Network Working Group G. Hellstrom
Request for Comments: 4351 Omnitor AB
Category: Historic P. Jones
Cisco Systems, Inc.
January 2006
Network Working Group G. Hellstrom
Request for Comments: 4351 Omnitor AB
Category: Historic P. Jones
Cisco Systems, Inc.
January 2006
Real-Time Transport Protocol (RTP) Payload for Text Conversation Interleaved in an Audio Stream
实时传输协议（RTP）用于音频流中交织的文本对话的有效负载
Status of This Memo
关于下段备忘
This memo defines a Historic Document for the Internet community. It does not specify an Internet standard of any kind. Distribution of this memo is unlimited.
此备忘录定义了互联网社区的历史文档。它没有规定任何类型的互联网标准。本备忘录的分发不受限制。
Copyright Notice
版权公告
Copyright (C) The Internet Society (2006).
版权所有（C）互联网协会（2006年）。
Abstract
摘要
This memo describes how to carry real-time text conversation session contents in RTP packets. Text conversation session contents are specified in ITU-T Recommendation T.140.
本备忘录描述了如何在RTP数据包中携带实时文本会话内容。ITU-T建议T.140中规定了文本对话会话内容。
One payload format is described for transmitting audio and text data within a single RTP session.
描述了一种用于在单个RTP会话内传输音频和文本数据的有效载荷格式。
This RTP payload description recommends a method to include redundant text from already transmitted packets in order to reduce the risk of text loss caused by packet loss.
该RTP有效负载描述推荐了一种方法，以包括来自已经传输的分组的冗余文本，从而降低由分组丢失引起的文本丢失的风险。
Table of Contents
目录
1. Introduction ....................................................3
2. Conventions Used in This Document ...............................4
3. Usage of RTP ....................................................4
3.1. Motivations and Rationale ..................................4
3.2. Payload Format for Transmission of audio/t140c Data ........4
3.3. The "T140block" ............................................5
3.4. Synchronization of Text with Other Media ...................5
3.5. Synchronization Considerations for the audio/t140c Format ..5
3.6. RTP Packet Header ..........................................6
4. Protection against Loss of Data .................................7
4.1. Payload Format When Using Redundancy .......................7
4.2. Using Redundancy with the audio/t140c Format ...............8
5. Recommended Procedure ...........................................8
5.1. Recommended Basic Procedure ................................8
5.2. Transmission before and after "Idle Periods" ...............9
5.3. Detection of Lost Text Packets .............................9
5.4. Compensation for Packets Out of Order .....................10
6. Parameter for Character Transmission Rate ......................10
7. Examples .......................................................11
7.1. RTP Packetization Examples for the audio/t140c Format .....11
7.2. SDP Examples ..............................................12
8. Security Considerations ........................................13
8.1. Confidentiality ...........................................13
8.2. Integrity .................................................13
8.3. Source Authentication .....................................13
9. Congestion Considerations ......................................14
10. IANA Considerations ...........................................15
10.1. Registration of MIME Media Type audio/t140c ..............15
10.2. SDP Mapping of MIME Parameters ...........................16
10.3. Offer/Answer Consideration ...............................17
11. Acknowledgements ..............................................17
12. Normative References ..........................................17
13. Informative References ........................................18
1. Introduction ....................................................3
2. Conventions Used in This Document ...............................4
3. Usage of RTP ....................................................4
3.1. Motivations and Rationale ..................................4
3.2. Payload Format for Transmission of audio/t140c Data ........4
3.3. The "T140block" ............................................5
3.4. Synchronization of Text with Other Media ...................5
3.5. Synchronization Considerations for the audio/t140c Format ..5
3.6. RTP Packet Header ..........................................6
4. Protection against Loss of Data .................................7
4.1. Payload Format When Using Redundancy .......................7
4.2. Using Redundancy with the audio/t140c Format ...............8
5. Recommended Procedure ...........................................8
5.1. Recommended Basic Procedure ................................8
5.2. Transmission before and after "Idle Periods" ...............9
5.3. Detection of Lost Text Packets .............................9
5.4. Compensation for Packets Out of Order .....................10
6. Parameter for Character Transmission Rate ......................10
7. Examples .......................................................11
7.1. RTP Packetization Examples for the audio/t140c Format .....11
7.2. SDP Examples ..............................................12
8. Security Considerations ........................................13
8.1. Confidentiality ...........................................13
8.2. Integrity .................................................13
8.3. Source Authentication .....................................13
9. Congestion Considerations ......................................14
10. IANA Considerations ...........................................15
10.1. Registration of MIME Media Type audio/t140c ..............15
10.2. SDP Mapping of MIME Parameters ...........................16
10.3. Offer/Answer Consideration ...............................17
11. Acknowledgements ..............................................17
12. Normative References ..........................................17
13. Informative References ........................................18
This document defines a payload type for carrying text conversation session contents in RTP [2] packets. Text conversation session contents are specified in ITU-T Recommendation T.140 [1]. Text conversation is used alone or in connection to other conversational facilities, such as video and voice, to form multimedia conversation services. Text in multimedia conversation sessions is sent character-by-character as soon as it is available, or with a small delay for buffering.
本文档定义了在RTP[2]数据包中承载文本会话内容的有效负载类型。ITU-T建议T.140[1]中规定了文本对话会话内容。文本会话单独使用或与其他会话设施（如视频和语音）结合使用，以形成多媒体会话服务。多媒体对话会话中的文本在可用时会立即逐字符发送，或者会有一个小的缓冲延迟。
The text is intended to be entered by human users from a keyboard, handwriting recognition, voice recognition, or any other input method. The rate of character entry is usually at a level of a few characters per second or less. In general, only one or a few new characters are expected to be transmitted with each packet. Small blocks of text may be prepared by the user and pasted into the user interface for transmission during the conversation, occasionally causing packets to carry more payload.
文本由人类用户通过键盘、手写识别、语音识别或任何其他输入方法输入。字符输入速率通常为每秒几个字符或更少。通常，每个数据包只需要传输一个或几个新字符。用户可以准备小的文本块，并将其粘贴到用户界面中以便在会话期间传输，偶尔会导致数据包携带更多的有效载荷。
T.140 specifies that text and other T.140 elements must be transmitted in ISO 10646-1[5] code with UTF-8 [6] transformation. That makes it easy to implement internationally useful applications and to handle the text in modern information technology environments. The payload of an RTP packet following this specification consists of text encoded according to T.140 without any additional framing. A common case will be a single ISO 10646 character, UTF-8 encoded.
T.140规定文本和其他T.140元素必须通过UTF-8[6]转换以ISO10646-1[5]代码传输。这使得实现国际上有用的应用程序和在现代信息技术环境中处理文本变得容易。遵循本规范的RTP数据包的有效载荷由根据T.140编码的文本组成，没有任何额外的帧。常见的情况是单个ISO10646字符，UTF-8编码。
T.140 requires the transport channel to provide characters without duplication and in original order. Text conversation users expect that text will be delivered with no or a low level of lost information.
T.140要求传输通道以原始顺序提供不重复的字符。文本对话用户希望文本在传递时不会丢失或丢失少量信息。
Therefore a mechanism based on RTP is specified here. It gives text arrival in correct order, without duplication, and with detection and indication of loss. It also includes an optional possibility to repeat data for redundancy to lower the risk of loss. Since packet overhead is usually much larger than the T.140 contents, the increase in bandwidth with the use of redundancy is minimal.
因此，这里指定了一种基于RTP的机制。它使文本以正确的顺序到达，没有重复，并带有丢失检测和指示。它还包括重复数据以实现冗余的可选可能性，以降低丢失风险。由于数据包开销通常比T.140内容大得多，因此使用冗余时带宽的增加最小。
By using RTP for text transmission in a multimedia conversation application, uniform handling of text and other media can be achieved in, as examples, conferencing systems, firewalls, and network translation devices. This, in turn, eases the design and increases the possibility for prompt and proper media delivery.
通过在多媒体会话应用程序中使用RTP进行文本传输，例如，可以在会议系统、防火墙和网络翻译设备中实现文本和其他媒体的统一处理。这反过来又简化了设计，并增加了及时和正确交付媒体的可能性。
This document introduces a method of transporting text interleaved with voice within the same RTP session.
本文档介绍了在同一RTP会话中传输与语音交织的文本的方法。
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 [4].
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[4]中所述进行解释。
The payload format for real-time text transmission with RTP [2] described in this memo is intended for use between Public Switched Telephone Network (PSTN) gateways and is called audio/t140c.
本备忘录中描述的RTP[2]实时文本传输的有效负载格式旨在用于公共交换电话网络（PSTN）网关之间，称为音频/t140c。
The audio/t140c payload specification is intended to allow gateways that are interconnecting two PSTN networks to interleave, through a single RTP session, audio and text data received on the PSTN circuit. This is comparable to the way in which dual-tone multifrequency (DTMF) is extracted and transmitted within an RTP session [14].
音频/t140c有效负载规范旨在允许互连两个PSTN网络的网关通过单个RTP会话交错PSTN电路上接收的音频和文本数据。这与在RTP会话中提取和传输双音多频（DTMF）的方式相当[14]。
The audio/t140c format SHALL NOT be used for applications other than PSTN gateway applications. In such applications, a specific profiling document MAY make it REQUIRED for a specific application. The reason to prefer to use audio/t140c could be for gateway application where the ports are a limited and scarce resource. Applications SHOULD use RFC 4103 [15] for real-time text communication that falls outside the limited scope of this specification.
音频/t140c格式不得用于PSTN网关应用以外的应用。在这些应用程序中，特定的分析文档可能会使特定的应用程序需要它。选择使用audio/t140c的原因可能是因为网关应用程序的端口有限且稀缺。应用程序应使用RFC 4103[15]进行实时文本通信，该通信不属于本规范的限制范围。
An audio/t140c conversation RTP payload format consists of a 16-bit "T140block counter" carried in network byte order (see RFC 791 [11] Annex B), followed by one and only one "T140block" (see section 3.3). The fields in the RTP header are set as defined in section 3.6.
音频/t140c对话RTP有效负载格式由一个16位“T140block计数器”组成，该计数器按网络字节顺序携带（见RFC 791[11]附录B），后跟一个且仅一个“T140block”（见第3.3节）。RTP标题中的字段按照第3.6节中的定义进行设置。
The T140block counter MUST be initialized to zero the first time that a packet containing a T140block is transmitted and MUST be incremented by 1 each time that a new block is transmitted. Once the counter reaches the value 0xFFFF, the counter is reset to 0 the next time the counter is incremented. This T140block counter is used to detect lost blocks and to avoid duplication of blocks.
T140块计数器必须在第一次发送包含T140块的数据包时初始化为零，并且每次发送新块时必须递增1。计数器达到值0xFFFF后，下次计数器递增时，计数器将重置为0。该T140块计数器用于检测丢失的块并避免重复块。
For the purposes of readability, the remainder of this document refers only to the T140block without making explicit reference to the T140block counter. Readers should understand that when using the audio/t140c format, the T140block counter MUST always precede the actual T140block, including redundant data transmissions.
为便于阅读，本文件其余部分仅参考T140块，未明确参考T140块计数器。读者应了解，当使用音频/t140c格式时，T140block计数器必须始终位于实际T140block之前，包括冗余数据传输。
T.140 text is UTF-8 coded as specified in T.140 with no extra framing. The T140block contains one or more T.140 code elements as specified in [1]. Most T.140 code elements are single ISO 10646 [5] characters, but some are multiple-character sequences. Each character is UTF-8 encoded [6] into one or more octets. Each block MUST contain an integral number of UTF-8-encoded characters regardless of the number of octets per character. Any composite character sequence (CCS) SHOULD be placed within one block.
T.140文本按照T.140中的规定进行UTF-8编码，无额外框架。T140块包含[1]中规定的一个或多个T.140代码元素。大多数T.140代码元素是单个ISO10646[5]字符，但有些是多字符序列。每个字符都经过UTF-8编码[6]为一个或多个八位字节。每个块必须包含整数个UTF-8编码字符，而不考虑每个字符的八位字节数。任何复合字符序列（CCS）都应放在一个块内。
Usually, each medium in a session utilizes a separate RTP stream. As such, if synchronization of the text and other media packets is important, the streams MUST be associated when the sessions are established and the streams MUST share the same reference clock (refer to the description of the timestamp field as it relates to synchronization in section 5.1 of RFC 3550). Association of RTP streams can be done through the CNAME field of RTP Control Protocol (RTCP) SDES function. It is dependent on the particular application and is outside the scope of this document.
通常，会话中的每个媒体都使用单独的RTP流。因此，如果文本和其他媒体分组的同步很重要，则在建立会话时必须关联流，并且流必须共享相同的参考时钟（参考RFC 3550第5.1节中与同步相关的时间戳字段的描述）。RTP流的关联可以通过RTP控制协议（RTCP）SDES功能的CNAME字段完成。它取决于特定的应用，不在本文件的范围内。
The audio/t140c packets are generally transmitted as interleaved packets between voice packets or other kinds of audio packets with the intention to create one common audio signal in the receiving equipment to be used for alternating between text and voice. The audio/t140c payload is then used to play out audio signals according to a PSTN textphone coding method (usually a modem).
音频/t140c分组通常作为语音分组或其他种类的音频分组之间的交织分组来发送，目的是在接收设备中创建一个用于在文本和语音之间交替的公共音频信号。然后，音频/t140c有效载荷用于根据PSTN文本电话编码方法（通常是调制解调器）播放音频信号。
One should observe the RTP timestamps of the voice, text, or other audio packets in order to reproduce the stream correctly when playing out the audio. Also, note that incoming text from a PSTN circuit might be at a higher bit-rate than can be played out on an egress PSTN circuit. As such, it is possible that, on the egress side, a gateway may not complete the play out of the text packets before it is time to play the next voice packet. Given that this application is primarily for the benefit of users of PSTN textphone devices, it is strongly RECOMMENDED that all received text packets be properly reproduced on the egress gateway before considering any other subsequent audio packets.
人们应该观察语音、文本或其他音频包的RTP时间戳，以便在播放音频时正确再现流。另外，请注意，来自PSTN电路的输入文本的比特率可能高于在出口PSTN电路上播放的比特率。因此，在出口侧，网关可能在播放下一语音分组的时间到来之前不完成文本分组的播放。鉴于此应用程序主要是为了PSTN textphone设备用户的利益，强烈建议在考虑任何其他后续音频包之前，在出口网关上正确复制所有接收到的文本包。
If necessary, voice and other audio packets should be discarded in order to properly reproduce the text signals on the PSTN circuit, even if the text packets arrive late.
如有必要，应丢弃语音和其他音频数据包，以便在PSTN电路上正确再现文本信号，即使文本数据包到达较晚。
The PSTN textphone users commonly use turn-taking indicators in the text stream, so it can be expected that as long as text is transmitted, it is valid text and should be given priority over voice.
PSTN textphone用户通常在文本流中使用话轮转换指示器，因此可以预期，只要传输文本，它就是有效文本，应该优先于语音。
Note that the usual RTP semantics apply with regards to switching payload formats within an RTP session. A sender MAY switch between "audio/t140c" and some other format within an RTP session, but MUST NOT send overlapping data using two different audio formats within an RTP session. This does not prohibit an implementation from being split into two logical parts to send overlapping data, each part using a different synchronization source (SSRC) and sending its own RTP and RTCP (such an endpoint will appear to others in the session as two participants with different SSRCs, but the same RTCP SDES CNAME). Further details around using multiple payloads in an RTP session can be found in RFC 3550 [2].
请注意，通常的RTP语义适用于RTP会话中的有效负载格式切换。发送方可以在RTP会话中在“音频/t140c”和其他格式之间切换，但不得在RTP会话中使用两种不同的音频格式发送重叠数据。这并不禁止将实现拆分为两个逻辑部分以发送重叠数据，每个部分使用不同的同步源（SSRC）并发送自己的RTP和RTCP（这样的端点在会话中会显示为具有不同SSRC但相同RTCP SDES CNAME的两个参与者）。有关在RTP会话中使用多个有效载荷的更多详细信息，请参见RFC 3550[2]。
Each RTP packet starts with a fixed RTP header. The following fields of the RTP fixed header are specified for T.140 text streams:
每个RTP数据包都以一个固定的RTP报头开始。为T.140文本流指定RTP固定标题的以下字段：
Payload Type (PT): The assignment of an RTP payload type is specific to the RTP profile under which this payload format is used. For profiles that use dynamic payload type number assignment, this payload format can be identified by the MIME type "audio/t140c" (see section 10). If redundancy is used per RFC 2198, another payload type number needs to be provided for the redundancy format. The MIME type for identifying RFC 2198 is available in RFC 3555 [17].
有效负载类型（PT）：RTP有效负载类型的分配特定于使用此有效负载格式的RTP配置文件。对于使用动态有效负载类型编号分配的配置文件，可以通过MIME类型“audio/t140c”来标识此有效负载格式（请参阅第10节）。如果按照RFC 2198使用冗余，则需要为冗余格式提供另一个有效负载类型编号。RFC 3555[17]中提供了用于识别RFC 2198的MIME类型。
Sequence number: The definition of sequence numbers is available in RFC 3550 [2]. Character loss is detected through the T140block counter when using the audio/t140c payload format.
序列号：序列号的定义见RFC 3550[2]。使用音频/t140c有效负载格式时，通过T140block计数器检测到字符丢失。
Timestamp: The RTP Timestamp encodes the approximate instance of entry of the primary text in the packet. For audio/t140c, the clock frequency MAY be set to any value, and SHOULD be set to the same value as for any audio packets in the same RTP stream in order to avoid RTP timestamp rate switching. The value SHOULD be set by out of band mechanisms. Sequential packets MUST NOT use the same timestamp. Since packets do not represent any constant duration, the timestamp cannot be used to directly infer packet loss.
时间戳：RTP时间戳对数据包中主文本输入的近似实例进行编码。对于音频/t140c，时钟频率可以设置为任何值，并且应该设置为与相同RTP流中的任何音频分组相同的值，以避免RTP时间戳速率切换。该值应由带外机制设置。连续数据包不得使用相同的时间戳。由于数据包不代表任何恒定的持续时间，时间戳不能用于直接推断数据包丢失。
M-bit: The M-bit MUST be included. The first packet in a session, and the first packet after an idle period, SHOULD be distinguished by setting the marker bit in the RTP data header to one. The
M位：必须包括M位。会话中的第一个数据包和空闲期后的第一个数据包应通过将RTP数据头中的标记位设置为1来区分。这个
marker bit in all other packets MUST be set to zero. The reception of the marker bit MAY be used for refined methods for detection of loss.
所有其他数据包中的标记位必须设置为零。标记比特的接收可用于检测丢失的改进方法。
Consideration must be devoted to keeping loss of text caused by packet loss within acceptable limits. (See ITU-T F.703 [16].)
必须考虑将数据包丢失导致的文本丢失保持在可接受的范围内。（见ITU-T F.703[16]。）
The default method that MUST be used when no other method is explicitly selected is redundancy in accordance with RFC 2198 [3]. When this method is used, the original text and two redundant generations SHOULD be transmitted if the application or end-to-end conditions do not call for other levels of redundancy to be used.
根据RFC 2198[3]，当没有明确选择其他方法时，必须使用的默认方法是冗余。使用此方法时，如果应用程序或端到端条件不要求使用其他冗余级别，则应传输原始文本和两个冗余生成。
Other protection methods MAY be used. Forward Error Correction mechanisms as per RFC 2733 [8] or any other mechanism with the purpose of increasing the reliability of text transmission MAY be used as an alternative or complement to redundancy. Text data MAY be sent without additional protection if end-to-end network conditions allow the text quality requirements specified in ITU-T F.703 [16] to be met in all anticipated load conditions.
可使用其他保护方法。根据RFC 2733[8]的前向纠错机制或任何其他旨在提高文本传输可靠性的机制可作为冗余的替代或补充。如果端到端网络条件允许在所有预期负载条件下满足ITU-T F.703[16]中规定的文本质量要求，则可以发送文本数据而无需额外保护。
When using the format with redundant data, the transmitter may select a number of T140block generations to retransmit in each packet. A higher number introduces better protection against loss of text but marginally increases the data rate.
当使用具有冗余数据的格式时，发送器可以选择在每个分组中重新传输的T140块生成数。数字越大，可以更好地防止文本丢失，但会略微提高数据速率。
The RTP header is followed by one or more redundant data block headers, one for each redundant data block to be included. Each of these headers provides the timestamp offset and length of the corresponding data block plus a payload type number indicating the payload format audio/t140c.
RTP报头后面是一个或多个冗余数据块报头，每个冗余数据块包含一个。这些报头中的每一个都提供了相应数据块的时间戳偏移量和长度，以及指示有效负载格式audio/t140c的有效负载类型号。
After the redundant data block headers follows the redundant data fields carrying T140blocks from previous packets, and finally the new (primary) T140block for this packet.
冗余数据块报头之后是冗余数据字段，其中包含来自先前数据包的T140块，最后是该数据包的新（主）T140块。
Redundant data that would need a timestamp offset higher than 16383 due to its age at transmission MUST NOT be included in transmitted packets.
传输的数据包中不得包含因传输时的使用年限而需要大于16383的时间戳偏移量的冗余数据。
Since sequence numbers are not provided in the redundant header and since the sequence number space is shared by all audio payload types within an RTP session, a sequence number in the form of a T140block counter is added to the T140block for transmission. This allows the redundant T140block data corresponding to missing primary data to be retrieved and used properly into the stream of received T140block data when using the audio/t140c payload format.
由于冗余报头中不提供序列号，并且由于序列号空间由RTP会话中的所有音频有效负载类型共享，因此将T140块计数器形式的序列号添加到T140块以进行传输。这允许在使用音频/t140c有效负载格式时检索与丢失的主数据相对应的冗余T140block数据，并将其正确地用于接收到的T140block数据流中。
All non-empty redundant data blocks MUST contain the same data as a T140block previously transmitted as primary data, and be identified with a T140block counter equating to the original T140block counter for that T140block.
所有非空冗余数据块必须包含与先前作为主数据传输的T140块相同的数据，并用T140块计数器标识，该计数器等于该T140块的原始T140块计数器。
The T140block counters preceding the text in the T140block enables the ordering by the receiver. If there is a gap in the T140block counter value of received audio/t140c packets, and if there are redundant T140blocks with T140block counters matching those that are missing, the redundant T140blocks may be substituted for the missing T140blocks.
T140块中文本前面的T140块计数器启用接收方的排序。如果接收到的音频/t140c分组的T140block计数器值中存在间隙，并且如果存在冗余t140块，其中t140块计数器与缺失的计数器匹配，则冗余t140块可以替换缺失的t140块。
The value of the length field in the redundant header indicates the length of the concatenated T140block counter and the T140block.
冗余报头中长度字段的值表示连接的T140块计数器和T140块的长度。
This section contains RECOMMENDED procedures for usage of the payload format. Based on the information in the received packets, the receiver can:
本节包含使用有效负载格式的建议步骤。基于所接收的分组中的信息，接收机可以：
- reorder text received out of order. - mark where text is missing because of packet loss. - compensate for lost packets by using redundant data.
- 按顺序重新排列收到的文本。-标记由于数据包丢失而丢失文本的位置。-通过使用冗余数据补偿丢失的数据包。
Packets are transmitted when there is valid T.140 data to transmit.
当有有效的T.140数据要传输时，数据包被传输。
T.140 specifies that T.140 data MAY be buffered for transmission with a maximum buffering time of 500 ms. A buffering time of 300 ms is RECOMMENDED when the application or end-to-end network conditions are not known to require another value.
T.140规定，T.140数据可缓冲传输，最大缓冲时间为500 ms。当应用或端到端网络条件未知时，建议缓冲时间为300 ms。
If no new data is available for a longer period than the buffering time, the transmission process is in an idle period.
如果在比缓冲时间更长的时间内没有新数据可用，则传输过程处于空闲期。
When new text is available for transmission after an idle period, it is RECOMMENDED to send it as soon as possible. After this transmission, it is RECOMMENDED to buffer T.140 data in buffering time intervals until next idle period. This is done in order to keep the maximum bit-rate usage for text at a reasonable level. The buffering time MUST be selected so that text users will perceive a real-time text flow.
当新文本在空闲时间后可供传输时，建议尽快发送。在该传输之后，建议以缓冲时间间隔缓冲T.140数据，直到下一个空闲周期。这样做是为了将文本的最大比特率使用保持在合理水平。必须选择缓冲时间，以便文本用户能够感知实时文本流。
When valid T.140 data has been sent and no new T.140 data is available for transmission after the selected buffering time, an empty T140block SHOULD be transmitted. This situation is regarded to be the beginning of an idle period. The procedure is recommended in order to more rapidly detect potentially missing text before an idle period or when the audio stream switches from the transmission of audio/t140c to some other form of audio.
当已发送有效的T.140数据且在选定的缓冲时间后没有新的T.140数据可用于传输时，应传输空的T140块。这种情况被认为是闲置期的开始。建议使用该程序，以便在空闲时间之前或当音频流从音频/t140c传输切换到其他形式的音频时，更快速地检测可能丢失的文本。
An empty T140block contains no data, neither T.140 data nor a T140block counter.
空T140块不包含数据，既不包含T.140数据，也不包含T140块计数器。
When redundancy is used, transmission continues with a packet at every transmission timer expiration and insertion of an empty T.140block as primary, until the last non-empty T140block has been transmitted as primary and as redundant data with all intended generations of redundancy. The last packet before an idle period will contain only one non-empty T140block as redundant data, and the empty primary T140block.
当使用冗余时，在每次传输定时器到期时，传输继续进行，并插入一个空的T.140块作为主数据块，直到最后一个非空的T140块已作为主数据块和冗余数据传输，并具有所有预期的冗余代数。空闲时间段之前的最后一个数据包将只包含一个非空T140块作为冗余数据，以及空的主T140块。
When using the audio/t140c payload format, empty T140blocks sent as primary data SHOULD NOT be included as redundant T140blocks, as it would simply be a waste of bandwidth to send them and it would introduce a risk of false detection of loss.
当使用音频/t140c有效负载格式时，作为主数据发送的空T140block不应作为冗余T140block包括在内，因为发送它们只会浪费带宽，并且会引入错误检测丢失的风险。
After an idle period, the transmitter SHOULD set the M-bit to one in the first packet with new text.
空闲时间过后，发送器应在具有新文本的第一个数据包中将M位设置为1。
Receivers detect the loss of an audio/t140c packet by observing the value of the T140block counter in a subsequent audio/t140c packet.
接收机通过观察后续音频/t140c数据包中T140block计数器的值来检测音频/t140c数据包的丢失。
Missing data SHOULD be marked by insertion of a missing text marker in the received stream for each missing T140block, as specified in ITU-T T.140 Addendum 1 [1].
根据ITU-T T.140附录1[1]的规定，对于每个缺失T140块，应通过在接收流中插入缺失文本标记来标记缺失数据。
Procedures based on detection of the packet with the M-bit set to one MAY be used to reduce the risk for introducing false markers of loss.
基于M位设置为1的分组检测的过程可用于降低引入错误丢失标记的风险。
False detection will also be avoided when using audio/t140c by observing the value of the T140block counter value.
在使用audio/t140c时，通过观察T140block计数器值，也可以避免错误检测。
If two successive packets have the same number of redundant generations, it SHOULD be treated as the general redundancy level for the session. Change of the general redundancy level SHOULD only be done after an idle period.
如果两个连续数据包具有相同数量的冗余生成，则应将其视为会话的一般冗余级别。一般冗余级别的更改只能在空闲时间后进行。
For protection against packets arriving out of order, the following procedure MAY be implemented in the receiver. If analysis of a received packet reveals a gap in the sequence and no redundant data is available to fill that gap, the received packet SHOULD be kept in a buffer to allow time for the missing packet(s) to arrive. It is RECOMMENDED that the waiting time be limited to 1 second.
为了防止分组无序到达，可以在接收机中实现以下过程。如果对接收到的数据包的分析显示序列中存在间隙，并且没有可用的冗余数据来填补该间隙，则应将接收到的数据包保存在缓冲器中，以留出时间让丢失的数据包到达。建议将等待时间限制为1秒。
If a packet with a T140block belonging to the gap arrives before the waiting time expires, this T140block is inserted into the gap and then consecutive T140blocks from the leading edge of the gap may be consumed. Any T140block that does not arrive before the time limit expires should be treated as lost and a missing text marker inserted (see section 5.3).
如果具有属于该间隙的t140块的分组在等待时间到期之前到达，则将该t140块插入该间隙中，然后可以消耗该间隙前缘的连续t140块。任何未在时限到期前到达的T140块应视为丢失，并插入缺失的文本标记（见第5.3节）。
In some cases, it is necessary to limit the rate at which characters are transmitted. For example, when a PSTN gateway is interworking between an IP device and a PSTN textphone, it may be necessary to limit the character rate from the IP device in order to avoid throwing away characters in case of buffer overflow at the PSTN gateway.
在某些情况下，有必要限制字符的传输速率。例如，当PSTN网关在IP设备和PSTN文本电话之间互通时，可能需要限制来自IP设备的字符速率，以避免在PSTN网关发生缓冲区溢出时丢弃字符。
To control the character transmission rate, the MIME parameter "cps" in the "fmtp" attribute [7] is defined (see section 10). It is used in Session Description Protocol (SDP) with the following syntax:
为了控制字符传输速率，定义了“fmtp”属性[7]中的MIME参数“cps”（参见第10节）。它在会话描述协议（SDP）中使用，语法如下：
a=fmtp:<format> cps=<integer>
a=fmtp:<format> cps=<integer>
The <format> field is populated with the payload type that is used for text. The <integer> field contains an integer representing the maximum number of characters that may be received per second. The value shall be used as a mean value over any 10-second interval. The default value is 30.
<format>字段填充用于文本的有效负载类型。<integer>字段包含一个整数，表示每秒可接收的最大字符数。该值应作为任何10秒间隔的平均值。默认值为30。
In receipt of this parameter, devices MUST adhere to the request by transmitting characters at a rate at or below the specified <integer> value. Examples of use in SDP are found in section 7.2.
在接收到该参数时，设备必须通过以等于或低于指定的<integer>值的速率传输字符来遵守请求。SDP中的使用示例见第7.2节。
Below is an example of an audio/t140c RTP packet without redundancy.
下面是没有冗余的音频/t140c RTP包的示例。
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0 |M| T140c PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp (8000Hz) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| T140block counter | T.140 encoded data |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0 |M| T140c PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp (8000Hz) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| T140block counter | T.140 encoded data |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Below is an example of an RTP packet with one redundant T140block using audio/t140c payload format. The primary data block is empty, which is the case when transmitting a packet for the sole purpose of forcing the redundant data to be transmitted in the absence of any new data. Note that since this is the audio/t140c payload format, the redundant block of T.140 data is immediately preceded with a T140block counter.
下面是一个RTP数据包的示例，其中包含一个使用音频/t140c有效负载格式的冗余T140block。主数据块是空的，这是在没有任何新数据的情况下仅为了强制发送冗余数据而发送分组时的情况。注意，由于这是音频/t140c有效负载格式，T.140数据的冗余块前面紧跟着一个T140block计数器。
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0 |M| "RED" PT | sequence number of primary |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp of primary encoding "P" |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| T140c PT | timestamp offset of "R" | "R" block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| T140c PT | "R" T140block counter | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
| "R" T.140 encoded redundant data |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0 |M| "RED" PT | sequence number of primary |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp of primary encoding "P" |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| T140c PT | timestamp offset of "R" | "R" block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| T140c PT | "R" T140block counter | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
| "R" T.140 encoded redundant data |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
As a follow-on to the previous example, the example below shows the next RTP packet in the sequence that does contain a new real T140block when using the audio/t140c payload format. This example has 2 levels of redundancy and one primary data block. Since the previous primary block was empty, no redundant data is included for that block. This is because when using the audio/t140c payload format, any previously transmitted "empty" T140blocks are NOT included as redundant data in subsequent packets.
作为上一示例的后续，下面的示例显示了序列中的下一个RTP数据包，该数据包在使用音频/t140c有效负载格式时确实包含新的实T140block。此示例具有两个冗余级别和一个主数据块。由于上一个主块为空，因此该块不包含冗余数据。这是因为当使用音频/t140c有效负载格式时，任何先前传输的“空”t140c块都不会作为冗余数据包含在后续数据包中。
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0 |M| "RED" PT | sequence number of primary |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp of primary encoding "P" |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| T140c PT | timestamp offset of "R1" | "R1" block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| T140c PT | "R1" T140block counter | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
| "R1" T.140 encoded redundant data |
+ +---------------+
| | "P" T140block |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| counter | "P" T.140 encoded primary data |
+-+-+-+-+-+-+-+-+ +
| |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC=0 |M| "RED" PT | sequence number of primary |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp of primary encoding "P" |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|1| T140c PT | timestamp offset of "R1" | "R1" block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|0| T140c PT | "R1" T140block counter | |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +
| "R1" T.140 encoded redundant data |
+ +---------------+
| | "P" T140block |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| counter | "P" T.140 encoded primary data |
+-+-+-+-+-+-+-+-+ +
| |
+ +---------------+
| |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Below is an example of SDP describing RTP text interleaved with G.711 audio packets within the same RTP session from port 7200 and at a maximum text rate of 6 characters per second:
下面是SDP的示例，描述了在同一RTP会话中，从端口7200以每秒6个字符的最大文本速率与G.711音频包交织的RTP文本：
m=audio 7200 RTP/AVP 0 98
a=rtpmap:98 t140c/8000
a=fmtp:98 cps=6
m=audio 7200 RTP/AVP 0 98
a=rtpmap:98 t140c/8000
a=fmtp:98 cps=6
Below is an example using RFC 2198 to provide the recommended two levels of redundancy to the text packets in an RTP session with interleaving text and G.711 at a text rate no faster than 20 characters per second:
下面是使用RFC 2198为RTP会话中的文本包提供建议的两级冗余的示例，其中文本和G.711以不超过每秒20个字符的文本速率交错：
m=audio 7200 RTP/AVP 0 98 100
a=rtpmap:98 t140c/8000
a=fmtp:98 cps=20
a=rtpmap:100 red/8000
a=fmtp:100 98/98/98
m=audio 7200 RTP/AVP 0 98 100
a=rtpmap:98 t140c/8000
a=fmtp:98 cps=20
a=rtpmap:100 red/8000
a=fmtp:100 98/98/98
Note: While these examples utilize the RTP/AVP profile, it is not intended to limit the scope of this memo to use with only that profile. Rather, any appropriate profile may be used in conjunction with this memo.
注：虽然这些示例使用RTP/AVP配置文件，但并不打算将本备忘录的范围限制为仅与该配置文件一起使用。相反，任何适当的配置文件都可以与本备忘录一起使用。
All of the security considerations from section 14 of RFC 3550 [2] apply.
RFC 3550[2]第14节中的所有安全注意事项均适用。
Since the intention of the described payload format is to carry text in a text conversation, security measures in the form of encryption are of importance. The amount of data in a text conversation session is low, and therefore any encryption method MAY be selected and applied to T.140 session contents or to the whole RTP packets. Secure Realtime Transport Protocol (SRTP) [13] provides a suitable method for ensuring confidentiality.
由于所述有效载荷格式的目的是在文本对话中携带文本，因此以加密形式的安全措施非常重要。文本会话会话中的数据量较低，因此可以选择任何加密方法并将其应用于T.140会话内容或整个RTP分组。安全实时传输协议（SRTP）[13]为确保机密性提供了合适的方法。
It may be desirable to protect the text contents of an RTP stream against manipulation. SRTP [13] provides methods for providing integrity that MAY be applied.
可能希望保护RTP流的文本内容不受操纵。SRTP[13]提供了提供完整性的方法，这些方法可以应用。
Measures to make sure that the source of text is the intended one can be accomplished by a combination of methods.
确保文本来源为预期来源的措施可以通过多种方法的组合来实现。
Text streams are usually used in a multimedia control environment. Security measures for authentication are available and SHOULD be applied in the registration and session establishment procedures, so that the identity of the sender of the text stream is reliably associated with the person or device setting up the session. Once established, SRTP [13] mechanisms MAY be applied to ascertain that the source is maintained the same during the session.
文本流通常用于多媒体控制环境。身份验证的安全措施可用，并应在注册和会话建立过程中应用，以便文本流发送者的身份与设置会话的人员或设备可靠关联。一旦建立，SRTP[13]机制可用于确定源在会话期间保持不变。
The congestion considerations from section 10 of RFC 3550 [2], section 6 of RFC 2198 [3], and any used profile (e.g., the part about congestion in section 2 of RFC 3551 [10]) apply with the following application-specific considerations.
RFC 3550[2]第10节、RFC 2198[3]第6节以及任何使用的配置文件（例如，RFC 3551[10]第2节中关于拥塞的部分）中的拥塞注意事项适用于以下特定于应用的注意事项。
Automated systems MUST NOT use this format to send large amounts of text at a rate significantly above that which a human user could enter.
自动化系统不得使用此格式发送大量文本，发送速度远远高于人类用户可以输入的速度。
Even if the network load from users of text conversation is usually very low, for best-effort networks an application MUST monitor the packet loss rate and take appropriate actions to reduce its sending rate if this application sends at higher rate than what TCP would achieve over the same path. The reason is that this application, due to its recommended usage of two or more redundancy levels, is very robust against packet loss. At the same time, due to the low bit-rate of text conversations, if one considers the discussion in RFC 3714 [12], this application will experience very high packet loss rates before it needs to perform any reduction in the sending rate.
即使来自文本会话用户的网络负载通常很低，对于尽力而为的网络，如果此应用程序的发送速率高于TCP在同一路径上的发送速率，则应用程序必须监控数据包丢失率，并采取适当措施降低其发送速率。原因是，由于推荐使用两个或更多冗余级别，该应用程序对数据包丢失非常健壮。同时，由于文本对话的低比特率，如果考虑RFC 3714[12]中的讨论，该应用程序在需要执行任何发送速率降低之前将经历非常高的丢包率。
If the application needs to reduce its sending rate, it SHOULD NOT reduce the number of redundancy levels below the default amount specified in section 4. Instead, the following actions are RECOMMENDED in order of priority:
如果应用程序需要降低其发送速率，则不应将冗余级别的数量降低到第4节中指定的默认数量以下。相反，建议按优先顺序采取以下措施：
- Increase the shortest time between transmissions described in section 5.1 from the recommended 300 ms to 500 ms that is the highest value allowable according to T.140.
- 将第5.1节中描述的两次传输之间的最短时间从建议的300 ms增加到500 ms，这是根据T.140允许的最高值。
- Limit the maximum rate of characters transmitted.
- 限制传输字符的最大速率。
- Increase the shortest time between transmissions to a higher value, not higher than 5 seconds. This will cause unpleasant delays in transmission, beyond what is allowed according to T.140, but text will still be conveyed in the session with some usability.
- 将两次传输之间的最短时间增加到更高的值，不超过5秒。这将导致令人不快的传输延迟，超出T.140允许的范围，但文本仍将在会话中传输，具有一定的可用性。
- Exclude participants from the session.
- 将参与者排除在会话之外。
Please note that if the reduction in bit-rate achieved through the above measures is not sufficient, the only remaining action is to terminate the session.
请注意，如果通过上述措施实现的比特率降低还不够，唯一剩下的操作是终止会话。
As guidance, some load figures are provided here as examples based on use of IPv4, including the load from IP, UDP, and RTP headers without compression.
作为指导，这里提供了一些基于IPv4使用的负载数据作为示例，包括来自IP、UDP和RTP报头的负载（无压缩）。
- Experience tells that a common mean character transmission rate during a complete PSTN text telephony session in reality is around 2 characters per second.
- 经验告诉我们，在一个完整的PSTN文本电话会话中，一个常见的平均字符传输速率实际上大约为每秒2个字符。
- A maximum performance of 20 characters per second is enough even for voice-to-text applications.
- 即使对于语音到文本应用程序，每秒最多20个字符的性能也足够了。
- With the (unusually high) load of 20 characters per second, in a language that make use of three-octet UTF-8 characters, two redundant levels, and 300 ms between transmissions, the maximum load of this application is 3500 bits/s.
- 在使用三个八位UTF-8字符、两个冗余级别和300毫秒传输间隔的语言中，每秒20个字符的负载（异常高），此应用程序的最大负载为3500位/秒。
- When the restrictions mentioned above are applied, limiting transmission to 10 characters per second, using 5 s between transmissions, the maximum load of this application in a language that uses one octet per UTF-8 character is 300 bits/s.
- 当应用上述限制时，将传输限制为每秒10个字符，在传输之间使用5秒，此应用程序在每UTF-8字符使用一个八位字节的语言中的最大负载为300比特/秒。
Note also, that this payload can be used in a congested situation as a last resort to maintain some contact when audio and video media need to be stopped. The availability of one low bit-rate stream for text in such adverse situations may be crucial for maintaining some communication in a critical situation.
还要注意的是，在拥挤的情况下，当音频和视频媒体需要停止时，此有效负载可以作为最后手段来保持联系。在这种不利情况下，文本的一个低比特率流的可用性对于在危急情况下维持某些通信可能是至关重要的。
This document defines one RTP payload format named "t140" and an associated MIME type "audio/t140c". They have been registered by the IANA.
本文档定义了一种名为“t140”的RTP有效负载格式和一种相关的MIME类型“audio/t140c”。它们已由IANA注册。
MIME media type name: audio
MIME媒体类型名称：音频
MIME subtype name: t140c
MIME子类型名称：t140c
Required parameters: rate: The RTP timestamp clock rate, which is equal to the sampling rate. This parameter SHOULD have the same value as for any audio codec packets interleaved in the same RTP stream.
所需参数：速率：RTP时间戳时钟速率，等于采样速率。此参数的值应与在同一RTP流中交织的任何音频编解码器数据包的值相同。
Optional parameters: cps: The maximum number of characters that may be received per second. The default value is 30.
可选参数：cps：每秒可接收的最大字符数。默认值为30。
Encoding considerations: T.140 text can be transmitted with RTP as specified in RFC 4351.
编码注意事项：T.140文本可以按照RFC 4351中的规定使用RTP传输。
Security considerations: See section 8 of RFC 4351.
安全注意事项：见RFC 4351第8节。
Interoperability considerations: None
互操作性注意事项：无
Published specification: ITU-T T.140 Recommendation. RFC 4351.
已发布规范：ITU-T.140建议。RFC 4351。
Applications which use this media type: Text communication systems and text conferencing tools that transmit text associated with audio and within the same RTP session as the audio, such as PSTN gateways that transmit audio and text signals between two PSTN textphone users over an IP network.
使用此媒体类型的应用程序：文本通信系统和文本会议工具，它们在与音频相关的RTP会话中传输与音频相关的文本，例如通过IP网络在两个PSTN textphone用户之间传输音频和文本信号的PSTN网关。
Additional information: This type is only defined for transfer via RTP.
附加信息：此类型仅为通过RTP传输而定义。
Magic number(s): None
File extension(s): None
Macintosh File Type Code(s): None
Magic number(s): None
File extension(s): None
Macintosh File Type Code(s): None
Person & email address to contact for further information: Paul E. Jones E-mail: paulej@packetizer.com
联系人和电子邮件地址以获取更多信息：Paul E.Jones电子邮件：paulej@packetizer.com
Intended usage: COMMON
预期用途：普通
Author / Change controller: Paul E. Jones | IETF avt WG delegated from the IESG paulej@packetizer.com |
作者/变更控制员：IESG委派的Paul E.Jones | IETF avt工作组paulej@packetizer.com |
The information carried in the MIME media type specification has a specific mapping to fields in the Session Description Protocol (SDP) [7], which is commonly used to describe RTP sessions. When SDP is used to specify sessions employing the audio/t140c format, the mapping is as follows:
MIME媒体类型规范中包含的信息具有到会话描述协议（SDP）[7]中字段的特定映射，该协议通常用于描述RTP会话。当使用SDP指定采用音频/t140c格式的会话时，映射如下：
- The MIME type ("audio") goes in SDP "m=" as the media name.
- MIME类型（“音频”）以SDP“m=”作为媒体名称。
- The MIME subtype (payload format name) goes in SDP "a=rtpmap" as the encoding name. For audio/t140c, the clock rate MAY be set to any value, and SHOULD be set to the same value as for any audio packets in the same RTP stream.
- MIME子类型（有效负载格式名称）以SDP“a=rtpmap”作为编码名称。对于音频/t140c，时钟速率可以设置为任何值，并且应该设置为与相同RTP流中的任何音频分组相同的值。
- The parameter "cps" goes in SDP "a=fmtp" attribute.
- 参数“cps”位于SDP“a=fmtp”属性中。
- When the payload type is used with redundancy according to RFC 2198, the level of redundancy is shown by the number of elements in the slash-separated payload type list in the "fmtp" parameter of the redundancy declaration as defined in RFC 2198 [3].
- 根据RFC 2198，当有效负载类型与冗余一起使用时，冗余级别通过RFC 2198[3]中定义的冗余声明的“fmtp”参数中斜杠分隔的有效负载类型列表中的元素数量来显示。
In order to achieve interoperability within the framework of the offer/answer model [9], the following consideration should be made:
为了在提供/应答模型[9]的框架内实现互操作性，应考虑以下因素：
- The "cps" parameter is declarative. Both sides may provide a value, which is independent of the other side.
- The "cps" parameter is declarative. Both sides may provide a value, which is independent of the other side.translate error, please retry
The authors want to thank Stephen Casner, Magnus Westerlund, and Colin Perkins for valuable support with reviews and advice on creation of this document; Mickey Nasiri at Ericsson Mobile Communication for providing the development environment; Michele Mizarro for verification of the usability of the payload format for its intended purpose; and Andreas Piirimets for editing support.
作者要感谢Stephen Casner、Magnus Westerlund和Colin Perkins对本文件创建的评论和建议的宝贵支持；爱立信移动通信公司的Mickey Nasiri提供开发环境；Michele Mizarro，用于验证有效载荷格式在其预期用途中的可用性；Andreas Piirimets提供编辑支持。
[1] ITU-T Recommendation T.140 (1998) - Text conversation protocol for multimedia application, with amendment 1, (2000).
[1] ITU-T建议T.140（1998）-多媒体应用的文本对话协议，修订件1，（2000年）。
[2] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003.
[2] 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。
[3] 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, September 1997.
[3] 帕金斯，C.，库维拉斯，I.，霍德森，O.，哈德曼，V.，汉德利，M.，博洛特，J.，维加·加西亚，A.，和S.福斯·帕里斯，“冗余音频数据的RTP有效载荷”，RFC 21981997年9月。
[4] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997.
[4] Bradner，S.，“RFC中用于表示需求水平的关键词”，BCP 14，RFC 2119，1997年3月。
[5] ISO/IEC 10646-1: (1993), Universal Multiple Octet Coded Character Set.
[5] ISO/IEC 10646-1:（1993），通用多八位编码字符集。
[6] Yergeau, F., "UTF-8, a transformation format of ISO 10646", STD 63, RFC 3629, November 2003.
[6] Yergeau，F.，“UTF-8，ISO 10646的转换格式”，STD 63，RFC 3629，2003年11月。
[7] Handley, M. and V. Jacobson, "SDP: Session Description Protocol", RFC 2327, April 1998.
[7] Handley，M.和V.Jacobson，“SDP：会话描述协议”，RFC 2327，1998年4月。
[8] Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format for Generic Forward Error Correction", RFC 2733, December 1999.
[8] Rosenberg，J.和H.Schulzrinne，“通用前向纠错的RTP有效载荷格式”，RFC 2733，1999年12月。
[9] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with Session Description Protocol (SDP)", RFC 3264, June 2002.
[9] 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。
[10] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
[10] 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。
[11] Postel, J., "Internet Protocol", STD 5, RFC 791, September 1981.
[11] Postel，J.，“互联网协议”，STD 5，RFC 7911981年9月。
[12] Floyd, S. and J. Kempf, "IAB Concerns Regarding Congestion Control for Voice Traffic in the Internet", RFC 3714, March 2004.
[12] Floyd，S.和J.Kempf，“IAB对互联网语音流量拥塞控制的关注”，RFC 3714，2004年3月。
[13] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004.
[13] 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。
[14] Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals", RFC 2833, May 2000.
[14] Schulzrinne，H.和S.Petrack，“DTMF数字、电话音和电话信号的RTP有效载荷”，RFC 28332000年5月。
[15] Hellstrom, G. and P. Jones, "RTP Payload for Text Conversation", RFC 4103, June 2005.
[15] Hellstrom，G.和P.Jones，“文本对话的RTP有效载荷”，RFC 4103，2005年6月。
[16] ITU-T Recommendation F.703, Multimedia Conversational Services, Nov 2000.
[16] ITU-T建议F.703，多媒体会话服务，2000年11月。
[17] Casner, S. and P. Hoschka, "MIME Type Registration of RTP Payload Formats", RFC 3555, July 2003.
[17] Casner，S.和P.Hoschka，“RTP有效载荷格式的MIME类型注册”，RFC 3555，2003年7月。
Authors' Addresses
作者地址
Gunnar Hellstrom Omnitor AB Renathvagen 2 SE-121 37 Johanneshov Sweden
Gunnar Hellstrom Omnitor AB Renathvagen 2 SE-121 37瑞典约翰尼绍夫
Phone: +46 708 204 288 / +46 8 556 002 03
Fax: +46 8 556 002 06
EMail: gunnar.hellstrom@omnitor.se
Phone: +46 708 204 288 / +46 8 556 002 03
Fax: +46 8 556 002 06
EMail: gunnar.hellstrom@omnitor.se
Paul E. Jones Cisco Systems, Inc. 7025 Kit Creek Rd. Research Triangle Park, NC 27709 USA
Paul E.Jones Cisco Systems，Inc.美国北卡罗来纳州三角研究公园Kit Creek路7025号，邮编：27709
Phone: +1 919 392 6948
EMail: paulej@packetizer.com
Phone: +1 919 392 6948
EMail: paulej@packetizer.com
Full Copyright Statement
完整版权声明
Copyright (C) The Internet Society (2006).
版权所有（C）互联网协会（2006年）。
This document is subject to the rights, licenses and restrictions contained in BCP 78, and except as set forth therein, the authors retain all their rights.
本文件受BCP 78中包含的权利、许可和限制的约束，除其中规定外，作者保留其所有权利。
This document and the information contained herein are provided on an "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
本文件及其包含的信息是按“原样”提供的，贡献者、他/她所代表或赞助的组织（如有）、互联网协会和互联网工程任务组不承担任何明示或暗示的担保，包括但不限于任何保证，即使用本文中的信息不会侵犯任何权利，或对适销性或特定用途适用性的任何默示保证。
Intellectual Property
知识产权
The IETF takes no position regarding the validity or scope of any Intellectual Property Rights or other rights that might be claimed to pertain to the implementation or use of the technology described in this document or the extent to which any license under such rights might or might not be available; nor does it represent that it has made any independent effort to identify any such rights. Information on the procedures with respect to rights in RFC documents can be found in BCP 78 and BCP 79.
IETF对可能声称与本文件所述技术的实施或使用有关的任何知识产权或其他权利的有效性或范围，或此类权利下的任何许可可能或可能不可用的程度，不采取任何立场；它也不表示它已作出任何独立努力来确定任何此类权利。有关RFC文件中权利的程序信息，请参见BCP 78和BCP 79。
Copies of IPR disclosures made to the IETF Secretariat and any assurances of licenses to be made available, or the result of an attempt made to obtain a general license or permission for the use of such proprietary rights by implementers or users of this specification can be obtained from the IETF on-line IPR repository at http://www.ietf.org/ipr.
向IETF秘书处披露的知识产权副本和任何许可证保证，或本规范实施者或用户试图获得使用此类专有权利的一般许可证或许可的结果，可从IETF在线知识产权存储库获取，网址为http://www.ietf.org/ipr.
The IETF invites any interested party to bring to its attention any copyrights, patents or patent applications, or other proprietary rights that may cover technology that may be required to implement this standard. Please address the information to the IETF at ietf-ipr@ietf.org.
IETF邀请任何相关方提请其注意任何版权、专利或专利申请，或其他可能涵盖实施本标准所需技术的专有权利。请将信息发送至IETF的IETF-ipr@ietf.org.
Acknowledgement
确认
Funding for the RFC Editor function is provided by the IETF Administrative Support Activity (IASA).
RFC编辑器功能的资金由IETF行政支持活动（IASA）提供。