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

Abstract

  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.

Status of This Memo

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

  This document is a product of the Internet Engineering Task Force
  (IETF).  It represents the consensus of the IETF community.  It has
  received public review and has been approved for publication by the
  Internet Engineering Steering Group (IESG).  Not all documents
  approved by the IESG are a candidate for any level of Internet
  Standard; see Section 2 of RFC 5741.

  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.




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Copyright Notice

  Copyright (c) 2015 IETF Trust and the persons identified as the
  document authors.  All rights reserved.

  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.





































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Table of Contents

  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

  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



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  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.

  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.

  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.

  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.

  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.










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1.1.  Scope

  The scope of this document is summarized as follows:

  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  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  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  Regarding SIP signaling in presence of media:

     *  The media performance is not benchmarked.

     *  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.

     *  The type of DUT dictates whether the associated media streams
        traverse the DUT.  Both scenarios are within the scope of this
        document.

     *  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.

  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.



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  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  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  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  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  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  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.

2.  Terminology

  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.




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  For the sake of clarity and continuity, this document adopts the
  template for definitions set out in Section 2 of RFC 1242 [RFC1242].

  The term "Device Under Test (DUT)" is defined in Section 3.1.1 of RFC
  2285 [RFC2285].

  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].

  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  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  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.

3.  Term Definitions

3.1.  Protocol Components

3.1.1.  Session

  Definition:
     The combination of signaling and media messages and associated
     processing that enable a single SIP-based audio or video call, or
     SIP registration.

  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.



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  Measurement Units:
     N/A.

  Issues:
     None.

  See Also:
     Media Plane
     Signaling Plane
     Associated Media

3.1.2.  Signaling Plane

  Definition:
     The plane in which SIP messages [RFC3261] are exchanged between
     SIP agents [RFC3261].

  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.

  Measurement Units:
     N/A.

  Issues:
     None.

  See Also:
     Media Plane
     Emulated Agent

3.1.3.  Media Plane

  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.

  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.




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  Measurement Units:
     N/A.

  Issues:
     None.

  See Also:
     Signaling Plane

3.1.4.  Associated Media

  Definition:
     Media that corresponds to an 'm' line in the SDP payload of the
     Signaling Plane.

  Discussion:
     The format of the media is determined by the SDP attributes for
     the corresponding 'm' line.

  Measurement Units:
     N/A.

  Issues:
     None.

3.1.5.  Overload

  Definition:
     Overload is defined as the state where a SIP server does not have
     sufficient resources to process all incoming SIP messages
     [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.




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  Measurement Units:
     N/A.

3.1.6.  Session Attempt

  Definition:
     A SIP INVITE or REGISTER request sent by the EA that has not
     received a final response.

  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.

  Measurement Units:
     N/A.

  Issues:
     None.

  See Also:
     Session
     Session Attempt Rate

3.1.7.  Established Session

  Definition:
     A SIP session for which the EA acting as the UA has received a 200
     OK message.

  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.

  Measurement Units:
     N/A.

  Issues:
     None.




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  See Also:
     None.

3.1.8.  Session Attempt Failure

  Definition:
     A session attempt that does not result in an Established Session.

  Discussion:
     The session attempt failure may be indicated by the following
     observations at the EA:

     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).

  Measurement Units:
     N/A.

  Issues:
     None.

  See Also:
     Session Attempt

3.2.  Test Components

3.2.1.  Emulated Agent

  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.

  Discussion:
     The EA functions in the Signaling and Media Planes.  The Tester
     may act as multiple EAs.

  Measurement Units:
     N/A.

  Issues:
     None.






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  See Also:
     Media Plane
     Signaling Plane
     Established Session
     Associated Media

3.2.2.  Signaling Server

  Definition:
     Device in the test topology that facilitates the creation of
     sessions between EAs.  This device is the DUT.

  Discussion:
     The DUT is a network intermediary that is RFC 3261 capable such as
     a Registrar, Redirect Server, Stateful Proxy, B2BUA, or SBC.

  Measurement Units:
     N/A.

  Issues:
     None.

  See Also:
     Signaling Plane

3.2.3.  SIP Transport Protocol

  Definition:
     The protocol used for transport of the Signaling Plane messages.

  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.

  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.

  Issues:
     None.



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  See Also:
     None.

3.3.  Test Setup Parameters

3.3.1.  Session Attempt Rate

  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.

  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.

  Measurement Units:
     Session Attempts per second

  Issues:
     None.

  See Also:
     Session
     Session Attempt

3.3.2.  Establishment Threshold Time

  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.

  Discussion:
     This time duration is test dependent.

     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].

  Measurement Units:
     seconds

  Issues:
     None.





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  See Also:
     None.

3.3.3.  Session Duration

  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.

  Discussion:
     The time at which the BYE is sent will control the Session
     Duration.

  Measurement Units:
     seconds

  Issues:
     None.

  See Also:
     None.

3.3.4.  Media Packet Size

  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.

  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.

     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.

  Measurement Units:
     An integer number of frames or samples, depending on whether a
     hybrid- or sample-based codec is used, respectively.




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  Issues:
     None.

  See Also:
     None.

3.3.5.  Codec Type

  Definition:
     The name of the codec used to generate the media session.

  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.

  Measurement Units:
     This is a textual name (alphanumeric) assigned to uniquely
     identify the codec.

  Issues:
     None.
  See Also:
     None.

3.4.  Benchmarks

3.4.1.  Session Establishment Rate

  Definition:
     The maximum value of the Session Attempt Rate that the DUT can
     handle for an extended, predefined period with zero failures.

  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.

  Measurement Units:
     sessions per second (sps)

  Issues:
     None.





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  See Also:
     Session Attempt Rate

3.4.2.  Registration Rate

  Definition:
     The maximum value of the Registration Attempt Rate that the DUT
     can handle for an extended, predefined period with zero failures.

  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.

     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.

  Measurement Units:
     registrations per second (rps)

  Issues:
     None.

  See Also:
     None.








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3.4.3.  Registration Attempt Rate

  Definition:
     Configuration of the EA for the number of registrations per second
     that the EA attempts to send to the DUT.

  Discussion:
     The Registration Attempt Rate is the number of registration
     requests per second that the EA sends toward the DUT.

  Measurement Units:
     registrations per second (rps)

  Issues:
     None.

  See Also:
     None.

4.  Security Considerations

  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.




















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5.  References

5.1.  Normative References

  [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>.

  [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>.

  [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>.

  [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>.

  [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>.

  [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>.

5.2.  Informative References

  [RFC2285]  Mandeville, R., "Benchmarking Terminology for LAN
             Switching Devices", RFC 2285, February 1998,
             <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>.

  [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>.



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  [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>.

  [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>.

  [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>.

Acknowledgments

  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.

























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Authors' Addresses

  Carol Davids
  Illinois Institute of Technology
  201 East Loop Road
  Wheaton, IL  60187
  United States

  Phone: +1 630 682 6024
  EMail: [email protected]


  Vijay K. Gurbani
  Bell Laboratories, Alcatel-Lucent
  1960 Lucent Lane
  Rm 9C-533
  Naperville, IL  60566
  United States

  Phone: +1 630 224 0216
  EMail: [email protected]


  Scott Poretsky
  Allot Communications
  300 TradeCenter, Suite 4680
  Woburn, MA  08101
  United States

  Phone: +1 508 309 2179
  EMail: [email protected]




















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