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Versions: (draft-johnston-sipping-rtcp-summary) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 RFC 6035

SIPPING Working Group                                      A. Pendleton
Internet Draft                                                   Nortel
                                                               A. Clark
                                                              Telchemy
                                                           A. Johnston
                                                                 Avaya
                                                          H. Sinnreich
                                                                 Adobe
Intended status: Informational                            July 12, 2008
Expires: January 2008



      Session Initiation Protocol Package for Voice Quality Reporting
                                   Event


                    draft-ietf-sipping-rtcp-summary-03


Status of this Memo

   By submitting this Internet-Draft, each author represents that
   any applicable patent or other IPR claims of which he or she is
   aware have been or will be disclosed, and any of which he or she
   becomes aware will be disclosed, in accordance with Section 6 of
   BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   This Internet-Draft will expire on January 2009.

Copyright Notice




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   Copyright (C) The IETF Trust (2008).

Abstract

   This document defines a SIP event package that enables the collection
   and reporting of metrics that measure the quality for Voice over
   Internet Protocol (VoIP) sessions.

Table of Contents

   1. Introduction...................................................3
      1.1. Usage Scenarios...........................................3
   2. Terminology....................................................5
   3. SIP Events for Voice over IP Performance Reporting.............5
      3.1. SUBSCRIBE/NOTIFY method...................................5
      3.2. PUBLISH method............................................6
      3.3. Multi-Party and Multi-Segment Calls.......................6
      3.4. Overload Avoidance........................................6
   4. Event Package Formal Definition................................7
      4.1. Event Package Name........................................7
      4.2. Event Package Parameters..................................7
      4.3. SUBSCRIBE Bodies..........................................7
      4.4. Subscription Duration.....................................7
      4.5. NOTIFY and PUBLISH Bodies.................................7
      4.6. Voice Quality Event Syntax and Semantics..................8
         4.6.1. ABNF Syntax Definition...............................8
         4.6.2. Parameter Definitions and Mappings..................23
      4.7. Message Flow and Syntax Examples.........................31
         4.7.1. End of Session Report using NOTIFY..................31
         4.7.2. Mid Session Threshold Violation using NOTIFY........35
         4.7.3. End of Session Report using PUBLISH.................40
         4.7.4. Alert Report using PUBLISH..........................44
      4.8. Configuration Dataset for vq-rtcpxr Events...............47
   5. IANA Considerations...........................................49
      5.1. SIP Event Package Registration...........................50
      5.2. application/vq-rtcp-xr MIME Registration.................50
   6. Security Considerations.......................................51
   7. Contributors..................................................51
   8. Normative References..........................................51
   Author's Addresses...............................................52
   Intellectual Property Statement..................................53
   Disclaimer of Validity...........................................53







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1. Introduction

   Real time communications over IP networks use SIP for signaling with
   RTP/RTCP for media transport and reporting respectively. These
   protocols are very flexible and can support an extremely wide
   spectrum of usage scenarios. For this reason, extensions to these
   protocols must be specified in the context of a specific usage
   scenario. In this memo, extensions to SIP are proposed to support the
   reporting of RTCP XR VoIP metrics (RFC3611).

1.1. Usage Scenarios

   The usage scenarios addressed in this memo are situations where a SIP
   user agent can easily report the voice quality since it communicates
   with a small number of other endpoints:

   1. Point-to-point VoIP conversations. These can include small
      telephony type multiparty scenarios, such as when using call
      transfer.

   2. Conference calls using a central conferencing server when each
      SIP can report on the quality of their leg to the central
      conferencing server.

   3. Multicast VoIP calls using source specific multicast (SSM). This
      is somewhat similar to the central conferencing scenario.

   Distributed conferences with audio mixing in the endpoints may
   require reporting on too many call legs and may be therefore less
   practical if there are more than 3-4 participants.

   The usage scenarios 1, 2, and 3 provide voice quality reports that
   are most closely related to the user experience, since the reporting
   application resides in the endpoints, such as in SIP UAs (UA). Many
   SIP UAs however may have limitations as to the footprint of the
   software and as a result frugal reporting capabilities are
   preferable.

   RTCP reports are usually sent to other participating endpoints in a
   session which can make collection of performance information by
   administration or management systems more complex. In the usage
   scenarios addressed in this memo, the data contained in RTCP XR VoIP
   metrics reports (RFC3611) are forwarded to central collection server
   systems using SIP.

   Applications residing in the server or elsewhere can aid in network
   management to alleviate bandwidth constraints and also to support


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   customer service by identifying and acknowledging calls of poor
   quality. Specifying such applications are however beyond the scope of
   this paper.

   Keeping it Simple

   There are a large portfolio of quality parameters that can be
   associated with VoIP, but only a minimal necessary number of
   parameters are included on the RTCP-XR reports:

       1. The codec type, as resulting from the SDP offer-answer
          negotiation in SIP,

       2. The burst gap loss density and max gap duration, since voice
          cut-outs are the most annoying quality impairment in VoIP,

       3. Round trip delay because it is critical to conversational
          quality,

       4. Conversational quality as a catch-all for other voice quality
          impairments, such as random distributed packet loss, jitter,
          annoying silent suppression effects, etc.

   In specific usage scenarios where other parameters are required,
   designers can include other parameters beyond the scope of this
   paper.

   RTCP reports are best effort only, and though very useful have a
   number of limitations as discussed in [4]. This must be considered
   when using RTCP reports in managed networks.

   This document defines a new SIP event package, vq-rtcpxr, and a new
   MIME type, application/vq-rtcpxr, that enable the collection and
   reporting of metrics that measure quality for RTP [3] sessions. The
   definitions of the metrics used in the event package are based on
   RTCP Extended Reports [4] and RTCP [3]; a mapping between the SIP
   event parameters and the parameters within the forementioned RFC's
   is defined within this document in section 4.6.2.

   Monitoring of voice quality is believed to be the highest priority
   for usage of this mechanism and as such, the metrics in the event
   package are largely tailored for voice quality measurements. The
   event package is designed to be extensible. However the negotiation
   of such extensions is not defined in this document.





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   The event package supports reporting both the voice quality metrics
   for both inbound and outbound directions.  Voice quality metrics for
   the inbound direction can generally be computed locally by the
   reporting endpoint however voice quality metrics for the outbound
   direction are computed by the remote endpoint and sent to the
   reporting endpoint using the RTCP Extended Reports [4].

   Configuration of usage of the event package is not covered in this
   document. It is the recommendation of this document that the SIP
   configuration framework [8] be used.  The authors have defined a
   configuration dataset that would facilitate this support in section
   5.8.

   The event package SHOULD be used with the SUBSCRIBE/NOTIFY method
   however MAY be used with the PUBLISH method for backward
   compatibility with some existing implementations. Message flow
   examples for both methods are provided in this document.

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
   RFC 2119 [1].


3. SIP Events for Voice over IP Performance Reporting

      This document defines a SIP events package [5] for Voice over IP
   performance reporting. A SIP UA can send these events to an entity
   which can make the information available to other applications. For
   purposes of illustration, the entities involved in SIP vq-rtcpxr
   event reporting will be referred to as follows:

   o REPORTER is an entity involved in the measurement and reporting
     of media quality i.e. the SIP UA involved in a media session.

   o COLLECTOR is an entity that receives SIP vq-rtcpxr events. A
     COLLECTOR may be a proxy server or another entity that is capable
     of supporting SIP vq-rtcpxr events.

3.1. SUBSCRIBE/NOTIFY method

   The REPORTER SHOULD send the voice quality metric reports using the
   NOTIFY method.  The COLLECTOR SHALL send a SUBSCRIBE to the REPORTER
   to explicitly establish the relationship. Configuration of an Address
   of Resource (AOR) of the COLLECTOR is not needed.


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   The REPORTER MUST NOT send any vq-rtcpxr events if a COLLECTOR AOR
   has not been configured.

   The REPORTER SHALL populate the Request-URI of the NOTIFY method
   with the address of the resource (AOR) of the COLLECTOR.

   The COLLECTOR MAY send a SUBSCRIBE to a SIP Proxy acting on behalf
   ofthe reporting SIP UA's.

3.2. PUBLISH method

   A SIP UA that supports this specification MAY also send the service
   quality metric reports using the PUBLISH method, however this
   approach SHOULD NOT be used in unmanaged Internet services. The
   Publish method MAY be supported for backward compatibility with
   existing implementations.

   The REPORTER MAY therefore populate the Request-URI of the PUBLISH
   method with the AOR of the COLLECTOR. To ensure security of SIP
   proxies and the COLLECTOR, the REPORTER must be configured with the
   AOR of the COLLECTOR, preferably using the SIP UA configuration
   framework [8], as described in section 5.8.

   It is recommended that the REPORTER send an OPTIONS message to the
   COLLECTOR to ensure support of the PUBLISH message.

3.3. Multi-Party and Multi-Segment Calls

   A voice quality metric report may be sent for each session
   terminating at the REPORTER and may contain multiple report bodies.
   For a multi-party call the report MAY contain report bodies for the
   session between the reporting endpoint and each remote endpoint for
   which there was an RTP session during the call.

   Supplementary services such as call hold and call transfer can result
   in the user participating in a series of concatenated sessions,
   potentially with different choices of codec or sample rate, although
   these may be perceived by the user as a single call.  A REPORTER MAY
   send a voice quality metric report at the end of each session or MAY
   send a single voice quality metric report containing a report body
   for each segment of the call.

3.4. Overload Avoidance

   To avoid overload of SIP Proxies or COLLECTORS it is important to do
   capacity planning and to minimize the number of reports that are
   sent.


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      Approaches to avoiding overload include:

      a. Send only one report at the end of each call

      b. Use interval reports only on "problem" calls that are being
        closely monitored

   Limit the number of alerts that can be sent to a maximum of one per
   call

   Additionally, it is recommended that COLLECTORS that receive these
   reports use the 503 ''Service Unavailable'' error response code to
   limit unwanted reports and include the Retry-after header with an
   appropriate time delay, depending on the needs of the COLLECTOR.

4. Event Package Formal Definition

4.1. Event Package Name

   This document defines a SIP Event Package as defined in RFC 3265 [5].

4.2. Event Package Parameters

   No event package parameters are defined.

4.3. SUBSCRIBE Bodies

   No SUBSCRIBE bodies are described by this specification.

4.4. Subscription Duration

   Subscriptions to this event package MAY range from minutes to weeks.
   Subscriptions in hours or days are more typical and are RECOMMENDED.
   The default subscription duration for this event package is one hour.

4.5. NOTIFY and PUBLISH Bodies

   There are three notify bodies: a session report, an interval session
   report, and a alert report.

   The session report is used for end of session reporting. This can be
   generated when a voice media session terminates or when a media
   change occurs, such as a codec change or a session forks. This report
   is intended to allow cumulative metric reporting. The session reports
   will populate the metrics with values that are measured over the
   interval explicitly defined by the "start" and "stop" timestamps.



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   The interval report is used for periodic or interval reporting. This
   report is intended to capture short duration metric reporting.
   Interval reports will populate the metrics with values that are
   measured over the interval explicitly defined by the "start" and
   "stop" timestamps.

   The alert report is used when voice quality degrades during a
   session.  The session report parameters are also included in the
   alert report to provide all of the necessary diagnostic information.
   Like the interval report, the metrics in the alert reports will be
   populated with values that are measured over the interval explicitly
   defined by the "start" and "stop" timestamps.

   This specification defines a new MIME type application/vq-rtcpxr
   which is a text encoding of the RTCP and RTCP-XR statistics, along
   with some additional metrics and correlation information.

4.6. Voice Quality Event Syntax and Semantics

   This section describes the syntax extensions required for event
   publication in SIP. The formal syntax definitions described in this
   section are expressed in the Augmented BNF [6] format used in SIP
   [2], and contains references to elements defined therein.
   Additionally, the definition of the timestamp format is provided in
   [7]. Note that most of the parameters are optional. In practice, most
   implementations will send a subset of the parameters. It is not the
   intention of this document to define what parameters may or may not
   be useful for monitoring the quality of a voice session, but to
   enable reporting of voice quality.  As such, the syntax allows the
   implementer to choose which metrics are most appropriate for their
   solution.  As there are no "invalid", "unknown", or "not applicable"
   values in the syntax, the intention is to exclude any parameters for
   which values are not available, not applicable, or unknown.

   Additionally, the authors recognize that implementers may need to add
   new parameter lines to the reports and new metrics to the existing
   parameter lines. The extension tokens are intended to fulfill this
   need.

4.6.1.  ABNF Syntax Definition

      VQReportEvent  =  AlertReport /  SessionReport / IntervalReport







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      SessionReport = "VQSessionReport" CRLF

                  LocalMetrics [CRLF RemoteMetrics]

                  [DialogID]

      IntervalReport = "VQIntervalReport" CRLF

                  LocalMetrics [CRLF RemoteMetrics]

                  [DialogID]

      LocalMetrics  = "LocalMetrics" COLON CRLF Metrics

      RemoteMetrics = "RemoteMetrics" COLON CRLF Metrics

      AlertReport   = "VQAlertReport" COLON

            MetricType WSP Severity WSP Direction CRLF

            "Metrics:" CRLF

            Metrics

            [CRLF "RemoteMetrics:" CRLF Metrics]

            [DialogID]

      Metrics = TimeStamps CRLF

         [SessionDescription CRLF]

         CallID CRLF

         FromID CRLF

         ToID CRLF

         LocalAddr CRLF

         RemoteAddr CRLF

         [JitterBuffer CRLF]

         [PacketLoss CRLF]

         [BurstGapLoss CRLF]


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         [Delay CRLF]

         [Signal CRLF]

         [QualityEstimates CRLF]

         *(Extension CRLF)

      ; Timestamps are provided in Coordinated Universal Time (UTC)

      ; using the ABNF format provided in RFC3339,

      ;  "Date and Time on the Internet: Timestamps"

      ; These timestamps should reflect, as closely as

      ; possible, the actual time during which the media session

      ; was running to enable correlation to events occurring

      ; in the network infrastructure and to accounting or billing

      ; records

      TimeStamps = "Timestamps" COLON StartTime WSP StopTime

      StartTime  = "START" EQUAL date-time

      StopTime   = "STOP" EQUAL date-time

      ; SessionDescription provides a shortened version of the

      ; session SDP but contains only the relevant parameters for

      ; session quality reporting purposes

      SessionDescription  = "SessionDesc" COLON

         [PayloadType WSP]

         [PayloadDesc WSP]

         [SampleRate WSP]

         [FrameDuration WSP]

         [FrameOctets WSP]


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         [FramesPerPacket WSP]

         [PacketsPerSecond WSP]

         [FmtpOptions WSP]

         [PacketLossConcealment WSP]

         [SilenceSuppressionState]

         *(WSP Extension)

      ; PayloadType provides the PT parameter used in the RTP packets

      ; i.e. the codec used for decoding received RTP packets

      ; It is recommended that IANA registered values are used

      ; where possible.

      PayloadType  = "PT" EQUAL (1*3DIGIT)

      ; PayloadDesc provides a text description of the codec

      ; It is recommended that the abbreviated Standard name for

      ; the codec is used (for example "G.729A")

      PayloadDesc  = "PD" EQUAL (word / DQUOTE word-plus DQUOTE)

      ; SampleRate provides the rate at which voice was sampled

      ; in the case of narrowband codecs, the value will typically be
   8000.

      ; For codecs that are able to change sample rates the lowest and

      ; highest sample rates shall be reported (e.g. 8000;16000).

      SampleRate = "SR" EQUAL (1*5DIGIT) *(SEMI (1*5DIGIT))

      ; FrameDuration can be combined with the FramesPerPacket to
   determine

      ; the packetization rate; the units for this are milliseconds.

      FrameDuration = "FD" EQUAL (1*3DIGIT)


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      ; FrameOctets provides the number of octets in each frame

      ; Used where FrameDuration is not available

      FrameOctets  = "FO" EQUAL (1*4DIGIT)

      ; FramesPerPacket provides the number of frames in each RTP packet

      FramesPerPacket = "FPP" EQUAL (1*2DIGIT)

      ; Packets per second provides the number of packets, including one
   or

      ; more frames within each, that are transmitted per second

      PacketsPerSecond = "PPS" EQUAL (1*5DIGIT)

      ; FMTP options from SDP.  Note that the parameter is deliniated

      ; by " " to avoid parsing issues in transitioning between SDP and

      ; SIP parsing

      FmtpOptions = "FMTP" EQUAL DQUOTE word-plus DQUOTE

      ; PacketLossConcealment indicates whether a PLC algorithm was

      ; or is being used for the session.  The values follow the same

      ; numbering convention as RFC 3611.  For more details,

      ; please refer to RFC 3611, RTCP XR

      ; 0 - unspecified

      ; 1 - disabled

      ; 2 - enhanced

      ; 3 - standard

      PacketLossConcealment  = "PLC" EQUAL ("0" / "1" / "2" / "3")

      ; SilenceSuppressionState indicates whether silence suppression,

      ; also known as Voice Activity Detection (VAD) is enabled.



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      SilenceSuppressionState  = "SSUP" EQUAL ("on" / "off")

      ; CallId provides the call id from the SIP header

      CallID  =  "CallID" COLON Call-ID-Parm

      ; FromID provides the identification of the reporting endpoint

      ; of the media session [2].

      FromID = "FromID" COLON from-spec

      ; ToID provides the identification of the remote endpoint

      ; of the media session [2].

      ToID = "ToID" COLON (name-addr/addr-spec)

      ; LocalAddr provides the IP address, port and ssrc for the

      ; session from the perspective of the endpoint/UA which is

      ; generating the metrics.

      LocalAddr   = "LocalAddr" COLON IPAddress WSP Port WSP Ssrc

      ; RemoteAddr provides the IP address, port and ssrc for the

      ; session from the perspective of the peer of the endpoint/UA

      ; that is generating the metrics.

      RemoteAddr  = "RemoteAddr" COLON IPAddress WSP Port WSP Ssrc

      ; For clarification, the LocalAddr in the LocalMetrics report

      ; would be the RemoteAddr in the RemoteMetrics report.

      IPAddress   = "IP" EQUAL IPv6address / IPv4address

      Port        = "PORT" EQUAL 1*DIGIT

      Ssrc        = "SSRC" EQUAL ( ''0x'' 1*8HEXDIG)

      JitterBuffer = "JitterBuffer" COLON

         [JitterBufferAdaptive WSP]


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         [JitterBufferRate WSP]

         [JitterBufferNominal WSP]

         [JitterBufferMax WSP]

         [JitterBufferAbsMax]

         *(WSP Extension)

      ; JitterBufferAdaptive indicates whether the jitter buffer in the

      ; endpoint is adaptive, static, or unknown.

      ; The values follow the same numbering convention as RFC 3611.

      ; For more details, please refer to that document.

      ; 0 - unknown

      ; 1 - reserved

      ; 2 - non-adaptive

      ; 3 - adaptive

      JitterBufferAdaptive  = "JBA" EQUAL ("0" / "1" / "2" / "3")

      ; JitterBuffer metric definitions are provided in RTCP XR, RFC
   3611

      JitterBufferRate      = "JBR" EQUAL (1*2DIGIT) ;0-15

      JitterBufferNominal   = "JBN" EQUAL (1*5DIGIT) ;0-65535

      JitterBufferMax       = "JBM" EQUAL (1*5DIGIT) ;0-65535

      JitterBufferAbsMax    = "JBX" EQUAL (1*5DIGIT) ;0-65535

      ; PacketLoss metric definitions are provided in RTCP XR, RFC 3611

      PacketLoss = "PacketLoss" COLON

                 [NetworkPacketLossRate WSP]

                 [JitterBufferDiscardRate]



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                 *(WSP Extension)

      NetworkPacketLossRate =

        "NLR" EQUAL (1*3DIGIT ["." 1*2DIGIT]) ;percentage

      JitterBufferDiscardRate =

        "JDR" EQUAL (1*3DIGIT ["." 1*2DIGIT]) ;percentage

      ; BurstGapLoss metric definitions are provided in RTCP XR, RFC
   3611

      BurstGapLoss = "BurstGapLoss" COLON

         [BurstLossDensity WSP]

         [BurstDuration WSP]

         [GapLossDensity WSP]

         [GapDuration WSP]

         [MinimumGapThreshold]

         *(WSP Extension)

      BurstLossDensity =

       "BLD" EQUAL (1*3DIGIT ["." 1*2DIGIT]) ;percentage

      BurstDuration =

       "BD" EQUAL (1*7DIGIT) ;0-3,600,000 -- milliseconds

      GapLossDensity =

       "GLD" EQUAL (1*3DIGIT ["." 1*2DIGIT]) ;percentage

      GapDuration =

       "GD" EQUAL (1*7DIGIT) ;0-3,600,000 -- milliseconds

      MinimumGapThreshold =

       "GMIN" EQUAL (1*3DIGIT) ;1-255



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      Delay = "Delay" COLON

         [RoundTripDelay WSP]

         [EndSystemDelay WSP]

         [OneWayDelay WSP]

         [SymmOneWayDelay WSP]

         [InterarrivalJitter WSP]

         [MeanAbsoluteJitter]

         *(WSP Extension)

      ; RoundTripDelay is recommended to be measured as defined in

      ; RTCP, RFC 3550.

      RoundTripDelay = "RTD" EQUAL (1*5DIGIT) ;0-65535

      ; EndSystemDelay metric is defined in RTCP XR, RFC 3611

      EndSystemDelay = "ESD" EQUAL (1*5DIGIT) ;0-65535

      ; OneWayDelay is defined in RFC2679

      OneWayDelay = "OWD" EQUAL (1*5DIGIT) ;0-65535

      ; SymmOneWayDelay is defined as half the sum of RoundTripDelay

      ; and the EndSystemDelay values for both endpoints.

      SymmOneWayDelay = ''SOWD'' EQUAL (1*5DIGIT); 0-65535

      ; Interarrival Jitter is measured as defined RFC 3550

      InterarrivalJitter = "IAJ" EQUAL (1*5DIGIT) ;0-65535

      ; Mean Absolute Jitter is measured as defined

      ; by ITU-T G.1020 where it is known as MAPDV

      MeanAbsoluteJitter = "MAJ" EQUAL (1*5DIGIT);0-65535




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      ; Signal metrics definitions are provided in RTCP XR, RFC 3611

      Signal = "Signal" COLON

         [SignalLevel WSP]

         [NoiseLevel WSP]

         [ResidualEchoReturnLoss]

         *(WSP Extension)

      ; SignalLevel will normally be a negative value

      ; the absence of the negative sign indicates a positive value

      ; where the signal level is negative, the sign must be included.

      ; This metric applies to the speech signal decoded from the

      ; received packet stream.

      SignalLevel = "SL" EQUAL (["-"] 1*2DIGIT)

      ; NoiseLevel will normally be negative but to align with the

      ; the encoding of SignalLevel, the sign must be explicitly
   included.

      ; The absence of a sign indicates a positive value

      ; This metric applies to the speech signal decoded from the

      ; received packet stream.

      NoiseLevel  = "NL" EQUAL (["-"] 1*2DIGIT)

      ; Residual Echo Return Loss (RERL) the ratio between

      ; the original signal and the echo level as measured after

      ; echo cancellation or suppression has been applied.

      ; Expressed in decibels (dB). This is positive value, echo

      ; return loss levels less than or equal to 0 dB shall be indicated
   as 0 dB.


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      ; This metric relates to the proportion of the speech signal

      ; decoded from the received packet stream that is reflected

      ; back in the encoded speech signal output in the transmitted

      ; packet stream.

      ResidualEchoReturnLoss = "RERL" EQUAL (1*3DIGIT)

      ; Voice Quality estimation metrics

      ; The definition of these metrics are provided in RTCP XR and

      ; the new High Resolution proposal, RTCP HR.

      ; Each quality estmiate has an optional associated algorithm.

      ; These fields permit the implementation to use a variety

      ; of different calculation methods for each type of metric

      QualityEstimates  = "QualityEst" COLON

         [ListeningQualityR WSP]

         [RLQEstAlg WSP]

         [ConversationalQualityR WSP]

         [RCQEstAlg WSP]

         [ExternalR-In WSP]

         [ExtRInEstAlg WSP]

         [ExternalR-Out WSP]

         [ExtROutEstAlg WSP]

         [MOS-LQ WSP]

         [MOSLQEstAlg WSP]

         [MOS-CQ WSP]

         [MOSCQEstAlg WSP]


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         [QoEEstAlg]

         *(WSP Extension)

      ListeningQualityR = "RLQ" EQUAL (1*3DIGIT) ; 0 - 120

      RLQEstAlg = "RLQEstAlg" EQUAL word ; "P.564", or other

      ConversationalQualityR = "RCQ" EQUAL (1*3DIGIT) ; 0 - 120

      RCQEstAlg = "RCQEstAlg" EQUAL word ; "P.564", or other

      ; ExternalR-In is measured by the local endpoint for incoming

      ; connection on "other" side of this endpoint

      ;   e.g. PhoneA <---> Bridge <----> Phone B

      ;   ListeningQualityR = quality for PhoneA ----> Bridge path

      ;   ExternalR-In = quality for Bridge <---- PhoneB path

      ExternalR-In = "EXTRI" EQUAL (1*3DIGIT) ; 0 - 120

      ExtRInEstAlg = "ExtRIEstAlg" EQUAL word ; "P.564" or other

      ; ExternalR-Out is copied from RTCP XR message received from the

      ; remote endpoint on "other" side of this endpoint

      ;   e.g. PhoneA <---> Bridge <----> Phone B

      ;   ExternalR-Out = quality for Bridge -----> PhoneB path

      ExternalR-Out = "EXTRO" EQUAL (1*3DIGIT) ; 0 - 120

      ExtROutEstAlg = "ExtROEstAlg" EQUAL word ; "P.564" or other

      MOS-LQ = "MOSLQ" EQUAL (DIGIT ["." 1*3DIGIT]) ; 0.0 - 4.9

      MOSLQEstAlg = "MOSLQEstAlg" EQUAL word ; "P.564" or other

      MOS-CQ = "MOSCQ" EQUAL (DIGIT ["." 1*3DIGIT])  ; 0.0 - 4.9

      MOSCQEstAlg = "MOSCQEstAlg" EQUAL word ; "P.564" or other




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      ; alternative to the separate estimation algorithms

      ; for use when the same algorithm is used for all measurements

      QoEEstAlg = "QoEEstAlg" EQUAL word; "P.564" or other

      ; DialogID provides the identification of the dialog with

      ; which the media session is related.  This value is taken

      ; from the SIP header.

      DialogID  = "DialogID" COLON Call-ID-Parm *(SEMI did-parm)

      did-parm  = to-tag / from-tag / word

      to-tag    = "to-tag" EQUAL token

      from-tag  = "from-tag" EQUAL token

      ; MetricType provides the metric on which a notification of

      ; threshold violation was based.  The more commonly used metrics

      ; for alerting purposes are included here explicitly and the

      ; token parameter allows for extension

      MetricType = "Type" EQUAL "RLQ" / "RCQ" / "EXTR" /

         "MOSLQ" / "MOSCQ" /

         "BD" / "NLR" / "JDR" /

         "RTD" / "ESD" / "IAJ" /

         "RERL" / ''SL'' / ''NL / Extension

      Direction = "Dir" EQUAL "local" / "remote"

      Severity  = "Severity" EQUAL "Warning" / "Critical" /

         "Clear"

      Call-ID-Parm =  word [ "@" word ]




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      ; miscellaneous needs for ABNF

      ; some of these are pulled out of RFC2234 or RFC3261

      ; where this is the case, it is noted.

      ; taken from RFC2234

      CRLF =  %x0D.0A

      DIGIT =  %x30-39

      WSP   =  SP / HTAB ; white space

      SP    =  " "

      HTAB  =  %x09 ; horizontal tab

      HEXDIG =  DIGIT / "A" / "B" / "C" / "D" / "E" / "F" /

                   "a" / "b" / "c" / "d" / "e" / "f"

      DQUOTE  =  %x22 ; " (Double Quote)

      ALPHA   =  %x41-5A / %x61-7A   ; A-Z / a-z

      ; taken from RFC3261

      alphanum  =  ALPHA / DIGIT

      LWS  =  [*WSP CRLF] 1*WSP ; linear whitespace

      SWS  =  [LWS] ; sep whitespace

      SEMI =  SWS ";" SWS ; semicolon

      EQUAL   =  SWS "=" SWS ; equal

      COLON   =  SWS ":" SWS ; colon

      token       =  1*(alphanum / "-" / "." / "!" / "%" / "*"

                        / "_" / "+" / "`" / "'" / "~" )



      IPv4address    =  1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT


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      IPv6address    =  hexpart [ ":" IPv4address ]

      hexpart        =  hexseq / hexseq "::" [ hexseq ] / "::" [ hexseq
   ]

      hexseq         =  hex4 *( ":" hex4)

      hex4           =  1*4HEXDIG

      ; DATE-TIME format

      ; taken from RFC3339, refer for more information

      date-fullyear   = 4DIGIT ; e.g. 2006

      date-month      = 2DIGIT ; e.g. 01 or 11

      date-mday       = 2DIGIT ; e.g. 02 or 22

      time-hour       = 2DIGIT ; e.g. 01 or 13

      time-minute     = 2DIGIT ; e.g. 03 or 55

      time-second     = 2DIGIT ; e.g. 01 or 59

      time-secfrac    = "." 1*DIGIT

      time-numoffset  = ("+" / "-") time-hour ":" time-minute

      time-offset     = "Z" / time-numoffset

      partial-time = time-hour ":" time-minute ":" time-second

   [time-secfrac]

      full-date    = date-fullyear "-" date-month "-" date-mday

      full-time    = partial-time time-offset

      date-time       = full-date "T" full-time

      ;

      ; Miscellaneous definitions for the syntax

      ;



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      Extension = word-plus

      word  =  1*(alphanum / "-" / "." / "!" / "%" / "*" /

         "_" / "+" / "`" / "'" / "~" /

         "(" / ")" / "<" / ">" /

         ":" / "\" / DQUOTE /

         "/" / "[" / "]" / "?" )

      word-plus =  1*(alphanum /  "-"  /  "."  /  "!"  / "%" / "*" /

         "_"  /  "+"  /  "`"  /  "'"  /  "~"  /

         "("  /  ")"  /  "<"  /  ">"  /  ":"  /

         "\"  /  "/"  /  "["  /  "]"  /  "?"  /

         "{"  /  "}"  /  "="  /  " ")



4.6.2. Parameter Definitions and Mappings

4.6.2.1. General mapping percentages from 8 bit, fixed point numbers

   RFC3611 uses an 8 bit, fixed point number with the binary point at
   the left edge of the field.  This value is calculated by dividing the
   total number of packets lost by the total number of packets expected
   and multiplying the result by 256, and taking the integer part.

   For any RTCP XR parameter in this format, to map into the equivalent
   SIP vq-rtcpxr parameter, simply reverse the equation i.e. divide by
   256 and taking the integer part.

4.6.2.2. Timestamps

   Following SIP and other IETF convention, timestamps are provided in
   Coordinated Universal Time (UTC) using the ABNF format provided in
   IETF RFC3339 [7]. These timestamps should reflect, as closely
   aspossible, the actual time during which the media session was




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   running to enable correlation to related events occurring in the
   network and to accounting or billing records.



4.6.2.3. SessionDescription

   The parameters in this field provide a shortened version of the
   session SDP(s), containing only the relevant parameters for session
   quality reporting purposes.

      Payload Type

   This is the "payload type" parameter used in the RTP packets i.e. the
   codec. This field can also be mapped from the SDP "rtpmap" attribute
   field "payload type". IANA registered types SHOULD be used.

      Payload Desc

   This parameter provides a text name for the codec used in this
   session.

      Sample Rate

   This parameter is mapped from the SDP "rtpmap"  attribute field
   "clock rate". The field provides the rate at which voice was sampled,
   measured in Hertz (Hz).

      Frame Duration

   This parameter is not contained in RTP or SDP but can usually be
   obtained from the device codec. The field reflects the amount of
   voice content in each frame within the RTP payload, measured in
   milliseconds. Note this value can be combined with the
   FramesPerPacket to determine the packetization rate.

      Frame Octets

   This parameter is not contained in RTP or SDP but is usually provided
   by the device codec. The field provides the number of octets in each
   frame within the RTP payload. This field is usually not provided when
   FrameDuration is provided.

      Frames Per Packet

   This parameter is not contained in RTP or SDP but can usually be
   obtained from the device codec.  This field provides the number of


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   frames in each RTP packet. Note this value can be combined with the
   FrameDuration to determine the packetization rate.

      Packets Per Second

   This parameter is not contained in RTP or SDP but can usually be
   obtained from the device codec. Packets per second provides the
   number of RTP packets that are transmitted per second.

      FMTP

   This parameter is taken directly from the SDP attribute "fmtp".



      Silence Suppression State

   This parameter does not correspond to SDP, RTP, or RTCP XR. It
   indicates whether silence suppression, also known as Voice Activity
   Detection (VAD) is enabled for the identified session.



      Packet Loss Concealment

   This value corresponds to "PLC" in RFC3611 in the VoIP Metrics Report
   Block. The values defined by RFC3611 are reused by this
   recommendation and therefore no mapping is required.

4.6.2.4. LocalAddr

   This field provides the IP address, port and synchronization source
   (SSRC) for the session from the perspective of the endpoint that is
   sending the event.  The IPAddress can be IPv4 or IPv6 format. The
   SSRC is taken from SDP, RTCP, or RTCP XR input paramters.

   In the presence of NAT, the MAPPED-ADDRESS as reported by the STUN
   [9] server (RFC 3489) MUST be reported, since the internal IP address
   is not visible to the network operator.

4.6.2.5. RemoteAddr

   This field provides the IP address, port and ssrc of the session peer
   from the perspective of the remote endpoint sending the event. In the
   presence of NAT, the MAPPED-ADDRESS as reported by the STUN [9]
   server (RFC 3489bis) MUST be reported, since the internal IP address
   is not visible to the network operator.


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4.6.2.6. Jitter Buffer Parameters

      Jitter Buffer Adaptive

   This value corresponds to "JBA" in RFC3611 in the VoIP Metrics Report
   Block. The values defined by RFC3611 are reused by this ecommendation
   and therefore no mapping is required.

      Jitter Buffer Rate

   This value corresponds to "JB rate" in RFC3611 in the VoIP Metrics
   Report Block. The parameter does not require any mapping alculations.

      Jitter Buffer Nominal

   This value corresponds to "JB nominal" in RFC3611 in the VoIP Metrics
   Report Block. The parameter does not require any mapping
   calculations.

      Jitter Buffer Max

   This value corresponds to "JB maximum" in RFC3611 in the VoIP Metrics
   Report Block. The parameter does not require any conversion.

      Jitter Buffer Abs Max

   This value corresponds to "JB abs max" in RFC3611 in the VoIP Metrics
   Report Block. The parameter does not require any conversion.

4.6.2.7. Packet Loss Parameters

      Network Packet Loss Rate

   This value corresponds to "loss rate" in RFC3611 in the VoIP Metrics
   Report Block. For conversion, see "General mapping percentages from 8
   bit, fixed point numbers".

      Jitter Buffer Discard Rate

   This value corresponds to "discard rate" in RFC3611 in the VoIP
   Metrics Report Block.  For conversion, see "General mapping
   percentages from 8 bit, fixed point numbers".







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4.6.2.8. Burst/Gap Parameters

      Burst Loss Density

   This value corresponds to "burst density" in RFC3611 in the VoIP
   Metrics Report Block. For conversion, see "General mapping
   percentages from 8 bit, fixed point numbers".

      Burst Duration

   This value corresponds to "burst duration" in RFC3611 in the VoIP
   Metrics Report Block. This value requires no conversion; the exact
   value sent in an RTCP XR VoIP Metrics Report Block can be included in
   the SIP vq-rtcpxr parameter.

      Gap Loss Density

   This value corresponds to "gap density" in RFC3611 in the VoIP
   metrics Report Block.

   See "General mapping percentages from 8 bit, fixed point numbers".

      Gap Duration

   This value corresponds to "gap duration" in RFC3611 in the VoIP
   Metrics Report Block.  This value requires no conversion; the exact
   value sent in an RTCP XR VoIP Metrics Report Block can be included in
   the SIP vq-rtcpxr parameter.

      Minimum Gap Threshold

   This value corresponds to "Gmin" in RFC3611 in the VoIP Metrics
   Report Block. This value requires no conversion; the exact value sent
   in an RTCP XR VoIP Metrics Report Block can be included in the SIP
   vq-rtcpxr parameter.

4.6.2.9. Delay Parameters

      Round Trip Delay

This value corresponds to "round trip delay" in RFC3611 in the VoIP
Metrics Report Block and may be measured using the method defined in
RFC3550. The parameter is expressed in milliseconds.

      End System Delay




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   This value corresponds to "end system delay" in RFC3611 in the
   VoIP Metrics Report Block. This parameter does not require any
   conversion. The parameter is expressed in milliseconds.

      One Way Delay

   This value SHOULD be measured using the methods defined in IETF
   RFC2679. The parameter is expressed in milliseconds.

      Interarrival Jitter

   Inter-arrival jitter is defined in RFC 3550. The parameter is
   expressed in milliseconds.

      Mean Absolute Jitter

   It is recommended that MAJ be measured as defined by ITU-T G.1020.
   This parameter is often referred to as MAPDV. The parameter is
   expressed in milliseconds.

4.6.2.10. Signal-related Parameters

    Signal Level

   This field corresponds to "signal level" in RFC3611 in the VoIP
   Metrics Report Block. This field provides the voice signal
   relative level is defined as the ratio of the signal level to a 0
   dBm0 reference, expressed in decibels. This value can be used
   directly without extra conversion.

      Noise Level

   This field corresponds to "noise level" in RFC3611 in the VoIP
   Metrics Report Block. This field provide the ratio of the silent
   period background noise level to a 0 dBm0 reference, expressed in
   secibels.  This value can be used directly without extra conversion.

      Residual Echo Return Loss (RERL)

   This field corresponds to "RERL" in RFC3611 in the VoIP Metrics
   Report Block.  This field provides the ratio between the original
   signal and the echo level in decibels, as measured after echo
   cancellation or suppression has been applied. This value can be used
   directly without extra conversion.





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4.6.2.11. Quality Scores

   ListeningQualityR

   This field reports the listening quality expressed as an R factor
   (per G.107).  This does not include the effects of echo or delay.
   The range of R is 0-95 for narrowband calls and 0-120 for wideband
   calls. Algorithms for computing this value SHOULD be compliant with
   ITU-T Recommendation P.564.

      RLQEstAlg

   This field provides a text name for the algorithm used to estimate
   ListeningQualityR however may report "P.564" for algorithms that have
   been tested against ITU-T Recommendation P.564.

      ConversationalQualityR

   This field corresponds to "R factor" in RFC3611 in the VoIP Metrics
   Report Block. This parameter provides a cumulative measurement of
   voice quality from the start of the session to the reporting time.
   The range of R is 0-95 for narrowband calls and 0-120 for wideband
   calls. Algorithms for computing this value SHOULD be compliant with
   ITU-T Recommendation P.564. Within RFC3611 a reported R factor of 127
   indicates that this parameter is unavailable; in this case the
   ConversationalQualityR parameter SHALL be omitted from the vq-rtcpxr
   event.

      RCQEstAlg

   This field provides a text name for the algorithm used to estimate
   ConversationalQualityR however may report "P.564" for algorithms that
   have been tested against ITU-T Recommendation P.564.

      ExternalR-In

   This field corresponds to "ext. R factor" in RFC3611 in the VoIP
   Metrics Report Block.  This parameter reflects voice quality as
   measured by the local endpoint for incoming connection on "other"
   side (refer to RFC3611 for a more detailed explanation). The range of
   R is 0-95 for narrowband calls and 0-120 for wideband calls.
   Algorithms for computing this value SHOULD be compliant with ITU-T
   Recommendation P.564. Within RFC3611 a reported R factor of 127
   indicates that this parameter is unavailable; in this case the
   ConversationalQualityR parameter SHALL be omitted from the vq-rtcpxr
   event.



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      ExtRInEstAlg

   This field provides a text name for the algorithm used to estimate
   ExternalR-In however may report "P.564" for algorithms that have been
   tested against ITU-T Recommendation P.564.

      ExternalR-Out

   This field corresponds to "ext. R factor" in RFC3611 in the VoIP
   Metrics Report Block.  Here, the value is copied from RTCP XR message
   received from the remote endpoint on "other" side of this endpoint
   refer to RFC3611 for a more detailed explanation). The range of R is
   0-95 for narrowband calls and 0-120 for wideband calls.  Algorithms
   for computing this value SHOULD be compliant with ITU-T
   Recommendation P.564. Within RFC3611 a reported R factor of 127
   indicates that this parameter is unavailable; in this case the
   ConversationalQualityR parameter SHALL be omitted from the vq-rtcpxr
   event.

      ExtROutEstAlg

   This field provides a text name for the algorithm used to estimate
   ExternalR-Out however may report "P.564" for algorithms that have
   been tested against ITU-T Recommendation P.564.

      MOS-LQ

   This field corresponds to "MOSLQ" in RFC3611 in the VoIP Metrics
   Report Block.  This parameter is the estimated mean opinion score for
   listening voice quality on a scale from 1 to 5, in which 5 represents
   "Excellent" and 1 represents "Unacceptable". The RFC3611 format is
   expressed as an integer in the range 10 to 50, corresponding to MOS x
   10. Therefore the value should be divided by 10 to convert to vq-
   rtcpxr format. Additionally, a value of 127 in RFC3611 indicates
   that the parameter is unavailable and SHOULD NOT be included in the
   vq-rtcpxr event.

      MOSLQEstAlg

   This field provides a text name for the algorithm used to estimate
   MOS-LQ however may report "P.564" for algorithms that have been
   tested against ITU-T Recommendation P.564.

      MOS-CQ

   This field corresponds to "MOSCQ" in RFC3611 in the VoIP Metrics
   Report Block.  This parameter is the estimated mean opinion score for


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   conversation voice quality on a scale from 1 to 5, in which 5
   represents excellent and 1 represents unacceptable. The RFC3611
   format is expressed as an integer in the range 10 to 50,
   corresponding to MOS x 10. Therefore the value SHALL be divided by 10
   to convert to vq-rtcpxr format. Additionally, a value of 127 in
   RFC3611 indicates that the parameter is unavailable and SHOULD NOT be
   included in the vq-rtcpxr event.

      MOSCQEstAlg

   This field provides a text name for the algorithm used to estimate
   MOS-CQ however may report "P.564" for algorithms that have been
   tested against ITU-T Recommendation P.564.

      QoEEstAlg

   This field provides a text description of the algorithm used to
   estimate all voice quality metrics. This parameter is provided as an
   alternative to the separate estimation algorithms for use when the
   same algorithm is used for all measurements.

4.7. Message Flow and Syntax Examples

   This section shows a number of message flow examples showing how the
   event package works.

4.7.1. End of Session Report using NOTIFY

      Alice            Proxy/Registrar        Collector             Bob

      |                    |                    |                    |

      |                    |                    |                    |

      | REGISTER Allow-Event:vq-rtcpxr F1       |                    |

      |------------------->|                    |                    |

      |      200 OK F2     |                    |                    |

      |<-------------------|                    |                    |

      |                    |  SUBSCRIBE Event:vq-rtcpxr F3           |

      |                    |<-------------------|                    |

      | SUBSCRIBE Event:vq-rtcpxr F4            |                    |


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      |<-------------------|                    |                    |

      |     200 OK F5      |                    |                    |

      |------------------->|                    |                    |

      |                    |   200 OK F6        |                    |

      |                    |------------------->|                    |

      |      INVITE F7     |                    |                    |

      |------------------->|                    |                    |

      |                    |      INVITE F8     |                    |

      |                    |---------------------------------------->|

      |                    |      200 OK F9     |                    |

      |                    |<----------------------------------------|

      |     200 OK F10     |                    |                    |

      |<-------------------|                    |                    |

      |        ACK F11     |                    |                    |

      |------------------->|                    |                    |

      |                    |      ACK F12       |                    |

      |                    |---------------------------------------->|

      |        RTP         |                    |                    |

      |<============================================================>|

      |        RTCP, RTCP XR                    |                    |

      |<============================================================>|

      |                    |                    |                    |

      |    BYE F13         |                    |                    |

      |------------------->|      BYE F14       |                    |


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      |                    |---------------------------------------->|

      |                    |     200 OK F15     |                    |

      |                    |<----------------------------------------|

      |     200 OK F16     |                    |                    |

      |<-------------------|                    |                    |

      |  NOTIFY Event:vq-rtcpxr F17             |                    |

      |------------------->|                    |                    |

      |                    | NOTIFY Event:vq-rtcpxr F18              |

      |                    |------------------->|                    |

      |                    |     200 OK F19     |                    |

      |                    |<-------------------|                    |

      |     200 OK F20     |                    |                    |

      |<-------------------|                    |                    |



   Figure 1. Summary report with NOTIFY sent after session termination.



   In the call flow depicted in Figure 1, the following message format
   is sent in F17:

      NOTIFY sip:collector@example.org SIP/2.0

      Via: SIP/2.0/UDP pc22.example.org;branch=z9hG4bK3343d7

      Max-Forwards: 70

      To: <sip:collector@example.org>;tag=43524545

      From: Alice <sip:alice@example.org>;tag=a3343df32

      Call-ID: 1890463548@alice.example.org



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      CSeq: 4321 NOTIFY

      Contact: <sip:alice@pc22.example.org>

      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,

      SUBSCRIBE, NOTIFY

      Event: vq-rtcpxr

      Accept: application/sdp, message/sipfrag

      Subscription-State: active;expires=3600

      Content-Type: application/vq-rtcpxr

      Content-Length: ...



      VQSessionReport

      LocalMetrics:

      Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z

      SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50

                      PLC=3 SSUP=on

      CallID:1890463548@alice.example.org

      FromID: Alice <sip:alice@example.org>

      ToID: Bill <sip:bill@elpmaxe.org>

      LocalAddr:IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d

      RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=2468abcd

      JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120

      PacketLoss:NLR=5.0 JDR=2.0

      BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16

      Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10


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      Signal:SL=2 NL=-10 RERL=55

      QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=3.4 MOSCQ=3.3
   QoEEstAlg=P.564

      RemoteMetrics:

      Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z

      SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50

                      PLC=3 SSUP=on

      CallID:1890463548@alice.example.org

      LocalAddr:IP=11.1.1.150 PORT=5002 SSRC=2468abcd

      RemoteAddr:IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d

      JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120

      PacketLoss:NLR=5.0 JDR=2.0

      BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16

      Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10

      Signal:SL=2 NL=-10 RERL=55

      QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=3.4 MOSCQ=3.3
   QoEEstAlg=P.564

      DialogID:1890463548@alice.example.org;to-tag=8472761;
   from-tag=9123dh311

4.7.2. Mid Session Threshold Violation using NOTIFY

      Alice            Proxy/Registrar        Collector             Bob

      |                    |                    |                    |

      |                    |                    |                    |

      | REGISTER Allow-Event:vq-rtcpxr F1       |                    |

      |------------------->|                    |                    |



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      |      200 OK F2     |                    |                    |

      |<-------------------|                    |                    |

      |                    |  SUBSCRIBE Event:vq-rtcpxr F3           |

      |                    |<-------------------|                    |

      | SUBSCRIBE Event:vq-rtcpxr F4            |                    |

      |<-------------------|                    |                    |

      |     200 OK F5      |                    |                    |

      |------------------->|                    |                    |

      |                    |   200 OK F6        |                    |

      |                    |------------------->|                    |

      |      INVITE F7     |                    |                    |

      |------------------->|                    |                    |

      |                    |      INVITE F8     |                    |

      |                    |---------------------------------------->|

      |                    |      200 OK F9     |                    |

      |                    |<----------------------------------------|

      |     200 OK F10     |                    |                    |

      |<-------------------|                    |                    |

      |        ACK F11     |                    |                    |

      |------------------->|                    |                    |

      |                    |      ACK F12       |                    |

      |                    |---------------------------------------->|

      |        RTP         |                    |                    |

      |<============================================================>|


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      |        RTCP, RTCP XR                    |                    |

      |<============================================================>|

      |  NOTIFY Event:vq-rtcpxr F17             |                    |

      |------------------->|                    |                    |

      |                    | NOTIFY Event:vq-rtcpxr F18              |

      |                    |------------------->|                    |

      |                    |     200 OK F19     |                    |

      |                    |<-------------------|                    |

      |     200 OK F20     |                    |                    |

      |<-------------------|                    |                    |

      |                    |                    |                    |

      |    BYE F13         |                    |                    |

      |------------------->|      BYE F14       |                    |

      |                    |---------------------------------------->|

      |                    |     200 OK F15     |                    |

      |                    |<----------------------------------------|

      |     200 OK F16     |                    |                    |

      |<-------------------|                    |                    |

      |  NOTIFY Event:vq-rtcpxr F17             |                    |

      |------------------->|                    |                    |

      |                    | NOTIFY Event:vq-rtcpxr F18              |

      |                    |------------------->|                    |

      |                    |     200 OK F19     |                    |

      |                    |<-------------------|                    |


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      |     200 OK F20     |                    |                    |

      |<-------------------|                    |                    |



      Figure 2. Summary report sent during session with alert report.



   In the call flow depicted in Figure 2, the following message format
   is sent in F17:

      NOTIFY sip:collector@example.org SIP/2.0

      Via: SIP/2.0/UDP pc22.example.org;branch=z9hG4bK3343d7

      Max-Forwards: 70

      To: <sip:collector@example.org>

      From: Alice <sip:alice@example.org>;tag=a3343df32

      Call-ID: 1890463548@alice.example.org

      CSeq: 4321 PUBLISH

      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,

      SUBSCRIBE, NOTIFY

      Event: vq-rtcpxr

      Accept: application/sdp, message/sipfrag

      Content-Type: application/vq-rtcpxr

      Content-Length: ...



      VQAlertReport: Type=RLQ Severity=Warning Dir=local

      Metrics:

      Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z



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      SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50

              PLC=3 SSUP=on

      CallID:1890463548@alice.example.org

      FromID: Alice <sip:alice@example.org>

      ToID: Bill <sip:bill@elpmaxe.org>

      LocalAddr:IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d

      RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=2468abcd

      JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120

      PacketLoss:NLR=5.0 JDR=2.0

      BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16

      Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10

      Signal:SL=2 NL=-10 RERL=55

      QualityEst:RLQ=60 RCQ=55 EXTR=90 MOSLQ=2.4 MOSCQ=2.3
   QoEEstAlg=P.564

      RemoteMetrics:

      Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z

      SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50

                      PLC=3 SSUP=on

      CallID:1890463548@alice.example.org

      LocalAddr:IP=11.1.1.150 PORT=5002 SSRC=2468abcd

      RemoteAddr:IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d

      JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120

      PacketLoss:NLR=5.0 JDR=2.0

      BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=10



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      Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10

      Signal:SL=2 NL=-10 RERL=55

      QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=3.4 MOSCQ=3.3
   QoEEstAlg=P.564

      DialogID:1890463548@alice.example.org;to-tag=8472761;

              from-tag=9123dh31111

4.7.3.  End of Session Report using PUBLISH

      Alice            Proxy/Registrar        Collector             Bob

      |                    |                    |                    |

      |                    |                    |                    |

      | REGISTER Allow-Event:vq-rtcpxr  F1      |                    |

      |------------------->|                    |                    |

      |      200 OK F2     |                    |                    |

      |<-------------------|                    |                    |

      |      INVITE F3     |                    |                    |

      |------------------->|                    |                    |

      |                    |      INVITE F4     |                    |

      |                    |---------------------------------------->|

      |                    |      200 OK F5     |                    |

      |                    |<----------------------------------------|

      |     200 OK F6      |                    |                    |

      |<-------------------|                    |                    |

      |        ACK F7      |                    |                    |

      |------------------->|                    |                    |



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      |                    |      ACK F8        |                    |

      |                    |---------------------------------------->|

      |        RTP         |                    |                    |

      |<============================================================>|

      |        RTCP        |                    |                    |

      |<============================================================>|

      |                    |                    |                    |

      |    BYE F9          |                    |                    |

      |------------------->|      BYE F10       |                    |

      |                    |---------------------------------------->|

      |                    |     200 OK F11     |                    |

      |                    |<----------------------------------------|

      |     200 OK F12     |                    |                    |

      |<-------------------|                    |                    |

      |  PUBLISH Event:vq-rtcpxr F13            |                    |

      |------------------->|                    |                    |

      |                    | PUBLISH Event:vq-rtcpxr F14             |

      |                    |------------------->|                    |

      |                    |     200 OK F15     |                    |

      |                    |<-------------------|                    |

      |     200 OK F16     |                    |                    |

      |<-------------------|                    |                    |






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      Figure 3. End of session report sent after session termination.



   In the message flow depicted in Figure 3, the following message is
   sent in F13.

      PUBLISH sip:collector@example.org SIP/2.0

      Via: SIP/2.0/UDP pc22.example.org;branch=z9hG4bK3343d7

      Max-Forwards: 70

      To: <sip:proxy@example.org>

      From: Alice <sip:alice@example.org>;tag=a3343df32

      Call-ID: 1890463548@alice.example.org

      CSeq: 4331 PUBLISH

      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,

      SUBSCRIBE, NOTIFY

      Event: vq-rtcpxr

      Accept: application/sdp, message/sipfrag

      Content-Type: application/vq-rtcpxr

      Content-Length: ...



      VQSessionReport

      LocalMetrics:

      Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z

      SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50

                      FMTP="annexb=no" PLC=3 SSUP=on

      CallID:1890463548@alice.example.org



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      FromID: Alice <sip:alice@example.org>

      ToID: Bill <sip:bill@elpmaxe.org>

      LocalAddr:IP=10.10.1.100 PORT=5000 SSRC=2468abcd

      RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=1357efff

      JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120

      PacketLoss:NLR=5.0 JDR=2.0

      BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16

      Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10

      Signal:SL=2 NL=-10 RERL=55

      QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=3.4 MOSCQ=3.3
   QoEEstAlg=P.564

      RemoteMetrics:

      Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z

      SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50

                      FMTP="annexb=no" PLC=3 SSUP=on

      CallID:1890463548@alice.example.org

      LocalAddr:IP=11.1.1.150 PORT=5002 SSRC=1357efff

      RemoteAddr:IP=10.10.1.100 PORT=5000 SSRC=2468abcd

      JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120

      PacketLoss:NLR=5.0 JDR=2.0

      BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16

      Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10

      Signal:SL=2 NL=-10 RERL=55

      QualityEst:RLQ=90 RCQ=85 MOSLQ=3.4 MOSCQ=3.3 QoEEstAlg=P.564



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      DialogID:1890463548@alice.example.org;to-tag=8472761;
   from-tag=9123dh311

4.7.4. Alert Report using PUBLISH

      Alice            Proxy/Registrar        Collector             Bob

      |                    |                    |                    |

      |      INVITE F1     |                    |                    |

      |------------------->|                    |                    |

      |                    |      INVITE F2     |                    |

      |                    |---------------------------------------->|

      |                    |      200 OK F3     |                    |

      |                    |<----------------------------------------|

      |     200 OK F4      |                    |                    |

      |<-------------------|                    |                    |

      |        ACK F5      |                    |                    |

      |------------------->|                    |                    |

      |                    |      ACK F6        |                    |

      |                    |---------------------------------------->|

      |        RTP         |                    |                    |

      |<============================================================>|

      |        RTCP        |                    |                    |

      |<============================================================>|

      |  PUBLISH Event:vq-rtcpxr F7             |                    |

      |------------------->|                    |                    |

      |                    | PUBLISH Event:vq-rtcpxr F8              |



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      |                    |------------------->|                    |

      |                    |     200 OK F9      |                    |

      |                    |<-------------------|                    |

      |     200 OK F10     |                    |                    |

      |<-------------------|                    |                    |

      |                    |                    |                    |

      |      BYE F12       |                    |                    |

      |------------------->|      BYE F13       |                    |

      |                    |---------------------------------------->|

      |                    |     200 OK F14     |                    |

      |                    |<----------------------------------------|

      |     200 OK F15     |                    |                    |

      |<-------------------|                    |                    |



      Figure 4. Alert report message flow

   In the message flow depicted in Figure 4, the following message is
   sent in F7:

      PUBLISH sip:collector@example.org SIP/2.0

      Via: SIP/2.0/UDP pc22.example.org;branch=z9hG4bK3343d7

      Max-Forwards: 70

      To: <sip:collector@example.org>

      From: Alice <sip:alice@example.org>;tag=a3343df32

      Call-ID: 1890463548@alice.example.org

      CSeq: 4321 PUBLISH



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      Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,

      SUBSCRIBE, NOTIFY

      Event: vq-rtcpxr

      Accept: application/sdp, message/sipfrag

      Content-Type: application/vq-rtcpxr

      Content-Length: ...



      VQAlertReport: Type=RLQ Severity=Warning Dir=local

      Metrics:

      Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z

      SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50

                      PLC=3 SSUP=on

      CallID:1890463548@alice.example.org

      FromID: Alice <sip:alice@example.org>

      ToID: Bill <sip:bill@elpmaxe.org>

      LocalAddr:IP=10.10.1.100 PORT=5000 SSRC=2a4b6c8d

      RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=9f7e5d3c

      JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120

      PacketLoss:NLR=5.0 JDR=2.0

      BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16

      Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10

      Signal:SL=2 NL=-10 RERL=55

      QualityEst:RLQ=60 RCQ=55 EXTR=90 MOSLQ=2.4 MOSCQ=2.3
   QoEEstAlg=P.564



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      RemoteMetrics:

      Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z

      SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50

                      PLC=3 SSUP=on

      CallID:1890463548@alice.example.rog

      LocalAddr:IP=11.1.1.150 PORT=5002 SSRC=9f7e5d3c

      RemoteAddr:IP=10.10.1.100 PORT=5000 SSRC=2a4b6c8d

      JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120

      PacketLoss:NLR=5.0 JDR=2.0

      BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16

      Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10

      Signal:SL=2 NL=-10 RERL=55

      QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=3.4 MOSCQ=3.3
   QoEEstAlg=P.564

      DialogID:1890463548@alice.example.org;to-tag=8472761;

              from-tag=9123dh3111

4.8.   Configuration Dataset for vq-rtcpxr Events

   It is the suggestion of the authors that the SIP configuration

   framework [8] be used to establish the necessary parameters for

   usage of vq-rtcpxr events.  A dataset for this purpose is provided

   below:



   <?xml version="1.0" encoding="UTF-8"?>

      <xs:schema
   targetNamespace="urn:ietf:params:xml:ns:vqrtcpxrdataset"


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            xmlns:tns="urn:ietf:params:xml:ns:vqrtcpxrdataset"

            xmlns:xs="http://www.w3.org/2001/XMLSchema"



        <xs:element name="rtcpxr-collector">

         <xs:complexType>

                <xs:sequence>

                  <xs:element name="address" type=xs:string/>

                  <xs:element name="port" type=xs:integer/>

                </xs:sequence>

           </xs:complexType>

        </xs:element>



      </xs:schema>

          <xs:complexType>

            <xs:sequence>

              <xs:element name="threshold-parameter">

                <xs:sequence>

                  <xs:element name="parameter-name" type=xs:string/>

                  <xs:element name="warning-level" type=xs:integer/>

                  <xs:element name="excessive-level" type=xs:integer/>

                </xs:sequence>

              </xs:element>

              </xs:sequence>

            </xs:complexType>


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        </xs:element>



        <xs:element name="session-report-settings">

          <xs:complexType>

            <xs:sequence>

              <xs:element name="enable" type=xs:boolean/>

              <xs:element name="remote-report" type=xs:boolean/>

            </xs:sequence>

          </xs:complexType>

        </xs:element>



        <xs:element name="interval-report-settings">

          <xs:complexType>

            <xs:sequence>

              <xs:element name="enable" type=xs:boolean/>

              <xs:element name="remote-report" type=xs:boolean/>

              <xs:element name="timer" type=xs:integer/>

            </xs:sequence>

          </xs:complexType>

        </xs:element>



      </xs:schema>

5. IANA Considerations

   This document registers a new SIP Event Package and a new MIME type.


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5.1. SIP Event Package Registration

   Package name: vq-rtcpx

   Type: package

   Contact: Amy Pendleton <aspen@nortel.com>

   Published Specification: This document

5.2. application/vq-rtcp-xr MIME Registration

   MIME media type name: application

   MIME subtype name: vq-rtcpxr

   Mandatory parameters: none

   Optional parameters: none

   Encoding considerations: text

   Security considerations: See next section.

   Interoperability considerations: none.

   Published specification: This document.

   Applications which use this media type: This document type is
   being used in notifications of VoIP quality reports.

   Additional Information:

      Magic Number: None

      File Extension: None

      Macintosh file type code: "TEXT"

   Personal and email address for further information: Amy Pendleton
   <aspen@nortel.com>

   Intended usage: COMMON

   Author/Change controller: The IETF.




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6. Security Considerations

   RTCP reports can contain sensitive information since they can provide
   information about the nature and duration of a session established
   between two or more endpoints.  As a result, any third party wishing
   to obtain this information SHOULD be properly authenticated by the
   SIP UA using standard SIP mechanisms and according to the
   recommendations in [5].  Additionally the event content MAY be
   encrypted to ensure confidentiality; the mechanisms for providing
   confidentiality are detailed in [2].

7. Contributors

   The authors would like to thank Rajesh Kumar, Dave Oran, Tom
   Redman and Shane Holthaus for their comments and input.

8. Normative References

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

   [2] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
   Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:
   Session Initiation Protocol", RFC 3261, June 2002.

   [3] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
   "RTP: A Transport Protocol for Real-Time Applications", STD 64,
   RFC 3550, July 2003.

   [4] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol
   Extended Reports (RTCP XR)", RFC 3611, November 2003.

   [5] Huitema, C., "Real Time Control Protocol (RTCP) attribute in
   Session Description Protocol (SDP)", RFC 3605, October 2003.

   [5] Roach, A., "Session Initiation Protocol (SIP)-Specific Event

   Notification", RFC 3265, June 2002.

   [6] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax
   Specifications: ABNF", RFC 5234, January 2008.

   [7] Klyne, G. and C. Newman, "Date and Time on the Internet:
   Timestamps", RFC 3339, July 2002.





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   [8] Petrie, D., "A Framework for Session Initiation Protocol User
   Agent Profile Delivery", draft-ietf-sipping-config-framework-15,
   (work in progress), February 2008.

   [9] Rosenberg, J., J. Weinberger, C. Huitema, and R. Mahy, "STUN -                                                                         -
   Simple Traversal of User Datagram Protocol (UDP) Through Network
   Address Translators (NATs)," RFC 3489, March 2003.



   This document was prepared using 2-Word-v2.0.template.dot.

Author's Addresses

      Amy Pendleton

      Nortel

      2380 Performance Drive

      Richardson, TX  75081, USA

      Email: aspen@nortel.com



      Alan Clark

      Telchemy Incorporated

      2905 Premiere Parkway, Suite 280

      Duluth, GA 30097, USA

      Email: alan@telchemy.com



      Alan Johnston

      Avaya

      St. Louis, MO 63124, USA

      Email: alan@sipstation.com




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      Henry Sinnreich

      Adobe Systems, Inc.

      601 Townsend Street,San Francisco, CA 94103, USA

      Email: henrys@adobe.com



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Disclaimer of Validity

   This document and the information contained herein are provided on an
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Copyright Statement

   Copyright (C) The IETF Trust (2008).


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   This document is subject to the rights, licenses and restrictions
   contained in BCP 78, and except as set forth therein, the authors
   retain all their rights.

Acknowledgment

   Funding for the RFC Editor function is currently provided by the
   Internet Society.









































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