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Versions: (draft-friedman-avt-rtcp-report-extns) 00 01 02 03 04 05 06 RFC 3611

INTERNET-DRAFT                                               19 May 2003
Internet Engineering Task Force                Expires: 19 November 2003
Audio/Video Transport Working Group

                                                 Timur Friedman, Paris 6
                                                 Ramon Caceres, ShieldIP
                                                 Alan Clark, Telchemy
                                                 Editors

            RTP Control Protocol Extended Reports (RTCP XR)

                draft-ietf-avt-rtcp-report-extns-06.txt


Status of this Memo

   This document is an Internet-Draft and is subject to all provisions
   of Section 10 of RFC2026.

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Copyright Notice

   Copyright (C) The Internet Society (2003).  All Rights Reserved.

Abstract

   This document defines the Extended Report (XR) packet type for the
   RTP Control Protocol (RTCP), and defines how the use of XR packets
   can be signaled by an application if it employs the Session
   Description Protocol (SDP).  XR packets are composed of report
   blocks, and seven block types are defined here.  The purpose of the
   extended reporting format is to convey information that supplements
   the six statistics that are contained in the report blocks used by



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   RTCP's Sender Report (SR) and Receiver Report (RR) packets.  Some
   applications, such as multicast inference of network characteristics
   (MINC) or voice over IP (VoIP) monitoring, require other and more
   detailed statistics.  In addition to the block types defined here,
   additional block types may be defined in the future by adhering to
   the framework that this document provides.

Table of Contents

   1.     Introduction ..............................................  3
   1.1    Applicability .............................................  4
   1.2    Terminology ...............................................  7
   2.     XR Packet Format ..........................................  7
   3.     Extended Report Block Framework ...........................  8
   4.     Extended Report Blocks ....................................  9
   4.1    Loss RLE Report Block .....................................  9
   4.1.1  Run Length Chunk .......................................... 15
   4.1.2  Bit Vector Chunk .......................................... 15
   4.1.3  Terminating Null Chunk .................................... 15
   4.2    Duplicate RLE Report Block ................................ 16
   4.3    Packet Receipt Times Report Block ......................... 17
   4.4    Receiver Reference Time Report Block ...................... 20
   4.5    DLRR Report Block ......................................... 21
   4.6    Statistics Summary Report Block ........................... 22
   4.7    VoIP Metrics Report Block ................................. 25
   4.7.1  Packet Loss and Discard Metrics ........................... 26
   4.7.2  Burst Metrics ............................................. 27
   4.7.3  Delay Metrics ............................................. 30
   4.7.4  Signal Related Metrics .................................... 30
   4.7.5  Call Quality or Transmission Quality Metrics .............. 33
   4.7.6  Configuration Parameters .................................. 34
   4.7.7  Jitter Buffer Parameters .................................. 35
   5.     SDP Signaling ............................................. 36
   5.1    The SDP Attribute ......................................... 37
   5.2    Usage in Offer/Answer ..................................... 40
   5.3    Usage Outside of Offer/Answer ............................. 41
   6.     IANA Considerations ....................................... 42
   6.1    XR Packet Type ............................................ 42
   6.2    RTCP XR Block Type Registry ............................... 42
   6.3    The "rtcp-xr" SDP Attribute ............................... 43
   7.     Security Considerations ................................... 44
   A.     Algorithms ................................................ 45
   A.1    Sequence Number Interpretation ............................ 45
   A.2    Example Burst Packet Loss Calculation ..................... 46
          Intellectual Property ..................................... 48
          Full Copyright Statement .................................. 49
          Acknowledgments ........................................... 49
          Contributors .............................................. 50



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          Authors' Addresses ........................................ 50
          References ................................................ 52
          Normative References ...................................... 52
          Non-Normative References .................................. 53

1. Introduction

   This document defines the Extended Report (XR) packet type for the
   RTP Control Protocol (RTCP) [9], and defines how the use of XR
   packets can be signaled by an application if it employs the Session
   Description Protocol (SDP) [4].  XR packets convey information beyond
   that already contained in the reception report blocks of RTCP's
   sender report (SR) or Receiver Report (RR) packets.  The information
   is of use across RTP profiles, and so is not appropriately carried in
   SR or RR profile-specific extensions.  Information used for network
   management falls into this category, for instance.

   The definition is broken out over the three sections that follow the
   Introduction.  Section 2 defines the XR packet as consisting of an
   eight octet header followed by a series of components called report
   blocks.  Section 3 defines the common format, or framework,
   consisting of a type and a length field, required for all report
   blocks.  Section 4 defines several specific report block types.
   Other block types can be defined in future documents as the need
   arises.

   The report block types defined in this document fall into three
   categories.  The first category consists of packet-by-packet reports
   on received or lost RTP packets.  Reports in the second category
   convey reference time information between RTP participants.  In the
   third category, reports convey metrics relating to packet receipts,
   that are summary in nature but that are more detailed, or of a
   different type, than that conveyed in existing RTCP packets.

   All told, seven report block formats are defined by this document.
   Of these, three are packet-by-packet block types:

   - Loss RLE Report Block (Section 4.1): Run length encoding of reports
   concerning the losses and receipts of RTP packets.

   - Duplicate RLE Report Block (Section 4.2): Run length encoding of
   reports concerning duplicates of received RTP packets.

   - Packet Receipt Times Report Block (Section 4.3): A list of
   reception timestamps of RTP packets.

   There are two reference time related block types:




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   - Receiver Reference Time Report Block (Section 4.4): Receiver-end
   wallclock timestamps.  Together with the DLRR Report Block mentioned
   next, these allow non-senders to calculate round-trip times.

   - DLRR Report Block (Section 4.5): The delay since the last Receiver
   Reference Time Report Block was received.  An RTP data sender that
   receives a Receiver Reference Time Report Block can respond with a
   DLRR Report Block, in much the same way as, in the mechanism already
   defined for RTCP [9, Section 6.3.1], an RTP data receiver that
   receives a sender's NTP timestamp can respond by filling in the DLSR
   field of an RTCP reception report block.

   Finally, this document defines two summary metric block types:

   - Statistics Summary Report Block (Section 4.6): Statistics on RTP
   packet sequence numbers, losses, duplicates, jitter, and TTL or Hop
   Limit values.

   - VoIP Metrics Report Block (Section 4.7): Metrics for monitoring
   Voice over IP (VoIP) calls.

   Before proceeding to the XR packet and report block definitions, this
   document provides an applicability statement (Section 1.1) that
   describes the contexts in which these report blocks can be used.  It
   also defines (Section 1.2) the normative use of key words, such as
   MUST and SHOULD, as they are employed in this document.

   Following the definitions of the various report blocks, this document
   describes how applications that employ SDP can signal their use
   (Section 5).  The document concludes with a discussion (Section 6) of
   numbering considerations for the Internet Assigned Numbers Authority
   (IANA), of security considerations (Section 7), and with appendices
   that provide examples of how to implement algorithms discussed in the
   text.

1.1 Applicability

   The XR packets are useful across multiple applications, and for that
   reason are not defined as profile-specific extensions to RTCP sender
   or Receiver Reports [9, Section 6.4.3].  Nonetheless, they are not of
   use in all contexts.  In particular, the VoIP metrics report block
   (Section 4.7) is specific to voice applications, though it can be
   employed over a wide variety of such applications.

   The VoIP metrics report block can be applied to any one-to-one or
   one-to-many voice application for which the use of RTP and RTCP is
   specified.  The use of conversational metrics (Section 4.7.5),
   including the R factor (as described by the E Model defined in [3])



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   and the mean opinion score for conversational quality (MOS-CQ), in
   applications other than simple two party calls is not defined, and
   hence these metrics should be identified as unavailable in multicast
   conferencing applications.

   The packet-by-packet report block types, Loss RLE (Section 4.1),
   Duplicate RLE (Section 4.2), and Packet Receipt Times (Section 4.3),
   have been defined with network tomography applications, such as
   multicast inference of network characteristics (MINC) [11], in mind.
   MINC requires detailed packet receipt traces from multicast session
   receivers in order to infer the gross structure of the multicast
   distribution tree and the parameters, such as loss rates and delays,
   that apply to paths between the branching points of that tree.

   Any real time multicast multimedia application can use the packet-by-
   packet report block types.  Such an application could employ a MINC
   inference subsystem that would provide it with multicast tree
   topology information.  One potential use of such a subsystem would be
   for the identification of high loss regions in the multicast tree and
   the identification of multicast session participants well situated to
   provide retransmissions of lost packets.

   Detailed packet-by-packet reports do not necessarily have to consume
   disproportionate bandwidth with respect to other RTCP packets.  An
   application can cap the size of these blocks.  A mechanism called
   "thinning" is provided for these report blocks, and can be used to
   ensure that they adhere to a size limit by restricting the number of
   packets reported upon within any sequence number interval.  The
   rationale for, and use of this mechanism is described in [13].
   Furthermore, applications might not require report blocks from all
   receivers in order to answer such important questions as where in the
   multicast tree there are paths that exceed a defined loss rate
   threshold.  Intelligent decisions regarding which receivers send
   these report blocks can further restrict the portion of RTCP
   bandwidth that they consume.

   The packet-by-packet report blocks can also be used by dedicated
   network monitoring applications.  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.

   Nothing in the packet-by-packet block types restricts their use to
   multicast applications.  In particular, they could be used for
   network tomography similar to MINC, but using striped unicast
   packets.  In addition, if it were found useful, they could be used
   for applications limited to two participants.




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   One use to which the packet-by-packet reports are not immediately
   suited is for data packet acknowledgments as part of a packet
   retransmission mechanism.  The reason is that the packet accounting
   technique suggested for these blocks differs from the packet
   accounting normally employed by RTP.  In order to favor measurement
   applications, an effort is made to interpret as little as possible at
   the data receiver, and leave the interpretation as much as possible
   to participants that receive the report blocks.  Thus, for example, a
   packet with an anomalous SSRC ID or an anomalous sequence number
   might be excluded by normal RTP accounting, but would be reported
   upon for network monitoring purposes.

   The Statistics Summary Report Block (Section 4.6) has also been
   defined with network monitoring in mind.  This block type can be used
   equally well for reporting on unicast and multicast packet reception.

   The reference time related block types were conceived for receiver-
   based TCP-friendly multicast congestion control [18].  By allowing
   data receivers to calculate their round trip times to senders, they
   help the receivers estimate the downstream bandwidth they should
   request.  Note that if every receiver is to send Receiver Reference
   Time Report Blocks (Section 4.4), a sender might potentially send a
   number of DLRR Report Blocks (Section 4.5) equal to the number of
   receivers whose RTCP packets have arrived at the sender within its
   reporting interval.  As the number of participants in a multicast
   session increases, an application should use discretion regarding
   which participants send these blocks, and how frequently.

   XR packets supplement the existing RTCP packets, and may be stacked
   with other RTCP packets to form compound RTCP packets [9, Section 6].
   The introduction of XR packets into a session in no way changes the
   rules governing the calculation of the RTCP reporting interval [9,
   Section 6.2].  As XR packets are RTCP packets, they count as such for
   bandwidth calculations.  As a result, the addition of extended
   reporting information may tend to increase the average RTCP packet
   size, and thus the average reporting interval.  This increase may be
   limited by limiting the size of XR packets.

   The SDP signaling defined for XR packets in this document (Section 5)
   was done so with three use scenarios in mind: a Real Time Streaming
   Protocol (RTSP) controlled streaming application, a one-to-many
   multicast multimedia application such as a course lecture with
   enhanced feedback, and a Session Initiation Protocol (SIP) controlled
   conversational session involving two parties.  Applications that
   employ SDP are free to use additional SDP signaling for cases not
   covered here.  In addition, applications are free to use signaling
   mechanisms other than SDP.




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1.2 Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [1] and
   indicate requirement levels for compliance with this specification.

2. XR Packet Format

   An XR packet consists of a header of two 32-bit words, followed by a
   number, possibly zero, of extended report blocks.  This type of
   packet is laid out in a manner consistent with other RTCP packets, as
   concerns the essential version, packet type, and length information.
   XR packets are thus backwards compatible with RTCP receiver
   implementations that do not recognize them, but that ought to be able
   to parse past them using the length information.  A padding field and
   an SSRC field are also provided in the same locations that they
   appear in other RTCP packets, for simplicity.  The format is as
   follows:


    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
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=2|P|reserved |   PT=XR=207   |             length            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                              SSRC                             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                         report blocks                         :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   version (V): 2 bits
        Identifies the version of RTP.  This specification applies to
        RTP version two.

   padding (P): 1 bit
        If the padding bit is set, this XR packet contains some
        additional padding octets at the end.  The semantics of this
        field are identical to the semantics of the padding field in the
        SR packet, as defined by the RTP specification.

   reserved: 5 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   packet type (PT): 8 bits



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        Contains the constant 207 to identify this as an RTCP XR packet.
        This value is registered with the Internet Assigned Numbers
        Authority (IANA), as described in Section 5.1.

   length: 16 bits
        As described for the RTCP Sender Report (SR) packet (see Section
        6.3.1 of the RTP specification [9]).  Briefly, the length of
        this XR packet in 32-bit words minus one, including the header
        and any padding.

   SSRC: 32 bits
        The synchronization source identifier for the originator of this
        XR packet.

   report blocks: variable length.
        Zero or more extended report blocks.  In keeping with the
        extended report block framework defined below, each block MUST
        consist of one or more 32-bit words.

3. Extended Report Block Framework

   Extended report blocks are stacked, one after the other, at the end
   of an XR packet.  An individual block's length is a multiple of 4
   octets.  The XR header's length field describes the total length of
   the packet, including these extended report blocks.

   Each block has block type and length fields that facilitate parsing.
   A receiving application can demultiplex the blocks based upon their
   type, and can use the length information to locate each successive
   block, even in the presence of block types it does not recognize.

   An extended 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       | type-specific |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :             type-specific block contents                      :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        Identifies the block format.  Seven block types are defined in
        Section 4.  Additional block types may be defined in future
        specifications.  This field's name space is managed by the
        Internet Assigned Numbers Authority (IANA), as described in



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        Section 5.2.

   type-specific: 8 bits
        The use of these bits is determined by the block type
        definition.

   block length: 16 bits
        The length of this report block including the header, in 32-bit
        words minus one.  If the block type definition permits, zero is
        an acceptable value, signifying a block that consists of only
        the BT, type-specific, and block length fields, with a null
        type-specific block contents field.

   type-specific block contents: variable length
        The use of this field is defined by the particular block type,
        subject to the constraint that it MUST be a multiple of 32 bits
        long.  If the block type definition permits, It MAY be zero bits
        long.

4. Extended Report Blocks

   This section defines seven extended report blocks: block types for
   reporting upon received packet losses and duplicates, packet
   reception times, receiver reference time information, receiver inter-
   report delays, detailed reception statistics, and voice over IP
   (VoIP) metrics.  An implementation SHOULD ignore incoming blocks with
   types either not relevant or unknown to it. Additional block types
   MUST be registered with the Internet Assigned Numbers Authority
   (IANA) [16], as described in Section 5.2.

4.1 Loss RLE Report Block

   This block type permits detailed reporting upon individual packet
   receipt and loss events.  Such reports can be used, for example, for
   multicast inference of network characteristics (MINC) [11].  With
   MINC, one can discover the topology of the multicast tree used for
   distributing a source's RTP packets, and of the loss rates along
   links within that tree.  Or they could be used to provide raw data to
   a network management application.

   Since a Boolean trace of lost and received RTP packets is potentially
   lengthy, this block type permits the trace to be compressed through
   run length encoding.  To further reduce block size, loss event
   reports can be systematically dropped from the trace in a mechanism
   called thinning that is described below and that is studied in [13].

   A participant that generates a Loss RLE Report Block should favor
   accuracy in reporting on observed events over interpretation of those



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   events whenever possible.  Interpretation should be left to those who
   observe the report blocks.  Following this approach implies that
   accounting for Loss RLE Report Blocks will differ from the accounting
   for the generation of the SR and RR packets described in the RTP
   specification [9] in the following two areas: per-sender accounting
   and per-packet accounting.

   In its per-sender accounting, an RTP session participant SHOULD NOT
   make the receipt of a threshold minimum number of RTP packets a
   condition for reporting upon the sender of those packets.  This
   accounting technique differs from the technique described in Section
   6.2.1 and Appendix A.1 of the RTP specification that allows a
   threshold to determine whether a sender is considered valid.

   In its per-packet accounting, an RTP session participant SHOULD treat
   all sequence numbers as valid.  This accounting technique differs
   from the technique described in Appendix A.1 of the RTP specification
   that suggests ruling a sequence number valid or invalid on the basis
   of its contiguity with the sequence numbers of previously received
   packets.

   Sender validity and sequence number validity are interpretations of
   the raw data.  Such interpretations are justified in the interest,
   for example, of excluding the stray old packet from an unrelated
   session from having an effect upon the calculation of the RTCP
   transmission interval.  The presence of stray packets might, on the
   other hand, be of interest to a network monitoring application.

   One accounting interpretation that is still necessary is for a
   participant to decide whether the 16 bit sequence number has rolled
   over.  Under ordinary circumstances this is not a difficult task.
   For example, if packet number 65,535 (the highest possible sequence
   number) is followed shortly by packet number 0, it is reasonable to
   assume that there has been a rollover.  However it is possible that
   the packet is an earlier one (from 65,535 packets earlier).  It is
   also possible that the sequence numbers have rolled over multiple
   times, either forward or backward.  The interpretation becomes more
   difficult when there are large gaps between the sequence numbers,
   even accounting for rollover, and when there are long intervals
   between received packets.

   The per-packet accounting technique mandated here is for a
   participant to keep track of the sequence number of the packet most
   recently received from a sender.  For the next packet that arrives
   from that sender, the sequence number MUST be judged to fall no more
   than 32,768 packets ahead or behind the most recent one, whichever
   choice places it closer.  In the event that both choices are equally
   distant (only possible when the distance is 32,768), the choice MUST



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   be the one that does not require a rollover.  Appendix A.1 presents
   an algorithm that implements this technique.

   Each block reports on a single RTP data packet source, identified by
   its SSRC.  The receiver that is supplying the report is identified in
   the header of the RTCP packet.

   Choice of beginning and ending RTP packet sequence numbers for the
   trace is left to the application.  These values are reported in the
   block.  The last sequence number in the trace MAY differ from the
   sequence number reported on in any accompanying SR or RR report.

   Note that because of sequence number wraparound the ending sequence
   number MAY be less than the beginning sequence number.  A Loss RLE
   Report Block MUST NOT be used to report upon a range of 65,534 or
   greater in the sequence number space, as there is no means to
   identify multiple wraparounds.

   The trace described by a Loss RLE report consists of a sequence of
   Boolean values, one for each sequence number of the trace.  A value
   of one represents a packet receipt, meaning that one or more packets
   having that sequence number have been received since the most recent
   wraparound of sequence numbers (or since the beginning of the RTP
   session if no wraparound has been judged to have occurred).  A value
   of zero represents a packet loss, meaning that there has been no
   packet receipt for that sequence number as of the time of the report.
   If a packet with a given sequence number is received after a report
   of a loss for that sequence number, a later Loss RLE report MAY
   report a packet receipt for that sequence number.

   The encoding itself consists of a series of 16 bit units called
   chunks that describe sequences of packet receipts or losses in the
   trace.  Each chunk either specifies a run length or a bit vector, or
   is a null chunk.  A run length describes between 1 and 16,383 events
   that are all the same (either all receipts or all losses).  A bit
   vector describes 15 events that may be mixed receipts and losses.  A
   null chunk describes no events, and is used to round out the block to
   a 32 bit word boundary.

   The mapping from a sequence of lost and received packets into a
   sequence of chunks is not necessarily unique.  For example, the
   following trace covers 45 packets, of which the 22nd and 24th have
   been lost and the others received:


   1111 1111 1111 1111 1111 1010 1111 1111 1111 1111 1111 1





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   One way to encode this would be:


   bit vector 1111 1111 1111 111
   bit vector 1111 1101 0111 111
   bit vector 1111 1111 1111 111
   null chunk


   Another way to encode this would be:


   run of 21 receipts
   bit vector 0101 1111 1111 111
   run of 9 receipts
   null chunk


   The choice of encoding is left to the application.  As part of this
   freedom of choice, applications MAY terminate a series of run length
   and bit vector chunks with a bit vector chunk that runs beyond the
   sequence number space described by the report block.  For example, if
   the 44th packet in the same sequence were lost:


   1111 1111 1111 1111 1111 1010 1111 1111 1111 1111 1110 1


   This could be encoded as:


   run of 21 receipts
   bit vector 0101 1111 1111 111
   bit vector 1111 1110 1000 000
   null chunk


   In this example, the last five bits of the second bit vector describe
   a part of the sequence number space that extends beyond the last
   sequence number in the trace.  These bits have been set to zero.

   All bits in a bit vector chunk that describe a part of the sequence
   number space that extends beyond the last sequence number in the
   trace MUST be set to zero, and MUST be ignored by the receiver.

   A null packet MUST appear at the end of a Loss RLE Report Block if
   the number of run length plus bit vector chunks is odd.  The null
   chunk MUST NOT appear in any other context.



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   Caution should be used in sending Loss RLE Report Blocks because,
   even with the compression provided by run length encoding, they can
   easily consume bandwidth out of proportion with normal RTCP packets.
   The block type includes a mechanism, called thinning, that allows an
   application to limit report sizes.

   A thinning value, T, selects a subset of packets within the sequence
   number space: those with sequence numbers that are multiples of 2^T.
   Packet reception and loss reports apply only to those packets.  T can
   vary between 0 and 15.  If T is zero then every packet in the
   sequence number space is reported upon.  If T is fifteen then one in
   every 32,768 packets is reported upon.

   Suppose that the trace just described begins at sequence number
   13,821.  The last sequence number in the trace is 13,865.  If the
   trace were to be thinned with a thinning value of T=2, then the
   following sequence numbers would be reported upon: 13,824, 13,828,
   13,832, 13,836, 13,840, 13,844, 13,848, 13,852, 13,856, 13,860,
   13,864.  The thinned trace would be as follows:


      1    1    1    1    1    0    1    1    1    1    0


   This could be encoded as follows:


   bit vector 1111 1011 1100 000
   null chunk


   The last four bits in the bit vector, representing sequence numbers
   13,868, 13,872, 13,876, and 13,880, extend beyond the trace and are
   thus set to zero and are ignored by the receiver.  With thinning, the
   loss of the 22nd packet goes unreported because its sequence number,
   13,842, is not a multiple of four.  Packet receipts for all sequence
   numbers that are not multiples of four also go unreported.  However,
   in this example thinning has permitted the Loss RLE Report Block to
   be shortened by one 32 bit word.

   Choice of the thinning value is left to the application.

   The Loss RLE 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=1      | rsvd. |   T   |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk 1              |             chunk 2           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                              ...                              :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk n-1            |             chunk n           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Loss RLE Report Block is identified by the constant 1.

   rsvd.: 4 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   thinning (T): 4 bits
        The amount of thinning performed on the sequence number space.
        Only those packets with sequence numbers 0 mod 2^T are reported
        on by this block.  A value of 0 indicates that there is no
        thinning, and all packets are reported on.  The maximum thinning
        is one packet in every 32,768 (amounting to two packets within
        each 16-bit sequence space).

   block length: 16 bits
        Defined in Section 3.

   SSRC of source: 32 bits
        The SSRC of the RTP data packet source being reported upon by
        this report block.

   begin_seq: 16 bits
        The first sequence number that this block reports on.

   end_seq: 16 bits
        The last sequence number that this block reports on plus one.

   chunk i: 16 bits
        There are three chunk types: run length, bit vector, and



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        terminating null, defined in the following sections.  If the
        chunk is all zeroes then it is a terminating null chunk.
        Otherwise, the leftmost bit of the chunk determines its type: 0
        for run length and 1 for bit vector.

4.1.1 Run Length Chunk


    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |C|R|        run length         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   chunk type (C): 1 bit
        A zero identifies this as a run length chunk.

   run type (R): 1 bit
        Zero indicates a run of 0s.  One indicates a run of 1s.

   run length: 14 bits
        A value between 1 and 16,383.  The value MUST not be zero for a
        run length chunk (zeroes in both the run type and run length
        fields would make the chunk a terminating null chunk).  Run
        lengths of 15 or less MAY be described with a run length chunk
        despite the fact that they could also be described as part of a
        bit vector chunk.

4.1.2 Bit Vector Chunk


    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |C|        bit vector           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   chunk type (C): 1 bit
        A one identifies this as a bit vector chunk.

   bit vector: 15 bits
        The vector is read from left to right, in order of increasing
        sequence number (with the appropriate allowance for wraparound).

4.1.3 Terminating Null Chunk




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   This chunk is all zeroes.


    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0|
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


4.2 Duplicate RLE Report Block

   This block type permits per-sequence-number reports on duplicates in
   a source's RTP packet stream.  Such information can be used for
   network diagnosis, and provide an alternative to packet losses as a
   basis for multicast tree topology inference.

   The Duplicate RLE Report Block format is identical to the Loss RLE
   Report Block format.  Only the interpretation is different, in that
   the information concerns packet duplicates rather than packet losses.
   The trace to be encoded in this case also consists of zeros and ones,
   but a zero here indicates the presence of duplicate packets for a
   given sequence number, whereas a one indicates that no duplicates
   were received.

   The existence of a duplicate for a given sequence number is
   determined over the entire reporting period.  For example, if packet
   number 12,593 arrives, followed by other packets with other sequence
   numbers, the arrival later in the reporting period of another packet
   numbered 12,593 counts as a duplicate for that sequence number.  The
   duplicate does not need to follow immediately upon the first packet
   of that number.  Care must be taken that a report does not cover a
   range of 65,534 or greater in the sequence number space.

   No distinction is made between the existence of a single duplicate
   packet and multiple duplicate packets for a given sequence number.
   Note also that since there is no duplicate for a lost packet, a loss
   is encoded as a one in a Duplicate RLE Report Block.

   The Duplicate RLE 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=2      | rsvd. |   T   |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk 1              |             chunk 2           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                              ...                              :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          chunk n-1            |             chunk n           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Duplicate RLE Report Block is identified by the constant 2.

   rsvd.: 4 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   thinning (T): 4 bits
        As defined in Section 4.1.

   block length: 16 bits
        Defined in Section 3.

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

   begin_seq: 16 bits
        As defined in Section 4.1.

   end_seq: 16 bits
        As defined in Section 4.1.

   chunk i: 16 bits
        As defined in Section 4.1.

4.3 Packet Receipt Times Report Block

   This block type permits per-sequence-number reports on packet receipt
   times for a given source's RTP packet stream.  Such information can
   be used for MINC inference of the topology of the multicast tree used



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   to distribute the source's RTP packets, and of the delays along the
   links within that tree.  It can also be used to measure partial path
   characteristics and to model distributions for packet jitter.

   Packet receipt times are expressed in the same units as used in the
   RTP timestamps of data packets.  This is so that, for each packet,
   one can establish both the send time and the receipt time in
   comparable terms.  Note, however, that as an RTP sender ordinarily
   initializes its time to a value chosen at random, there can be no
   expectation that reported send and receipt times will differ by an
   amount equal to the one-way delay between sender and receiver.  The
   reported times can nonetheless be useful for the purposes mentioned
   above.

   At least one packet MUST have been received for each sequence number
   reported upon in this block.  If this block type is used to report
   receipt times for a series of sequence numbers that includes lost
   packets, several blocks are required.  If duplicate packets have been
   received for a given sequence number, and those packets differ in
   their receipt times, any time other than the earliest MUST NOT be
   reported.  This is to ensure consistency among reports.

   Times reported in RTP timestamp format consume more bits than loss or
   duplicate information, and do not lend themselves to run length
   encoding.  The use of thinning is encouraged to limit the size of
   Packet Receipt Times Report Blocks.

   The Packet Receipt Times 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=3      | rsvd. |   T   |         block length          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       Receipt time of packet begin_seq                        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       Receipt time of packet (begin_seq + 1) mod 65536        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   :                              ...                              :
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       Receipt time of packet (end_seq - 1) mod 65536          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Packet Receipt Times Report Block is identified by the
        constant 3.

   rsvd.: 4 bits
        This field is reserved for future definition.  In the absence of
        such definition, the bits in this field MUST be set to zero and
        MUST be ignored by the receiver.

   thinning (T): 4 bits
        As defined in Section 4.1.

   block length: 16 bits
        Defined in Section 3.

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

   begin_seq: 16 bits
        As defined in Section 4.1.

   end_seq: 16 bits
        As defined in Section 4.1.

   Packet i receipt time: 32 bits
        The receipt time of the packet with sequence number i at the
        receiver.  The modular arithmetic shown in the packet format
        diagram is to allow for sequence number rollover.  It is
        preferable for the time value to be established at the link



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        layer interface, or in any case as close as possible to the wire
        arrival time.  Units and format are the same as for the
        timestamp in RTP data packets.  As opposed to RTP data packet
        timestamps, in which nominal values may be used instead of
        system clock values in order to convey information useful for
        periodic playout, the receipt times should reflect the actual
        time as closely as possible.  For a session, if the RTP
        timestamp is chosen at random, the first receipt time value
        SHOULD also be chosen at random, and subsequent timestamps
        offset from this value.  On the other hand, if the RTP timestamp
        is meant to reflect the reference time at the sender, then the
        receipt time SHOULD be as close as possible to the reference
        time at the receiver.

4.4 Receiver Reference Time Report Block

   This block extends RTCP's timestamp reporting so that non-senders may
   also send timestamps.  It recapitulates the NTP timestamp fields from
   the RTCP Sender Report [9, Sec. 6.3.1].  A non-sender may estimate
   its round trip time (RTT) to other participants, as proposed in [18],
   by sending this report block and receiving DLRR Report Blocks (see
   next section) in reply.


    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=4      |   reserved    |       block length = 2        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |              NTP timestamp, most significant word             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |             NTP timestamp, least significant word             |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Receiver Reference Time Report Block is identified by the
        constant 4.

   reserved: 8 bits
        This field is reserved for future definition.  In the absence of
        such 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.




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   NTP timestamp: 64 bits
        Indicates the wallclock time when this block was sent so that it
        may be used in combination with timestamps returned in DLRR
        Report Blocks (see next section) from other receivers to measure
        round-trip propagation to those receivers.  Receivers should
        expect that the measurement accuracy of the timestamp may be
        limited to far less than the resolution of the NTP timestamp.
        The measurement uncertainty of the timestamp is not indicated as
        it may not be known. A report block sender that can keep track
        of elapsed time but has no notion of wallclock time may use the
        elapsed time since joining the session instead. This is assumed
        to be less than 68 years, so the high bit will be zero.  It is
        permissible to use the sampling clock to estimate elapsed
        wallclock time. A report sender that has no notion of wallclock
        or elapsed time may set the NTP timestamp to zero.

4.5 DLRR Report Block

   This block extends RTCP's delay since last Sender Report (DLSR)
   mechanism [9, Sec. 6.3.1] so that non-senders may also calculate
   round trip times, as proposed in [18].  It is termed DLRR for delay
   since last Receiver Report, and may be sent in response to a Receiver
   Timestamp Report Block (see previous section) from a receiver to
   allow that receiver to calculate its round trip time to the
   respondent.  The report consists of one or more 3 word sub-blocks:
   one sub-block per Receiver Report.


    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=5      |   reserved    |         block length          |
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   |                 SSRC_1 (SSRC of first receiver)               | sub-
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
   |                         last RR (LRR)                         |   1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                   delay since last RR (DLRR)                  |
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
   |                 SSRC_2 (SSRC of second receiver)              | sub-
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block
   :                               ...                             :   2
   +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+


   block type (BT): 8 bits
        A DLRR Report Block is identified by the constant 5.




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

   block length: 16 bits
        Defined in Section 3.

   last RR timestamp (LRR): 32 bits
        The middle 32 bits out of 64 in the NTP timestamp (as explained
        in the previous section) received as part of a Receiver
        Reference Time Report Block from participant SSRC_n. If no such
        block has been received, the field is set to zero.

   delay since last RR (DLRR): 32 bits
        The delay, expressed in units of 1/65536 seconds, between
        receiving the last Receiver Reference Time Report Block from
        participant SSRC_n and sending this DLRR Report Block.  If no
        Receiver Reference Time Report Block has been received yet from
        SSRC_n, the DLRR field is set to zero (or the DLRR is omitted
        entirely). Let SSRC_r denote the receiver issuing this DLRR
        Report Block. Participant SSRC_n can compute the round-trip
        propagation delay to SSRC_r by recording the time A when this
        Receiver Timestamp Report Block is received.  It calculates the
        total round-trip time A-LRR using the last RR timestamp (LRR)
        field, and then subtracting this field to leave the round-trip
        propagation delay as A-LRR-DLRR. This is illustrated in [9, Fig.
        2].

4.6 Statistics Summary Report Block

   This block reports statistics beyond the information carried in the
   standard RTCP packet format, but not as fine grained as that carried
   in the report blocks previously described.  Information is recorded
   about lost packets, duplicate packets, jitter measurements, and TTL
   or Hop Limit values.  Such information can be useful for network
   management.

   The report block contents are dependent upon a series of flags bit
   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
   Statistics Summary Report Block with a non-zero value in any field
   flagged as unreported.

   The 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=6      |L|D|J|ToH|rsvd.|       block length = 9        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          begin_seq            |             end_seq           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        lost_packets                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        dup_packets                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         min_jitter                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         max_jitter                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         mean_jitter                           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                         dev_jitter                            |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | min_ttl_or_hl | max_ttl_or_hl |mean_ttl_or_hl | dev_ttl_or_hl |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A Statistics Summary Report Block is identified by the constant
        6.

   loss report flag (L): 1 bit
        Bit set to 1 if the lost_packets field contains a report, 0
        otherwise.

   duplicate report flag (D): 1 bit
        Bit set to 1 if the dup_packets field contains a report, 0
        otherwise.

   jitter flag (J): 1 bit
        Bit set to 1 if the min_jitter, max_jitter, mean_jitter, and
        dev_jitter fields all contain reports, 0 if none of them do.

   TTL or Hop Limit flag (ToH): 2 bits
        This field is set to 0 if none of the fields min_ttl_or_hl,
        max_ttl_or_hl, mean_ttl_or_hl, or dev_ttl_or_hl contain reports.
        If the field is non-zero then all of these fields contain
        reports.  The value 1 signifies that they report on IPv4 TTL
        values.  The value 2 signifies that they report on IPv6 Hop
        Limit values.  This value 3 is undefined and MUST NOT be used.



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

   block length: 16 bits
        The constant 9, in accordance with the definition of this field
        in Section 3.

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

   begin_seq: 16 bits
        As defined in Section 4.1.

   end_seq: 16 bits
        As defined in Section 4.1.

   lost_packets: 32 bits
        Number of lost packets in the above sequence number interval.

   dup_packets: 32 bits
        Number of duplicate packets in the above sequence number
        interval.

   min_jitter: 32 bits
        The minimum relative transit time between two packets in the
        above sequence number interval.  All jitter values are measured
        as the difference between a packet's RTP timestamp and the
        reporter's clock at the time of arrival, measured in the same
        units.

   max_jitter: 32 bits
        The maximum relative transit time between two packets in the
        above sequence number interval.

   mean_jitter: 32 bits
        The mean relative transit time between each two packet series in
        the above sequence number interval, rounded to the nearest value
        expressible as an RTP timestamp.

   dev_jitter: 32 bits
        The standard deviation of the relative transit time between each
        two packet series in the above sequence number interval.

   min_ttl_or_hl: 8 bits
        The minimum TTL or Hop Limit value of data packets in the
        sequence number range.



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   max_ttl_or_hl: 8 bits
        The maximum TTL or Hop Limit value of data packets in the
        sequence number range.

   mean_ttl_or_hl: 8 bits
        The mean TTL or Hop Limit value of data packets in the sequence
        number range, rounded to the nearest integer.

   dev_ttl_or_hl: 8 bits
        The standard deviation of TTL or Hop Limit values of data
        packets in the sequence number range.

4.7 VoIP Metrics Report Block

   The VoIP Metrics Report Block provides metrics for monitoring voice
   over IP (VoIP) calls.  These metrics include packet loss and discard
   metrics, delay metrics, analog metrics, and voice quality 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 call quality.

   In order to properly assess the quality of a Voice over IP call it is
   desirable to consider the degree of burstiness of packet loss [14].
   Following a Gilbert-Elliott model [3], a period of time, bounded by
   lost and/or discarded packets, with a high rate of losses and/or
   discards is a "burst," and a period of time between two bursts is a
   "gap."  Bursts correspond to periods of time during which the packet
   loss rate is high enough to produce noticeable degradation in audio
   quality.  Gaps correspond to periods of time during which only
   isolated lost packets may occur, and in general these can be masked
   by packet loss concealment.  Delay reports include the transit delay
   between RTP end points and the VoIP end system processing delays,
   both of which contribute to the user perceived delay.  Additional
   metrics include signal, echo, noise, and distortion levels.  Call
   quality metrics include R factors (as described by the E Model
   defined in [6,3]) and mean opinion scores (MOS scores).

   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 seven 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=7      |   reserved    |       block length = 8        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |                        SSRC of source                         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   loss rate   | discard rate  | burst density |  gap density  |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |       burst duration          |         gap duration          |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     round trip delay          |       end system delay        |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   | signal level  |  noise level  |     RERL      |     Gmin      |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   R factor    | ext. R factor |    MOS-LQ     |    MOS-CQ     |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |   RX config   |   reserved    |          JB nominal           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |          JB maximum           |          JB abs max           |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   block type (BT): 8 bits
        A VoIP Metrics Report Block is identified by the constant 7.

   reserved: 8 bits
        This field is reserved for future definition.  In the absence of
        such 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.

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

   The remaining fields are described in the following six sections:
   Packet Loss and Discard Metrics, Delay Metrics, Signal Related
   Metrics, Call Quality or Transmission Quality Metrics, Configuration
   Metrics, and Jitter Buffer Parameters.

4.7.1 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 voice stream however having separate counts helps



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   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: 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 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: 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 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.

4.7.2 Burst Metrics

   A burst is a period during which a high proportion of packets are
   either lost or discarded due to late arrival.  A burst is defined, in
   terms of a value Gmin, as a longest sequence that (a) starts with a
   lost or discarded packet, (b) does not contain any occurrences of
   Gmin or more consecutive received (and not discarded) packets, and
   (c) ends with a lost or discarded packet.

   A gap, informally, is a period of low packet losses and/or discards.
   Formally, a gap is defined as any of the following: (a) the period
   from the start of an RTP session to the receipt time of the last
   received packet before the first burst, (b) the period from the end
   of the last burst to either the time of the report or the end of the
   RTP session, whichever comes first, or (c) the period of time between
   two bursts.




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   For the purpose of determining if a lost or discarded packet near the
   start or end of an RTP session is within a gap or a burst it is
   assumed that the RTP session is preceded and followed by at least
   Gmin received packets, and that the time of the report is followed by
   at least Gmin received packets.

   A gap has the property that any lost or discarded packets within the
   gap must be preceded and followed by at least Gmin packets that were
   received and not discarded.  This gives a maximum loss/discard rate
   within a gap of: 1 / (Gmin + 1).

   A Gmin value of 16 is RECOMMENDED as it results in gap
   characteristics that correspond to good quality (i.e. low packet loss
   rate, a minimum distance of 16 received packets between lost packets)
   and hence differentiates nicely between good and poor quality
   periods.

   For example, a 1 denotes a received, 0 a lost, and X a discarded
   packet in the following pattern covering 64 packets:


   11110111111111111111111X111X1011110111111111111111111X111111111
   |---------gap----------|--burst---|------------gap------------|


   The burst consists of the twelve packets indicated above, starting at
   a discarded packet and ending at a lost packet.  The first gap starts
   at the beginning of the session and the second gap ends at the time
   of the report.

   If the packet spacing is 10 ms and the Gmin value is the recommended
   value of 16, the burst duration is 120 ms, the burst density 0.33,
   the gap duration 230 ms + 290 ms = 520 ms, and the gap density 0.04.

   This would result in reported values as follows (see field
   descriptions for semantics and details on how these are calculated):

   loss rate             12, which corresponds to 5%
   discard rate          12, which corresponds to 5%
   burst density         84, which corresponds to 33%
   gap density           10, which corresponds to 4%
   burst duration       120, value in milliseconds
   gap duration         520, value in milliseconds

   burst density: 8 bits
        The fraction of RTP data packets within burst periods since the
        beginning of reception that were either lost or discarded.  This
        value is expressed as a fixed point number with the binary point



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        at the left edge of the field.  It is calculated by dividing the
        total number of packets lost or discarded (excluding duplicate
        packet discards) within burst periods by the total number of
        packets expected within the burst periods, multiplying the
        result of the division by 256, limiting the maximum value to 255
        (to avoid overflow), and taking the integer part.  This field
        MUST be populated and MUST be set to zero if no packets have
        been received.

   gap density: 8 bits
        The fraction of RTP data packets within inter-burst gaps since
        the beginning of reception that were either lost or discarded.
        The 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 packets lost or discarded
        (excluding duplicate packet discards) within gap periods by the
        total number of packets expected within the gap periods,
        multiplying the result of the division by 256, limiting the
        maximum value to 255 (to avoid overflow), and taking the integer
        part.  This field MUST be populated and MUST be set to zero if
        no packets have been received.

   burst duration: 16 bits
        The mean duration, expressed in milliseconds, of the burst
        periods that have occurred since the beginning of reception.
        The duration of each period is calculated based upon the packets
        that mark the beginning and end of that period.  It is equal to
        the timestamp of the end packet, plus the duration of the end
        packet, minus the timestamp of the beginning packet.  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.

   gap duration: 16 bits
        The mean duration, expressed in milliseconds, of the gap periods
        that have occurred since the beginning of reception.  The
        duration of each period is calculated based upon the packet that
        marks the end of the prior burst and the packet that marks the
        beginning of the subsequent burst. It is equal to the timestamp
        of the subsequent burst packet, minus the timestamp of the prior
        burst packet, plus the duration of the prior burst packet.  If
        the actual values are not available, estimated values MUST be
        used.  In the case of a gap that occurs at the beginning of
        reception, the sum of the timestamp of the prior burst packet
        and the duration of the prior burst packet are replaced by the
        reception start time.  In the case of a gap that occurs at the
        end of reception, the timestamp of the subsequent burst packet
        is replaced by the reception end time.  If there have been no



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        gap periods, the gap duration value MUST be zero.

4.7.3 Delay Metrics

   For the purpose of the following definitions, the RTP interface is
   the interface between the RTP instance and the voice application
   (i.e.  FEC, de-interleaving, de-multiplexing, jitter buffer). For
   example, the time delay due to RTP payload multiplexing would be
   considered to be part of the voice application or end-system delay
   whereas delay due to multiplexing RTP frames within a UDP frame would
   be considered part of the RTP reported delay.  This distinction is
   consistent with the use of RTCP for delay measurements.

   round trip delay: 16 bits
        The most recently calculated round trip time between RTP
        interfaces, expressed in milliseconds. This value MAY be
        measured using RTCP, the DLRR method defined in Section 4.5 of
        this document, in which case it is necessary to convert the
        units of measurement from NTP timestamp values to milliseconds,
        or other approaches.  If RTCP is used then the reported delay
        value is the time of receipt of the most recent RTCP packet from
        source SSRC, minus the LSR (last SR) time reported in its SR
        (Sender Report), minus the DLSR (delay since last SR) reported
        in its SR.  A non-zero LSR value is required in order to
        calculate round trip delay. A value of 0 is permissible, however
        this field MUST be populated as soon as a delay estimate is
        available.

   end system delay: 16 bits
        The most recently estimated end system delay, expressed in
        milliseconds.  End system delay is defined as the sum of the
        total sample accumulation and encoding delay associated with the
        sending direction and the jitter buffer, decoding, and playout
        buffer delay associated with the receiving direction.  This
        delay MAY be estimated or measured.  This value SHOULD be
        provided in all VoIP metrics reports.  If an implementation is
        unable to provide the data, the value 0 MUST be used.

   Note that the one way symmetric VoIP segment delay may be calculated
   from the round trip and end system delays as follows.  If the round
   trip delay is denoted RTD and the end system delays associated with
   the two endpoints are ESD(A) and ESD(B) then:


   one way symmetric voice path delay  =  ( RTD + ESD(A) + ESD(B) ) / 2


4.7.4 Signal Related Metrics



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   The following metrics are intended to provide real time information
   related to the non-packet elements of the voice over IP system to
   assist with the identification of problems affecting call quality.
   The values identified below must be determined for the received audio
   signal. The information required to populate these fields may not be
   available in all systems, although it is strongly recommended that
   this data SHOULD be provided to support problem diagnosis.

   signal level: 8 bits
        The voice signal relative level is defined as the ratio of the
        signal level to a 0 dBm0 reference [10], expressed in decibels
        as a signed integer in two's complement form.  This is measured
        only for packets containing speech energy.  The intent of this
        metric is not to provide a precise measurement of the signal
        level but to provide a real time indication that the signal
        level may be excessively high or low.

        signal level = 10 Log10 ( rms talkspurt power (mW) )

        A value of 127 indicates that this parameter is unavailable.
        Typical values should be generally in the -15 to -20 dBm range.

   noise level: 8 bits
        The noise level is defined as the ratio of the silent period
        background noise level to a 0 dBm0 reference, expressed in
        decibels as a signed integer in two's complement form.

        noise level = 10 Log10 ( rms silence power (mW) )

        A value of 127 indicates that this parameter is unavailable.

   residual echo return loss (RERL): 8 bits
        The residual echo return loss value may be measured directly by
        the VoIP end system's echo canceller or may be estimated by
        adding the echo return loss (ERL) and echo return loss
        enhancement (ERLE) values reported by the echo canceller.

        RERL(dB) = ERL (dB) + ERLE (dB)

        In the case of a VoIP gateway the source of echo is typically
        line echo that occurs at 2-4 wire conversion points in the
        network.  This can be in the 8-12 dB range.  A line echo
        canceler can provide an ERLE of 30 dB or more and hence reduce
        this to 40-50 dB.  In the case of an IP phone this could be
        acoustic coupling between handset speaker and microphone or
        residual acoustic echo from speakerphone operation, and may more
        correctly be termed terminal coupling loss (TCL).  A typical
        handset would result in 40-50 dB of echo loss due to acoustic



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        feedback.

        Examples:

        - IP gateway connected to circuit switched network with 2 wire
        loop.  Without echo cancellation, typical 2-4 wire converter ERL
        of 12 dB.  RERL = ERL + ERLE = 12 + 0 = 12 dB.

        - IP gateway connected to circuit switched network with 2 wire
        loop.  With echo canceler that improves echo by 30 dB.  RERL =
        ERL + ERLE = 12 + 30 = 42 dB.

        - IP phone with conventional handset.  Acoustic coupling from
        handset speaker to microphone (terminal coupling loss) is
        typically 40 dB.  RERL = TCL = 40 dB.

        If we denote the local end of the VoIP path as A and the remote
        end as B and if the sender loudness rating (SLR) and receiver
        loudness rating (RLR) are known for A (default values 8 dB and 2
        dB respectively), then the echo loudness level at end A (talker
        echo loudness rating or TELR) is given by:

        TELR(A) = SRL(A) + ERL(B) + ERLE(B) + RLR(A)

        TELR(B) = SRL(B) + ERL(A) + ERLE(A) + RLR(B)

        Hence in order to incorporate echo into a voice quality estimate
        at the A end of a VoIP connection it is desirable to send the
        ERL + ERLE value from B to A using a format such as RTCP XR.

        Echo related information may not be available in all VoIP end
        systems.  As echo does have a significant effect on
        conversational quality it is recommended that estimated values
        for echo return loss and terminal coupling loss be provided (if
        sensible estimates can be reasonably determined).

        Typical values for end systems are given below to provide
        guidance:

        - IP Phone with handset: typically 45 dB.

        - PC softphone or speakerphone: extremely variable, consider
        reporting "undefined" (127).

        - IP gateway with line echo canceller: typically has ERL and
        ERLE available.

        - IP gateway without line echo canceller: frequently a source of



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        echo related problems, consider reporting either a low value (12
        dB) or "undefined" (127).

   Gmin
        See Configuration Parameters (Section 4.7.6, below).

4.7.5 Call Quality or Transmission Quality Metrics

   The following metrics are direct measures of the call quality or
   transmission quality, and incorporate the effects of codec type,
   packet loss, discard, burstiness, delay etc.  These metrics may not
   be available in all systems however SHOULD be provided in order to
   support problem diagnosis.

   R factor: 8 bits
        The R factor is a voice quality metric describing the segment of
        the call that is carried over this RTP session.  It is expressed
        as an integer in the range 0 to 100, with a value of 94
        corresponding to "toll quality" and values of 50 or less
        regarded as unusable.  This metric is defined as including the
        effects of delay, consistent with ITU-T G.107 [6] and ETSI TS
        101 329-5 [3].

        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.

   ext. R factor: 8 bits
        The external R factor is a voice quality metric describing the
        segment of the call that is carried over a network segment
        external to the RTP segment, for example a cellular network. Its
        values are interpreted in the same manner as for the RTP R
        factor.  This metric is defined as including the effects of
        delay, consistent with ITU-T G.107 [6] and ETSI TS 101 329-5
        [3], and relates to the outward voice path from the Voice over
        IP termination for which this metrics block applies.

        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.

   Note that an overall R factor may be estimated from the RTP segment R
   factor and the external R factor, as follows:

   R total = RTP R factor + ext. R factor - 94

   MOS-LQ: 8 bits
        The estimated mean opinion score for listening quality (MOS-LQ)



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        is a voice 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 delay and can be
        compared to MOS scores obtained from listening quality (ACR)
        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-CQ: 8 bits
        The estimated mean opinion score for conversational quality
        (MOS-CQ) is defined as including the effects of delay and other
        effects that would affect conversational quality.  The metric
        may be calculated by converting an R factor determined according
        to ITU-T G.107 [6] or ETSI TS 101 329-5 [3] into an estimated
        MOS using the equation specified in G.107.  It is expressed as
        an integer in the range 10 to 50, corresponding to MOS x 10, as
        for MOS-LQ.

        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.

4.7.6 Configuration Parameters

   Gmin: 8 bits
        The gap threshold.  This field contains the value used for this
        report block to determine if a gap exists.  The recommended
        value of 16 corresponds to a burst period having a minimum
        density of 6.25% of lost or discarded packets, which may cause
        noticeable degradation in call quality; during gap periods, if
        packet loss or discard occurs, each lost or discarded packet
        would be preceded by and followed by a sequence of at least 16
        received non-discarded packets.  Note that lost or discarded
        packets that occur within Gmin packets of a report being
        generated may be reclassified as being part of a burst or gap in
        later reports.  ETSI TS 101 329-5 [3] defines a computationally
        efficient algorithm for measuring burst and gap density using a
        packet loss/discard event driven approach.  This algorithm is
        reproduced in Appendix A.2 of the present document.  Gmin MUST
        not be zero and MUST be provided and MUST remain constant across
        VoIP Metrics report blocks for the duration of the RTP session.

   receiver configuration byte (RX config): 8 bits
        This byte consists of the following fields:



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         0 1 2 3 4 5 6 7
        +-+-+-+-+-+-+-+-+
        |PLC|JBA|JB rate|
        +-+-+-+-+-+-+-+-+


        packet loss concealment (PLC): 2 bits
             Standard (11) / enhanced (10) / disabled (01) / unspecified
             (00).  When PLC = 11 then a simple replay or interpolation
             algorithm is being used to fill-in the missing packet; this
             approach is typically able to conceal isolated lost packets
             at low packet loss rates.  When PLC = 10 then an enhanced
             interpolation algorithm is being used; algorithms of this
             type are able to conceal high packet loss rates
             effectively.  When PLC = 01 then silence is being inserted
             in place of lost packets.  When PLC = 00 then no
             information is available concerning the use of PLC, however
             for some codecs this may be inferred.

        jitter buffer adaptive (JBA): 2 bits
             Adaptive (11) / non-adaptive (10) / reserved (01)/ unknown
             (00).  When the jitter buffer is adaptive then its size is
             being dynamically adjusted to deal with varying levels of
             jitter.  When non-adaptive, the jitter buffer size is
             maintained at a fixed level.  When either adaptive or non-
             adaptive modes are specified then the jitter buffer size
             parameters below MUST be specified.

        jitter buffer rate (JB rate): 4 bits
             J = adjustment rate (0-15). This represents the
             implementation specific adjustment rate of a jitter buffer
             in adaptive mode. This parameter is defined in terms of the
             approximate time taken to fully adjust to a step change in
             peak to peak jitter from 30 ms to 100 ms such that:

             adjustment time = 2 * J * frame size (ms)

             This parameter is intended only to provide a guide to the
             degree of "aggressiveness" of a an adaptive jitter buffer
             and may be estimated. A value of 0 indicates that the
             adjustment time is unknown for this implementation.

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

4.7.7 Jitter Buffer Parameters



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   The values reported in these fields SHOULD be the most recently
   obtained values at the time of reporting.

   jitter buffer nominal delay (JB nominal): 16 bits
        This is the current nominal jitter buffer delay in milliseconds,
        which corresponds to the nominal jitter buffer delay for packets
        that arrive exactly on time.  This parameter MUST be provided
        for both fixed and adaptive jitter buffer implementations.

   jitter buffer maximum delay (JB maximum): 16 bits
        This is the current maximum jitter buffer delay in milliseconds
        which corresponds to the earliest arriving packet that would not
        be discarded.  In simple queue implementations this may
        correspond to the nominal size. In adaptive jitter buffer
        implementations this value may dynamically vary up to JB abs max
        (see below).  This parameter MUST be provided for both fixed and
        adaptive jitter buffer implementations.

   jitter buffer absolute maximum delay (JB abs max): 16 bits
        This is the absolute maximum delay in milliseconds that the
        adaptive jitter buffer can reach under worst case conditions.
        If this value exceeds 65535 milliseconds then this field SHALL
        convey the value 65535.  This parameter MUST be provided for
        adaptive jitter buffer implementations and its value MUST be set
        to JB maximum for fixed jitter buffer implementations.

5. SDP Signaling

   This section defines Session Description Protocol (SDP) [4] signaling
   for XR blocks that can be employed by applications that utilize SDP.
   This signaling is defined to be used either by applications that
   implement the SDP Offer/Answer model [8] or by applications that use
   SDP to describe media and transport configurations in connection with
   such protocols as the Session Announcement Protocol (SAP) [15] or the
   Real Time Streaming Protocol (RTSP) [17].  There exist other
   potential signaling methods, which are not defined here.

   The XR blocks MAY be used without prior signaling.  This is
   consistent with the rules governing other RTCP packet types, as
   described in [9].  An example in which signaling would not be used is
   an application that always requires the use of one or more XR blocks.
   However for applications that are configured at session initiation,
   the use of some type of signaling is recommended.

   Note that, although the use of SDP signaling for XR blocks may be
   optional, if used it MUST be used as defined here.  If SDP signaling
   is used in an environment where XR blocks are only implemented by
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   blocks will ignore the SDP attribute.

5.1 The SDP Attribute

   This section defines one new SDP attribute "rtcp-xr" that can be used
   to signal participants in a media session that they should use the
   specified XR blocks.  This attribute can be easily extended in the
   future with new parameters to cover any new report blocks.

   The RTCP XR blocks SDP attribute is defined below in Augmented
   Backus-Naur Form (ABNF) [2].  It is both a session and a media level
   attribute.  When specified at session level, it applies to all media
   level blocks in the session.  Any media level specification MUST
   replace a session level specification, if one is present, for that
   media block.

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


   xr-format = pkt-loss-rle
             / pkt-dup-rle
             / pkt-rcpt-times
             / rcvr-rtt
             / stat-summary
             / voip-metrics
             / format-ext


   pkt-loss-rle   = "pkt-loss-rle" ["=" max-size]
   pkt-dup-rle    = "pkt-dup-rle" ["=" max-size]
   pkt-rcpt-times = "pkt-rcpt-times" ["=" max-size]
   rcvr-rtt       = "rcvr-rtt" "=" rcvr-rtt-mode [":" max-size]
   rcvr-rtt-mode  = "all"
                  / "sender"
   stat-summary   = "stat-summary" ["=" stat-flag *("," stat-flag)]
   stat-flag      = "loss"
                  / "dup"
                  / "jitt"
                  / "TTL"
                  / "HL"
   voip-metrics   = "voip-metrics"
   max-size       = 1*DIGIT ; maximum block size in octets
   DIGIT          = %x30-39
   format-ext     = non-ws-string

   non-ws-string  = 1*(%x21-FF)
   CRLF           = %d13.10




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   The "rtcp-xr" attribute contains zero, one, or more XR block related
   parameters.  Each parameter signals functionality for an XR block, or
   a group of XR blocks.  The attribute is extensible so that parameters
   can be defined for any future XR block (and a parameter should be
   defined for every future block).

   Each "rtcp-xr" parameter belongs to one of two categories.  The first
   category, the unilateral parameters, are for report blocks that
   simply report on the RTP stream and related metrics.  The second
   category, collaborative parameters, are for XR blocks that involve
   actions by more than one party in order to carry out their functions.

   Round trip time (RTT) measurement is an example of collaborative
   functionality.  An RTP data packet receiver sends a Receiver
   Reference Time Report Block (Section 4.4).  A participant that
   receives this block sends a DLRR Report Block (Section 4.5) in
   response, allowing the receiver to calculate its RTT to that
   participant.  As this example illustrates, collaborative
   functionality may be implemented by two or more different XR blocks.
   The collaborative functionality of several XR blocks may be governed
   by a single "rtcp-xr" parameter.

   For the unilateral category, this document defines the following
   parameters.  The parameter names and their corresponding XR formats
   are as follows:

   Parameter name    XR block (block type and name)
   --------------    ------------------------------------
   pkt-loss-rle      1  Loss RLE Report Block
   pkt-dup-rle       2  Duplicate RLE Report Block
   pkt-rcpt-times    3  Packet Receipt Times Report Block
   stat-summary      6  Statistics Summary Report Block
   voip-metrics      7  VoIP Metrics Report Block


   The "pkt-loss-rle", "pkt-dup-rle", and "pkt-rcpt-times" parameters
   MAY specify an integer value.  This value indicates the largest size
   the whole report block SHOULD have in octets.  This shall be seen as
   an indication that thinning shall be applied if necessary to meet the
   target size.

   The "stat-summary" parameter contains a list indicating which fields
   SHOULD be included in the Statistics Summary report blocks that are
   sent.  The list is a comma separated list, containing one or more
   field indicators.  The space character (0x20) SHALL NOT be present
   within the list.  Field indicators represent the flags defined in
   section 4.6.  The field indicators and their respective flags are as
   follows:



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   Indicator    Flag
   ---------    ---------------------------
   loss         loss report flag (L)
   dup          duplicate report flag (D)
   jitt         jitter flag (J)
   TTL          TTL or Hop Limit flag (ToH)
   HL           TTL or Hop Limit flag (ToH)


   For "loss", "dup", and "jitt", the presence of the indicator
   indicates that the corresponding flag should be set to 1 in the
   Statistics Summary report blocks that are sent.  The presence of
   "TTL" indicates that the corresponding flag should be set to 1.  The
   presence of "HL" indicates that the corresponding flag should be set
   to 2.  The indicators "TTL" and "HL" MUST NOT be signaled together.

   Blocks in the collaborative category are classified as initiator
   blocks or response blocks.  Signaling SHOULD indicate which
   participants are required to respond to the initiator block.  A party
   that wishes to receive response blocks from those participants can
   trigger this by sending an initiator block.

   The collaborative category currently consists only of one
   functionality, namely the RTT measurement mechanism for RTP data
   receivers.  The collective functionality of the Receiver Reference
   Time Report Block and DLRR Report Block is represented by the "rcvr-
   rtt" parameter.  This parameter takes as its arguments a mode value
   and, optionally, a maximum size for the DLRR report block.  The mode
   value "all" indicates that both RTP data senders and data receivers
   MAY send DLRR blocks, while the mode value "sender" indicates that
   only active RTP senders MAY send DLRR blocks, i.e. non RTP senders
   SHALL NOT send DLRR blocks.  If a maximum size in octets is included,
   any DLRR Report Blocks that are sent SHALL NOT exceed the specified
   size.  If size limitations mean that a DLRR Report Block sender
   cannot report in one block upon all participants from which it has
   received a Receiver Reference Time Report Block then it SHOULD report
   on participants in a round robin fashion across several report
   intervals.

   The "rtcp-xr" attributes parameter list MAY be empty.  This is useful
   in cases in which an application needs to signal that it understands
   the SDP signaling but does not wish to avail itself of XR
   functionality.  For example, an application in a SIP controlled
   session could signal that it wishes to stop using all XR blocks by
   removing all applicable SDP parameters in a re-INVITE message that it
   sends.  If XR blocks are not to be used at all from the beginning of
   a session, it is RECOMMENDED that the "rtcp-xr" attribute not be
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   When the "rtcp-xr" attribute is present, participants SHOULD NOT send
   XR blocks other than the ones indicated by the parameters.  This
   means that inclusion of a "rtcp-xr" attribute without any parameters
   tells a participant that it SHOULD NOT send any XR blocks at all.
   The purpose is to conserve bandwidth.  This is especially important
   when collaborative parameters are applied to a large multicast group:
   the sending of an initiator block could potentially trigger responses
   from all participants.  There are, however, contexts in which it
   makes sense to send an XR block in the absence of a parameter
   signaling its use.  For instance, an application might be designed so
   as to send certain report blocks without negotiation, while using SDP
   signaling to negotiate the use of other blocks.

5.2 Usage in Offer/Answer

   In the Offer/Answer context [8], the interpretation of SDP signaling
   for XR packets depends upon the direction attribute that is signaled:
   "recvonly", "sendrecv", or "sendonly" [4].  If no direction attribute
   is supplied then "sendrecv" is assumed.  This section applies only to
   unicast media streams, except where noted.  Discussion of unilateral
   parameters is followed by discussion of collaborative parameters in
   this section.

   For "sendonly" and "sendrecv" media stream offers that specify
   unilateral "rtcp-xr" attribute parameters, the answerer SHOULD send
   the corresponding XR blocks.  For "sendrecv" offers, the answerer MAY
   include the "rtcp-xr" attribute in its response, and specify any
   unilateral parameters in order to request that the offerer send the
   corresponding XR blocks.  The offerer SHOULD send these blocks.

   For "recvonly" media stream offers, the offerer's use of the "rtcp-
   xr" attribute in connection with unilateral parameters indicates that
   the offerer is capable of sending the corresponding XR blocks.  If
   the answerer responds with an "rtcp-xr" attribute, the offerer SHOULD
   send XR blocks for each specified unilateral parameter that was in
   its offer.

   For multicast media streams, the inclusion of an "rtcp-xr" attribute
   with unilateral parameters means that every media recipient SHOULD
   send the corresponding XR blocks.

   An SDP offer with a collaborative parameter declares the offerer
   capable of receiving the corresponding initiator and replying with
   the appropriate responses.  For example, an offer that specifies the
   "rcvr-rtt" parameter means that the offerer is prepared to receive
   Receiver Reference Time Report Blocks and to send DLRR Report Blocks.
   An offer of a collaborative parameter means that the answerer MAY
   send the initiator, and, having received the initiator, the offerer



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   SHOULD send the responses.

   There are exceptions to the rule that an offerer of a collaborative
   parameter should send responses.  For instance, the collaborative
   parameter might specify a mode that excludes the offerer.  Or
   congestion control or maximum transmission unit considerations might
   militate against the offerer's response.

   By including a collaborative parameter in its answer, the answerer
   declares its ability to receive initiators and to send responses.
   The offerer MAY then send initiators, to which the answerer SHOULD
   reply with responses.  As for the offer of a collaborative parameter,
   there are exceptions to the rule that the answerer should reply.

   When making an SDP offer of a collaborative parameter for a multicast
   media stream, the offerer SHOULD specify which participants are to
   respond to a received initiator.  A participant that is not specified
   SHOULD NOT send responses.  Otherwise, undue bandwidth might be
   consumed.  The offer indicates that each participant that is
   specified SHOULD respond if it receives an initiator.  It also
   indicates that a specified participant MAY send an initiator block.

   An SDP answer for a multicast media stream SHOULD include all
   collaborative parameters that are present in the offer and that are
   supported by the answerer.  It SHOULD NOT include any collaborative
   parameter that is absent from the offer.

   If a participant receives an SDP offer and understands the "rtcp-xr"
   attribute but does not wish to implement XR functionality offered,
   its answer SHOULD include an "rtcp-xr" attribute without parameters.
   By doing so, the party declares that at a minimum that it is capable
   of understanding the signaling.

5.3 Usage Outside of Offer/Answer

   SDP can be employed outside of the Offer/Answer context, for instance
   for multimedia sessions that are announced through the Session
   Announcement Protocol (SAP) [15], or streamed through the Real Time
   Streaming Protocol (RTSP) [17].  The signaling model is simpler, as
   the sender does not negotiate parameters, but the functionality that
   is expected from specifying the "rtcp-xr" attribute is the same as in
   Offer/Answer.

   When a unilateral parameter is specified for the "rtcp-xr" attribute
   associated with a media stream, the receiver of that stream SHOULD
   send the corresponding XR block.  When a collaborative parameter is
   specified, only the participants indicated by the mode value in the
   collaborative parameter are concerned.  Each such participant that



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   receives an initiator block SHOULD send the corresponding response
   block.  Each such participant MAY also send initiator blocks.

6. IANA Considerations

   This document defines a new RTCP packet type, the Extended Report
   (XR) type, within the existing Internet Assigned Numbers Authority
   (IANA) registry of RTP RTCP Control Packet Types.  This document also
   defines a new IANA registry: the registry of RTCP XR Block Types.
   Within this new registry, this document defines an initial set of
   seven block types and describes how the remaining types are to be
   allocated.

   Further, this document defines a new SDP attribute, "rtcp-xr", within
   the existing IANA registry of SDP Parameters.  It defines a new IANA
   registry, the registry of RTCP XR SDP Parameters, and an initial set
   of six parameters, and describes how additional parameters are to be
   allocated.

6.1 XR Packet Type

   The XR packet type defined by this document is registered with the
   IANA as packet type 207 in the registry of RTP RTCP Control Packet
   types (PT).

6.2 RTCP XR Block Type Registry

   This document creates an IANA registry called the RTCP XR Block Type
   Registry to cover the name space of the Extended Report block type
   (BT) field specified in Section 3. The BT field contains eight bits,
   allowing 256 values.  The RTCP XR Block Type Registry is to be
   managed by the IANA according to the Specification Required policy of
   RFC 2434 [7].  Future specifications SHOULD attribute block type
   values in strict numeric order following the values attributed in
   this document:


   BT  name
   --  ----
    1  Loss RLE Report Block
    2  Duplicate RLE Report Block
    3  Packet Receipt Times Report Block
    4  Receiver Reference Time Report Block
    5  DLRR Report Block
    6  Statistics Summary Report Block
    7  VoIP Metrics Report Block





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   The BT value 255 is reserved for future extensions.

   Furthermore, future specifications SHOULD avoid the value 0.  Doing
   so facilitates packet validity checking, since an all-zeros field
   might commonly be found in an ill-formed packet.

   Any registration MUST contain the following information:

   - Contact information of the one doing the registration, including at
     least name, address, and email.

   - The format of the block type being registered, consistent with the
     extended report block format described in Section 3.

   - A description of what the block type represents and how it shall be
     interpreted, detailing this information for each of its fields.

6.3 The "rtcp-xr" SDP Attribute

   This SDP attribute "rtcp-xr" defined by this document is registered
   with the IANA registry of SDP Parameters as follows:

   SDP Attribute ("att-field"):

     Attribute name:     rtcp-xr
     Long form:          RTP Control Protocol Extended Report Parameters
     Type of name:       att-field
     Type of attribute:  session and media level
     Subject to charset: no
     Purpose:            see Section 5 of this document
     Reference:          this document
     Values:             see this document and registrations below

   The attribute has an extensible parameter field and therefore a
   registry for these parameters is required.  This document creates an
   IANA registry called the RTCP XR SDP Parameters Registry.  It
   contains the six parameters defined in Section 5.1: "pkt-loss-rle",
   "pkt-dup-rle", "pkt-rcpt-times", "stat-summary", "voip-metrics", and
   "recv-rtt".

   Additional parameters are to be added to this registry in accordance
   with the Specification Required policy of RFC 2434 [7].  Any
   registration MUST contain the following information:

   - Contact information of the one doing the registration, including at
     least name, address, and email.

   - An Augmented Backus-Naur Form (ABNF) [2] definition of the



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     parameter, in accordance with the "format-ext" definition of
     Section 5.1.

   - A description of what the parameter represents and how it shall be
     interpreted, both normally and in Offer/Answer.

7. Security Considerations

   This document extends the RTCP reporting mechanism.  The security
   considerations that apply to RTCP reports [9, Appendix B] also apply
   to XR reports.  This section details the additional security
   considerations that apply to the extensions.

   The extensions introduce heightened confidentiality concerns.
   Standard RTCP reports contain a limited number of summary statistics.
   The information contained in XR reports is both more detailed and
   more extensive (covering a larger number of parameters).  The per-
   packet report blocks and the VoIP Metrics Report Block provide
   examples.

   The per-packet information contained in Loss RLE, Duplicate RLE, and
   Packet Receipt Times Report Blocks facilitates multicast inference of
   network characteristics (MINC) [11].  Such inference can reveal the
   gross topology of a multicast distribution tree, as well as
   parameters, such as the loss rates and delays, along paths between
   branching points in that tree.  Such information might be considered
   sensitive to autonomous system administrators.

   The VoIP Metrics Report Block provides information on the quality of
   ongoing voice calls.  Though such information might be carried in
   application specific format in standard RTP sessions, making it
   available in a standard format here makes it more available to
   potential eavesdroppers.

   No new mechanisms are introduced in this document to ensure
   confidentiality.  Encryption procedures, such as those being
   suggested for a Secure RTCP (SRTCP) [12] at the time that this
   document was written, can be used when confidentiality is a concern
   to end hosts.  Given that RTCP traffic can be encrypted by the end
   hosts, autonomous systems must be prepared for the fact that certain
   aspects of their network topology can be revealed.

   Any encryption or filtering of XR report blocks entails a loss of
   monitoring information to third parties.  For example, a network that
   establishes a tunnel to encrypt VoIP Report Blocks denies that
   information to the service providers traversed by the tunnel.  The
   service providers cannot then monitor or respond to the quality of
   the VoIP calls that they carry, potentially creating problems for the



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   network's users.  As a default, XR packets should not be encrypted or
   filtered.

   The extensions also make certain denial of service attacks easier.
   This is because of the potential to create RTCP packets much larger
   than average with the per packet reporting capabilities of the Loss
   RLE, Duplicate RLE, and Timestamp Report Blocks.  Because of the
   automatic bandwidth adjustment mechanisms in RTCP, if some session
   participants are sending large RTCP packets, all participants will
   see their RTCP reporting intervals lengthened, meaning they will be
   able to report less frequently.  To limit the effects of large
   packets, even in the absence of denial of service attacks,
   applications SHOULD place an upper limit on the size of the XR report
   blocks they employ.  The "thinning" techniques described in Section
   4.1 permit the packet-by-packet report blocks to adhere to a
   predefined size limit.

A. Algorithms

A.1 Sequence Number Interpretation

   This the algorithm suggested by Section 4.1 for keeping track of the
   sequence numbers from a given sender.  It implements the accounting
   practice required for the generation of Loss RLE Report Blocks.

   This algorithm keeps track of 16 bit sequence numbers by translating
   them into a 32 bit sequence number space.  The first packet received
   from a source is considered to have arrived roughly in the middle of
   that space.  Each packet that follows is placed either ahead or
   behind the prior one in this 32 bit space, depending upon which
   choice would place it closer (or, in the event of a tie, which choice
   would not require a rollover in the 16 bit sequence number).

   // The reference sequence number is an extended sequence number
   // that serves as the basis for determining whether a new 16 bit
   // sequence number comes earlier or later in the 32 bit sequence
   // space.
   u_int32 _src_ref_seq;
   bool    _uninitialized_src_ref_seq;

   // Place seq into a 32-bit sequence number space based upon a
   // heuristic for its most likely location.
   u_int32 extend_seq(const u_int16 seq) {

           u_int32 extended_seq, seq_a, seq_b, diff_a, diff_b;
           if(_uninitialized_src_ref_seq) {

                   // This is the first sequence number received.  Place



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                   // it in the middle of the extended sequence number
                   // space.
                   _src_ref_seq                = seq | 0x80000000u;
                   _uninitialized_src_ref_seq  = false;
                   extended_seq                = _src_ref_seq;
           }
           else {

                   // Prior sequence numbers have been received.
                   // Propose two candidates for the extended sequence
                   // number: seq_a is without wraparound, seq_b with
                   // wraparound.
                   seq_a = seq | (_src_ref_seq & 0xFFFF0000u);
                   if(_src_ref_seq < seq_a) {
                           seq_b  = seq_a - 0x00010000u;
                           diff_a = seq_a - _src_ref_seq;
                           diff_b = _src_ref_seq - seq_b;
                   }
                   else {
                           seq_b  = seq_a + 0x00010000u;
                           diff_a = _src_ref_seq - seq_a;
                           diff_b = seq_b - _src_ref_seq;
                   }

                   // Choose the closer candidate.  If they are equally
                   // close, the choice is somewhat arbitrary: we choose
                   // the candidate for which no rollover is necessary.
                   if(diff_a < diff_b) {
                           extended_seq = seq_a;
                   }
                   else {
                           extended_seq = seq_b;
                   }

                   // Set the reference sequence number to be this most
                   // recently-received sequence number.
                   _src_ref_seq = extended_seq;
           }

           // Return our best guess for a 32-bit sequence number that
           // corresponds to the 16-bit number we were given.
           return extended_seq;
   }

   A.2 Example Burst Packet Loss Calculation.

   This is an algorithm for measuring the burst characteristics for the
   VoIP Metrics Report Block (Section 4.7).  The algorithm, which has



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   been verified against a working implementation for correctness, is
   reproduced from ETSI TS 101 329-5 [3].  The algorithm as described
   here takes precedence over any change that might eventually be made
   to the algorithm in future ETSI documents.

   This algorithm is event driven and hence extremely computationally
   efficient.

   Given the following definition of states:


   state 1 = received a packet during a gap
   state 2 = received a packet during a burst
   state 3 = lost a packet during a burst
   state 4 = lost an isolated packet during a gap


   The "c" variables below correspond to state transition counts, i.e.
   c14 is the transition from state 1 to state 4. It is possible to
   infer one of a pair of state transition counts to an accuracy of 1
   which is generally sufficient for this application.

   "pkt" is the count of packets received since the last packet was
   declared lost or discarded and "lost" is the number of packets lost
   within the current burst.  "packet_lost" and "packet_discarded" are
   Boolean variables that indicate if the event that resulted in this
   function being invoked was a lost or discarded packet.

   if(packet_lost) {
           loss_count++;
   }
   if(packet_discarded) {
           discard_count++;
   }
   if(!packet_lost && !packet_discarded) {
           pkt++;
   }
   else {
           if(pkt >= gmin) {
                   if(lost == 1) {
                           c14++;
                   }
                   else {
                           c13++;
                   }
                   lost = 1;
                   c11 += pkt;
           }



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           else {
                   lost++;
                   if(pkt == 0) {
                           c33++;
                   }
                   else {
                           c23++;
                           c22 += (pkt - 1);
                   }
           }
           pkt = 0;
   }

   At each reporting interval the burst and gap metrics can be
   calculated as follows.

   // Calculate additional transition counts.
   c31 = c13;
   c32 = c23;
   ctotal = c11 + c14 + c13 + c22 + c23 + c31 + c32 + c33;

   // Calculate burst and densities.
   p32 = c32 / (c31 + c32 + c33);
   if((c22 + c23) < 1) {
           p23 = 1;
   }
   else {
           p23 = 1 - c22/(c22 + c23);
   }
   burst_density = 256 * p23 / (p23 + p32);
   gap_density = 256 * c14 / (c11 + c14);

   // Calculate burst and gap durations in ms
   m = frameDuration_in_ms * framesPerRTPPkt;
   gap_length = (c11 + c14 + c13) * m / c13;
   burst_length = ctotal * m / c13 - lgap;

   /* calculate loss and discard rates */
   loss_rate = 256 * loss_count / ctotal;
   discard_rate = 256 * discard_count / ctotal;

Intellectual Property

   The IETF takes no position regarding the validity or scope of any
   intellectual property or other rights that might be claimed to
   pertain to the implementation or use of the technology described in
   this document or the extent to which any license under such rights
   might or might not be available; neither does it represent that it



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   has made any effort to identify any such rights.  Information on the
   IETF's procedures with respect to rights in standards-track and
   standards-related documentation can be found in BCP 11 [5].  Copies
   of claims of rights made available for publication and any assurances
   of licenses to be made available, or the result of an attempt made to
   obtain a general license or permission for the use of such
   proprietary rights by implementors or users of this specification can
   be obtained from the IETF Secretariat.

   The IETF invites any interested party to bring to its attention any
   copyrights, patents or patent applications, or other proprietary
   rights which may cover technology that may be required to practice
   this standard.  Please address the information to the IETF Executive
   Director.

Full Copyright Statement

   Copyright (C) The Internet Society (2003). All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implementation may be prepared, copied, published
   and distributed, in whole or in part, without restriction of any
   kind, provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works.  However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Acknowledgments

   We thank the following people: Colin Perkins, Steve Casner, and
   Henning Schulzrinne for their considered guidance; Sue Moon for
   helping foster collaboration between the authors; Mounir Benzaid for



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   drawing our attention to the reporting needs of MLDA; Dorgham Sisalem
   and Adam Wolisz for encouraging us to incorporate MLDA block types;
   and Jose Rey for valuable review of the SDP Signaling section.

Contributors

   The following people are the authors of this document:

   Kevin Almeroth, UCSB
   Ramon Caceres, ShieldIP
   Alan Clark, Telchemy
   Robert Cole, AT&T Labs
   Nick Duffield, AT&T Labs-Research
   Timur Friedman, Paris 6
   Kaynam Hedayat, Brix Networks
   Kamil Sarac, UT Dallas
   Magnus Westerlund, Ericsson

   The principal people to contact regarding the individual report
   blocks described in this document are as follows:


   sec. report block                          principal contributors
   ---- ------------                          ----------------------
   4.1  Loss RLE Report Block                 Friedman, Caceres, Duffield
   4.2  Duplicate RLE Report Block            Friedman, Caceres, Duffield
   4.3  Packet Receipt Times Report Block     Friedman, Caceres, Duffield
   4.4  Receiver Reference Time Report Block  Friedman
   4.5  DLRR Report Block                     Friedman
   4.6  Statistics Summary Report Block       Almeroth, Sarac
   4.7  VoIP Metrics Report Block             Clark, Cole, Hedayat


   The principal person to contact regarding the SDP signaling described
   in this document is Magnus Westerlund.

Authors' Addresses

      Kevin Almeroth <almeroth@cs.ucsb.edu>
      Department of Computer Science
      University of California
      Santa Barbara, CA 93106 USA

      Ramon Caceres <ramon@shieldip.com>
      ShieldIP, Inc.
      11 West 42nd Street, 31st Floor
      New York, NY 10036 USA




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      Alan Clark <alan@telchemy.com>
      Telchemy Incorporated
      3360 Martins Farm Road, Suite 200
      Suwanee, GA 30024 USA
      Tel: +1 770 614 6944
      Fax: +1 770 614 3951

      Robert Cole <rgcole@att.com>
      AT&T Labs
      330 Saint Johns Street,
      2nd Floor
      Havre de Grace, MD 21078 USA
      Tel: +1 410 939 8732
      Fax: +1 410 939 8732

      Nick Duffield <duffield@research.att.com>
      AT&T Labs-Research
      180 Park Avenue, P.O. Box 971
      Florham Park, NJ 07932-0971 USA
      Tel: +1 973 360 8726
      Fax: +1 973 360 8050

      Timur Friedman <timur.friedman@lip6.fr>
      Universite Pierre et Marie Curie (Paris 6)
      Laboratoire LiP6-CNRS
      8, rue du Capitaine Scott
      75015 PARIS France
      Tel: +33 1 44 27 71 06
      Fax: +33 1 44 27 74 95

      Kaynam Hedayat <khedayat@brixnet.com>
      Brix Networks
      285 Mill Road
      Chelmsford, MA 01824 USA
      Tel: +1 978 367 5600
      Fax: +1 978 367 5700

      Kamil Sarac <ksarac@utdallas.edu>
      Department of Computer Science (ES 4.207)
      Eric Jonsson School of Engineering & Computer Science
      University of Texas at Dallas
      Richardson, TX 75083-0688 USA
      Tel: +1 972 883 2337
      Fax: +1 972 883 2349







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      Magnus Westerlund <magnus.westerlund@era.ericsson.se>
      Ericsson Research, Corporate Unit
      Ericsson Radio Systems AB
      SE-164 80 Stockholm
      Sweden
      Tel: +46 8 404 82 87
      Fax: +46 8 757 55 50

References

   The references are divided into normative references and non-
   normative references.

Normative References

   [1] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," BCP 14, RFC 2119, IETF, March 1997.

   [2] D. Crocker, P. Overell, "Augmented BNF for Syntax Specifications:
   ABNF", RFC 2234, Internet Engineering Task Force, November 1997.

   [3] ETSI, "Quality of Service (QoS) measurement methodologies," ETSI
   TS 101 329-5 V1.1.1 (2000-11), November 2000.

   [4] M. Handley, V. Jacobson, "SDP: Session Description Protocol", RFC
   2327, Internet Engineering Task Force, April 1998.

   [5] R. Hovey and S. Bradner, "The Organizations Involved in the IETF
   Standards Process," best current practice BCP 11, RFC 2028, IETF,
   October 1996.

   [6] ITU-T, "The E-Model, a computational model for use in
   transmission planning," Recommendation G.107 (05/00), May 2000.

   [7] T. Narten and H. Alvestrand, "Guidelines for Writing an IANA
   Considerations Section in RFCs," best current practice BCP 26, RFC
   2434, IETF, October 1998.

   [8] J. Rosenberg, H. Schulzrinne, "An Offer/Answer Model with the
   Session Description Protocol (SDP)", RFC 3264, Internet Engineering
   Task Force, June 2002.

   [9] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
   transport protocol for real-time applications," RFC 1889, IETF,
   February 1996.

   [10] TIA/EIA-810-A Transmission Requirements for Narrowband Voice
   over IP and Voice over PCM Digital Wireline Telephones, December



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   2000.

Non-Normative References

   [11] A. Adams, T. Bu, R. Caceres, N. G. Duffield, T. Friedman, J.
   Horowitz, F. Lo Presti, S. B. Moon, V. Paxson, and D. Towsley, "The
   Use of End-to-End Multicast Measurements for Characterizing Internal
   Network Behavior," IEEE Communications Magazine, May 2000.

   [12] Baugher, McGrew, Oran, Blom, Carrara, Naslund, and Norrman, "The
   Secure Real-time Transport Protocol," Internet-Draft draft-ietf-avt-
   srtp-05.txt, June 2002.  Note: this is is a work in progress.

   [13] R. Caceres, N. G. Duffield, and T. Friedman, "Impromptu
   measurement infrastructures using RTP," Proc. IEEE Infocom 2002.

   [14] A. D. Clark, "Modeling the Effects of Burst Packet Loss and
   Recency on Subjective Voice Quality," Proc. IP Telephony Workshop
   2001.

   [15] M. Handley, C. Perkins, E. Whelan, "Session Announcement
   Protocol", RFC 2974, Internet Engineering Task Force, October 2000.

   [16] J. Reynolds, "Assigned Numbers: RFC 1700 is Replaced by an On-
   line Database", RFC 3232, IETF, January 2002.

   [17] H. Schulzrinne, R. Lanphier, and A. Rao, "Real time streaming
   protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
   1998.

   [18] D. Sisalem and A. Wolisz, "MLDA: A TCP-friendly Congestion
   Control Framework for Heterogeneous Multicast Environments", Proc.
   IWQoS 2000.


















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