Internet Engineering Task Force (IETF) C. Davids
Request for Comments: 7501 Illinois Institute of Technology
Category: Informational V. Gurbani
ISSN: 2070-1721 Bell Laboratories, Alcatel-Lucent
S. Poretsky
Allot Communications
April 2015
Internet Engineering Task Force (IETF) C. Davids
Request for Comments: 7501 Illinois Institute of Technology
Category: Informational V. Gurbani
ISSN: 2070-1721 Bell Laboratories, Alcatel-Lucent
S. Poretsky
Allot Communications
April 2015
Terminology for Benchmarking Session Initiation Protocol (SIP) Devices: Basic Session Setup and Registration
会话启动协议（SIP）设备基准术语：基本会话设置和注册
Abstract
摘要
This document provides a terminology for benchmarking the Session Initiation Protocol (SIP) performance of devices. Methodology related to benchmarking SIP devices is described in the companion methodology document (RFC 7502). Using these two documents, benchmarks can be obtained and compared for different types of devices such as SIP Proxy Servers, Registrars, and Session Border Controllers. The term "performance" in this context means the capacity of the Device Under Test (DUT) to process SIP messages. Media streams are used only to study how they impact the signaling behavior. The intent of the two documents is to provide a normalized set of tests that will enable an objective comparison of the capacity of SIP devices. Test setup parameters and a methodology are necessary because SIP allows a wide range of configurations and operational conditions that can influence performance benchmark measurements. A standard terminology and methodology will ensure that benchmarks have consistent definitions and were obtained following the same procedures.
本文档提供了设备会话启动协议（SIP）性能基准测试术语。与SIP设备基准测试相关的方法在配套方法文档（RFC 7502）中进行了描述。使用这两个文档，可以获得并比较不同类型设备（如SIP代理服务器、注册器和会话边界控制器）的基准测试。本上下文中的术语“性能”是指被测设备（DUT）处理SIP消息的能力。媒体流仅用于研究它们如何影响信令行为。这两份文件的目的是提供一组规范化的测试，以便对SIP设备的容量进行客观比较。测试设置参数和方法是必要的，因为SIP允许范围广泛的配置和操作条件，这些配置和操作条件可能会影响性能基准测量。标准术语和方法将确保基准具有一致的定义，并按照相同的程序获得。
Status of This Memo
关于下段备忘
This document is not an Internet Standards Track specification; it is published for informational purposes.
本文件不是互联网标准跟踪规范；它是为了提供信息而发布的。
This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Not all documents approved by the IESG are a candidate for any level of Internet Standard; see Section 2 of RFC 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。并非IESG批准的所有文件都适用于任何级别的互联网标准；见RFC 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/rfc7501.
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/rfc7501.
Copyright Notice
版权公告
Copyright (c) 2015 IETF Trust and the persons identified as the document authors. All rights reserved.
版权所有（c）2015 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。
Table of Contents
目录
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Scope . . . . . . . . . . . . . . . . . . . . . . . . . . 5
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. Term Definitions . . . . . . . . . . . . . . . . . . . . . . 7
3.1. Protocol Components . . . . . . . . . . . . . . . . . . . 7
3.1.1. Session . . . . . . . . . . . . . . . . . . . . . . . 7
3.1.2. Signaling Plane . . . . . . . . . . . . . . . . . . . 8
3.1.3. Media Plane . . . . . . . . . . . . . . . . . . . . . 8
3.1.4. Associated Media . . . . . . . . . . . . . . . . . . 9
3.1.5. Overload . . . . . . . . . . . . . . . . . . . . . . 9
3.1.6. Session Attempt . . . . . . . . . . . . . . . . . . . 10
3.1.7. Established Session . . . . . . . . . . . . . . . . . 10
3.1.8. Session Attempt Failure . . . . . . . . . . . . . . . 11
3.2. Test Components . . . . . . . . . . . . . . . . . . . . . 11
3.2.1. Emulated Agent . . . . . . . . . . . . . . . . . . . 11
3.2.2. Signaling Server . . . . . . . . . . . . . . . . . . 12
3.2.3. SIP Transport Protocol . . . . . . . . . . . . . . . 12
3.3. Test Setup Parameters . . . . . . . . . . . . . . . . . . 13
3.3.1. Session Attempt Rate . . . . . . . . . . . . . . . . 13
3.3.2. Establishment Threshold Time . . . . . . . . . . . . 13
3.3.3. Session Duration . . . . . . . . . . . . . . . . . . 14
3.3.4. Media Packet Size . . . . . . . . . . . . . . . . . . 14
3.3.5. Codec Type . . . . . . . . . . . . . . . . . . . . . 15
3.4. Benchmarks . . . . . . . . . . . . . . . . . . . . . . . 15
3.4.1. Session Establishment Rate . . . . . . . . . . . . . 15
3.4.2. Registration Rate . . . . . . . . . . . . . . . . . . 16
3.4.3. Registration Attempt Rate . . . . . . . . . . . . . . 17
4. Security Considerations . . . . . . . . . . . . . . . . . . . 17
5. References . . . . . . . . . . . . . . . . . . . . . . . . . 18
5.1. Normative References . . . . . . . . . . . . . . . . . . 18
5.2. Informative References . . . . . . . . . . . . . . . . . 18
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 19
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 20
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
1.1. Scope . . . . . . . . . . . . . . . . . . . . . . . . . . 5
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 6
3. Term Definitions . . . . . . . . . . . . . . . . . . . . . . 7
3.1. Protocol Components . . . . . . . . . . . . . . . . . . . 7
3.1.1. Session . . . . . . . . . . . . . . . . . . . . . . . 7
3.1.2. Signaling Plane . . . . . . . . . . . . . . . . . . . 8
3.1.3. Media Plane . . . . . . . . . . . . . . . . . . . . . 8
3.1.4. Associated Media . . . . . . . . . . . . . . . . . . 9
3.1.5. Overload . . . . . . . . . . . . . . . . . . . . . . 9
3.1.6. Session Attempt . . . . . . . . . . . . . . . . . . . 10
3.1.7. Established Session . . . . . . . . . . . . . . . . . 10
3.1.8. Session Attempt Failure . . . . . . . . . . . . . . . 11
3.2. Test Components . . . . . . . . . . . . . . . . . . . . . 11
3.2.1. Emulated Agent . . . . . . . . . . . . . . . . . . . 11
3.2.2. Signaling Server . . . . . . . . . . . . . . . . . . 12
3.2.3. SIP Transport Protocol . . . . . . . . . . . . . . . 12
3.3. Test Setup Parameters . . . . . . . . . . . . . . . . . . 13
3.3.1. Session Attempt Rate . . . . . . . . . . . . . . . . 13
3.3.2. Establishment Threshold Time . . . . . . . . . . . . 13
3.3.3. Session Duration . . . . . . . . . . . . . . . . . . 14
3.3.4. Media Packet Size . . . . . . . . . . . . . . . . . . 14
3.3.5. Codec Type . . . . . . . . . . . . . . . . . . . . . 15
3.4. Benchmarks . . . . . . . . . . . . . . . . . . . . . . . 15
3.4.1. Session Establishment Rate . . . . . . . . . . . . . 15
3.4.2. Registration Rate . . . . . . . . . . . . . . . . . . 16
3.4.3. Registration Attempt Rate . . . . . . . . . . . . . . 17
4. Security Considerations . . . . . . . . . . . . . . . . . . . 17
5. References . . . . . . . . . . . . . . . . . . . . . . . . . 18
5.1. Normative References . . . . . . . . . . . . . . . . . . 18
5.2. Informative References . . . . . . . . . . . . . . . . . 18
Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . . . 19
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 20
Service Providers and IT organizations deliver Voice Over IP (VoIP) and multimedia network services based on the IETF Session Initiation Protocol (SIP) [RFC3261]. SIP is a signaling protocol originally intended to be used to dynamically establish, disconnect, and modify streams of media between end users. As it has evolved, it has been adopted for use in a growing number of services and applications. Many of these result in the creation of a media session, but some do not. Examples of this latter group include text messaging and subscription services. The set of benchmarking terms provided in this document is intended for use with any SIP-enabled device
服务提供商和IT组织基于IETF会话发起协议（SIP）[RFC3261]提供IP语音（VoIP）和多媒体网络服务。SIP是一种信令协议，最初用于在最终用户之间动态建立、断开和修改媒体流。随着它的发展，它已被越来越多的服务和应用程序采用。其中许多导致创建媒体会话，但有些则没有。后一组的示例包括短信和订阅服务。本文档中提供的基准术语集旨在用于任何支持SIP的设备
performing SIP functions in the interior of the network, whether or not these result in the creation of media sessions. The performance of end-user devices is outside the scope of this document.
在网络内部执行SIP功能，无论这些功能是否导致创建媒体会话。最终用户设备的性能不在本文档的范围内。
A number of networking devices have been developed to support SIP-based VoIP services. These include SIP servers, Session Border Controllers (SBCs), and Back-to-back User Agents (B2BUAs). These devices contain a mix of voice and IP functions whose performance may be reported using metrics defined by the equipment manufacturer or vendor. The Service Provider or IT organization seeking to compare the performance of such devices will not be able to do so using these vendor-specific metrics, whose conditions of test and algorithms for collection are often unspecified.
已经开发了许多网络设备来支持基于SIP的VoIP服务。这些包括SIP服务器、会话边界控制器（SBC）和背靠背用户代理（B2BUA）。这些设备包含语音和IP功能的混合，其性能可以使用设备制造商或供应商定义的指标进行报告。寻求比较此类设备性能的服务提供商或IT组织将无法使用这些特定于供应商的指标进行比较，这些指标的测试条件和收集算法通常未指定。
SIP functional elements and the devices that include them can be configured many different ways and can be organized into various topologies. These configuration and topological choices impact the value of any chosen signaling benchmark. Unless these conditions of test are defined, a true comparison of performance metrics across multiple vendor implementations will not be possible.
SIP功能元件和包括它们的设备可以以多种不同的方式配置，并且可以组织成各种拓扑。这些配置和拓扑选择会影响所选信令基准的价值。除非定义了这些测试条件，否则就不可能在多个供应商实施中对性能指标进行真正的比较。
Some SIP-enabled devices terminate or relay media as well as signaling. The processing of media by the device impacts the signaling performance. As a result, the conditions of test must include information as to whether or not the Device Under Test processes media. If the device processes media during the test, a description of the media must be provided. This document and its companion methodology document [RFC7502] provide a set of black-box benchmarks for describing and comparing the performance of devices that incorporate the SIP User Agent Client and Server functions and that operate in the network's core.
一些支持SIP的设备终止或中继媒体以及信令。设备对媒体的处理会影响信令性能。因此，测试条件必须包括被测设备是否处理介质的信息。如果设备在测试期间处理介质，则必须提供介质说明。本文档及其配套方法文档[RFC7502]提供了一组黑盒基准，用于描述和比较包含SIP用户代理客户端和服务器功能以及在网络核心中运行的设备的性能。
The definition of SIP performance benchmarks necessarily includes definitions of Test Setup Parameters and a test methodology. These enable the Tester to perform benchmarking tests on different devices and to achieve comparable results. This document provides a common set of definitions for Test Components, Test Setup Parameters, and Benchmarks. All the benchmarks defined are black-box measurements of the SIP signaling plane. The Test Setup Parameters and Benchmarks defined in this document are intended for use with the companion methodology document.
SIP性能基准的定义必然包括测试设置参数和测试方法的定义。这使得测试人员能够在不同的设备上执行基准测试，并获得可比较的结果。本文档提供了测试组件、测试设置参数和基准的一组通用定义。定义的所有基准都是SIP信令平面的黑盒测量。本文件中定义的测试设置参数和基准旨在与配套方法文件一起使用。
The scope of this document is summarized as follows:
本文件的范围总结如下：
o This terminology document describes SIP signaling performance benchmarks for black-box measurements of SIP networking devices. Stress conditions and debugging scenarios are not addressed in this document.
o 本术语文档描述了SIP网络设备黑盒测量的SIP信令性能基准。本文档中不讨论压力条件和调试场景。
o The DUT must be network equipment that is RFC 3261 capable. This may be a Registrar, Redirect Server, or Stateful Proxy. This document does not require the intermediary to assume the role of a stateless proxy. A DUT may also act as a B2BUA or take the role of an SBC.
o DUT必须是具有RFC 3261功能的网络设备。这可能是注册器、重定向服务器或有状态代理。本文档不要求中介机构承担无状态代理的角色。DUT也可以充当B2BUA或SBC的角色。
o The Tester acts as multiple Emulated Agents (EAs) that initiate (or respond to) SIP messages as session endpoints and source (or receive) associated media for established connections.
o 测试仪充当多个模拟代理（EA），启动（或响应）SIP消息作为会话端点，并为已建立的连接提供（或接收）相关媒体。
o Regarding SIP signaling in presence of media:
o 关于存在媒体时的SIP信令：
* The media performance is not benchmarked.
* 媒体表现没有基准。
* Some tests require media, but the use of media is limited to observing the performance of SIP signaling. Tests that require media will annotate the media characteristics as a condition of test.
* 有些测试需要介质，但介质的使用仅限于观察SIP信令的性能。需要介质的测试将标注介质特性作为测试条件。
* The type of DUT dictates whether the associated media streams traverse the DUT. Both scenarios are within the scope of this document.
* DUT的类型指示相关联的媒体流是否穿过DUT。这两种情况都在本文档的范围内。
* SIP is frequently used to create media streams; the signaling plane and media plane are treated as orthogonal to each other in this document. While many devices support the creation of media streams, benchmarks that measure the performance of these streams are outside the scope of this document and its companion methodology document [RFC7502]. Tests may be performed with or without the creation of media streams. The presence or absence of media streams MUST be noted as a condition of the test, as the performance of SIP devices may vary accordingly. Even if the media is used during benchmarking, only the SIP performance will be benchmarked, not the media performance or quality.
* SIP经常用于创建媒体流；在本文档中，信令平面和媒体平面被视为彼此正交。虽然许多设备支持创建媒体流，但测量这些流性能的基准不在本文档及其附带方法文档[RFC7502]的范围内。可以在创建或不创建媒体流的情况下执行测试。媒体流的存在或不存在必须作为测试的一个条件加以注意，因为SIP设备的性能可能会相应地变化。即使在基准测试期间使用了介质，也只会对SIP性能进行基准测试，而不会对介质性能或质量进行基准测试。
o Both INVITE and non-INVITE scenarios (registrations) are addressed in this document. However, benchmarking SIP presence or subscribe-notify extensions is not a part of this document.
o 本文档介绍了邀请和非邀请场景（注册）。但是，对SIP状态或订阅通知扩展进行基准测试不是本文档的一部分。
o Different transport -- such as UDP, TCP, SCTP, or TLS -- may be used. The specific transport mechanism MUST be noted as a condition of the test, as the performance of SIP devices may vary accordingly.
o 可以使用不同的传输，例如UDP、TCP、SCTP或TLS。特定的传输机制必须作为测试的一个条件加以说明，因为SIP设备的性能可能会相应地发生变化。
o REGISTER and INVITE requests may be challenged or remain unchallenged for authentication purposes. Whether or not the REGISTER and INVITE requests are challenged is a condition of test that will be recorded along with other such parameters that may impact the SIP performance of the device or system under test.
o 出于身份验证目的，注册和邀请请求可能会受到质疑或保持不变。寄存器和INVITE请求是否被质疑是一个测试条件，将与可能影响被测设备或系统的SIP性能的其他此类参数一起记录。
o Re-INVITE requests are not considered within the scope of this document since the benchmarks for INVITEs are based on the dialog created by the INVITE and not on the transactions that take place within that dialog.
o 重新邀请请求不在本文件范围内，因为邀请基准基于邀请创建的对话框，而不是该对话框中发生的事务。
o Only session establishment is considered for the performance benchmarks. Session disconnect is not considered within the scope of this document. This is because our goal is to determine the maximum capacity of the device or system under test, that is, the number of simultaneous SIP sessions that the device or system can support. It is true that there are BYE requests being created during the test process. These transactions do contribute to the load on the device or system under test and thus are accounted for in the metric we derive. We do not seek a separate metric for the number of BYE transactions a device or system can support.
o 性能基准只考虑建立会话。会话断开不在本文档的范围内。这是因为我们的目标是确定被测设备或系统的最大容量，即设备或系统可以支持的同时SIP会话数。在测试过程中确实创建了BYE请求。这些事务确实会增加被测设备或系统的负载，因此在我们导出的度量中进行了说明。对于设备或系统能够支持的BYE事务数，我们不寻求单独的度量标准。
o Scenarios that are specific to the IP Multimedia Subsystem (IMS) are not considered, but test cases can be applied with 3GPP-specific SIP signaling and the Proxy-Call Session Control Function (P-CSCF) as a DUT.
o 不考虑特定于IP多媒体子系统（IMS）的场景，但是测试用例可以与3GPP特定SIP信令和代理呼叫会话控制功能（P-CSCF）一起应用，作为DUT。
o The benchmarks described in this document are intended for a laboratory environment and are not intended to be used on a production network. Some of the benchmarks send enough traffic that a denial-of-service attack is possible if used in production networks.
o 本文件中描述的基准适用于实验室环境，不适用于生产网络。一些基准测试发送的流量足够大，如果在生产网络中使用，就有可能发生拒绝服务攻击。
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, RFC2119 [RFC2119]. RFC 2119 defines the use of these key words to help make the intent of Standards Track documents as clear as possible. While this document uses these keywords, this document is not a Standards Track document.
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、RFC2119[RFC2119]中的说明进行解释。RFC 2119定义了这些关键词的使用，以帮助尽可能明确标准跟踪文档的意图。虽然本文档使用这些关键字，但本文档不是标准跟踪文档。
For the sake of clarity and continuity, this document adopts the template for definitions set out in Section 2 of RFC 1242 [RFC1242].
为了清晰和连续性，本文件采用RFC 1242[RFC1242]第2节规定的定义模板。
The term "Device Under Test (DUT)" is defined in Section 3.1.1 of RFC 2285 [RFC2285].
RFC 2285[RFC2285]第3.1.1节定义了术语“被测设备（DUT）”。
Many commonly used SIP terms in this document are defined in RFC 3261 [RFC3261]. For convenience, the most important of these are reproduced below. Use of these terms in this document is consistent with their corresponding definition in the base SIP specification [RFC3261] as amended by [RFC4320], [RFC5393], and [RFC6026].
RFC 3261[RFC3261]中定义了本文档中许多常用的SIP术语。为方便起见，以下转载了其中最重要的部分。本文件中这些术语的使用与经[RFC4320]、[RFC5393]和[RFC6026]修订的基本SIP规范[RFC3261]中相应的定义一致。
o Call Stateful: A proxy is call stateful if it retains state for a dialog from the initiating INVITE to the terminating BYE request. A call stateful proxy is always transaction stateful, but the converse is not necessarily true.
o Call Stateful：如果代理保留从发起邀请到终止BYE请求的对话框的状态，那么它就是Call Stateful。调用有状态代理始终是事务有状态的，但反之不一定成立。
o Stateful Proxy: A logical entity, as defined by [RFC3261], that maintains the client and server transaction state machines during the processing of a request. (Also known as a transaction stateful proxy.) The behavior of a stateful proxy is further defined in Section 16 of RFC 3261 [RFC3261] . A transaction stateful proxy is not the same as a call stateful proxy.
o 有状态代理：由[RFC3261]定义的逻辑实体，在处理请求期间维护客户端和服务器事务状态机。（也称为事务有状态代理。）有状态代理的行为在RFC 3261[RFC3261]第16节中有进一步定义。事务有状态代理与调用有状态代理不同。
o Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a logical entity that receives a request and processes it as a user agent server (UAS). In order to determine how the request should be answered, it acts as a user agent client (UAC) and generates requests. Unlike a proxy server, it maintains dialog state and must participate in all requests sent on the dialogs it has established. Since it is a concatenation of a UAC and a UAS, no explicit definitions are needed for its behavior.
o 背靠背用户代理：背靠背用户代理（B2BUA）是接收请求并将其作为用户代理服务器（UAS）处理的逻辑实体。为了确定应该如何响应请求，它充当用户代理客户端（UAC）并生成请求。与代理服务器不同，它维护对话框状态，并且必须参与在其建立的对话框上发送的所有请求。由于它是UAC和UAS的串联，因此其行为不需要显式定义。
Definition: The combination of signaling and media messages and associated processing that enable a single SIP-based audio or video call, or SIP registration.
定义：信令和媒体消息以及相关处理的组合，支持单个基于SIP的音频或视频呼叫或SIP注册。
Discussion: The term "session" commonly implies a media session. In this document the term is extended to cover the signaling and any media specified and invoked by the corresponding signaling.
讨论：“会议”一词通常指媒体会议。在本文件中，该术语扩展至涵盖信令以及相应信令指定和调用的任何媒体。
Measurement Units: N/A.
计量单位：不适用。
Issues: None.
问题：没有。
See Also: Media Plane Signaling Plane Associated Media
另请参见：媒体平面信令平面关联媒体
Definition: The plane in which SIP messages [RFC3261] are exchanged between SIP agents [RFC3261].
定义：SIP代理[RFC3261]之间交换SIP消息[RFC3261]的平面。
Discussion: SIP messages are used to establish sessions in several ways: directly between two User Agents [RFC3261], through a Proxy Server [RFC3261], or through a series of Proxy Servers. The Session Description Protocol (SDP) is included in the Signaling Plane.
讨论：SIP消息以几种方式用于建立会话：直接在两个用户代理[RFC3261]之间、通过代理服务器[RFC3261]或通过一系列代理服务器。会话描述协议（SDP）包括在信令平面中。
Measurement Units: N/A.
计量单位：不适用。
Issues: None.
问题：没有。
See Also: Media Plane Emulated Agent
另请参见：媒体平面模拟代理
Definition: The data plane in which one or more media streams and their associated media control protocols (e.g., RTCP [RFC3550]) are exchanged between User Agents after a media connection has been created by the exchange of signaling messages in the Signaling Plane.
定义：一个或多个媒体流及其相关媒体控制协议（如RTCP[RFC3550]）在通过信令平面中的信令消息交换创建媒体连接后在用户代理之间交换的数据平面。
Discussion: Media may also be known as the "bearer channel". The Media Plane MUST include the media control protocol, if one is used, and the media stream(s). Examples of media are audio and video. The media streams are described in the SDP of the Signaling Plane.
讨论：媒体也被称为“承载频道”。媒体平面必须包括媒体控制协议（如果使用）和媒体流。媒体的例子有音频和视频。在信令平面的SDP中描述媒体流。
Measurement Units: N/A.
计量单位：不适用。
Issues: None.
问题：没有。
See Also: Signaling Plane
另见：信号机
Definition: Media that corresponds to an 'm' line in the SDP payload of the Signaling Plane.
定义：与信令平面SDP有效载荷中的“m”线相对应的媒体。
Discussion: The format of the media is determined by the SDP attributes for the corresponding 'm' line.
讨论：媒体的格式由相应“m”行的SDP属性决定。
Measurement Units: N/A.
计量单位：不适用。
Issues: None.
问题：没有。
Definition: Overload is defined as the state where a SIP server does not have sufficient resources to process all incoming SIP messages [RFC6357].
定义：重载定义为SIP服务器没有足够资源处理所有传入SIP消息的状态[RFC6357]。
Discussion: The distinction between an overload condition and other failure scenarios is outside the scope of black-box testing and of this document. Under overload conditions, all or a percentage of Session Attempts will fail due to lack of resources. In black-box testing, the cause of the failure is not explored. The fact that a failure occurred for whatever reason will trigger the tester to reduce the offered load, as described in the companion methodology document [RFC7502]. SIP server resources may include CPU processing capacity, network bandwidth, input/output queues, or disk resources. Any combination of resources may be fully utilized when a SIP server (the DUT) is in the overload condition. For proxy-only (or intermediary) devices, it is expected that the proxy will be driven into overload based on the delivery rate of signaling requests.
讨论：过载条件和其他故障场景之间的区别不在黑盒测试和本文档的范围内。在过载情况下，由于资源不足，所有或一定百分比的会话尝试都将失败。在黑盒测试中，未探索故障原因。如配套方法文件[RFC7502]所述，无论出于何种原因发生故障，都会触发测试仪降低提供的负载。SIP服务器资源可能包括CPU处理能力、网络带宽、输入/输出队列或磁盘资源。当SIP服务器（DUT）处于过载状态时，可以充分利用任何资源组合。对于仅代理（或中间）设备，预计代理将根据信令请求的传递速率而过载。
Measurement Units: N/A.
计量单位：不适用。
Definition: A SIP INVITE or REGISTER request sent by the EA that has not received a final response.
定义：EA发送的尚未收到最终响应的SIP邀请或注册请求。
Discussion: The attempted session may be either an invitation to an audio/ video communication or a registration attempt. When counting the number of session attempts, we include all requests that are rejected for lack of authentication information. The EA needs to record the total number of session attempts including those attempts that are routinely rejected by a proxy that requires the UA to authenticate itself. The EA is provisioned to deliver a specific number of session attempts per second. But the EA must also count the actual number of session attempts per given time interval.
讨论：尝试的会话可能是音频/视频通信邀请，也可能是注册尝试。在计算会话尝试次数时，我们将包括因缺少身份验证信息而被拒绝的所有请求。EA需要记录会话尝试的总数，包括那些需要UA进行自我验证的代理通常拒绝的尝试。EA被配置为每秒提供特定数量的会话尝试。但是EA还必须计算每个给定时间间隔的实际会话尝试次数。
Measurement Units: N/A.
计量单位：不适用。
Issues: None.
问题：没有。
See Also: Session Session Attempt Rate
另请参见：会话尝试率
Definition: A SIP session for which the EA acting as the UA has received a 200 OK message.
定义：充当UA的EA已收到200 OK消息的SIP会话。
Discussion: An Established Session may be either an invitation to an audio/ video communication or a registration attempt. Early dialogs for INVITE requests are out of scope for this work.
讨论：已建立的会话可以是音频/视频通信邀请，也可以是注册尝试。INVITE请求的早期对话框不在此工作范围内。
Measurement Units: N/A.
计量单位：不适用。
Issues: None.
问题：没有。
See Also: None.
另见：无。
Definition: A session attempt that does not result in an Established Session.
定义：未导致已建立会话的会话尝试。
Discussion: The session attempt failure may be indicated by the following observations at the EA:
讨论：EA的以下观察结果可能表明会话尝试失败：
1. Receipt of a SIP 3xx-, 4xx-, 5xx-, or 6xx-class response to a Session Attempt. 2. The lack of any received SIP response to a Session Attempt within the Establishment Threshold Time (cf. Section 3.3.2).
1. 接收对会话尝试的SIP 3xx、4xx、5xx或6xx类响应。2.在建立阈值时间内，没有收到对会话尝试的任何SIP响应（参见第3.3.2节）。
Measurement Units: N/A.
计量单位：不适用。
Issues: None.
问题：没有。
See Also: Session Attempt
另请参见：会话尝试
Definition: A device in the test topology that initiates/responds to SIP messages as one or more session endpoints and, wherever applicable, sources/receives Associated Media for Established Sessions.
定义：测试拓扑中的一种设备，作为一个或多个会话端点启动/响应SIP消息，并在适用的情况下，为已建立的会话提供/接收相关媒体。
Discussion: The EA functions in the Signaling and Media Planes. The Tester may act as multiple EAs.
讨论：EA在信令和媒体平面中的功能。测试仪可以充当多个EA。
Measurement Units: N/A.
计量单位：不适用。
Issues: None.
问题：没有。
See Also: Media Plane Signaling Plane Established Session Associated Media
另请参见：媒体平面信令平面已建立会话关联媒体
Definition: Device in the test topology that facilitates the creation of sessions between EAs. This device is the DUT.
定义：测试拓扑中的设备，有助于在EA之间创建会话。这个设备就是DUT。
Discussion: The DUT is a network intermediary that is RFC 3261 capable such as a Registrar, Redirect Server, Stateful Proxy, B2BUA, or SBC.
讨论：DUT是具有RFC 3261功能的网络中介，例如注册器、重定向服务器、有状态代理、B2BUA或SBC。
Measurement Units: N/A.
计量单位：不适用。
Issues: None.
问题：没有。
See Also: Signaling Plane
另见：信号机
Definition: The protocol used for transport of the Signaling Plane messages.
定义：用于传输信令平面消息的协议。
Discussion: Performance benchmarks may vary for the same SIP networking device depending upon whether TCP, UDP, TLS, SCTP, websockets [RFC7118], or any future transport-layer protocol is used. For this reason, it is necessary to measure the SIP Performance Benchmarks using these various transport protocols. Performance Benchmarks MUST report the SIP Transport Protocol used to obtain the benchmark results.
讨论：根据是否使用TCP、UDP、TLS、SCTP、WebSocket[RFC7118]或任何未来的传输层协议，同一SIP网络设备的性能基准可能会有所不同。因此，有必要使用这些不同的传输协议来测量SIP性能基准。性能基准必须报告用于获得基准测试结果的SIP传输协议。
Measurement Units: While these are not units of measure, they are attributes that are one of many factors that will contribute to the value of the measurements to be taken. TCP, UDP, SCTP, TLS over TCP, TLS over UDP, TLS over SCTP, and websockets are among the possible values to be recorded as part of the test.
度量单位：虽然这些不是度量单位，但它们是属性，是许多因素之一，这些因素将有助于测量值。TCP、UDP、SCTP、TCP上的TLS、UDP上的TLS、SCTP上的TLS和WebSocket都是作为测试一部分记录的可能值。
Issues: None.
问题：没有。
See Also: None.
另见：无。
Definition: Configuration of the EA for the number of sessions per second (sps) that the EA attempts to establish using the services of the DUT.
定义：EA使用DUT的服务尝试建立的每秒会话数（SP）的EA配置。
Discussion: The Session Attempt Rate is the number of sessions per second that the EA sends toward the DUT. Some of the sessions attempted may not result in a session being established.
讨论：会话尝试率是EA每秒向DUT发送的会话数。尝试的某些会话可能不会导致建立会话。
Measurement Units: Session Attempts per second
度量单位：每秒会话尝试次数
Issues: None.
问题：没有。
See Also: Session Session Attempt
另请参见：会话尝试
Definition: Configuration of the EA that represents the amount of time that an EA client will wait for a response from an EA server before declaring a Session Attempt Failure.
定义：EA的配置，表示EA客户端在声明会话尝试失败之前等待EA服务器响应的时间量。
Discussion: This time duration is test dependent.
讨论：此持续时间取决于测试。
It is RECOMMENDED that the Establishment Threshold Time value be set to Timer B or Timer F as specified in RFC 3261, Table 4 [RFC3261].
建议将建立阈值时间值设置为RFC 3261表4[RFC3261]中规定的计时器B或计时器F。
Measurement Units: seconds
测量单位：秒
Issues: None.
问题：没有。
See Also: None.
另见：无。
Definition: Configuration of the EA that represents the amount of time that the SIP dialog is intended to exist between the two EAs associated with the test.
定义：EA的配置，表示SIP对话框在与测试相关的两个EA之间存在的时间量。
Discussion: The time at which the BYE is sent will control the Session Duration.
讨论：发送BYE的时间将控制会话持续时间。
Measurement Units: seconds
测量单位：秒
Issues: None.
问题：没有。
See Also: None.
另见：无。
Definition: Configuration on the EA for a fixed number of frames or samples to be sent in each RTP packet of the media stream when the test involves Associated Media.
定义：EA上的配置，当测试涉及相关媒体时，在媒体流的每个RTP数据包中发送固定数量的帧或样本。
Discussion: This document describes a method to measure SIP performance. If the DUT is processing media as well as SIP messages the media processing will potentially slow down the SIP processing and lower the SIP performance metric. The tests with associated media are designed for audio codecs, and the assumption was made that larger media packets would require more processor time. This document does not define parameters applicable to video codecs.
讨论：本文档描述了一种测量SIP性能的方法。如果DUT正在处理媒体以及SIP消息，则媒体处理可能会减慢SIP处理并降低SIP性能指标。与相关媒体的测试是为音频编解码器设计的，并且假设较大的媒体数据包需要更多的处理器时间。本文档未定义适用于视频编解码器的参数。
For a single benchmark test, media sessions use a defined number of samples or frames per RTP packet. If two SBCs, for example, used the same codec but one puts more frames into the RTP packet, this might cause variation in the performance benchmark results.
对于单个基准测试，媒体会话使用每个RTP数据包定义数量的样本或帧。例如，如果两个SBC使用相同的编解码器，但其中一个将更多帧放入RTP数据包，这可能会导致性能基准测试结果发生变化。
Measurement Units: An integer number of frames or samples, depending on whether a hybrid- or sample-based codec is used, respectively.
测量单位：帧数或采样数的整数，具体取决于使用的是混合编解码器还是基于采样的编解码器。
Issues: None.
问题：没有。
See Also: None.
另见：无。
Definition: The name of the codec used to generate the media session.
定义：用于生成媒体会话的编解码器的名称。
Discussion: For a single benchmark test, all sessions use the same size packet for media streams. The size of packets can cause a variation in the performance benchmark measurements.
讨论：对于单个基准测试，所有会话对媒体流使用相同大小的数据包。数据包的大小可能会导致性能基准测量值发生变化。
Measurement Units: This is a textual name (alphanumeric) assigned to uniquely identify the codec.
度量单位：这是指定用于唯一标识编解码器的文本名称（字母数字）。
Issues: None. See Also: None.
问题：没有。另见：无。
Definition: The maximum value of the Session Attempt Rate that the DUT can handle for an extended, predefined period with zero failures.
定义：DUT在无故障的延长预定义时间段内可处理的会话尝试率的最大值。
Discussion: This benchmark is obtained with zero failure. The Session Attempt Rate provisioned on the EA is raised and lowered as described in the algorithm in the accompanying methodology document [RFC7502], until a traffic load over the period of time necessary to attempt N sessions completes without failure, where N is a parameter specified in the algorithm and recorded in the Test Setup Report.
讨论：该基准是在零故障的情况下获得的。按照随附的方法文件[RFC7502]中的算法所述，提高和降低EA上设置的会话尝试率，直到尝试N个会话所需的时间段内的流量负载无故障完成，其中N是算法中指定的参数，并记录在测试设置报告中。
Measurement Units: sessions per second (sps)
测量单位：每秒会话数（sps）
Issues: None.
问题：没有。
See Also: Session Attempt Rate
另请参见：会话尝试率
Definition: The maximum value of the Registration Attempt Rate that the DUT can handle for an extended, predefined period with zero failures.
定义：DUT在无故障的延长预定义周期内可处理的注册尝试率的最大值。
Discussion: This benchmark is obtained with zero failures. The registration rate provisioned on the Emulated Agent is raised and lowered as described in the algorithm in the companion methodology document [RFC7502], until a traffic load consisting of registration attempts at the given attempt rate over the period of time necessary to attempt N registrations completes without failure, where N is a parameter specified in the algorithm and recorded in the Test Setup Report.
讨论：该基准是在零故障的情况下获得的。按照配套方法文件[RFC7502]中的算法所述，提高和降低在仿真代理上设置的注册率，直到在尝试N次注册所需的时间段内，以给定的尝试率完成注册尝试所构成的流量负载无故障完成，其中N是算法中指定的参数，并记录在测试设置报告中。
This benchmark is described separately from the Session Establishment Rate (Section 3.4.1), although it could be considered a special case of that benchmark, since a REGISTER request is a request for a session that is not initiated by an INVITE request. It is defined separately because it is a very important benchmark for most SIP installations. An example demonstrating its use is an avalanche restart, where hundreds of thousands of endpoints register simultaneously following a power outage. In such a case, an authoritative measurement of the capacity of the device to register endpoints is useful to the network designer. Additionally, in certain controlled networks, there appears to be a difference between the registration rate of new endpoints and the registering rate of existing endpoints (register refreshes). This benchmark can capture these differences as well.
该基准与会话建立率（第3.4.1节）分开描述，尽管它可以被视为该基准的特例，因为注册请求是对会话的请求，而该会话不是由INVITE请求发起的。它是单独定义的，因为它是大多数SIP安装的一个非常重要的基准。演示其使用的一个示例是雪崩重启，断电后数十万个端点同时注册。在这种情况下，设备注册端点的能力的权威性度量对于网络设计者是有用的。此外，在某些受控网络中，新端点的注册率与现有端点的注册率（寄存器刷新）之间似乎存在差异。这个基准也可以捕捉这些差异。
Measurement Units: registrations per second (rps)
测量单位：每秒注册数（rps）
Issues: None.
问题：没有。
See Also: None.
另见：无。
Definition: Configuration of the EA for the number of registrations per second that the EA attempts to send to the DUT.
定义：EA每秒尝试发送给DUT的注册数的EA配置。
Discussion: The Registration Attempt Rate is the number of registration requests per second that the EA sends toward the DUT.
讨论：注册尝试率是EA每秒向DUT发送的注册请求数。
Measurement Units: registrations per second (rps)
测量单位：每秒注册数（rps）
Issues: None.
问题：没有。
See Also: None.
另见：无。
Documents of this type do not directly affect the security of the Internet or corporate networks as long as benchmarking is not performed on devices or systems connected to production networks. Security threats and how to counter these in SIP and the media layer are discussed in RFC 3261 [RFC3261], RFC 3550 [RFC3550], and RFC 3711 [RFC3711]. This document attempts to formalize a set of common terminology for benchmarking SIP networks. Packets with unintended and/or unauthorized DSCP or IP precedence values may present security issues. Determining the security consequences of such packets is out of scope for this document.
只要不在连接到生产网络的设备或系统上执行基准测试，此类文档不会直接影响互联网或公司网络的安全性。RFC 3261[RFC3261]、RFC 3550[RFC3550]和RFC 3711[RFC3711]讨论了SIP和媒体层中的安全威胁以及如何应对这些威胁。本文档试图将SIP网络基准测试的一组通用术语形式化。具有意外和/或未经授权的DSCP或IP优先级值的数据包可能会出现安全问题。确定此类数据包的安全后果超出了本文档的范围。
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997, <http://www.rfc-editor.org/info/rfc2119>.
[RFC2119]Bradner，S.，“RFC中用于表示需求水平的关键词”，BCP 14，RFC 2119，1997年3月<http://www.rfc-editor.org/info/rfc2119>.
[RFC3261] 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, <http://www.rfc-editor.org/info/rfc3261>.
[RFC3261]Rosenberg，J.，Schulzrinne，H.，Camarillo，G.，Johnston，A.，Peterson，J.，Sparks，R.，Handley，M.，和E.Schooler，“SIP：会话启动协议”，RFC 3261，2002年6月<http://www.rfc-editor.org/info/rfc3261>.
[RFC5393] Sparks, R., Ed., Lawrence, S., Hawrylyshen, A., and B. Campen, "Addressing an Amplification Vulnerability in Session Initiation Protocol (SIP) Forking Proxies", RFC 5393, December 2008, <http://www.rfc-editor.org/info/rfc5393>.
[RFC5393]Sparks，R.，Ed.，Lawrence，S.，Hawrylyshen，A.，和B.Campen，“解决会话启动协议（SIP）分叉代理中的放大漏洞”，RFC 5393，2008年12月<http://www.rfc-editor.org/info/rfc5393>.
[RFC4320] Sparks, R., "Actions Addressing Identified Issues with the Session Initiation Protocol's (SIP) Non-INVITE Transaction", RFC 4320, January 2006, <http://www.rfc-editor.org/info/rfc4320>.
[RFC4320]Sparks，R.“解决会话启动协议（SIP）非邀请事务中已识别问题的措施”，RFC 4320，2006年1月<http://www.rfc-editor.org/info/rfc4320>.
[RFC6026] Sparks, R. and T. Zourzouvillys, "Correct Transaction Handling for 2xx Responses to Session Initiation Protocol (SIP) INVITE Requests", RFC 6026, September 2010, <http://www.rfc-editor.org/info/rfc6026>.
[RFC6026]Sparks，R.和T.Zourzouvillys，“会话启动协议（SIP）邀请请求2xx响应的正确事务处理”，RFC 60262010年9月<http://www.rfc-editor.org/info/rfc6026>.
[RFC7502] Davids, C., Gurbani, V., and S. Poretsky, "Terminology for Benchmarking Session Initiation Protocol (SIP) Devices: Basic Session Setup and Registration", RFC 7502, April 2015, <http://www.rfc-editor.org/info/rfc7502>.
[RFC7502]Davids，C.，Gurbani，V.，和S.Poretsky，“会话启动协议（SIP）设备基准术语：基本会话设置和注册”，RFC 7502，2015年4月<http://www.rfc-editor.org/info/rfc7502>.
[RFC2285] Mandeville, R., "Benchmarking Terminology for LAN Switching Devices", RFC 2285, February 1998, <http://www.rfc-editor.org/info/rfc2285>.
[RFC2285]Mandeville，R.，“局域网交换设备的基准术语”，RFC 22852998年2月<http://www.rfc-editor.org/info/rfc2285>.
[RFC1242] Bradner, S., "Benchmarking Terminology for Network Interconnection Devices", RFC 1242, July 1991, <http://www.rfc-editor.org/info/rfc1242>.
[RFC1242]Bradner，S.，“网络互连设备的基准术语”，RFC 1242，1991年7月<http://www.rfc-editor.org/info/rfc1242>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3550]Schulzrinne，H.，Casner，S.，Frederick，R.，和V.Jacobson，“RTP：实时应用的传输协议”，STD 64，RFC 35502003年7月<http://www.rfc-editor.org/info/rfc3550>.
[RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norrman, "The Secure Real-time Transport Protocol (SRTP)", RFC 3711, March 2004, <http://www.rfc-editor.org/info/rfc3711>.
[RFC3711]Baugher，M.，McGrew，D.，Naslund，M.，Carrara，E.，和K.Norrman，“安全实时传输协议（SRTP）”，RFC 37112004年3月<http://www.rfc-editor.org/info/rfc3711>.
[RFC6357] Hilt, V., Noel, E., Shen, C., and A. Abdelal, "Design Considerations for Session Initiation Protocol (SIP) Overload Control", RFC 6357, August 2011, <http://www.rfc-editor.org/info/rfc6357>.
[RFC6357]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”，RFC 6357，2011年8月<http://www.rfc-editor.org/info/rfc6357>.
[RFC7118] Baz Castillo, I., Millan Villegas, J., and V. Pascual, "The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP)", RFC 7118, January 2014, <http://www.rfc-editor.org/info/rfc7118>.
[RFC7118]Baz Castillo，I.，Millan Villegas，J.，和V.Pascual，“作为会话启动协议（SIP）传输的WebSocket协议”，RFC 7118，2014年1月<http://www.rfc-editor.org/info/rfc7118>.
Acknowledgments
致谢
The authors would like to thank Keith Drage, Cullen Jennings, Daryl Malas, Al Morton, and Henning Schulzrinne for invaluable contributions to this document. Dale Worley provided an extensive review that lead to improvements in the documents. We are grateful to Barry Constantine, William Cerveny, and Robert Sparks for providing valuable comments during the documents' last calls and expert reviews. Al Morton and Sarah Banks have been exemplary working group chairs; we thank them for tracking this work to completion.
作者要感谢Keith Drage、Cullen Jennings、Daryl Malas、Al Morton和Henning Schulzrinne对本文件的宝贵贡献。Dale Worley进行了广泛的审查，从而改进了文件。我们感谢Barry Constantine、William Cerveny和Robert Sparks在文件的最后通话和专家审查期间提供了宝贵的意见。Al Morton和Sarah Banks是工作组的模范主席；我们感谢他们跟踪这项工作直到完成。
Authors' Addresses
作者地址
Carol Davids Illinois Institute of Technology 201 East Loop Road Wheaton, IL 60187 United States
卡罗尔·戴维斯伊利诺伊理工学院，美国伊利诺伊州惠顿东环路201号，邮编60187
Phone: +1 630 682 6024
EMail: davids@iit.edu
Phone: +1 630 682 6024
EMail: davids@iit.edu
Vijay K. Gurbani Bell Laboratories, Alcatel-Lucent 1960 Lucent Lane Rm 9C-533 Naperville, IL 60566 United States
Vijay K.Gurbani Bell实验室，阿尔卡特朗讯1960朗讯巷，美国伊利诺伊州纳珀维尔9C-533室，邮编：60566
Phone: +1 630 224 0216
EMail: vkg@bell-labs.com
Phone: +1 630 224 0216
EMail: vkg@bell-labs.com
Scott Poretsky Allot Communications 300 TradeCenter, Suite 4680 Woburn, MA 08101 United States
Scott Poretsky Allot Communications 300交易中心，美国马萨诸塞州沃本4680室，邮编：08101
Phone: +1 508 309 2179
EMail: sporetsky@allot.com
Phone: +1 508 309 2179
EMail: sporetsky@allot.com