Internet Engineering Task Force A. Clark
Internet-Draft Telchemy Incorporated
Expires: 4 January 2008 A. Pendleton
Nortel
R. Kumar
K. Connor
Cisco Systems
G. Hunt
BT
July 2007
RTCP HR - High Resolution VoIP Metrics Report Blocks
draft-ietf-avt-rtcphr-01.txt
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Copyright Notice
Copyright (C) The IETF Trust (2007).
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 [3].
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Definitions . . . . . . . . . . . . . . . . . . . . . . . . 2
3. High Resolution VoIP Metrics Report Block . . . . . . . . 4
4. RTCP HR Configuration Block . . . . . . . . . . . . . . . . 19
5. Application of RTCP HR to RTP translators . . . . . . . . . 22
6. RTCP HR Block Multiplexing . . . . . . . . . . . . . . . . . 24
7. SDP Signaling . . . . . . . . . . . . . . . . . . . . . . 28
8. Practical applications . . . . . . . . . . . . . . . . . . 28
9. IANA Considerations . . . . . . . . . . . . . . . . . . . . 31
10. Security Considerations . . . . . . . . . . . . . . . . . . 31
11. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 31
12. Informative References . . . . . . . . . . . . . . . . . . . 31
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . 32
Intellectual Property and Copyright Statements . . . . . . . 33
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 [3] VoIP Metrics block or RFC3550 [2] 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[3].
The new block types defined in this draft are the High Resolution
VoIP Metrics Report Block, and the High Resolution VoIP Metrics
Configuration Block.
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.
2. Definitions
2.1 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.
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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.
2.2 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.
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 the media stream was lost through packet loss and
discard. The term Severely Concealed Seconds defines a count of
seconds during which the proportion of the media stream lost through
packet loss and discardeds a specified threshold.
2.3 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 to
255.996 with a granulatity of 0.0039.
S7:8 would represent the range -127.996 to +127.996.
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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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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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 | SCS Threshold |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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/Peak PDV | Pos PDV Percentile |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Neg Threshold/Peak PDV | Neg PDV Percentile |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| PDV Type | JB/PLC config | JB nominal |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| JB maximum | JB abs max |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| JB high water mark | JB low water mark |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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| 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
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
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 [3], see clause 6. 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
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2 Concealed Seconds Metrics block
3 Call Quality Metrics
4 Vendor specific extension block
5-7 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.
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 Loss Proportion
Proportion of frames lost within the network expressed as a binary
fraction in 0:16 format. Duplicate frames shall be disregarded.
3.3.2 Discard Proportion
Proportion of voice frames received but discarded due to late or
early arrival, expressed as a binary fraction in 0:16 format.
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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
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.
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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.
3.4.4 Burst Loss/Discard Proportion
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 Proportion
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
The duration, in milliseconds, of the on-time playout duration
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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).
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-
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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.
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.
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.
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.
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For clarification, the count of Concealed Seconds MUST
include the count of Severely Concealed Seconds.
3.6.3 Severely Concealed Seconds
A count of the number of Severely Concealed Seconds.
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.
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.
3.7.2 End System Delay (ms)
The End System Delay is the internal round trip delay within the
reporting 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 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 IP 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 MAPDV this value is generated according to ITU-T G.1020.
For interval reports the MAPDV value is reset at the start of
the interval.
For PPDV the value reported is the value of J(i) calculated
according to RFC3550 at the time the report is generated.
(16 bit, S11:4 format) expressed in milliseconds
(ii) Positive Threshold/Peak PDV - the PDV associated with the
Positive PDV percentile (16 bit, S11:4 format) expressed in
milliseconds. The term Positive is associated with packets
arriving later than the expected time.
(iii) Negative Threshold/Peak PDV - the PDV associated with the
Negative PDV percentile (16 bit, S11:4 format) expressed in
milliseconds. The term Negative is associated with packets
arriving earler than the expected time.
(iv) Positive PDV Percentile - the percentage of packets on the
call for which individual packet delays were less than the
Positive Threshold PDV expressed in 8:8 format.
(v) Negative PDV Percentile - the percentage of packets on the call
for which individual packet delays were more than the Negative
Threshold PDV expressed in 8:8 format.
If the PDV Type indicated is IPDV and the Positive and Negative PDV
Percentiles are set to 100.0 then the Positive and Negative
Threshold/Peak PDV values are the peak values measured during the
reporting interval (which may be from the start of the call for
cumulative reports). In this case, the difference between the
Positive and Negative Threshold/Peak values defines the range of
IPDV.
3.7.5 PDV Type
Indicates the type of algorithm used to calculate PDV:
PPDV (0) according to RFC3550 [2],
MAPDV (1) according to ITU-T G.1020 [4],
IPDV (2) according to ITU-T Y.1540 [6]
Other values reserved
For example:-
(a) To report PPDV (RFC3550):
Mean PDV = PPDV
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Threshold PDV = Undefined (FF FF)
PDV Percentile = Undefined (FFF)
PDV type = 0 PPDV
(b) To report 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 (except of course in the extreme case of a 100%
percentile).
IPDV, according to Y.1540 is the difference in delay between the
i-th packet and the first packet of the stream. If the sending
and receiving clocks are not synchronized, this metrics includes
the effect of relative timing drift.
3.7.6 Jitter Buffer / PLC Configuration
Indicates the configuration of the jitter buffer and the type of PLC
algorithm in use.
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.
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3.8.1 Listening and Conversation Quality R Factors - R-LQ, R-CQ
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 RTCP HR Configuration block
(see Section 4).
3.8.2 Listening and Conversation Quality MOS - MOS-LQ, MOS-CQ
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 RTCP HR Configuration block.
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 Ext In and Out.
These parameters provide call quality information for external
networks - for example an external PCM or cellular network - or for
a reporting call quality from the "other" side of a transcoding
device or mixer - for example a conference bridge.
R-LQ Ext In - measured by this endpoint for incoming
connection on "other" side of this endpoint
R-LQ Ext Out - copied from RTCP XR message received from
remote endpoint on "other" side of this endpoint
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
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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
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
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
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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
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 "C"
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
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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.
3.9 Vendor Extension sub-block
One or more Vendor Extension sub-blocks MAY be present and if present
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
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enterprise numbers (www.iana.org/assignments/enterprise-numbers)
can be accommodated within two octets. The additional two octets
provide for future growth.
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
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Configuration report.
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
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 Correlation Tag
1 Algorithm Descriptor 1
2 Algorithm Descriptor 2
3 Algorithm Descriptor 3
4 Algorithm Descriptor 4
5 Vendor Specific Extension
6-7 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 to 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.
The Tag Type indicates the use of the correlation tag. The following
values are defined:
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0: IMS Charging Identity (ICID) subfield of the
P-Charging-Vector header specified in RFC 3455 [7].
1: Globally unique ID as specified in ITU-T H.225.0
(Table 20/H.225.0) [8].
2: Conference Identifier, per ITU-T H.225.0
(Table 20/H.225.0) [8].
3: SIP Call-ID as defined in RFC 3261 [9].
4: PacketCable Billing Call ID (BCID) [10].
5: Text string using the US-ASCII character set [11].
6: Octet string.
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.
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
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Descriptor = "Alg X" 0x00 - description
Call quality estimation algorithms may be defined for listening or
conversational quality MOS or R factor.
4.5 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 [2] 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 [2] 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.
For each session, 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 [2] 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.
Where the RTP translator makes its own measurements of the incoming
packet stream, section 7 of RFC3550 [2] 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
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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 (including XR and HR) MAY be configurable differently
for each network interface, and may be chosen differently for each
session if the translator is capable of this level of control. 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.
The following behaviours are potentially useful at a translator:
o RTCP [2] 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.
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".
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NOTE: Behaviour "a", "b", "c" and "d" may be further classified in
a) transparent forwarding ("a_t", "b_t", "c_t", "d_t") and
b) forwarding with modification of RTCP fields ("a_m", "b_m",
c_m", "d_t"), dependent on the specific, VoIP call-individual
mode of operation of the RTP translator (see also below note
at the end of clause 5.1.1). The detailed behaviour is
inherent to the RTP translator configuration, thus orthogonal
to the applied measurement mode.
Material in section 6 illustrates how policies may be applied to
control these behaviours and hence achieve potentially useful end-
to-end reporting modes.
6. RTCP HR Report Multiplexing
For RTP sessions that traverse one or more RTP translators,
implementations MAY concatenate RTCP HR Report Blocks. The order
of concatenated RTCP HR blocks shall be the same as the order
in which network segments are traversed.
This behaviour is described below with respect to the following
example of a bidirectional unicast connection between RTP end
systems A and Z using three 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 |
---- ----- ----- ----- ----
[Editors notes:
(a) In above example, if segment A->J does not support HR then
reports from K, L, Z should not be forwarded to A - for
further study.
(b) Suggest three options -
- local only, i.e. (A->J) only
- endpoint only, .e. (A->Z) only
- all segments (A->J) and (J->K)....]
In the first example (Figure 1) translators implement policies "a",
"b", "c1+c2" and "d" of section 5. The result is that the connection
operates in a mode called "Global Reporting" in which all
measurements by translators are forwarded to end systems. Without
concatenation, this set of policies results in a large number of
separate RTCP packets.
In Figure 2, concatenation is applied to reduce the number of
separate RTCP packets, here to 3. The packet labelled "RTCP(A.R)"
is the report from the end system, identified because its SSRC is
the same as the SSRC of the RTP flow in the same direction. This is
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forwarded without concatenation. The packet labelled RTCP(J.1R,
K.1R, L.1R) contains translator reports of measurements made on the
RTP flow from A to Z, identified because they report the measured
flow having SSRC equal to the SSRC of A. The packet labelled RTCP
(J.2R,K.2R,L.2R) contains translator reports of measurements
made on the RTP flow from Z to A, identified because they report
the measured flow having SSRC equal to the SSRC of Z.
A.R/A.T------>J.1R/J.2T---->K.1R/K.2T---->L.1R/L.2T------>Z.R/Z.T
| RTP | | | |
|<------------|-------------|-------------|------------->|
| | | | |
| RTCP(A.R) | RTCP(A.R) | RTCP(A.R) | RTCP(A.R) |
|- - - - - - >|- - - - - - >|- - - - - - >|- - - - - - > |
| | RTCP(J.1R) | RTCP(J.1R) | RTCP(J.1R) |
| |- - - - - - >|- - - - - - >|- - - - - - > |
| | RTCP(J.2R) | RTCP(J.2R) | RTCP(J.2R) |
| |- - - - - - >|- - - - - - >|- - - - - - > |
| | | RTCP(K.1R) | RTCP(K.1R) |
| | |- - - - - - >|- - - - - - > |
| | | RTCP(K.2R) | RTCP(K.2R) |
| | |- - - - - - >|- - - - - - > |
| | | | RTCP(L.2R) |
| | | |- - - - - - > |
| | | | RTCP(L.2R) |
| | | |- - - - - - > |
| | | | |
A.T/A.R<------J.1T/J.2R<----K.1T/K.2R<----L.1T/L.2R<------Z.T/Z.R
| RTP | | | |
|<------------|-------------|-------------|------------->|
| | | | |
| RTCP(Z.R) | RTCP(Z.R) | RTCP(Z.R) | RTCP(Z.R) |
|< - - - - - -|< - - - - - -|< - - - - - -|< - - - - - - |
| RTCP(L.1R) | RTCP(L.1R) | RTCP(L.1R) | |
|< - - - - - -|< - - - - - -|< - - - - - -| |
| RTCP(L.2R) | RTCP(L.2R) | RTCP(L.2R) | |
|< - - - - - -|< - - - - - -|< - - - - - -| |
| RTCP(K.1R) | RTCP(K.1R) | | |
|< - - - - - -|< - - - - - -| | |
| RTCP(K.2R) | RTCP(K.2R) | | |
|< - - - - - -|< - - - - - -| | |
| RTCP(J.1R) | | | |
|< - - - - - -| | | |
| RTCP(J.2R) | | | |
|< - - - - - -| | | |
| | | | |
Figure 1 The Global reporting mode
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A.R/A.T------>J.1R/J.2T---->K.1R/K.2T---->L.1R/L.2T------>Z.R/Z.T
| RTP | | | |
|<------------|-------------|-------------|------------->|
| | | | |
| RTCP(A.R) | RTCP(A.R) | RTCP(A.R) | RTCP(A.R) |
|- - - - - - >|- - - - - - >|- - - - - - >|- - - - - - > |
| | | | RTCP(J.1R, |
| | | RTCP(J.1R, | K.1R, |
| | RTCP(J.1R) | K.1R) | L.1R) |
| |- - - - - - >|- - - - - - >|- - - - - - > |
| | | | RTCP(J.2R, |
| | | RTCP(J.2R, | K.2R, |
| | RTCP(J.2R) | K.2R) | L.2R) |
| |- - - - - - >|- - - - - - >|- - - - - - > |
A.T/A.R<------J.1T/J.2R<----K.1T/K.2R<----L.1T/L.2R<------Z.T/Z.R
| RTP | | | |
|<------------|-------------|-------------|------------->|
| | | | |
| RTCP(Z.R) | RTCP(Z.R) | RTCP(Z.R) | RTCP(Z.R) |
|< - - - - - -|< - - - - - -|< - - - - - -|< - - - - - - |
| RTCP(L.1R, | | | |
| K.1R, | RTCP(L.1R, | | |
| J.1R) | K.1R) | RTCP(L.1R) | |
|< - - - - - -|< - - - - - -|< - - - - - -| |
| RTCP(L.2R, | | | |
| K.2R, | RTCP(L.2R, | | |
| J.2R) | K.2R) | RTCP(L.2R) | |
|< - - - - - -|< - - - - - -|< - - - - - -| |
| | | | |
Figure 2 Proposed multiplexing scheme of the Global reporting mode
In the second example, (Figure 3) translators implement policies
"a", "b", and "c1+c2" of section 5. The result is that the connection
operates in a mode called "Local Reporting" in which measurements by
each translator are sent to the nearest-neighbour RTP system in each
direction but are not forwarded further. Even without concatenation,
this set of policies results in a maximum of three RTCP packets in
each direction on each link (per RTCP measurement cycle).
Concatenation does not occur in this mode.
A.R/A.T------>J.1R/J.2T---->K.1R/K.2T---->L.1R/L.2T------>Z.R/Z.T
| RTP | | | |
|<------------|-------------|-------------|------------->|
| | | | |
| RTCP(A.R) | RTCP(A.R) | RTCP(A.R) | RTCP(A.R) |
|- - - - - - >|- - - - - - >|- - - - - - >|- - - - - - > |
Clark [Page 26]
draft-ietf-avt-rtcphr-01.txt July 2007
| | RTCP(J.1R) | RTCP(K.1R) | RTCP(L.1R) |
| |- - - - - - >|- - - - - - >|- - - - - - > |
| | RTCP(J.2R) | RTCP(K.2R) | RTCP(L.2R) |
| |- - - - - - >|- - - - - - >|- - - - - - > |
| | | | |
| | | | |
A.T/A.R<------J.1T/J.2R<----K.1T/K.2R<----L.1T/L.2R<------Z.T/Z.R
| RTP | | | |
|<------------|-------------|-------------|------------->|
| | | | |
| RTCP(Z.R) | RTCP(Z.R) | RTCP(Z.R) | RTCP(Z.R) |
|< - - - - - -|< - - - - - -|< - - - - - -|< - - - - - - |
| RTCP(J.1R) | RTCP(K.1R) | RTCP(L.1R) | |
|< - - - - - -|< - - - - - -|< - - - - - -| |
| RTCP(J.2R) | RTCP(K.2R) | RTCP(L.2R) | |
|< - - - - - -|< - - - - - -|< - - - - - -| |
| | | | |
Figure 3 Local reporting mode
Figure 4 depicts an example of the RTCP reports configuration of
the proposed multiplexing scheme.
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Header |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| Sender info |
| |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| Report block of SSRC 1 |
| |
| |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| Chunk 1 |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| HR header |
| |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| Report blocks (Chunk 1) |
| |
| |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| |
| ... |
| |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| Chunk N |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| HR header |
Clark [Page 27]
draft-ietf-avt-rtcphr-01.txt July 2007
| |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| Report blocks (Chunk N) |
| |
| |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
Figure 4 An example of the multiplexed RTCP reports packet
7. SDP Signaling
7.1 Block type values: assignment of initial value "mmm"
The assignment of the initial value "mmm" is under responsibility of
"non-RTP means" (definition see clause 3/RFC 3550), which is
typically SDP signalling.
7.2 Applicability for single port RTP/RTCP sessions
The selection of a value range for HR block types is independent of
the fact, whether RTP and RTCP using different ports or a single port
(see draft-ietf-avt-rtp-and-rtcp-mux).
Single port RTP/RTCP applications must ensure that RTP payload type
79 is NOT used (because in conflict with RTCP XR packet type 207,
see clause 4/draft-ietf-avt-rtp-and-rtcp-mux).
7.3 The SDP attribute "rtcp-hr"
The "rtcp-hr" attribute is complementing/extending the "rtcp-xr" SDP
attribute as defined in clause 5.1/RFC3611.
<Note: this clause is so far a placeholder, assuming that
the rtcp-xr attribute is not sufficient for RTCP HR. For further
study.>
7.4 Usage in Offer/Answer
See clause 5.2/RFC3611.
<Note: a correspondent modification would be required
dependent on "rtcp-hr" attribute. For further study.>
7.5 Usage Outside of Offer/Answer
See clause 5.3/RFC3611.
<Note: a correspondent modification would be required dependent on
"rtcp-hr" attribute. For further study.>
8. Practical Applications
8.1 Overview
Clark [Page 28]
draft-ietf-avt-rtcphr-01.txt July 2007
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.
8.2 Supplementary Services: Call Hold and Transfer
8.2.1 General
Supplementary services are under control of call/session control
protocols like SIP. Such signaling protocols are acting also as
"non-RTP means" (definition see clause 3/RFC 3550) in such service
scenarios.
The "northbound" served user instance for RTCP HR data is typically
"co-located" to the served user instance of the call/session
control protocol controlling the supplementary service. This allows
to correlate in principle supplementary service control events with
RTCP HR measurements in such network elements (like a SIP UA, SIP
proxy, application server, etc.).
Thus, the correlation between RTP/RTCP session control and
supplementary service control allows basically the minimization of
potential ambiguity.
Below sub-clause providing some additional notes dependent on
specific supplementary services.
8.2.2 Supplementary Service: Call Transfer
A successful call transfer means that an initial call between A and
B is transferred to a call between C and B. This means that the RTP
end system A is "replaced" by RTP end system C, accompanied by all
correspondent changes in a RTP/RTCP endpoint (e.g., SSRC for A
"replaced" by SSRC for B).
In the scope of RTCP HR, it is therefore recommended to consider
the two call phases (1st phase: call A-B, 2nd phase: call C-B) as
separate measurement phases.
Separate measurement phases could be e.g. based on interval metrics
and the derivation of call phase-individual cumulative metrics by
the "northbound" served user instance of RTCP HR, or by "resetting"
the cumulative metrics in each call phase.
8.2.3 Supplementary Service: Call Hold
Call hold enables the served (holding) user A to put user B (with
whom user A has an active call) into a hold condition (held user)
and subsequently to retrieve that user again.
During this hold condition, user B may be provided with media on
hold (MoH). The served (holding) user A may perform other actions
while user B is being held, e.g. consulting with another user C.
In the scope of RTCP HR, it is recommended to consider the
different call phases firstly as separate measurement phases (see
also 8.2.2).
Clark [Page 29]
draft-ietf-avt-rtcphr-01.txt July 2007
8.3 Bitrate efficiency improvements: VAD/Silence Suppression based
on Voice Activity Detection (VAD)Elimination
A VoIP call is either enabled or disabled for silence suppression.
This is typically a call-individual configuration parameter,
negotiated during call establishment phase, and not changed anymore
during the remaining call phase.
An enabled silence suppression mode is basically affecting almost
all high resolution VoIP metrics.
The "northbound" served user instance of RTCP HR may require access
to the information, whether silence suppression was enabled or
disabled for that call, in order to indicate that mode of operation
in the VoIP measurement data.
8.4 Endpoint configuration changes mid-call
An endpoint relates to an RTP end system, which can be either
a) located in VoIP user/terminal equipment (e.g. SIP UA), or
b) located in VoIP gateway equipment (e.g. PSTN-to-RTP H.248
media gateway), or
c) located in VoIP media server equipment.
8.4.1 Changes due to mid-call transitions between different voice
codec types
Voice codec type changes are reflected in RTP payload type changes,
which are visible in the Call Quality metrics sub-block (see clause
3.1).
8.4.2 Changes due to mid-call transitions from VoIP to RTP-based
VBDoIP
There might be mid-call transitions from VoIP to dedicated modes of
operation for voiceband data services support in case that at least
one RTP end system is located in type (b) equipment.
Mode transitions should be again reflected in RTP payload type
changes in case of RTP-based VBD transport (e.g. like ITU-T Rec.
V.152 for VBDoIP).
Details are for further study.
8.4.3 Changes due to mid-call transitions from VoIP to non-RTP
-based VBDoIP
UDPTL/UDP based realtime facsimile according ITU-T Rec. T.38 is an
example for RTP-less transport of facsimile/modem signals.
Any mid-call transition to T.38 would inherently terminated the
RTP/RTCP session, thus the measurement phase.
Details are for further study.
8.5 SSRC changes mid-call
An SSRC change may be e.g. the consequence of a mid-call transport
address change.
Details are for further study.
Clark [Page 30]
draft-ietf-avt-rtcphr-01.txt July 2007
9. IANA Considerations
This document defines a series of new RTCP Extended Report (XR)
block types within the existing Internet Assigned Numbers
Authority (IANA) registry of RTP RTCP XR block types. In
addition, this document defines the need for an IANA registry
of correlation tag types (Section 4.3)
10. 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.
11. Contributors
The authors gratefully acknowledge the comments and contributions
made by Jim Frauenthal, Mike Ramalho, Paul Jones, Claus Dahm, Bob
Biskner, Mohamed Mostafa, Tom Hock, Albert Higashi, Shane Holthaus,
Amit Arora, Bruce Adams, Albrecht Schwarz, Keith Lantz, Randy Ethier,
Philip Arden, Ravi Raviraj and Hideaki Yamada.
12. 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)",
Clark [Page 31]
draft-ietf-avt-rtcphr-01.txt July 2007
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.
[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.
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
Full Copyright Statement
Clark [Page 32]
draft-ietf-avt-rtcphr-01.txt July 2007
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Clark [Page 33]
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