PMOL                                                            D. Malas
Internet-Draft                                                 CableLabs
Intended status: Standards Track                               A. Morton
Expires: November 7, 2010                                      AT&T Labs
                                                             May 6, 2010


           Basic Telephony SIP End-to-End Performance Metrics
                  draft-ietf-pmol-sip-perf-metrics-05


   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
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   Without obtaining an adequate license from the person(s) controlling
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   it for publication as an RFC or to translate it into languages other
   than English.

Abstract

   This document defines a set of metrics and their usage to evaluate
   the performance of end-to-end Session Initiation Protocol (SIP) for
   telephony services in both production and testing environments.  The
   purpose of this document is to combine a standard set of common
   metrics, allowing interoperable performance measurements, easing the
   comparison of industry implementations.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on November 7, 2010.



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Copyright Notice

   Copyright (c) 2010 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
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   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 and Scope . . . . . . . . . . . . . . . . . . . .  4
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  5
   3.  Time Interval Measurement and Reporting  . . . . . . . . . . .  6
   4.  SIP Performance Metrics  . . . . . . . . . . . . . . . . . . .  7
     4.1.  Registration Request Delay (RRD) . . . . . . . . . . . . .  8
     4.2.  Ineffective Registration Attempts (IRA)  . . . . . . . . .  9
     4.3.  Post Dial Delay (PDD)  . . . . . . . . . . . . . . . . . . 11
       4.3.1.  Successful Session Setup PDD . . . . . . . . . . . . . 11
       4.3.2.  Failed Session Setup PDD . . . . . . . . . . . . . . . 12
     4.4.  Session Disconnect Delay (SDD) . . . . . . . . . . . . . . 13
     4.5.  Call Hold Time (CHT) . . . . . . . . . . . . . . . . . . . 15
       4.5.1.  Successful session duration CHT  . . . . . . . . . . . 15
       4.5.2.  Failed session completion CHT  . . . . . . . . . . . . 17
     4.6.  Answer Seizure Ratio (ASR) . . . . . . . . . . . . . . . . 18
     4.7.  Network Effectiveness Ratio (NER)  . . . . . . . . . . . . 19
     4.8.  Ineffective Session Attempts (ISA) . . . . . . . . . . . . 20
     4.9.  Session Completion Ratio (SCR) . . . . . . . . . . . . . . 20
       4.9.1.  Successful Session Completion  . . . . . . . . . . . . 21
       4.9.2.  Failed Session Completion  . . . . . . . . . . . . . . 22
   5.  Additional Considerations  . . . . . . . . . . . . . . . . . . 22
     5.1.  Metric Correlations  . . . . . . . . . . . . . . . . . . . 22
     5.2.  Back-to-back User Agent (B2BUA)  . . . . . . . . . . . . . 23
     5.3.  Authorization and Authentication . . . . . . . . . . . . . 23
     5.4.  Forking  . . . . . . . . . . . . . . . . . . . . . . . . . 23
     5.5.  Data Collection  . . . . . . . . . . . . . . . . . . . . . 24
     5.6.  Testing Documentation  . . . . . . . . . . . . . . . . . . 24
   6.  Conclusions  . . . . . . . . . . . . . . . . . . . . . . . . . 24
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 25
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 25
   9.  Contributors . . . . . . . . . . . . . . . . . . . . . . . . . 25
   10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 25
   11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 25
     11.1. Normative References . . . . . . . . . . . . . . . . . . . 25
     11.2. Informative References . . . . . . . . . . . . . . . . . . 26
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27














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1.  Introduction and Scope

   SIP has become a widely-used standard among many service providers,
   vendors, and end users in the telecommunications industry.  Although
   there are many different standards for measuring the performance of
   telephony signaling protocols, such as SS7, none of the metrics
   specifically address SIP.

   The scope of this document is limited to the definitions of a
   standard set of metrics for measuring and reporting SIP performance
   from an end-to-end perspective in a telephony environment.  The
   metrics introduce a common foundation for understanding and
   quantifying performance expectations between service providers,
   vendors, and the users of services based on SIP.  The intended
   audience for this document can be found among network operators, who
   often collect information on the responsiveness of the network to
   customer requests for services.

   Measurements of the metrics described in this document are affected
   by variables external to SIP.  The following is a non-exhaustive list
   of examples:

   o  Network connectivity

   o  Switch and router performance

   o  Server processes and hardware performance

   This document defines a list of pertinent metrics for varying aspects
   of a telephony environment.  They may be used individually or as a
   set based on the usage of SIP within the context of a given
   telecommunications service.

   The metrics defined in this document DO NOT take into consideration
   the impairment or failure of actual application processing of a
   request or response.  The metrics do not distinguish application
   processing time from other sources of delay, such as packet transfer
   delay.

   Metrics designed to quantify single device application processing
   performance are beyond the scope of this document.

   This document does not provide any numerical objectives or acceptance
   threshold values for the SIP performance metrics defined below, as
   these items are beyond the scope of IETF activities, in general.

   The metrics defined in this document are applicable in scenarios
   where the SIP messages launched (into a network under test) are



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   dedicated messages for testing purposes, or where the messages are
   user-initiated and a portion of the live traffic present.  These two
   scenarios are sometimes referred to as active and passive
   measurement, respectively.


2.  Terminology

   The following terms and conventions will be used throughout this
   document:

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [RFC2119].

   End-to-End - This is described as two or more elements utilized for
   initiating a request, receiving the request, and responding to the
   request.  It encompasses elements as necessary to be involved in a
   session dialog between the originating user agent client (UAC),
   destination user agent server (UAS), and any interim proxies (may
   also include back-to-back user agent's (B2BUA's)).  This may be
   relative to a single operator's set of elements or extend to
   encompass all elements (if beyond a single operator's network)
   associated with a session.

   Session - As described in RFC 3261 [RFC3261], SIP is used primarily
   to request, create, and conclude sessions.  "These sessions include
   Internet telephone calls, multimedia distribution, and multimedia
   conferences."  The metrics within this document measure the
   performance associated with the SIP dialogs necessary to establish
   these sessions; therefore, they are titled as: Session Request Delay,
   Session Disconnect Delay, etc.  Although the titles of many of the
   metrics include this term, they are specifically measuring the
   signaling aspects only.  Each session is identified by a unique
   Call-ID, "To" and "From" header field tag.

   Session Establishment - Session establishment occurs when a 200 OK
   response from the target UA has been received, in response to the
   originating UA's INVITE setup request, indicating the session setup
   request was successful.

   Session Setup - As referenced within the sub-sections of 4.2 in this
   document, session setup is the set of messages and included
   parameters directly related to the process of a UA requesting to
   establish a session with a corresponding UA.  This is also described
   as a set of steps in order to establish "ringing" [RFC3261].





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3.  Time Interval Measurement and Reporting

   Many of the metrics defined in this memo utilize a clock to assess
   the time interval between two events.  This section defines time-
   related terms and reporting requirements.

   t1 - start time

   This is the time instant (when a request is sent) that begins a
   continuous time interval.  T1 occurs when the designated request has
   been processed by the SIP application and the first bit of the
   request packet has been sent from the UA or proxy (and is externally
   observable at some logical or physical interface).

   T1 represents the time at which each request-response test begins,
   and SHALL be used to designate the time-of-day when a particular
   measurement was conducted (e.g., The Session Request Delay at "t1"
   and (some specific UA interface) was measured to be X ms.)

   t4 - end time

   This is the time instant that concludes the continuous time interval
   begun when the related request is sent. t4 occurs when the last bit
   of the designated response is received by the SIP application at the
   requesting device (and is externally observable at some logical or
   physical interface).

   Note: The designations t2 and t3 are reserved for future use at
   another interface involved in satisfying a request.

   Section 10.1 of [RFC2330] describes time-related issues in
   measurements, and defines the errors that can be attributed to the
   clock themselves.  These definitions are used in the material below.

   Time of Day Accuracy

   As defined above, t1 is associated with the start of a request and
   also serves as the time-of-day stamp associated with a single
   specific measurement.  The clock offset [RFC2330] is the difference
   between t1 and a recognized primary source of time, such as UTC
   (offset = t1 - UTC).

   When measurement results will be correlated with other results or
   information using time-of-day stamps, then the time clock that
   supplies t1 SHOULD be synchronized to a primary time source, to
   minimize the clock's offset.  The clocks used at the different
   measurement points SHOULD be synchronized to each other, to minimize
   the relative offset (as defined in RFC2330).  The clock's offset and



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   the relative offset MUST be reported with each measurement.

   Time Interval Accuracy

   The accuracy of the t4-t1 interval is also critical to maintain and
   report.  The difference between a clock's offsets at t1 and t4 is one
   source of error for the measurement and is associated with the
   clock's skew [RFC2330].

   A stable and reasonably accurate clock is needed to make the time
   interval measurements required by this memo.  This source of error
   SHOULD be constrained to less than +/- 1 ms, implying 1 part per 1000
   frequency accuracy for a 1 second interval.  This implies greater
   stability is required as the length of the t4-t1 increases, in order
   to constrain the error to be less than +/- 1ms.

   There are several other important aspects of clock operation:

   1.  Synchronization protocols require some ability to make
       adjustments to the local clock.  However, these adjustments
       (clock steps or slewing) can cause large errors if they occur
       during the t1 to t4 measurement interval.  Clock correction
       SHOULD be suspended during a t1 to t4 measurement interval,
       unless the time interval accuracy requirement above will be met.
       Alternatively, a measurement SHOULD NOT be performed during clock
       correction, unless the time interval accuracy requirement above
       will be met.

   2.  If a free-running clock is used to make the time interval
       measurement, then the time of day reported with the measurement
       (which is normally timestamp t1) SHOULD be derived from a
       different clock that meets the time of day accuracy requirements
       described above.

   The physical operation of reading time from a clock may be
   constrained by the delay to service the interrupt.  Therefore, if the
   accuracy of the time stamp read at t1 or t4 includes the interrupt
   delay, this source of error SHOULD be known and included in the error
   assessment.


4.  SIP Performance Metrics

   In regards to all of the following metrics, t1 begins with the first
   associated SIP message in a dialog sent by either UA, and is not
   reset if the UA must retransmit the same message, within the same
   transaction, multiple times.  The first associated SIP message
   indicates the t1 associated with the user or application expectation



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   relative to the request.

   Some metrics are calculated based on the final response message.
   These metrics do not take into consideration route advances (e.g.
   SIP redirects) or negotiations, such as SIP Require, session timers
   and re-attempts (e.g.  SIP 488 Not Acceptable Here).  In these unique
   cases, the final response related to the initial setup attempt SHOULD
   be utilized for input to the metric.

   The authentication method used to establish the SIP dialog will
   change the message exchanges.  The example message exchanges used do
   not attempt to describe all of the various authentication types.
   Since, authentication is frequently used, SIP Digest authentication
   was used for example purposes.

   In regards to all of the metrics, the accuracy and granularity of the
   output values are related to the accuracy and granularity of the
   input values.

   While these metrics do not specify the sample size.  This should be
   taken into consideration.  A small sample size, such as 1 or 2
   attempts, could result in a negative calculation for some of the
   metrics.  These metrics will provide a better indication of
   performance with larger sample sets.  For example, some SIP Service
   Providers (SSPs) may choose to collect input over an hour, daily,
   weekly or monthly timeframe, while another SSP may choose to perform
   metric calculations over a varying set of SIP dialogs.

4.1.  Registration Request Delay (RRD)

   Registration Request Delay (RRD) is a measurement of the delay in
   responding to a UA REGISTER request.  RRD SHALL be measured and
   reported only for successful REGISTER requests, while Ineffective
   Registration Attempts (Section 4.2) SHALL be reported for failures.
   This metric is measured at the UA.  The output value of this metric
   is numerical and SHOULD be stated in units of milliseconds.  The RRD
   is calculated using the following formula:

   RRD = Time of Final Response - Time of REGISTER Request

   In a successful registration attempt, RRD is defined as the time
   interval from the first bit of the initial REGISTER message
   containing the necessary information is passed by the originating UA
   to the intended registrar until the last bit of the 200 OK is
   received indicating the registration attempt has completed
   successfully.  This dialog includes an expected authentication
   challenge prior to receiving the 200 OK as described in the following
   registration flow examples.



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   The following message exchange provides an example of identifiable
   events necessary for inputs in calculating RRD during a successful
   registration completion:


                  UA1                 Registrar
                   |                      |
                   |REGISTER              |
            t1---->|--------------------->|
               /\  |                   401|
               ||  |<---------------------|
              RRD  |REGISTER              |
               ||  |--------------------->|
               \/  |                   200|
            t4---->|<---------------------|
                   |                      |


   Note: Networks with elements using primarily Digest authentication
   will exhibit different RRD characteristics than networks with
   elements primarily using other authentication mechanisms (such as
   Identity).  Operators monitoring RRD in networks with a mixture of
   authentications schemes should take note that the RRD measurements
   will likely have a multimodal distribution.

4.2.  Ineffective Registration Attempts (IRA)

   Ineffective registration attempts are utilized to detect failures or
   impairments causing an inability for a registrar to receive a UA
   REGISTER request.  This metric is measured at the UA.  The output
   value of this metric is numerical and SHOULD be reported as a
   percentage of registration attempts.

   This metric is calculated as a percentage of total REGISTER requests.
   The IRA is calculated using the following formula:


                        # of IRA
      IRA % = ----------------------------- x 100
               Total # of REGISTER Requests


   A failed registration attempt is defined as a final failure response
   to the initial REGISTER request.  It usually indicates a failure
   received from the destination registrar, interim proxies, or due to a
   timeout of the REGISTER request at the originating UA.  A failure
   response is described as a 4XX (excluding 401, 402, and 407 non-
   failure challenge response codes), 5XX, or possible 6XX message.  A



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   timeout failure is identified by the timer F expiring.  IRA may be
   used to detect problems in downstream signaling functions, which may
   be impairing the REGISTER message from reaching the intended
   registrar; or, it may indicate a registrar has become overloaded and
   is unable to respond to the request.

   The following message exchange provides a timeout example of an
   identifiable event necessary for input as a failed registration
   attempt:


                  UA1                Registrar
                   |                      |
                   |REGISTER              |
                   |--------------------->|
                   |REGISTER              |
                   |--------------------->|
                   |REGISTER              |
                   |--------------------->|
                   |                      |
      Failure ---->|***Timer F Expires    |
                   |                      |


   In the previous message exchange UA1 retries a REGISTER request
   multiple times before the timer, indicating the failure, expires.
   Only the first REGISTER request MUST used for input to the
   calculation and an IRA.  Subsequent REGISTER retries are identified
   by the same transaction identifier (same topmost Via header field
   branch parameter value) and MUST be ignored for purposes of metric
   calculation.  This ensures an accurate representation of the metric
   output.

   The following message exchange provides a registrar servicing failure
   example of an identifiable event necessary for input as a failed
   registration attempt:


                  UA1                Registrar
                   |                      |
                   |REGISTER              |
                   |--------------------->|
                   |                      |
                   |                      |
                   |                      |
                   |                      |
                   |                   503|
      Failure ---->|<---------------------|



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                   |                      |

4.3.  Post Dial Delay (PDD)

   Post Dial Delay (PDD) is utilized to detect failures or impairments
   causing delays in responding to a UA session request.  PDD is
   measured for both successful and failed session setup requests as
   this metric usually relates to a user experience; however, PDD for
   session requests ending in a failure MUST NOT be combined in the same
   result with successful requests.  The duration associated with
   success and failure responses will likely vary substantially, and the
   desired output time associated with each will be significantly
   different in many cases.  This metric is similar to Post-Selection
   Delay [E.721], and it is measured at the originating UA only.  The
   output value of this metric MUST indicate whether the output is for
   successful or failed session requests and SHOULD be stated in units
   of seconds.  The PDD is calculated using the following formula:

   PDD = Time of Status Indicative Response - Time of INVITE

4.3.1.  Successful Session Setup PDD

   In a successful request attempt, PDD is defined as the time interval
   from the first bit of the initial INVITE message containing the
   necessary information is sent by the originating user agent to the
   intended mediation or destination agent until the last bit of the
   first provisional response is received indicating an audible or
   visual status of the initial session setup request.  In SIP, the
   message indicating status would be a non-100 Trying provisional
   message received in response to an INVITE request.  In some cases, a
   non-100 Trying provisional message is not received, but rather a 200
   message is received as the first status message instead.  In these
   situations, the 200 message would be used to calculate the interval.
   In most circumstances, this metric relies on receiving a non-100
   Trying message.  The use of PRACK [RFC3262] MAY improve the quality
   and consistency of the results.

   The following message exchange provides an example of identifiable
   events necessary for inputs in calculating PDD during a successful
   session setup request without a redirect (i.e. 3XX message):











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                  UA1                    UA2
                   |                      |
                   |INVITE                |
            t1---->|--------------------->|
               /\  |                      |
               ||  |                      |
              PDD  |                      |
               ||  |                      |
               \/  |                   180|
            t4---->|<---------------------|
                   |                      |


   The following message exchange provides an example of identifiable
   events necessary for inputs in calculating PDD during a successful
   session setup with a redirect (e.g. 302 Moved Temporarily):


                  UA1             Redirect Server              UA2
                   |                      |                     |
                   |INVITE                |                     |
            t1---->|--------------------->|                     |
               /\  |                   302|                     |
               ||  |<---------------------|                     |
               ||  |ACK                   |                     |
              PDD  |--------------------->|                     |
               ||  |INVITE                                      |
               ||  |------------------------------------------->|
               \/  |                                         180|
            t4---->|<-------------------------------------------|


4.3.2.  Failed Session Setup PDD

   In a failed request attempt, PDD is defined as the time interval from
   the first bit of the initial INVITE message containing the necessary
   information sent by the originating agent or user to the intended
   mediation or destination agent until the last bit of the first
   provisional response or a failure indication response.  A failure
   response is described as a 4XX (excluding 401, 402, and 407 non-
   failure challenge response codes), 5XX, or possible 6XX message.  PDD
   may be used to detect problems in downstream signaling functions,
   which may be impairing the INVITE message from reaching the intended
   UA.  While this metric calculates the delay associated with a failed
   session request, the metric Ineffective Session Attempts (Section
   4.10) is used for calculating a ratio of session attempt failures.

   The following message exchange provides an example of identifiable



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   events necessary for inputs in calculating PDD during a failed
   session setup attempt without a redirect (i.e. 3XX message):


                  UA1                    UA2
                   |                      |
                   |INVITE                |
            t1---->|--------------------->|
               /\  |                      |
               ||  |                      |
              PDD  |                      |
               ||  |                      |
               \/  |                   480|
            t4---->|<---------------------|
                   |                      |


   The following message exchange provides an example of identifiable
   events necessary for inputs in calculating PDD during a failed
   session setup attempt with a redirect (e.g. 302 Moved Temporarily):


                  UA1             Redirect Server              UA2
                   |                      |                     |
                   |INVITE                |                     |
            t1---->|--------------------->|                     |
               /\  |                   302|                     |
               ||  |<---------------------|                     |
               ||  |ACK                   |                     |
              PDD  |--------------------->|                     |
               ||  |INVITE                                      |
               ||  |------------------------------------------->|
               \/  |                                         480|
            t4---->|<-------------------------------------------|


4.4.  Session Disconnect Delay (SDD)

   This metric is utilized to detect failures or impairments delaying
   the time necessary to end a session.  It can be measured from either
   end-point UAs involved in the SIP dialog.  SDD is measured for both
   successful and failed session completions.  The output value of this
   metric is numerical and SHOULD be stated in units of milliseconds.
   The SDD is calculated using the following formula:

   SDD = Time of 2XX or Timeout - Time of Completion Message (BYE)

   SDD is defined as the interval between the first bit of the sent



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   session completion message, such as a BYE, and the last bit of the
   subsequently received 2XX acknowledgment.  The following message
   exchanges provide an example of identifiable events necessary for
   inputs in calculating SDD during a successful session completion:

   Measuring SDD at the originating UA (UA1) -


                  UA1                    UA2
                   |                      |
                   |INVITE                |
                   |--------------------->|
                   |                   180|
                   |<---------------------|
                   |                   200|
                   |<---------------------|
                   |ACK                   |
                   |--------------------->|
                   |BYE                   |
            t1---->|--------------------->|
               /\  |                      |
               ||  |                      |
              SDD  |                      |
               ||  |                      |
               \/  |                   200|
            t4---->|<---------------------|


   Measuring SDD at the target UA (UA2) -


                  UA1                    UA2
                   |                      |
                   |INVITE                |
                   |--------------------->|
                   |                   180|
                   |<---------------------|
                   |                   200|
                   |<---------------------|
                   |ACK                   |
                   |--------------------->|
                   |                   BYE|
                   |<---------------------|<----t1
                   |                      |  /\
                   |                      |  ||
                   |                      | SDD
                   |                      |  ||
                   |200                   |  \/



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                   |--------------------->|<----t4


   In some cases, no response is received after a session completion
   message is sent and potentially retried.  In this case, the
   completion message, such as a BYE, results in a Timer F expiration.
   Sessions ending in this manner SHOULD be excluded from the metric
   calculation.

4.5.  Call Hold Time (CHT)

   This metric is used to detect problems (e.g. poor audio quality)
   causing short session durations.  CHT is measured for both successful
   and failed session completions.  It can be measured from both a UAC
   and UAS perspective.  This metric is traditionally calculated as
   Average Call Hold Time (ACHT) in telephony applications of SIP.  The
   output value of this metric is numerical and SHOULD be stated in
   units of seconds.  The CHT is calculated using the following formula:

   CHT = Time of BYE or Timeout - Time of 200 OK response to INVITE

   This metric does not calculate the duration of sessions leveraging
   early media.  For example, some automated response systems only use
   early media by responding with a SIP 183 Session in Progress message
   with SDP connecting the originating UA with the the automated
   message.  Usually, in these sessions the originating UA never
   receives a 200 OK, and the message exchange ends with the originating
   UA sending a CANCEL.

4.5.1.  Successful session duration CHT

   In a successful session completion, CHT is calculated as an average
   and is defined as the duration of a dialog defined by the interval
   from receipt of the first bit of a 200 OK response to an INVITE and
   receipt of the last bit of an associated BYE message indicating
   dialog completion.

   The following message exchanges provide an example of identifiable
   events necessary for inputs in calculating CHT during a successful
   session completion (The message message exchanges are changed between
   the originating and target UAs to provide varying examples.):

   Measuring CHT at the originating UA (UA1) -








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                  UA1                    UA2
                   |                      |
                   |INVITE                |
                   |--------------------->|
                   |                   180|
                   |<---------------------|
                   |                   200|
            t1---->|<---------------------|
               /\  |ACK                   |
               ||  |--------------------->|
               ||  |                      |
              CHT  |                      |
               ||  |                      |
               ||  |                      |
               \/  |BYE                   |
            t4---->|--------------------->|
                   |                      |


   When measuring CHT at the target UA (UA2), it is defined by the
   interval from sending the first bit of a 200 OK response to an INVITE
   and receipt of the last bit of an associated BYE message indicating
   dialog completion.  If UA2 initiates the BYE, then it is defined by
   the interval from sending the first bit of a 200 OK response to an
   INVITE and sending the first bit of an associated BYE message
   indicating dialog completion.  This is illustrated in the following
   example message exchange -


                  UA1                    UA2
                   |                      |
                   |INVITE                |
                   |--------------------->|
                   |                   180|
                   |<---------------------|
                   |                   200|
                   |<---------------------|<----t1
                   |ACK                   |  /\
                   |--------------------->|  ||
                   |                      |  ||
                   |                      |  CHT
                   |                      |  ||
                   |                      |  ||
                   |                   BYE|  \/
                   |<---------------------|<----t4
                   |                      |





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   (In these two examples, t1 is the same even if either UA receives the
   BYE instead of sending it.)

4.5.2.  Failed session completion CHT

   In some cases, no response is received after a session completion
   message is sent and potentially retried.  In this case, CHT is
   defined as the interval between receiving the first bit of a 200 OK
   response to an INVITE, and the resulting Timer F expiration.  The
   following message exchanges provide an example of identifiable events
   necessary for inputs in calculating CHT during a failed session
   completion attempt:

   Measuring CHT at the originating UA (UA1) -


                  UA1                    UA2
                   |                      |
                   |INVITE                |
                   |--------------------->|
                   |                   180|
                   |<---------------------|
                   |                   200|
            t1---->|<---------------------|
               /\  |BYE                   |
               ||  |--------------------->|
               ||  |BYE                   |
              CHT  |--------------------->|
               ||  |BYE                   |
               ||  |--------------------->|
               \/  |                      |
            t4---->|***Timer F Expires    |


   When measuring CHT at UA2, SDT is defined as the interval between
   sending the first bit of a 200 OK response to an INVITE, and the
   resulting Timer F expiration.  This is illustrated in the following
   example message exchange -













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                  UA1                    UA2
                   |                      |
                   |INVITE                |
                   |--------------------->|
                   |                   180|
                   |<---------------------|
                   |                   200|
                   |<---------------------|<----t1
                   |                   BYE|  /\
                   |<---------------------|  ||
                   |                   BYE|  ||
                   |<---------------------|  CHT
                   |                   BYE|  ||
                   |<---------------------|  ||
                   |                      |  \/
                   |    Timer F Expires***|<----t4


4.6.  Answer Seizure Ratio (ASR)

   This metric is used to detect the ability of a terminating UA or
   downstream proxy to successfully establish sessions per new session
   INVITE requests.  SER is defined as the number of new session INVITE
   requests resulting in a 200 OK response, to the total number of
   attempted INVITE requests less INVITE requests resulting in a 3XX
   response.  This metric is similar to Answer Seizure Ratio (ASR)
   defined in [E.411].  It is measured at the originating UA only.  The
   output value of this metric is numerical and SHOULD be adjusted to
   indicate a percentage of successfully established sessions.  The ASR
   is calculated using the following formula:


                 # of INVITE Requests w/ associated 200 OK
   ASR = --------------------------------------------------------- x 100
     (Total # of INVITE Requests)-(# of INVITE Requests w/ 3XX Response)


   The following message exchange provides an example of identifiable
   events necessary for inputs in determining session establishment as
   described above:











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                           UA1                 UA2
                            |                   |
                            |INVITE             |
               +----------->|------------------>|
               |            |                180|
               |            |<------------------|
      Session Established   |                   |
               |            |                   |
               |            |                200|
               +----------->|<------------------|
                            |                   |


4.7.  Network Effectiveness Ratio (NER)

   This metric is complimentary to ASR, but is intended to exclude the
   potential effects of the target UA from the metric.  NER is defined
   as the number of INVITE requests resulting in a 200 OK response and
   INVITE requests resulting in a 480, 486, or 600; to the total number
   of attempted INVITE requests less INVITE requests resulting in a 3XX,
   401, 402, and 407 response.  These response codes were chosen,
   because they indicate an acceptable UA effect without the interaction
   of an individual user of the UA.  If future response codes provide
   similar effects, they should be considered in addition with this
   metric.  This metric is similar to Network Effectiveness Ratio (NER)
   defined in [E.411].  It is measured at the originating UA only.  The
   output value of this metric is numerical and SHOULD be adjusted to
   indicate a percentage of successfully established sessions less
   common UAS failures.

   In order to simplify the formula, the following variable is used to
   summarize multiple SIP responses:

   a = 3XX, 401, 402, and 407

   The NER is calculated using the following formula:


        # of INVITE Requests w/ associated 200 OK, 480, 486, or 600
   NER = -------------------------------------------------------- x 100
     (Total # of INVITE Requests)-(# of INVITE Requests w/ 'a' Response)


   Reference the example flow is Section 4.7.







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4.8.  Ineffective Session Attempts (ISA)

   Ineffective session attempts occur when a proxy or agent internally
   releases a setup request with a failed or overloaded condition.  This
   metric is similar to Ineffective Machine Attempts (IMA) in telephony
   applications of SIP, and was adopted from Telcordia GR-512-CORE
   [GR-512].  The output value of this metric is numerical and SHOULD be
   adjusted to indicate a percentage of ineffective session attempts.
   The following failure responses provide a guideline for this
   criterion:

   o  408 Request Timeout

   o  500 Server Internal Error

   o  503 Service Unavailable

   o  504 Server Timeout

   This set was derived in a similar manner as described in Section 4.9,
   in addition 408 failure responses may indicate an overloaded state
   with a downstream element; however, there are situations other than
   overload, which may cause an increase in 408 responses.

   This metric is calculated as a percentage of total session setup
   requests.  The ISA is calculated using the following formula:


                        # of ISA
      ISA % = ----------------------------- x 100
               Total # of Session Requests


4.9.  Session Completion Ratio (SCR)

   A session completion is defined as a SIP dialog, which completes
   without failing due to a lack of response from an intended proxy or
   UA.  This metric is only used when at least one proxy is involved in
   the dialog.  This metric is similar to Call Completion Ratio (CCR) in
   telephony applications of SIP.  The output value of this metric is
   numerical and SHOULD be adjusted to indicate a percentage of
   successfully completed sessions.

   This metric is calculated as a percentage of total sessions completed
   successfully.  The SCR is calculated using the following formula:


                # of Successfully Completed Sessions



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      SCR % = --------------------------------------- x 100
                     Total # of Session Requests


4.9.1.  Successful Session Completion

   A session completes successfully when it begins with a setup request
   and ends with a session completion message.

   The following dialog [RFC3665] provides an example describing the
   necessary message exchanges of a successful session completion:


          UA1           Proxy 1          Proxy 2             UA2
           |                |                |                |
           |INVITE          |                |                |
           |--------------->|                |                |
           |             407|                |                |
           |<---------------|                |                |
           |ACK             |                |                |
           |--------------->|                |                |
           |INVITE          |                |                |
           |--------------->|INVITE          |                |
           |             100|--------------->|INVITE          |
           |<---------------|             100|--------------->|
           |                |<---------------|                |
           |                |                |             180|
           |                |            180 |<---------------|
           |             180|<---------------|                |
           |<---------------|                |             200|
           |                |             200|<---------------|
           |             200|<---------------|                |
           |<---------------|                |                |
           |ACK             |                |                |
           |--------------->|ACK             |                |
           |                |--------------->|ACK             |
           |                |                |--------------->|
           |                Both Way RTP Media                |
           |<================================================>|
           |                |                |             BYE|
           |                |             BYE|<---------------|
           |             BYE|<---------------|                |
           |<---------------|                |                |
           |200             |                |                |
           |--------------->|200             |                |
           |                |--------------->|200             |
           |                |                |--------------->|
           |                |                |                |



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4.9.2.  Failed Session Completion

   Session completion fails when an INVITE is sent from the originating
   UA, but there is no indication the INVITE reached the target UA.
   This can also occur if the target UA does not respond to the INVITE
   or the response never reaches the target UA associated with the
   session.

   The following dialog provides an example describing the necessary
   message exchanges of an unsuccessful session completion:


          UA1           Proxy 1          Proxy 2             UA2
           |                |                |                |
           |INVITE          |                |                |
           |--------------->|                |                |
           |             407|                |                |
           |<---------------|                |                |
           |ACK             |                |                |
           |--------------->|                |                |
           |INVITE          |                |                |
           |--------------->|INVITE          |                |
           |             100|--------------->|INVITE          |
           |<---------------|             100|--------------->|
           |                |<---------------|                |
           |                |                |INVITE          |
           |                |                |--------------->|
           |                |                |                |
           |                |                |INVITE          |
           |                |                |--------------->|
           |                |                |                |
           |                |             408|                |
           |             408|<---------------|                |
           |<---------------|ACK             |                |
           |                |--------------->|                |
           |ACK             |                |                |
           |--------------->|                |                |



5.  Additional Considerations

5.1.  Metric Correlations

   These metrics may be used to determine the performance of a domain
   and/or user.  This would be to provide a metric relative to one or
   more dimensions.  The following is an example subset of dimensions
   for providing further granularity per metric:



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   o  To "user"

   o  From "user"

   o  Bi-direction "user"

   o  To "domain"

   o  From "domain"

   o  Bi-direction "domain"

5.2.  Back-to-back User Agent (B2BUA)

   A B2BUA may impact the ability to collect these metrics with an end-
   to-end perspective.  It is necessary to realize a B2BUA may act as an
   originating UAC and terminating UAS or it may act as a proxy.  In
   some cases, it may be necessary to consider information collected
   from both sides of the B2BUA in order to determine the end-to-end
   perspective.  In other cases, the B2BUA may act simply as a proxy
   allowing data to be derived as necessary for the input into any of
   the listed calculations.

5.3.  Authorization and Authentication

   During the process of setting up a SIP dialog, various authentication
   methods may be utilized.  These authentication methods will add to
   the duration as measured by the metrics, and the length of time will
   vary based on those methods.  The failures of these authentication
   methods will also be captured by these metrics, since SIP is
   ultimately used to indicate the success or failure of the
   authorization and/or authentication attempt.  The metrics in section
   3 are inclusive of the duration associated with this process, even if
   the method is external to the SIP protocol.  This was included
   purposefully, due to its inherent impact on the protocol and the
   subsequent SIP dialogs.

5.4.  Forking

   Forking SHOULD be considered when determining the messages associated
   with the input values for the described metrics.  If all of the
   forked dialogs were used in the metric calculations, the numbers
   would skew dramatically.  There are two different points of forking,
   which MUST be considered.  First, forking may occur at a proxy
   downstream from the UA that is being used for metric input values.
   The downstream proxy is responsible for forking a message.  Then,
   this proxy will send provisional (e.g. 180) messages received from
   the requests and send the accepted (e.g. 200) response to the UA.



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   Second, in the cases where the originating UA or proxy is forking the
   messages, then it MUST parse the message exchanges necessary for
   input into the metrics For example, it MAY utilize the first INVITE
   or set of INVITE messages sent and the first accepted 200 OK.  Tags
   will identify this dialog as distinct from the other 200 OK
   responses, which are acknowledged and an immediate BYE is sent.  The
   application responsible for capturing and/or understanding the input
   values MUST utilize these tags to distinguish between dialog
   requests.

   Note that if an INVITE is forked before reaching its destination,
   multiple early dialogs are likely and multiple confirmed dialogs are
   possible (though unlikely).  When this occurs, an SRD measurement
   should be taken for each dialog that is created (early or confirmed).

5.5.  Data Collection

   The input necessary for these calculations may be collected in a
   number of different manners.  It may be collected or retrieved from
   call detail records (CDR) or raw signaling information generated by a
   proxy or UA.  When using records, time synchronization MUST be
   considered between applicable elements.

   The information may also be transmitted through the use of network
   management protocols like Simple Network Management Protocol (SNMP)
   and via future extensions to the SIP Management Information Base
   (MIB) modules [RFC4780], or through a potential undefined new
   performance metric event package [RFC3265] retrieved via SUBSCRIBE
   requests.

   Data may be collected for a sample of calls or all calls, and may
   also be derived from test call scenarios.  These metrics are flexible
   based on the needs of the application.

5.6.  Testing Documentation

   In some cases, these metrics will be used to provide output values to
   signify the performance level of a specific SIP-based element.  When
   using these metrics in a test environment, the environment MUST be
   accurately documented for the purposes of replicating any output
   values in future testing and/or validation.


6.  Conclusions

   The proposed guideline provides a description of common performance
   metrics, and their defined use with SIP.  The use of these metrics
   will provide a common viewpoint across all vendors, service



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   providers, and users.  These metrics will likely be utilized in
   production SIP environments for providing input regarding Key
   Performance Indicators (KPI) and Service Level Agreement (SLA)
   indications; however, they may also be used for testing end-to-end
   SIP-based service environments.


7.  IANA Considerations

   There are no IANA considerations at this time.


8.  Security Considerations

   Security SHOULD be considered in the aspect of securing the relative
   data utilized in providing input to the above calculations.  All
   other aspects of security SHOULD be considered as described in RFC
   3261 [RFC3261].


9.  Contributors

   The following people made substantial contributions to this work:


      Carol Davids         Illinois Institute of Technology
      Marian Delkinov      Ericsson
      Adam Uzelac          Global Crossing
      Jean-Francois Mule   CableLabs
      Rich Terpstra        Level 3 Communications



10.  Acknowledgements

   We would like to thank John Hearty and Dean Bayless for their efforts
   in reviewing the draft and providing insight regarding clarification
   of certain aspects described throughout the draft.  We also thank Dan
   Romascanu for his insightful comments and Vijay Gurbani for agreeing
   to perform the role of document shepherd.


11.  References

11.1.  Normative References

   [E.411]    ITU-T, "Series E: Overall Network Operation, Telephone
              Service, Service Operation and Human Factors", E.411 ,



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              March 2000.

   [E.721]    ITU-T, "Series E: Overall Network Operation, Telephone
              Service, Service Operation and Human Factors", E.411 ,
              May 1999.

   [GR-512]   Telcordia, "LSSGR: Reliability, Section 12", GR-512-
              CORE Issue 2, January 1998.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [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.

   [RFC3262]  Rosenberg, J. and H. Schulzrinne, "Reliability of
              Provisional Responses in Session Initiation Protocol
              (SIP)", RFC 3262, June 2002.

   [RFC3265]  Roach, A., "Session Initiation Protocol (SIP)-Specific
              Event Notification", RFC 3265, June 2002.

   [RFC3326]  Schulzrinne, H., Oran, D., and G. Camarillo, "The Reason
              Header Field for the Session Initiation Protocol (SIP)",
              RFC 3326, December 2002.

   [RFC3428]  Campbell, B., Rosenberg, J., Schulzrinne, H., Huitema, C.,
              and D. Gurle, "Session Initiation Protocol (SIP) Extension
              for Instant Messaging", RFC 3428, December 2002.

   [RFC3665]  Johnston, A., Donovan, S., Sparks, R., Cunningham, C., and
              K. Summers, "Session Initiation Protocol (SIP) Basic Call
              Flow Examples", BCP 75, RFC 3665, December 2003.

   [RFC4780]  Lingle, K., Mule, J-F., Maeng, J., and D. Walker,
              "Management Information Base for the Session Initiation
              Protocol (SIP)", RFC 4780, April 2007.

11.2.  Informative References

   [RFC2330]  Paxson, V., Almes, G., Mahdavi, J., and M. Mathis,
              "Framework for IP Performance Metrics", RFC 2330,
              May 1998.

   [RFC3398]  Camarillo, G., Roach, A., Peterson, J., and L. Ong,
              "Integrated Services Digital Network (ISDN) User Part



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              (ISUP) to Session Initiation Protocol (SIP) Mapping",
              RFC 3398, December 2002.


Authors' Addresses

   Daryl Malas
   CableLabs
   858 Coal Creek Circle
   Louisville, CO  80027
   US

   Phone: +1 303 661 3302
   Email: d.malas@cablelabs.com


   Al Morton
   AT&T Labs
   200 Laurel Avenue South
   Middletown, NJ  07748
   US

   Phone: +1 732 420 1571
   Email: acmorton@att.com



























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