Internet Engineering Task Force (IETF) A. Pendleton
Request for Comments: 6035 A. Clark
Category: Standards Track Telchemy Incorporated
ISSN: 2070-1721 A. Johnston
Avaya
H. Sinnreich
Unaffiliated
November 2010
Internet Engineering Task Force (IETF) A. Pendleton
Request for Comments: 6035 A. Clark
Category: Standards Track Telchemy Incorporated
ISSN: 2070-1721 A. Johnston
Avaya
H. Sinnreich
Unaffiliated
November 2010
Session Initiation Protocol Event Package for Voice Quality Reporting
用于语音质量报告的会话启动协议事件包
Abstract
摘要
This document defines a Session Initiation Protocol (SIP) event package that enables the collection and reporting of metrics that measure the quality for Voice over Internet Protocol (VoIP) sessions. Voice call quality information derived from RTP Control Protocol Extended Reports (RTCP-XR) and call information from SIP is conveyed from a User Agent (UA) in a session, known as a reporter, to a third party, known as a collector. A registration for the application/ vq-rtcpxr media type is also included.
本文档定义了一个会话启动协议（SIP）事件包，该事件包支持收集和报告度量Internet语音协议（VoIP）会话质量的指标。来自RTP控制协议扩展报告（RTCP-XR）的语音呼叫质量信息和来自SIP的呼叫信息在会话（称为报告者）中从用户代理（UA）传送到第三方（称为收集器）。还包括应用程序/vq rtcpxr介质类型的注册。
Status of This Memo
关于下段备忘
This is an Internet Standards Track document.
这是一份互联网标准跟踪文件。
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). Further information on Internet Standards is available in Section 2 of RFC 5741.
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。有关互联网标准的更多信息，请参见RFC 5741第2节。
Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at http://www.rfc-editor.org/info/rfc6035.
有关本文件当前状态、任何勘误表以及如何提供反馈的信息，请访问http://www.rfc-editor.org/info/rfc6035.
Copyright Notice
版权公告
Copyright (c) 2010 IETF Trust and the persons identified as the document authors. All rights reserved.
版权所有（c）2010 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 (http://trustee.ietf.org/license-info) 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(http://trustee.ietf.org/license-info)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。
This document may contain material from IETF Documents or IETF Contributions published or made publicly available before November 10, 2008. The person(s) controlling the copyright in some of this material may not have granted the IETF Trust the right to allow modifications of such material outside the IETF Standards Process. Without obtaining an adequate license from the person(s) controlling the copyright in such materials, this document may not be modified outside the IETF Standards Process, and derivative works of it may not be created outside the IETF Standards Process, except to format it for publication as an RFC or to translate it into languages other than English.
本文件可能包含2008年11月10日之前发布或公开的IETF文件或IETF贡献中的材料。控制某些材料版权的人员可能未授予IETF信托允许在IETF标准流程之外修改此类材料的权利。在未从控制此类材料版权的人员处获得充分许可的情况下，不得在IETF标准流程之外修改本文件，也不得在IETF标准流程之外创建其衍生作品，除了将其格式化以RFC形式发布或将其翻译成英语以外的其他语言。
Table of Contents
目录
1. Introduction ....................................................4
1.1. Applicability Statement ....................................4
1.2. Use of the Mechanism .......................................4
2. Terminology .....................................................6
3. SIP Events for VoIP Quality Reporting ...........................6
3.1. SUBSCRIBE NOTIFY Method ....................................6
3.2. PUBLISH Method .............................................7
3.3. Multi-Party and Multi-Segment Calls ........................7
3.4. Overload Avoidance .........................................7
4. Event Package Formal Definition .................................8
4.1. Event Package Name .........................................8
4.2. Event Package Parameters ...................................8
4.3. SUBSCRIBE Bodies ...........................................8
4.4. Subscribe Duration .........................................8
4.5. NOTIFY Bodies ..............................................8
4.6. Voice Quality Event and Semantics .........................10
4.6.1. ABNF Syntax Definition .............................10
4.6.2. Parameter Definitions and Mappings .................21
4.7. Message Flow and Syntax Examples ..........................29
4.7.1. End of Session Report Using NOTIFY .................29
4.7.2. Midsession Threshold Violation Using NOTIFY ........32
4.7.3. End of Session Report Using PUBLISH ................35
4.7.4. Alert Report Using PUBLISH .........................37
4.8. Configuration Dataset for vq-rtcpxr Events ................39
5. IANA Considerations ............................................39
5.1. SIP Event Package Registration ............................39
5.2. application/vq-rtcpxr Media Type Registration .............39
6. Security Considerations ........................................40
7. Contributors ...................................................40
8. References .....................................................40
8.1. Normative References ......................................40
8.2. Informative References ....................................41
1. Introduction ....................................................4
1.1. Applicability Statement ....................................4
1.2. Use of the Mechanism .......................................4
2. Terminology .....................................................6
3. SIP Events for VoIP Quality Reporting ...........................6
3.1. SUBSCRIBE NOTIFY Method ....................................6
3.2. PUBLISH Method .............................................7
3.3. Multi-Party and Multi-Segment Calls ........................7
3.4. Overload Avoidance .........................................7
4. Event Package Formal Definition .................................8
4.1. Event Package Name .........................................8
4.2. Event Package Parameters ...................................8
4.3. SUBSCRIBE Bodies ...........................................8
4.4. Subscribe Duration .........................................8
4.5. NOTIFY Bodies ..............................................8
4.6. Voice Quality Event and Semantics .........................10
4.6.1. ABNF Syntax Definition .............................10
4.6.2. Parameter Definitions and Mappings .................21
4.7. Message Flow and Syntax Examples ..........................29
4.7.1. End of Session Report Using NOTIFY .................29
4.7.2. Midsession Threshold Violation Using NOTIFY ........32
4.7.3. End of Session Report Using PUBLISH ................35
4.7.4. Alert Report Using PUBLISH .........................37
4.8. Configuration Dataset for vq-rtcpxr Events ................39
5. IANA Considerations ............................................39
5.1. SIP Event Package Registration ............................39
5.2. application/vq-rtcpxr Media Type Registration .............39
6. Security Considerations ........................................40
7. Contributors ...................................................40
8. References .....................................................40
8.1. Normative References ......................................40
8.2. Informative References ....................................41
Real-time communications over IP networks use SIP for signaling with RTP/RTCP for media transport and reporting, respectively. These protocols are very flexible and can support an extremely wide spectrum of usage scenarios. For this reason, extensions to these protocols must be specified in the context of a specific usage scenario. In this memo, extensions to SIP are proposed to support the reporting of RTP Control Protocol Extended Reports [4] metrics.
IP网络上的实时通信使用SIP信令，RTP/RTCP分别用于媒体传输和报告。这些协议非常灵活，可以支持非常广泛的使用场景。因此，必须在特定使用场景的上下文中指定这些协议的扩展。在本备忘录中，建议对SIP进行扩展，以支持RTP控制协议扩展报告[4]指标的报告。
RTP is utilized in many different architectures and topologies. RFC 5117 [13] lists and describes the following topologies: point-to-point, point-to-multipoint using multicast, point-to-multipoint using the translator from RFC 3550, point-to-multipoint using the mixer model from RFC 3550, point-to-multipoint using video-switching Multipoint Control Units (MCUs), point-to-multipoint using RTCP-terminating MCU, and non-symmetric mixer/translators. As the Abstract of this document points out, this specification is for reporting quality of Voice over Internet Protocol (VoIP) sessions. As such, only the first topology, point to point, is currently supported by this specification. This reflects both current VoIP deployments, which are predominantly point to point using unicast, and the state of research in the area of quality.
RTP被用于许多不同的体系结构和拓扑中。RFC 5117[13]列出并描述了以下拓扑：点对点、使用多播的点对多点、使用RFC 3550中的转换器的点对多点、使用RFC 3550中的混音器模型的点对多点、使用视频切换多点控制单元（MCU）的点对多点、使用RTCP端接MCU的点对多点，和非对称混频器/转换器。正如本文档摘要所指出的，本规范用于报告Internet协议语音（VoIP）会话的质量。因此，本规范目前仅支持第一种拓扑，即点对点拓扑。这反映了当前主要采用点对点单播的VoIP部署和质量领域的研究状况。
How to accurately report the quality of a multipart conference or a session involving multiple hops through translators and mixers is currently an area of research in the industry. However, this mechanism can easily be used for centrally mixed conference calls, in which each leg of the conferences is just a point-to-point call. This mechanism could be extended to cover additional RTP topologies in the future once these topics progress out of the realm of research and into actual Internet deployments.
如何通过翻译器和混音器准确地报告多部分会议或涉及多跳的会议的质量是当前业界的一个研究领域。然而，这种机制可以很容易地用于集中混合的会议呼叫，其中会议的每个分支只是一个点对点呼叫。一旦这些主题从研究领域发展到实际的互联网部署，这种机制可以扩展到未来的其他RTP拓扑。
RTCP reports are usually sent to other participating endpoints in a session. This can make the collection of performance information by an administrator or management system quite complex to implement. In the usage scenarios addressed in this memo, the data contained in RTCP XR VoIP metrics reports (RFC 3611 [4]) are forwarded to a central collection server systems using SIP.
RTCP报告通常发送到会话中的其他参与端点。这会使管理员或管理系统收集性能信息变得非常复杂，难以实现。在本备忘录所述的使用场景中，RTCP XR VoIP度量报告（RFC 3611[4]）中包含的数据使用SIP转发到中央采集服务器系统。
Applications residing in the server or elsewhere can aid in network management to alleviate bandwidth constraints and also to support customer service by identifying and acknowledging calls of poor quality. However, specifying such applications is beyond the scope of this paper.
驻留在服务器或其他地方的应用程序可以帮助网络管理，以减轻带宽限制，并通过识别和确认质量较差的呼叫来支持客户服务。然而，指定此类应用超出了本文的范围。
There is a large portfolio of quality parameters that can be associated with VoIP, but only a minimal necessary number of parameters are included on the RTCP-XR reports:
有大量可与VoIP关联的质量参数组合，但RTCP-XR报告中仅包含最低数量的必要参数：
1. The codec type, as resulting from the Session Description Protocol (SDP) offer-answer negotiation in SIP,
1. 由会话描述协议（SDP）产生的编解码器类型在SIP中提供应答协商，
2. The burst gap loss density and max gap duration, since voice cut-outs are the most annoying quality impairment in VoIP,
2. 突发间隔丢失密度和最大间隔持续时间，因为语音中断是VoIP中最烦人的质量损害，
3. Round-trip delay, because it is critical to conversational quality,
3. 往返延迟，因为它对会话质量至关重要，
4. Conversational quality as a catch-all for other voice quality impairments, such as randomly distributed packet loss, jitter, annoying silent suppression effects, etc.
4. 会话质量是其他语音质量损害的总括，如随机分布的数据包丢失、抖动、恼人的静音抑制效果等。
In specific usage scenarios where other parameters are required, designers can include other parameters beyond the scope of this paper.
在需要其他参数的特定使用场景中，设计者可以包括本文范围之外的其他参数。
RTCP reports are best effort only, and though they are very useful, they have a number of limitations as discussed in [3]. This must be considered when using RTCP reports in managed networks.
RTCP报告仅尽最大努力，尽管它们非常有用，但如[3]所述，它们有许多局限性。在托管网络中使用RTCP报告时必须考虑这一点。
This document defines a new SIP event package, vq-rtcpxr, and a new MIME type, application/vq-rtcpxr, that enable the collection and reporting of metrics that measure quality for RTP [3] sessions. The definitions of the metrics used in the event package are based on RTCP Extended Reports [4] and RTCP [3]; a mapping between the SIP event parameters and the parameters within the aforementioned RFCs is defined within this document in Section 4.6.2.
本文档定义了一个新的SIP事件包vq rtcpxr和一个新的MIME类型application/vq rtcpxr，用于收集和报告衡量RTP[3]会话质量的指标。事件包中使用的度量定义基于RTCP扩展报告[4]和RTCP[3]；本文件第4.6.2节定义了SIP事件参数与上述RFC内参数之间的映射。
Monitoring of voice quality is believed to be the highest priority for usage of this mechanism, and as such, the metrics in the event package are largely tailored for voice quality measurements. The event package is designed to be extensible. However, the negotiation of such extensions is not defined in this document.
语音质量监控被认为是使用该机制的最高优先级，因此，事件包中的度量主要针对语音质量测量进行定制。事件包被设计为可扩展的。但是，本文件未对此类延期的协商进行定义。
The event package supports reporting the voice quality metrics for both the inbound and outbound directions. Voice quality metrics for the inbound direction can generally be computed locally by the
事件包支持报告入站和出站方向的语音质量指标。入站方向的语音质量度量通常可以由
reporting endpoint; however, voice quality metrics for the outbound direction are computed by the remote endpoint and sent to the reporting endpoint using the RTCP Extended Reports [4].
报告终点；但是，出站方向的语音质量度量由远程端点计算，并使用RTCP扩展报告发送到报告端点[4]。
The configuration of the usage of this event package is not covered in this document. It is the recommendation of this document that the SIP configuration framework [15] be used. This is discussed in Section 4.8.
本文档不包括此事件包的使用配置。本文件建议使用SIP配置框架[15]。第4.8节对此进行了讨论。
The event package SHOULD be used with the SUBSCRIBE/NOTIFY method; however, it MAY also be used with the PUBLISH method [8] for backward compatibility with some existing implementations. Message flow examples for both methods are provided in this document.
事件包应与SUBSCRIBE/NOTIFY方法一起使用；但是，它也可以与发布方法[8]一起使用，以向后兼容某些现有实现。本文提供了这两种方法的消息流示例。
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 BCP 14, RFC 2119 [1].
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BCP 14、RFC 2119[1]中的描述进行解释。
This document defines a SIP events package [5] for Voice over IP performance reporting. A SIP UA can send these events to an entity that can make the information available to other applications. For purposes of illustration, the entities involved in SIP vq-rtcpxr event reporting will be referred to as follows:
本文档定义了用于IP语音性能报告的SIP事件包[5]。SIP UA可以将这些事件发送给一个实体，该实体可以将信息提供给其他应用程序。为便于说明，SIP vq rtcpxr事件报告中涉及的实体将被称为：
o REPORTER: an entity involved in the measurement and reporting of media quality, i.e., the SIP UA involved in a media session.
o 记者：参与媒体质量测量和报告的实体，即参与媒体会议的SIP UA。
o COLLECTOR: an entity that receives SIP vq-rtcpxr events. A COLLECTOR may be a proxy server or another entity that is capable of supporting SIP vq-rtcpxr events.
o 收集器：接收SIP vq rtcpxr事件的实体。收集器可以是代理服务器或能够支持SIP vq rtcpxr事件的另一个实体。
The COLLECTOR SHALL send a SUBSCRIBE to the REPORTER to explicitly establish the relationship. The REPORTER SHOULD send the voice quality metric reports using the NOTIFY method. The REPORTER MUST NOT send any vq-rtcpxr events if a COLLECTOR address has not been configured. The REPORTER populates the Request-URI according to the rules for an in-dialog request. The COLLECTOR MAY send a SUBSCRIBE to a SIP Proxy acting on behalf of the reporting SIP UAs.
催收员应向报告人发送订阅，以明确建立关系。报告者应使用NOTIFY方法发送语音质量度量报告。如果尚未配置收集器地址，则报告程序不得发送任何vq rtcpxr事件。报告器根据对话框内请求的规则填充请求URI。收集器可以向代表报告SIP ua的SIP代理发送订阅。
A SIP UA that supports this specification MAY also send the service quality metric reports using the PUBLISH method [8]; however, this approach SHOULD NOT be used, in general, on the public Internet. The PUBLISH method MAY be supported for backward compatibility with existing implementations.
支持此规范的SIP UA还可以使用发布方法[8]发送服务质量度量报告；然而，这种方法一般不应在公共互联网上使用。为了与现有实现向后兼容，可能支持PUBLISH方法。
The REPORTER MAY therefore populate the Request-URI of the PUBLISH method with the address of the COLLECTOR. To ensure security of SIP proxies and the COLLECTOR, the REPORTER MUST be configured with the address of the COLLECTOR, preferably using the SIP UA configuration framework [15], as described in Section 5.8.
因此，报告器可以使用收集器的地址填充发布方法的请求URI。为了确保SIP代理和收集器的安全，必须使用收集器的地址配置报告器，最好使用SIP UA配置框架[15]，如第5.8节所述。
It is RECOMMENDED that the REPORTER send an OPTIONS message to the COLLECTOR to ensure support of the PUBLISH message.
建议报告者向收集器发送选项消息，以确保对发布消息的支持。
If PUBLISH is not supported, then the REPORTER can only wait for a SUBSCRIBE request from the COLLECTOR and then deliver the information in NOTIFYs. If a REPORTER sends a PUBLISH to a COLLECTOR that does not support or allow this method, a 501 Not Implemented or a 405 Method Not Allowed response will be received, and the REPORTER will stop publication.
如果不支持发布，则报告者只能等待收集器的订阅请求，然后在NOTIFY中传递信息。如果报告器向不支持或不允许此方法的收集器发送发布，将收到501 not Implemented或405 method not Allowed响应，报告器将停止发布。
A voice quality metric report may be sent for each session terminating at the REPORTER, and it may contain multiple report bodies. For a multi-party call, the report MAY contain report bodies for the session between the reporting endpoint and each remote endpoint for which there was an RTP session during the call.
对于在报告者处终止的每个会话，可以发送语音质量度量报告，并且它可以包含多个报告主体。对于多方调用，报告可能包含报告端点和每个远程端点之间会话的报告主体，在调用过程中存在RTP会话。
Multi-party services such as call hold and call transfer can result in the user participating in a series of concatenated sessions, potentially with different choices of codec or sample rate, although these may be perceived by the user as a single call. A REPORTER MAY send a voice quality metric report at the end of each session or MAY send a single voice quality metric report containing an application/ vq-rtcpxr body for each segment of the call.
多方服务（例如呼叫保持和呼叫转移）可导致用户参与一系列串联会话，可能具有不同的编解码器或采样率选择，尽管这些可能被用户视为单个呼叫。报告者可以在每个会话结束时发送语音质量度量报告，也可以发送单个语音质量度量报告，其中包含呼叫每个段的应用程序/vq rtcpxr主体。
Users of this extension should ensure that they implement general SIP mechanisms for avoiding overload. For instance, an overloaded proxy or COLLECTOR MUST send a 503 Service Unavailable or other 5xx response with an appropriate Retry-After time specified. REPORTERs MUST act on these responses and respect the Retry-After time
此扩展的用户应确保实现通用SIP机制以避免过载。例如，重载代理或收集器必须在指定的时间后发送503服务不可用或其他5xx响应，并进行适当的重试。记者必须根据这些反应采取行动，并尊重事后重试
interval. In addition, future SIP extensions to better handle overload as covered in [14] should be followed as they are standardized.
间隔此外，应遵循[14]中所述的未来SIP扩展以更好地处理过载，因为它们是标准化的。
To avoid overload of SIP Proxies or COLLECTORS, it is important to do capacity planning and to minimize the number of reports that are sent.
为了避免SIP代理或收集器过载，必须进行容量规划并尽量减少发送的报告数量。
Approaches to avoiding overload include:
避免过载的方法包括：
a. Send only one report at the end of each call.
a. 每次通话结束时只发送一份报告。
b. Use interval reports only on "problem" calls that are being closely monitored.
b. 仅对被密切监视的“问题”呼叫使用间隔报告。
c. Limit the number of alerts that can be sent to a maximum of one per call.
c. 将每次呼叫可发送的警报数量限制为最多一个。
This document defines a SIP Event Package. SIP Event Packages were originally defined in RFC 3265 [5].
本文档定义了一个SIP事件包。SIP事件包最初是在RFC 3265[5]中定义的。
No event package parameters are defined.
未定义任何事件包参数。
SUBSCRIBE bodies are described by this specification.
订阅主体由本规范描述。
Subscriptions to this event package MAY range from minutes to weeks. Subscriptions in hours or days are more typical and are RECOMMENDED. The default subscription duration for this event package is one hour.
对此事件包的订阅可能在几分钟到几周之间。以小时或天为单位的订阅更为典型，建议使用。此事件包的默认订阅持续时间为一小时。
There are three notify bodies: a Session report, an Interval report, and an Alert report.
有三个通知主体：会话报告、间隔报告和警报报告。
The Session report SHOULD be used for reporting when a voice media session terminates, when a media change occurs, such as a codec change or a session fork, or when a session terminates due to no media packets being received and MUST NOT be used for reporting at
会话报告应用于报告语音媒体会话何时终止，发生媒体更改（如编解码器更改或会话分叉）时，或由于未接收到媒体数据包而终止会话时，并且不得用于报告
arbitrary points in time. This report MUST be used for cumulative metric reporting and the report timestamps MUST be from the start of a media session to the time at which the report is generated.
任意时间点。此报告必须用于累积度量报告，并且报告时间戳必须是从媒体会话开始到生成报告的时间。
The Interval report SHOULD be used for periodic or interval reporting and MUST NOT be used for reporting of the complete media session. This report is intended to capture short duration metric reporting and the report intervals SHOULD be non-overlapping time windows.
间隔报告应用于定期或间隔报告，不得用于报告整个媒体会话。此报告旨在捕获短期度量报告，报告间隔应为非重叠时间窗口。
The Alert report MAY be used when voice quality degrades during a session. The time window to which an Alert report relates MAY be a short time interval or from the start of the call to the point the alert is generated; this time window SHOULD be selected to provide the most useful information to support problem diagnosis.
当会话期间语音质量下降时，可以使用警报报告。警报报告所涉及的时间窗口可以是短时间间隔，也可以是从调用开始到警报生成点的时间窗口；应选择此时间窗口，以提供支持问题诊断的最有用信息。
Session, Interval, and Alert reports MUST populate the metrics with values that are measured over the interval explicitly defined by the "start" and "stop" timestamps.
会话、间隔和警报报告必须使用在“开始”和“停止”时间戳明确定义的间隔内测量的值填充度量。
Voice quality summary reports reference only one codec (payload type). This payload type SHOULD be the main voice payload, not comfort noise or telephone event payloads. For applications that consistently and rapidly switch codecs, the most used codec should be reported. All values in the report, such as IP addresses, synchronization source (SSRC), etc., represent those values as received by the REPORTER. In some scenarios, these may not be the same on either end of the session -- the COLLECTOR will need logic to be able to put these sessions together. The values of parameters such as sample rate, frame duration, frame octets, packets per second, round-trip delay, etc., depend on the type of report in which they are present. If present in a Session or an Interval report, they represent average values over the session or interval. If present in an Alert report, they represent instantaneous values.
语音质量摘要报告仅引用一个编解码器（负载类型）。该有效载荷类型应为主要语音有效载荷，而不是舒适性噪声或电话事件有效载荷。对于一致且快速切换编解码器的应用程序，应报告使用最多的编解码器。报告中的所有值，如IP地址、同步源（SSRC）等，都表示报告者接收到的值。在某些场景中，会话的两端可能不相同——收集器需要逻辑才能将这些会话放在一起。诸如采样率、帧持续时间、帧八位字节、每秒数据包、往返延迟等参数的值取决于它们所在的报告类型。如果存在于会话或间隔报告中，则它们表示会话或间隔的平均值。如果出现在警报报告中，则表示瞬时值。
The REPORTER always includes local quality reporting information and should, if possible, share remote quality reporting information to the COLLECTOR. This remote quality could be available from received RTCP-XR reports or other sources. Reporting this is useful in cases where the other end might support RTCP-XR but not this voice quality reporting.
报告人始终包括本地质量报告信息，如果可能，应向催收员共享远程质量报告信息。此远程质量可从收到的RTCP-XR报告或其他来源获得。报告在另一端可能支持RTCP-XR但不支持此语音质量报告的情况下，此功能非常有用。
This specification defines a new MIME type, application/vq-rtcpxr, which is a text encoding of the RTCP and RTCP-XR statistics with some additional metrics and correlation information.
本规范定义了一种新的MIME类型application/vq rtcpxr，它是RTCP和RTCP-XR统计数据的文本编码，带有一些额外的度量和相关信息。
This section describes the syntax extensions required for event publication in SIP. The formal syntax definitions described in this section are expressed in the Augmented BNF [6] format used in SIP [2] and contain references to elements defined therein.
本节描述SIP中事件发布所需的语法扩展。本节中描述的正式语法定义以SIP[2]中使用的扩充BNF[6]格式表示，并包含对其中定义元素的引用。
Additionally, the definition of the timestamp format is provided in [7]. Note that most of the parameters are optional. In practice, most implementations will send a subset of the parameters. It is not the intention of this document to define what parameters may or may not be useful for monitoring the quality of a voice session, but to enable reporting of voice quality. As such, the syntax allows the implementer to choose which metrics are most appropriate for their solution. As there are no "invalid", "unknown", or "not applicable" values in the syntax, the intention is to exclude any parameters for which values are not available, not applicable, or unknown.
此外，时间戳格式的定义见[7]。请注意，大多数参数都是可选的。实际上，大多数实现都会发送参数的子集。本文档的目的不是定义哪些参数可能对监控语音会话的质量有用，哪些参数可能不有用，而是为了能够报告语音质量。因此，语法允许实现者选择最适合其解决方案的度量。由于语法中没有“无效”、“未知”或“不适用”值，因此目的是排除任何值不可用、不适用或未知的参数。
The authors recognize that implementers may need to add new parameter lines to the reports and new metrics to the existing parameter lines. The extension tokens are intended to fulfill this need.
作者认识到，实施者可能需要在报告中添加新的参数行，并在现有参数行中添加新的度量。扩展令牌旨在满足这一需求。
VQReportEvent = AlertReport / SessionReport / IntervalReport
VQReportEvent = AlertReport / SessionReport / IntervalReport
SessionReport = "VQSessionReport" [ HCOLON "CallTerm" ] CRLF SessionInfo CRLF LocalMetrics [ CRLF RemoteMetrics ] [ CRLF DialogID ]
SessionReport=“VQSessionReport”[HCOLON“CallTerm”]CRLF SessionInfo CRLF LocalMetrics[CRLF RemoteMetrics][CRLF DialogID]
; CallTerm indicates the final report of a session.
; CallTerm表示会话的最终报告。
IntervalReport = "VQIntervalReport" [ HCOLON "CallTerm" ] CRLF SessionInfo CRLF LocalMetrics [ CRLF RemoteMetrics ] [ CRLF DialogID ]
IntervalReport=“VQIntervalReport”[HCOLON“CallTerm”]CRLF SessionInfo CRLF LocalMetrics[CRLF RemoteMetrics][CRLF DialogID]
LocalMetrics = "LocalMetrics" HCOLON CRLF Metrics
LocalMetrics=“LocalMetrics”HCOLON CRLF Metrics
RemoteMetrics = "RemoteMetrics" HCOLON CRLF Metrics
RemoteMetrics=“RemoteMetrics”HCOLON CRLF度量
AlertReport = "VQAlertReport" HCOLON MetricType WSP Severity WSP Direction CRLF SessionInfo CRLF LocalMetrics [ CRLF RemoteMetrics ] [ DialogID ]
AlertReport=“VQAlertReport”HCOLON MetricType WSP严重性WSP方向CRLF SessionInfo CRLF LocalMetrics[CRLF RemoteMetrics][DialogID]
SessionInfo = CallID CRLF LocalID CRLF RemoteID CRLF OrigID CRLF LocalAddr CRLF RemoteAddr CRLF LocalGroupID CRLF RemoteGroupID CRLF [ LocalMACAddr CRLF ] [ RemoteMACAddr CRLF ]
SessionInfo=CallID CRLF LocalID CRLF RemoteID CRLF OrigID CRLF LocalAddr CRLF RemoteAddr CRLF LocalGroupID CRLF RemoteGroupID CRLF[localmacadr CRLF][remotemacadr CRLF]
Metrics = TimeStamps CRLF
[ SessionDescription CRLF ]
[ JitterBuffer CRLF ]
[ PacketLoss CRLF ]
[ BurstGapLoss CRLF ]
[ Delay CRLF ]
[ Signal CRLF ]
[ QualityEstimates CRLF ]
*(Extension CRLF)
Metrics = TimeStamps CRLF
[ SessionDescription CRLF ]
[ JitterBuffer CRLF ]
[ PacketLoss CRLF ]
[ BurstGapLoss CRLF ]
[ Delay CRLF ]
[ Signal CRLF ]
[ QualityEstimates CRLF ]
*(Extension CRLF)
; Timestamps are provided in Coordinated Universal Time (UTC)
; using the ABNF format provided in RFC 3339,
; "Date and Time on the Internet: Timestamps"
; These timestamps SHOULD reflect, as closely as
; possible, the actual time during which the media session
; was running to enable correlation to events occurring
; in the network infrastructure and to accounting records.
; Time zones other than "Z" are not allowed.
; Timestamps are provided in Coordinated Universal Time (UTC)
; using the ABNF format provided in RFC 3339,
; "Date and Time on the Internet: Timestamps"
; These timestamps SHOULD reflect, as closely as
; possible, the actual time during which the media session
; was running to enable correlation to events occurring
; in the network infrastructure and to accounting records.
; Time zones other than "Z" are not allowed.
TimeStamps = "Timestamps" HCOLON StartTime WSP StopTime StartTime = "START" EQUAL date-time StopTime = "STOP" EQUAL date-time
TimeStamps=“TimeStamps”HCOLON StartTime WSP StopTime StartTime=“START”等日期时间StopTime=“STOP”等日期时间
; SessionDescription provides a shortened version of the
; session SDP but contains only the relevant parameters for
; session quality reporting purposes.
; SessionDescription provides a shortened version of the
; session SDP but contains only the relevant parameters for
; session quality reporting purposes.
SessionDescription = "SessionDesc" HCOLON
[ PayloadType WSP ]
[ PayloadDesc WSP ]
[ SampleRate WSP ]
[ PacketsPerSecond WSP ]
[ FrameDuration WSP ]
[ FrameOctets WSP ]
[ FramesPerPacket WSP ]
[ FmtpOptions WSP ]
SessionDescription = "SessionDesc" HCOLON
[ PayloadType WSP ]
[ PayloadDesc WSP ]
[ SampleRate WSP ]
[ PacketsPerSecond WSP ]
[ FrameDuration WSP ]
[ FrameOctets WSP ]
[ FramesPerPacket WSP ]
[ FmtpOptions WSP ]
[ PacketLossConcealment WSP ] [ SilenceSuppressionState ] *(WSP Extension)
[PacketLossConcealment WSP][silenesuppressionstate]*（WSP扩展）
; PayloadType provides the PT parameter used in the RTP packets.
; PayloadType提供RTP数据包中使用的PT参数。
PayloadType = "PT" EQUAL (1*3DIGIT)
PayloadType=“PT”等于（1*3数字）
; PayloadDesc provides a text description of the codec.
; This parameter SHOULD use the IANA registry for
; media-type names defined by RFC 4855 where it unambiguously
; defines the codec. Refer to the "Audio Media Types"
; registry on http://www.iana.org.
; PayloadDesc provides a text description of the codec.
; This parameter SHOULD use the IANA registry for
; media-type names defined by RFC 4855 where it unambiguously
; defines the codec. Refer to the "Audio Media Types"
; registry on http://www.iana.org.
PayloadDesc = "PD" EQUAL (word / DQUOTE word-plus DQUOTE)
PayloadDesc=“PD”相等（word/DQUOTE word加DQUOTE）
; SampleRate reports the rate at which a voice was sampled
; in the case of narrowband codecs, this value will typically
; be 8000.
; For codecs that are able to change sample rates, the lowest and
; highest sample rates MUST be reported (e.g., 8000;16000).
; SampleRate reports the rate at which a voice was sampled
; in the case of narrowband codecs, this value will typically
; be 8000.
; For codecs that are able to change sample rates, the lowest and
; highest sample rates MUST be reported (e.g., 8000;16000).
SampleRate = "SR" EQUAL (1*6DIGIT) *(SEMI (1*66DIGIT))
SampleRate = "SR" EQUAL (1*6DIGIT) *(SEMI (1*66DIGIT))
; FrameDuration can be combined with the FramesPerPacket
; to determine the packetization rate; the units for
; FrameDuration are milliseconds. NOTE: for frame-based codecs,
; each frame constitutes a single frame; for sample-based codecs,
; a "frame" refers to the set of samples carried in an RTP packet.
; FrameDuration can be combined with the FramesPerPacket
; to determine the packetization rate; the units for
; FrameDuration are milliseconds. NOTE: for frame-based codecs,
; each frame constitutes a single frame; for sample-based codecs,
; a "frame" refers to the set of samples carried in an RTP packet.
FrameDuration = "FD" EQUAL (1*4DIGIT)
FrameDuration=“FD”等于（1*4数字）
; FrameOctets provides the number of octets in each frame
; at the time the report is generated (i.e., last value).
; This MAY be used where FrameDuration is not available.
; NOTE: for frame-based codecs, each frame constitutes a single frame;
; for sample-based codecs, a "frame" refers to the set of samples
; carried in an RTP packet.
; FrameOctets provides the number of octets in each frame
; at the time the report is generated (i.e., last value).
; This MAY be used where FrameDuration is not available.
; NOTE: for frame-based codecs, each frame constitutes a single frame;
; for sample-based codecs, a "frame" refers to the set of samples
; carried in an RTP packet.
FrameOctets = "FO" EQUAL (1*5DIGIT)
FrameOctets=“FO”等于（1*5位）
; FramesPerPacket provides the number of frames in each RTP
; packet at the time the report is generated.
; NOTE: for frame-based codecs, each frame constitutes a single frame;
; for sample-based codecs, a "frame" refers to the set of samples
; carried in an RTP packet.
; FramesPerPacket provides the number of frames in each RTP
; packet at the time the report is generated.
; NOTE: for frame-based codecs, each frame constitutes a single frame;
; for sample-based codecs, a "frame" refers to the set of samples
; carried in an RTP packet.
FramesPerPacket = "FPP" EQUAL (1*2DIGIT)
FramesPerPacket=“FPP”相等（1*2数字）
; Packets per second provides the average number of packets
; that are transmitted per second, as at the time the report is
; generated.
; Packets per second provides the average number of packets
; that are transmitted per second, as at the time the report is
; generated.
PacketsPerSecond = "PPS" EQUAL (1*5DIGIT)
PacketsPerSecond=“PPS”相等（1*5位）
; FMTP options from SDP. Note that the parameter is delineated
; by " " to avoid parsing issues in transitioning between SDP
; and SIP parsing.
; FMTP options from SDP. Note that the parameter is delineated
; by " " to avoid parsing issues in transitioning between SDP
; and SIP parsing.
FmtpOptions = "FMTP" EQUAL DQUOTE word-plus DQUOTE
FmtpOptions=“FMTP”等于DQUOTE单词加DQUOTE
; PacketLossConcealment indicates whether a PLC algorithm was
; or is being used for the session. The values follow the same
; numbering convention as RFC 3611 [4].
; 0 - unspecified
; 1 - disabled
; 2 - enhanced
; 3 - standard
; PacketLossConcealment indicates whether a PLC algorithm was
; or is being used for the session. The values follow the same
; numbering convention as RFC 3611 [4].
; 0 - unspecified
; 1 - disabled
; 2 - enhanced
; 3 - standard
PacketLossConcealment = "PLC" EQUAL ("0" / "1" / "2" / "3")
PacketLossConcealment = "PLC" EQUAL ("0" / "1" / "2" / "3")
; SilenceSuppressionState indicates whether silence suppression, ; also known as Voice Activity Detection (VAD) is enabled.
; 沉默抑制状态表示沉默抑制是否为；启用也称为语音活动检测（VAD）。
SilenceSuppressionState = "SSUP" EQUAL ("on" / "off")
SilenceSuppressionState = "SSUP" EQUAL ("on" / "off")
; CallId provides the call id from the SIP dialog.
; CallId提供SIP对话框中的呼叫id。
CallID = "CallID" HCOLON Call-ID-Parm
CallID=“CallID”HCOLON调用ID参数
; LocalID identifies the reporting endpoint for the media session [2].
; LocalID标识媒体会话的报告端点[2]。
LocalID = "LocalID" HCOLON (name-addr/addr-spec)
LocalID=“LocalID”HCOLON（名称addr/addr spec）
; RemoteID identifies the remote endpoint of the media session [2].
; RemoteID标识媒体会话的远程端点[2]。
RemoteID = "RemoteID" HCOLON (name-addr/addr-spec)
RemoteID=“RemoteID”HCOLON（名称addr/addr spec）
; OrigID identifies the endpoint which originated the session.
; OrigID标识发起会话的端点。
OrigID = "OrigID" HCOLON (name-addr/addr-spec)
OrigID=“OrigID”HCOLON（名称addr/addr spec）
; LocalAddr provides the IP address, port, and SSRC of the
; endpoint/UA, which is the receiving end of the stream being
; measured.
; LocalAddr provides the IP address, port, and SSRC of the
; endpoint/UA, which is the receiving end of the stream being
; measured.
LocalAddr = "LocalAddr" HCOLON IPAddress WSP Port WSP Ssrc
LocalAddr=“LocalAddr”HCOLON IPAddress WSP端口WSP Ssrc
; RemoteAddr provides the IP address, port, and SSRC of the ; the source of the stream being measured.
; RemoteAddr提供服务器的IP地址、端口和SSRC；正在测量的流的源。
RemoteAddr = "RemoteAddr" HCOLON IPAddress WSP Port WSP Ssrc
RemoteAddr=“RemoteAddr”HCOLON IPAddress WSP端口WSP Ssrc
; LocalMACAddr provides the Media Access Control (MAC) address ; of the local SIP device.
; LocalMACAddr提供媒体访问控制（MAC）地址；本地SIP设备的。
LocalMACAddr = "LocalMAC" HCOLON hex2 *(":" hex2)
LocalMACAddr = "LocalMAC" HCOLON hex2 *(":" hex2)
; RemoteMACAddr provides the MAC address ; of the remote SIP device.
; RemoteMACAddr提供MAC地址；远程SIP设备的。
RemoteMACAddr = "RemoteMAC" HCOLON hex2 *(":" hex2)
RemoteMACAddr = "RemoteMAC" HCOLON hex2 *(":" hex2)
; LocalGroupID provides the identification for the purposes ; of aggregation for the local endpoint.
; LocalGroupID提供用于以下目的的标识：；本地终结点的聚合。
LocalGroupID = "LocalGroup" HCOLON word-plus
LocalGroupID=“LocalGroup”HCOLON word plus
; RemoteGroupID provides the identification for the purposes ; of aggregation for the remote endpoint.
; RemoteGroupID提供用于以下目的的标识：；远程终结点的聚合类型。
RemoteGroupID = "RemoteGroup" HCOLON word-plus
RemoteGroupID=“RemoteGroup”HCOLON word plus
; For clarification, the LocalAddr in the LocalMetrics report ; MUST be the RemoteAddr in the RemoteMetrics report.
; 为了澄清，LocalMetrics报告中的LocalAddr；必须是RemoteMetrics报告中的RemoteAddr。
IPAddress = "IP" EQUAL IPv6address / IPv4address
Port = "PORT" EQUAL 1*DIGIT
Ssrc = "SSRC" EQUAL ( %x30.78 1*8HEXDIG)
IPAddress = "IP" EQUAL IPv6address / IPv4address
Port = "PORT" EQUAL 1*DIGIT
Ssrc = "SSRC" EQUAL ( %x30.78 1*8HEXDIG)
JitterBuffer = "JitterBuffer" HCOLON
[ JitterBufferAdaptive WSP ]
[ JitterBufferRate WSP ]
[ JitterBufferNominal WSP ]
[ JitterBufferMax WSP ]
[ JitterBufferAbsMax ]
*(WSP Extension)
JitterBuffer = "JitterBuffer" HCOLON
[ JitterBufferAdaptive WSP ]
[ JitterBufferRate WSP ]
[ JitterBufferNominal WSP ]
[ JitterBufferMax WSP ]
[ JitterBufferAbsMax ]
*(WSP Extension)
; JitterBufferAdaptive indicates whether the jitter buffer in
; the endpoint is adaptive, static, or unknown.
; The values follow the same numbering convention as RFC 3611 [4].
; For more details, please refer to that document.
; 0 - unknown
; 1 - reserved
; 2 - non-adaptive
; 3 - adaptive
; JitterBufferAdaptive indicates whether the jitter buffer in
; the endpoint is adaptive, static, or unknown.
; The values follow the same numbering convention as RFC 3611 [4].
; For more details, please refer to that document.
; 0 - unknown
; 1 - reserved
; 2 - non-adaptive
; 3 - adaptive
JitterBufferAdaptive = "JBA" EQUAL ("0" / "1" / "2" / "3")
JitterBufferAdaptive = "JBA" EQUAL ("0" / "1" / "2" / "3")
; JitterBuffer metric definitions are provided in RFC 3611 [4].
; RFC 3611[4]中提供了JitterBuffer度量定义。
JitterBufferRate = "JBR" EQUAL (1*2DIGIT) ;0-15
JitterBufferNominal = "JBN" EQUAL (1*5DIGIT) ;0-65535
JitterBufferMax = "JBM" EQUAL (1*5DIGIT) ;0-65535
JitterBufferAbsMax = "JBX" EQUAL (1*5DIGIT) ;0-65535
JitterBufferRate = "JBR" EQUAL (1*2DIGIT) ;0-15
JitterBufferNominal = "JBN" EQUAL (1*5DIGIT) ;0-65535
JitterBufferMax = "JBM" EQUAL (1*5DIGIT) ;0-65535
JitterBufferAbsMax = "JBX" EQUAL (1*5DIGIT) ;0-65535
; PacketLoss metric definitions are provided in RFC 3611 [4].
; RFC 3611[4]中提供了包装损耗度量定义。
PacketLoss = "PacketLoss" HCOLON [ NetworkPacketLossRate WSP ] [ JitterBufferDiscardRate ] *(WSP Extension)
PacketLoss=“PacketLoss”HCOLON[NetworkPacketLossRate WSP][JitterBufferDiscardRate]*（WSP扩展）
NetworkPacketLossRate =
"NLR" EQUAL (1*3DIGIT [ "." 1*2DIGIT ]) ;percentage
NetworkPacketLossRate =
"NLR" EQUAL (1*3DIGIT [ "." 1*2DIGIT ]) ;percentage
JitterBufferDiscardRate =
"JDR" EQUAL (1*3DIGIT [ "." 1*2DIGIT ]) ;percentage
JitterBufferDiscardRate =
"JDR" EQUAL (1*3DIGIT [ "." 1*2DIGIT ]) ;percentage
; BurstGapLoss metric definitions are provided in RFC 3611 [4].
; RFC 3611[4]中提供了突发性概率度量定义。
BurstGapLoss = "BurstGapLoss" HCOLON
[ BurstLossDensity WSP ]
[ BurstDuration WSP ]
[ GapLossDensity WSP ]
[ GapDuration WSP ]
[ MinimumGapThreshold ]
*(WSP Extension)
BurstGapLoss = "BurstGapLoss" HCOLON
[ BurstLossDensity WSP ]
[ BurstDuration WSP ]
[ GapLossDensity WSP ]
[ GapDuration WSP ]
[ MinimumGapThreshold ]
*(WSP Extension)
BurstLossDensity =
"BLD" EQUAL (1*3DIGIT [ "." 1*2DIGIT ]) ;percentage
BurstLossDensity =
"BLD" EQUAL (1*3DIGIT [ "." 1*2DIGIT ]) ;percentage
BurstDuration =
"BD" EQUAL (1*7DIGIT) ;0-3,600,000 -- milliseconds
BurstDuration =
"BD" EQUAL (1*7DIGIT) ;0-3,600,000 -- milliseconds
GapLossDensity =
"GLD" EQUAL (1*3DIGIT [ "." 1*2DIGIT ]) ;percentage
GapLossDensity =
"GLD" EQUAL (1*3DIGIT [ "." 1*2DIGIT ]) ;percentage
GapDuration =
"GD" EQUAL (1*7DIGIT) ;0-3,600,000 -- milliseconds
GapDuration =
"GD" EQUAL (1*7DIGIT) ;0-3,600,000 -- milliseconds
MinimumGapThreshold = "GMIN" EQUAL (1*3DIGIT) ;1-255
最小gapthreshold=“GMIN”等于（1*3位数）；1-255
Delay = "Delay" HCOLON
[ RoundTripDelay WSP ]
[ EndSystemDelay WSP ]
[ OneWayDelay WSP ]
[ SymmOneWayDelay WSP ]
[ InterarrivalJitter WSP ]
[ MeanAbsoluteJitter ]
*(WSP Extension)
Delay = "Delay" HCOLON
[ RoundTripDelay WSP ]
[ EndSystemDelay WSP ]
[ OneWayDelay WSP ]
[ SymmOneWayDelay WSP ]
[ InterarrivalJitter WSP ]
[ MeanAbsoluteJitter ]
*(WSP Extension)
; RoundTripDelay SHALL be measured as defined in RFC 3550 [3].
; 应按照RFC 3550[3]的规定测量往返延迟。
RoundTripDelay = "RTD" EQUAL (1*5DIGIT) ;0-65535
RoundTripDelay = "RTD" EQUAL (1*5DIGIT) ;0-65535
; EndSystemDelay metric is defined in RFC 3611 [4].
; RFC 3611[4]中定义了EndSystemDelay度量。
EndSystemDelay = "ESD" EQUAL (1*5DIGIT) ;0-65535
EndSystemDelay = "ESD" EQUAL (1*5DIGIT) ;0-65535
; OneWayDelay is defined in RFC 2679 [12].
; 单向延迟在RFC 2679[12]中定义。
OneWayDelay = "OWD" EQUAL (1*5DIGIT) ;0-65535
OneWayDelay = "OWD" EQUAL (1*5DIGIT) ;0-65535
; SymmOneWayDelay is defined as half the sum of RoundTripDelay ; and the EndSystemDelay values for both endpoints.
; SymmOneWayDelay定义为RoundTripDelay之和的一半；以及两个端点的EndSystemDelay值。
SymmOneWayDelay = "SOWD" EQUAL (1*5DIGIT); 0-65535
SymmOneWayDelay = "SOWD" EQUAL (1*5DIGIT); 0-65535
; Interarrival Jitter is calculated as defined RFC 3550 [3] ; and converted into milliseconds.
; 根据RFC 3550[3]的定义计算到达间抖动；并转换成毫秒。
InterarrivalJitter = "IAJ" EQUAL (1*5DIGIT) ;0-65535 ms
InterarrivalJitter = "IAJ" EQUAL (1*5DIGIT) ;0-65535 ms
; Mean Absolute Jitter is measured as defined ; by ITU-T G.1020 [9] where it is known as MAPDV.
; 根据定义测量平均绝对抖动；由ITU-T G.1020[9]编制，其中称为MAPDV。
MeanAbsoluteJitter = "MAJ" EQUAL (1*5DIGIT);0-65535
MeanAbsoluteJitter = "MAJ" EQUAL (1*5DIGIT);0-65535
; Signal metrics definitions are provided in RFC 3611 [4].
; RFC 3611[4]中提供了信号度量定义。
Signal = "Signal" HCOLON
[ SignalLevel WSP ]
[ NoiseLevel WSP ]
[ ResidualEchoReturnLoss ]
*(WSP Extension)
Signal = "Signal" HCOLON
[ SignalLevel WSP ]
[ NoiseLevel WSP ]
[ ResidualEchoReturnLoss ]
*(WSP Extension)
; SignalLevel will normally be a negative value.
; 信号电平通常为负值。
; The absence of the negative sign indicates a positive value.
; Where the signal level is negative, the sign MUST be
; included. This metric applies to the speech signal decoded
; from the received packet stream.
; The absence of the negative sign indicates a positive value.
; Where the signal level is negative, the sign MUST be
; included. This metric applies to the speech signal decoded
; from the received packet stream.
SignalLevel = "SL" EQUAL ([ "-" ] 1*2DIGIT)
SignalLevel=“SL”等于（[“-”]1*2数字）
; NoiseLevel will normally be negative and the sign MUST be
; explicitly included.
; The absence of a sign indicates a positive value.
; This metric applies to the speech signal decoded from the
; received packet stream.
; NoiseLevel will normally be negative and the sign MUST be
; explicitly included.
; The absence of a sign indicates a positive value.
; This metric applies to the speech signal decoded from the
; received packet stream.
NoiseLevel = "NL" EQUAL ([ "-" ] 1*2DIGIT)
NoiseLevel=“NL”等于（[“-”]1*2数字）
; Residual Echo Return Loss (RERL) is the ratio between
; the original signal and the echo level as measured after
; echo cancellation or suppression has been applied.
; Expressed in decibels (dB). This is typically a positive
; value.
; This metric relates to the proportion of the speech signal
; decoded from the received packet stream that is reflected
; back in the encoded speech signal output in the transmitted
; packet stream (i.e., will affect the REMOTE user's
; conversational quality). To support the diagnosis of echo-
; related problems experienced by the local user of the device
; generating a report according to this document, the value of
; RERL reported via the RTCP XR VoIP Metrics payload SHOULD be
; reported in the RemoteMetrics set of data.
; Residual Echo Return Loss (RERL) is the ratio between
; the original signal and the echo level as measured after
; echo cancellation or suppression has been applied.
; Expressed in decibels (dB). This is typically a positive
; value.
; This metric relates to the proportion of the speech signal
; decoded from the received packet stream that is reflected
; back in the encoded speech signal output in the transmitted
; packet stream (i.e., will affect the REMOTE user's
; conversational quality). To support the diagnosis of echo-
; related problems experienced by the local user of the device
; generating a report according to this document, the value of
; RERL reported via the RTCP XR VoIP Metrics payload SHOULD be
; reported in the RemoteMetrics set of data.
ResidualEchoReturnLoss = "RERL" EQUAL (1*3DIGIT)
剩余电量等于（1*3位数）
; Voice Quality estimation metrics.
; Each quality estimate has an optional associated algorithm.
; These fields permit the implementation to use a variety
; of different calculation methods for each type of metric.
; Voice Quality estimation metrics.
; Each quality estimate has an optional associated algorithm.
; These fields permit the implementation to use a variety
; of different calculation methods for each type of metric.
QualityEstimates = "QualityEst" HCOLON
[ ListeningQualityR WSP ]
[ RLQEstAlg WSP ]
[ ConversationalQualityR WSP ]
[ RCQEstAlg WSP ]
[ ExternalR-In WSP ]
[ ExtRInEstAlg WSP ]
[ ExternalR-Out WSP ]
[ ExtROutEstAlg WSP ]
[ MOS-LQ WSP ]
[ MOSLQEstAlg WSP ]
QualityEstimates = "QualityEst" HCOLON
[ ListeningQualityR WSP ]
[ RLQEstAlg WSP ]
[ ConversationalQualityR WSP ]
[ RCQEstAlg WSP ]
[ ExternalR-In WSP ]
[ ExtRInEstAlg WSP ]
[ ExternalR-Out WSP ]
[ ExtROutEstAlg WSP ]
[ MOS-LQ WSP ]
[ MOSLQEstAlg WSP ]
[ MOS-CQ WSP ]
[ MOSCQEstAlg WSP ]
[ QoEEstAlg ]
*(WSP Extension)
[ MOS-CQ WSP ]
[ MOSCQEstAlg WSP ]
[ QoEEstAlg ]
*(WSP Extension)
ListeningQualityR = "RLQ" EQUAL (1*3DIGIT) ; 0 - 120
ListeningQualityR = "RLQ" EQUAL (1*3DIGIT) ; 0 - 120
RLQEstAlg = "RLQEstAlg" EQUAL word ; "P.564" [10], or other
RLQEstAlg=“RLQEstAlg”等字；“第564页”[10]，或其他
ConversationalQualityR = "RCQ" EQUAL (1*3DIGIT) ; 0 - 120
ConversationalQualityR = "RCQ" EQUAL (1*3DIGIT) ; 0 - 120
RCQEstAlg = "RCQEstAlg" EQUAL word ; "P.564", or other
RCQEstAlg=“RCQEstAlg”等字；“第564页”，或其他
; ExternalR-In is measured by the local endpoint for incoming
; connection on the "other" side of this endpoint. For example,
; Phone A <---> Bridge <----> Phone B
; ListeningQualityR = quality for Phone A ----> Bridge path
; ExternalR-In = quality for Bridge <---- Phone B path
; ExternalR-In is measured by the local endpoint for incoming
; connection on the "other" side of this endpoint. For example,
; Phone A <---> Bridge <----> Phone B
; ListeningQualityR = quality for Phone A ----> Bridge path
; ExternalR-In = quality for Bridge <---- Phone B path
ExternalR-In = "EXTRI" EQUAL (1*3DIGIT) ; 0 - 120
ExternalR-In = "EXTRI" EQUAL (1*3DIGIT) ; 0 - 120
ExtRInEstAlg = "ExtRIEstAlg" EQUAL word ; "P.564" or other
ExtRInEstAlg=“ExtRIEstAlg”等字；“第564页”或其他
; ExternalR-Out is copied from the RTCP XR message received from the
; remote endpoint on the "other" side of this endpoint. For example,
; Phone A <---> Bridge <----> Phone B
; ExternalR-Out = quality for Bridge -----> Phone B path
; ExternalR-Out is copied from the RTCP XR message received from the
; remote endpoint on the "other" side of this endpoint. For example,
; Phone A <---> Bridge <----> Phone B
; ExternalR-Out = quality for Bridge -----> Phone B path
ExternalR-Out = "EXTRO" EQUAL (1*3DIGIT) ; 0 - 120
ExternalR-Out = "EXTRO" EQUAL (1*3DIGIT) ; 0 - 120
ExtROutEstAlg = "ExtROEstAlg" EQUAL word ; "P.564" or other
ExtROEstAlg=“ExtROEstAlg”等字；“第564页”或其他
MOS-LQ = "MOSLQ" EQUAL (DIGIT [ "." 1*3DIGIT ]) ; 0.0 - 4.9
MOS-LQ = "MOSLQ" EQUAL (DIGIT [ "." 1*3DIGIT ]) ; 0.0 - 4.9
MOSLQEstAlg = "MOSLQEstAlg" EQUAL word ; "P.564" or other
MOSLQEstAlg=“MOSLQEstAlg”等字；“第564页”或其他
MOS-CQ = "MOSCQ" EQUAL (DIGIT [ "." 1*3DIGIT ]) ; 0.0 - 4.9
MOS-CQ = "MOSCQ" EQUAL (DIGIT [ "." 1*3DIGIT ]) ; 0.0 - 4.9
MOSCQEstAlg = "MOSCQEstAlg" EQUAL word ; "P.564" or other
moscquestalg=“moscquestalg”等字；“第564页”或其他
; QoEEstAlg provides an alternative to the separate
; estimation algorithms for use when the same algorithm
; is used for all measurements.
; QoEEstAlg provides an alternative to the separate
; estimation algorithms for use when the same algorithm
; is used for all measurements.
QoEEstAlg = "QoEEstAlg" EQUAL word ; "P.564" or other
QoEEstAlg=“QoEEstAlg”等字；“第564页”或其他
; DialogID provides the identification of the dialog with
; which the media session is related. This value is taken
; from the SIP header.
; DialogID provides the identification of the dialog with
; which the media session is related. This value is taken
; from the SIP header.
DialogID = "DialogID" COLON Call-ID-Parm *(SEMI did-parm)
DialogID=“DialogID”冒号调用ID Parm*（半did Parm）
did-parm = to-tag / from-tag / word
did-parm = to-tag / from-tag / word
to-tag = "to-tag" EQUAL token
to tag=“to tag”相等标记
from-tag = "from-tag" EQUAL token
from tag=“from tag”相等标记
; MetricType provides the metric on which a notification of
; threshold violation was based. The more commonly used metrics
; for alerting purposes are included here explicitly, using the
; character encoding that represents the parameter in
; this ABNF. The Extension parameter can be used to provide
; metrics that are not defined by this document.
; MetricType provides the metric on which a notification of
; threshold violation was based. The more commonly used metrics
; for alerting purposes are included here explicitly, using the
; character encoding that represents the parameter in
; this ABNF. The Extension parameter can be used to provide
; metrics that are not defined by this document.
MetricType = "Type" EQUAL "RLQ" / "RCQ" / "EXTR" /
"MOSLQ" / "MOSCQ" /
"BD" / "NLR" / "JDR" /
"RTD" / "ESD" / "IAJ" /
"RERL" / "SL" / "NL" / Extension
MetricType = "Type" EQUAL "RLQ" / "RCQ" / "EXTR" /
"MOSLQ" / "MOSCQ" /
"BD" / "NLR" / "JDR" /
"RTD" / "ESD" / "IAJ" /
"RERL" / "SL" / "NL" / Extension
Direction = "Dir" EQUAL "local" / "remote"
Severity = "Severity" EQUAL "Warning" / "Critical" /
"Clear"
Direction = "Dir" EQUAL "local" / "remote"
Severity = "Severity" EQUAL "Warning" / "Critical" /
"Clear"
Call-ID-Parm = word [ "@" word ]
呼叫ID Parm=word[“@”word]
; General ABNF notation from RFC 5234.
; RFC 5234中的一般ABNF符号。
CRLF = %x0D.0A
DIGIT = %x30-39
WSP = SP / HTAB ; white space
SP = " "
HTAB = %x09 ; horizontal tab
HEXDIG = DIGIT / "A" / "B" / "C" / "D" / "E" / "F" /
"a" / "b" / "c" / "d" / "e" / "f"
DQUOTE = %x22 ; " (Double Quote)
ALPHA = %x41-5A / %x61-7A ; A-Z / a-z
CRLF = %x0D.0A
DIGIT = %x30-39
WSP = SP / HTAB ; white space
SP = " "
HTAB = %x09 ; horizontal tab
HEXDIG = DIGIT / "A" / "B" / "C" / "D" / "E" / "F" /
"a" / "b" / "c" / "d" / "e" / "f"
DQUOTE = %x22 ; " (Double Quote)
ALPHA = %x41-5A / %x61-7A ; A-Z / a-z
; ABNF notation from RFC 3261.
; RFC3261中的ABNF符号。
alphanum = ALPHA / DIGIT
LWS = [ *WSP CRLF ] 1*WSP ; linear whitespace
SWS = [ LWS ] ; sep whitespace
SEMI = SWS ";" SWS ; semicolon
EQUAL = SWS "=" SWS ; equal
COLON = SWS ":" SWS ; colon
HCOLON = *( SP / HTAB ) ":" SWS
alphanum = ALPHA / DIGIT
LWS = [ *WSP CRLF ] 1*WSP ; linear whitespace
SWS = [ LWS ] ; sep whitespace
SEMI = SWS ";" SWS ; semicolon
EQUAL = SWS "=" SWS ; equal
COLON = SWS ":" SWS ; colon
HCOLON = *( SP / HTAB ) ":" SWS
token = 1*(alphanum / "-" / "." / "!" / "%" / "*"
/ "_" / "+" / "`" / "'" / "~" )
token = 1*(alphanum / "-" / "." / "!" / "%" / "*"
/ "_" / "+" / "`" / "'" / "~" )
IPv4address = 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT
IPv6address = hexpart [ ":" IPv4address ]
hexpart = hexseq / hexseq "::" [ hexseq ] / "::"
[ hexseq ]
hexseq = hex4 *( ":" hex4)
hex4 = 1*4HEXDIG
hex2 = 2HEXDIG
IPv4address = 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT
IPv6address = hexpart [ ":" IPv4address ]
hexpart = hexseq / hexseq "::" [ hexseq ] / "::"
[ hexseq ]
hexseq = hex4 *( ":" hex4)
hex4 = 1*4HEXDIG
hex2 = 2HEXDIG
; ABNF notation from RFC 3339.
; RFC3339中的ABNF符号。
date-fullyear = 4DIGIT ; e.g. 2006
date-month = 2DIGIT ; e.g. 01 or 11
date-mday = 2DIGIT ; e.g. 02 or 22
time-hour = 2DIGIT ; e.g. 01 or 13
time-minute = 2DIGIT ; e.g. 03 or 55
time-second = 2DIGIT ; e.g. 01 or 59
time-secfrac = "." 1*DIGIT
time-numoffset = ("+" / "-") time-hour ":" time-minute
time-offset = "Z" / time-numoffset
partial-time = time-hour ":" time-minute ":" time-second [ time-secfrac]
full-date = date-fullyear "-" date-month "-" date-mday
full-time = partial-time time-offset
date-time = full-date "T" full-time
date-fullyear = 4DIGIT ; e.g. 2006
date-month = 2DIGIT ; e.g. 01 or 11
date-mday = 2DIGIT ; e.g. 02 or 22
time-hour = 2DIGIT ; e.g. 01 or 13
time-minute = 2DIGIT ; e.g. 03 or 55
time-second = 2DIGIT ; e.g. 01 or 59
time-secfrac = "." 1*DIGIT
time-numoffset = ("+" / "-") time-hour ":" time-minute
time-offset = "Z" / time-numoffset
partial-time = time-hour ":" time-minute ":" time-second [ time-secfrac]
full-date = date-fullyear "-" date-month "-" date-mday
full-time = partial-time time-offset
date-time = full-date "T" full-time
; Miscellaneous definitions ;
; 杂项定义；
Extension = word-plus
Extension = word-plus
word = 1*(alphanum / "-" / "." / "!" / "%" / "*" /
"_" / "+" / "`" / "'" / "~" /
"(" / ")" / "<" / ">" /
":" / "\" / DQUOTE /
"/" / "[" / "]" / "?" )
word = 1*(alphanum / "-" / "." / "!" / "%" / "*" /
"_" / "+" / "`" / "'" / "~" /
"(" / ")" / "<" / ">" /
":" / "\" / DQUOTE /
"/" / "[" / "]" / "?" )
word-plus = 1*(alphanum / "-" / "." / "!" / "%" / "*" /
"_" / "+" / "`" / "'" / "~" /
"(" / ")" / "<" / ">" / ":" /
"\" / "/" / "[" / "]" / "?" /
"{" / "}" / "=" / " ")
word-plus = 1*(alphanum / "-" / "." / "!" / "%" / "*" /
"_" / "+" / "`" / "'" / "~" /
"(" / ")" / "<" / ">" / ":" /
"\" / "/" / "[" / "]" / "?" /
"{" / "}" / "=" / " ")
Parameter values, codec types, and other aspects of the endpoints may change dynamically during a session. The reported values of metrics and configuration parameters SHALL be the current value at the time the report is generated.
在会话期间，端点的参数值、编解码器类型和其他方面可能会动态更改。度量和配置参数的报告值应为生成报告时的当前值。
The Packet Loss Rate and Packet Discard Rate parameters are calculated over the period between the starting and ending timestamps for the report. These are normally calculated from a count of the number of lost or discarded packets divided by the count of the number of packets, and hence are based on the current values of these counters at the time the report was generated.
数据包丢失率和数据包丢弃率参数是在报告的开始和结束时间戳之间的时间段内计算的。这些计数器通常根据丢失或丢弃的数据包数量除以数据包数量的计数来计算，因此基于生成报告时这些计数器的当前值。
Packet delay variation, signal level, noise level, and echo level are computed as running or interval averages, based on the appropriate standard, e.g., RFC 3550 for Packet Delay Variation (PDV), and the sampled value of these running averages is reported. Delay, packet size, jitter buffer size, and codec-related data may change during a session and the current value of these parameters is reported as sampled at the time the report is generated.
分组延迟变化、信号电平、噪声电平和回波电平根据适当的标准（例如分组延迟变化（PDV）的RFC 3550）计算为运行或间隔平均值，并报告这些运行平均值的采样值。延迟、数据包大小、抖动缓冲区大小和编解码器相关数据可能在会话期间发生变化，这些参数的当前值在生成报告时报告为采样值。
RFC 3611 uses an 8-bit, fixed-point number with the binary point at the left edge of the field. This value is calculated by dividing the total number of packets lost by the total number of packets expected and multiplying the result by 256, and then taking the integer part.
RFC3611使用8位定点数字，二进制点位于字段的左边缘。该值的计算方法是将丢失的数据包总数除以预期的数据包总数，并将结果乘以256，然后取整数部分。
For any RTCP XR parameter in this format, to map into the equivalent SIP vq-rtcpxr parameter, simply reverse the equation, i.e., divide by 256 and take the integer part.
对于此格式的任何RTCP XR参数，要映射到等效的SIP vq rtcpxr参数，只需反转方程，即除以256并取整数部分。
Following SIP and other IETF conventions, timestamps are provided in Coordinated Universal Time (UTC) using the ABNF format provided in RFC 3339 [7]. These timestamps SHOULD reflect, as closely as possible, the actual time during which the media session was running to enable correlation to related events occurring in the network and to accounting or billing records.
按照SIP和其他IETF约定，使用RFC 3339[7]中提供的ABNF格式以协调世界时（UTC）的形式提供时间戳。这些时间戳应尽可能准确地反映媒体会话运行的实际时间，以便与网络中发生的相关事件以及记帐或计费记录相关联。
The parameters in this field provide a shortened version of the session SDP(s), containing only the relevant parameters for session quality reporting purposes. Where values may change during a session, for example, a codec may change rate, then the most-recent value of the parameter is reported.
此字段中的参数提供会话SDP的缩短版本，仅包含用于会话质量报告目的的相关参数。例如，在会话期间值可能改变的情况下，编解码器可能改变速率，然后报告参数的最新值。
This is the "payload type" parameter used in the RTP packets, i.e., the codec. This field can also be mapped from the SDP "rtpmap" attribute field "payload type". IANA-registered types SHOULD be used.
这是RTP数据包（即编解码器）中使用的“有效负载类型”参数。此字段也可以从SDP“rtpmap”属性字段“有效负载类型”映射。应使用IANA注册的类型。
This parameter is a text description of the codec. This parameter SHOULD use the IANA registry for media-type names where it unambiguously defines the codec. Refer to the "Audio Media Types" registry on http://www.iana.org.
此参数是编解码器的文本描述。此参数应使用IANA注册表作为媒体类型名称，并在其中明确定义编解码器。请参阅上的“音频媒体类型”注册表http://www.iana.org.
This parameter is mapped from the SDP "rtpmap" attribute field "clock rate". The field provides the rate at which a voice was sampled, measured in Hertz (Hz).
此参数从SDP“rtpmap”属性字段“时钟频率”映射而来。该字段提供语音采样的速率，以赫兹（Hz）为单位。
This parameter is not contained in RTP or SDP but can usually be obtained from the device codec. Packets per second provides the (rounded) number of RTP packets that are transmitted per second.
此参数不包含在RTP或SDP中，但通常可以从设备编解码器获得。Packets per second提供每秒传输的RTP数据包的（舍入）数量。
This parameter is not contained in RTP or SDP but can usually be obtained from the device codec. The field reflects the amount of voice content in each frame within the RTP payload, measured in milliseconds. Note that this value can be combined with the FramesPerPacket to determine the packetization rate. Also, where a sample-based codec is used, a "frame" refers to the set of samples carried in an RTP packet.
此参数不包含在RTP或SDP中，但通常可以从设备编解码器获得。该字段反映RTP有效负载内每个帧中的语音内容量，以毫秒为单位。请注意，此值可以与FramesPerPacket组合以确定打包速率。此外，在使用基于样本的编解码器的情况下，“帧”指RTP分组中携带的样本集。
This parameter is not contained in RTP or SDP but is usually provided by the device codec. The field provides the number of octets in each frame within the RTP payload. This field is usually not provided when the FrameDuration is provided. Also, where a sample-based codec is used, a "frame" refers to the set of samples carried in an RTP packet.
此参数不包含在RTP或SDP中，但通常由设备编解码器提供。该字段提供RTP有效负载内每个帧中的八位字节数。当提供FrameDuration时，通常不提供此字段。此外，在使用基于样本的编解码器的情况下，“帧”指RTP分组中携带的样本集。
This parameter is not contained in RTP or SDP but can usually be obtained from the device codec. This field provides the number of frames in each RTP packet. Note that this value can be combined with the FrameDuration to determine the packetization rate. Also, where a sample-based codec is used, a "frame" refers to the set of samples carried in an RTP packet.
此参数不包含在RTP或SDP中，但通常可以从设备编解码器获得。此字段提供每个RTP数据包中的帧数。请注意，此值可以与FrameDuration组合以确定打包速率。此外，在使用基于样本的编解码器的情况下，“帧”指RTP分组中携带的样本集。
This parameter is taken directly from the SDP attribute "fmtp" defined in RFC 4566.
此参数直接取自RFC 4566中定义的SDP属性“fmtp”。
This parameter does not correspond to SDP, RTP, or RTCP XR. It indicates whether silence suppression, also known as Voice Activity Detection (VAD), is enabled for the identified session.
此参数与SDP、RTP或RTCP XR不对应。它指示是否为已识别的会话启用静音抑制，也称为语音活动检测（VAD）。
This value corresponds to "PLC" in RFC 3611 in the VoIP Metrics Report Block. The values defined by RFC 3611 are reused by this recommendation and therefore no mapping is required.
该值对应于VoIP度量报告块中RFC 3611中的“PLC”。本建议重用RFC 3611定义的值，因此无需映射。
This field provides the IP address, port, and synchronization source (SSRC) for the session from the perspective of the endpoint that is measuring performance. The IPAddress MAY be in IPv4 or IPv6 format. The SSRC is taken from SDP, RTCP, or RTCP XR input parameters.
此字段从测量性能的端点的角度提供会话的IP地址、端口和同步源（SSRC）。IP地址可以是IPv4或IPv6格式。SSRC取自SDP、RTCP或RTCP XR输入参数。
In the presence of NAT and where a NAT-traversal mechanism such as Session Traversal Utilities for NAT (STUN) [16] is used, the external IP address can be reported, since the internal IP address is not visible to the network operator.
在存在NAT的情况下，如果使用NAT遍历机制，例如NAT会话遍历实用程序（STUN）[16]，则可以报告外部IP地址，因为网络运营商看不到内部IP地址。
This field provides the IP address, port, and SSRC of the session peer from the perspective of the remote endpoint measuring performance. In the presence of NAT and where a NAT-traversal mechanism such as STUN [16] is used, the external IP address can be reported, since the internal IP address is not visible to the network operator.
该字段从远程端点测量性能的角度提供会话对等方的IP地址、端口和SSRC。在存在NAT的情况下，如果使用了NAT穿越机制（如STUN[16]），则可以报告外部IP地址，因为网络运营商看不到内部IP地址。
This value corresponds to "JBA" in RFC 3611 in the VoIP Metrics Report Block. The values defined by RFC 3611 are unchanged and therefore no mapping is required.
该值对应于VoIP度量报告块中RFC 3611中的“JBA”。RFC 3611定义的值不变，因此不需要映射。
This value corresponds to "JB rate" in RFC 3611 in the VoIP Metrics Report Block. The parameter does not require any conversion.
该值对应于VoIP度量报告块中RFC 3611中的“JB速率”。该参数不需要任何转换。
This value corresponds to "JB nominal" in RFC 3611 in the VoIP Metrics Report Block. The parameter does not require any conversion.
该值对应于VoIP度量报告块中RFC 3611中的“JB NORMAL”。该参数不需要任何转换。
This value corresponds to "JB maximum" in RFC 3611 in the VoIP Metrics Report Block. The parameter does not require any conversion.
该值对应于VoIP度量报告块中RFC 3611中的“JB最大值”。该参数不需要任何转换。
This value corresponds to "JB abs max" in RFC 3611 in the VoIP Metrics Report Block. The parameter does not require any conversion.
该值对应于VoIP度量报告块中RFC 3611中的“JB abs max”。该参数不需要任何转换。
This value corresponds to "loss rate" in RFC 3611 in the VoIP Metrics Report Block. For conversion, see Section 4.6.2.1. A loss rate of 100% MAY be reported if media packets were expected but none had been received at the time of session termination.
该值对应于VoIP度量报告块中RFC 3611中的“丢失率”。有关转换，请参见第4.6.2.1节。如果预期会有媒体数据包，但在会话终止时未收到任何媒体数据包，则可能会报告100%的丢失率。
This value corresponds to "discard rate" in RFC 3611 in the VoIP Metrics Report Block. For conversion, see Section 4.6.2.1.
该值对应于VoIP度量报告块中RFC 3611中的“丢弃率”。有关转换，请参见第4.6.2.1节。
This value corresponds to "burst density" in RFC 3611 in the VoIP Metrics Report Block. For conversion, see Section 4.6.2.1.
该值对应于VoIP度量报告块中RFC 3611中的“突发密度”。有关转换，请参见第4.6.2.1节。
This value corresponds to "burst duration" in RFC 3611 in the VoIP Metrics Report Block. This value requires no conversion; the exact value sent in an RTCP XR VoIP Metrics Report Block can be included in the SIP vq-rtcpxr parameter.
该值对应于VoIP度量报告块中RFC 3611中的“突发持续时间”。此值不需要转换；在RTCP XR VoIP Metrics报告块中发送的确切值可以包含在SIP vq rtcpxr参数中。
This value corresponds to "gap density" in RFC 3611 in the VoIP metrics Report Block.
该值对应于VoIP度量报告块中RFC 3611中的“间隙密度”。
This value corresponds to "gap duration" in RFC 3611 in the VoIP Metrics Report Block. This value requires no conversion; the exact value sent in an RTCP XR VoIP Metrics Report Block can be reported.
该值对应于VoIP度量报告块中RFC 3611中的“间隙持续时间”。此值不需要转换；可以报告RTCP XR VoIP Metrics报告块中发送的确切值。
This value corresponds to "Gmin" in RFC 3611 in the VoIP Metrics Report Block. This value requires no conversion; the exact value sent in an RTCP XR VoIP Metrics Report Block can be reported.
该值对应于VoIP度量报告块中RFC 3611中的“Gmin”。此值不需要转换；可以报告RTCP XR VoIP Metrics报告块中发送的确切值。
This value corresponds to "round trip delay" in RFC 3611 in the VoIP Metrics Report Block and may be measured using the method defined in RFC 3550. The parameter is expressed in milliseconds.
该值对应于VoIP度量报告块中RFC 3611中的“往返延迟”，可以使用RFC 3550中定义的方法进行测量。参数以毫秒表示。
This value corresponds to "end system delay" in RFC 3611 in the VoIP Metrics Report Block. This parameter does not require any conversion. The parameter is expressed in milliseconds.
该值对应于VoIP度量报告块中RFC 3611中的“结束系统延迟”。此参数不需要任何转换。参数以毫秒表示。
This value is computed by adding Round-Trip Delay to the local and remote End System Delay and dividing by two.
该值通过将往返延迟加上本地和远程端系统延迟并除以2来计算。
This value SHOULD be measured using the methods defined in IETF RFC 2679 [12]. The parameter is expressed in milliseconds.
应使用IETF RFC 2679[12]中定义的方法测量该值。参数以毫秒表示。
Inter-arrival jitter is calculated as defined in RFC 3550 and converted into milliseconds.
到达间抖动按照RFC 3550中的定义进行计算，并转换为毫秒。
It is recommended that MAJ be measured as defined in ITU-T G.1020 [9]. This parameter is often referred to as MAPDV (Mean Absolute Packet Delay Variation). The parameter is expressed in milliseconds.
建议按照ITU-T G.1020[9]中的定义测量MAJ。此参数通常称为MAPDV（平均绝对数据包延迟变化）。参数以毫秒表示。
This field corresponds to "signal level" in RFC 3611 in the VoIP Metrics Report Block. This field provides the voice signal relative level is defined as the ratio of the signal level to a 0 dBm0 reference, expressed in decibels. This value can be used directly without extra conversion.
此字段对应于VoIP度量报告块中RFC 3611中的“信号电平”。此字段提供语音信号相对电平，其定义为信号电平与0 dBm0参考的比率，以分贝表示。此值可直接使用，无需额外转换。
This field corresponds to "noise level" in RFC 3611 in the VoIP Metrics Report Block. This field provides the ratio of the silent period background noise level to a 0 dBm0 reference, expressed in decibels. This value can be used directly without extra conversion.
此字段对应于VoIP度量报告块中RFC 3611中的“噪声级”。此字段提供静默期背景噪声水平与0 dBm0参考的比率，以分贝表示。此值可直接使用，无需额外转换。
This field corresponds to "RERL" in RFC 3611 in the VoIP Metrics Report Block. This field provides the ratio between the original signal and the echo level in decibels, as measured after echo cancellation or suppression has been applied. This value can be used directly without extra conversion.
此字段对应于VoIP度量报告块中RFC 3611中的“RERL”。此字段提供原始信号与回声电平之间的比率（分贝），如在应用回声消除或抑制后测量的那样。此值可直接使用，无需额外转换。
This field reports the listening quality expressed as an R factor (per G.107). This does not include the effects of echo or delay. The range of R is 0-95 for narrowband calls and 0-120 for wideband calls. Algorithms for computing this value SHOULD be compliant with ITU-T Recommendations P.564 [10] and G.107 [11].
该字段报告以R系数表示的收听质量（根据G.107）。这不包括回声或延迟的影响。对于窄带呼叫，R的范围为0-95，对于宽带呼叫，R的范围为0-120。计算该值的算法应符合ITU-T建议P.564[10]和G.107[11]。
This field provides a text name for the algorithm used to estimate ListeningQualityR. This field will be free form text and not necessarily reflective of any standards or recommendations.
此字段为用于估计ListingQualityr的算法提供文本名称。该字段为自由格式文本，不一定反映任何标准或建议。
This field corresponds to "R factor" in RFC 3611 in the VoIP Metrics Report Block. This parameter provides a cumulative measurement of voice quality from the start of the session to the reporting time. The range of R is 0-95 for narrowband calls and 0-120 for wideband calls. Algorithms for computing this value SHOULD be compliant with ITU-T Recommendations P.564 and G.107. Within RFC 3611, a reported R factor of 127 indicates that this parameter is unavailable; in this case, the ConversationalQualityR parameter MUST be omitted from the vq-rtcpxr event.
此字段对应于VoIP度量报告块中RFC 3611中的“R因子”。此参数提供从会话开始到报告时间的语音质量累积测量值。对于窄带呼叫，R的范围为0-95，对于宽带呼叫，R的范围为0-120。计算该值的算法应符合ITU-T建议P.564和G.107。在RFC 3611中，报告的R系数127表示该参数不可用；在这种情况下，vq rtcpxr事件中必须忽略ConversationalQualityR参数。
This field provides a text name for the algorithm used to estimate ConversationalQualityR. This field will be free form text and not necessarily reflective of any standards or recommendations.
此字段为用于估计ConversationalQualityR的算法提供文本名称。该字段为自由格式文本，不一定反映任何标准或建议。
This field corresponds to "ext. R factor" in RFC 3611 in the VoIP Metrics Report Block. This parameter reflects voice quality as measured by the local endpoint for incoming connection on "other" side (refer to RFC 3611 for a more-detailed explanation). The range
此字段对应于VoIP度量报告块中RFC 3611中的“ext.R factor”。此参数反映了本地端点在“另一方”的传入连接上测量的语音质量（有关更详细的说明，请参阅RFC 3611）。靶场
of R is 0-95 for narrowband calls and 0-120 for wideband calls. Algorithms for computing this value SHOULD be compliant with ITU-T Recommendations P.564 and G.107. Within RFC 3611, a reported R factor of 127 indicates that this parameter is unavailable; in this case, the ConversationalQualityR parameter MUST be omitted from the vq-rtcpxr event.
对于窄带呼叫，R的值为0-95，对于宽带呼叫，R的值为0-120。计算该值的算法应符合ITU-T建议P.564和G.107。在RFC 3611中，报告的R系数127表示该参数不可用；在这种情况下，vq rtcpxr事件中必须忽略ConversationalQualityR参数。
This field provides a text name for the algorithm used to estimate ExternalR-In. This field will be free-form text and not necessarily reflective of any standards or recommendations.
此字段为用于在中估计ExternalR的算法提供文本名称。该字段为自由格式文本，不一定反映任何标准或建议。
This field corresponds to "ext. R factor" in RFC 3611 in the VoIP Metrics Report Block. Here, the value is copied from RTCP XR message received from the remote endpoint on the "other" side of this endpoint; refer to RFC 3611 for a more detailed explanation). The range of R is 0-95 for narrowband calls and 0-120 for wideband calls. Algorithms for computing this value SHOULD be compliant with ITU-T Recommendations P.564 and G.107. Within RFC 3611, a reported R factor of 127 indicates that this parameter is unavailable; in this case, the ConversationalQualityR parameter SHALL be omitted from the vq-rtcpxr event.
此字段对应于VoIP度量报告块中RFC 3611中的“ext.R factor”。此处，从该端点“另一端”的远程端点接收的RTCP XR消息复制该值；有关更详细的说明，请参阅RFC 3611）。对于窄带呼叫，R的范围为0-95，对于宽带呼叫，R的范围为0-120。计算该值的算法应符合ITU-T建议P.564和G.107。在RFC 3611中，报告的R系数127表示该参数不可用；在这种情况下，vq rtcpxr事件中应忽略ConversationalQualityR参数。
This field provides a text name for the algorithm used to estimate ExternalR-Out. This field will be free-form text and not necessarily reflective of any standards or recommendations.
此字段为用于估计外部输出的算法提供文本名称。该字段为自由格式文本，不一定反映任何标准或建议。
Conversion of RFC 3611 reported mean opinion scores (MOSs) for use in reporting MOS-LQ and MOS-CQ MUST be performed by dividing the RFC 3611 reported value by 10 if this value is less than or equal to 50 or omitting the MOS-xQ parameter if the RFC 3611 reported value is 127 (which indicates unavailable).
如果RFC 3611报告值小于或等于50，则必须将RFC 3611报告值除以10，或者如果RFC 3611报告值为127（表示不可用），则忽略MOS xQ参数，以转换用于报告MOS-LQ和MOS-CQ的RFC 3611报告平均意见分数（MOS）。
This field corresponds to "MOSLQ" in RFC 3611 in the VoIP Metrics Report Block. This parameter is the estimated mean opinion score for listening voice quality on a scale from 1 to 5, in which 5 represents "Excellent" and 1 represents "Unacceptable". Algorithms for
此字段对应于VoIP度量报告块中RFC 3611中的“MOSLQ”。该参数是在1到5的范围内对收听语音质量的估计平均意见分数，其中5表示“优秀”，1表示“不可接受”。算法
computing this value SHOULD be compliant with ITU-T Recommendation P.564 [10]. This field provides a text name for the algorithm used to estimate MOS-LQ.
计算该值应符合ITU-T建议P.564[10]。此字段为用于估计MOS-LQ的算法提供文本名称。
This field corresponds to "MOSCQ" in RFC 3611 in the VoIP Metrics Report Block. This parameter is the estimated mean opinion score for conversation voice quality on a scale from 1 to 5, in which 5 represents excellent and 1 represents unacceptable. Algorithms for computing this value SHOULD be compliant with ITU-T Recommendation P.564 with regard to the listening quality element of the computed MOS score.
此字段对应于VoIP度量报告块中RFC 3611中的“MOSCQ”。此参数是会话语音质量的估计平均意见分数，评分范围为1到5，其中5表示优秀，1表示不可接受。计算该值的算法应符合ITU-T建议P.564关于计算MOS分数的收听质量元素的规定。
This field provides a text name for the algorithm used to estimate MOS-CQ. This field will be free-form text and not necessarily reflective of any standards or recommendations.
此字段为用于估计MOS-CQ的算法提供文本名称。该字段为自由格式文本，不一定反映任何标准或建议。
This field provides a text description of the algorithm used to estimate all voice quality metrics. This parameter is provided as an alternative to the separate estimation algorithms for use when the same algorithm is used for all measurements. This field will be free-form text and not necessarily reflective of any standards or recommendations.
此字段提供用于估计所有语音质量指标的算法的文本说明。当对所有测量使用相同的算法时，提供该参数作为单独估计算法的替代方案。该字段为自由格式文本，不一定反映任何标准或建议。
This section shows a number of message flow examples showing how the event package works.
本节展示了一些消息流示例，展示了事件包的工作方式。
Alice Proxy/Registrar Collector Bob
| | | |
| | | |
| REGISTER Allow-Event:vq-rtcpxr F1 | |
|------------------->| | |
| 200 OK F2 | | |
|<-------------------| | |
| | SUBSCRIBE Event:vq-rtcpxr F3 |
| |<-------------------| |
| SUBSCRIBE Event:vq-rtcpxr F4 | |
|<-------------------| | |
| 200 OK F5 | | |
|------------------->| | |
| | 200 OK F6 | |
| |------------------->| |
| INVITE F7 | | |
|------------------->| | |
| | INVITE F8 | |
| |---------------------------------------->|
| | 200 OK F9 | |
| |<----------------------------------------|
| 200 OK F10 | | |
|<-------------------| | |
| ACK F11 | | |
|------------------->| | |
| | ACK F12 | |
| |---------------------------------------->|
| RTP | | |
|<============================================================>|
| RTCP, RTCP XR | |
|<============================================================>|
| | | |
| BYE F13 | | |
|------------------->| BYE F14 | |
| |---------------------------------------->|
| | 200 OK F15 | |
| |<----------------------------------------|
| 200 OK F16 | | |
|<-------------------| | |
| NOTIFY Event:vq-rtcpxr F17 | |
|------------------->| | |
| | NOTIFY Event:vq-rtcpxr F18 |
| |------------------->| |
| | 200 OK F19 | |
| |<-------------------| |
| 200 OK F20 | | |
|<-------------------| | |
Alice Proxy/Registrar Collector Bob
| | | |
| | | |
| REGISTER Allow-Event:vq-rtcpxr F1 | |
|------------------->| | |
| 200 OK F2 | | |
|<-------------------| | |
| | SUBSCRIBE Event:vq-rtcpxr F3 |
| |<-------------------| |
| SUBSCRIBE Event:vq-rtcpxr F4 | |
|<-------------------| | |
| 200 OK F5 | | |
|------------------->| | |
| | 200 OK F6 | |
| |------------------->| |
| INVITE F7 | | |
|------------------->| | |
| | INVITE F8 | |
| |---------------------------------------->|
| | 200 OK F9 | |
| |<----------------------------------------|
| 200 OK F10 | | |
|<-------------------| | |
| ACK F11 | | |
|------------------->| | |
| | ACK F12 | |
| |---------------------------------------->|
| RTP | | |
|<============================================================>|
| RTCP, RTCP XR | |
|<============================================================>|
| | | |
| BYE F13 | | |
|------------------->| BYE F14 | |
| |---------------------------------------->|
| | 200 OK F15 | |
| |<----------------------------------------|
| 200 OK F16 | | |
|<-------------------| | |
| NOTIFY Event:vq-rtcpxr F17 | |
|------------------->| | |
| | NOTIFY Event:vq-rtcpxr F18 |
| |------------------->| |
| | 200 OK F19 | |
| |<-------------------| |
| 200 OK F20 | | |
|<-------------------| | |
Figure 1. Summary report with NOTIFY sent after session termination. In the call flow depicted in Figure 1, the following message format is sent in F17:
图1。会话终止后发送带有通知的摘要报告。在图1所示的调用流中，以下消息格式在F17中发送：
NOTIFY sip:collector@example.org SIP/2.0 Via: SIP/2.0/UDP pc22.example.org;branch=z9hG4bK3343d7 Max-Forwards: 70 To: <sip:collector@example.org>;tag=43524545 From: Alice <sip:alice@example.org>;tag=a3343df32 Call-ID: 1890463548 CSeq: 4321 NOTIFY Contact: <sip:alice@pc22.example.org> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Event: vq-rtcpxr Accept: application/sdp, message/sipfrag Subscription-State: active;expires=3600 Content-Type: application/vq-rtcpxr Content-Length: ...
通知sip：collector@example.orgSIP/2.0 Via:SIP/2.0/UDP pc22.example.org；分支=Z9HG4BK343D7最大转发：70到：<sip:collector@example.org>;tag=43524545 From:Alice<sip:alice@example.org>;tag=a3343df32呼叫ID:1890463548 CSeq:4321通知联系人：<sip:alice@pc22.example.org>允许：邀请、确认、取消、选项、再见、引用、订阅、通知事件：vq rtcpxr接受：应用程序/sdp、消息/sipfrag订阅状态：活动；expires=3600内容类型：应用程序/vq rtcpxr内容长度：。。。
VQSessionReport: CallTerm
CallID: 6dg37f1890463
LocalID: Alice <sip:alice@example.org>
RemoteID: Bill <sip:bill@example.net>
OrigID: Alice <sip:alice@example.org>
LocalGroup: example-phone-55671
RemoteGroup: example-gateway-09871
LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d
LocalMAC: 00:1f:5b:cc:21:0f
RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd
RemoteMAC: 00:26:08:8e:95:02
LocalMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-18 NL=-50 RERL=55
QualityEst:RLQ=88 RCQ=85 EXTRI=90 MOSLQ=4.1 MOSCQ=4.0
QoEEstAlg=P.564
RemoteMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
VQSessionReport: CallTerm
CallID: 6dg37f1890463
LocalID: Alice <sip:alice@example.org>
RemoteID: Bill <sip:bill@example.net>
OrigID: Alice <sip:alice@example.org>
LocalGroup: example-phone-55671
RemoteGroup: example-gateway-09871
LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d
LocalMAC: 00:1f:5b:cc:21:0f
RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd
RemoteMAC: 00:26:08:8e:95:02
LocalMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-18 NL=-50 RERL=55
QualityEst:RLQ=88 RCQ=85 EXTRI=90 MOSLQ=4.1 MOSCQ=4.0
QoEEstAlg=P.564
RemoteMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-21 NL=-45 RERL=55
QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=4.3 MOSCQ=4.2
QoEEstAlg=P.564
DialogID:1890463548@alice.example.org;to-tag=8472761;
from-tag=9123dh311
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-21 NL=-45 RERL=55
QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=4.3 MOSCQ=4.2
QoEEstAlg=P.564
DialogID:1890463548@alice.example.org;to-tag=8472761;
from-tag=9123dh311
Alice Proxy/Registrar Collector Bob
| | | |
| | | |
| REGISTER Allow-Event:vq-rtcpxr F1 | |
|------------------->| | |
| 200 OK F2 | | |
|<-------------------| | |
| | SUBSCRIBE Event:vq-rtcpxr F3 |
| |<-------------------| |
| SUBSCRIBE Event:vq-rtcpxr F4 | |
|<-------------------| | |
| 200 OK F5 | | |
|------------------->| | |
| | 200 OK F6 | |
| |------------------->| |
| INVITE F7 | | |
|------------------->| | |
| | INVITE F8 | |
| |---------------------------------------->|
| | 200 OK F9 | |
| |<----------------------------------------|
| 200 OK F10 | | |
|<-------------------| | |
| ACK F11 | | |
|------------------->| | |
| | ACK F12 | |
| |---------------------------------------->|
| RTP | | |
|<============================================================>|
| RTCP, RTCP XR | |
|<============================================================>|
| NOTIFY Event:vq-rtcpxr F13 | |
|------------------->| | |
| | NOTIFY Event:vq-rtcpxr F14 |
| |------------------->| |
| | 200 OK F15 | |
| |<-------------------| |
| 200 OK F16 | | |
Alice Proxy/Registrar Collector Bob
| | | |
| | | |
| REGISTER Allow-Event:vq-rtcpxr F1 | |
|------------------->| | |
| 200 OK F2 | | |
|<-------------------| | |
| | SUBSCRIBE Event:vq-rtcpxr F3 |
| |<-------------------| |
| SUBSCRIBE Event:vq-rtcpxr F4 | |
|<-------------------| | |
| 200 OK F5 | | |
|------------------->| | |
| | 200 OK F6 | |
| |------------------->| |
| INVITE F7 | | |
|------------------->| | |
| | INVITE F8 | |
| |---------------------------------------->|
| | 200 OK F9 | |
| |<----------------------------------------|
| 200 OK F10 | | |
|<-------------------| | |
| ACK F11 | | |
|------------------->| | |
| | ACK F12 | |
| |---------------------------------------->|
| RTP | | |
|<============================================================>|
| RTCP, RTCP XR | |
|<============================================================>|
| NOTIFY Event:vq-rtcpxr F13 | |
|------------------->| | |
| | NOTIFY Event:vq-rtcpxr F14 |
| |------------------->| |
| | 200 OK F15 | |
| |<-------------------| |
| 200 OK F16 | | |
|<-------------------| | |
| | | |
| BYE F17 | | |
|------------------->| BYE F18 | |
| |---------------------------------------->|
| | 200 OK F19 | |
| |<----------------------------------------|
| 200 OK F20 | | |
|<-------------------| | |
| NOTIFY Event:vq-rtcpxr F21 | |
|------------------->| | |
| | NOTIFY Event:vq-rtcpxr F22 |
| |------------------->| |
| | 200 OK F23 | |
| |<-------------------| |
| 200 OK F24 | | |
|<-------------------| | |
|<-------------------| | |
| | | |
| BYE F17 | | |
|------------------->| BYE F18 | |
| |---------------------------------------->|
| | 200 OK F19 | |
| |<----------------------------------------|
| 200 OK F20 | | |
|<-------------------| | |
| NOTIFY Event:vq-rtcpxr F21 | |
|------------------->| | |
| | NOTIFY Event:vq-rtcpxr F22 |
| |------------------->| |
| | 200 OK F23 | |
| |<-------------------| |
| 200 OK F24 | | |
|<-------------------| | |
Figure 2. An alert report is sent during the session. In the call flow depicted in Figure 2, the following message format is sent in F13:
图2。会话期间会发送警报报告。在图2所示的调用流中，以下消息格式在F13中发送：
NOTIFY sip:collector@example.org SIP/2.0 Via: SIP/2.0/UDP pc22.example.org;branch=z9hG4bK3343d7 Max-Forwards: 70 To: <sip:proxy@example.org> From: Alice <sip:alice@example.org>;tag=a3343df32 Call-ID: 1890463548 CSeq: 4331 PUBLISH Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Event: vq-rtcpxr Accept: application/sdp, message/sipfrag Content-Type: application/vq-rtcpxr Content-Length: ...
通知sip：collector@example.orgSIP/2.0 Via:SIP/2.0/UDP pc22.example.org；分支=Z9HG4BK343D7最大转发：70到：<sip:proxy@example.org>发件人：Alice<sip:alice@example.org>;tag=a3343df32呼叫ID:1890463548 CSeq:4331发布允许：邀请、确认、取消、选项、再见、引用、订阅、通知事件：vq rtcpxr接受：应用程序/sdp，消息/sipfrag内容类型：应用程序/vq rtcpxr内容长度：。。。
VQAlertReport: Type=NLR Severity=Critical Dir=local
CallID: 6dg37f1890463
LocalID: Alice <sip:alice@example.org>
RemoteID: Bill <sip:bill@example.org>
OrigID: Alice <sip:alice@example.org>
LocalGroup: example-phone-55671
RemoteGroup: example-gateway-09871
LocalAddr:IP=10.10.1.100 PORT=5000 SSRC=0x2468abcd
LocalMAC: 00:1f:5b:cc:21:0f
VQAlertReport: Type=NLR Severity=Critical Dir=local
CallID: 6dg37f1890463
LocalID: Alice <sip:alice@example.org>
RemoteID: Bill <sip:bill@example.org>
OrigID: Alice <sip:alice@example.org>
LocalGroup: example-phone-55671
RemoteGroup: example-gateway-09871
LocalAddr:IP=10.10.1.100 PORT=5000 SSRC=0x2468abcd
LocalMAC: 00:1f:5b:cc:21:0f
RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=1357efff
RemoteMAC: 00:26:08:8e:95:02
LocalMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
FMTP="annexb=no" PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=10.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-21 NL=-50 RERL=55
QualityEst:RLQ=80 RCQ=85 EXTRI=90 MOSLQ=3.5 MOSCQ=3.7
QoEEstAlg=P.564
RemoteMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
FMTP="annexb=no" PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-21 NL=-45 RERL=55
QualityEst:RLQ=90 RCQ=85 MOSLQ=4.3 MOSCQ=4.2 QoEEstAlg=P.564
DialogID:1890463548@alice.example.org;to-tag=8472761;
from-tag=9123dh311
RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=1357efff
RemoteMAC: 00:26:08:8e:95:02
LocalMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
FMTP="annexb=no" PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=10.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-21 NL=-50 RERL=55
QualityEst:RLQ=80 RCQ=85 EXTRI=90 MOSLQ=3.5 MOSCQ=3.7
QoEEstAlg=P.564
RemoteMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
FMTP="annexb=no" PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-21 NL=-45 RERL=55
QualityEst:RLQ=90 RCQ=85 MOSLQ=4.3 MOSCQ=4.2 QoEEstAlg=P.564
DialogID:1890463548@alice.example.org;to-tag=8472761;
from-tag=9123dh311
Alice Proxy/Registrar Collector Bob
| | | |
| | | |
| REGISTER Allow-Event:vq-rtcpxr F1 | |
|------------------->| | |
| 200 OK F2 | | |
|<-------------------| | |
| INVITE F3 | | |
|------------------->| | |
| | INVITE F4 | |
| |---------------------------------------->|
| | 200 OK F5 | |
| |<----------------------------------------|
| 200 OK F6 | | |
|<-------------------| | |
| ACK F7 | | |
|------------------->| | |
| | ACK F8 | |
| |---------------------------------------->|
| RTP | | |
|<============================================================>|
| RTCP | | |
|<============================================================>|
| | | |
| BYE F9 | | |
|------------------->| BYE F10 | |
| |---------------------------------------->|
| | 200 OK F11 | |
| |<----------------------------------------|
| 200 OK F12 | | |
|<-------------------| | |
| PUBLISH Event:vq-rtcpxr F13 | |
|------------------->| | |
| | PUBLISH Event:vq-rtcpxr F14 |
| |------------------->| |
| | 200 OK F15 | |
| |<-------------------| |
| 200 OK F16 | | |
|<-------------------| | |
Alice Proxy/Registrar Collector Bob
| | | |
| | | |
| REGISTER Allow-Event:vq-rtcpxr F1 | |
|------------------->| | |
| 200 OK F2 | | |
|<-------------------| | |
| INVITE F3 | | |
|------------------->| | |
| | INVITE F4 | |
| |---------------------------------------->|
| | 200 OK F5 | |
| |<----------------------------------------|
| 200 OK F6 | | |
|<-------------------| | |
| ACK F7 | | |
|------------------->| | |
| | ACK F8 | |
| |---------------------------------------->|
| RTP | | |
|<============================================================>|
| RTCP | | |
|<============================================================>|
| | | |
| BYE F9 | | |
|------------------->| BYE F10 | |
| |---------------------------------------->|
| | 200 OK F11 | |
| |<----------------------------------------|
| 200 OK F12 | | |
|<-------------------| | |
| PUBLISH Event:vq-rtcpxr F13 | |
|------------------->| | |
| | PUBLISH Event:vq-rtcpxr F14 |
| |------------------->| |
| | 200 OK F15 | |
| |<-------------------| |
| 200 OK F16 | | |
|<-------------------| | |
Figure 3. End of session report sent after session termination. In the message flow depicted in Figure 3, the following message is sent in F13.
图3。会话终止后发送的会话结束报告。在图3所示的消息流中，以下消息在F13中发送。
PUBLISH sip:collector@example.org SIP/2.0 Via: SIP/2.0/UDP pc22.example.org;branch=z9hG4bK3343d7 Max-Forwards: 70 To: <sip:proxy@example.org> From: Alice <sip:alice@example.org>;tag=a3343df32 Call-ID: 1890463548 CSeq: 4331 PUBLISH Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Event: vq-rtcpxr Accept: application/sdp, message/sipfrag Content-Type: application/vq-rtcpxr Content-Length: ...
发布sip：collector@example.orgSIP/2.0 Via:SIP/2.0/UDP pc22.example.org；分支=Z9HG4BK343D7最大转发：70到：<sip:proxy@example.org>发件人：Alice<sip:alice@example.org>;tag=a3343df32呼叫ID:1890463548 CSeq:4331发布允许：邀请、确认、取消、选项、再见、引用、订阅、通知事件：vq rtcpxr接受：应用程序/sdp，消息/sipfrag内容类型：应用程序/vq rtcpxr内容长度：。。。
VQSessionReport: CallTerm
CallID: 6dg37f1890463
LocalID: Alice <sip:alice@example.org>
RemoteID: Bill <sip:bill@example.net>
OrigID: Alice <sip:alice@example.org>
LocalGroup: example-phone-55671
RemoteGroup: example-gateway-09871
LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d
LocalMAC: 00:1f:5b:cc:21:0f
RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd
RemoteMAC: 00:26:08:8e:95:02
LocalMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
FMTP="annexb=no" PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-21 NL=-50 RERL=55
QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=4.2 MOSCQ=4.3
QoEEstAlg=P.564
RemoteMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
FMTP="annexb=no" PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-21 NL=-45 RERL=55
QualityEst:RLQ=90 RCQ=85 MOSLQ=4.3 MOSCQ=4.2 QoEEstAlg=P.564
DialogID:1890463548@alice.example.org;to-tag=8472761;
from-tag=9123dh311
VQSessionReport: CallTerm
CallID: 6dg37f1890463
LocalID: Alice <sip:alice@example.org>
RemoteID: Bill <sip:bill@example.net>
OrigID: Alice <sip:alice@example.org>
LocalGroup: example-phone-55671
RemoteGroup: example-gateway-09871
LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d
LocalMAC: 00:1f:5b:cc:21:0f
RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd
RemoteMAC: 00:26:08:8e:95:02
LocalMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
FMTP="annexb=no" PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-21 NL=-50 RERL=55
QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=4.2 MOSCQ=4.3
QoEEstAlg=P.564
RemoteMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
FMTP="annexb=no" PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-21 NL=-45 RERL=55
QualityEst:RLQ=90 RCQ=85 MOSLQ=4.3 MOSCQ=4.2 QoEEstAlg=P.564
DialogID:1890463548@alice.example.org;to-tag=8472761;
from-tag=9123dh311
Alice Proxy/Registrar Collector Bob
| | | |
| INVITE F1 | | |
|------------------->| | |
| | INVITE F2 | |
| |---------------------------------------->|
| | 200 OK F3 | |
| |<----------------------------------------|
| 200 OK F4 | | |
|<-------------------| | |
| ACK F5 | | |
|------------------->| | |
| | ACK F6 | |
| |---------------------------------------->|
| RTP | | |
|<============================================================>|
| RTCP | | |
|<============================================================>|
| PUBLISH Event:vq-rtcpxr F7 | |
|------------------->| | |
| | PUBLISH Event:vq-rtcpxr F8 |
| |------------------->| |
| | 200 OK F9 | |
| |<-------------------| |
| 200 OK F10 | | |
|<-------------------| | |
| | | |
| BYE F11 | | |
|------------------->| BYE F12 | |
| |---------------------------------------->|
| | 200 OK F13 | |
| |<----------------------------------------|
| 200 OK F14 | | |
|<-------------------| | |
Alice Proxy/Registrar Collector Bob
| | | |
| INVITE F1 | | |
|------------------->| | |
| | INVITE F2 | |
| |---------------------------------------->|
| | 200 OK F3 | |
| |<----------------------------------------|
| 200 OK F4 | | |
|<-------------------| | |
| ACK F5 | | |
|------------------->| | |
| | ACK F6 | |
| |---------------------------------------->|
| RTP | | |
|<============================================================>|
| RTCP | | |
|<============================================================>|
| PUBLISH Event:vq-rtcpxr F7 | |
|------------------->| | |
| | PUBLISH Event:vq-rtcpxr F8 |
| |------------------->| |
| | 200 OK F9 | |
| |<-------------------| |
| 200 OK F10 | | |
|<-------------------| | |
| | | |
| BYE F11 | | |
|------------------->| BYE F12 | |
| |---------------------------------------->|
| | 200 OK F13 | |
| |<----------------------------------------|
| 200 OK F14 | | |
|<-------------------| | |
Figure 4. Alert report message flow
图4。警报报告消息流
In the message flow depicted in Figure 4, the following message is sent in F7:
在图4所示的消息流中，以下消息在F7中发送：
PUBLISH sip:collector@example.org SIP/2.0
Via: SIP/2.0/UDP pc22.example.org;branch=z9hG4bK3343d7
Max-Forwards: 70
To: <sip:collector@example.org>
From: Alice <sip:alice@example.org>;tag=a3343df32
Call-ID: 1890463548
PUBLISH sip:collector@example.org SIP/2.0
Via: SIP/2.0/UDP pc22.example.org;branch=z9hG4bK3343d7
Max-Forwards: 70
To: <sip:collector@example.org>
From: Alice <sip:alice@example.org>;tag=a3343df32
Call-ID: 1890463548
CSeq: 4321 PUBLISH Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Event: vq-rtcpxr Accept: application/sdp, message/sipfrag Content-Type: application/vq-rtcpxr Content-Length: ...
CSeq:4321发布允许：邀请、确认、取消、选项、再见、引用、订阅、通知事件：vq rtcpxr接受：应用程序/sdp、消息/sipfrag内容类型：应用程序/vq rtcpxr内容长度：。。。
VQAlertReport: Type=RLQ Severity=Warning Dir=local
CallID: 6dg37f1890463
LocalID: Alice <sip:alice@example.org>
RemoteID: Bill <sip:bill@example.org>
OrigID: Alice <sip:alice@example.org>
LocalGroup: example-phone-55671
RemoteGroup: example-gateway-09871
LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d
LocalMAC: 00:1f:5b:cc:21:0f
RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd
RemoteMAC: 00:26:08:8e:95:02
Metrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-12 NL=-30 RERL=55
QualityEst:RLQ=60 RCQ=55 EXTR=90 MOSLQ=2.4 MOSCQ=2.3
QoEEstAlg=P.564
RemoteMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-23 NL=-60 RERL=55
QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=4.2 MOSCQ=4.3
QoEEstAlg=P.564
DialogID:1890463548@alice.example.org;to-tag=8472761;
from-tag=9123dh3111
VQAlertReport: Type=RLQ Severity=Warning Dir=local
CallID: 6dg37f1890463
LocalID: Alice <sip:alice@example.org>
RemoteID: Bill <sip:bill@example.org>
OrigID: Alice <sip:alice@example.org>
LocalGroup: example-phone-55671
RemoteGroup: example-gateway-09871
LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d
LocalMAC: 00:1f:5b:cc:21:0f
RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd
RemoteMAC: 00:26:08:8e:95:02
Metrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-12 NL=-30 RERL=55
QualityEst:RLQ=60 RCQ=55 EXTR=90 MOSLQ=2.4 MOSCQ=2.3
QoEEstAlg=P.564
RemoteMetrics:
Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
PLC=3 SSUP=on
JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
PacketLoss:NLR=5.0 JDR=2.0
BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
Signal:SL=-23 NL=-60 RERL=55
QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=4.2 MOSCQ=4.3
QoEEstAlg=P.564
DialogID:1890463548@alice.example.org;to-tag=8472761;
from-tag=9123dh3111
It is the suggestion of the authors that the SIP configuration framework [15] be used to establish the necessary parameters for usage of vq-rtcpxr events. A dataset for this purpose should be designed and documented in a separate document upon completion of the framework.
作者建议使用SIP配置框架[15]为vq rtcpxr事件的使用建立必要的参数。在框架完成后，应设计用于此目的的数据集，并将其记录在单独的文件中。
This document registers a new SIP Event Package and a new media type.
本文档注册了一个新的SIP事件包和一个新的媒体类型。
Package name: vq-rtcpxr
Type: package
Contact: Amy Pendleton <aspen@telchemy.com>
Published Specification: This document
Package name: vq-rtcpxr
Type: package
Contact: Amy Pendleton <aspen@telchemy.com>
Published Specification: This document
Type name: application Subtype name: vq-rtcpxr Required parameters: none Optional parameters: none Encoding considerations: 7 bit Security considerations: See next section. Interoperability considerations: none. Published specification: This document.
类型名称：应用程序子类型名称：vq rtcpxr必需参数：无可选参数：无编码注意事项：7位安全注意事项：请参阅下一节。互操作性考虑：无。已发布规范：本文件。
Applications that use this media type: This document type is being used in notifications of VoIP quality reports.
使用此媒体类型的应用程序：此文档类型用于VoIP质量报告的通知。
Additional Information:
其他信息：
Magic Number: None File Extension: None Macintosh file type code: "TEXT"
幻数：无文件扩展名：无Macintosh文件类型代码：“文本”
Person and email address for further information: Amy Pendleton
<aspen@telchemy.com>
Person and email address for further information: Amy Pendleton
<aspen@telchemy.com>
Intended usage: COMMON
预期用途：普通
Author / Change controller: The IETF.
作者/变更控制者：IETF。
RTCP reports can contain sensitive information since they can provide information about the nature and duration of a session established between two or more endpoints. As a result, any third party wishing to obtain this information SHOULD be properly authenticated by the SIP UA using standard SIP mechanisms and according to the recommendations in [5]. Additionally, the event content MAY be encrypted to ensure confidentiality; the mechanisms for providing confidentiality are detailed in [2].
RTCP报告可以包含敏感信息，因为它们可以提供有关在两个或多个端点之间建立的会话的性质和持续时间的信息。因此，希望获得该信息的任何第三方都应通过SIP UA使用标准SIP机制并根据[5]中的建议进行适当认证。此外，可对事件内容进行加密以确保机密性；提供保密性的机制详见[2]。
The authors would like to thank Rajesh Kumar, Dave Oran, Tom Redman, Shane Holthaus, and Jack Ford for their comments and input.
作者要感谢Rajesh Kumar、Dave Oran、Tom Redman、Shane Holthaus和Jack Ford的评论和意见。
[1] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997.
[1] Bradner，S.，“RFC中用于表示需求水平的关键词”，BCP 14，RFC 2119，1997年3月。
[2] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A., Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP: Session Initiation Protocol", RFC 3261, June 2002.
[2] 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 3261，2002年6月。
[3] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003.
[3] 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。
[4] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol Extended Reports (RTCP XR)", RFC 3611, November 2003.
[4] Friedman，T.，Caceres，R.，和A.Clark，“RTP控制协议扩展报告（RTCP XR）”，RFC 36112003年11月。
[5] Roach, A., "Session Initiation Protocol (SIP)-Specific Event Notification", RFC 3265, June 2002.
[5] Roach，A.，“会话启动协议（SIP）-特定事件通知”，RFC3265，2002年6月。
[6] Crocker, D. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, January 2008.
[6] Crocker，D.和P.Overell，“语法规范的扩充BNF:ABNF”，STD 68，RFC 5234，2008年1月。
[7] Klyne, G., Ed. and C. Newman, "Date and Time on the Internet: Timestamps", RFC 3339, July 2002.
[7] Klyne，G.，Ed.和C.Newman，“互联网上的日期和时间：时间戳”，RFC 33392002年7月。
[8] Niemi, A., "Session Initiation Protocol (SIP) Extension for Event State Publication", RFC 3903, October 2004.
[8] Niemi，A.，“事件状态发布的会话启动协议（SIP）扩展”，RFC 3903，2004年10月。
[9] ITU-T G.1020, "Performance parameter definitions for quality of speech and other voiceband applications utilizing IP networks".
[9] ITU-T G.1020，“利用IP网络的语音质量和其他声带应用的性能参数定义”。
[10] ITU-T P.564, "Conformance testing for voice over IP transmission quality assessment models".
[10] ITU-T P.564，“IP语音传输质量评估模型的一致性测试”。
[11] ITU-T G.107, "The E-model, a computational model for use in transmission planning".
[11] ITU-T G.107，“E模型，用于输电规划的计算模型”。
[12] Almes, G., Kalidindi, S., and M. Zekauskas, "A One-way Delay Metric for IPPM", RFC 2679, September 1999.
[12] Almes，G.，Kalidini，S.，和M.Zekauskas，“IPPM的单向延迟度量”，RFC 2679，1999年9月。
[13] Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117, January 2008.
[13] Westerlund，M.和S.Westerlund，S.Wenger，“RTP拓扑”，RFC 51172008年1月。
[14] Hilt, V., Noel, E., Shen, C., and A. Abdelal, "Design Considerations for Session Initiation Protocol (SIP) Overload Control", Work in Progress, July 2009.
[14] Hilt，V.，Noel，E.，Shen，C.，和A.Abdelal，“会话启动协议（SIP）过载控制的设计考虑”，正在进行的工作，2009年7月。
[15] Petrie, D. and S. Channabasappa, "A Framework for Session Initiation Protocol User Agent Profile Delivery", Work in Progress, October 2010.
[15] Petrie，D.和S.Channabasappa，“会话启动协议用户代理配置文件交付框架”，正在进行的工作，2010年10月。
[16] Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session Traversal Utilities for NAT (STUN)", RFC 5389, October 2008.
[16] Rosenberg，J.，Mahy，R.，Matthews，P.，和D.Wing，“NAT（STUN）的会话遍历实用程序”，RFC 5389，2008年10月。
Authors' Addresses
作者地址
Amy Pendleton Telchemy Incorporated EMail: aspen@telchemy.com
Amy Pendleton Telchemy公司电子邮件：aspen@telchemy.com
Alan Clark Telchemy Incorporated EMail: alan.d.clark@telchemy.com
Alan Clark Telchemy公司电子邮件：Alan.d。clark@telchemy.com
Alan Johnston Avaya St. Louis, MO 63124 EMail: alan.b.johnston@gmail.com
密苏里州圣路易斯市艾伦·约翰斯顿·阿瓦亚63124电子邮件：Alan.b。johnston@gmail.com
Henry Sinnreich Unaffiliated EMail: henry.sinnreich@gmail.com
Henry Sinnreich非附属电子邮件：Henry。sinnreich@gmail.com