Network Working Group                                              Q. Wu
Internet-Draft                                                    Huawei
Intended status: Standards Track                                 G. Zorn
Expires: September 3, 2011                                   Network Zen
                                                               R. Schott
                                           Deutsche Telekom Laboratories
                                                           March 2, 2011


RTP Control Protocol Extended Reports (RTCP XR) Report Blocks for Real-
                  time Application Quality Monitoring
             draft-wu-xrblock-rtcp-xr-quality-monitoring-01

Abstract

   This document defines a set of RTP Control Protocol Extended Reports
   (RTCP XR) Report Blocks and associated SDP parameters allowing the
   report of real time application 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
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   This Internet-Draft will expire on September 3, 2011.

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   Copyright (c) 2011 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

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   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.

   This document may contain material from IETF Documents or IETF
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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  . . . . . . . . . . . . . . . . . . .  6
     4.1.  RTP Flows Initial Synchronization Delay Report Block . . .  6
     4.2.  RTP Flows General Synchronization Offset Metrics Block . .  7
     4.3.  Layered Streams Statistics Metrics Block . . . . . . . . .  8
   5.  Application Layer Metrics  . . . . . . . . . . . . . . . . . . 10
     5.1.  Transport Streams Statistics Summary Report Block  . . . . 10
     5.2.  Transport Stream Loss and Discard Metrics Block  . . . . . 12
     5.3.  Transport Stream Burst Metrics Block . . . . . . . . . . . 14
     5.4.  Synthetical Multimedia Quality Metrics Block . . . . . . . 16
   6.  SDP Signaling  . . . . . . . . . . . . . . . . . . . . . . . . 18
   7.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 20
   8.  Security Considerations  . . . . . . . . . . . . . . . . . . . 21
   9.  Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 21
   10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 21
     10.1. Normative References . . . . . . . . . . . . . . . . . . . 21
     10.2. Informative References . . . . . . . . . . . . . . . . . . 22
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 23










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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
   RTP or IP, to ensure that all end users obtain acceptable video/audio
   quality.  The main drives come from operators and enterprises, 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 real
   time application (e.g., 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 real time data
   quality, since the ability to analyze the real time data 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 real time application
   quality for operators when transporting real time contents across a
   network, the synthentical multimedia quality Metrics is highly
   required which can be conveyed in the RTCP XR packets[RFC3611] and
   may have the following three benefits:

      *  Tuning the content encoder algorithm to satisfy real time data
         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




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      *  Verifying the continued correct operation of an existing system

   RFC 3611 [RFC3611] defines seven report block formats for network
   management and quality monitoring.  However, some of these metrics
   are mostly for multicast inference of network characteristics (MINC)
   or voice over IP (VoIP) monitoring and not widely applicable to other
   applications, e.g., video quality monitoring.  This document focuses
   on specifying new additional report block types used to convey QoE
   related parameters that is genericly designed for use in voice, audio
   and video services.

   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 real time application 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
      *  RTP Flows General Synchronization Offset Metrics Block
      *  Layered Streams Statistics Metrics Block

   The other four are application layer metrics:

      *  Transport Stream Statistics Summary Report Block
      *  Transport Stream Loss and Discard Metrics Block
      *  Transport 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.



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   Picture Type

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

   RTP Flows General Synchronization Offset Metrics Block in 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 RTP Flows Initial 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 measure
   synchronization between different media layers for the same
   multimedia session.

   The Transport Stream Loss and Discard Metrics Report Block, Transport
   Stream Burst Metrics report Block, Transport Statistics Summary
   Report Block and Layered Streams 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



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   AV application.


4.  Transport Layer Metrics

4.1.  RTP Flows Initial Synchronization Delay Report Block

   This block reports Initial synchronization delay beyond the
   information carried in the standard RTCP packet format.  Information
   is recorded about the the difference between the start of RTP
   sessions and the time the RTP receiver acquires all components of RTP
   sessions [RFC6051].  The components of RTP session are referred to as
   one RTP session for each media type or the media content in each
   layer contained in RTP Control Protocol (RTCP) sender report (SR)
   packets [RFC3550].  For unicast session, the delay due to negotiation
   of NAT pinholes, firewall holes, quality-of-service, and media
   security keys is contributed to such initial synchronization delay.
   For multicast session, the initial synchronization delay varies with
   the session bandwidth and the number of members, the number of
   senders in the session.  In the absence of packet loss, the initial
   synchronisation delay equals to the average time taken to receive the
   first RTCP packet in the RTP session with the longest RTCP reporting
   interval.In the presence of packet loss, the media synchronization
   needs to wait until the reporting interval has passed, and the next
   RTCP SR packet is sent.

   The RTP Flows Initial 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.




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   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 [RFC6051].  The value is calculated
      as follows:

         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 [RFC6051]

4.2.  RTP Flows 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 status 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.  For layered session or multimedia
   session,the first RTP packet can be chosen as the basic packet of
   reference RTP stream.  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
      <RFGSO>.







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   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.

   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 represents 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
      [RFC6051] 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.




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   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                |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   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].






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   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.  Transport 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
   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 Transport Streams Statistics Summary Report 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    |T|P|    rsd.   |        block length           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         lost_frames                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          dup frames                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        partial_lost_frames                    |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        partial_dup_frames                     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                  key frames impairement proportion            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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

   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.

   P: 1 bit
      Bit set to 1 if the partial_lost_frames field or the partial_dup_
      frames 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].







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   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.

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

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

   key frames impairment proportion:32bits
      The proportion of key frame impaired by packet loss,discard and
      duplication.


5.2.  Transport 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
   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:





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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    |T |  reserved  |        block length           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          Loss rate            |        Discard rate           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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

   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
      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



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      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.  Transport 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.

   The block is encoded as six 32-bit words:










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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    |  Reserved     |        block length           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          Loss Distance        |          Loss Period          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |      Max Loss Duration        |           Reserved.           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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

   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
      that marks the beginning of the subsequent burst loss.  It is
      equal to the timestamp of the subsequent burst frame, minus the



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      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 application performance or quality
   metrics beyond the information carried in the standard RTCP packet
   format.  Information is recorded about multimedia application QoE
   metric which is expressed as a MOS ("Mean Opinion Score"), MOS is on
   a scale from 1 to 5, in which 5 represents excellent and 1 represents
   unacceptable.  MOS scores are usually obtained using subjective
   testing or using objective algorithm to estimate the multimedia
   quality.  However Subjective testing is not suitable for measuring
   the multimedia quality since the results may vary from test to test.
   Therefore using objective algorithm to calculate MOS scores is
   recommended.  ITU-T recommendation [G.1082][P.NAMS][P.NBAMS] defines
   a methodology for verifying the performance of QoE estimation
   algorithms for video and audio.  Hence this document recommends
   vendors and implementers to use International Telecommunication Union
   (ITU)-specified methodologies to measure parameters when possible.

   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



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   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:

    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|   MC  | Rsd.|        block length           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         MOS Value                             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   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 Class (MC): 4 bits
      This field is used to indicate the MOS type to be reported.  The
      MOS type is defined as follows:
         0000 MOS-A - Audio Quality MOS [G.107][P.564].
         0001 MOS-V - Video Quality MOS [P.NAMS][P.NBAMS].
         0010 MOS-AV - Audio-Video Quality MOS[P.NAMS][P.NBAMS].
         0100~1111 - Reserved

   Rsd.: 7 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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   SSRC of source: 32 bits
      As defined in Section 4.1 of [RFC3611].

   MOS Value: Variable Length

      The estimated mean opinion score for Audio Qulity, Video Quality
      or Audio-Video quality is defined as including the effects of
      delay and other effects that would affect Audio-Video quality
      [G.1082][P.NAMS][P.NBAMS].  It is expressed as an integer in the
      range 10 to 50, corresponding to MOS x 10, as for MOS.  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.



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-init-syn
                    / RTP-flows-general-syn
                    / multimedia-quality-metrics
                    / transport-stream-loss-metrics
                    / transport-stream-burst-metrics
                    / transport-stat-summary
                    / layered-stream-stat-metrics

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

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

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

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

       transport-stream-stat-summary = "transport-stream-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 Metrics"
                           /"Cumulative metrics"

   Refer to Section 5.1 of RFC 3611 [RFC3611] for a detailed description
   and the full syntax of the "rtcp-xr" attribute.





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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:       RFGSO
      Long Name:  RTP Flows General Synchronization Offset Metrics Block
      Value       <RFGSO>
      Reference:  Section 4.2

      Name:       TSSS
      Long Name:  Transport Stream Statistics Summary
      Value       <TSSS>
      Reference:  Section 5.1

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

      Name:       TSLDM
      Long Name:  Transport Stream Loss and Discard Metrics
      Value       <TSLDM>
      Reference:  Section 5.2

      Name:       TSBM
      Long Name:  Transport Stream Burst Metrics
      Value       <TSBM>
      Reference:  Section 5.3

      Name:       SMQM
      Long Name:  Synthetical Multimedia Quality Metric
      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:





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      *  "RTP-flows-init-syn"
      *  "RTP-flows-general-syn"
      *  "multimedia-quality-metrics"
      *  "transport-stream-loss-metrics"
      *  "transport-stream-burst-metrics"
      *  "transport-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

   The new RTCP XR report blocks proposed in this document introduces no
   new security considerations beyond those described in [RFC3611].


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-rtp-svc]
              Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
              "RTP Payload Format for Scalable Video Coding",
              draft-ietf-avt-rtp-svc-27 (work in progress),
              February 2011.

   [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.



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   [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.

   [RFC6051]  Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
              Flows", RFC 6051, November 2010.

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.107]    ITU-T, "The E Model, a computational model for use in
              transmission planning", ITU-T Recommendation G.107,
              April 2009.

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

   [I-D.ietf-pmol-metrics-framework-02]
              Clark, A., "Framework for Performance Metric Development".

   [IEEE]     IEEE, "Human Perception of Jitter and Media
              Synchronization", IEEE Journal on Selected Areas in
              Communications Vol. 14, No. 1, January 1996.

   [P.564]    ITU-T, "Conformance testing for narrowband Voice over IP
              transmission quality assessment models", ITU-T
              Recommendation P.564, July 2006.

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



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              Recommendation P.NAMS, November 2009.

   [P.NBAMS]  ITU-T, "non-intrusive bit-stream model for assessment of
              performance of multimedia streaming", ITU-T
              Recommendation P.NBAMS, 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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