One document matched: draft-wu-avt-rtcp-xr-quality-monitoring-01.txt

Differences from draft-wu-avt-rtcp-xr-quality-monitoring-00.txt




Network Working Group                                              Q. Wu
Internet-Draft                                                    Huawei
Intended status: Standards Track                                 G. Zorn
Expires: January 10, 2011                                    Network Zen
                                                            July 9, 2010


RTP Control Protocol Extended Reports (RTCP XR) Report Blocks for Real-
                     time Video Quality Monitoring
               draft-wu-avt-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 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 January 10, 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
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as



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   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2.  Terminology  . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.1.  Standards Language . . . . . . . . . . . . . . . . . . . .  4
     2.2.  Acronyms . . . . . . . . . . . . . . . . . . . . . . . . .  4
   3.  Applicability  . . . . . . . . . . . . . . . . . . . . . . . .  4
   4.  RTP Flows Synchronization Delay Report Block . . . . . . . . .  5
   5.  Audio-Video Synchronization Report Block . . . . . . . . . . .  6
   6.  Video Statistics Summary Report Block  . . . . . . . . . . . .  7
   7.  TR 101 290 Decodability Metrics Report Block . . . . . . . . . 10
   8.  Video Stream Metrics Report Block  . . . . . . . . . . . . . . 13
     8.1.  Packet Loss and Discard Metrics  . . . . . . . . . . . . . 15
     8.2.  Burst Metrics  . . . . . . . . . . . . . . . . . . . . . . 16
     8.3.  Delay Metrics  . . . . . . . . . . . . . . . . . . . . . . 17
     8.4.  Configuration Parameter Metrics  . . . . . . . . . . . . . 17
     8.5.  Jitter Buffer Parameters . . . . . . . . . . . . . . . . . 18
   9.  Perceptual Quality Metrics Report Block  . . . . . . . . . . . 18
   10. SDP Signaling  . . . . . . . . . . . . . . . . . . . . . . . . 21
   11. IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 22
   12. Security Considerations  . . . . . . . . . . . . . . . . . . . 24
   13. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 24
   14. References . . . . . . . . . . . . . . . . . . . . . . . . . . 24
     14.1. Normative References . . . . . . . . . . . . . . . . . . . 24
     14.2. Informative References . . . . . . . . . . . . . . . . . . 25
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 25






















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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 network-
   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 its 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 network-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, conveying video quality
   metrics in RTCP XR packets [RFC3611] has 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 synchronization
   information on received RTP packets.  The report blocks in the second
   category convey metrics relating to packet receipts defined in RFC
   3611 [RFC3611] that are summary in nature but include data that is
   more detailed, or of a different type, than that conveyed in existing
   RTCP packets.

   Six report block formats are defined by this document.  Of these, two
   are synchronization information blocks:

      *  RTP Flows Synchronization Delay Report Block
      *  Audio-Video Synchronization Report Block

   The other four are summary metrics blocks:

      *  Statistics Summary Report Block
      *  TR 101 290 Decodability Metrics Block [ETSI]
      *  Video Stream Metrics Report Block
      *  Perceptual 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].

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



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   [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 Audio-Video Synchronization Report Block Section 5 has been
   defined for video conferencing applications.  Such applications can
   use this report block to monitor A/V component 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 Stream Metrics Report Block and Statistics Summary Report Block
   can be applied to any real time video application, while the TR 101
   290 Decodability Metrics Report Block and Perceptual Quality Metrics
   Report Block can be used in any real-time AV application [ETSI].


4.  RTP Flows Synchronization Delay Report Block

   This block reports synchronization delay between RTP sessions of the
   same video stream sent using Multi-Session Transmission
   [I-D.ietf-avt-rtp-svc] beyond the information carried in the standard
   RTCP packet format.  Information is recorded about session bandwidth
   and synchronization delay.

   The RTP Flows 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                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                     Session Bandwidth                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |               Initial Synchronization Delay                   |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+





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

   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]).

   Session Bandwidth: 32 bits
      This field indicates the maximum bandwidth allocated to each RTP
      Session.  The value in this field represents the bandwidth
      allocation per RTP Session in kilobits per second, with values in
      the range 0 <= BW < 65536.

   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:

         Initial Synchronization Delay = max((the time interval between
         receiving the first RTP packet with synchronization metadata
         and the start of a session), (the time interval between
         receiving the first RTCP packet in the RTP session with the
         longest RTCP reporting interval and the start of a session)).


5.  Audio-Video Synchronization Report Block

   This block reports the audio-video synchronization requirements
   between audio and video components beyond the information carried in
   the standard RTCP packet format.  Information is recorded about
   tolerant audio lead video time and tolerant audio lag video time.
   The Audio-Video Synchronization 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    |L|  Reserved   |         Block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |        Audio Lead Video       |       Audio Lag Video         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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

   Audio Lead/Lag Video flag (L): 1 bit
      Bit set to 1 if Audio Lead Video field contains a report, 0 if
      Audio Lag Video field contains a report.

   Reserved: 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.

   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].

   Audio Lead Video time: 16 bits
      This field indicates the tolerant audio lead video time in
      milliseconds [DSLF].  This value is calculated based on the
      interarrival time between previous video RTP packet and the next
      first audio RTP packet and timestamps of both previous video RTP
      packet and the next first audio packet.

   Audio Lag Video time: 16 bits
      This field indicates the tolerant audio lag video time in
      milliseconds [DSLF].  This value is calculated based on the
      interarrival time between previous video RTP packet and the next
      first audio RTP packet and timestamps of both previous video RTP
      packet and the next first audio packet.


6.  Video Statistics Summary Report Block

   This block reports statistics beyond the information carried in the



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   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 Video 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    |L|D|FT |  LT |P|        block length           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         lost_frames                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                          dup frames                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                 lost_layered component packets                |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                  dup layered component_packets                |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                    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>.

   loss report flag (L): 1 bit
      Bit set to 1 if the lost_frame packets field, the
      lost_layered_component field or the lost partial frame packets
      field contains a report, 0 otherwise.





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   duplicate report flag (D): 1 bit
      Bit set to 1 of the dup_frame packets field, the
      dup_layered_component packets field or the dup partial frame
      packets field contains a report, 0 otherwise.

   Frame type indicator (FT): 2 bits
      This field is used to indicate the frame type to be reported.
      Bits set to 01 if the lost_frames field or dup_frames field
      contain a I_frame report, 10 if the lost_frames field and
      dup_frames field contain a P frame report, 11 if the lost_frames
      field and dup_frames field contain a B frame report, 00 otherwise.

   Layer Type indicator (LT): 3 bits
      This field is used to indicate the layer type of layered video to
      be reported.  LT is set to 001 if the loss_component_packet field
      and dup_component packet contain the base layer packet in SVC
      [I-D.ietf-avt-rtp-svc], 010 if the loss_component packet field and
      dup_component packet contain enhancement layer 1 packet in SVC,
      011 if the loss_component packet field and dup_component packet
      contain the enhancement layer 2 packet, 000 otherwise.

   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.

   Block length: 16 bits
      The constant 8, 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.








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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.

   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.



7.  TR 101 290 Decodability Metrics Report Block

   This block reports decodability metrics statistics beyond the
   information carried in the standard RTCP packet format.  Information
   is recorded about the number of Transport Stream Synchronization
   Losses, Sync byte errors, Continuity count errors, Transport errors,
   Program Clock Reference (PCR) errors, PCR repetition errors, PCR
   discontinuity indicator errors, and Presentation Time Stamp (PTS)
   errors [ETSI].  Such information can be useful for network management
   and video quality monitoring.

   Note that this is only applicable to MPEG-2 RTP streams [RFC2250].
   and not to any other video codec.

   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
   Decodability Metrics Block with a non-zero value in any field flagged
   as unreported.

   The Decodability 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    |L|B|C|T|P|S|rvd|         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       Number of packets       |         Number of TSs         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |             Transport Stream Synchronization Losses           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                      Sync byte errors                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                  Continuity count errors                      |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                      Transport errors                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         PCR errors                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                   PCR repetition errors                       |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |            PCR discontinuity indicator errors                 |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         PTS errors                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

   block type (BT): 8 bits
      A TR 101 290 decodability metrics report block is identified by
      the constant <TDM>.

   Transport Stream Synchronization Loss flag (L): 1 bit
      Bit set to 1 if the Transport Stream Synchronization Loss field
      contains a report, 0 otherwise.

   Sync byte error flag (B): 1 bit
      Bit set to 1 if the Sync byte error field contains a report, 0
      otherwise.

   Continuity count error flag (C): 1 bit
      Bit set to 1 if the Continuity count error field contains a
      report, 0 otherwise.








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   Transport error flag (T): 1 bit
      Bit set to 1 if the Transport error field contains a report, 0
      otherwise.

   PCR related error flag (P): 1 bit
      Bit set to 1 if the PCR error field, PCR repetition error field
      and PCR discontinuity indicator error fields contain a report, 0
      otherwise.

   PTS error flag (S): 1 bit
      Bit set to 1 if the PTS error field contains a report, 0
      otherwise.

   rvd: 2 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 10, 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 [RFC3611].

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

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

   Number of Packets: 16 bits
      Number of RTP packets in the above sequence number interval.

   Number of TSs: 16 bits
      Number of Transport Streams in the above sequence number interval.

   Transport Stream Synchronization Losses: 32 bits
      Number of Transport Stream Synchronization Losses in the above
      sequence number interval.

   Sync byte errors: 32 bits
      Number of Transport sync byte errors in the above sequence number
      interval.







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   Continuity count error: 32 bits
      Number of Transport Continuity count errors in the above sequence
      number interval.

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

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

   PCR repetition errors: 32 bits
      Number of Transport PCR repetition errors in the above sequence
      number interval.

   PCR discontinuity indicator errors: 32 bits
      Number of PCR discontinuity indicator errors in the above sequence
      number interval.

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



8.  Video Stream Metrics Report Block

   The Video Stream Metrics Report Block provides metrics for monitoring
   the quality of video stream.  This metrics includes Loss and discard
   metrics, Burst metrics, Delay metrics for I-Frame packets, B-Frame
   packets and P-Frame packets, Configuration parameter metrics.  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 also 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.



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   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    |FT |  reserved |        block length           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   Loss rate   |  Discard rate |          Loss Period          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |         Loss Distance         |        Max Loss Duration      |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     Round trip delay          |       End system delay        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     Gmin      |   RX config   |          JB nominal           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          JB maximum           |          JB abs max           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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

   Frame type indicator (FT): 2 bits
      This field is used to indicate the frame type to be reported.
      Bits set to 01 if the Loss rate field and discard rate field
      contain a I_frame report, 10 if the Loss rate field and discard
      rate field contain a P frame report, 11 if the Loss rate field and
      discard rate field contain a B frame report, 00 otherwise.

   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 8, 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].
   The remaining fields are described in the following four sections:



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   Packet Loss and Discard Metrics, Delay Metrics, Configuration
   Metrics, and Jitter Buffer Parameters.

8.1.  Packet Loss and Discard Metrics

   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.

   Loss rate for I-Frame packets: 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 I frame packets lost
      (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 for I-Frame packets: 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 I-Frame packets discarded (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.

   Loss rate for P-Frame packets: 8 bits
      The loss rate for P-Frame packets is similar to the loss rate for
      I-Frame packets.  The value is calculated by dividing the total
      number of P frame packets lost (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.







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   Discard rate for P-Frame packets: 8 bits
      The discard rate for P-Frame packets is similar to the discard
      rate for I-Frame packets.  The value is calculated by dividing the
      total number of P-Frame packets discarded (excluding duplicate
      packet discards) by the total number of packets expected,
      multiplying the result of the division by 256.

   Loss rate for B-Frame packets: 8 bits
      The loss rate for B-Frame packets is similar to the loss rate for
      P-Frame packets.  The value is calculated by dividing the total
      number of B frame packets lost (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.

   Discard rate for P-Frame packets: 8 bits
      The discard rate for B-Frame packets is similar to the discard
      rate for P-Frame packets.  The value is calculated by dividing the
      total number of B-Frame packet discarded (excluding duplicate
      packet discards) by the total number of packets expected,
      multiplying the result of the division by 256.

8.2.  Burst Metrics

   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
      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



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      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.

8.3.  Delay Metrics

   Similar to the delay metrics for audio stream defined in the section
   4.7.3 of [RFC3611], the delay metrics for video stream fall into two
   categories: end-system delay due to RTP payload multiplexing and
   round trip delay due to multiplexing RTP frames within a UDP frame.

   Round trip delay: 16 bits
      As defined in Section 4.7.3 of [RFC3611].

   End system delay: 16 bits
      As defined in Section 4.7.3 of [RFC3611].


8.4.  Configuration Parameter Metrics

   Similar to the configuration metrics defined in the section 4.7.6 of
   [RFC3611], the configuration metrics for video stream include Gmin,
   receiver configuration byte, packet loss concealment, jitter buffer
   adaptive, and jitter buffer rate.

   Gmin: 8 bits
      As defined in Section 4.7.6 of [RFC3611].

   Receiver configuration byte (RX config): 8 bits
      As defined in Section 4.7.6 of [RFC3611].

   Jitter buffer adaptive (JBA): 2 bits
      As defined in Section 4.7.6 of [RFC3611].

   Jitter buffer rate (JB rate): 4 bits
      As defined in Section 4.7.6 of [RFC3611].







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8.5.  Jitter Buffer Parameters

   Similar to the jitter buffer parameters defined in the section 4.7.7
   of [RFC3611], the Jitter Buffer Parameters for video streams include
   jitter buffer nominal delay, jitter buffer maximum delay and jitter
   buffer absolute maximum delay.

   Jitter buffer nominal delay (JB nominal): 16 bits
      As defined in Section 4.7.7 of[RFC3611].

   Jitter buffer maximum delay (JB maximum): 16 bits
      As defined in Section 4.7.7 of[RFC3611].

   Jitter buffer absolute maximum delay (JB abs max): 16 bits
      As defined in Section 4.7.7 of[RFC3611].



9.  Perceptual Quality Metrics Report Block

   This block reports perceptual quality metrics beyond the information
   carried in the standard RTCP packet format.  Information is recorded
   about Video MOS, Audio Video MOS, Video Service Transmission Quality,
   Video Service Audio Quality, Video Service Multimedia Quality and
   Video Service Picture Quality.

   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 Perceptual 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|S|T|A|M|P|Rsd|         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              MOS-V            |           MOS-AV              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              VSTQ             |           VSAQ                |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              VSMQ             |           VSPQ                |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |        Loss Period VSPQ       |    Loss Distance VSPQ         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

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

   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 (S): 1 bit
      Bit set to 1 if the MOS-V field and 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.

   Video Service Audio Quality flag (A): 1 bit
      Bit set to 1 if the VSAQ field contains a report, 0 otherwise.

   Video Service Multimedia Quality flag (M): 1 bit
      Bit set to 1 if the VSMQ field contains a report, 0 otherwise.

   Video Service Picture Quality flag (P): 1 bit
      Bit set to 1 if the VSPQ field, Loss Period VSPQ, Loss Distance
      VSPQ contains a report, 0 otherwise.







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   Rsd.: 2 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].

   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.  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-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.  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)

   VSAQ: 16 bits
      Video Service Audio Quality (TBC)

   VSMQ: 16 bits
      Video Service Multimedia Quality (TBC)

   VSPQ: 16 bits
      Video Service Picture Quality (TBC)







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   Loss Period VSPQ: 16 bits
      Video Service Picture Quality during Loss Period (TBC)

   Loss Distance VSPQ: 16 bits
      Video Service Picture Quality during Loss Distance (TBC)


10.  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-syn
                  / percept-quality-metrics
                  / video-stream-metric
                  / video-stat-summary
                  / decodability-metric

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

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

         percept-quality-metrics = "percept-quality-metrics"
                                   ["=" max-size]
            max-size = 1*DIGIT ; maximum block size in octets

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

         video-stat-summary = "video-stat-summary"
                              ["=" stat-flag *("," stat-flag)]
            stat-flag = "I Frame loss and duplication"
                        / "P Frame loss and duplication"
                        / "B Frame loss and duplication"

         decodability-metric = "decodability-metric"
                               ["=" stat-flag *("," stat-flag)]
            stat-flag = "Interval Metric"
                        / "MOS-V and MOS-AV"
                        / "VSTQ"
                        / "VSAQ"
                        / "VSMQ"
                        / "VSPQ"

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


11.  IANA Considerations

   New report block types for RTCP XR are subject to IANA registration.



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   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:       RFSD
      Long Name:  RTP Flows Synchronization Delay
      Value       <RFSD>
      Reference:  Section 4

      Name:       AVS
      Long Name:  Audio-Video Synchronization
      Value       <AVS>
      Reference:  Section 5

      Name:       VSS
      Long Name:  Video Statistics Summary
      Value       <VSS>
      Reference:  Section 6

      Name:       TDM
      Long Name:  TR 101 290 Decodability Metrics
      Value       <TDM>
      Reference:  Section 7

      Name:       VSM
      Long Name:  Video Stream Metrics
      Value       <VSM>
      Reference:  Section 8

      Name:       PQM
      Long Name:  Perceptual Quality Metric
      Value       <PQM>
      Reference:  Section 9

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

      *  "RTP-flows-syn"
      *  "audio-video-syn"
      *  "percept-quality-metrics"
      *  "video-stream-metric"
      *  "video-stat-summary"
      *  "decodability-metric"

   The contact information for the registrations is:




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


12.  Security Considerations

   TBC


13.  Acknowledgements

   The authors would like to thank Youqing Yang, Wenxiao Yu and Yinliang
   Hu for their valuable comments and suggestions on this document.


14.  References

14.1.  Normative References

   [ETSI]     ETSI, "Digital Video Broadcasting (DVB); Measurement
              guidelines for DVB systems", Technical Report TR 101 290,
              2001.

   [I-D.ietf-avt-rapid-rtp-sync]
              Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
              Flows", draft-ietf-avt-rapid-rtp-sync-11 (work in
              progress), May 2010.

   [I-D.ietf-avt-rtp-svc]
              Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
              "RTP Payload Format for SVC Video",
              draft-ietf-avt-rtp-svc-21 (work in progress), April 2010.

   [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.




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   [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.

14.2.  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.

   [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







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