bradner, S., "IETF Working Group Guidelines and Procedures", BCP 25, RFC 2418, DOI 10.17487/RFC2418, September 1998, <>.

[RFC2418]Bradner,S.,“IETF工作组指南和程序”,BCP 25,RFC 2418,DOI 10.17487/RFC2418,1998年9月<>.

[RFC2475] Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z., and W. Weiss, "An Architecture for Differentiated Services", RFC 2475, DOI 10.17487/RFC2475, December 1998, <>.

[RFC2475]Blake,S.,Black,D.,Carlson,M.,Davies,E.,Wang,Z.,和W.Weiss,“差异化服务架构”,RFC 2475,DOI 10.17487/RFC2475,1998年12月<>.

[RFC2508] Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP Headers for Low-Speed Serial Links", RFC 2508, DOI 10.17487/RFC2508, February 1999, <>.

[RFC2508]Casner,S.和V.Jacobson,“压缩低速串行链路的IP/UDP/RTP报头”,RFC 2508,DOI 10.17487/RFC2508,1999年2月<>.

[RFC2733] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for Generic Forward Error Correction", RFC 2733, DOI 10.17487/RFC2733, December 1999, <>.

[RFC2733]Rosenberg,J.和H.Schulzrinne,“通用前向纠错的RTP有效载荷格式”,RFC 2733,DOI 10.17487/RFC2733,1999年12月<>.

[RFC2736] Handley, M. and C. Perkins, "Guidelines for Writers of RTP Payload Format Specifications", BCP 36, RFC 2736, DOI 10.17487/RFC2736, December 1999, <>.

[RFC2736]Handley,M.和C.Perkins,“RTP有效载荷格式规范编写者指南”,BCP 36,RFC 2736,DOI 10.17487/RFC2736,1999年12月<>.

[RFC2959] Baugher, M., Strahm, B., and I. Suconick, "Real-Time Transport Protocol Management Information Base", RFC 2959, DOI 10.17487/RFC2959, October 2000, <>.

[RFC2959]Baugher,M.,Strahm,B.,和I.Suconick,“实时传输协议管理信息库”,RFC 2959,DOI 10.17487/RFC2959,2000年10月<>.

[RFC2974] Handley, M., Perkins, C., and E. Whelan, "Session Announcement Protocol", RFC 2974, DOI 10.17487/RFC2974, October 2000, <>.

[RFC2974]Handley,M.,Perkins,C.,和E.Whelan,“会话公告协议”,RFC 2974,DOI 10.17487/RFC2974,2000年10月<>.

[RFC3009] Rosenberg, J. and H. Schulzrinne, "Registration of parityfec MIME types", RFC 3009, DOI 10.17487/RFC3009, November 2000, <>.

[RFC3009]Rosenberg,J.和H.Schulzrinne,“parityfec MIME类型的注册”,RFC 3009,DOI 10.17487/RFC3009,2000年11月<>.

[RFC3095] Bormann, C., Burmeister, C., Degermark, M., Fukushima, H., Hannu, H., Jonsson, L-E., Hakenberg, R., Koren, T., Le, K., Liu, Z., Martensson, A., Miyazaki, A., Svanbro, K., Wiebke, T., Yoshimura, T., and H. Zheng, "RObust Header Compression (ROHC): Framework and four profiles: RTP, UDP, ESP, and uncompressed", RFC 3095, DOI 10.17487/RFC3095, July 2001, <>.

[RFC3095]Bormann,C.,Burmeister,C.,Degermark,M.,Fukushima,H.,Hannu,H.,Jonsson,L-E.,Hakenberg,R.,Koren,T.,Le,K.,Liu,Z.,Martenson,A.,Miyazaki,A.,Svanbro,K.,Wiebke,T.,Yoshimura,T.,和H.Zheng,“鲁棒头压缩(ROHC):框架和四个配置文件:RTP,UDP,ESP,和未压缩”,RFC 3095,内政部10.17487/RFC30952001年7月<>.

[RFC3261] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, DOI 10.17487/RFC3261, June 2002, <>.

[RFC3261]Rosenberg,J.,Schulzrinne,H.,Camarillo,G.,Johnston,A.,Peterson,J.,Sparks,R.,Handley,M.,和E.Schooler,“SIP:会话启动协议”,RFC 3261,DOI 10.17487/RFC3261,2002年6月<>.

[RFC3264] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, DOI 10.17487/RFC3264, June 2002, <>.

[RFC3264]Rosenberg,J.和H.Schulzrinne,“具有会话描述协议(SDP)的提供/应答模型”,RFC 3264,DOI 10.17487/RFC3264,2002年6月<>.

[RFC3410] Case, J., Mundy, R., Partain, D., and B. Stewart, "Introduction and Applicability Statements for Internet-Standard Management Framework", RFC 3410, DOI 10.17487/RFC3410, December 2002, <>.

[RFC3410]Case,J.,Mundy,R.,Partain,D.,和B.Stewart,“互联网标准管理框架的介绍和适用性声明”,RFC 3410,DOI 10.17487/RFC3410,2002年12月<>.

[RFC3497] Gharai, L., Perkins, C., Goncher, G., and A. Mankin, "RTP Payload Format for Society of Motion Picture and Television Engineers (SMPTE) 292M Video", RFC 3497, DOI 10.17487/RFC3497, March 2003, <>.

[RFC3497]Gharai,L.,Perkins,C.,Goncher,G.,和A.Mankin,“电影电视工程师学会(SMPTE)292M视频RTP有效载荷格式”,RFC 3497,DOI 10.17487/RFC3497,2003年3月<>.

[RFC3545] Koren, T., Casner, S., Geevarghese, J., Thompson, B., and P. Ruddy, "Enhanced Compressed RTP (CRTP) for Links with High Delay, Packet Loss and Reordering", RFC 3545, DOI 10.17487/RFC3545, July 2003, <>.

[RFC3545]Koren,T.,Casner,S.,Geevarghese,J.,Thompson,B.,和P.Ruddy,“具有高延迟、数据包丢失和重新排序的链路的增强压缩RTP(CRTP)”,RFC 3545,DOI 10.17487/RFC3545,2003年7月<>.

[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, 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 3550,DOI 10.17487/RFC3550,2003年7月<>.

[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, DOI 10.17487/RFC3551, July 2003, <>.

[RFC3551]Schulzrinne,H.和S.Casner,“具有最小控制的音频和视频会议的RTP配置文件”,STD 65,RFC 3551,DOI 10.17487/RFC3551,2003年7月<>.

[RFC3558] Li, A., "RTP Payload Format for Enhanced Variable Rate Codecs (EVRC) and Selectable Mode Vocoders (SMV)", RFC 3558, DOI 10.17487/RFC3558, July 2003, <>.

[RFC3558]Li,A.“增强型变速率编解码器(EVRC)和可选模式声码器(SMV)的RTP有效载荷格式”,RFC 3558,DOI 10.17487/RFC3558,2003年7月<>.

[RFC3569] Bhattacharyya, S., Ed., "An Overview of Source-Specific Multicast (SSM)", RFC 3569, DOI 10.17487/RFC3569, July 2003, <>.

[RFC3569]Bhattacharyya,S.,编辑,“源特定多播(SSM)概述”,RFC 3569,DOI 10.17487/RFC3569,2003年7月<>.

[RFC3577] Waldbusser, S., Cole, R., Kalbfleisch, C., and D. Romascanu, "Introduction to the Remote Monitoring (RMON) Family of MIB Modules", RFC 3577, DOI 10.17487/RFC3577, August 2003, <>.

[RFC3577]Waldbusser,S.,Cole,R.,Kalbflish,C.,和D.Romascanu,“MIB模块远程监控(RMON)系列介绍”,RFC 3577,DOI 10.17487/RFC3577,2003年8月<>.

[RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed., "RTP Control Protocol Extended Reports (RTCP XR)", RFC 3611, DOI 10.17487/RFC3611, November 2003, <>.

[RFC3611]Friedman,T.,Ed.,Caceres,R.,Ed.,和A.Clark,Ed.,“RTP控制协议扩展报告(RTCP XR)”,RFC 3611,DOI 10.17487/RFC36112003年11月<>.

[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, DOI 10.17487/RFC3711, March 2004, <>.

[RFC3711]Baugher,M.,McGrew,D.,Naslund,M.,Carrara,E.,和K.Norrman,“安全实时传输协议(SRTP)”,RFC 3711,DOI 10.17487/RFC3711,2004年3月<>.

[RFC3828] Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., Ed., and G. Fairhurst, Ed., "The Lightweight User Datagram Protocol (UDP-Lite)", RFC 3828, DOI 10.17487/RFC3828, July 2004, <>.

[RFC3828]Larzon,L-A.,Degermark,M.,Pink,S.,Jonsson,L-E.,Ed.,和G.Fairhurst,Ed.,“轻量级用户数据报协议(UDP Lite)”,RFC 3828,DOI 10.17487/RFC3828,2004年7月<>.

[RFC3984] Wenger, S., Hannuksela, M., Stockhammer, T., Westerlund, M., and D. Singer, "RTP Payload Format for H.264 Video", RFC 3984, DOI 10.17487/RFC3984, February 2005, <>.

[RFC3984]Wenger,S.,Hannuksela,M.,Stockhammer,T.,Westerlund,M.,和D.Singer,“H.264视频的RTP有效载荷格式”,RFC 3984,DOI 10.17487/RFC3984,2005年2月<>.

[RFC4103] Hellstrom, G. and P. Jones, "RTP Payload for Text Conversation", RFC 4103, DOI 10.17487/RFC4103, June 2005, <>.

[RFC4103]Hellstrom,G.和P.Jones,“文本对话的RTP有效载荷”,RFC 4103,DOI 10.17487/RFC4103,2005年6月<>.

[RFC4170] Thompson, B., Koren, T., and D. Wing, "Tunneling Multiplexed Compressed RTP (TCRTP)", BCP 110, RFC 4170, DOI 10.17487/RFC4170, November 2005, <>.

[RFC4170]Thompson,B.,Koren,T.,和D.Wing,“隧道多路复用压缩RTP(TCRTP)”,BCP 110,RFC 4170,DOI 10.17487/RFC4170,2005年11月<>.

[RFC4175] Gharai, L. and C. Perkins, "RTP Payload Format for Uncompressed Video", RFC 4175, DOI 10.17487/RFC4175, September 2005, <>.

[RFC4175]Gharai,L.和C.Perkins,“未压缩视频的RTP有效载荷格式”,RFC 4175,DOI 10.17487/RFC4175,2005年9月<>.

[RFC4352] Sjoberg, J., Westerlund, M., Lakaniemi, A., and S. Wenger, "RTP Payload Format for the Extended Adaptive Multi-Rate Wideband (AMR-WB+) Audio Codec", RFC 4352, DOI 10.17487/RFC4352, January 2006, <>.

[RFC4352]Sjoberg,J.,Westerlund,M.,Lakaniemi,A.,和S.Wenger,“扩展自适应多速率宽带(AMR-WB+)音频编解码器的RTP有效载荷格式”,RFC 4352,DOI 10.17487/RFC4352,2006年1月<>.

[RFC4396] Rey, J. and Y. Matsui, "RTP Payload Format for 3rd Generation Partnership Project (3GPP) Timed Text", RFC 4396, DOI 10.17487/RFC4396, February 2006, <>.

[RFC4396]Rey,J.和Y.Matsui,“第三代合作伙伴关系项目(3GPP)定时文本的RTP有效载荷格式”,RFC 4396,DOI 10.17487/RFC4396,2006年2月<>.

[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, DOI 10.17487/RFC4566, July 2006, <>.

[RFC4566]Handley,M.,Jacobson,V.,和C.Perkins,“SDP:会话描述协议”,RFC 4566,DOI 10.17487/RFC4566,2006年7月<>.

[RFC4571] Lazzaro, J., "Framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) Packets over Connection-Oriented Transport", RFC 4571, DOI 10.17487/RFC4571, July 2006, <>.

[RFC4571]Lazzaro,J.,“面向连接传输上的帧实时传输协议(RTP)和RTP控制协议(RTCP)数据包”,RFC 4571,DOI 10.17487/RFC4571,2006年7月<>.

[RFC4573] Even, R. and A. Lochbaum, "MIME Type Registration for RTP Payload Format for H.224", RFC 4573, DOI 10.17487/RFC4573, July 2006, <>.

[RFC4573]Even,R.和A.Lochbaum,“H.224 RTP有效载荷格式的MIME类型注册”,RFC 4573,DOI 10.17487/RFC4573,2006年7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, DOI 10.17487/RFC4585, 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 4585,DOI 10.17487/RFC4585,2006年7月<>.

[RFC4587] Even, R., "RTP Payload Format for H.261 Video Streams", RFC 4587, DOI 10.17487/RFC4587, August 2006, <>.

[RFC4587]偶数,R.,“H.261视频流的RTP有效载荷格式”,RFC 4587,DOI 10.17487/RFC4587,2006年8月<>.

[RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. Hakenberg, "RTP Retransmission Payload Format", RFC 4588, DOI 10.17487/RFC4588, July 2006, <>.

[RFC4588]Rey,J.,Leon,D.,Miyazaki,A.,Varsa,V.,和R.Hakenberg,“RTP重传有效载荷格式”,RFC 4588,DOI 10.17487/RFC4588,2006年7月<>.

[RFC4648] Josefsson, S., "The Base16, Base32, and Base64 Data Encodings", RFC 4648, DOI 10.17487/RFC4648, October 2006, <>.

[RFC4648]Josefsson,S.,“Base16、Base32和Base64数据编码”,RFC 4648,DOI 10.17487/RFC4648,2006年10月<>.

[RFC4844] Daigle, L., Ed. and Internet Architecture Board, "The RFC Series and RFC Editor", RFC 4844, DOI 10.17487/RFC4844, July 2007, <>.

[RFC4844]Daigle,L.,Ed.和互联网架构委员会,“RFC系列和RFC编辑器”,RFC 4844,DOI 10.17487/RFC4844,2007年7月<>.

[RFC4855] Casner, S., "Media Type Registration of RTP Payload Formats", RFC 4855, DOI 10.17487/RFC4855, February 2007, <>.

[RFC4855]Casner,S.,“RTP有效载荷格式的媒体类型注册”,RFC 4855,DOI 10.17487/RFC4855,2007年2月<>.

[RFC4867] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, "RTP Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs", RFC 4867, DOI 10.17487/RFC4867, April 2007, <>.

[RFC4867]Sjoberg,J.,Westerlund,M.,Lakaniemi,A.,和Q.Xie,“自适应多速率(AMR)和自适应多速率宽带(AMR-WB)音频编解码器的RTP有效载荷格式和文件存储格式”,RFC 4867,DOI 10.17487/RFC4867,2007年4月<>.

[RFC4975] Campbell, B., Ed., Mahy, R., Ed., and C. Jennings, Ed., "The Message Session Relay Protocol (MSRP)", RFC 4975, DOI 10.17487/RFC4975, September 2007, <>.

[RFC4975]Campbell,B.,Ed.,Mahy,R.,Ed.,和C.Jennings,Ed.,“消息会话中继协议(MSRP)”,RFC 4975,DOI 10.17487/RFC4975,2007年9月<>.

[RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error Correction", RFC 5109, DOI 10.17487/RFC5109, December 2007, <>.

[RFC5109]Li,A.,Ed.“通用前向纠错的RTP有效载荷格式”,RFC 5109,DOI 10.17487/RFC5109,2007年12月<>.

[RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC 5124, DOI 10.17487/RFC5124, February 2008, <>.

[RFC5124]Ott,J.和E.Carrara,“基于实时传输控制协议(RTCP)的反馈扩展安全RTP配置文件(RTP/SAVPF)”,RFC 5124DOI 10.17487/RFC5124,2008年2月<>.

[RFC5234] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, DOI 10.17487/RFC5234, January 2008, <>.

[RFC5234]Crocker,D.,Ed.和P.Overell,“语法规范的扩充BNF:ABNF”,STD 68,RFC 5234,DOI 10.17487/RFC5234,2008年1月<>.

[RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July 2008, <>.

[RFC5285]Singer,D.和H.Desneni,“RTP报头扩展的一般机制”,RFC 5285,DOI 10.17487/RFC5285,2008年7月<>.

[RFC5385] Touch, J., "Version 2.0 Microsoft Word Template for Creating Internet Drafts and RFCs", RFC 5385, DOI 10.17487/RFC5385, February 2010, <>.

[RFC5385]Touch,J.,“用于创建互联网草稿和RFC的2.0版Microsoft Word模板”,RFC 5385,DOI 10.17487/RFC5385,2010年2月<>.

[RFC5484] Singer, D., "Associating Time-Codes with RTP Streams", RFC 5484, DOI 10.17487/RFC5484, March 2009, <>.

[RFC5484]Singer,D.,“将时间码与RTP流相关联”,RFC 5484,DOI 10.17487/RFC54842009年3月<>.

[RFC5583] Schierl, T. and S. Wenger, "Signaling Media Decoding Dependency in the Session Description Protocol (SDP)", RFC 5583, DOI 10.17487/RFC5583, July 2009, <>.

[RFC5583]Schierl,T.和S.Wenger,“会话描述协议(SDP)中的信令媒体解码依赖性”,RFC 5583,DOI 10.17487/RFC5583,2009年7月<>.

[RFC5795] Sandlund, K., Pelletier, G., and L-E. Jonsson, "The RObust Header Compression (ROHC) Framework", RFC 5795, DOI 10.17487/RFC5795, March 2010, <>.

[RFC5795]Sandlund,K.,Pelletier,G.和L-E.Jonsson,“鲁棒头压缩(ROHC)框架”,RFC 5795,DOI 10.17487/RFC5795,2010年3月<>.

[RFC5905] Mills, D., Martin, J., Ed., Burbank, J., and W. Kasch, "Network Time Protocol Version 4: Protocol and Algorithms Specification", RFC 5905, DOI 10.17487/RFC5905, June 2010, <>.

[RFC5905]Mills,D.,Martin,J.,Ed.,Burbank,J.,和W.Kasch,“网络时间协议版本4:协议和算法规范”,RFC 5905,DOI 10.17487/RFC59052010年6月<>.

[RFC6015] Begen, A., "RTP Payload Format for 1-D Interleaved Parity Forward Error Correction (FEC)", RFC 6015, DOI 10.17487/RFC6015, October 2010, <>.

[RFC6015]Begen,A.,“用于1-D交错奇偶校验前向纠错(FEC)的RTP有效载荷格式”,RFC 6015,DOI 10.17487/RFC6015,2010年10月<>.

[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP Flows", RFC 6051, DOI 10.17487/RFC6051, November 2010, <>.

[RFC6051]Perkins,C.和T.Schierl,“RTP流的快速同步”,RFC 6051,DOI 10.17487/RFC6051,2010年11月<>.

[RFC6184] Wang, Y., Even, R., Kristensen, T., and R. Jesup, "RTP Payload Format for H.264 Video", RFC 6184, DOI 10.17487/RFC6184, May 2011, <>.

[RFC6184]Wang,Y.,Even,R.,Kristensen,T.,和R.Jesup,“H.264视频的RTP有效载荷格式”,RFC 6184,DOI 10.17487/RFC6184,2011年5月<>.

[RFC6190] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, "RTP Payload Format for Scalable Video Coding", RFC 6190, DOI 10.17487/RFC6190, May 2011, <>.

[RFC6190]Wenger,S.,Wang,Y.,Schierl,T.,和A.Eleftheriadis,“可伸缩视频编码的RTP有效载荷格式”,RFC 6190,DOI 10.17487/RFC6190,2011年5月<>.

[RFC6295] Lazzaro, J. and J. Wawrzynek, "RTP Payload Format for MIDI", RFC 6295, DOI 10.17487/RFC6295, June 2011, <>.

[RFC6295]Lazzaro,J.和J.Wawrzynek,“MIDI的RTP有效载荷格式”,RFC 6295,DOI 10.17487/RFC6295,2011年6月<>.

[RFC6354] Xie, Q., "Forward-Shifted RTP Redundancy Payload Support", RFC 6354, DOI 10.17487/RFC6354, August 2011, <>.

[RFC6354]谢,Q,“前向移位RTP冗余有效载荷支持”,RFC 6354,DOI 10.17487/RFC6354,2011年8月<>.

[RFC6363] Watson, M., Begen, A., and V. Roca, "Forward Error Correction (FEC) Framework", RFC 6363, DOI 10.17487/RFC6363, October 2011, <>.

[RFC6363]Watson,M.,Begen,A.和V.Roca,“前向纠错(FEC)框架”,RFC 6363,DOI 10.17487/RFC6363,2011年10月<>.

[RFC6410] Housley, R., Crocker, D., and E. Burger, "Reducing the Standards Track to Two Maturity Levels", BCP 9, RFC 6410, DOI 10.17487/RFC6410, October 2011, <>.

[RFC6410]Housley,R.,Crocker,D.,和E.Burger,“将标准轨道降低到两个成熟度水平”,BCP 9,RFC 6410,DOI 10.17487/RFC6410,2011年10月<>.

[RFC6562] Perkins, C. and JM. Valin, "Guidelines for the Use of Variable Bit Rate Audio with Secure RTP", RFC 6562, DOI 10.17487/RFC6562, March 2012, <>.

[RFC6562]Perkins,C.和JM。Valin,“带安全RTP的可变比特率音频使用指南”,RFC 6562,DOI 10.17487/RFC6562,2012年3月<>.

[RFC6597] Downs, J., Ed. and J. Arbeiter, Ed., "RTP Payload Format for Society of Motion Picture and Television Engineers (SMPTE) ST 336 Encoded Data", RFC 6597, DOI 10.17487/RFC6597, April 2012, <>.

[RFC6597]Downs,J.,Ed.和J.Arbeiter,Ed.,“电影和电视工程师学会(SMPTE)ST 336编码数据的RTP有效载荷格式”,RFC 6597,DOI 10.17487/RFC6597,2012年4月<>.

[RFC6679] Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P., and K. Carlberg, "Explicit Congestion Notification (ECN) for RTP over UDP", RFC 6679, DOI 10.17487/RFC6679, August 2012, <>.

[RFC6679]Westerlund,M.,Johansson,I.,Perkins,C.,O'Hanlon,P.,和K.Carlberg,“UDP上RTP的显式拥塞通知(ECN)”,RFC 6679,DOI 10.17487/RFC66792012年8月<>.

[RFC6682] Watson, M., Stockhammer, T., and M. Luby, "RTP Payload Format for Raptor Forward Error Correction (FEC)", RFC 6682, DOI 10.17487/RFC6682, August 2012, <>.

[RFC6682]Watson,M.,Stockhammer,T.和M.Luby,“猛禽前向纠错(FEC)的RTP有效载荷格式”,RFC 6682,DOI 10.17487/RFC6682,2012年8月<>.

[RFC6701] Farrel, A. and P. Resnick, "Sanctions Available for Application to Violators of IETF IPR Policy", RFC 6701, DOI 10.17487/RFC6701, August 2012, <>.

[RFC6701]Farrel,A.和P.Resnick,“适用于违反IETF知识产权政策者的制裁”,RFC 6701,DOI 10.17487/RFC6701,2012年8月<>.

[RFC6838] Freed, N., Klensin, J., and T. Hansen, "Media Type Specifications and Registration Procedures", BCP 13, RFC 6838, DOI 10.17487/RFC6838, January 2013, <>.

[RFC6838]Freed,N.,Klensin,J.和T.Hansen,“介质类型规范和注册程序”,BCP 13,RFC 6838,DOI 10.17487/RFC6838,2013年1月<>.

[RFC7160] Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple Clock Rates in an RTP Session", RFC 7160, DOI 10.17487/RFC7160, April 2014, <>.

[RFC7160]Petit Huguenin,M.和G.Zorn,Ed.,“在RTP会话中支持多个时钟速率”,RFC 7160,DOI 10.17487/RFC7160,2014年4月<>.

[RFC7164] Gross, K. and R. Brandenburg, "RTP and Leap Seconds", RFC 7164, DOI 10.17487/RFC7164, March 2014, <>.

[RFC7164]Gross,K.和R.Brandenburg,“RTP和闰秒”,RFC 7164,DOI 10.17487/RFC7164,2014年3月<>.

[RFC7201] Westerlund, M. and C. Perkins, "Options for Securing RTP Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014, <>.

[RFC7201]Westerlund,M.和C.Perkins,“保护RTP会话的选项”,RFC 7201,DOI 10.17487/RFC7201,2014年4月<>.

[RFC7202] Perkins, C. and M. Westerlund, "Securing the RTP Framework: Why RTP Does Not Mandate a Single Media Security Solution", RFC 7202, DOI 10.17487/RFC7202, April 2014, <>.

[RFC7202]Perkins,C.和M.Westerlund,“保护RTP框架:为什么RTP不要求单一媒体安全解决方案”,RFC 7202,DOI 10.17487/RFC7202,2014年4月<>.

[RFC7231] Fielding, R., Ed. and J. Reschke, Ed., "Hypertext Transfer Protocol (HTTP/1.1): Semantics and Content", RFC 7231, DOI 10.17487/RFC7231, June 2014, <>.

[RFC7231]Fielding,R.,Ed.和J.Reschke,Ed.,“超文本传输协议(HTTP/1.1):语义和内容”,RFC 7231,DOI 10.17487/RFC72312014年6月<>.

[RFC7273] Williams, A., Gross, K., van Brandenburg, R., and H. Stokking, "RTP Clock Source Signalling", RFC 7273, DOI 10.17487/RFC7273, June 2014, <>.

[RFC7273]Williams,A.,Gross,K.,van Brandenburg,R.,和H.Stokking,“RTP时钟源信令”,RFC 7273,DOI 10.17487/RFC7273,2014年6月<>.

[RFC7322] Flanagan, H. and S. Ginoza, "RFC Style Guide", RFC 7322, DOI 10.17487/RFC7322, September 2014, <>.

[RFC7322]Flanagan,H.和S.Ginoza,“RFC风格指南”,RFC 7322,DOI 10.17487/RFC7322,2014年9月<>.

[RFC7587] Spittka, J., Vos, K., and JM. Valin, "RTP Payload Format for the Opus Speech and Audio Codec", RFC 7587, DOI 10.17487/RFC7587, June 2015, <>.

[RFC7587]Spittka,J.,Vos,K.,和JM。Valin,“Opus语音和音频编解码器的RTP有效载荷格式”,RFC 7587,DOI 10.17487/RFC7587,2015年6月<>.

[RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms for Real-Time Transport Protocol (RTP) Sources", RFC 7656, DOI 10.17487/RFC7656, November 2015, <>.

[RFC7656]Lennox,J.,Gross,K.,Nandakumar,S.,Salgueiro,G.,和B.Burman,Ed.,“实时传输协议(RTP)源的语义和机制分类”,RFC 7656,DOI 10.17487/RFC7656,2015年11月<>.

[RFC7667] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 7667, DOI 10.17487/RFC7667, November 2015, <>.

[RFC7667]Westerlund,M.和S.Westerlund,M.和S.Wenger,“RTP拓扑”,RFC 7667,DOI 10.17487/RFC7667,2015年11月<>.

[RFC7741] Westin, P., Lundin, H., Glover, M., Uberti, J., and F. Galligan, "RTP Payload Format for VP8 Video", RFC 7741, DOI 10.17487/RFC7741, March 2016, <>.

[RFC7741]威斯汀,P.,伦丁,H.,格洛弗,M.,尤伯蒂,J.,和F.加里根,“VP8视频的RTP有效载荷格式”,RFC 7741,DOI 10.17487/RFC77412016年3月<>.

[RFC7798] Wang, Y., Sanchez, Y., Schierl, T., Wenger, S., and M. Hannuksela, "RTP Payload Format for High Efficiency Video Coding (HEVC)", RFC 7798, DOI 10.17487/RFC7798, March 2016, <>.

[RFC7798]Wang,Y.,Sanchez,Y.,Schierl,T.,Wenger,S.,和M.Hannuksela,“高效视频编码(HEVC)的RTP有效载荷格式”,RFC 7798,DOI 10.17487/RFC7798,2016年3月<>.

[RFC7826] Schulzrinne, H., Rao, A., Lanphier, R., Westerlund, M., and M. Stiemerling, Ed., "Real-Time Streaming Protocol Version 2.0", RFC 7826, DOI 10.17487/RFC7826, December 2016, <>.

[RFC7826]Schulzrinne,H.,Rao,A.,Lanphier,R.,Westerlund,M.,和M.Stiemering,Ed.,“实时流协议版本2.0”,RFC 7826,DOI 10.17487/RFC78262016年12月<>.

[RFC7990] Flanagan, H., "RFC Format Framework", RFC 7990, DOI 10.17487/RFC7990, December 2016, <>.

[RFC7990]Flanagan,H.,“RFC格式框架”,RFC 7990,DOI 10.17487/RFC7990,2016年12月<>.

[RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions", RFC 8083, DOI 10.17487/RFC8083, March 2017, <>.

[RFC8083]Perkins,C.和V.Singh,“多媒体拥塞控制:单播RTP会话的断路器”,RFC 8083,DOI 10.17487/RFC8083,2017年3月<>.

[TAO] Hoffman, P., Ed., "The Tao of IETF: A Novice's Guide to the Internet Engineering Task Force", November 2012, <>.


[TRACKER] "IETF Datatracker", <>.


Appendix A. RTP Payload Format Template

This section contains a template for writing an RTP payload format in the form of an Internet-Draft. Text within [...] are instructions and must be removed from the draft itself. Some text proposals that are included are conditional. "..." is used to indicate where further text should be written.


A.1. Title
A.1. 标题

[The title shall be descriptive but as compact as possible. RTP is allowed and recommended abbreviation in the title]


RTP payload format for ...


A.2. Front-Page Boilerplate
A.2. 头版样板

Status of this Memo


[Insert the IPR notice and copyright boilerplate from BCP 78 and 79 that applies to this draft.]

[插入BCP 78和79中适用于本草案的知识产权声明和版权模板。]

[Insert the current Internet-Draft document explanation. At the time of publishing it was:]


Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet-Drafts is at


Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress."


A.3. Abstract
A.3. 摘要

[A payload format abstract should mention the capabilities of the format, for which media format is used, and a little about that codec formats capabilities. Any abbreviation used in the payload format must be spelled out here except the very well known like RTP. No citations are allowed, and no use of language from RFC 2119 either.]

[有效负载格式摘要应提及使用媒体格式的格式的功能,以及编解码器格式的一些功能。有效负载格式中使用的任何缩写必须在此处拼写,但众所周知的RTP除外。不允许引用,也不允许使用RFC 2119中的语言。]

A.4. Table of Contents
A.4. 目录

[If your draft is approved for publication as an RFC, a Table of Contents is required, per [RFC7322].]


A.5. Introduction
A.5. 介绍

[The Introduction should provide a background and overview of the payload format's capabilities. No normative language in this section, i.e., no MUST, SHOULDs etc.]


A.6. Conventions, Definitions, and Abbreviations
A.6. 约定、定义和缩写

[Define conventions, definitions, and abbreviations used in the document in this section. The most common definition used in RTP payload formats are the RFC 2119 definitions of the uppercase normative words, e.g., MUST and SHOULD.]

[定义本节文件中使用的约定、定义和缩写。RTP有效载荷格式中最常用的定义是RFC 2119大写规范性词语的定义,例如必须和应该。]

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.

本文件中的关键词“必须”、“不得”、“要求”、“应”、“不得”、“应”、“不应”、“建议”、“可”和“可选”应按照RFC 2119中的说明进行解释。

A.7. Media Format Description
A.7. 媒体格式说明

[The intention of this section is to enable reviewers and persons to get an overview of the capabilities and major properties of the media format. It should be kept short and concise and is not a complete replacement for reading the media format specification.]


A.8. Payload Format
A.8. 有效载荷格式

[Overview of payload structure]


A.8.1. RTP Header Usage
A.8.1. RTP头使用

[RTP header usage needs to be defined. The fields that absolutely need to be defined are timestamp and marker bit. Further fields may be specified if used. All the rest should be left to their RTP specification definition.]


The remaining RTP header fields are used as specified in RTP [RFC3550].


A.8.2. Payload Header
A.8.2. 有效载荷头

[Define how the payload header, if it exists, is structured and used.]


A.8.3. Payload Data
A.8.3. 有效载荷数据

[The payload data, i.e., what the media codec has produced. Commonly done through reference to the media codec specification, which defines how the data is structured. Rules for padding may need to be defined to bring data to octet alignment.]


A.9. Payload Examples
A.9. 有效载荷示例

[One or more examples are good to help ease the understanding of the RTP payload format.]


A.10. Congestion Control Considerations
A.10. 拥塞控制考虑因素

[This section is to describe the possibility to vary the bitrate as a response to congestion. Below is also a proposal for an initial text that reference RTP and profiles definition of congestion control.]


Congestion control for RTP SHALL be used in accordance with RFC 3550 [RFC3550], and with any applicable RTP profile: e.g., RFC 3551 [RFC3551]. An additional requirement if best-effort service is being used is users of this payload format MUST monitor packet loss to ensure that the packet loss rate is within acceptable parameters. Circuit Breakers [RFC8083] is an update to RTP [RFC3550] that defines criteria for when one is required to stop sending RTP Packet Streams. The circuit breakers is to be implemented and followed.

RTP的拥塞控制应根据RFC 3550[RFC3550]和任何适用的RTP配置文件使用:例如,RFC 3551[RFC3551]。如果使用尽力而为服务,另一个要求是此有效负载格式的用户必须监控数据包丢失,以确保数据包丢失率在可接受的参数范围内。断路器[RFC8083]是对RTP[RFC3550]的更新,它定义了何时需要停止发送RTP数据包流的标准。应实施并遵循断路器。

A.11. Payload Format Parameters
A.11. 有效载荷格式参数

This RTP payload format is identified using the ... media type, which is registered in accordance with RFC 4855 [RFC4855] and using the template of RFC 6838 [RFC6838].

此RTP有效负载格式使用。。。媒体类型,根据RFC 4855[RFC4855]注册,并使用RFC 6838[RFC6838]模板。

A.11.1. Media Type Definition
A.11.1. 媒体类型定义

[Here the media type registration template from RFC 6838 is placed and filled out. This template is provided with some common RTP boilerplate.]

[此处放置并填写来自RFC 6838的媒体类型注册模板。此模板随一些常见RTP样板提供。]

Type name:


Subtype name:


Required parameters:


Optional parameters:


Encoding considerations:


This media type is framed and binary; see Section 4.8 in RFC 6838 [RFC6838].

这种媒体类型是框架和二进制的;参见RFC 6838[RFC6838]中的第4.8节。

Security considerations:


Please see the Security Considerations section in RFC XXXX

请参阅RFC XXXX中的安全注意事项部分

Interoperability considerations:


Published specification:


Applications that use this media type:


Additional information:


Deprecated alias names for this type:


[Only applicable if there exists widely deployed alias for this media type; see Section 4.2.9 of [RFC6838]. Remove or use N/A otherwise.]


Magic number(s):


[Only applicable for media types that has file format specification. Remove or use N/A otherwise.]


File extension(s):


[Only applicable for media types that has file format specification. Remove or use N/A otherwise.]


Macintosh file type code(s):


[Only applicable for media types that has file format specification. Even for file formats they can be skipped as they are not relied on after Mac OS 9.X. Remove or use N/A otherwise.]

[仅适用于具有文件格式规范的媒体类型。即使对于文件格式,也可以跳过它们,因为它们在Mac OS 9.X之后不受依赖。否则请删除或使用N/A。]

Person & email address to contact for further information:


Intended usage:




Restrictions on usage:


[The below text is for media types that is only defined for RTP payload formats. There exist certain media types that are defined both as RTP payload formats and file transfer. The rules for such types are documented in RFC 4855 [RFC4855].]

[以下文本适用于仅为RTP有效负载格式定义的媒体类型。存在同时定义为RTP有效负载格式和文件传输的某些媒体类型。RFC 4855[RFC4855]中记录了此类类型的规则。]

This media type depends on RTP framing and, hence, is only defined for transfer via RTP [RFC3550]. Transport within other framing protocols is not defined at this time.




Change controller:


IETF Payload working group delegated from the IESG.


Provisional registration? (standards tree only):



(Any other information that the author deems interesting may be added below this line.)


[From RFC 6838:

[来自RFC 6838:

"N/A", written exactly that way, can be used in any field if desired to emphasize the fact that it does not apply or that the question was not omitted by accident. Do not use 'none' or other words that could be mistaken for a response.


Limited-use media types should also note in the applications list whether or not that list is exhaustive.]


A.11.2. Mapping to SDP
A.11.2. 映射到SDP

The mapping of the above defined payload format media type and its parameters SHALL be done according to Section 3 of RFC 4855 [RFC4855].

应根据RFC 4855[RFC4855]第3节的规定,映射上述定义的有效负载格式媒体类型及其参数。

[More specific rules only need to be included if some parameter does not match these rules.]


A.11.2.1. Offer/Answer Considerations
A.11.2.1. 报价/答复注意事项

[Here write your Offer/Answer considerations section; please see Section for help.]


A.11.2.2. Declarative SDP Considerations
A.11.2.2. 声明性SDP注意事项

[Here write your considerations for declarative SDP, please see Section for help.]


A.12. IANA Considerations
A.12. IANA考虑

This memo requests that IANA registers [insert media type name here] as specified in Appendix A.11.1. The media type is also requested to be added to the IANA registry for "RTP Payload Format MIME types" <>.


[See Section 7.4 and consider if any of the parameter needs a registered name space.]


A.13. Security Considerations
A.13. 安全考虑

[See Section 7.2.]


RTP packets using the payload format defined in this specification are subject to the security considerations discussed in the RTP specification [RFC3550] , and in any applicable RTP profile such as RTP/AVP [RFC3551], RTP/AVPF [RFC4585], RTP/SAVP [RFC3711], or RTP/ SAVPF [RFC5124]. However, as "Securing the RTP Protocol Framework: Why RTP Does Not Mandate a Single Media Security Solution" [RFC7202] discusses, it is not an RTP payload format's responsibility to discuss or mandate what solutions are used to meet the basic security goals like confidentiality, integrity, and source authenticity for RTP in general. This responsibility lays on anyone using RTP in an application. They can find guidance on available security mechanisms and important considerations in "Options for Securing RTP Sessions" [RFC7201]. Applications SHOULD use one or more appropriate strong security mechanisms. The rest of this Security Considerations section discusses the security impacting properties of the payload format itself.


This RTP payload format and its media decoder do not exhibit any significant non-uniformity in the receiver-side computational complexity for packet processing, and thus are unlikely to pose a denial-of-service threat due to the receipt of pathological data. Nor does the RTP payload format contain any active content.


[The previous paragraph may need editing due to the format breaking either of the statements. Fill in here any further potential security threats created by the payload format itself.]


A.14. RFC Editor Considerations
A.14. RFC编辑器注意事项

Note to RFC Editor: This section may be removed after carrying out all the instructions of this section.


RFC XXXX is to be replaced by the RFC number this specification receives when published.

RFC XXXX将替换为本规范发布时收到的RFC编号。

A.15. References
A.15. 工具书类

[References must be classified as either normative or informative and added to the relevant section. References should use descriptive reference tags.]


A.15.1. Normative References
A.15.1. 规范性引用文件

[Normative references are those that are required to be used to correctly implement the payload format. Also, when requirements language is used, as in the sample text for "Congestion Control Considerations" above, there should be a normative reference to [RFC2119].]


A.15.2. Informative References
A.15.2. 资料性引用

[All other references.]


A.16. Authors' Addresses
A.16. 作者地址

[All authors need to include their name and email address as a minimum: postal mail and possibly phone numbers are included commonly.]


[The Template Ends Here!]




The author would like to thank the individuals who have provided input to this document. These individuals include Richard Barnes, Ali C. Begen, Bo Burman, Ross Finlayson, Russ Housley, John Lazzaro, Jonathan Lennox, Colin Perkins, Tom Taylor, Stephan Wenger, and Qin Wu.




The author would like to thank Tom Taylor for the editing pass of the whole document and contributing text regarding proprietary RTP payload formats. Thanks also goes to Thomas Schierl who contributed text regarding Media Scalability features in payload formats (Section 5.1.5). Stephan Wenger has contributed text on the need to understand the media coding (Section 3.1) as well as joint development of payload format with the media coding (Section 4.4).

作者要感谢Tom Taylor编辑了整个文档,并提供了有关专有RTP有效负载格式的文本。还要感谢Thomas Schierl,他提供了有关有效负载格式的媒体可伸缩性特性的文本(第5.1.5节)。Stephan Wenger就需要理解媒体编码(第3.1节)以及有效载荷格式与媒体编码的联合开发(第4.4节)提供了文本。

Author's Address


Magnus Westerlund Ericsson Farogatan 2 SE-164 80 Kista Sweden

Magnus Westerlund Ericsson Farogatan 2 SE-164 80瑞典基斯塔

   Phone: +46 10 714 82 87
   Phone: +46 10 714 82 87