Internet DRAFT - draft-clark-avt-rtcphr
draft-clark-avt-rtcphr
Internet Engineering Task Force A. Clark
Internet-Draft Telchemy Incorporated
Expires: 22 December 2006 A. Pendleton
Nortel
R. Kumar
K. Connor
Cisco Systems
G. Hunt
BT
June 2006
RTCP HR - High Resolution VoIP Metrics Report Blocks
draft-clark-avt-rtcphr-02
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Copyright Notice
Copyright (C) The Internet Society (2006).
Abstract
This document defines extensions to the RTCP XR extended report
packet type blocks to support Voice over IP (VoIP) monitoring
for services that require higher resolution or more detailed
metrics than those supported by RFC3611 [13].
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . 2
3. High Resolution VoIP Metrics Report Block . . . . . . . . 5
4. RTCP HR Configuration Block . . . . . . . . . . . . . . . . 21
5. Application of RTCP HR to RTP translators. . . . . . . . . . 24
6. Practical applications . . . . . . . . . . . . . . . . . . 32
7. Summary . . . . . . . . . . . . . . . . . . . . . . . . . . 32
8. IANA Considerations . . . . . . . . . . . . . . . . . . . . 32
9. Security Considerations . . . . . . . . . . . . . . . . . . 33
10. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 33
11. Informative References . . . . . . . . . . . . . . . . . . . 33
Authors' Addresses . . . . . . . . . . . . . . . . . . . . .
Intellectual Property and Copyright Statements . . . . . . .
1. Introduction
This draft defines several new block types to augment those defined
in RFC3611 for use in Quality of Service reporting for Voice over IP.
The new block types support the reporting of metrics to a higher
resolution to support certain applications, for example carrier
backbone networks.
For certain types of VoIP service it is desirable to report VoIP
performance metrics to a higher resolution than provided in the
RFC3611 [13] VoIP Metrics block or RFC3550 [12] Receiver Reports.
The report blocks described in this section provide both interval
based and cumulative metrics with a higher resolution than that
provided in the RFC3611 VoIP metrics report block.
The new block types defined in this draft are the High Resolution
VoIP Metrics Report Block, and the High Resolution VoIP Metrics
Configuration Block.
2. Definitions
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2.1 Reporting modes
RTCP HR report blocks may be produced by any of the types of RTP
systems which are defined in RFC3550 [12] as capable of receiving RTP
packets, that is, by an RTP end system, by an RTP mixer, or by an
RTP translator. For purposes of sending reports about distribution
quality, a mixer is equivalent to an end system, since mixers
resynchronize audio packets before transmission and do not forward
reception reports from contributing sources towards the destination
of the mixed output RTP stream. However an RTP translator has RTP
interfaces in more than one transport-level "cloud" RFC3550 [12],
participating in the same session, and does not resynchronize audio
packets in transit between "clouds". A translator may:-
- forward RTCP reports generated by RTP systems in one cloud towards
RTP systems in another cloud, possibly with modification
- generate its own RTCP reports and send them to one or more of the
clouds to which it is connected.
This document defines three modes for reporting by translators.
These modes are defined briefly here and described in more detail in
section 5 below.
"End system peering" is a mode in which a translator forwards RTCP
reports originating from an RTP end system towards other RTP end
systems (possibly with some modification consistent with the
translation which it performs RFC3550 [12]). RTCP reports received
from an end system are forwarded towards the same destination(s) as
the destination(s) of the RTP packets received at the same interface
as that at which the RTCP reports were received. This is the minimum
which a translator must do to maintain the integrity of RTCP
communication between the end systems. In end system peering mode,
the translator does not originate any RTCP reports based on any
measurements which it may make locally.
"Local reporting" is a mode in which the translator forwards the end
systems' reports as in "end system peering" mode, to maintain RTCP
communication between end systems. In addition, the translator makes
measurements on the RTP stream arriving at each receive transport
interface, creating associated RTCP reports. These RTCP reports are
sent by the translator both towards the source and destination(s)
of the monitored RTP stream. However, the translator does not forward
RTCP reports generated by other translators. "Local reporting" mode
allows translators to assist in localising transport faults, whilst
keeping RTCP traffic and processing bounded in systems which use
large numbers of translators.
"Global reporting" is a mode in which the translator forwards the
end systems' reports as in "end system peering" mode, to maintain
RTCP communication between end systems. In addition the translator
makes measurements on the RTP stream arriving at each receive
transport interface, creating associated RTCP reports. These RTCP
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reports are sent by the translator both towards the source and
destination(s) of the RTP stream. Finally, the translator also
forwards RTCP reports generated by other translators, but only
towards the same destination(s) as the destination(s) of the RTP
stream received at the same interface as that at which the RTCP
reports were received. In particular, no RTCP report received at a
translator from a system S is ever sent back towards S. "Global
reporting" provides complete transport metrics to every RTP system
involved in a session. In particular, end systems receive reports of
measurements made on RTP streams at all receive interfaces of every
translator. However, the volume of RTCP traffic and RTCP processing
may become excessively large in systems using large numbers of
translators.
In "local reporting" and "global reporting" modes, RTCP reports are
generated based on measurements made on RTP streams received by
translators. Typically translators are not able to make all of the
measurements which an end system is able to make. For example, a
translator does not contain a jitter buffer so cannot report on
discarded packets. However, translators may make useful measurements
of packet loss, round trip delay, and packet delay variation.
2.2 Cumulative and Interval Metrics
Cumulative metrics relate to the entire duration of the call to the
point at which metrics are determined and reported, and are typically
used to report call quality. Cumulative metrics generally result in
a lower volume of data that may need to be stored, as each report
supersedes earlier reports.
Interval metrics relate to the period since the last Interval report.
Interval data may be easier to correlate with specific network events
for which timing is known, and may also be used as a basis for
threshold crossing alerts.
Note that interval metrics for the start and end of calls may be
unreliable due to factors such as irregular start and end interval
length and the difficulty in knowing when packet transmission started
and ended.
2.3 Bursts, Gaps, and Concealed Seconds
The terms Burst and Gap are used in a manner consistent with that of
RTCP XR (RFC3611). RTCP XR views a call as being divided into bursts,
which are periods during which the combined packet loss and discard
rate is high enough to cause noticeable call quality degradation
(generally over 5 percent loss/discard rate), and gaps, which are
periods during which lost or discarded packets are infrequent and
hence call quality is generally acceptable.
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The recommended value for Gmin in RFC3611 results in a Burst being a
period of time during which the call quality is degraded to a similar
extent to a typical PCM Severely Errored Second.
The term Concealed Seconds defines a count of seconds during which
some proportion of time was lost through packet loss and discard. The
term Severely Concealed Seconds defines a count of seconds during
which the proportion of time lost through packet loss and discard
exceeds a specified threshold.
2.4 Numeric formats
This report block makes use of binary fractions. The terminology
used is
S X:Y, where S indicates a signed representation,
X the number of bits prior to the decimal place and
Y the number of bits after the decimal place.
Hence 8:8 represents an unsigned number in the range 0.0, 0.0039 to
255.996. S7:8 would represent the range -127.996 to +127.996.
3 High Resolution VoIP Metrics Report Block
3.1 Block Description
This block comprises a header and a series of sub-blocks. The
Map field in the header defines which sub-blocks are present.
Header sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=N | Map | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Duration |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Basic Loss/Discard Metrics sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Loss Proportion | Discard Proportion |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Number of frames expected |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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Burst/Gap metrics sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Threshold | Burst Duration (ms) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Gap Duration (ms) |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Burst Loss/Disc Proportion | Gap Loss/Disc Proportion |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Playout metrics sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| On-time Playout Duration |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| On-time Active Speech Playout Duration |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Loss Concealment Duration |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Buffer Adjustment Concealment Duration |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Concealed Seconds metrics sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Unimpaired Seconds |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Concealed Seconds |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Severely Concealed Seconds | RESERVED | threshold |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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Delay and PDV metrics sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Network Round Trip Delay | End System Delay |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| External Delay | Mean PDV |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Pos Threshold PDV | Pos PDV Percentile |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Neg Threshold PDV | Neg PDV Percentile |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| JB config | JB Metric | JB nominal |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| JB maximum | JB abs max |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Call Quality metrics sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| R-LQ | R-CQ |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MOS-LQ | MOS-CQ |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| R-LQ Ext In | R-LQ Ext Out |RFC3550 Payload| Media Type |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RxSigLev (IP) |RxNoiseLev (IP)| Local RERL | Remote RERL |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RxSigLev (Ext)|RxNoiseLev(Ext)| Metric Status |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Vendor specific extension sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Vendor ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Vendor ID Src |Manuf Code Src | Extension Block Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Vendor-specific extension data |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
3.2 Header
Implementations MUST send the Header block within each High
Resolution Metrics report.
3.2.1 Block type
Nine High Resolution VoIP Metrics blocks are defined
mmm = HR Metrics- Cumulative, Locally Generated
mmm+1 = HR Metrics- Cumulative, Relayed from Remote IP Endpoint
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mmm+2 = HR Metrics- Cumulative, Relayed from Remote Ext Endpoint
mmm+3 = HR Metrics- Cumulative,TBD
mmm+4 = HR Metrics- Interval, Locally Generated
mmm+5 = HR Metrics- Interval, Relayed from Remote IP Endpoint
mmm+6 = HR Metrics- Interval, Relayed from Remote Ext Endpoint
mmm+7 = HR Metrics- Interval, TBD
mmm+7 = HR Metrics- Alert, Locally Generated
mmm+9 = HR Metrics- Alert, Relayed from Remote IP Endpoint
mmm+10 = HR Metrics- Alert, Relayed from Remote Ext Endpoint
mmm+11 = HR Metrics- Alert, TBD
<Authors' note: The number of report types is being reviewed and
may be reduced to three - Cumulative, Interval, Alert>
The time interval associated with these report blocks is left to the
implementation. Spacing of RTCP reports should be in accordance
with RFC3550. The specific timing of RTCP HR reports may be
determined in response to an internally derived alert such as a
threshold violation however the spacing of RTCP HR reports must not
exceed that defined in RFC3550.
Note that interval data may be derived by subtracting successive
cumulative reports, which provides increased tolerance to potential
loss of RTCP reports.
When these block types are used in SDP offer-answer, semantics are
as defined in RFC3611 [13]. In particular, for "sendrecv" unicast
connections, a block type in an offer indicates the offerer's wish
to receive the specified block type. A block type in an answer
indicates the answerer's wish to receive the specified block type.
<Authors' note: need to review idea of negotiation of sub-blocks by
SDP offer-answer?>
3.2.2 Map field
A Map field indicates the optional sub-blocks present in this
report. A 1 indicates that the sub-block is present, and a 0 that
the block is absent. If present, the sub-blocks must be in the
sequence defined in this document.
The bits have the following definitions:
0 Burst/Gap Metrics block
1 Playout Metrics block
2 Concealed Seconds Metrics block
3 Call Quality Metrics
4 Vendor specific extension block
5-15 Reserved, set to 0
3.2.3 Block Length
The block length indicates the length of this report in 32 bit
words and includes the header and any extension octets.
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3.2.4 SSRC
The SSRC of the stream to which this report relates. The value
of this field shall follow the rules defined in RFC3550 with
regard to the forwarding of RTP and RTCP messages.
3.2.5 Duration
The duration of time for which this report applies expressed in
milliseconds. For cumulative reports this would be the call
duration. For interval reports this would be the duration of the
interval.
3.3 Basic Loss/ Discard Metrics
The Basic Loss/Discard Metrics sub-block MUST be present.
This block reports the proportion of frames lost by the network and
the proportion of frames discarded due to jitter.
For sample-based codecs such as G.711, a frame shall be defined as
an RTP frame. For endpoints that incorporate jitter buffers capable
of fractional frame discard the proportion of frames discarded MAY
be determined on the basis of the proportion of samples discarded.
If Voice Activity Detection is used then the proportion of frames
lost and discarded shall be determined based on transmitted packets,
i.e. frames that contained silence and were not transmitted shall
not be considered.
A frame shall be regarded as lost if it fails to arrive within an
implementation-specific time window. A frame that arrives within
this time window but is too early or late to be played out shall
be regarded as discarded. A frame shall be classified as one of
received (or OK), discarded or lost.
The Loss and Discard metrics are determined after the effects of
FEC, redundancy (RFC2198) or other similar process.
3.3.1 Proportion of frames lost
Proportion of frames lost within the network expressed as a binary
fraction in 0:16 format. Duplicate frames shall be disregarded.
3.3.2 Proportion of frames discarded
Proportion of voice frames received but discarded due to late or
early arrival, expressed as a binary fraction in 0:16 format.
3.3.3 Dead connection detection
If no RTP, SID or RTP no-op packets have been received for a pre-
specified time interval(for example ten seconds) then an RTCP HR
dead connection indication MUST be sent. This time interval may
be changed during the connection, for example being longer at the
start of a call. Dead connection detection may be temporarily
disabled during silence periods if VAD is used.
A dead connection is indicated by setting both the frames lost
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and frames discarded fields above to 0xFFFF (equivalent to
indicating 99.99% of packets have been both lost and discarded).
Receipt of an RTCP HR block with either field set to a value other
than 0xFFFF shall indicate that the remote endpoint HAS received
some valid RTP packets.
3.3.4 Number of frames expected
A count of the number of frames expected, estimated if necessary.
If no frames have been received then this count shall be set to
zero, however if a dead connection is indicated then this count
shall be regarded as undefined.
3.4 Burst/Gap metrics sub-block
The Burst/Gap metrics sub-block MAY be present and if present MUST
be indicated in the Map field.
This block provides information on transient IP problems and is
able to represent the combined effect of packet loss and packet
discard. Burst/Gap metrics are typically used in Cumulative reports
however MAY be used in Interval reports.
The definition of Burst and Gap is consistent with that defined in
the RFC3611 VoIP Metrics block, with the clarification that Loss
and Discard are defined in terms of frames (as described in 3.3
above). To accomodate the range of jitter buffer algorithms and
packet discard logic that may be used by implementors, the method
used to distinguish between bursts and gaps may be an equivalent
method to that defined in RFC3611. The method used SHOULD produce
the same result as that defined in RFC3611 for conditions of burst
packet loss, but MAY produce different results for conditions of
time varying jitter.
If Voice Activity Detection is used the Burst and Gap Duration
shall be determined as if silence frames had been sent, i.e. a
period of silence in excess of Gmin frames MUST terminate a burst
condition.
The Burst/Gap Metrics sub-block contains the following elements.
3.4.1 Threshold
The Threshold is equivalent to Gmin in RFC3611, i.e. the number of
successive frames that must be received and not discarded prior to
and following a lost or discarded frame in order for this lost or
discarded frame to be regarded as part of a gap.
3.4.2 Burst Duration (ms)
The average duration of a burst of lost and discarded frames.
3.4.3 Gap Duration (ms)
The average duration of periods between bursts.
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3.4.4 Burst Loss/Discard Rate
The proportion of Lost and Discarded frames during Bursts expressed
as a binary fraction expressed in 0:16 format.
3.4.5 Gap Loss/Discard Rate
The proportion of Lost and Discarded frames during Gaps expressed
as a binary fraction expressed in 0:16 format.
3.5 Playout Metrics sub-block
The Playout Duration metrics sub-block MAY be present and
if present MUST be indicated in the Map field.
At any instant, the audio output at a receiver may be classified
as either 'normal' or 'concealed'. 'Normal' refers to playout of
audio payload received from the remote end, and also includes locally
generated signals such as announcements, tones and comfort noise.
Concealment refers to playout of locally-generated signals used to
mask the impact of network impairments or to reduce the audibility
of jitter buffer adaptations.
This sub-block accounts for the source of the output audio, in
millisecond units. The on-time and active speech playout durations
allow calculation of the voice activity fraction. The on-time, and
concealment durations allow calculation of concealment ratios. This
sub-block distinguishes between reactive (due to effective packet
loss) and proactive (due to buffer adaptation) concealment.
3.5.1 On-time Playout Duration
'On-time' playout is the uninterrupted, in-sequence playout of valid
decoded audio information originating from the remote endpoint.
This includes comfort noise during periods of remote talker silence,
if VAD is used, and locally generated or regenerated tones and
announcements.
An equivalent definition is that on-time playout is playout of any
signal other than those used for concealment.
On-time playout duration MUST include both speech and silence
intervals, whether VAD is used or not. This duration is reported
in millisecond units.
3.5.2 On-time Active Speech Playout Duration (optional)
The duration, in milliseconds, of the on-time playout duration
corresponding to playout of active speech signals, if known. If
not known, then this field is set to all ones (0x FFFF FFFF).
In the absence of silence suppression, on-time active speech playout
equals on-time playout (section 3.5.1).
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3.5.3 Loss Concealment Duration
The duration, in milliseconds, of audio playout corresponding to
Loss-type concealment.
Loss-type concealment is reactive insertion or deletion of samples
in the audio playout stream due to effective frame loss at the audio
decoder. "Effective frame loss" is the event in which a frame of
coded audio is simply not present at the audio decoder when
required. In this case, substitute audio samples are generally
formed, at the decoder or elsewhere, to reduce audible impairment.
Only loss-type concealment is necessary to form Concealed and
Severely Concealed Seconds counts, in Section 3.6.
3.5.4 Buffer Adjustment Concealment Duration (optional)
The duration, in milliseconds, of audio playout corresponding to
Buffer Adjustment-type concealment, if known. If not known, then
this field is set to all ones (0x FFFF FFFF).
Buffer Adjustment-type concealment is proactive or controlled
insertion or deletion of samples in the audio playout stream due to
jitter buffer adaptation, re-sizing or re-centering decisions within
the endpoint.
Because this insertion is controlled, rather than occurring randomly
in response to losses, it is typically less audible than loss-type
concealment (section 3.5.3). For example, jitter buffer adaptation
events may be constrained to occur during periods of talker silence,
in which case only silence duration is affected, or sophisticated
time-stretching methods for insertion/deletion during favorable
periods in active speech may be employed. For these reasons, buffer
adjustment-type concealment MAY be exempted from inclusion in
calculations of Concealed Seconds and Severely Concealed Seconds.
However, an implementation SHOULD include buffer-type concealment in
counts of Concealed Seconds and Severely Concealed Seconds if the
event occurs at an 'inopportune' moment, with an emergency or large,
immediate adaptation during active speech, or for unsophisticated
adaptation during speech without regard for the underlying signal, in
which cases the assumption of low-audibility cannot hold. In other
words, jitter buffer adaptation events which may be presumed to be
audible SHOULD be included in Concealed Seconds and Severely
Concealed Seconds counts.
Concealment events which cannot be classified as Buffer Adjustment-
type MUST be classified as Loss-type.
3.6 Concealed Seconds metrics sub-block
The Concealed Seconds metrics sub-block MAY be present and if
present MUST be indicated in the Map field.
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This sub-block provides a description of potentially audible
impairments due to lost and discarded packets at the endpoint,
expressed on a time basis analogous to a traditional PSTN T1/E1
errored seconds metric.
The following metrics are based on successive one second intervals as
declared by a local clock. This local clock does NOT need to be
synchronized to any external time reference. The starting time of
this clock is unspecified. Note that this implies that the same loss
pattern could result in slightly different count values, depending on
where the losses occur relative to the particular one-second
demarcation points. For example, two loss events occurring 50ms apart
could result in either one concealed second or two, depending on the
particular 1000 ms boundaries used.
The seconds in this sub-block are not necessarily calendar seconds.
At the tail end of a call, periods of time of less than 1000ms shall
be incorporated into these counts if they exceed 500mS and shall be
disregarded if they are less than 500mS.
3.6.1 Unimpaired Seconds
A count of the number of unimpaired Seconds that have occurred on
this call.
An unimpaired Second is defined as a continuous period of
1000ms during which no frame loss or discard due to late arrival has
occurred. Every second in a call must be classified as either
OK or Concealed.
Normal playout of comfort noise or other silence concealment
signal during periods of talker silence, if VAD is used, shall be
counted as unimpaired seconds.
3.6.2 Concealed Seconds
A count of the number of Concealed Seconds that have occurred on
this call.
A Concealed Second is defined as a continuous period of 1000ms
during which any frame loss or discard due to late arrival has
occurred.
Equivalently, a concealed second is one in which
some Loss-type concealment (defined in section 3.6) has occurred.
Buffer adjustment-type concealment SHALL not cause Concealed
Seconds to be incremented, with the following exception. An
implementation MAY cause Concealed Seconds to be incremented for
'emergency' buffer adjustments made during talkspurts.
For clarification, the count of Concealed Seconds MUST
include the count of Severely Concealed Seconds.
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3.6.3 Severely Concealed Seconds
A count of the number of Severely Concealed Seconds that have
occurred on this call.
A Severely Concealed Second is defined as a non-overlapping period
of 1000 ms during which the cumulative amount of time that has been
subject to frame loss or discard due to late arrival, exceeds the
SCS Threshold.
3.6.4 SCS Threshold
The SCS Threshold defines the amount of time corresponding to lost
or discarded frames that must occur within a one second period in
order for the second to be classified as a Severely Concealed
Second. This is expressed in milliseconds and hence can represent
a range of 0.1 to 25.5 percent loss/ discard.
A default threshold of 50ms (5% effective frame loss per second)
is suggested.
3.7 Delay and Packet Delay Variation (PDV) metrics sub-block
The Delay and PDV metrics sub-block MUST be present. This sub-block
contains a number of parameters related to overall delay (latency),
delay variation and the current jitter buffer configuration.
[Authors' note - need to add high and low water marks]
3.7.1 Network Round Trip Delay (ms)
The Network Round Trip Delay is the most recently measured value
of the RTP-to-RTP interface round trip delay, typically determined
using RTCP SR/RR. If no measured delay is available then this
field shall be set to 0xFFFF
[Note - potentially add one way delay]
3.7.2 End System Delay (ms)
The End System Delay is the internal round trip delay within the
endpoint, calculated using the nominal value of the jitter buffer
delay plus the accumulation/ encoding and decoding / playout delay
associated with the codec being used.
3.7.3 External Network Delay (ms)
The External Network Delay parameter indicates external network
round trip delay through cellular, satellite or other types of
network with significant delay impact, if known. A value of
0xFFFF shall indicate that the delay is unknown.
If the external network is VoIP based then this parameter is
typically determined using RTCP SR/RR. If the external network delay
is known and does not vary materially then this value may be
provisioned.
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3.7.4 PDV/ Jitter Metrics
Jitter metrics defined are:
(i) Mean PDV - for cumulative reports this is a running average of
PDV and for interval reports this is an interval average
(16 bit, S11:4 format) expressed in milliseconds
(ii) Threshold PDV - the PDV associated with the PDV percentile
(16 bit, S11:4 format) expressed in milliseconds, both positive
and negative thresholds are given
(iii)PDV Percentile - the percentage of packets on the call for
which individual packet delays were outside the Threshold PDV,
expressed in 8:8 format. Both positive and negative percentiles
are given.
3.7.5 PDV Type
Indicates the type of algorithm used to calculate PDV:
PPDV (0) according to RFC3550,
MAPDV (1) according to ITU-T G.1020,
IPDV (2) according to ITU-T Y.1540
Other values reserved
For example:-
(a) To report PPDV (RFC3550):
Mean PDV = PPDV
Threshold PDV = Undefined (FF FF)
PDV Percentile = Undefined (FFF)
PDV type = 0 PPDV
(b) To report 95th percentile MAPDV (G.1020):
Mean PDV = average MAPDV
Pos Threshold PDV = 50.0
Pos PDV Percentile = 95.3
Neg Threshold PDV = 50.0 (note - implies -50ms)
Neg PDV Percentile = 98.4
PDV type = 1 MAPDV
Note that implementations may either fix the reported percentile
and calculate the associated PDV level OR may fix a threshold PDV
level and calculate the associated percentile. From a practical
implementation perspective it is simpler to use the second of these
approaches.
3.7.6 Jitter Buffer and PLC Configuration
Indicates the configuration of the jitter buffer and the type of PLC
algorithm in use.
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bits 0-3
0 = silence insertion
1 = simple replay, no attenuation
2 = simple replay, with attenuation
3 = enhanced
? = ?
Other values reserved
bits 4-7
0 = Fixed jitter buffer
1 = Adaptive jitter buffer
Other values reserved
3.7.7 Jitter Buffer Size parameters
Current nominal, maximum and absolute maximum jitter buffer size
expressed in milliseconds, as defined in RFC3611.
3.8 Call Quality Metrics sub-block
The Call Quality Metrics sub-block MAY be present and if present
MUST be indicated in the Header Map field. This sub-block reports
call quality metrics and estimates of signal, noise and echo levels.
Signal, noise and echo metrics should be long term averages and
should not be instantaneous values.
3.8.1 Listening and Conversation Quality R Factors
Expresses listening and conversational quality in terms of R factor,
a 0-120 scaled parameter in 8:8 format. The algorithm used to
calculate R factor MAY be defined in the Voice/Data Quality Metric
Algorithm block.
3.8.2 Listening and Conversation Quality MOS Scores
Expresses listening and conversational quality in terms of MOS,
a 1-5 scaled parameter in 8:8 format. The algorithm used to
calculate MOS MAY be defined in the Voice/Data Quality Metric
Algorithm block (see Section 4).
Note that R factors and MOS scores may be defined for both narrow
and wide-band VoIP calls. R Factors are continuous for narrow and
wideband, hence the R factor for a wideband call may be higher than
that for a narrowband call. MOS scores are scaled relative to
reference conditions and hence both narrow and wideband MOS occupy
the same 1-5 scale; this can lead to a wideband MOS being lower than
a narrowband MOS even though the listening quality may be higher.
3.8.3 R-LQ External In and Out.
R-LQ External In - measured by this endpoint for incoming
connection on "other" side of this endpoint
R-LQ External Out - copied from RTCP XR message received from
remote endpoint on "other" side of this endpoint
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e.g. Phone A <---> Bridge <----> Phone B
In XR message from Bridge to Phone A:-
- R-LQ = quality for PhoneA ----> Bridge path
- R-LQ-ExtIn = quality for Bridge <---- Phone B path
- R-LQ-ExtOut = quality for Bridge -----> Phone B path
This allows PhoneA to assess
(i) received quality from the combination of
R-LQ measured at A
and
R-LQ-ExtIn reported by the Bridge to A
(ii) remote endpoint quality from the combination of
R-LQ reported by the Bridge
and
R-LQ-ExtOut reported by the Bridge
3.8.4 RFC3550 RTP Payload Type
The RTP Payload type field - as per RFC3551 and
http://www.iana.org/assignments/rtp-parameters
If the RFC3550 Payload type indicates a dynamic payload then
it is strongly recommended that the Extended Payload Descriptor
sub-block (Section 4.4) be sent in order that mid-stream management
systems can identify the codec type used.
3.8.5 Media Type
Media type -
0 = No media present
1 = Narrowband audio
2 = Wideband audio
3.8.6 Received Signal and Noise Levels - IP side
The received signal level during talkspurts and the noise level
expressed in dBm0, for the decoded packet stream.
3.8.7 Received Signal and Noise Levels - External
The received signal level during talkspurts and the noise level
expressed in dBm0, for the PCM side of a gateway, audio input
from a handset or decoded packet stream for an IP-to-IP gateway.
3.8.8 Local and Remote Residual Echo Return Loss
The Local and Remote Residual Echo Return Loss (RERL) expressed in
dB. The Local RERL is the echo level that would be reflected into the
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IP path due to line echo on the circuit switched element side of this
IP endpoint if a gateway or acoustic echo if a handset or wireless
terminal.
The Remote RERL is the echo level that would be reflected into the
remote IP endpoint from the network "behind" it, and would typically
be measured at and reported from the remote endpoint. This value is
included as it may be used in calculating the R-CQ and MOS-CQ values
expressed in this report block.
3.8.9 Metric Status
Indicates the source of parameter values used in call quality
calculation:
Bit Description Source
0-1 Local IP side Signal/Noise Levels measured on the incoming
decoded VoIP stream to this endpoint
00 = assumed
01 = measured for this call
10 = measured across multiple calls on this port
11 = measured across multiple ports
2-3 Remote IP side Signal/Noise Levels reported by the remote
IP endpoint through RTCP XR or equivalent
00 = assumed
01 = measured for this call
10 = measured across multiple calls on this port
11 = measured across multiple ports
4-5 Local Trunk side Signal/Noise Levels measured on the incoming
PCM, Audio or non-IP side of this endpoint
00 = assumed
01 = measured for this call
10 = measured across multiple calls on this port
11 = measured across multiple ports
6-7 Local Echo level measured in the incoming line/ trunk/
handset direction at this endpoint after the effects of echo
cancellation
00 = assumed
01 = measured for this call
10 = measured across multiple calls on this port
11 = measured across multiple ports
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8 Remote Echo level measured in the incoming line/ trunk/
handset direction at the remote endpoint after the effects of
echo cancellation and reported to this endpoint via RTCP XR
or equivalent.
0 = assumed
1 = reported from remote endpoint
9-15 Reserved
For example, if this endpoint is "C" in the diagram below then the
following definitions would apply.
Endpoint B -----RTP-------> Gateway C <-----PCM-------> D
"Remote" "Local" "Trunk/PCM/External"
Reporting endpoint is "B"
Local IP side signal/noise metrics relate to signal/noise levels
from decoded RTP packets received by C from B
Remote IP side signal/noise metrics relate to signal/noise levels
from decoded RTP packets received by B from C, and reported by B
to C through RTCP XR or RTCP HR VoIP Metrics blocks
Local Trunk side signal/noise metrics relate to signal/noise
levels from the PCM signal received by C from D
Local Echo level relates to the proportion of the signal passing
from B to C to D that is reflected back to C at some point
between C and D or on the far side of D. This would typically be
electrical echo or acoustic echo.
Remote Echo level relates to the proportion of the signal passing
from D to C to B that is reflected back to B at some point
between B and the user. This echo level is typically measured at
B and reported to C via RTCP XR or RTCP HR VoIP Metrics blocks.
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3.9 Vendor Extension sub-block
One or more Vendor Extension sub-blocks MAY be present. Their
presence MUST be indicated in the Header Map field. Note that the map
field does not indicate the number of vendor extension sub-blocks.
This must be deduced from the length of the overall report block and
the lengths of the Vendor Extension sub-blocks.
Each vendor extension sub-block consists of an extension header and
vendor-specific extension data. The extension header has the form
Vendor ID, Vendor ID Source and Extension Block Length. The Extension
Block Length is defined as including these extension header and
extension data octets but does not include any subsequent vendor
extension sub-blocks. An implementation can skip over a
vendor extension sub-block that it does not understand.
The Vendor ID Source field indicates whether the four-octet Vendor ID
is based on ITU T.35 or is an IANA private enterprise number. The
designated values for these options are 1 and 2 respectively. If the
Vendor ID Source field is assigned a value of 0, then all of the
fields in the vendor extension sub-block with the exception of the
Vendor ID Source field and the Extension Block Length field have
proprietary definitions.
If the Vendor ID is based on ITU-T Recommendation T.35, its first two
octets either contain a country code from Annex A of ITU-T Rec. T.35
followed by 0x00, or an escape code of 0xFF followed by a country
code from Annex B of ITU-T Rec. T.35. The next two octets comprise a
terminal provider code allocated by a national assignment authority
(http://people.itu.int/~campos/t35/t35db.htm). This field is padded
with leading zeros if necessary. If a vendor has multiple
terminal provider codes in different registries (e.g. the H-series,
T-series and V-series registries for the USA), then this provenance
shall be indicated in the Manufacturer Code Source field. Possible
values of the Manufacturer Code Source field are:
0 Unspecified/don't care (may be used if there is no conflict)
1 G-series registry
2 H-series registry
3 T-series registry
4 V-series registry
5-255 Reserved.
If the four-octet Vendor ID is an IANA private enterprise number,
then it is padded with leading zeros as necessary. Current private
enterprise numbers (www.iana.org/assignments/enterprise-numbers)
can be accommodated within two octets. The additional two octets
provide for future growth.
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4. RTCP HR Configuration Block
This block type provides a flexible means to describe the algorithms
used for call quality calculation and other data. This block need
only be exchanged occasionally, for example sent once at the start
of a call.
Header sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=N | Map | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Correlation Tag sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Tag Type | Tag length | Correlation Tag... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... Correlation Tag |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Algorithm sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Alg type | Descriptor len| Algorithm descriptor... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| ... Algorithm descriptor |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Vendor Specific Extensions sub-block
0 1 2 3
0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7 0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Vendor ID |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Vendor ID Src |Manuf Code Src | Extension Block Length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Vendor-specific extension data |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
4.1 Header
Implementations MUST send the Header block within each RTCP HR
Configuration report.
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4.2.1 Block type
One RTCP HR Configuration block is defined
mmm+12 = RTCP HR Configuration Block
The time interval associated with these report blocks is left to the
implementation. Spacing of RTCP reports should be in accordance
with RFC3550 however the specific timing of RTCP HR reports may
be determined in response to an internally derived alert such as a
threshold crossing.
4.2.2 Map field
An Map field indicates the optional sub-blocks present in this
report. A '1' indicates that the sub-block is present, and a '0' that
the block is absent. If present, the sub-blocks must be in the
sequence defined in this document.
The bits have the following definitions:
0 Correlation Tag
1 Algorithm Descriptor 1
2 Algorithm Descriptor 2
3 Algorithm Descriptor 3
4 Algorithm Descriptor 4
5 Algorithm Descriptor 5
6 Algorithm Descriptor 6
7 Algorithm Descriptor 7
8 Algorithm Descriptor 8
9 Vendor Specific Extension
10-15 Reserved, set to '0'
4.2.3 Block Length
The block length indicates the length of this report in 32 bit
words and includes the header and any extension octets.
4.2.4 SSRC
The SSRC of the stream to which this report relates.
4.3 Correlation Tag
The Correlation Tag sub-block MAY be present and if present
MUST be indicated in the map field. This tag facilitates the
correlation of the high resolution VoIP metrics report blocks
with other call-related data, session-related data or endpoint data.
An example use case is for an endpoint may convey its version of a
call identifier or a global call identifier via this tag. A
flow measurement tool (sniffer) that is not call-aware can then
forward the RTCP-HR reports along with this correlation tag to
network management. Network management can then use this tag to
correlate this report with other diagnostic information such as
call detail records.
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The Tag Type indicates the use of the correlation tag. The following
values are defined:
0: IMS Charging Identity (ICID) subfield of the
P-Charging-Vector header specified in RFC 3455 [1].
1: Globally unique ID as specified in ITU-T H.225.0
(Table 20/H.225.0) [2].
2: Conference Identifier, per ITU-T H.225.0
(Table 20/H.225.0) [2].
3: SIP Call-ID as defined in RFC 3261 [3].
4: PacketCable Billing Call ID (BCID) [4].
5: Text string using the US-ASCII character set [5].
6: Octet sting.
7-255: Future growth.
Although the intent of this RFC is to list all currently known
values of usable correlation tags, it is possible that new values
may be defined in the future. An IANA registry of correlation
tags is recommended.
The tag length indicates the overall length of the sub-block in
32 bit words and includes the tag type and length fields.
4.4 Algorithm descriptor
The Algorithm Type sub-block MAY be present and if present
MUST be indicated in the map field
The Algorithm Type is a bit field which indicates which
algorithm is being described. The bits are defined as:-
Bit 0: MOS-LQ Algorithm
Bit 1: MOS-CQ Algorithm
Bit 2: R-LQ Algorithm
Bit 3: R-CQ Algorithm
Bit 4-7: Reserved and set to '0'
The descriptor length gives the overall length of the descriptor in
32 bit words and includes the algorithm descriptor and length fields.
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The algorithm descriptor is a text field that contains the
description or name of the algorithm. If the algorithm name is
shorter than the length of the field then the trailing octets
must be set to 0x00.
For example, an implementation may report:
Algorithm descriptor = 0xF0 - R and MOS algorithms
Descriptor length = 3 - 3 words
Descriptor = "Alg X" 0x00 - description
Call quality estimation algorithms may be defined for listening or
conversational quality MOS or R factor.
4.4 Vendor Extension field
One or more vendor specific extension blocks may be added, as defined
in Section 3.10
5. Application of RTCP HR to RTP translators
RFC3550 [12] defines three types of RTP systems which may source and
sink RTP streams. These types are RTP end systems, RTP mixers, and
RTP translators. For purposes of reporting connection quality to
other RTP systems, RTP mixers and RTP end systems are very similar.
Mixers resynchronize audio packets and do not relay RTCP reports
received from one cloud towards other cloud(s). Translators do not
resynchronize packets and SHOULD forward certain RTCP reports
between clouds. Translators have a wide range of possible reporting
behaviours. This section describes in more detail the application of
RTCP HR to RTP translators.
An RTP translator is defined in RFC3550 [12] as "An intermediate
system that forwards RTP packets with their synchronization source
identifier intact". RFC3550 gives the following examples of
translators: devices that convert encodings without mixing;
replicators from multicast to unicast; and application-level filters
in firewalls.
An RTP translator has at least two RTP connections via different
logical interfaces to network clouds, with logical separation based
on a transport layer identifier (e.g. by UDP port) or at a lower
layer (e.g. IP address, Ethernet VLAN identifier, ATM VCI, or
physical interface). RTP traffic streams flowing via these two
connections may be unicast or multicast, and may be unidirectional
or bidirectional. Section 7 of RFC3550 [12] defines the RTCP
processing in the translator which is required to maintain correct
semantics for the RTCP communication between the RTP end systems
involved in the connection.
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Where the RTP translator makes its own measurements of the incoming
packet stream, section 7 of RFC3550 [12] allows it to send RTCP
reception reports about what it has received. To do this, it is
allowed to allocate an SSRC and a CNAME for itself.
An RTP translator has several potential sources of information about
transport and application quality in a session. These include
o RTCP (including XR and HR) reports received from RTP end
systems and mixers
o RTCP (including XR and HR) reports received from other
translators
o Measurements made locally on incoming RTP streams at the
translators' interfaces
In general, it is a matter of policy whether each of these types of
information should be reported or forwarded by the translator towards
neighbouring RTP systems involved in the same session. The objective
of this policy may be to ensure that sufficient information is
available to support network and service management at RTP systems,
whilst avoiding excessive RTCP processing.
Where a translator forms a boundary between service providers'
domains, a provider may wish to apply policy to restrict the release
of network performance information towards the provider's peer. This
suggests that policy controlling reporting and forwarding of RTCP
information MAY be configurable differently for each network
interface. This allows certain types of RTCP reports for the session
to be sent outwards via one subset of the translator's interfaces
involved in the session, whilst restricting reporting and forwarding
outwards via another subset.
<Authors' note: Need to review need for an RTP system to echo
reception or transmission metrics from any RTCP report back towards
the RTP system which originated it. This behaviour increases RTCP
traffic without increasing the information available at any RTP
system - however DOES make information available to routers along
the paths between RTP systems, which may be asymmetric.>
The following behaviours are potentially useful:
o RTCP [12] reports received from an upstream RTP end system
or mixer SHOULD be forwarded towards the downstream RTP end
systems or mixers involved in the session. This behaviour is
RECOMMENDED to maintain the integrity of the RTCP connection
between the RTP end systems involved in the connection. We call
this behaviour "a" to aid discussion below of the examples of
end-to-end capabilities.
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o RTCP XR and RTCP HR reports received from an upstream RTP end
system or mixer MAY be forwarded towards the downstream RTP
end systems or mixers involved in the session. We call this
behaviour "b".
o The results of measurements made on an RTP stream received at
one of the translator's network interfaces MAY be sent out
as RTCP reports towards neighbouring RTP systems, either
upstream in the direction towards the RTP system which
originated the RTP stream (behaviour "c1"), or downstream in the
direction towards the RTP system(s) which will receive the
translated RTP stream (behaviour "c2"), or both upstream and
downstream (described as behaviour "c1+c2").
o RTCP XR and RTCP HR reports received from an upstream RTP
translator MAY be forwarded towards the downstream RTP
end systems or mixers involved in the session. We call this
behaviour "d".
In the next section we illustrate how policies may be applied to
control these behaviours and hence achieve three end-to-end
reporting modes "end system peering", "local reporting" and "global
reporting" for a hypothetical bidirectional unicast connection
using translators. These three modes provide different compromises
between the amount of information available to the RTP systems, and
the amount of RTCP traffic and processing. Other modes are possible,
for example where policy restricts forwarding of reports on network
performance to another operator, but these are out of the scope of
this document.
To help discussion of these example modes, we name the translator's
left-side connections to the first cloud as 1T and 1R, where 1T is
the connection on which the translator transmits packets to the
first cloud and 1R is the connection on which the translator
receives packets from the first cloud. Similarly the translator's
right-side connections to the second cloud are named 2T and 2R.
5.1 Examples of measurement modes involving translators
The three modes are described with reference to the following
example, a bidirectional unicast connection which includes two RTP
end systems A and Z connected through 3 translators J, K and L:
---- ----- ----- ----- ----
| T|-->|1R 2T|-->|1R 2T|-->|1R 2T|-->|R |
| A | | J | | K | | L | | Z |
| R|<--|1T 2R|<--|1T 2R|<--|1T 2R|<--|T |
---- ----- ----- ----- ----
The notation X.nR is used below for the receive interface of RTP
system X facing a cloud identified by an index n, and similarly the
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notation X.nT is used for X's transmit interface facing the same
cloud. The index n is local to the RTP system, that is, a cloud
identified as "1" by one translator may be identified as "2" by a
neighbouring translator, including one connected to the same cloud.
Where an RTP system is connected to only one cloud (as for RTP end
-systems), the index n is dropped.
5.1.1 End system peering mode
In the first mode, called "end system peering", a translator
forwards RTCP reports generated by end systems but does not send any
report of its own measurements (if any) to peer RTP/RTCP devices.
RTCP reports received at 1R are sent out from 2T, and reports
received at 2R are sent out from 1T.
This end-to-end mode is achieved by policy to permit behaviours "a"
and "b" at all translators involved in the connection.
Note that translators operating in "end system peering" mode MAY
make measurements on incoming RTP streams, and MAY report the
measurements via "northbound" interfaces to management or session
control systems, but such measurement and reporting is out of scope
of this document.
The following illustrates "end system peering" using the example
connection. In "end system peering" mode, no measurements made at
translators are sent in RTCP. Measurements made on the RTP stream
received at end-system A's interface A.R are sent in RTCP from A.T,
received at J.1R and relayed from J.2T. Following this pattern they
are then successively received at K.1R and relayed from K.2T,
received at L.1R and relayed from L.2T, and finally received at A's
peer, the end-system Z, on interface Z.R. In the reverse direction
(assuming a unicast bidirectional connection), measurements made on
the RTP stream received at interface Z.R are sent in RTCP from Z.T,
received at L.2R and relayed from L.1T, and follow this pattern
until they are finally relayed from J.1T and received at Z's peer,
the end-system A, on interface A.R.
A translator may identify RTCP packets sent by an end system because
they have the same SSRC as the RTP packets received at the same
interface. Hence a translator behaving according to "a", or "a" and
"b", (thus operating in "end system peering" mode) MAY police the
mode (reducing the load on downstream systems) by discarding RTCP
packets which do not have the same SSRC as the RTP traffic.
The advantage of "end system peering" mode is that it minimises
processing in translators whilst allowing compliance to section 7.2
of RFC3550 [12]. The amount of RTCP traffic is not increased by the
introduction of translators into the connection.
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The disadvantage of the "end system peering" mode is that RTP
systems making up the connection are unable to identify the network
segment in which an impairment occurred, because the RTP translators
which form the boundaries between networks do not report
measurements of impairments to their peer RTP systems.
Note that in order to be compliant to [12], any processing of
RTP packets by the translator must be accompanied by appropriate
modification of RTCP fields such that the RTCP information remains
consistent with the modified RTP traffic and will make sense to the
RTP end-system which receives the translated stream. Hence a
translator's forwarding of RTCP packets may include modification of
the RTCP.
5.1.2 Local Reporting mode
In the second mode, called "local reporting", a translator forwards
RTCP reports generated by end systems, and also sends reports of its
own measurements for both directions of transmission to both
neighbouring RTP/RTCP devices (which may also be translators, but
may instead be mixers or end systems). However a translator does not
forward reports received from an RTP device which is not an end
system or mixer. Specifically, RTCP reports originated by an RTP
end system or mixer, and received at 1R, are sent out from 2T. RTCP
reports originated by an RTP end system or mixer, and received at
2R, are sent out from 1T. In addition, reports based on measurements
made by the translator itself on the packet stream incoming at 1R
are sent out via both 1T and 2T. Similarly, reports based on
measurements made by the translator itself on the packet stream
incoming at 2R are sent out via both 1T and 2T. However, RTCP
reports originated by a neighbouring translator and received at 1R
are not forwarded either to 1T or to 2T, and similarly RTCP reports
originated by a neighbouring translator and received at 2R are not
forwarded either to 1T or to 2T. In "local reporting" mode, a
translator must have a CNAME and must allocate an SSRC with which to
label reports containing its own measurements.
This end-to-end mode is achieved by policy to permit behaviours "a",
"b", "c1", and "c2" at all translators involved in the connection.
For "local reporting" mode, it is necessary to determine whether an
RTCP report originated at an end-system or at a translator. This can
be achieved by comparing the SSRC of the RTCP report with the SSRC
of the RTP packets comprising the traffic stream. If the SSRC of the
RTCP report is the same as that of the RTP packets, the RTCP report
originated from an end system; otherwise, the RTCP report originated
at a translator and should not be forwarded further in RTCP.
The following illustrates "local reporting" using the example
connection. In "local reporting" mode, measurements made at
end-systems are forwarded along the whole chain, as for "end system
peering". In addition, measurements made at an interface of a
translator are sent to both the translator's neighbour RTP systems.
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However, measurements made by a translator and received at another
RTP system are not forwarded further in RTCP. In detail,
measurements made on the RTP stream received at end-system A's
interface A.R are sent in RTCP from A.T, received at J.1R and
relayed from J.2T as for "end system peering" mode. However for
"local reporting" mode, measurements made at J.1R and J.2R are also
sent out from J.2T and received at K.1R. Hence K receives (at K.1R)
measurements made at all of A.R, J.1R, and J.2R. Of these,
translator K forwards measurements from A.R out via K.2T. Translator
K also sends its own measurements made at K.1R and K.2R out via
K.2T, such that L receives (at L.1R) measurements made at all of
A.R, K.1R and K.2R. Finally similar processing occurs in L such that
Z receives (at Z.R) measurements made at A.R, L.1R and L.2R.
In the reverse direction, measurements made on the RTP stream
received at end-system Z's interface Z.R are sent in RTCP from Z.T,
received at L.2R and relayed from L.1T as for "end system peering"
mode. However for "local reporting" mode, measurements made at L.1R
and L.2R are also sent out from L.1T and received at K.2R. Hence K
receives (at K.2R) measurements made at all of Z.R, L.1R, and L.2R.
Of these, translator K forwards measurements from Z.R out via K.1T.
Translator K also sends its own measurements made at K.1R and K.2R
out via K.1T, such that J receives (at J.2R) measurements made at
all of Z.R, K.1R and K.2R. Finally similar processing occurs in J
such that A receives (at A.R) measurements made at Z.R, J.1R and
J.2R.
The first advantage of "local reporting" mode is that it permits a
translator to learn via RTCP the results of measurements made at
both its neighbours. In addition, it may observe RTCP HR reports of
the results of measurements made at the terminating end system.
From this information, a translator has the information which it
requires to localise a single impairment to one of the locations
"left of the cloud to my left", "in the cloud to my left", "in the
cloud to my right", or "right of the cloud to my right".
For example, K sees reports from J.1R and L.1R, and also has its own
measurements made at K.1R. Finally it may observe RTCP reports of
Z's measurements made at Z.R. Suppose a transport impairment is
detected at Z.R. If a related impairment is detected at L.1R but not
at K.1R, K may deduce that there is probably a fault in the network
cloud between K and L.
The second advantage of the "local reporting" mode (relative to the
"global reporting" mode) is that the amount of RTCP traffic on any
link is independent of the number of translators involved in the
connection. The total amount of RTCP processing is proportional to
the number of translators but this processing is distributed among
the translators. Hence the mode scales to large numbers of
translators, for unicast connections.
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5.1.3 Global Reporting mode
In the third mode, called "global reporting", measurements made at
end-systems are forwarded along the whole chain, as for "end system
peering". In addition, measurements made at an interface of a
translator are sent to both the translator's neighbour RTP systems.
Measurements made by a translator and received at another translator
are also forwarded further in RTCP, but only in the direction away
from the source of the measurement - that is, information received
over RTCP from one cloud is sent to the other but is not copied back
to any RTP system in the cloud which sent the measurement. In
"global reporting" mode, a translator must have a CNAME and must
allocate an SSRC with which to label reports containing its own
measurements. Specifically, RTCP reports originated by an RTP end
system or mixer, and received at 1R, are sent out from 2T. RTCP
reports originated by an RTP end system or mixer, and received at
2R, are sent out from 1T. In addition, reports based on measurements
made by the translator itself on the packet stream incoming at 1R
and 2R are sent out via both 1T and 2T. In addition, RTCP reports
either originated or forwarded by a neighbouring translator and
received at 1R are forwarded out via 2T, and similarly RTCP reports
either originated or forwarded by a neighbouring translator and
received at 2R are forwarded out via 1T.
This end-to-end mode is achieved by policy to permit behaviours "a",
"b", "c1", "c2", and "d" at all translators involved in the
connection.
The following illustrates "global reporting" using the example
connection. Measurements made on the RTP stream received at
end-system A's interface A.R are sent in RTCP from A.T, received at
J.1R and relayed from J.2T as for "end system peering" mode. However
for "global reporting" mode, measurements made at J.1R and J.2R are
also sent out from J.2T and received at K.1R. Hence K receives (at
K.1R) measurements made at all of A.R, J.1R, and J.2R. Translator K
forwards all of these measurements out via K.2T. Translator K also
sends its own measurements made at K.1R and K.2R out via K.2T, such
that L receives (at L.1R) measurements made at all of A.R, J.1R,
J.2R, K.1R and K.2R. Finally similar processing occurs in L such
that Z receives (at Z.R) measurements made at A.R, J.1R, J.2R, K.1R,
K.2R, L.1R and L.2R.
In the reverse direction, measurements made on the RTP stream
received at end-system Z's interface Z.R are sent in RTCP from Z.T,
received at L.2R and relayed from L.1T as for "end system peering"
mode. However for "global reporting" mode, measurements made at
L.1R and L.2R are also sent out from L.1T and received at K.2R.
Hence K receives (at K.2R) measurements made at all of Z.R, L.1R,
and L.2R. Translator K forwards all of these measurements out via
K.1T. Translator K also sends its own measurements made at K.1R and
K.2R out via K.1T, such that J receives (at J.2R) measurements made
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at all of Z.R, L.1R, L.2R, K.1R and K.2R. Finally similar processing
occurs in J such that A receives (at A.R) measurements made at Z.R,
L.1R, L.2R, K.1R, K.2R, J.1R and J.2R.
In "global reporting" mode, all RTCP packets received at an incoming
interface should be assumed to be either from an end system or from
a translator on the path from the end system, and should be
forwarded.
The advantage of the "global reporting" mode is that every RTP
system (translators and end-systems) has access to all the
information to allow localisation of impairments to the faulty
cloud.
The disadvantage of the "global reporting" mode is that the amount
of RTCP traffic on every link is proportional to the number of
translators in the connection, and the total amount of RTCP
processing, though distributed among translators, is proportional to
the square of the number of translators. Hence the "global
reporting" mode does not scale well for connections which may
involve large numbers of translators.
5.1.4 Summary of operating modes
End System Peering - Measurements are made on received RTP streams
at RTP endpoints and relayed via translators. Any measurements made
by translators are not forwarded but MAY be sent "northbound" (out
of scope of this document).
Advantages - Minimises translator processing. RTCP traffic is
not increased by additional translators.
Disadvantages - RTP systems are unable to identify where
impairments are occurring.
Local mode - Translators forward end system reports (End System
Peering), and also send reports of their own measurements to their
neighbours. They do not forward reports from other translators or
mixers.
Advantages - Allows the translator to learn the results of
neighbour measurements via RTCP. RTCP traffic is independent
of the number of translators.
RTCP traffic processing is proportional to the number of
translators, but processing is distributed, so this mode scales
better than Global mode.
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Disadvantages - Not all RTP systems have access to all the
information, so this mode is less powerful than Global mode
for impairment localisation.
Global mode Translators forward both end system reports (End System
Peering) and measurements sent from a neighbour translator, but only
in the direction away from the measurement source. Measurements
made at a translator interface are also sent to both the
translator's neighbour systems.
Advantages - Every RTP system (translators and end-systems) has
access to all the information to allow impairment localisation.
Disadvantages - RTCP traffic in every link is proportional to
the number of translators in the connection. The total RTCP
processing is proportional to the square of the number of
translators.
6. Practical Applications
6.1 Overview
The objective of this section is to identify a number of cases in
which there could potentially be some ambiguity in the application
of the report blocks defined above or some exceptions to the
defined operation of the metrics.
6.2 Call Hold and Transfer
6.3 VAD/Silence Elimination
6.4 Endpoint configuration changes mid-call
6.5 SSRC changes mid-call
7. Summary
8. IANA Considerations
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9. Security Considerations
RTCP reports can contain sensitive information since they can provide
information about the nature and duration of a session established
between two endpoints. As a result, any third party wishing to
obtain this information should be properly authenticated and the
information transferred securely.
10. Contributors
The authors gratefully acknowledge the comments and contributions
made by Jim Frauenthal, Mike Ramalho, Kevin Connor, Paul Jones,
Claus Dahm, Bob Biskner, Mohamed Mostafa, Tom Hock, Albert Higashi,
Shane Holthaus, Amit Arora, Bruce Adams, Geoff Hunt, Albrecht
Schwarz, Keith Lantz and Randy Ethier.
11. Informative References
[1] Bradner, S., "Key words for use in RFCs to Indicate Requirement
Levels", BCP 14, RFC 2119, June 1997.
[2] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", STD 64,
RFC 3550, July 2003.
[3] Friedman, T., Caceres, R. and A. Clark, "RTP Control Protocol
Extended Reports (RTCP XR)", RFC 3611, November 2003.
[4] "Performance parameter definitions for quality of speech and
other voiceband applications utilizing IP networks"
ITU-T Rec. G.1020, November 2003
[5] Annex A to G.1020: "VoIP Gateway specific reference points and
performance parameters", Amendment 1 to ITU-T Rec. G.1020
May 2004
[6] "Internet protocol data communication service - IP packet
transfer and availability performance parameters:,
ITU-T Rec. Y.1540, December 2002
[7] M. Garcia-Martin, E. Henrikson and D. Mills, "Private Header
(P-Header) Extensions to the Session Initiation Protocol
(SIP) for the 3rd-Generation Partnership Project (3GPP)",
RFC 3455, January 2003.
[8] ITU-T H.225.0, "Call signalling protocols and media stream
packetization for packet-based multimedia communication
systems", July 2003.
[9] J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston,
J. Peterson, J. Peterson, R. Sparks, M. Handley, E. Schooler,
"SIP: Session Initiation Protocol", June 2002.
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[10] PacketCable(TM) Multimedia Specification,
PKT-SP-MM-I02-040930, September 2004.
[11] Coded Character Set--7-Bit American Standard Code for
Information Interchange, ANSI X3.4-1986.
[12] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
"RTP: A Transport Protocol for Real-Time Applications", RFC
3550, July 2003.
[13] Friedman, T., Caceres, R., Clark, A., "RTP Control Protocol
Extended Reports (RTCP XR)", RFC 3611, November 2003
Authors' Addresses
Alan Clark
Telchemy Incorporated
2905 Premiere Parkway, Suite 280
Duluth, GA 30097
Email: alan@telchemy.com
Amy Pendleton
Nortel
2380 Performance Drive
Richardson, TX 75081
Email: aspen@nortel.com
Rajesh Kumar
Cisco Systems
170 West Tasman Drive
San Jose, CA 95134
Email: rkumar@cisco.com
Kevin Connor
Cisco Systems
5590 Whitehorn Way
Blaine, WA 98230
Email: kconnor@cisco.com
Geoff Hunt
BT
BT Adastral Park
Martlesham Heath
Ipswich IP5 3RE
UK
Email: geoff.hunt@bt.com
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Full Copyright Statement
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