    Internet Engineering Task Force (IETF)                      A. Pendleton
    Request for Comments: 6035                                      A. Clark
    Category: Standards Track                          Telchemy Incorporated
    ISSN: 2070-1721                                              A. Johnston
                                                                       Avaya
                                                                H. Sinnreich
                                                                Unaffiliated
                                                               November 2010

     Session Initiation Protocol Event Package for Voice Quality Reporting

    Abstract

       This document defines a Session Initiation Protocol (SIP) event
       package that enables the collection and reporting of metrics that
       measure the quality for Voice over Internet Protocol (VoIP) sessions.
       Voice call quality information derived from RTP Control Protocol
       Extended Reports (RTCP-XR) and call information from SIP is conveyed
       from a User Agent (UA) in a session, known as a reporter, to a third
       party, known as a collector.  A registration for the application/ vq-
       rtcpxr media type is also included.

    Status of This Memo

       This is an Internet Standards Track document.

       This document is a product of the Internet Engineering Task Force
       (IETF).  It represents the consensus of the IETF community.  It has
       received public review and has been approved for publication by the
       Internet Engineering Steering Group (IESG).  Further information on
       Internet Standards is available in Section 2 of RFC 5741.

       Information about the current status of this document, any errata,
       and how to provide feedback on it may be obtained at
       http://www.rfc-editor.org/info/rfc6035.

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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
       to this document.  Code Components extracted from this document must
       include Simplified BSD License text as described in Section 4.e of
       the Trust Legal Provisions and are provided without warranty as
       described in the Simplified BSD License.

       This document may contain material from IETF Documents or IETF
       Contributions published or made publicly available before November
       10, 2008.  The person(s) controlling the copyright in some of this
       material may not have granted the IETF Trust the right to allow
       modifications of such material outside the IETF Standards Process.
       Without obtaining an adequate license from the person(s) controlling
       the copyright in such materials, this document may not be modified
       outside the IETF Standards Process, and derivative works of it may
       not be created outside the IETF Standards Process, except to format
       it for publication as an RFC or to translate it into languages other
       than English.

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    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

    Table of Contents

       1. Introduction ....................................................4
          1.1. Applicability Statement ....................................4
          1.2. Use of the Mechanism .......................................4
       2. Terminology .....................................................6
       3. SIP Events for VoIP Quality Reporting ...........................6
          3.1. SUBSCRIBE NOTIFY Method ....................................6
          3.2. PUBLISH Method .............................................7
          3.3. Multi-Party and Multi-Segment Calls ........................7
          3.4. Overload Avoidance .........................................7
       4. Event Package Formal Definition .................................8
          4.1. Event Package Name .........................................8
          4.2. Event Package Parameters ...................................8
          4.3. SUBSCRIBE Bodies ...........................................8
          4.4. Subscribe Duration .........................................8
          4.5. NOTIFY Bodies ..............................................8
          4.6. Voice Quality Event and Semantics .........................10
               4.6.1. ABNF Syntax Definition .............................10
               4.6.2. Parameter Definitions and Mappings .................21
          4.7. Message Flow and Syntax Examples ..........................29
               4.7.1. End of Session Report Using NOTIFY .................29
               4.7.2. Midsession Threshold Violation Using NOTIFY ........32
               4.7.3. End of Session Report Using PUBLISH ................35
               4.7.4. Alert Report Using PUBLISH .........................37
          4.8. Configuration Dataset for vq-rtcpxr Events ................39
       5. IANA Considerations ............................................39
          5.1. SIP Event Package Registration ............................39
          5.2. application/vq-rtcpxr Media Type Registration .............39
       6. Security Considerations ........................................40
       7. Contributors ...................................................40
       8. References .....................................................40
          8.1. Normative References ......................................40
          8.2. Informative References ....................................41

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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 RTP Control Protocol Extended Reports [4] metrics.

    1.1.  Applicability Statement

       RTP is utilized in many different architectures and topologies.  RFC
       5117 [13] lists and describes the following topologies: point-to-
       point, point-to-multipoint using multicast, point-to-multipoint using
       the translator from RFC 3550, point-to-multipoint using the mixer
       model from RFC 3550, point-to-multipoint using video-switching
       Multipoint Control Units (MCUs), point-to-multipoint using RTCP-
       terminating MCU, and non-symmetric mixer/translators.  As the
       Abstract of this document points out, this specification is for
       reporting quality of Voice over Internet Protocol (VoIP) sessions.
       As such, only the first topology, point to point, is currently
       supported by this specification.  This reflects both current VoIP
       deployments, which are predominantly point to point using unicast,
       and the state of research in the area of quality.

       How to accurately report the quality of a multipart conference or a
       session involving multiple hops through translators and mixers is
       currently an area of research in the industry.  However, this
       mechanism can easily be used for centrally mixed conference calls, in
       which each leg of the conferences is just a point-to-point call.
       This mechanism could be extended to cover additional RTP topologies
       in the future once these topics progress out of the realm of research
       and into actual Internet deployments.

    1.2.  Use of the Mechanism

       RTCP reports are usually sent to other participating endpoints in a
       session.  This can make the collection of performance information by
       an administrator or management system quite complex to implement.  In
       the usage scenarios addressed in this memo, the data contained in
       RTCP XR VoIP metrics reports (RFC 3611 [4]) are forwarded to a
       central collection server systems using SIP.

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    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

       Applications residing in the server or elsewhere can aid in network
       management to alleviate bandwidth constraints and also to support
       customer service by identifying and acknowledging calls of poor
       quality.  However, specifying such applications is beyond the scope
       of this paper.

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

       1.  The codec type, as resulting from the Session Description
           Protocol (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 randomly 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 they are very useful,
       they have a number of limitations as discussed in [3].  This must be
       considered when using RTCP reports in managed networks.

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

       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.

       The event package supports reporting the voice quality metrics for
       both the inbound and outbound directions.  Voice quality metrics for
       the inbound direction can generally be computed locally by the

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       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].

       The configuration of the usage of this event package is not covered
       in this document.  It is the recommendation of this document that the
       SIP configuration framework [15] be used.  This is discussed in
       Section 4.8.

       The event package SHOULD be used with the SUBSCRIBE/NOTIFY method;
       however, it MAY also be used with the PUBLISH method [8] for backward
       compatibility with some existing implementations.  Message flow
       examples for both methods are provided in this document.

    2.  Terminology

       The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
       "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
       document are to be interpreted as described in BCP 14, RFC 2119 [1].

    3.  SIP Events for VoIP Quality 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
       that can make the information available to other applications.  For
       purposes of illustration, the entities involved in SIP vq-rtcpxr
       event reporting will be referred to as follows:

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

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

    3.1.  SUBSCRIBE NOTIFY Method

       The COLLECTOR SHALL send a SUBSCRIBE to the REPORTER to explicitly
       establish the relationship.  The REPORTER SHOULD send the voice
       quality metric reports using the NOTIFY method.  The REPORTER MUST
       NOT send any vq-rtcpxr events if a COLLECTOR address has not been
       configured.  The REPORTER populates the Request-URI according to the
       rules for an in-dialog request.  The COLLECTOR MAY send a SUBSCRIBE
       to a SIP Proxy acting on behalf of the reporting SIP UAs.

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    3.2.  PUBLISH Method

       A SIP UA that supports this specification MAY also send the service
       quality metric reports using the PUBLISH method [8]; however, this
       approach SHOULD NOT be used, in general, on the public Internet.  The
       PUBLISH method MAY be supported for backward compatibility with
       existing implementations.

       The REPORTER MAY therefore populate the Request-URI of the PUBLISH
       method with the address of the COLLECTOR.  To ensure security of SIP
       proxies and the COLLECTOR, the REPORTER MUST be configured with the
       address of the COLLECTOR, preferably using the SIP UA configuration
       framework [15], as described in Section 5.8.

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

          If PUBLISH is not supported, then the REPORTER can only wait for a
          SUBSCRIBE request from the COLLECTOR and then deliver the
          information in NOTIFYs.  If a REPORTER sends a PUBLISH to a
          COLLECTOR that does not support or allow this method, a 501 Not
          Implemented or a 405 Method Not Allowed response will be received,
          and the REPORTER will stop publication.

    3.3.  Multi-Party and Multi-Segment Calls

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

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

    3.4.  Overload Avoidance

       Users of this extension should ensure that they implement general SIP
       mechanisms for avoiding overload.  For instance, an overloaded proxy
       or COLLECTOR MUST send a 503 Service Unavailable or other 5xx
       response with an appropriate Retry-After time specified.  REPORTERs
       MUST act on these responses and respect the Retry-After time

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       interval.  In addition, future SIP extensions to better handle
       overload as covered in [14] should be followed as they are
       standardized.

       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.

       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.

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

    4.  Event Package Formal Definition

    4.1.  Event Package Name

       This document defines a SIP Event Package.  SIP Event Packages were
       originally defined in RFC 3265 [5].

    4.2.  Event Package Parameters

       No event package parameters are defined.

    4.3.  SUBSCRIBE Bodies

       SUBSCRIBE bodies are described by this specification.

    4.4.  Subscribe 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 Bodies

       There are three notify bodies: a Session report, an Interval report,
       and an Alert report.

       The Session report SHOULD be used for reporting when a voice media
       session terminates, when a media change occurs, such as a codec
       change or a session fork, or when a session terminates due to no
       media packets being received and MUST NOT be used for reporting at

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       arbitrary points in time.  This report MUST be used for cumulative
       metric reporting and the report timestamps MUST be from the start of
       a media session to the time at which the report is generated.

       The Interval report SHOULD be used for periodic or interval reporting
       and MUST NOT be used for reporting of the complete media session.
       This report is intended to capture short duration metric reporting
       and the report intervals SHOULD be non-overlapping time windows.

       The Alert report MAY be used when voice quality degrades during a
       session.  The time window to which an Alert report relates MAY be a
       short time interval or from the start of the call to the point the
       alert is generated; this time window SHOULD be selected to provide
       the most useful information to support problem diagnosis.

       Session, Interval, and Alert reports MUST populate the metrics with
       values that are measured over the interval explicitly defined by the
       "start" and "stop" timestamps.

       Voice quality summary reports reference only one codec (payload
       type).  This payload type SHOULD be the main voice payload, not
       comfort noise or telephone event payloads.  For applications that
       consistently and rapidly switch codecs, the most used codec should be
       reported.  All values in the report, such as IP addresses,
       synchronization source (SSRC), etc., represent those values as
       received by the REPORTER.  In some scenarios, these may not be the
       same on either end of the session -- the COLLECTOR will need logic to
       be able to put these sessions together.  The values of parameters
       such as sample rate, frame duration, frame octets, packets per
       second, round-trip delay, etc., depend on the type of report in which
       they are present.  If present in a Session or an Interval report,
       they represent average values over the session or interval.  If
       present in an Alert report, they represent instantaneous values.

       The REPORTER always includes local quality reporting information and
       should, if possible, share remote quality reporting information to
       the COLLECTOR.  This remote quality could be available from received
       RTCP-XR reports or other sources.  Reporting this is useful in cases
       where the other end might support RTCP-XR but not this voice quality
       reporting.

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

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    4.6.  Voice Quality Event 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 contain 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.

       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

    SessionReport = "VQSessionReport" [ HCOLON "CallTerm" ] CRLF
                SessionInfo  CRLF
                LocalMetrics [ CRLF RemoteMetrics ]
                [ CRLF DialogID ]

    ; CallTerm indicates the final report of a session.

    IntervalReport = "VQIntervalReport" [ HCOLON "CallTerm" ] CRLF
                SessionInfo  CRLF
                LocalMetrics [ CRLF RemoteMetrics ]
                [ CRLF DialogID ]

    LocalMetrics  = "LocalMetrics" HCOLON CRLF Metrics

    RemoteMetrics = "RemoteMetrics" HCOLON CRLF Metrics

    AlertReport   = "VQAlertReport" HCOLON
          MetricType WSP Severity WSP Direction CRLF
          SessionInfo  CRLF
          LocalMetrics [ CRLF RemoteMetrics ]
          [ DialogID ]

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    SessionInfo =
       CallID CRLF
       LocalID CRLF
       RemoteID CRLF
       OrigID CRLF
       LocalAddr CRLF
       RemoteAddr CRLF
       LocalGroupID CRLF
       RemoteGroupID CRLF
       [ LocalMACAddr CRLF ]
       [ RemoteMACAddr CRLF ]

    Metrics = TimeStamps CRLF
       [ SessionDescription CRLF ]
       [ JitterBuffer CRLF ]
       [ PacketLoss CRLF ]
       [ BurstGapLoss CRLF ]
       [ Delay CRLF ]
       [ Signal CRLF ]
       [ QualityEstimates CRLF ]
       *(Extension CRLF)

    ; Timestamps are provided in Coordinated Universal Time (UTC)
    ; using the ABNF format provided in RFC 3339,
    ;  "Date and Time on the Internet: Timestamps"
    ; These timestamps SHOULD reflect, as closely as
    ; possible, the actual time during which the media session
    ; was running to enable correlation to events occurring
    ; in the network infrastructure and to accounting records.
    ; Time zones other than "Z" are not allowed.

    TimeStamps = "Timestamps" HCOLON 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" HCOLON
       [ PayloadType WSP ]
       [ PayloadDesc WSP ]
       [ SampleRate WSP ]
       [ PacketsPerSecond WSP ]
       [ FrameDuration WSP ]
       [ FrameOctets WSP ]
       [ FramesPerPacket WSP ]
       [ FmtpOptions WSP ]

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       [ PacketLossConcealment WSP ]
       [ SilenceSuppressionState ]
       *(WSP Extension)

    ; PayloadType provides the PT parameter used in the RTP packets.

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

    ; PayloadDesc provides a text description of the codec.
    ; This parameter SHOULD use the IANA registry for
    ; media-type names defined by RFC 4855 where it unambiguously
    ; defines the codec.  Refer to the "Audio Media Types"
    ; registry on http://www.iana.org.

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

    ; SampleRate reports the rate at which a voice was sampled
    ; in the case of narrowband codecs, this value will typically
    ; be 8000.
    ; For codecs that are able to change sample rates, the lowest and
    ; highest sample rates MUST be reported (e.g., 8000;16000).

    SampleRate = "SR" EQUAL (1*6DIGIT) *(SEMI (1*66DIGIT))

    ; FrameDuration can be combined with the FramesPerPacket
    ; to determine the packetization rate; the units for
    ; FrameDuration are milliseconds.  NOTE: for frame-based codecs,
    ; each frame constitutes a single frame; for sample-based codecs,
    ; a "frame" refers to the set of samples carried in an RTP packet.

    FrameDuration = "FD" EQUAL (1*4DIGIT)

    ; FrameOctets provides the number of octets in each frame
    ; at the time the report is generated (i.e., last value).
    ; This MAY be used where FrameDuration is not available.
    ; NOTE: for frame-based codecs, each frame constitutes a single frame;
    ; for sample-based codecs, a "frame" refers to the set of samples
    ; carried in an RTP packet.

    FrameOctets  = "FO" EQUAL (1*5DIGIT)

    ; FramesPerPacket provides the number of frames in each RTP
    ; packet at the time the report is generated.
    ; NOTE: for frame-based codecs, each frame constitutes a single frame;
    ; for sample-based codecs, a "frame" refers to the set of samples
    ; carried in an RTP packet.

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

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    ; Packets per second provides the average number of packets
    ; that are transmitted per second, as at the time the report is
    ; generated.

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

    ; FMTP options from SDP.  Note that the parameter is delineated
    ; by " " to avoid parsing issues in transitioning between SDP
    ; and SIP parsing.

    FmtpOptions = "FMTP" EQUAL DQUOTE word-plus DQUOTE

    ; PacketLossConcealment indicates whether a PLC algorithm was
    ; or is being used for the session.  The values follow the same
    ; numbering convention as RFC 3611 [4].
    ; 0 - unspecified
    ; 1 - disabled
    ; 2 - enhanced
    ; 3 - standard

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

    ; SilenceSuppressionState indicates whether silence suppression,
    ; also known as Voice Activity Detection (VAD) is enabled.

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

    ; CallId provides the call id from the SIP dialog.

    CallID  =  "CallID" HCOLON Call-ID-Parm

    ; LocalID identifies the reporting endpoint for the media session [2].

    LocalID = "LocalID" HCOLON (name-addr/addr-spec)

    ; RemoteID identifies the remote endpoint of the media session [2].

    RemoteID = "RemoteID" HCOLON (name-addr/addr-spec)

    ; OrigID identifies the endpoint which originated the session.

    OrigID = "OrigID" HCOLON (name-addr/addr-spec)

    ; LocalAddr provides the IP address, port, and SSRC of the
    ; endpoint/UA, which is the receiving end of the stream being
    ; measured.

    LocalAddr   = "LocalAddr" HCOLON IPAddress WSP Port WSP Ssrc

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    ; RemoteAddr provides the IP address, port, and SSRC of the
    ; the source of the stream being measured.

    RemoteAddr  = "RemoteAddr" HCOLON IPAddress WSP Port WSP Ssrc

    ; LocalMACAddr provides the Media Access Control (MAC) address
    ; of the local SIP device.

    LocalMACAddr   = "LocalMAC" HCOLON hex2 *(":" hex2)

    ; RemoteMACAddr provides the MAC address
    ; of the remote SIP device.

    RemoteMACAddr   = "RemoteMAC" HCOLON hex2 *(":" hex2)

    ; LocalGroupID provides the identification for the purposes
    ; of aggregation for the local endpoint.

    LocalGroupID = "LocalGroup" HCOLON word-plus

    ; RemoteGroupID provides the identification for the purposes
    ; of aggregation for the remote endpoint.

    RemoteGroupID = "RemoteGroup" HCOLON word-plus

    ; For clarification, the LocalAddr in the LocalMetrics report
    ; MUST be the RemoteAddr in the RemoteMetrics report.

    IPAddress   = "IP" EQUAL IPv6address / IPv4address
    Port        = "PORT" EQUAL 1*DIGIT
    Ssrc        = "SSRC" EQUAL ( %x30.78 1*8HEXDIG)

    JitterBuffer = "JitterBuffer" HCOLON
       [ JitterBufferAdaptive WSP ]
       [ JitterBufferRate WSP ]
       [ JitterBufferNominal WSP ]
       [ JitterBufferMax WSP ]
       [ JitterBufferAbsMax ]
       *(WSP Extension)

    ; JitterBufferAdaptive indicates whether the jitter buffer in
    ; the endpoint is adaptive, static, or unknown.
    ; The values follow the same numbering convention as RFC 3611 [4].
    ; For more details, please refer to that document.
    ; 0 - unknown
    ; 1 - reserved
    ; 2 - non-adaptive
    ; 3 - adaptive

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

    ; JitterBuffer metric definitions are provided in RFC 3611 [4].

    JitterBufferRate      = "JBR" EQUAL (1*2DIGIT) ;0-15
    JitterBufferNominal   = "JBN" EQUAL (1*5DIGIT) ;0-65535
    JitterBufferMax       = "JBM" EQUAL (1*5DIGIT) ;0-65535
    JitterBufferAbsMax    = "JBX" EQUAL (1*5DIGIT) ;0-65535

    ; PacketLoss metric definitions are provided in RFC 3611 [4].

    PacketLoss = "PacketLoss" HCOLON
               [ NetworkPacketLossRate WSP ]
               [ JitterBufferDiscardRate ]
               *(WSP Extension)

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

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

    ; BurstGapLoss metric definitions are provided in RFC 3611 [4].

    BurstGapLoss = "BurstGapLoss" HCOLON
       [ 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

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

    Delay = "Delay" HCOLON
       [ RoundTripDelay WSP ]
       [ EndSystemDelay WSP ]
       [ OneWayDelay WSP ]
       [ SymmOneWayDelay WSP ]
       [ InterarrivalJitter WSP ]
       [ MeanAbsoluteJitter ]
       *(WSP Extension)

    ; RoundTripDelay SHALL be measured as defined in RFC 3550 [3].

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

    ; EndSystemDelay metric is defined in RFC 3611 [4].

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

    ; OneWayDelay is defined in RFC 2679 [12].

    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 calculated as defined RFC 3550 [3]
    ; and converted into milliseconds.

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

    ; Mean Absolute Jitter is measured as defined
    ; by ITU-T G.1020 [9] where it is known as MAPDV.

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

    ; Signal metrics definitions are provided in RFC 3611 [4].

    Signal = "Signal" HCOLON
       [ SignalLevel WSP ]
       [ NoiseLevel WSP ]
       [ ResidualEchoReturnLoss ]
       *(WSP Extension)

    ; SignalLevel will normally be a negative value.

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    ; 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 and the sign MUST be
    ; explicitly included.
    ; The absence of a sign indicates a positive value.
    ; This metric applies to the speech signal decoded from the
    ; received packet stream.

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

    ; Residual Echo Return Loss (RERL) is the ratio between
    ; the original signal and the echo level as measured after
    ; echo cancellation or suppression has been applied.
    ; Expressed in decibels (dB).  This is typically a positive
    ; value.
    ; This metric relates to the proportion of the speech signal
    ; decoded from the received packet stream that is reflected
    ; back in the encoded speech signal output in the transmitted
    ; packet stream (i.e., will affect the REMOTE user's
    ; conversational quality).  To support the diagnosis of echo-
    ; related problems experienced by the local user of the device
    ; generating a report according to this document, the value of
    ; RERL reported via the RTCP XR VoIP Metrics payload SHOULD be
    ; reported in the RemoteMetrics set of data.

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

    ; Voice Quality estimation metrics.
    ; Each quality estimate has an optional associated algorithm.
    ; These fields permit the implementation to use a variety
    ; of different calculation methods for each type of metric.

    QualityEstimates  = "QualityEst" HCOLON
       [ ListeningQualityR WSP ]
       [ RLQEstAlg WSP ]
       [ ConversationalQualityR WSP ]
       [ RCQEstAlg WSP ]
       [ ExternalR-In WSP ]
       [ ExtRInEstAlg WSP ]
       [ ExternalR-Out WSP ]
       [ ExtROutEstAlg WSP ]
       [ MOS-LQ WSP ]
       [ MOSLQEstAlg WSP ]

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       [ MOS-CQ WSP ]
       [ MOSCQEstAlg WSP ]
       [ QoEEstAlg ]
       *(WSP Extension)

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

    RLQEstAlg = "RLQEstAlg" EQUAL word ; "P.564" [10], 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 the "other" side of this endpoint.  For example,
    ;   Phone A <---> Bridge <----> Phone B
    ;   ListeningQualityR = quality for Phone A ----> Bridge path
    ;   ExternalR-In = quality for Bridge <---- Phone B path

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

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

    ; ExternalR-Out is copied from the RTCP XR message received from the
    ; remote endpoint on the "other" side of this endpoint.  For example,
    ;   Phone A <---> Bridge <----> Phone B
    ;   ExternalR-Out = quality for Bridge -----> Phone B path

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

    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

    ; QoEEstAlg provides an alternative to the separate
    ; estimation algorithms for use when the same algorithm
    ; is used for all measurements.

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    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, using the
    ; character encoding that represents the parameter in
    ; this ABNF.  The Extension parameter can be used to provide
    ; metrics that are not defined by this document.

    MetricType = "Type" EQUAL "RLQ" / "RCQ" / "EXTR" /
       "MOSLQ" / "MOSCQ" /
       "BD" / "NLR" / "JDR" /
       "RTD" / "ESD" / "IAJ" /
       "RERL" / "SL" / "NL" / Extension

    Direction = "Dir" EQUAL "local" / "remote"
    Severity  = "Severity" EQUAL "Warning" / "Critical" /
       "Clear"

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

    ; General ABNF notation from RFC 5234.

    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

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    ; ABNF notation from RFC 3261.

    alphanum  =  ALPHA / DIGIT
    LWS  =  [ *WSP CRLF ] 1*WSP ; linear whitespace
    SWS  =  [ LWS ] ; sep whitespace
    SEMI =  SWS ";" SWS ; semicolon
    EQUAL   =  SWS "=" SWS ; equal
    COLON   =  SWS ":" SWS ; colon
    HCOLON  =  *( SP / HTAB ) ":" SWS

    token       =  1*(alphanum / "-" / "." / "!" / "%" / "*"
                      / "_" / "+" / "`" / "'" / "~" )

    IPv4address   =  1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT
    IPv6address   =  hexpart [ ":" IPv4address ]
    hexpart       =  hexseq / hexseq "::" [ hexseq ] / "::"
                          [ hexseq ]
    hexseq        =  hex4 *( ":" hex4)
    hex4          =  1*4HEXDIG
    hex2          =  2HEXDIG

    ; ABNF notation from RFC 3339.

    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
    ;

    Extension = word-plus

    word  =  1*(alphanum / "-" / "." / "!" / "%" / "*" /
       "_" / "+" / "`" / "'" / "~" /
       "(" / ")" / "<" / ">" /
       ":" / "\" / DQUOTE /
       "/" / "[" / "]" / "?" )

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    word-plus =  1*(alphanum /  "-"  /  "."  /  "!"  / "%" / "*" /
       "_"  /  "+"  /  "`"  /  "'"  /  "~"  /
       "("  /  ")"  /  "<"  /  ">"  /  ":"  /
       "\"  /  "/"  /  "["  /  "]"  /  "?"  /
       "{"  /  "}"  /  "="  /  " ")

    4.6.2.  Parameter Definitions and Mappings

       Parameter values, codec types, and other aspects of the endpoints may
       change dynamically during a session.  The reported values of metrics
       and configuration parameters SHALL be the current value at the time
       the report is generated.

       The Packet Loss Rate and Packet Discard Rate parameters are
       calculated over the period between the starting and ending timestamps
       for the report.  These are normally calculated from a count of the
       number of lost or discarded packets divided by the count of the
       number of packets, and hence are based on the current values of these
       counters at the time the report was generated.

       Packet delay variation, signal level, noise level, and echo level are
       computed as running or interval averages, based on the appropriate
       standard, e.g., RFC 3550 for Packet Delay Variation (PDV), and the
       sampled value of these running averages is reported.  Delay, packet
       size, jitter buffer size, and codec-related data may change during a
       session and the current value of these parameters is reported as
       sampled at the time the report is generated.

    4.6.2.1.  General Mapping Percentages from 8-bit, Fixed-Point Numbers

       RFC 3611 uses an 8-bit, fixed-point number with the binary point at
       the left edge of the field.  This value is calculated by dividing the
       total number of packets lost by the total number of packets expected
       and multiplying the result by 256, and then taking the integer part.

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

    4.6.2.2.  Timestamps

       Following SIP and other IETF conventions, timestamps are provided in
       Coordinated Universal Time (UTC) using the ABNF format provided in
       RFC 3339 [7].  These timestamps SHOULD reflect, as closely as
       possible, the actual time during which the media session was running
       to enable correlation to related events occurring in the network and
       to accounting or billing records.

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    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.  Where values may change during a
       session, for example, a codec may change rate, then the most-recent
       value of the parameter is reported.

    4.6.2.3.1.  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.

    4.6.2.3.2.  Payload Desc

       This parameter is a text description of the codec.  This parameter
       SHOULD use the IANA registry for media-type names where it
       unambiguously defines the codec.  Refer to the "Audio Media Types"
       registry on http://www.iana.org.

    4.6.2.3.3.  Sample Rate

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

    4.6.2.3.4.  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
       (rounded) number of RTP packets that are transmitted per second.

    4.6.2.3.5.  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 that this value can be combined with the
       FramesPerPacket to determine the packetization rate.  Also, where a
       sample-based codec is used, a "frame" refers to the set of samples
       carried in an RTP packet.

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    4.6.2.3.6.  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 the FrameDuration is provided.  Also, where a sample-based codec
       is used, a "frame" refers to the set of samples carried in an RTP
       packet.

    4.6.2.3.7.  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
       frames in each RTP packet.  Note that this value can be combined with
       the FrameDuration to determine the packetization rate.  Also, where a
       sample-based codec is used, a "frame" refers to the set of samples
       carried in an RTP packet.

    4.6.2.3.8.  FMTP Options

       This parameter is taken directly from the SDP attribute "fmtp"
       defined in RFC 4566.

    4.6.2.3.9.  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.

    4.6.2.3.10.  Packet Loss Concealment

       This value corresponds to "PLC" in RFC 3611 in the VoIP Metrics
       Report Block.  The values defined by RFC 3611 are reused by this
       recommendation and therefore no mapping is required.

    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
       measuring performance.  The IPAddress MAY be in IPv4 or IPv6 format.
       The SSRC is taken from SDP, RTCP, or RTCP XR input parameters.

       In the presence of NAT and where a NAT-traversal mechanism such as
       Session Traversal Utilities for NAT (STUN) [16] is used, the external
       IP address can be reported, since the internal IP address is not
       visible to the network operator.

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    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 measuring
       performance.  In the presence of NAT and where a NAT-traversal
       mechanism such as STUN [16] is used, the external IP address can be
       reported, since the internal IP address is not visible to the network
       operator.

    4.6.2.6.  Jitter Buffer Parameters

    4.6.2.6.1.  Jitter Buffer Adaptive

       This value corresponds to "JBA" in RFC 3611 in the VoIP Metrics
       Report Block.  The values defined by RFC 3611 are unchanged and
       therefore no mapping is required.

    4.6.2.6.2.  Jitter Buffer Rate

       This value corresponds to "JB rate" in RFC 3611 in the VoIP Metrics
       Report Block.  The parameter does not require any conversion.

    4.6.2.6.3.  Jitter Buffer Nominal

       This value corresponds to "JB nominal" in RFC 3611 in the VoIP
       Metrics Report Block.  The parameter does not require any conversion.

    4.6.2.6.4.  Jitter Buffer Max

       This value corresponds to "JB maximum" in RFC 3611 in the VoIP
       Metrics Report Block.  The parameter does not require any conversion.

    4.6.2.6.5.  Jitter Buffer Abs Max

       This value corresponds to "JB abs max" in RFC 3611 in the VoIP
       Metrics Report Block.  The parameter does not require any conversion.

    4.6.2.7.  Packet Loss Parameters

    4.6.2.7.1.  Network Loss Rate

       This value corresponds to "loss rate" in RFC 3611 in the VoIP Metrics
       Report Block.  For conversion, see Section 4.6.2.1.  A loss rate of
       100% MAY be reported if media packets were expected but none had been
       received at the time of session termination.

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    4.6.2.7.2.  Jitter Buffer Discard Rate

       This value corresponds to "discard rate" in RFC 3611 in the VoIP
       Metrics Report Block.  For conversion, see Section 4.6.2.1.

    4.6.2.8.  Burst/Gap Parameters

    4.6.2.8.1.  Burst Loss Density

       This value corresponds to "burst density" in RFC 3611 in the VoIP
       Metrics Report Block.  For conversion, see Section 4.6.2.1.

    4.6.2.8.2.  Burst Duration

       This value corresponds to "burst duration" in RFC 3611 in the VoIP
       Metrics Report Block.  This value requires no conversion; the exact
       value sent in an RTCP XR VoIP Metrics Report Block can be included in
       the SIP vq-rtcpxr parameter.

    4.6.2.8.3.  Gap Loss Density

       This value corresponds to "gap density" in RFC 3611 in the VoIP
       metrics Report Block.

    4.6.2.8.4.  Gap Duration

       This value corresponds to "gap duration" in RFC 3611 in the VoIP
       Metrics Report Block.  This value requires no conversion; the exact
       value sent in an RTCP XR VoIP Metrics Report Block can be reported.

    4.6.2.8.5.  Minimum Gap Threshold

       This value corresponds to "Gmin" in RFC 3611 in the VoIP Metrics
       Report Block.  This value requires no conversion; the exact value
       sent in an RTCP XR VoIP Metrics Report Block can be reported.

    4.6.2.9.  Delay Parameters

    4.6.2.9.1.  Round-Trip Delay

       This value corresponds to "round trip delay" in RFC 3611 in the VoIP
       Metrics Report Block and may be measured using the method defined in
       RFC 3550.  The parameter is expressed in milliseconds.

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    4.6.2.9.2.  End System Delay

       This value corresponds to "end system delay" in RFC 3611 in the VoIP
       Metrics Report Block.  This parameter does not require any
       conversion.  The parameter is expressed in milliseconds.

    4.6.2.9.3.  Symmetric One-Way Delay

       This value is computed by adding Round-Trip Delay to the local and
       remote End System Delay and dividing by two.

    4.6.2.9.4.  One-Way Delay

       This value SHOULD be measured using the methods defined in IETF RFC
       2679 [12].  The parameter is expressed in milliseconds.

    4.6.2.9.5.  Inter-Arrival Jitter

       Inter-arrival jitter is calculated as defined in RFC 3550 and
       converted into milliseconds.

    4.6.2.9.6.  Mean Absolute Jitter

       It is recommended that MAJ be measured as defined in ITU-T G.1020
       [9].  This parameter is often referred to as MAPDV (Mean Absolute
       Packet Delay Variation).  The parameter is expressed in milliseconds.

    4.6.2.10.  Signal-Related Parameters

    4.6.2.10.1.  Signal Level

       This field corresponds to "signal level" in RFC 3611 in the VoIP
       Metrics Report Block.  This field provides the voice signal relative
       level is defined as the ratio of the signal level to a 0 dBm0
       reference, expressed in decibels.  This value can be used directly
       without extra conversion.

    4.6.2.10.2.  Noise Level

       This field corresponds to "noise level" in RFC 3611 in the VoIP
       Metrics Report Block.  This field provides the ratio of the silent
       period background noise level to a 0 dBm0 reference, expressed in
       decibels.  This value can be used directly without extra conversion.

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    4.6.2.10.3.  Residual Echo Return Loss (RERL)

       This field corresponds to "RERL" in RFC 3611 in the VoIP Metrics
       Report Block.  This field provides the ratio between the original
       signal and the echo level in decibels, as measured after echo
       cancellation or suppression has been applied.  This value can be used
       directly without extra conversion.

    4.6.2.11.  Quality Scores

    4.6.2.11.1.  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 Recommendations P.564 [10] and G.107 [11].

    4.6.2.11.2.  RLQEstAlg

       This field provides a text name for the algorithm used to estimate
       ListeningQualityR.  This field will be free form text and not
       necessarily reflective of any standards or recommendations.

    4.6.2.11.3.  ConversationalQualityR

       This field corresponds to "R factor" in RFC 3611 in the VoIP Metrics
       Report Block.  This parameter provides a cumulative measurement of
       voice quality from the start of the session to the reporting time.
       The range of R is 0-95 for narrowband calls and 0-120 for wideband
       calls.  Algorithms for computing this value SHOULD be compliant with
       ITU-T Recommendations P.564 and G.107.  Within RFC 3611, a reported R
       factor of 127 indicates that this parameter is unavailable; in this
       case, the ConversationalQualityR parameter MUST be omitted from the
       vq-rtcpxr event.

    4.6.2.11.4.  RCQEstAlg

       This field provides a text name for the algorithm used to estimate
       ConversationalQualityR.  This field will be free form text and not
       necessarily reflective of any standards or recommendations.

    4.6.2.11.5.  ExternalR-In

       This field corresponds to "ext. R factor" in RFC 3611 in the VoIP
       Metrics Report Block.  This parameter reflects voice quality as
       measured by the local endpoint for incoming connection on "other"
       side (refer to RFC 3611 for a more-detailed explanation).  The range

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       of R is 0-95 for narrowband calls and 0-120 for wideband calls.
       Algorithms for computing this value SHOULD be compliant with ITU-T
       Recommendations P.564 and G.107.  Within RFC 3611, a reported R
       factor of 127 indicates that this parameter is unavailable; in this
       case, the ConversationalQualityR parameter MUST be omitted from the
       vq-rtcpxr event.

    4.6.2.11.6.  ExtRInEstAlg

       This field provides a text name for the algorithm used to estimate
       ExternalR-In.  This field will be free-form text and not necessarily
       reflective of any standards or recommendations.

    4.6.2.11.7.  ExternalR-Out

       This field corresponds to "ext. R factor" in RFC 3611 in the VoIP
       Metrics Report Block.  Here, the value is copied from RTCP XR message
       received from the remote endpoint on the "other" side of this
       endpoint; refer to RFC 3611 for a more detailed explanation).  The
       range of R is 0-95 for narrowband calls and 0-120 for wideband calls.
       Algorithms for computing this value SHOULD be compliant with ITU-T
       Recommendations P.564 and G.107.  Within RFC 3611, a reported R
       factor of 127 indicates that this parameter is unavailable; in this
       case, the ConversationalQualityR parameter SHALL be omitted from the
       vq-rtcpxr event.

    4.6.2.11.8.  ExtROutEstAlg

       This field provides a text name for the algorithm used to estimate
       ExternalR-Out.  This field will be free-form text and not necessarily
       reflective of any standards or recommendations.

    4.6.2.11.9.  MOS Reporting

       Conversion of RFC 3611 reported mean opinion scores (MOSs) for use in
       reporting MOS-LQ and MOS-CQ MUST be performed by dividing the RFC
       3611 reported value by 10 if this value is less than or equal to 50
       or omitting the MOS-xQ parameter if the RFC 3611 reported value is
       127 (which indicates unavailable).

    4.6.2.11.9.1.  MOS-LQ

       This field corresponds to "MOSLQ" in RFC 3611 in the VoIP Metrics
       Report Block.  This parameter is the estimated mean opinion score for
       listening voice quality on a scale from 1 to 5, in which 5 represents
       "Excellent" and 1 represents "Unacceptable".  Algorithms for

    Pendleton, et al.            Standards Track                   [Page 28]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

       computing this value SHOULD be compliant with ITU-T Recommendation
       P.564 [10].  This field provides a text name for the algorithm used
       to estimate MOS-LQ.

    4.6.2.11.9.2.  MOS-CQ

       This field corresponds to "MOSCQ" in RFC 3611 in the VoIP Metrics
       Report Block.  This parameter is the estimated mean opinion score for
       conversation voice quality on a scale from 1 to 5, in which 5
       represents excellent and 1 represents unacceptable.  Algorithms for
       computing this value SHOULD be compliant with ITU-T Recommendation
       P.564 with regard to the listening quality element of the computed
       MOS score.

    4.6.2.11.9.3.  MOSCQEstAlg

       This field provides a text name for the algorithm used to estimate
       MOS-CQ.  This field will be free-form text and not necessarily
       reflective of any standards or recommendations.

    4.6.2.11.10.  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.  This field will be
       free-form text and not necessarily reflective of any standards or
       recommendations.

    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

    Pendleton, et al.            Standards Track                   [Page 29]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

           Alice            Proxy/Registrar        Collector             Bob
           |                    |                    |                    |
           |                    |                    |                    |
           | REGISTER Allow-Event:vq-rtcpxr F1       |                    |
           |------------------->|                    |                    |
           |      200 OK F2     |                    |                    |
           |<-------------------|                    |                    |
           |                    |  SUBSCRIBE Event:vq-rtcpxr F3           |
           |                    |<-------------------|                    |
           | SUBSCRIBE Event:vq-rtcpxr F4            |                    |
           |<-------------------|                    |                    |
           |     200 OK F5      |                    |                    |
           |------------------->|                    |                    |
           |                    |   200 OK F6        |                    |
           |                    |------------------->|                    |
           |      INVITE F7     |                    |                    |
           |------------------->|                    |                    |
           |                    |      INVITE F8     |                    |
           |                    |---------------------------------------->|
           |                    |      200 OK F9     |                    |
           |                    |<----------------------------------------|
           |     200 OK F10     |                    |                    |
           |<-------------------|                    |                    |
           |        ACK F11     |                    |                    |
           |------------------->|                    |                    |
           |                    |      ACK F12       |                    |
           |                    |---------------------------------------->|
           |        RTP         |                    |                    |
           |<============================================================>|
           |        RTCP, RTCP XR                    |                    |
           |<============================================================>|
           |                    |                    |                    |
           |    BYE F13         |                    |                    |
           |------------------->|      BYE F14       |                    |
           |                    |---------------------------------------->|
           |                    |     200 OK F15     |                    |
           |                    |<----------------------------------------|
           |     200 OK F16     |                    |                    |
           |<-------------------|                    |                    |
           |  NOTIFY Event:vq-rtcpxr F17             |                    |
           |------------------->|                    |                    |
           |                    | NOTIFY Event:vq-rtcpxr F18              |
           |                    |------------------->|                    |
           |                    |     200 OK F19     |                    |
           |                    |<-------------------|                    |
           |     200 OK F20     |                    |                    |
           |<-------------------|                    |                    |

    Pendleton, et al.            Standards Track                   [Page 30]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

       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
           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: CallTerm
           CallID: 6dg37f1890463
           LocalID: Alice <sip:alice@example.org>
           RemoteID: Bill <sip:bill@example.net>
           OrigID: Alice <sip:alice@example.org>
           LocalGroup: example-phone-55671
           RemoteGroup: example-gateway-09871
           LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d
           LocalMAC: 00:1f:5b:cc:21:0f
           RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd
           RemoteMAC: 00:26:08:8e:95:02
           LocalMetrics:
           Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
           SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
                           PLC=3 SSUP=on
           JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
           PacketLoss:NLR=5.0 JDR=2.0
           BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
           Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
           Signal:SL=-18 NL=-50 RERL=55
           QualityEst:RLQ=88 RCQ=85 EXTRI=90 MOSLQ=4.1 MOSCQ=4.0
             QoEEstAlg=P.564
           RemoteMetrics:
           Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
           SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
                           PLC=3 SSUP=on
           JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
           PacketLoss:NLR=5.0 JDR=2.0

    Pendleton, et al.            Standards Track                   [Page 31]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

           BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
           Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
           Signal:SL=-21 NL=-45 RERL=55
           QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=4.3 MOSCQ=4.2
             QoEEstAlg=P.564
           DialogID:1890463548@alice.example.org;to-tag=8472761;
             from-tag=9123dh311

    4.7.2.  Midsession Threshold Violation 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            |                    |
        |<-------------------|                    |                    |
        |     200 OK F5      |                    |                    |
        |------------------->|                    |                    |
        |                    |   200 OK F6        |                    |
        |                    |------------------->|                    |
        |      INVITE F7     |                    |                    |
        |------------------->|                    |                    |
        |                    |      INVITE F8     |                    |
        |                    |---------------------------------------->|
        |                    |      200 OK F9     |                    |
        |                    |<----------------------------------------|
        |     200 OK F10     |                    |                    |
        |<-------------------|                    |                    |
        |        ACK F11     |                    |                    |
        |------------------->|                    |                    |
        |                    |      ACK F12       |                    |
        |                    |---------------------------------------->|
        |        RTP         |                    |                    |
        |<============================================================>|
        |        RTCP, RTCP XR                    |                    |
        |<============================================================>|
        |  NOTIFY Event:vq-rtcpxr F13             |                    |
        |------------------->|                    |                    |
        |                    | NOTIFY Event:vq-rtcpxr F14              |
        |                    |------------------->|                    |
        |                    |     200 OK F15     |                    |
        |                    |<-------------------|                    |
        |     200 OK F16     |                    |                    |

    Pendleton, et al.            Standards Track                   [Page 32]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

        |<-------------------|                    |                    |
        |                    |                    |                    |
        |    BYE F17         |                    |                    |
        |------------------->|      BYE F18       |                    |
        |                    |---------------------------------------->|
        |                    |     200 OK F19     |                    |
        |                    |<----------------------------------------|
        |     200 OK F20     |                    |                    |
        |<-------------------|                    |                    |
        |  NOTIFY Event:vq-rtcpxr F21             |                    |
        |------------------->|                    |                    |
        |                    | NOTIFY Event:vq-rtcpxr F22              |
        |                    |------------------->|                    |
        |                    |     200 OK F23     |                    |
        |                    |<-------------------|                    |
        |     200 OK F24     |                    |                    |
        |<-------------------|                    |                    |

       Figure 2.  An alert report is sent during the session.
       In the call flow depicted in Figure 2, the following message
       format is sent in F13:

           NOTIFY sip:collector@example.org SIP/2.0
           Via: SIP/2.0/UDP pc22.example.org;branch=z9hG4bK3343d7
           Max-Forwards: 70
           To: <sip:proxy@example.org>
           From: Alice <sip:alice@example.org>;tag=a3343df32
           Call-ID: 1890463548
           CSeq: 4331 PUBLISH
           Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
            SUBSCRIBE, NOTIFY
           Event: vq-rtcpxr
           Accept: application/sdp, message/sipfrag
           Content-Type: application/vq-rtcpxr
           Content-Length: ...

           VQAlertReport: Type=NLR Severity=Critical Dir=local
           CallID: 6dg37f1890463
           LocalID: Alice <sip:alice@example.org>
           RemoteID: Bill <sip:bill@example.org>
           OrigID: Alice <sip:alice@example.org>
           LocalGroup: example-phone-55671
           RemoteGroup: example-gateway-09871
           LocalAddr:IP=10.10.1.100 PORT=5000 SSRC=0x2468abcd
           LocalMAC: 00:1f:5b:cc:21:0f

    Pendleton, et al.            Standards Track                   [Page 33]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

           RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=1357efff
           RemoteMAC: 00:26:08:8e:95:02
           LocalMetrics:
           Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
           SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
                           FMTP="annexb=no" PLC=3 SSUP=on
           JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
           PacketLoss:NLR=10.0 JDR=2.0
           BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
           Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
           Signal:SL=-21 NL=-50 RERL=55
           QualityEst:RLQ=80 RCQ=85 EXTRI=90 MOSLQ=3.5 MOSCQ=3.7
                            QoEEstAlg=P.564
           RemoteMetrics:
           Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
           SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
                           FMTP="annexb=no" PLC=3 SSUP=on
           JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
           PacketLoss:NLR=5.0 JDR=2.0
           BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
           Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
           Signal:SL=-21 NL=-45 RERL=55
           QualityEst:RLQ=90 RCQ=85 MOSLQ=4.3 MOSCQ=4.2 QoEEstAlg=P.564
           DialogID:1890463548@alice.example.org;to-tag=8472761;
              from-tag=9123dh311

    Pendleton, et al.            Standards Track                   [Page 34]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

    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      |                    |                    |
           |------------------->|                    |                    |
           |                    |      ACK F8        |                    |
           |                    |---------------------------------------->|
           |        RTP         |                    |                    |
           |<============================================================>|
           |        RTCP        |                    |                    |
           |<============================================================>|
           |                    |                    |                    |
           |    BYE F9          |                    |                    |
           |------------------->|      BYE F10       |                    |
           |                    |---------------------------------------->|
           |                    |     200 OK F11     |                    |
           |                    |<----------------------------------------|
           |     200 OK F12     |                    |                    |
           |<-------------------|                    |                    |
           |  PUBLISH Event:vq-rtcpxr F13            |                    |
           |------------------->|                    |                    |
           |                    | PUBLISH Event:vq-rtcpxr F14             |
           |                    |------------------->|                    |
           |                    |     200 OK F15     |                    |
           |                    |<-------------------|                    |
           |     200 OK F16     |                    |                    |
           |<-------------------|                    |                    |

       Figure 3. End of session report sent after session termination.
       In the message flow depicted in Figure 3, the following message is
       sent in F13.

    Pendleton, et al.            Standards Track                   [Page 35]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

           PUBLISH sip:collector@example.org SIP/2.0
           Via: SIP/2.0/UDP pc22.example.org;branch=z9hG4bK3343d7
           Max-Forwards: 70
           To: <sip:proxy@example.org>
           From: Alice <sip:alice@example.org>;tag=a3343df32
           Call-ID: 1890463548
           CSeq: 4331 PUBLISH
           Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
            SUBSCRIBE, NOTIFY
           Event: vq-rtcpxr
           Accept: application/sdp, message/sipfrag
           Content-Type: application/vq-rtcpxr
           Content-Length: ...

           VQSessionReport: CallTerm
           CallID: 6dg37f1890463
           LocalID: Alice <sip:alice@example.org>
           RemoteID: Bill <sip:bill@example.net>
           OrigID: Alice <sip:alice@example.org>
           LocalGroup: example-phone-55671
           RemoteGroup: example-gateway-09871
           LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d
           LocalMAC: 00:1f:5b:cc:21:0f
           RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd
           RemoteMAC: 00:26:08:8e:95:02
           LocalMetrics:
           Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
           SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
                           FMTP="annexb=no" PLC=3 SSUP=on
           JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
           PacketLoss:NLR=5.0 JDR=2.0
           BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
           Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
           Signal:SL=-21 NL=-50 RERL=55
           QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=4.2 MOSCQ=4.3
             QoEEstAlg=P.564
           RemoteMetrics:
           Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
           SessionDesc:PT=18 PD=G729 SR=8000 FD=20 FO=20 FPP=2 PPS=50
                           FMTP="annexb=no" PLC=3 SSUP=on
           JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
           PacketLoss:NLR=5.0 JDR=2.0
           BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
           Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
           Signal:SL=-21 NL=-45 RERL=55
           QualityEst:RLQ=90 RCQ=85 MOSLQ=4.3 MOSCQ=4.2 QoEEstAlg=P.564
           DialogID:1890463548@alice.example.org;to-tag=8472761;
              from-tag=9123dh311

    Pendleton, et al.            Standards Track                   [Page 36]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

    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              |
           |                    |------------------->|                    |
           |                    |     200 OK F9      |                    |
           |                    |<-------------------|                    |
           |     200 OK F10     |                    |                    |
           |<-------------------|                    |                    |
           |                    |                    |                    |
           |      BYE F11       |                    |                    |
           |------------------->|      BYE F12       |                    |
           |                    |---------------------------------------->|
           |                    |     200 OK F13     |                    |
           |                    |<----------------------------------------|
           |     200 OK F14     |                    |                    |
           |<-------------------|                    |                    |

       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

    Pendleton, et al.            Standards Track                   [Page 37]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

           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
           CallID: 6dg37f1890463
           LocalID: Alice <sip:alice@example.org>
           RemoteID: Bill <sip:bill@example.org>
           OrigID: Alice <sip:alice@example.org>
           LocalGroup: example-phone-55671
           RemoteGroup: example-gateway-09871
           LocalAddr: IP=10.10.1.100 PORT=5000 SSRC=1a3b5c7d
           LocalMAC: 00:1f:5b:cc:21:0f
           RemoteAddr:IP=11.1.1.150 PORT=5002 SSRC=0x2468abcd
           RemoteMAC: 00:26:08:8e:95:02
           Metrics:
           Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
           SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
                           PLC=3 SSUP=on
           JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
           PacketLoss:NLR=5.0 JDR=2.0
           BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
           Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
           Signal:SL=-12 NL=-30 RERL=55
           QualityEst:RLQ=60 RCQ=55 EXTR=90 MOSLQ=2.4 MOSCQ=2.3
              QoEEstAlg=P.564
           RemoteMetrics:
           Timestamps:START=2004-10-10T18:23:43Z STOP=2004-10-01T18:26:02Z
           SessionDesc:PT=0 PD=PCMU SR=8000 FD=20 FO=160 FPP=1 PPS=50
                           PLC=3 SSUP=on
           JitterBuffer:JBA=3 JBR=2 JBN=40 JBM=80 JBX=120
           PacketLoss:NLR=5.0 JDR=2.0
           BurstGapLoss:BLD=0 BD=0 GLD=2.0 GD=500 GMIN=16
           Delay:RTD=200 ESD=140 SOWD=200 IAJ=2 MAJ=10
           Signal:SL=-23 NL=-60 RERL=55
           QualityEst:RLQ=90 RCQ=85 EXTRI=90 MOSLQ=4.2 MOSCQ=4.3
              QoEEstAlg=P.564
           DialogID:1890463548@alice.example.org;to-tag=8472761;
                   from-tag=9123dh3111

    Pendleton, et al.            Standards Track                   [Page 38]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

    4.8.  Configuration Dataset for vq-rtcpxr Events

       It is the suggestion of the authors that the SIP configuration
       framework [15] be used to establish the necessary parameters for
       usage of vq-rtcpxr events.  A dataset for this purpose should be
       designed and documented in a separate document upon completion of the
       framework.

    5.  IANA Considerations

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

    5.1.  SIP Event Package Registration

          Package name: vq-rtcpxr
          Type: package
          Contact: Amy Pendleton <aspen@telchemy.com>
          Published Specification: This document

    5.2.  application/vq-rtcpxr Media Type Registration

       Type name: application
       Subtype name: vq-rtcpxr
       Required parameters: none
       Optional parameters: none
       Encoding considerations: 7 bit
       Security considerations: See next section.
       Interoperability considerations: none.
       Published specification: This document.

       Applications that 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"

       Person and email address for further information: Amy Pendleton
          <aspen@telchemy.com>

       Intended usage: COMMON

       Author / Change controller: The IETF.

    Pendleton, et al.            Standards Track                   [Page 39]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

    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,
       Shane Holthaus, and Jack Ford for their comments and input.

    8.  References

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

       [6]   Crocker, D. and P. Overell, "Augmented BNF for Syntax
             Specifications: ABNF", STD 68, RFC 5234, January 2008.

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

       [8]   Niemi, A., "Session Initiation Protocol (SIP) Extension for
             Event State Publication", RFC 3903, October 2004.

       [9]   ITU-T G.1020, "Performance parameter definitions for quality of
             speech and other voiceband applications utilizing IP networks".

    Pendleton, et al.            Standards Track                   [Page 40]

    RFC 6035         SIP Package for Voice Quality Reporting   November 2010

       [10]  ITU-T P.564, "Conformance testing for voice over IP
             transmission quality assessment models".

       [11]  ITU-T G.107, "The E-model, a computational model for use in
             transmission planning".

       [12]  Almes, G., Kalidindi, S., and M. Zekauskas, "A One-way Delay
             Metric for IPPM", RFC 2679, September 1999.

    8.2.  Informative References

       [13]  Westerlund, M. and S. Wenger, "RTP Topologies", RFC 5117,
             January 2008.

       [14]  Hilt, V., Noel, E., Shen, C., and A. Abdelal, "Design
             Considerations for Session Initiation Protocol (SIP) Overload
             Control", Work in Progress, July 2009.

       [15]  Petrie, D. and S. Channabasappa, "A Framework for Session
             Initiation Protocol User Agent Profile Delivery", Work
             in Progress, October 2010.

       [16]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session
             Traversal Utilities for NAT (STUN)", RFC 5389, October 2008.

    Authors' Addresses

       Amy Pendleton
       Telchemy Incorporated
       EMail: aspen@telchemy.com

       Alan Clark
       Telchemy Incorporated
       EMail: alan.d.clark@telchemy.com

       Alan Johnston
       Avaya
       St. Louis, MO  63124
       EMail: alan.b.johnston@gmail.com

       Henry Sinnreich
       Unaffiliated
       EMail: henry.sinnreich@gmail.com

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