Network Working Group                                         R. Kreuter
Request for Comments: 4040                                    Siemens AG
Category: Standards Track                                     April 2005
        
Network Working Group                                         R. Kreuter
Request for Comments: 4040                                    Siemens AG
Category: Standards Track                                     April 2005
        

RTP Payload Format for a 64 kbit/s Transparent Call

64 kbit/s透明调用的RTP有效负载格式

Status of This Memo

关于下段备忘

This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards" (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited.

本文件规定了互联网社区的互联网标准跟踪协议,并要求进行讨论和提出改进建议。有关本协议的标准化状态和状态,请参考当前版本的“互联网官方协议标准”(STD 1)。本备忘录的分发不受限制。

Copyright Notice

版权公告

Copyright (C) The Internet Society (2005).

版权所有(C)互联网协会(2005年)。

Abstract

摘要

This document describes how to carry 64 kbit/s channel data transparently in RTP packets, using a pseudo-codec called "Clearmode". It also serves as registration for a related MIME type called "audio/clearmode".

本文档描述了如何使用名为“Clearmode”的伪编解码器在RTP数据包中透明地传输64 kbit/s通道数据。它还可以注册一个名为“audio/clearmode”的相关MIME类型。

"Clearmode" is a basic feature of VoIP Media Gateways.

“Clearmode”是VoIP媒体网关的基本功能。

Table of Contents

目录

   1.  Introduction..................................................  2
   2.  Conventions Used in This Document.............................  2
   3.  64 kbit/s Data Stream Handling and RTP Header Parameters......  3
   4.  IANA Considerations...........................................  3
   5.  Mapping to Session Description Protocol (SDP) Parameters......  5
   6.  Security Considerations.......................................  5
   7.  References....................................................  6
       7.1. Normative References.....................................  6
       7.2. Informative References...................................  6
   8.  Acknowledgements..............................................  7
        
   1.  Introduction..................................................  2
   2.  Conventions Used in This Document.............................  2
   3.  64 kbit/s Data Stream Handling and RTP Header Parameters......  3
   4.  IANA Considerations...........................................  3
   5.  Mapping to Session Description Protocol (SDP) Parameters......  5
   6.  Security Considerations.......................................  5
   7.  References....................................................  6
       7.1. Normative References.....................................  6
       7.2. Informative References...................................  6
   8.  Acknowledgements..............................................  7
        
1. Introduction
1. 介绍

Voice over IP (VoIP) Media Gateways need to carry all possible data streams generated by analog terminals or integrated services digital network (ISDN) terminals via an IP network. Within this document a

IP语音(VoIP)媒体网关需要通过IP网络承载模拟终端或综合业务数字网(ISDN)终端生成的所有可能的数据流。在本文件中

VoIP Media Gateway is a device that converts a (digital or analog) linear data stream to a digital packetized data stream or vice versa. Refer to RFC 2719 [10] for an introduction into the basic architecture of a Media Gateway based network.

VoIP媒体网关是一种将(数字或模拟)线性数据流转换为数字分组数据流的设备,反之亦然。有关基于媒体网关的网络的基本架构的介绍,请参阅RFC 2719[10]。

Usually a VoIP Media Gateway does some processing on the data it converts besides packetization or depacketization; i.e. echo cancellation or dual tone multifrequency (DTMF) detection, and especially a coding/decoding. But there is a class of data streams that does not rely on or allow any data processing within the VoIP Media Gateway except for packetization or depacketization. ISDN data terminals i.e. will produce data streams that are not compatible with a non-linear encoding as used for voice.

通常,VoIP媒体网关除了对其转换的数据进行打包或解除打包外,还对其转换的数据进行一些处理;i、 e.回声消除或双音多频(DTMF)检测,尤其是编码/解码。但有一类数据流不依赖或不允许VoIP媒体网关内的任何数据处理,除了打包或解除打包。ISDN数据终端,即将产生与用于语音的非线性编码不兼容的数据流。

For such applications, there is a necessity for a transparent relay of 64 kbit/s data streams in real-time transport protocol (RTP) [4] packets. This mode is often referred to as "clear-channel data" or "64 kbit/s unrestricted". No encoder/decoder is needed in that case, but a unique RTP payload type is necessary and a related MIME type is to be registered for signaling purposes.

对于此类应用,实时传输协议(RTP)[4]数据包中需要64 kbit/s数据流的透明中继。此模式通常称为“清除通道数据”或“64 kbit/s无限制”。在这种情况下不需要编码器/解码器,但需要唯一的RTP有效负载类型,并且为了信令目的,需要注册相关的MIME类型。

Clearmode is not restricted to the examples described above. It can be used by any application, that does not need a special encoding/decoding for transfer via a RTP connection.

Clearmode不限于上述示例。它可以被任何应用程序使用,不需要特殊的编码/解码来通过RTP连接进行传输。

This payload format document describes a pseudo-codec called "Clearmode", for sample oriented 64 kbit/s data streams with 8 bits per sample. It is in accordance with RFC 2736 [1], which provides a guideline for the specification of new RTP payload formats.

此有效负载格式文档描述了一个称为“Clearmode”的伪编解码器,用于每个样本8位的面向样本的64 kbit/s数据流。它符合RFC 2736[1],RFC 2736[1]为新RTP有效载荷格式的规范提供了指南。

Examples for the current use of Clearmode are the transfer of "ISDN 7 kHz voice" and "ISDN data" in VoIP Media Gateways.

当前使用Clearmode的示例包括在VoIP媒体网关中传输“ISDN 7 kHz语音”和“ISDN数据”。

This document also serves as the MIME type registration according to RFC 2045 [2] and RFC 2048 [3], which defines procedures for registration of new MIME types within the IETF tree.

根据RFC 2045[2]和RFC 2048[3],本文档还作为MIME类型注册,其定义了在IETF树中注册新MIME类型的过程。

2. Conventions Used in This Document
2. 本文件中使用的公约

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 [8].

本文件中的关键词“必须”、“不得”、“要求”、“应”、“不应”、“应”、“不应”、“建议”、“可”和“可选”应按照RFC 2119[8]中所述进行解释。

3. 64 kbit/s Data Stream Handling and RTP Header Parameters
3. 64 kbit/s数据流处理和RTP头参数

Clearmode does not use any encoding or decoding. It just provides packetization.

Clearmode不使用任何编码或解码。它只是提供了打包功能。

Clearmode assumes that the data to be handled is sample oriented with one octet (8bits) per sample. There is no restriction on the number of samples per packet other than the 64 kbyte limit imposed by the IP protocol. The number of samples SHOULD be less than the path maximum transmission unit (MTU) minus combined packet header length. If the environment is expected to have tunnels or security encapsulation as part of operation, the number of samples SHOULD be reduced to allow for the extra header space.

Clearmode假设要处理的数据是面向样本的,每个样本有一个八位字节(8bit)。除了IP协议施加的64 kbyte限制外,对每个数据包的采样数没有任何限制。采样数应小于路径最大传输单元(MTU)减去组合数据包报头长度。如果预期环境将隧道或安全封装作为操作的一部分,则应减少样本数量以允许额外的头空间。

The payload packetization/depacketization for Clearmode is similar to the Pulse Code Modulation (PCMU or PCMA) handling described in RFC 3551 [5]. Each Clearmode octet SHALL be octet-aligned in an RTP packet. The sign bit of each octet SHALL correspond to the most significant bit of the octet in the RTP packet.

Clearmode的有效载荷打包/解打包类似于RFC 3551[5]中描述的脉冲编码调制(PCMU或PCMA)处理。每个Clearmode八位字节应在RTP数据包中对齐。每个八位字节的符号位应对应于RTP数据包中八位字节的最高有效位。

A sample rate of 8000 Hz MUST be used. This calculates to a 64 kbit/s transmission rate per channel.

必须使用8000 Hz的采样率。这将计算每个通道64 kbit/s的传输速率。

The Timestamp SHALL be set as described in RFC 3550 [4].

时间戳的设置应符合RFC 3550[4]的规定。

The marker bit is always zero. Silence suppression is not applicable for Clearmode data streams.

标记位始终为零。静音抑制不适用于Clearmode数据流。

The payload type is dynamically assigned and is not presented in this document.

有效负载类型是动态分配的,不在本文档中显示。

RTP header fields not mentioned here SHALL be used as specified in RFC 3550 [4] and any applicable profile.

此处未提及的RTP标题字段应按照RFC 3550[4]和任何适用概要文件的规定使用。

This document specifies the use of RTP over unicast and multicast UDP as well as TCP. (This does not preclude the use of this definition when RTP is carried by other lower-layer protocols.)

本文档指定在单播和多播UDP以及TCP上使用RTP。(当RTP由其他较低层协议承载时,这并不排除使用此定义。)

4. IANA Considerations
4. IANA考虑

This document registers the following MIME subtype: audio/clearmode.

本文档注册了以下MIME子类型:audio/clearmode。

To: ietf-types@iana.org

致:ietf-types@iana.org

Subject: Registration of MIME media type audio/clearmode

主题:注册MIME媒体类型音频/清晰模式

MIME media type name: audio

MIME媒体类型名称:音频

MIME subtype name: clearmode

MIME子类型名称:clearmode

Required parameters: none

所需参数:无

Optional parameters: ptime, maxptime

可选参数:ptime、maxptime

"ptime" gives the length of time in milliseconds represented by the media in a packet, as described in RFC 2327 [6].

“ptime”给出数据包中媒体表示的时间长度(毫秒),如RFC 2327[6]所述。

"maxptime" represents the maximum amount of media, which can be encapsulated in each packet, expressed as time in milliseconds, as described in RFC 3267 [9].

“maxptime”表示可封装在每个数据包中的最大媒体量,以毫秒表示,如RFC 3267[9]中所述。

Encoding considerations:

编码注意事项:

This type is only defined for transfer via RTP [4].

此类型仅定义为通过RTP传输[4]。

Security considerations:

安全考虑:

See Section 6 of RFC 4040

参见RFC 4040第6节

Interoperability considerations: none

互操作性注意事项:无

Published specification: RFC 4040

已发布规范:RFC 4040

Applications, which use this media type:

使用此媒体类型的应用程序:

Voice over IP Media Gateways, transferring "ISDN 64 kb/s data", "ISDN 7 kHz voice", or other 64 kbit/s data streams via an RTP connection

IP媒体网关上的语音,通过RTP连接传输“ISDN 64 kb/s数据”、“ISDN 7 kHz语音”或其他64 kbit/s数据流

Note: the choice of the "audio" top-level MIME type was made because the dominant uses of this pseudo-codec are expected to telephony and voice-gateway-related. The "audio" type allows the use of sharing of the port in the SDP "m=" line with codecs such as audio/g711 [6], [7], for one example. This sharing is an important application and would not be possible otherwise.

注意:之所以选择“音频”顶级MIME类型,是因为该伪编解码器的主要用途预计与电话和语音网关相关。例如,“audio”类型允许使用诸如audio/g711[6]、[7]等编解码器共享SDP“m=”行中的端口。这种共享是一个重要的应用程序,否则是不可能的。

Additional information: none

其他信息:无

Intended usage: COMMON

预期用途:普通

Author/Change controller:

作者/变更控制员:

IETF Audio/Video transport working group delegated from the IESG

IESG授权的IETF音频/视频传输工作组

5. Mapping to Session Description Protocol (SDP) Parameters
5. 映射到会话描述协议(SDP)参数

Parameters are mapped to SDP [6] in a standard way.

参数以标准方式映射到SDP[6]。

o The MIME type (audio) goes in SDP "m=" as the media name.

o MIME类型(音频)以SDP“m=”作为媒体名称。

o The MIME subtype (clearmode) goes in SDP "a=rtpmap" as the encoding name.

o MIME子类型(clearmode)以SDP“a=rtpmap”作为编码名称。

o The optional parameters "ptime" and "maxptime" go in the SDP "a=ptime" and "a=maxptime" attributes, respectively.

o 可选参数“ptime”和“maxptime”分别位于SDP“a=ptime”和“a=maxptime”属性中。

An example mapping is as follows:

映射示例如下所示:

       audio/clearmode; ptime=10
        
       audio/clearmode; ptime=10
        
       m=audio 12345 RTP/AVP 97
       a=rtpmap:97 CLEARMODE/8000
       a=ptime:10
        
       m=audio 12345 RTP/AVP 97
       a=rtpmap:97 CLEARMODE/8000
       a=ptime:10
        

Note that the payload format (encoding) names defined in the RTP Profile are commonly shown in upper case. MIME subtypes are commonly shown in lower case. These names are case-insensitive in both places.

请注意,RTP配置文件中定义的有效负载格式(编码)名称通常以大写形式显示。MIME子类型通常以小写形式显示。这些名称在两个位置都不区分大小写。

6. Security Considerations
6. 安全考虑

Implementations using the payload format defined in this specification are subject to the security considerations discussed in the RFC 3550 [4]. The payload format described in this document does not specify any different security services. The primary function of this payload format is to add a transparent transport for a 64 kbit/s data stream.

使用本规范中定义的有效负载格式的实现应遵守RFC 3550[4]中讨论的安全注意事项。本文档中描述的有效负载格式未指定任何不同的安全服务。此有效负载格式的主要功能是为64 kbit/s数据流添加透明传输。

Confidentiality of the media streams is achieved by encryption, for example by application of the Secure RTP profile [11].

媒体流的机密性通过加密实现,例如通过应用安全RTP配置文件[11]。

As with any IP-based protocol, in some circumstances a receiver may be overloaded simply by the receipt of too many packets, either desired or undesired. Network-layer authentication MAY be used to discard packets from undesired sources, but the processing cost of the authentication itself may be too high. Overload can also occur, if the sender chooses to use a smaller packetization period, than the receiver can process. The ptime parameter can be used to negotiate an appropriate packetization during session setup.

与任何基于IP的协议一样,在某些情况下,接收机可能仅仅因为接收了太多的数据包而过载,不管是想要的还是不想要的。网络层认证可用于丢弃来自不希望的源的数据包,但认证本身的处理成本可能过高。如果发送方选择使用比接收方能够处理的更小的打包周期,也可能发生过载。ptime参数可用于在会话设置期间协商适当的打包。

In general RTP is not an appropriate transfer protocol for reliable octet streams. TCP is better in those cases. Besides that, packet loss due to congestion is as much an issue for clearmode, as for other payload formats. Refer to RFC 3551 [5], section 2, for a discussion of this issue.

一般来说,RTP不是可靠八位组流的合适传输协议。在这些情况下,TCP更好。除此之外,对于clearmode和其他有效负载格式而言,拥塞导致的数据包丢失同样是一个问题。有关此问题的讨论,请参阅RFC 3551[5],第2节。

7. References
7. 工具书类
7.1. Normative References
7.1. 规范性引用文件

[1] Handley, M. and C. Perkins, "Guidelines for Writers of RTP Payload Format Specifications", BCP 36, RFC 2736, December 1999.

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

[2] Freed, N. and N. Borenstein, "Multipurpose Internet Mail Extensions (MIME) Part One: Format of Internet Message Bodies", RFC 2045, November 1996.

[2] Freed,N.和N.Borenstein,“多用途互联网邮件扩展(MIME)第一部分:互联网邮件正文格式”,RFC 20451996年11月。

[3] Freed, N., Klensin, J., and J. Postel, "Multipurpose Internet Mail Extensions (MIME) Part Four: Registration Procedures", BCP 13, RFC 2048, November 1996.

[3] Freed,N.,Klensin,J.,和J.Postel,“多用途互联网邮件扩展(MIME)第四部分:注册程序”,BCP 13,RFC 2048,1996年11月。

[4] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003.

[4] Schulzrinne,H.,Casner,S.,Frederick,R.,和V.Jacobson,“RTP:实时应用的传输协议”,STD 64,RFC 35502003年7月。

[5] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, July 2003.

[5] 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 3551,2003年7月。

[6] Handley, M. and V. Jacobson, "SDP: Session Description Protocol", RFC 2327, April 1998.

[6] Handley,M.和V.Jacobson,“SDP:会话描述协议”,RFC 2327,1998年4月。

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

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

[8] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997.

[8] Bradner,S.,“RFC中用于表示需求水平的关键词”,BCP 14,RFC 2119,1997年3月。

[9] Sjoberg, J., Westerlund, M., Lakaniemi, A., and Q. Xie, "Real-Time Transport Protocol (RTP) Payload Format and File Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs", RFC 3267, June 2002.

[9] 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)音频编解码器的实时传输协议(RTP)有效载荷格式和文件存储格式”,RFC 3267,2002年6月。

7.2. Informative References
7.2. 资料性引用

[10] Ong, L., Rytina, I., Garcia, M., Schwarzbauer, H., Coene, L., Lin, H., Juhasz, I., Holdrege, M., and C. Sharp, "Framework Architecture for Signaling Transport", RFC 2719, October 1999.

[10] Ong,L.,Rytina,I.,Garcia,M.,Schwarzbauer,H.,Coene,L.,Lin,H.,Juhasz,I.,Holdrege,M.,和C.Sharp,“信号传输的框架架构”,RFC 2719,1999年10月。

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

[11] Baugher,M.,McGrew,D.,Naslund,M.,Carrara,E.,和K.Norrman,“安全实时传输协议(SRTP)”,RFC 37112004年3月。

8. Acknowledgements
8. 致谢

The editor would like to acknowledge the help of the IETF AVT Working Group and, in particular the help of Colin Perkins and Magnus Westerlund for their intensive reviews and comments.

编辑要感谢IETF AVT工作组的帮助,特别是Colin Perkins和Magnus Westerlund的帮助,感谢他们的深入评论和评论。

Author's Address

作者地址

Ruediger Kreuter Siemens AG 81730 Munich, Germany

德国慕尼黑Ruediger Kreuter西门子公司81730

   EMail: ruediger.kreuter@siemens.com
        
   EMail: ruediger.kreuter@siemens.com
        

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完整版权声明

Copyright (C) The Internet Society (2005).

版权所有(C)互联网协会(2005年)。

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Acknowledgement

确认

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