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rfc:rfc6035

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.

Pendleton, et al. Standards Track [Page 1] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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.

Pendleton, et al. Standards Track [Page 2] 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

Pendleton, et al. Standards Track [Page 3] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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.

Pendleton, et al. Standards Track [Page 4] 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

Pendleton, et al. Standards Track [Page 5] RFC 6035 SIP Package for Voice Quality Reporting November 2010

 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.

Pendleton, et al. Standards Track [Page 6] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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

Pendleton, et al. Standards Track [Page 7] RFC 6035 SIP Package for Voice Quality Reporting November 2010

 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

Pendleton, et al. Standards Track [Page 8] RFC 6035 SIP Package for Voice Quality Reporting November 2010

 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.

Pendleton, et al. Standards Track [Page 9] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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 ]

Pendleton, et al. Standards Track [Page 10] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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 ]

Pendleton, et al. Standards Track [Page 11] RFC 6035 SIP Package for Voice Quality Reporting November 2010

 [ 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)

Pendleton, et al. Standards Track [Page 12] RFC 6035 SIP Package for Voice Quality Reporting November 2010

; 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

Pendleton, et al. Standards Track [Page 13] RFC 6035 SIP Package for Voice Quality Reporting November 2010

; 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

Pendleton, et al. Standards Track [Page 14] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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

Pendleton, et al. Standards Track [Page 15] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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.

Pendleton, et al. Standards Track [Page 16] RFC 6035 SIP Package for Voice Quality Reporting November 2010

; 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 ]

Pendleton, et al. Standards Track [Page 17] RFC 6035 SIP Package for Voice Quality Reporting November 2010

 [ 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.

Pendleton, et al. Standards Track [Page 18] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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

Pendleton, et al. Standards Track [Page 19] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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

Pendleton, et al. Standards Track [Page 20] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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.

Pendleton, et al. Standards Track [Page 21] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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.

Pendleton, et al. Standards Track [Page 22] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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.

Pendleton, et al. Standards Track [Page 23] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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.

Pendleton, et al. Standards Track [Page 24] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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.

Pendleton, et al. Standards Track [Page 25] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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.

Pendleton, et al. Standards Track [Page 26] RFC 6035 SIP Package for Voice Quality Reporting November 2010

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

Pendleton, et al. Standards Track [Page 27] RFC 6035 SIP Package for Voice Quality Reporting November 2010

 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

Pendleton, et al. Standards Track [Page 41]

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