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Network Working Group                                              Q. Wu
Internet-Draft                                                    Huawei
Intended status: Standards Track                                 G. Zorn
Expires: April 16, 2011                                      Network Zen
                                                               R. Schott
                                           Deutsche Telekom Laboratories
                                                        October 13, 2010


RTP Control Protocol Extended Reports (RTCP XR) Report Blocks for Real-
                     time Video Quality Monitoring
               draft-wu-avt-rtcp-xr-quality-monitoring-04

Abstract

   This document defines a set of RTP Control Protocol Extended Reports
   (RTCP XR) Report Blocks and associated SDP parameters allowing the
   report of video quality metrics, primarily for video applications of
   RTP.

Status of this Memo

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

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

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

   This Internet-Draft will expire on April 16, 2011.

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



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   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.1.  Standards Language . . . . . . . . . . . . . . . . . . . .  4
     2.2.  Acronyms . . . . . . . . . . . . . . . . . . . . . . . . .  5
   3.  Applicability  . . . . . . . . . . . . . . . . . . . . . . . .  5
   4.  Transport Layer Metrics  . . . . . . . . . . . . . . . . . . .  5
     4.1.  RTP Flows Initial Synchronization Delay Report Block . . .  6
     4.2.  RTP Flow General Synchronization Offset Metrics Block  . .  7
     4.3.  Layered Streams Statistics Metrics Block . . . . . . . . .  8
   5.  Application Layer Metrics  . . . . . . . . . . . . . . . . . .  9
     5.1.  RTP Streams Statistics Summary Report Block  . . . . . . .  9
     5.2.  Video Stream Loss and Discard Metrics Block  . . . . . . . 11
     5.3.  Video Stream Burst Metrics Block . . . . . . . . . . . . . 13
     5.4.  Synthetical Multimedia Quality Metrics Block . . . . . . . 15
   6.  SDP Signaling  . . . . . . . . . . . . . . . . . . . . . . . . 17
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 19
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 20
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 20
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 20
     10.1. Normative References . . . . . . . . . . . . . . . . . . . 20
     10.2. Informative References . . . . . . . . . . . . . . . . . . 21
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 22






















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1.  Introduction

   Along with the wide deployment of broadband access and the
   development of new IPTV services (e.g., broadcast video, video on
   demand), there is increasing interest in monitoring and managing
   networks and applications that deliver real-time applications over
   IP, to ensure that all end users obtain acceptable video/audio
   quality.  The main drives come from operators, since offering
   performance monitoring capability can help diagnose network
   impairments, facilitate in root cause analysis and aid in verifying
   compliance with service level agreements (SLAs) between Internet
   Service Providers (ISPs) and content providers.

   The factors that affect real-time application quality can be split
   into two categories.  The first category consists of transport-
   dependent factors such as packet loss, delay and jitter (which also
   translates into losses in the playback buffer).  The factors in the
   second category are application-specific factors that affect video
   quality and are sensitivity to network errors.  These factors can be
   but not limited to video codec and loss recovery technique, coding
   bit rate, packetization scheme, and content characteristics.

   Compared with application-specific factors, the transport-dependent
   factors sometimes are not sufficient to measure video quality, since
   the ability to analyze the video in the application layer provides
   quantifiable measurements for subscriber Quality of Experience (QoE)
   that may not be captured in the transmission layers or from the RTP
   layer down.  In a typical scenario, monitoring of the transmission
   layers can produce statistics suggesting that quality is not an
   issue, such as the fact that network jitter is not excessive.
   However, problems may occur in the service layers leading to poor
   subscriber QoE.  Therefore monitoring using only network-level
   measurements may be insufficient when application layer video quality
   is required.

   In order to provide accurate measures of video quality for operators
   when transporting video across a network, the video quality Metrics
   is highly required which can be conveyed in the RTCP XR
   packets[RFC3611] and may have the following three benefits:

      *  Tuning the video encoder algorithm to satisfy video quality
         requirements
      *  Determining which system techniques to use in a given situation
         and when to switch from one technique to another as system
         parameters change
      *  Verifying the continued correct operation of an existing system

   RFC 3611 [RFC3611] defines seven report block formats for network



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   management and quality monitoring.  However, there are no block types
   specifically designed for conveying video quality metrics.  This
   document focuses on specifying new report block types used to convey
   video-specific quality metrics.

   The report block types defined in this document fall into two
   categories.  The first category consists of general information
   regarding transmission quality, to be generated and processed by the
   RTP transport.  The report blocks in the second category convey
   metrics above transport that affect video quality and are sensitivity
   to network errors.

   Seven report block formats are defined by this document.  Of these,
   three are transport layer metrics:

      *  RTP Flows Initial Synchronization Delay Report Block
      *  Audio-Video Playout Offset Report Block
      *  Layered Streams Statistics Metrics Block

   The other four are application layer metrics:

      *  Video Statistics Summary Report Block
      *  Video Stream Loss and Discard Metrics Block
      *  Video Stream Burst Metrics Block
      *  Synthetical Multimedia Quality Metrics Block


2.  Terminology

2.1.  Standards Language

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

   In addition, the following terms are defined:

   Layered Component Packet

      a RTP packet using layered codecs containing the specified layered
      component, e.g., encoded stream at the base layer or at the
      enhancement layer.

   Picture Type

      Picture types used in the different video algorithms compose of
      the key-frame and the Derivation frame.  Key-frame is also called
      as reference frame and used as a reference for predicting other



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      pictures.  It is coded without prediction from other pictures.
      The Derivation frame is derived from Key-frame using prediction
      from the reference frame.


2.2.  Acronyms

   SSRC
      Synchronization Source [RFC3550]

   TS
      Transport Stream [ISO-IEC.13818-1.2007]


3.  Applicability

   All the report blocks defined in this document could be used by
   dedicated network monitoring applications.  As specified in RFC 3611
   [RFC3611], for such an application it might be appropriate to allow
   more than 5% of RTP data bandwidth to be used for RTCP packets, thus
   allowing proportionately larger and more detailed report blocks.

   The Flows General Synchronization Offset Block Section 4.2 has been
   defined for various multimedia applications.  Such applications can
   use this report block to monitor offset between two RTP streams
   synchronization to ensure satisfactory QoE.  Tighter tolerances than
   typically used have been recommended for such applications.

   The Flows Synchronization Delay Report Block has been defined
   primarily for layered or multi-description video coding applications.
   When joining a layered video session in such an application, a
   receiver may not synchronize playout across the multimedia session
   until RTCP SR packets have been received on all of the component RTP
   sessions.  This report block can be used to ensure synchronization
   between different media layers for the same multimedia session.

   The Video Stream Loss and Discard Metrics Report Block, Video Stream
   Burst Metrics report Block, Video Statistics Summary Report Block and
   Layered Video Statistics Metrics Block can be applied to any real
   time video application, while Synthetical Multimedia Quality Metrics
   Report Block can be used in any real-time AV application .


4.  Transport Layer Metrics







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4.1.  RTP Flows Initial Synchronization Delay Report Block

   This block reports the initial synchronization delay between RTP
   sessions of the same media stream sent using Multi-Session
   Transmission [I-D.ietf-avt-rtp-svc] or the initial synchronization
   delay betwen RTP session of the different media types
   [I-D.ietf-avt-rapid-rtp-sync], which is beyond the information
   carried in the standard RTCP packet format.  Information is recorded
   about session bandwidth and synchronization delay.

   The RTP Flows Intial Synchronization Delay Report Block has the
   following format:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=TBD    |   Reserved    |          Block length         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                      SSRC of Sender                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Initial Synchronization Delay                   |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Block type (BT): 8 bits
      The Statistics Summary Report Block is identified by the constant
      <RFISD>.

   Reserved: 8 bits
      This field is reserved for future definition.  In the absence of
      such a definition, the bits in this field MUST be set to zero and
      MUST be ignored by the receiver.

   Block length: 16 bits
      The constant 3, in accordance with the definition of this field in
      Section 3 of RFC 3611 [RFC3611].

   SSRC of Sender: 32 bits
      The SSRC of the RTP data packet source being reported upon by this
      report block.  (Section 4.1 of [RFC3611]).

   Initial Synchronization Delay: 32 bits
      The average delay, expressed in units of 1/65536 seconds, between
      the RTCP packets received on all of the components RTP sessions
      and the beginning of session [I-D.ietf-avt-rapid-rtp-sync].  The
      value is calculated as follows:






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         The average time, expressed in units of 1/65536 seconds, taken
         to receive the first RTCP packet in the RTP session with the
         longest RTCP reporting interval [I-D.ietf-avt-rapid-rtp-sync]

4.2.  RTP Flow General Synchronization Offset Metrics Block

   In an RTP multimedia session, there can be an arbitrary number of
   streams, with the same RTCP CNAME.  This block reports the general
   Synchronization offset requirements of these RTP streams beyond the
   information carried in the standard RTCP packet format.  Information
   is recorded about the synchronization offset time of each RTP stream
   relative to the reference RTP stream with the same CNAME and General
   Synchronisation Offset of zero..  The RTP Flow General
   Synchronization Offset Report Block has the following format:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=TBD    |I|  Reserved   |         Block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                General Synchronization Offset                 |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Block type (BT): 8 bits
      The Statistics Summary Report Block is identified by the constant
      <AVPO>.

   Interval Metric flag (I): 1 bit

      This field is used to indicate whether the Audio-Video
      synchronization metrics are Interval or Cumulative metrics, that
      is, whether the reported values applies to the most recent
      measurement interval duration between successive metrics reports
      (I=1) (the Interval Duration) or to the accumulation period
      characteristic of cumulative measurements (I=0) (the Cumulative
      Duration).

   Reserved: 8 bits
      This field is reserved for future definition.  In the absence of
      such a definition, the bits in this field MUST be set to zero and
      MUST be ignored by the receiver.








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   Block length: 16 bits
      The constant 2, in accordance with the definition of this field in
      Section 3 of RFC 3611 [RFC3611].

   SSRC of source: 32 bits
      As defined in Section 4.1 of RFC 3611 [RFC3611].

   General synchronization offset: 32 bits
      This field indicates the synchronization offset time of one RTP
      stream in milliseconds relative to the reference RTP stream with
      the same CNAME and General Synchronisation Offset of zero
      [I-D.ietf-avt-rapid-rtp-sync] This value is calculated based on
      the interarrival time between arbitray RTP packet and the
      reference RTP packet with the same CNAME , and timestamps of this
      arbitray RTP packet and the reference RTP packet with the same
      CNAME.

4.3.  Layered Streams Statistics Metrics Block

   This block reports layered streams statistics beyond the information
   carried in the Statistics Summary Report Block RTCP packet specified
   in the section 4.6 of RFC 3611 [RFC3611].  Information is recorded
   about lost layered component packets, duplicated layered component
   packets.  Such information can be useful for network management and
   video quality monitoring.

   The report block contents are dependent upon a series of flag bits
   carried in the first part of the header.  Not all parameters need to
   be reported in each block.  Flags indicate which parameters are
   reported and which are not.  The fields corresponding to unreported
   parameters MUST be present, but are set to zero.  The receiver MUST
   ignore any Layered Streams Statistics Metrics Block with a non-zero
   value in any field flagged as unreported.

   The Layered Stream Statistics metrics Block has the following format:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=TBD    |T|     rsd.    |        block length           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 Lost_Layered Component Packets                |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                  Dup Layered Component_Packets                |



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   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Block type (BT): 8 bits
      The Layered stream Statistics Metrics Block is identified by the
      constant <LSSM>.

   Layer Type flag (T): 1 bits
      This field is used to indicate the Layer Type of layered video to
      be reported.  LT is set to 0 if the loss_component_packet field
      and dup_component packet contain the base layer packet in layered
      codecs,e.g, SVC in [I-D.ietf-avt-rtp-svc], 1 if the loss_component
      packet field and dup_component packet contain enhancement layer
      packet in layered codec.

   Rsd.: 3 bits
      This field is reserved for future definition.  In the absence of
      such a definition, the bits in this field MUST be set to zero and
      MUST be ignored by the receiver.

   Block length: 16 bits
      The constant 3, in accordance with the definition of this field in
      Section 3 of RFC 3611 [RFC3611].

   SSRC of source: 32 bits
      As defined in Section 4.1 of RFC 3611 [RFC3611].

   begin_seq: 16 bits
      As defined in Section 4.1 of RFC 3611 [RFC3611].

   end_seq: 16 bits
      As defined in Section 4.1 of RFC 3611 [RFC3611].

   Lost_Layered Component Packets: 32 bits
      Number of lost_component packets in the above sequence number
      interval.

   Dup_Layered Component Packets: 32 bits
      Number of dup_component packets in the above sequence number
      interval.



5.  Application Layer Metrics

5.1.  RTP Streams Statistics Summary Report Block

   This block reports statistics beyond the information carried in the
   Statistics Summary Report Block RTCP packet specified in the section



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   4.6 of RFC 3611 [RFC3611].  Information is recorded about lost frame
   packets, duplicated frame packets, lost layered component packets,
   duplicated layered component packets.  Such information can be useful
   for network management and video quality monitoring.

   The report block contents are dependent upon a series of flag bits
   carried in the first part of the header.  Not all parameters need to
   be reported in each block.  Flags indicate which parameters are
   reported and which are not.  The fields corresponding to unreported
   parameters MUST be present, but are set to zero.  The receiver MUST
   ignore any Video Statistics Summary Report Block with a non-zero
   value in any field flagged as unreported.

   The RTP Streams Statistics Summary Report Block has the following
   format:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=TBD    |T|P|    rsd.   |        block length           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         lost_frames                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          dup frames                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    lost_partial frame packets                 |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                     dup partial frame_packets                 |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Block type (BT): 8 bits
      The Video Statistics Summary Report Block is identified by the
      constant <VSS>.

   Picture type indicator (T): 1 bits
      Picture types used in the different video algorithms compose of
      key-frame and derivation frame.  This field is used to indicate
      the frame type to be reported.  Bits set to 0 if the lost_frames
      field or dup_frames field contain a key_frame report or reference
      frame report, 1 if the lost_frames field and dup_frames field
      contain other derivation frame report.






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   P: 1 bit
      Bit set to 1 if the lost_partial frame packets field or the
      dup_partial_frame packets field contains a report, 0 otherwise.

   Rsd.: 3 bits
      This field is reserved for future definition.  In the absence of
      such a definition, the bits in this field MUST be set to zero and
      MUST be ignored by the receiver.

   Block length: 16 bits
      The constant 5, in accordance with the definition of this field in
      Section 3 of RFC 3611 [RFC3611].

   SSRC of source: 32 bits
      As defined in Section 4.1 of RFC 3611 [RFC3611].

   begin_seq: 16 bits
      As defined in Section 4.1 of RFC 3611 [RFC3611].

   end_seq: 16 bits
      As defined in Section 4.1 of RFC 3611 [RFC3611].

   lost_frames: 32 bits
      Number of lost_frames in the above sequence number interval.

   dup_frames: 32 bits
      Number of dup_frames in the above sequence number interval.

   lost_partial frame packets: 32 bits
      Number of lost_partial frame packets in the above sequence number
      interval.

   dup_partial frame packets: 32 bits
      Number of dup_partial frame packets in the above sequence number
      interval.


5.2.  Video Stream Loss and Discard Metrics Block

   This block reports Loss and Discard metrics statistics beyond the
   information carried in the standard RTCP packet format.  The block
   reports separately on packets lost on the IP channel, and those that
   have been received but then discarded by the receiving jitter buffer.

   It is very useful to distinguish between packets lost by the network
   and those discarded due to jitter.  Both have equal effect on the
   quality of the video stream, however, having separate counts helps
   identify the source of quality degradation.  These fields MUST be



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   populated, and MUST be set to zero if no packets have been received.

   Implementations MUST provide values for all the fields defined here.
   For certain metrics, if the value is undefined or unknown, then the
   specified default or unknown field value MUST be provided.

   The block is encoded as six 32-bit words:

    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=TBD    |T |  reserved  |        block length           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          Loss rate            |        Discard rate           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   block type (BT): 8 bits
      A Video Stream Metrics Report Block is identified by the constant
      <VSLDM>.

   Picture type indicator (T): 1 bits
      Picture types used in the different video algorithms compose of
      key-frame and derivation frame.  This field is used to indicate
      the picture type to be reported.  Bits set to 0 if the Loss rate
      field and discard rate field contain a Key_frame report or
      reference frame report, 1 if the Loss rate field and discard rate
      field contain other derivation frame reports.

   reserved: 6 bits
      This field is reserved for future definition.  In the absence of
      such a definition, the bits in this field MUST be set to zero and
      MUST be ignored by the receiver.

   block length: 16 bits
      The constant 1, in accordance with the definition of this field in
      Section 3 of RFC 3611 [RFC3611].

   SSRC of source: 32 bits
      The SSRC of the RTP data packet source being reported upon by this
      report block. in accordance with the definition of this field in
      Section 3 of RFC 3611 [RFC3611].
   Loss rate: 8 bits
      The fraction of RTP data packets from the source lost since the
      beginning of reception, expressed as a fixed point number with the
      binary point at the left edge of the field.  This value is
      calculated by dividing the total number of lost packets containing



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      specified frame (e.g., Key frame) (after the effects of applying
      any error protection such as FEC) by the total number of packets
      expected, multiplying the result of the division by 256, limiting
      the maximum value to 255 (to avoid overflow), and taking the
      integer part.  The numbers of duplicated packets and discarded
      packets do not enter into this calculation.  Since receivers
      cannot be required to maintain unlimited buffers, a receiver MAY
      categorize late-arriving packets as lost.  The degree of lateness
      that triggers a loss SHOULD be significantly greater than that
      which triggers a discard.

   Discard rate: 8 bits
      The fraction of RTP data packets from the source that have been
      discarded since the beginning of reception, due to late or early
      arrival, under-run or overflow at the receiving jitter buffer.
      This value is expressed as a fixed point number with the binary
      point at the left edge of the field.  It is calculated by dividing
      the total number of discarded packets containing specified frame
      (e.g., Key Frame) (excluding duplicate packet discards) by the
      total number of packets expected, multiplying the result of the
      division by 256, limiting the maximum value to 255 (to avoid
      overflow), and taking the integer part.


5.3.  Video Stream Burst Metrics Block

   This block reports Burst metrics statistics beyond the information
   carried in the standard RTCP packet format.  It reports on the
   combined effect of losses and discards, as both have equal effect on
   video quality.

   In order to properly assess the quality of a video stream, it is
   desirable to consider the degree of burstiness of packet loss RFC
   3357 [RFC3357].  Following the one-way loss pattern sample metrics
   discussed in [RFC3357], a measure of the spacing between consecutive
   network packet loss or error events, is a "loss distance".  The loss
   distance metric captures the spacing between the loss periods.  The
   duration of a loss or error event (e.g. and how many packets are lost
   in that duration) is a "loss period", the loss period metric captures
   the frequency and length (burstiness) of loss once it starts.  Delay
   reports include the transit delay between RTP end points and the end
   system processing delays, both of which contribute to the user
   perceived delay.

   Implementations MUST provide values for all the fields defined here.
   For certain metrics, if the value is undefined or unknown, then the
   specified default or unknown field value MUST be provided.




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   The block is encoded as six 32-bit words:

   0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=TBD    |  Reserved     |        block length           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          Loss Distance        |          Loss Period          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |      Max Loss Duration        |           Reserved.           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   block type (BT): 8 bits
      A Video Stream Metrics Report Block is identified by the constant
      <VSBM>.

   reserved: 8 bits
      This field is reserved for future definition.  In the absence of
      such a definition, the bits in this field MUST be set to zero and
      MUST be ignored by the receiver.

   block length: 16 bits
      The constant 2, in accordance with the definition of this field in
      Section 3 of RFC 3611 [RFC3611].

   SSRC of source: 32 bits
      The SSRC of the RTP data packet source being reported upon by this
      report block. in accordance with the definition of this field in
      Section 3 of RFC 3611 [RFC3611].

   Loss Distance: 16 bits
      The mean duration, expressed in milliseconds, of the loss
      intervals that have occurred since the beginning of reception
      [DSLF].  The duration of each loss distance is calculated based
      upon the frames that mark the beginning and end of that period.
      It is equal to the timestamp of the end frame, plus the duration
      of the end frame, minus the timestamp of the beginning frame.  If
      the actual values are not available, estimated values MUST be
      used.  If there have been no burst periods, the burst duration
      value MUST be zero.

   Loss Period: 16 bits
      The mean duration, expressed in milliseconds, of the burst loss
      periods that have occurred since the beginning of reception
      [DSLF].  The duration of each period is calculated based upon the
      frame that marks the end of the prior burst loss and the frame



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      that marks the beginning of the subsequent burst loss.  It is
      equal to the timestamp of the subsequent burst frame, minus the
      timestamp of the prior burst packet, plus the duration of the
      prior burst packet.  If the actual values are not available,
      estimated values MUST be used.  In the case of a gap that occurs
      at the beginning of reception, the sum of the timestamp of the
      prior burst packet and the duration of the prior burst packet are
      replaced by the reception start time.  In the case of a gap that
      occurs at the end of reception, the timestamp of the subsequent
      burst packet is replaced by the reception end time.  If there have
      been no gap periods, the gap duration value MUST be zero.

   Max Loss Duration of a single error: 16 bits
      The maximum loss duration, expressed in milliseconds, of the loss
      periods that have occurred since the beginning of reception.  The
      recommended max loss duration is specified as less than 16 ms in
      [DSLF], which provides a balance between interleaver depth
      protection from xDSL errors induced by impulse noise, delay added
      to other applications and video service QoE requirements to reduce
      visible impairments.

   Reserved: 16 bits
      All bits SHALL be set to 0 by the sender and SHALL be ignored on
      reception.

   block length: 16 bits
      The constant 2, in accordance with the definition of this field in
      Section 3 of RFC 3611 [RFC3611].


5.4.  Synthetical Multimedia Quality Metrics Block

   This block reports the multimedia quality metrics beyond the
   information carried in the standard RTCP packet format.  Information
   is recorded about Video MOS, Audio MOS, Audio Video MOS, Video
   Service Transmission Quality [G.1082][P.NAMS].

   The report block contents are dependent upon a series of flag bits
   carried in the first part of the header.  Not all parameters need to
   be reported in each block.  Flags indicate which are and which are
   not reported.  The fields corresponding to unreported parameters MUST
   be present, but are set to zero.  The receiver MUST ignore any
   Perceptual Quality Metrics Block with a non-zero value in any field
   flagged as unreported.

   The Synthetical Multimedia Quality Metrics Block has the following
   format:




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    0                   1                   2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     BT=TBD    |I|V|A|M|T|Rsd. |        block length           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              MOS-V            |           MOS-A               |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              MOS-AV           |           VSTQ                |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   Block type (BT): 8 bits
      The Perceptual Quality Metrics Block is identified by the constant
      <SMQM>.

   Interval Metric flag (I): 1 bit
      This field is used to indicate whether the Basic Loss/Discard
      metrics are Interval or Cumulative metrics, that is, whether the
      reported values applies to the most recent measurement interval
      duration between successive metrics reports (I=1) (the Interval
      Duration) or to the accumulation period characteristic of
      cumulative measurements (I=0) (the Cumulative Duration).

   MOS-V flag (V): 1 bit
      Bit set to 1 if the MOS-V field and MOS-AV field contain a report,
      0 otherwise.

   MOS-A flag (A): 1 bit
      Bit set to 1 if the MOS-A field contain a report, 0 otherwise.

   MOS-AV flag (M): 1 bit
      Bit set to 1 if the MOS-AV field contain a report, 0 otherwise.

   Video Service Transmission Quality flag (T): 1 bit
      Bit set to 1 if the VSTQ field contains a report, 0 otherwise.

   Rsd.: 3 bits
      This field is reserved for future definition.  In the absence of
      such a definition, the bits in this field MUST be set to zero and
      MUST be ignored by the receiver.

   SSRC of source: 32 bits
      As defined in Section 4.1 of [RFC3611].







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   MOS-V: 16 bits
      The estimated mean opinion score for video quality (MOS-V) is a
      video quality metric on a scale from 1 to 5, in which 5 represents
      excellent and 1 represents unacceptable [G.1082][P.NAMS].  This
      metric is defined as not including the effects of audio
      impairments and can be compared to MOS scores obtained from video
      quality tests.  It is expressed as an integer in the range 10 to
      50, corresponding to MOS x 10.  For example, a value of 35 would
      correspond to an estimated MOS score of 3.5.

      A value of 127 indicates that this parameter is unavailable.
      Values other than 127 and the valid range defined above MUST NOT
      be sent and MUST be ignored by the receiving system.

   MOS-A: 16 bits
      The estimated mean opinion score for Audio quality (MOS-A) is
      defined as including the effects of delay and other effects that
      would affect Audio-Video quality [RFC3611].  It is expressed as an
      integer in the range 10 to 50, corresponding to MOS x 10, as for
      MOS-A.

      A value of 127 indicates that this parameter is unavailable.
      Values other than 127 and the valid range defined above MUST not
      be sent and MUST be ignored by the receiving system.

   MOS-AV: 16 bits

      The estimated mean opinion score for Audio-Video quality (MOS-AV)
      is defined as including the effects of delay and other effects
      that would affect Audio-Video quality [G.1082][P.NAMS].  It is
      expressed as an integer in the range 10 to 50, corresponding to
      MOS x 10, as for MOS-AV.  A value of 127 indicates that this
      parameter is unavailable.  Values other than 127 and the valid
      range defined above MUST NOT be sent and MUST be ignored by the
      receiving system.

   VSTQ: 16 bits
      Video Service Transmission Quality (TBC) .



6.  SDP Signaling

   Six new parameters are defined for the six report blocks defined in
   this document to be used with Session Description Protocol (SDP)
   [RFC4566] using the Augmented Backus-Naur Form (ABNF) [RFC5234].
   They have the following syntax within the "rtcp-xr" attribute
   [RFC3611]:



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   rtcp-xr-attrib =  "a=rtcp-xr:"
                     [xr-format *(SP xr-format)] CRLF

      xr-format = RTP-flows-syn
                  / audio-video-ofset
                  / multimedia-quality-metrics
                  / video-stream-loss-metrics
                  / video-stream-burst-metrics
                  / video-stat-summary
                  / layered-video-stat-metrics

         RTP-flows-syn = "RTP-flows-syn"
                         ["=" max-size]
            max-size = 1*DIGIT ; maximum block size in octets

         audio-video-ofset = "audio-video-ofset"
                           ["=" max-size]
            max-size = 1*DIGIT ; maximum block size in octets

         video-stream-burst-metrics = "video-stream-burst-metrics"
                               ["=" max-size]
            max-size = 1*DIGIT ; maximum block size in octets

         video-stream-loss-metrics = "video-stream-loss-metrics"
                                 ["=" stat-flag *("," stat-flag)]
               stat-flag = "key Frame loss and duplication"
                           / "derivation Frame loss and duplication"

         video-stat-summary = "video-stat-summary"
                                 ["=" stat-flag *("," stat-flag)]
               stat-flag = "key Frame loss and duplication"
                           / "derivation Frame loss and duplication"


         layered-stream-stat-metrics = "layered-stream-stat-metrics"
                              ["=" stat-flag *("," stat-flag)]
               stat-flag = "base layer packet"
                           / "enhancment layer packet"

         multimedia-quality-metrics = "multimedia-quality-metrics"
                               ["=" stat-flag *("," stat-flag)]
            stat-flag = "Interval Metric"
                        / "MOS-V"
                        / "MOS-A"
                        / "MOS-AV"
                        / "VSTQ"

   Refer to Section 5.1 of RFC 3611 [RFC3611] for a detailed description



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   and the full syntax of the "rtcp-xr" attribute.


7.  IANA Considerations

   New report block types for RTCP XR are subject to IANA registration.
   For general guidelines on IANA allocations for RTCP XR, refer to
   Section 6.2 of [RFC3611].

   This document assigns six new block type values in the RTCP XR Block
   Type Registry:

      Name:       RFISD
      Long Name:  RTP Flows Initial Synchronization Delay
      Value       <RFISD>
      Reference:  Section 4.1

      Name:       AVPO
      Long Name:  Audio-Video Playout Offset
      Value       <AVPO>
      Reference:  Section 4.2

      Name:       VSS
      Long Name:  Video Statistics Summary
      Value       <VSS>
      Reference:  Section 5.1

      Name:       LSSM
      Value       <LSSM>
      Long Name:  Layered Stream Statistics Metrics
      Reference:  Section 4.3

      Name:       VSLDM
      Long Name:  Video Stream Loss and Discard Metrics
      Value       <VSLDM>
      Reference:  Section 5.2

      Name:       VSBM
      Long Name:  Video Stream Burst Metrics
      Value       <VSBM>
      Reference:  Section 5.3

      Name:       SMQM
      Long Name:  Synthetical Multimedia Quality Metric







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      Value       <SMQM>
      Reference:  Section 5.4

   This document also registers seven SDP [RFC4566] parameters for the
   "rtcp-xr" attribute in the RTCP XR SDP Parameters Registry:

      *  "RTP-flows-syn"
      *  "audio-video-ofset"
      *  "multimedia-quality-metrics"
      *  "video-stream-loss-metrics"
      *  "video-stream-burst-metrics"
      *  "video-stat-summary"
      *  "layered-stream-stat-metrics"

   The contact information for the registrations is:

                    Qin Wu
                    sunseawq@huawei.com
                    101 Software Avenue, Yuhua District
                    Nanjing, JiangSu 210012 China


8.  Security Considerations

   TBC


9.  Acknowledgements

   The authors would like to thank Bill Ver Steeg, David R Oran, Ali
   Begen,Colin Perkins, Roni Even,Youqing Yang, Wenxiao Yu and Yinliang
   Hu for their valuable comments and suggestions on this document.


10.  References

10.1.  Normative References

   [I-D.ietf-avt-rapid-rtp-sync]
              Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
              Flows", draft-ietf-avt-rapid-rtp-sync-12 (work in
              progress), July 2010.

   [I-D.ietf-avt-rtp-svc]
              Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
              "RTP Payload Format for Scalable Video Coding",
              draft-ietf-avt-rtp-svc-23 (work in progress),
              October 2010.



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   [ISO-IEC.13818-1.2007]
              International Organization for Standardization,
              "Information technology - Generic coding of moving
              pictures and associated audio information: Systems",
              ISO International Standard 13818-1, October 2007.

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

   [RFC2250]  Hoffman, D., Fernando, G., Goyal, V., and M. Civanlar,
              "RTP Payload Format for MPEG1/MPEG2 Video", RFC 2250,
              January 1998.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3611]  Friedman, T., Caceres, R., and A. Clark, "RTP Control
              Protocol Extended Reports (RTCP XR)", RFC 3611,
              November 2003.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.

   [RFC5234]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234, January 2008.

10.2.  Informative References

   [DSLF]     Rahrer, T., Ed., Fiandra, Ed., and Wright, Ed., "Triple-
              play Services Quality of Experience (QoE) Requirements",
              DSL Forum Technical Report TR-126, December 2006.

   [G.1082]   ITU-T, "Measurement-based methods for improving the
              robustness of IPTV performance", ITU-T
              Recommendation G.1082, April 2009.

   [I-D.ietf-fecframe-interleaved-fec-scheme]
              Begen, A., "RTP Payload Format for 1-D Interleaved Parity
              FEC", draft-ietf-fecframe-interleaved-fec-scheme-09 (work
              in progress), January 2010.

   [I-D.ietf-fecframe-raptor]
              Waston, M., "Raptor FEC Schemes for FECFRAME",
              draft-ietf-fecframe-raptor-02 (work in progress),
              March 2010.

   [IEEE]     IEEE, "Human Perception of Jitter and Media



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              Synchronization", IEEE Journal on Selected Areas in
              Communications Vol. 14, No. 1, January 1996.

   [P.NAMS]   ITU-T, "Non-intrusive parametric model for the Assessment
              of performance of Multimedia Streaming", ITU-T
              Recommendation P.NAMS, November 2009.

   [RFC3357]  Koodli, R. and R. Ravikanth, "One-way Loss Pattern Sample
              Metrics", RFC 3357, August 2002.


Authors' Addresses

   Qin Wu
   Huawei
   101 Software Avenue, Yuhua District
   Nanjing, Jiangsu  210012
   China

   Email: sunseawq@huawei.com


   Glen Zorn
   Network Zen
   77/440 Soi Phoomjit, Rama IV Road
   Phra Khanong, Khlong Toie
   Bangkok  10110
   Thailand

   Phone: +66 (0) 87 502 4274
   Email: gwz@net-zen.net


   Roland Schott
   Deutsche Telekom Laboratories
   Deutsche-Telekom-Allee 7
   Darmstadt  64295
   Germany

   Email: Roland.Schott@telekom.de











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