INTERNET-DRAFT 24 January 2003
Internet Engineering Task Force Expires: 24 July 2003
Audio/Video Transport Working Group
Timur Friedman, Paris 6
Ramon Caceres, ShieldIP
Alan Clark, Telchemy
Editors
RTP Extended Reports (RTP XR)
draft-ietf-avt-rtcp-report-extns-02.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). 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
RTCP's sender report (SR) and receiver report (RR) packets. Some
applications, such as multicast inference of network characteristics
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(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 adhereing to
the framework that this document provides.
Table of Contents
1. Introduction .............................................. 2
1.1 Terminology ............................................... 3
2. XR Packet Format .......................................... 4
3. Extended Report Block Framework ........................... 5
4. Extended Report Blocks .................................... 6
4.1 Loss RLE Report Block ..................................... 6
4.1.1 Run Length Chunk .......................................... 12
4.1.2 Bit Vector Chunk .......................................... 12
4.1.3 Terminating Null Chunk .................................... 12
4.2 Duplicate RLE Report Block ................................ 13
4.3 Timestamp Report Block .................................... 13
4.4 Statistics Summary Report Block ........................... 16
4.5 Receiver Timestamp Report Block ........................... 19
4.6 DLRR Report Block ......................................... 20
4.7 VoIP Metrics Report Block ................................. 21
4.7.1 Packet Loss and Discard Metrics ........................... 23
4.7.2 Burst Metrics ............................................. 23
4.7.3 Delay Metrics ............................................. 26
4.7.4 Signal Related Metrics .................................... 26
4.7.5 Call Quality or Transmission Quality Metrics .............. 29
4.7.6 Configuration Parameters .................................. 30
4.7.7 Jitter Buffer Parameters .................................. 31
5. IANA Considerations ....................................... 32
5.1 XR Packet Type ............................................ 32
5.2 RTP XR Block Type Registry ................................ 32
6. Security Considerations ................................... 33
A. Algorithms ................................................ 34
A.1 Sequence Number Interpretation ............................ 34
A.2 Example Burst Packet Loss Calculation ..................... 35
Intellectual Property ..................................... 37
Full Copyright Statement .................................. 38
Acknowledegments .......................................... 38
Contributors .............................................. 39
Authors' Addresses ........................................ 39
References ................................................ 40
Normative References ...................................... 40
Non-Normative References .................................. 41
1. Introduction
This document defines the extended report (XR) packet type for the
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RTP control protocol (RTCP) [7]. 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 following sections of
this document, starting with a general framework and finishing with
the specific information conveyed. The framework defined by Section
2 contains common header information followed by a series of
components called report blocks. Section 3 defines the format common
to such blocks. Section 4 defines seven block types.
Seven report block formats are defined by this document:
- Loss RLE Report Block (Section 4.1): Run length encoding of RTP
packet loss reports.
- Duplicate RLE Report Block (Section 4.2): Run length encoding of
reports of RTP packet duplicates.
- Timestamp Report Block (Section 4.3): A list of timestamps of
received RTP packets.
- Statistics Summary Report Block (Section 4.4): Statistics on RTP
packet sequence numbers, losses, duplicates, jitter, and TTL values.
- Receiver Timestamp Report Block (Section 4.5): Receiver-end
timestamps that complement the sender-end timestamps already defined
for RTCP.
- DLRR Report Block (Section 4.6): The delay since the last Receiver
Timestamp Report Block was received, allowing non-senders to
calculate round-trip times.
- VoIP Metrics Report Block (Section 4.7): Metrics for monitoring
Voice over IP (VoIP) calls.
These blocks are defined within a minimal framework: a type field and
a length field are common to all XR blocks. The purpose is to
maintain flexibility and to keep overhead low. 0ther block formats,
beyond the seven defined here, may be defined within this framework
as the need arises.
1.1 Terminology
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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
The XR packet consists of a header of two 32-bit words, followed by a
number, possibly zero, of extended report blocks.
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=XP=207 | length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC/CSRC |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: 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
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
the receiver MUST ignore any XR packet with a non-zero value in
this field.
packet type (PT): 8 bits
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 RTP sender report (SR) packet (see Section
6.3.1 of the RTP specification [7]). Briefly, the length of
this XR packet in 32-bit words minus one, including the header
and any padding.
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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 data :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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
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
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the BT, type-specific, and block length fields, with a null
type-specific data field.
type-specific data: 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
losses, duplicates, packet reception timestamps, detailed reception
statistics, receiver timestamps, receiver inter-report delays, and
voice over IP (VoIP) metrics. An implementation MAY 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) [5], 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) [8]. 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 [9].
A participant that generates a Loss RLE Report Block should favor
accuracy in reporting on observed events over interpretation of those
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 [7] 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
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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 accouting 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
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 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.
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Note that because of sequence number wrap around 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 wrap arounds.
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
wrap around of sequence numbers (or since the beginning of the RTP
session if no wrap around 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 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
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:
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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.
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
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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
the receiver MUST ignore any Loss RLE Report Block with a non-
zero value in this field.
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.
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
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
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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 wrap
around).
4.1.3 Terminating Null Chunk
This chunk is all zeroes.
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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 existance 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
the receiver MUST ignore any Duplicate RLE Report Block with a
non-zero value in this field.
thinning (T): 4 bits
As defined in Section 4.1.
block length: 16 bits
Defined in Section 3.
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 Timestamp Report Block
This block type permits per-sequence-number reports on packet receipt
timestamps for a given source's RTP packet stream. Such information
can be used for MINC inference of the topology of the multicast tree
used 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
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path characteristics and to model distributions for packet jitter.
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
timestamps 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 receiver timestamps, any timestamp other than the earliest MUST
NOT be reported. This is to ensure consistency among reports.
Timestamps 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 Timestamp Report Blocks.
The Timestamp 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=3 | rsvd. | T | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP timestamp 1 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP timestamp 2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: ... :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RTP timestamp n |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
A Timestamp 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
the receiver MUST ignore any Timestamp Report Block with a non-
zero value in this field.
thinning (T): 4 bits
As defined in Section 4.1.
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block length: 16 bits
Defined in Section 3.
begin_seq: 16 bits
As defined in Section 4.1.
end_seq: 16 bits
As defined in Section 4.1.
RTP timestamp i: 32 bits
The timestamp reflects the packet arrival time at the receiver.
It is preferable for the timestamp to be established at the link
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 receiver timestamps should reflect the
actual time as closely as possible. The initial value of the
timestamp is random, and subsequent timestamps are offset from
this value.
4.4 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
values (TTL values being taken from the TTL field of IPv4 packets, if
the data packets are carried over IPv4). Such information can be
useful for network management.
The report block contents are dependent upon a bit vector 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 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=4 |L|D|J|T|resvd. | block length = 9 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| lost_packets |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| dup_packets |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| min_jitter |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| max_jitter |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| avg_jitter |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| dev_jitter |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| min_ttl | max_ttl | avg_ttl | dev_ttl |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
A Statistics Summary Report Block is identified by the constant
4.
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, avg_jitter, and
dev_jitter fields all contain reports, 0 if none of them do.
TTL flag (T): 1 bit
Bit set to 1 if the min_ttl, max_ttl, avg_ttl, and dev_ttl
fields all contain reports, 0 if none of them do.
resvd.: 4 bits
This field is reserved for future definition. In the absence of
such definition, all bits in this field MUST be set to zero, and
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the receiver MUST ignore any Statistics Summary Report Block
with a non-zero value in this field.
block length: 16 bits
The constant 9, in accordance with the definition of this field
in Section 3.
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.
avg_jitter: 32 bits
The average relative transit time between each two packet series
in the above sequence number interval.
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: 8 bits
The minimum TTL value of data packets in sequence number range.
max_ttl: 8 bits
The maximum TTL value of data packets in sequence number range.
avg_ttl: 8 bits
The average TTL value of data packets in sequence number range.
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dev_ttl: 8 bits
The standard deviation of TTL values of data packets in sequence
number range.
4.5 Receiver Timestamp 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 [7, Sec. 6.3.1]. A non-sender may estimate
its RTT to other participants, as proposed in [11], 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=5 | reserved | block length = 2 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NTP timestamp, most significant word |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| NTP timestamp, least significant word |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
A Receiver Timestamp Report Block is identified by the constant
5.
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
the receiver MUST ignore any Receiver Timestamp Report Block
with a non-zero value in this field.
block length: 16 bits
The constant 2, in accordance with the definition of this field
in Section 3.
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
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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.
4.6 DLRR Report Block
This block extends RTCP's delay since last sender report (DLSR)
mechanism [7, Sec. 6.3.1] so that non-senders may also calculate
round trip times, as proposed in [11]. 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
respondant. 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=6 | 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 6.
reserved: 8 bits
This field is reserved for future definition. In the absence of
such definition, all bits in this field MUST be set to zero, the
receiver MUST ignore any DLRR Report Block with a non-zero value
in this field.
block length: 16 bits
Defined in Section 3.
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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
Timestamp 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 Timestamp Report Block from
participant SSRC_n and sending this DLRR Report Block. If no
Receiver Timestamp 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-LSR using the last SR timestamp (LSR)
field, and then subtracting this field to leave the round-trip
propagation delay as A-LSR-DLSR. This is illustrated in [7, Fig.
2].
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 [10].
Following a Gilbert-Elliott model [2], 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 RTCP 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 [2]) and mean opinion scores (MOS scores).
An implementation that sends these blocks SHOULD send at least one
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every ten seconds for the duration of the call, SHOULD send one
whenever a CODEC type change or other significant change occurs,
SHOULD send one when significant call quality degradation is detected
and SHOULD send one upon call termination. 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:
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 = 6 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| loss rate | discard rate | burst density | gap density |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| burst duration | gap duration |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| round trip delay | end system delay |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| signal power | RERL | noise level | Gmin |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| R factor | ext. R factor | MOS-LQ | MOS-CQ |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RX config | 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, all bits in this field MUST be set to zero and
the receiver MUST ignore any VoIP Metrics Report Block with a
non-zero value in this field.
block length: 16 bits
The constant 6, in accordance with the definition of this field
in Section 3.
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.
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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
identify the source of quality degradation. These fields MUST be
populated.
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, informally, is a period of high packet losses and/or
discards. Formally, a burst is defined as a longest sequence of
packets bounded by lost or discarded packets with the constraint that
within a burst any sequence of successive packets that were received,
and not discarded due to delay variation, is of length less than a
value Gmin.
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
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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.
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
density 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 density 12, which corresponds to 5%
discard density 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
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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
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.
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.
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
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is replaced by the reception end time. If there have been no
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 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 during the first two or three RTCP
exchanges as insufficient data may be available to determine
delay however 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 total
encoding, decoding and jitter buffer delay determined at the
reporting endpoint. This is the time required for an RTP frame
to be buffered, decoded, converted to analog form, looped back
at the local analog interface, encoded, and made available for
transmission as an RTP frame. The manner in which this value is
estimated is implementation dependent. This parameter MUST be
provided in all VoIP metrics reports.
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 power: 8 bits
The voice signal relative level is defined as the ratio of the
signal level to overflow signal level, 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. If the full range
(overflow level) of the Vocoder's digital to analog conversion
function is +/- L and the value of a decoded sample during a
talkspurt is V then the signal level is given by:
signal level = 10 log10 ( mean( abs(V) / L ) )
A value of 127 indicates that this parameter is unavailable.
residual echo return loss (RERL): 8 bits
The residual echo return loss is defined as the sum of the
measured echo return loss (ERL) and the echo return loss
enhancement (ERLE) expressed in dB as a signed integer in two's
complement form. It defines the ratio of a transmitted voice
signal that is reflected back to the talker. A low level of
echo return loss (say less than 20 dB) in conjunction with some
delay can lead to hollowness or audible echo. A high level of
echo return loss (say over 40 dB) is preferable.
The ERL and ERLE parameters are often available directly from the
echo cancellor contained within the VoIP end system. They relate to
the echo on the external network attached to the end point.
In the case of a VoIP gateway this would be line echo that typically
occurs at 2-4 wire conversion points in the network. Echo return
loss from typical 2-4 wire conversions can be in the 8-12 dB range.
A line echo cancellor 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
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 due to acoustic feedback.
Typical values for RERL are as follows:
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(i) IP gateway connected to circuit switched network with 2 wire loop
Without echo cancellation, typical 2-4 wire convertor ERL of 12 dB
RERL = ERL + ERLE = 12 + 0 = 12 dB
(ii) IP gateway connected to circuit switched network with 2 wire loop
With echo cancellor that improves echo by 30 dB
RERL = ERL + ERLE = 12 + 30 = 42 dB
(iii) IP phone with conventional handset
Acoustic coupling from handset speaker to microphone 40 dB
Residual echo return loss = 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.
For an IP phone with handset this metric MUST be set to the designed
or measured terminal coupling loss, which would typically be 40-50
dB.
For a PC softphone or speakerphone this metric MUST be set to either
the value reported by the acoustic echo cancellor or to 127 to
indicate an undefined value.
For an IP gateway with ERL and ERLE measurements this metric MUST be
reported as ERL + ERLE.
For an IP gateway without ERL and ERLE measurement capability then
this metric MUST be reported as 12 dB if line echo cancellation is
disabled and 40 dB if line echo cancellation is enabled.
noise level: 8 bits
The noise level is defined as the ratio of the silent period
back ground noise level to overflow signal power, expressed in
decibels as a signed integer in two's complement form. If the
full range (overflow level) of the Vocoder's digital to analog
conversion function is +/- L and the value of a decoded sample
during a silence period is V then the noise level is given by
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noise level = 10 log10 ( mean( abs(V) / L ) )
A value of 127 indicates that this parameter is unavailable.
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 [4] and ETSI TS
101 329-5 [2].
A value of 127 indicates that this parameter is unavailable.
ext. R factor: 8 bits
The external R factor is a voice quality metric describing the
seg ment 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 [4] and ETSI TS 101 329-5
[2], 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.
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)
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
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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.
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 [4] or ETSI TS 101 329-5 [2] 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.
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 dis card 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 [2] 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.
receiver configuration byte (RX config): 8 bits
This byte consists of the following fields:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|PLC|JBA|JB rate|
+-+-+-+-+-+-+-+-+
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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 is typically able to conceal isolated lost packets
with loss rates under 3%. When PLC = 10 then an enhanced
interpolation algorithm is being used. This would
typically be able to conceal lost packets for loss rates of
10% or more. When PLC = 01 then silence is 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.
4.7.7 Jitter Buffer Parameters
jitter buffer nominal size in frames (JB nominal): 8 bits
This is the current nominal fill point within the jitter buffer,
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 size in frames (JB maximum): 8 bits
This is the current maximum jitter buffer level corresponding to
the earliest arriving packet that would not be discarded. In
simple queue implementations this may correspond to the nominal
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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 size in frames (JB abs max): 8 bits
This is the absolute maximum size that the adaptive jitter
buffer can reach under worst case jitter conditions. 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. IANA Considerations
This document defines a new RTP 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 RTP 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.
5.1 XR Packet Type
The IANA SHALL register the RTP extended report (XR) packet defined
by this document as packet type 207 in the registry of RTP RTCP
Control Packet types (PT).
5.2 RTP XR Block Type Registry
The IANA SHALL create the RTP XR Block Type Registry to cover the
name space of the extended report block type (BT) field specified in
Section 3 of this document. The BT field contains eight bits,
allowing 256 values. The IANA SHALL manage the RTP XR Block Type
Registry according to the Specification Required policy of RFC 2434
[6]. Future specifications SHOULD attribute block type values in
strict numeric order following the values attributed in this
document:
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BT name
-- ----
1 Loss RLE Report Block
2 Duplicate RLE Report Block
3 Timestamp Report Block
4 Statistics Summary Report Block
5 Receiver Timestamp Report Block
6 DLRR Report Block
7 VoIP Metrics Report Block
Furthermore, future specifications SHOULD avoid the values 0 and 255.
Doing so facilitates packet validity checking, since all-zeros and
all-ones are values that might commonly be found in ill-formed
packets.
6. Security Considerations
This document extends the RTCP reporting mechanism. The security
considerations that apply to RTCP reports 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 information contained in Loss RLE, Duplicate RLE, and
Timestamp Report Blocks facilitates MINC inference of multicast
distribution trees for RTP data packets, and inference of link
characteristics such as loss and delay. This inference reveals
information that might otherwise be considered confidential 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. Standard encryption procedures can be used when
confidentiality is a concern to end hosts. Autonomous system
administrators concerned about the loss of confidentiality regarding
their networks can encrypt traffic, or filter it to exclude RTCP
packets containing the XR report blocks concerned.
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
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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
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 limit the size of XR report blocks and employ the
thinning techniques described in this document in order to fit
reports into the space available.
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) {
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// This is the first sequence number received. Place
// 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
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VoIP Metrics Report Block (Section 4.7). It is reproduced from ETSI
TS 101 329-5 [2]. The 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;
}
else {
lost++;
if(pkt == 0) {
c33++;
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}
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 densities */
loss_density = 256 * loss_count / ctotal;
discard_density = 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
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 [3]. Copies
of claims of rights made available for publication and any assurances
Friedman, Caceres, and Clark, eds. [Page 37]
draft-ietf-avt-rtcp-report-extns-02.txt 24 January 2003
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.
Acknowledgements
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; Magnus Westerlund
for his detailed comments; Mounir Benzaid for drawing our attention
to the reporting needs of MLDA; and Dorgham Sisalem and Adam Wolisz
for encouraging us to incorporate MLDA block types.
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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
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 Timestamp Report Block Friedman, Caceres, Duffield
4.4 Statistics Summary Report Block Almeroth, Sarac
4.5 Receiver Timestamp Report Block Friedman
4.6 DLRR Report Block Friedman
4.7 VoIP Metrics Report Block Clark, Cole, Hedayat
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
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
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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
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.
Friedman, Caceres, and Clark, eds. [Page 40]
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[2] ETSI, "Quality of Service (QoS) measurement methodologies," ETSI
TS 101 329-5 V1.1.1 (2000-11), November 2000.
[3] R. Hovey and S. Bradner, "The Organizations Involved in the IETF
Standards Process," best current practice BCP 11, RFC 2028, IETF,
October 1996.
[4] ITU-T, "The E-Model, a computational model for use in
transmission planning," Recommendation G.107 (05/00), May 2000.
[5] J. Reynolds and J. Postel, "Assigned Numbers," standard STD 2,
RFC 1700, IETF, October 1994.
[6] T. Narten and H. Alvestrand, "Guidelines for Writing an IANA
Considerations Section in RFCs," best current practice BCP 26, RFC
2434, IETF, October 1998.
[7] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: A
transport protocol for real-time applications," RFC 1889, IETF,
February 1996.
Non-Normative References
[8] 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.
[9] R. Caceres, N. G. Duffield, and T. Friedman, "Impromptu
measurement infrastructures using RTP," Proc. IEEE Infocom 2002.
[10] A. D. Clark, "Modeling the Effects of Burst Packet Loss and
Recency on Subjective Voice Quality," Proc. IP Telephony Workshop
2001.
[11] D. Sisalem and A. Wolisz, "MLDA: A TCP-friendly Congestion
Control Framework for Heterogeneous Multicast Environments", Proc.
IWQoS 2000.
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