RFC 3611
RTP Control Protocol Extended Reports (RTCP XR)
4.7. VoIP Metrics Report Block
The VoIP Metrics Report Block provides metrics for monitoring voice over IP (VoIP) calls. These metrics include packet loss and discard metrics, delay metrics, analog metrics, and voice quality metrics. The block reports separately on packets lost on the IP channel, and those that have been received but then discarded by the receiving jitter buffer. It also reports on the combined effect of losses and discards, as both have equal effect on call quality. In order to properly assess the quality of a Voice over IP call, it is desirable to consider the degree of burstiness of packet loss [14]. Following a Gilbert-Elliott model [3], a period of time, bounded by lost and/or discarded packets with a high rate of losses and/or discards, is a "burst", and a period of time between two bursts is a "gap". Bursts correspond to periods of time during which the packet loss rate is high enough to produce noticeable degradation in audio quality. Gaps correspond to periods of time during which only isolated lost packets may occur, and in general these can be masked by packet loss concealment. Delay reports include the transit delay between RTP end points and the VoIP end system processing delays, both of which contribute to the user perceived delay. Additional metrics include signal, echo, noise, and distortion levels. Call quality metrics include R factors (as described by the E Model defined in [6,3]) and mean opinion scores (MOS scores). Implementations MUST provide values for all the fields defined here. For certain metrics, if the value is undefined or unknown, then the specified default or unknown field value MUST be provided.
The block is encoded as seven 32-bit words:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=7 | reserved | block length = 8 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| loss rate | discard rate | burst density | gap density |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| burst duration | gap duration |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| round trip delay | end system delay |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| signal level | noise level | RERL | Gmin |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| R factor | ext. R factor | MOS-LQ | MOS-CQ |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| RX config | reserved | JB nominal |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| JB maximum | JB abs max |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
A VoIP Metrics Report Block is identified by the constant 7.
reserved: 8 bits
This field is reserved for future definition. In the absence
of such a definition, the bits in this field MUST be set to
zero and MUST be ignored by the receiver.
block length: 16 bits
The constant 8, in accordance with the definition of this field
in Section 3.
SSRC of source: 32 bits
As defined in Section 4.1.
The remaining fields are described in the following six sections:
Packet Loss and Discard Metrics, Delay Metrics, Signal Related
Metrics, Call Quality or Transmission Quality Metrics, Configuration
Metrics, and Jitter Buffer Parameters.
4.7.1. Packet Loss and Discard Metrics
It is very useful to distinguish between packets lost by the network and those discarded due to jitter. Both have equal effect on the quality of the voice stream, however, having separate counts helps identify the source of quality degradation. These fields MUST be populated, and MUST be set to zero if no packets have been received. loss rate: 8 bits The fraction of RTP data packets from the source lost since the beginning of reception, expressed as a fixed point number with the binary point at the left edge of the field. This value is calculated by dividing the total number of packets lost (after the effects of applying any error protection such as FEC) by the total number of packets expected, multiplying the result of the division by 256, limiting the maximum value to 255 (to avoid overflow), and taking the integer part. The numbers of duplicated packets and discarded packets do not enter into this calculation. Since receivers cannot be required to maintain unlimited buffers, a receiver MAY categorize late-arriving packets as lost. The degree of lateness that triggers a loss SHOULD be significantly greater than that which triggers a discard. discard rate: 8 bits The fraction of RTP data packets from the source that have been discarded since the beginning of reception, due to late or early arrival, under-run or overflow at the receiving jitter buffer. This value is expressed as a fixed point number with the binary point at the left edge of the field. It is calculated by dividing the total number of packets discarded (excluding duplicate packet discards) by the total number of packets expected, multiplying the result of the division by 256, limiting the maximum value to 255 (to avoid overflow), and taking the integer part.4.7.2. Burst Metrics
A burst is a period during which a high proportion of packets are either lost or discarded due to late arrival. A burst is defined, in terms of a value Gmin, as the longest sequence that (a) starts with a lost or discarded packet, (b) does not contain any occurrences of Gmin or more consecutively received (and not discarded) packets, and (c) ends with a lost or discarded packet. A gap, informally, is a period of low packet losses and/or discards. Formally, a gap is defined as any of the following: (a) the period from the start of an RTP session to the receipt time of the last
received packet before the first burst, (b) the period from the end
of the last burst to either the time of the report or the end of the
RTP session, whichever comes first, or (c) the period of time between
two bursts.
For the purpose of determining if a lost or discarded packet near the
start or end of an RTP session is within a gap or a burst, it is
assumed that the RTP session is preceded and followed by at least
Gmin received packets, and that the time of the report is followed by
at least Gmin received packets.
A gap has the property that any lost or discarded packets within the
gap must be preceded and followed by at least Gmin packets that were
received and not discarded. This gives a maximum loss/discard rate
within a gap of: 1 / (Gmin + 1).
A Gmin value of 16 is RECOMMENDED, as it results in gap
characteristics that correspond to good quality (i.e., low packet
loss rate, a minimum distance of 16 received packets between lost
packets), and hence differentiates nicely between good and poor
quality periods.
For example, a 1 denotes a received packet, 0 a lost packet, and X a
discarded packet in the following pattern covering 64 packets:
11110111111111111111111X111X1011110111111111111111111X111111111
|---------gap----------|--burst---|------------gap------------|
The burst consists of the twelve packets indicated above, starting at
a discarded packet and ending at a lost packet. The first gap starts
at the beginning of the session and the second gap ends at the time
of the report.
If the packet spacing is 10 ms and the Gmin value is the recommended
value of 16, the burst duration is 120 ms, the burst density 0.33,
the gap duration 230 ms + 290 ms = 520 ms, and the gap density 0.04.
This would result in reported values as follows (see field
descriptions for semantics and details on how these are calculated):
loss rate 12, which corresponds to 5%
discard rate 12, which corresponds to 5%
burst density 84, which corresponds to 33%
gap density 10, which corresponds to 4%
burst duration 120, value in milliseconds
gap duration 520, value in milliseconds
burst density: 8 bits
The fraction of RTP data packets within burst periods since the
beginning of reception that were either lost or discarded.
This value is expressed as a fixed point number with the binary
point at the left edge of the field. It is calculated by
dividing the total number of packets lost or discarded
(excluding duplicate packet discards) within burst periods by
the total number of packets expected within the burst periods,
multiplying the result of the division by 256, limiting the
maximum value to 255 (to avoid overflow), and taking the
integer part. This field MUST be populated and MUST be set to
zero if no packets have been received.
gap density: 8 bits
The fraction of RTP data packets within inter-burst gaps since
the beginning of reception that were either lost or discarded.
The value is expressed as a fixed point number with the binary
point at the left edge of the field. It is calculated by
dividing the total number of packets lost or discarded
(excluding duplicate packet discards) within gap periods by the
total number of packets expected within the gap periods,
multiplying the result of the division by 256, limiting the
maximum value to 255 (to avoid overflow), and taking the
integer part. This field MUST be populated and MUST be set to
zero if no packets have been received.
burst duration: 16 bits
The mean duration, expressed in milliseconds, of the burst
periods that have occurred since the beginning of reception.
The duration of each period is calculated based upon the
packets that mark the beginning and end of that period. It is
equal to the timestamp of the end packet, plus the duration of
the end packet, minus the timestamp of the beginning packet.
If the actual values are not available, estimated values MUST
be used. If there have been no burst periods, the burst
duration value MUST be zero.
gap duration: 16 bits
The mean duration, expressed in milliseconds, of the gap
periods that have occurred since the beginning of reception.
The duration of each period is calculated based upon the packet
that marks the end of the prior burst and the packet that marks
the beginning of the subsequent burst. It is equal to the
timestamp of the subsequent burst packet, minus the timestamp
of the prior burst packet, plus the duration of the prior burst
packet. If the actual values are not available, estimated
values MUST be used. In the case of a gap that occurs at the
beginning of reception, the sum of the timestamp of the prior
burst packet and the duration of the prior burst packet are
replaced by the reception start time. In the case of a gap
that occurs at the end of reception, the timestamp of the
subsequent burst packet is replaced by the reception end time.
If there have been no gap periods, the gap duration value MUST
be zero.
4.7.3. Delay Metrics
For the purpose of the following definitions, the RTP interface is
the interface between the RTP instance and the voice application
(i.e., FEC, de-interleaving, de-multiplexing, jitter buffer). For
example, the time delay due to RTP payload multiplexing would be
considered part of the voice application or end-system delay, whereas
delay due to multiplexing RTP frames within a UDP frame would be
considered part of the RTP reported delay. This distinction is
consistent with the use of RTCP for delay measurements.
round trip delay: 16 bits
The most recently calculated round trip time between RTP
interfaces, expressed in milliseconds. This value MAY be
measured using RTCP, the DLRR method defined in Section 4.5 of
this document, where it is necessary to convert the units of
measurement from NTP timestamp values to milliseconds, or other
approaches. If RTCP is used, then the reported delay value is
the time of receipt of the most recent RTCP packet from source
SSRC, minus the LSR (last SR) time reported in its SR (Sender
Report), minus the DLSR (delay since last SR) reported in its
SR. A non-zero LSR value is required in order to calculate
round trip delay. A value of 0 is permissible; however, this
field MUST be populated as soon as a delay estimate is
available.
end system delay: 16 bits
The most recently estimated end system delay, expressed in
milliseconds. End system delay is defined as the sum of the
total sample accumulation and encoding delay associated with
the sending direction and the jitter buffer, decoding, and
playout buffer delay associated with the receiving direction.
This delay MAY be estimated or measured. This value SHOULD be
provided in all VoIP metrics reports. If an implementation is
unable to provide the data, the value 0 MUST be used.
Note that the one way symmetric VoIP segment delay may be calculated
from the round trip and end system delays is as follows; if the round
trip delay is denoted, RTD and the end system delays associated with
the two endpoints are ESD(A) and ESD(B) then:
one way symmetric voice path delay = ( RTD + ESD(A) + ESD(B) ) / 2
4.7.4. Signal Related Metrics
The following metrics are intended to provide real time information
related to the non-packet elements of the voice over IP system to
assist with the identification of problems affecting call quality.
The values identified below must be determined for the received audio
signal. The information required to populate these fields may not be
available in all systems, although it is strongly recommended that
this data SHOULD be provided to support problem diagnosis.
signal level: 8 bits
The voice signal relative level is defined as the ratio of the
signal level to a 0 dBm0 reference [10], expressed in decibels
as a signed integer in two's complement form. This is measured
only for packets containing speech energy. The intent of this
metric is not to provide a precise measurement of the signal
level but to provide a real time indication that the signal
level may be excessively high or low.
signal level = 10 Log10 ( rms talkspurt power (mW) )
A value of 127 indicates that this parameter is unavailable.
Typical values should generally be in the -15 to -20 dBm range.
noise level: 8 bits
The noise level is defined as the ratio of the silent period
background noise level to a 0 dBm0 reference, expressed in
decibels as a signed integer in two's complement form.
noise level = 10 Log10 ( rms silence power (mW) )
A value of 127 indicates that this parameter is unavailable.
residual echo return loss (RERL): 8 bits
The residual echo return loss value may be measured directly by
the VoIP end system's echo canceller or may be estimated by
adding the echo return loss (ERL) and echo return loss
enhancement (ERLE) values reported by the echo canceller.
RERL(dB) = ERL (dB) + ERLE (dB)
In the case of a VoIP gateway, the source of echo is typically
line echo that occurs at 2-4 wire conversion points in the
network. This can be in the 8-12 dB range. A line echo
canceler can provide an ERLE of 30 dB or more and hence reduce
this to 40-50 dB. In the case of an IP phone, this could be
acoustic coupling between handset speaker and microphone or
residual acoustic echo from speakerphone operation, and may
more correctly be termed terminal coupling loss (TCL). A
typical handset would result in 40-50 dB of echo loss due to
acoustic feedback.
Examples:
- IP gateway connected to circuit switched network with 2 wire
loop. Without echo cancellation, typical 2-4 wire converter
ERL of 12 dB. RERL = ERL + ERLE = 12 + 0 = 12 dB.
- IP gateway connected to circuit switched network with 2 wire
loop. With echo canceler that improves echo by 30 dB.
RERL = ERL + ERLE = 12 + 30 = 42 dB.
- IP phone with conventional handset. Acoustic coupling from
handset speaker to microphone (terminal coupling loss) is
typically 40 dB. RERL = TCL = 40 dB.
If we denote the local end of the VoIP path as A and the remote
end as B, and if the sender loudness rating (SLR) and receiver
loudness rating (RLR) are known for A (default values 8 dB and
2 dB respectively), then the echo loudness level at end A
(talker echo loudness rating or TELR) is given by:
TELR(A) = SRL(A) + ERL(B) + ERLE(B) + RLR(A)
TELR(B) = SRL(B) + ERL(A) + ERLE(A) + RLR(B)
Hence, in order to incorporate echo into a voice quality
estimate at the A end of a VoIP connection, it is desirable to
send the ERL + ERLE value from B to A using a format such as
RTCP XR.
Echo related information may not be available in all VoIP end
systems. As echo does have a significant effect on
conversational quality, it is recommended that estimated values
for echo return loss and terminal coupling loss be provided (if
sensible estimates can be reasonably determined).
Typical values for end systems are given below to provide
guidance:
- IP Phone with handset: typically 45 dB.
- PC softphone or speakerphone: extremely variable, consider
reporting "undefined" (127).
- IP gateway with line echo canceller: typically has ERL and
ERLE available.
- IP gateway without line echo canceller: frequently a source
of echo related problems, consider reporting either a low
value (12 dB) or "undefined" (127).
Gmin
See Configuration Parameters (Section 4.7.6, below).
4.7.5. Call Quality or Transmission Quality Metrics
The following metrics are direct measures of the call quality or
transmission quality, and incorporate the effects of codec type,
packet loss, discard, burstiness, delay etc. These metrics may not
be available in all systems, however, they SHOULD be provided in
order to support problem diagnosis.
R factor: 8 bits
The R factor is a voice quality metric describing the segment
of the call that is carried over this RTP session. It is
expressed as an integer in the range 0 to 100, with a value of
94 corresponding to "toll quality" and values of 50 or less
regarded as unusable. This metric is defined as including the
effects of delay, consistent with ITU-T G.107 [6] and ETSI TS
101 329-5 [3].
A value of 127 indicates that this parameter is unavailable.
Values other than 127 and the valid range defined above MUST
not be sent and MUST be ignored by the receiving system.
ext. R factor: 8 bits
The external R factor is a voice quality metric describing the
segment of the call that is carried over a network segment
external to the RTP segment, for example a cellular network.
Its values are interpreted in the same manner as for the RTP R
factor. This metric is defined as including the effects of
delay, consistent with ITU-T G.107 [6] and ETSI TS 101 329-5
[3], and relates to the outward voice path from the Voice over
IP termination for which this metrics block applies.
A value of 127 indicates that this parameter is unavailable.
Values other than 127 and the valid range defined above MUST
not be sent and MUST be ignored by the receiving system.
Note that an overall R factor may be estimated from the RTP segment R
factor and the external R factor, as follows:
R total = RTP R factor + ext. R factor - 94
MOS-LQ: 8 bits
The estimated mean opinion score for listening quality (MOS-LQ)
is a voice quality metric on a scale from 1 to 5, in which 5
represents excellent and 1 represents unacceptable. This
metric is defined as not including the effects of delay and can
be compared to MOS scores obtained from listening quality (ACR)
tests. It is expressed as an integer in the range 10 to 50,
corresponding to MOS x 10. For example, a value of 35 would
correspond to an estimated MOS score of 3.5.
A value of 127 indicates that this parameter is unavailable.
Values other than 127 and the valid range defined above MUST
not be sent and MUST be ignored by the receiving system.
MOS-CQ: 8 bits
The estimated mean opinion score for conversational quality
(MOS-CQ) is defined as including the effects of delay and other
effects that would affect conversational quality. The metric
may be calculated by converting an R factor determined
according to ITU-T G.107 [6] or ETSI TS 101 329-5 [3] into an
estimated MOS using the equation specified in G.107. It is
expressed as an integer in the range 10 to 50, corresponding to
MOS x 10, as for MOS-LQ.
A value of 127 indicates that this parameter is unavailable.
Values other than 127 and the valid range defined above MUST
not be sent and MUST be ignored by the receiving system.
4.7.6. Configuration Parameters
Gmin: 8 bits
The gap threshold. This field contains the value used for this
report block to determine if a gap exists. The recommended
value of 16 corresponds to a burst period having a minimum
density of 6.25% of lost or discarded packets, which may cause
noticeable degradation in call quality; during gap periods, if
packet loss or discard occurs, each lost or discarded packet
would be preceded by and followed by a sequence of at least 16
received non-discarded packets. Note that lost or discarded
packets that occur within Gmin packets of a report being
generated may be reclassified as part of a burst or gap in
later reports. ETSI TS 101 329-5 [3] defines a computationally
efficient algorithm for measuring burst and gap density using a
packet loss/discard event driven approach. This algorithm is
reproduced in Appendix A.2 of the present document. Gmin MUST
not be zero, MUST be provided, and MUST remain constant across
VoIP Metrics report blocks for the duration of the RTP session.
receiver configuration byte (RX config): 8 bits
This byte consists of the following fields:
0 1 2 3 4 5 6 7
+-+-+-+-+-+-+-+-+
|PLC|JBA|JB rate|
+-+-+-+-+-+-+-+-+
packet loss concealment (PLC): 2 bits
Standard (11) / enhanced (10) / disabled (01) / unspecified
(00). When PLC = 11, then a simple replay or interpolation
algorithm is being used to fill-in the missing packet; this
approach is typically able to conceal isolated lost packets at
low packet loss rates. When PLC = 10, then an enhanced
interpolation algorithm is being used; algorithms of this type
are able to conceal high packet loss rates effectively. When
PLC = 01, then silence is being inserted in place of lost
packets. When PLC = 00, then no information is available
concerning the use of PLC; however, for some codecs this may be
inferred.
jitter buffer adaptive (JBA): 2 bits
Adaptive (11) / non-adaptive (10) / reserved (01)/ unknown
(00). When the jitter buffer is adaptive, then its size is
being dynamically adjusted to deal with varying levels of
jitter. When non-adaptive, the jitter buffer size is
maintained at a fixed level. When either adaptive or non-
adaptive modes are specified, then the jitter buffer size
parameters below MUST be specified.
jitter buffer rate (JB rate): 4 bits
J = adjustment rate (0-15). This represents the implementation
specific adjustment rate of a jitter buffer in adaptive mode.
This parameter is defined in terms of the approximate time
taken to fully adjust to a step change in peak to peak jitter
from 30 ms to 100 ms such that:
adjustment time = 2 * J * frame size (ms)
This parameter is intended only to provide a guide to the
degree of "aggressiveness" of an adaptive jitter buffer and may
be estimated. A value of 0 indicates that the adjustment time
is unknown for this implementation.
reserved: 8 bits
This field is reserved for future definition. In the absence
of such a definition, the bits in this field MUST be set to
zero and MUST be ignored by the receiver.
4.7.7. Jitter Buffer Parameters
The values reported in these fields SHOULD be the most recently
obtained values at the time of reporting.
jitter buffer nominal delay (JB nominal): 16 bits
This is the current nominal jitter buffer delay in
milliseconds, which corresponds to the nominal jitter buffer
delay for packets that arrive exactly on time. This parameter
MUST be provided for both fixed and adaptive jitter buffer
implementations.
jitter buffer maximum delay (JB maximum): 16 bits
This is the current maximum jitter buffer delay in milliseconds
which corresponds to the earliest arriving packet that would
not be discarded. In simple queue implementations this may
correspond to the nominal size. In adaptive jitter buffer
implementations, this value may dynamically vary up to JB abs
max (see below). This parameter MUST be provided for both
fixed and adaptive jitter buffer implementations.
jitter buffer absolute maximum delay (JB abs max): 16 bits
This is the absolute maximum delay in milliseconds that the
adaptive jitter buffer can reach under worst case conditions.
If this value exceeds 65535 milliseconds, then this field SHALL
convey the value 65535. This parameter MUST be provided for
adaptive jitter buffer implementations and its value MUST be
set to JB maximum for fixed jitter buffer implementations.
5. SDP Signaling
This section defines Session Description Protocol (SDP) [4] signaling
for XR blocks that can be employed by applications that utilize SDP.
This signaling is defined to be used either by applications that
implement the SDP Offer/Answer model [8] or by applications that use
SDP to describe media and transport configurations in connection
with such protocols as the Session Announcement Protocol (SAP) [15] or the Real Time Streaming Protocol (RTSP) [17]. There exist other potential signaling methods that are not defined here. The XR blocks MAY be used without prior signaling. This is consistent with the rules governing other RTCP packet types, as described in [9]. An example in which signaling would not be used is an application that always requires the use of one or more XR blocks. However, for applications that are configured at session initiation, the use of some type of signaling is recommended. Note that, although the use of SDP signaling for XR blocks may be optional, if used, it MUST be used as defined here. If SDP signaling is used in an environment where XR blocks are only implemented by some fraction of the participants, the ones not implementing the XR blocks will ignore the SDP attribute.5.1. The SDP Attribute
This section defines one new SDP attribute "rtcp-xr" that can be used to signal participants in a media session that they should use the specified XR blocks. This attribute can be easily extended in the future with new parameters to cover any new report blocks. The RTCP XR blocks SDP attribute is defined below in Augmented Backus-Naur Form (ABNF) [2]. It is both a session and a media level attribute. When specified at session level, it applies to all media level blocks in the session. Any media level specification MUST replace a session level specification, if one is present, for that media block. rtcp-xr-attrib = "a=" "rtcp-xr" ":" [xr-format *(SP xr-format)] CRLF xr-format = pkt-loss-rle / pkt-dup-rle / pkt-rcpt-times / rcvr-rtt / stat-summary / voip-metrics / format-ext pkt-loss-rle = "pkt-loss-rle" ["=" max-size] pkt-dup-rle = "pkt-dup-rle" ["=" max-size] pkt-rcpt-times = "pkt-rcpt-times" ["=" max-size] rcvr-rtt = "rcvr-rtt" "=" rcvr-rtt-mode [":" max-size] rcvr-rtt-mode = "all" / "sender" stat-summary = "stat-summary" ["=" stat-flag *("," stat-flag)]
stat-flag = "loss"
/ "dup"
/ "jitt"
/ "TTL"
/ "HL"
voip-metrics = "voip-metrics"
max-size = 1*DIGIT ; maximum block size in octets
DIGIT = %x30-39
format-ext = non-ws-string
non-ws-string = 1*(%x21-FF)
CRLF = %d13.10
The "rtcp-xr" attribute contains zero, one, or more XR block related
parameters. Each parameter signals functionality for an XR block, or
a group of XR blocks. The attribute is extensible so that parameters
can be defined for any future XR block (and a parameter should be
defined for every future block).
Each "rtcp-xr" parameter belongs to one of two categories. The first
category, the unilateral parameters, are for report blocks that
simply report on the RTP stream and related metrics. The second
category, collaborative parameters, are for XR blocks that involve
actions by more than one party in order to carry out their functions.
Round trip time (RTT) measurement is an example of collaborative
functionality. An RTP data packet receiver sends a Receiver
Reference Time Report Block (Section 4.4). A participant that
receives this block sends a DLRR Report Block (Section 4.5) in
response, allowing the receiver to calculate its RTT to that
participant. As this example illustrates, collaborative
functionality may be implemented by two or more different XR blocks.
The collaborative functionality of several XR blocks may be governed
by a single "rtcp-xr" parameter.
For the unilateral category, this document defines the following
parameters. The parameter names and their corresponding XR formats
are as follows:
Parameter name XR block (block type and name)
-------------- ------------------------------------
pkt-loss-rle 1 Loss RLE Report Block
pkt-dup-rle 2 Duplicate RLE Report Block
pkt-rcpt-times 3 Packet Receipt Times Report Block
stat-summary 6 Statistics Summary Report Block
voip-metrics 7 VoIP Metrics Report Block
The "pkt-loss-rle", "pkt-dup-rle", and "pkt-rcpt-times" parameters MAY specify an integer value. This value indicates the largest size the whole report block SHOULD have in octets. This shall be seen as an indication that thinning shall be applied if necessary to meet the target size. The "stat-summary" parameter contains a list indicating which fields SHOULD be included in the Statistics Summary report blocks that are sent. The list is a comma separated list, containing one or more field indicators. The space character (0x20) SHALL NOT be present within the list. Field indicators represent the flags defined in Section 4.6. The field indicators and their respective flags are as follows: Indicator Flag --------- --------------------------- loss loss report flag (L) dup duplicate report flag (D) jitt jitter flag (J) TTL TTL or Hop Limit flag (ToH) HL TTL or Hop Limit flag (ToH) For "loss", "dup", and "jitt", the presence of the indicator indicates that the corresponding flag should be set to 1 in the Statistics Summary report blocks that are sent. The presence of "TTL" indicates that the corresponding flag should be set to 1. The presence of "HL" indicates that the corresponding flag should be set to 2. The indicators "TTL" and "HL" MUST NOT be signaled together. Blocks in the collaborative category are classified as initiator blocks or response blocks. Signaling SHOULD indicate which participants are required to respond to the initiator block. A party that wishes to receive response blocks from those participants can trigger this by sending an initiator block. The collaborative category currently consists only of one functionality, namely the RTT measurement mechanism for RTP data receivers. The collective functionality of the Receiver Reference Time Report Block and DLRR Report Block is represented by the "rcvr- rtt" parameter. This parameter takes as its arguments a mode value and, optionally, a maximum size for the DLRR report block. The mode value "all" indicates that both RTP data senders and data receivers MAY send DLRR blocks, while the mode value "sender" indicates that only active RTP senders MAY send DLRR blocks, i.e., non RTP senders SHALL NOT send DLRR blocks. If a maximum size in octets is included, any DLRR Report Blocks that are sent SHALL NOT exceed the specified size. If size limitations mean that a DLRR Report Block sender cannot report in one block upon all participants from which it has
received a Receiver Reference Time Report Block then it SHOULD report on participants in a round robin fashion across several report intervals. The "rtcp-xr" attributes parameter list MAY be empty. This is useful in cases in which an application needs to signal that it understands the SDP signaling but does not wish to avail itself of XR functionality. For example, an application in a SIP controlled session could signal that it wishes to stop using all XR blocks by removing all applicable SDP parameters in a re-INVITE message that it sends. If XR blocks are not to be used at all from the beginning of a session, it is RECOMMENDED that the "rtcp-xr" attribute not be supplied at all. When the "rtcp-xr" attribute is present, participants SHOULD NOT send XR blocks other than the ones indicated by the parameters. This means that inclusion of a "rtcp-xr" attribute without any parameters tells a participant that it SHOULD NOT send any XR blocks at all. The purpose is to conserve bandwidth. This is especially important when collaborative parameters are applied to a large multicast group: the sending of an initiator block could potentially trigger responses from all participants. There are, however, contexts in which it makes sense to send an XR block in the absence of a parameter signaling its use. For instance, an application might be designed so as to send certain report blocks without negotiation, while using SDP signaling to negotiate the use of other blocks.5.2. Usage in Offer/Answer
In the Offer/Answer context [8], the interpretation of SDP signaling for XR packets depends upon the direction attribute that is signaled: "recvonly", "sendrecv", or "sendonly" [4]. If no direction attribute is supplied, then "sendrecv" is assumed. This section applies only to unicast media streams, except where noted. Discussion of unilateral parameters is followed by discussion of collaborative parameters in this section. For "sendonly" and "sendrecv" media stream offers that specify unilateral "rtcp-xr" attribute parameters, the answerer SHOULD send the corresponding XR blocks. For "sendrecv" offers, the answerer MAY include the "rtcp-xr" attribute in its response, and specify any unilateral parameters in order to request that the offerer send the corresponding XR blocks. The offerer SHOULD send these blocks. For "recvonly" media stream offers, the offerer's use of the "rtcp- xr" attribute in connection with unilateral parameters indicates that the offerer is capable of sending the corresponding XR blocks. If
the answerer responds with an "rtcp-xr" attribute, the offerer SHOULD send XR blocks for each specified unilateral parameter that was in its offer. For multicast media streams, the inclusion of an "rtcp-xr" attribute with unilateral parameters means that every media recipient SHOULD send the corresponding XR blocks. An SDP offer with a collaborative parameter declares the offerer capable of receiving the corresponding initiator and replying with the appropriate responses. For example, an offer that specifies the "rcvr-rtt" parameter means that the offerer is prepared to receive Receiver Reference Time Report Blocks and to send DLRR Report Blocks. An offer of a collaborative parameter means that the answerer MAY send the initiator, and, having received the initiator, the offerer SHOULD send the responses. There are exceptions to the rule that an offerer of a collaborative parameter should send responses. For instance, the collaborative parameter might specify a mode that excludes the offerer; or congestion control or maximum transmission unit considerations might militate against the offerer's response. By including a collaborative parameter in its answer, the answerer declares its ability to receive initiators and to send responses. The offerer MAY then send initiators, to which the answerer SHOULD reply with responses. As for the offer of a collaborative parameter, there are exceptions to the rule that the answerer should reply. When making an SDP offer of a collaborative parameter for a multicast media stream, the offerer SHOULD specify which participants are to respond to a received initiator. A participant that is not specified SHOULD NOT send responses. Otherwise, undue bandwidth might be consumed. The offer indicates that each participant that is specified SHOULD respond if it receives an initiator. It also indicates that a specified participant MAY send an initiator block. An SDP answer for a multicast media stream SHOULD include all collaborative parameters that are present in the offer and that are supported by the answerer. It SHOULD NOT include any collaborative parameter that is absent from the offer. If a participant receives an SDP offer and understands the "rtcp-xr" attribute but does not wish to implement XR functionality offered, its answer SHOULD include an "rtcp-xr" attribute without parameters. By doing so, the party declares that, at a minimum, is capable of understanding the signaling.
5.3. Usage Outside of Offer/Answer
SDP can be employed outside of the Offer/Answer context, for instance for multimedia sessions that are announced through the Session Announcement Protocol (SAP) [15], or streamed through the Real Time Streaming Protocol (RTSP) [17]. The signaling model is simpler, as the sender does not negotiate parameters, but the functionality expected from specifying the "rtcp-xr" attribute is the same as in Offer/Answer. When a unilateral parameter is specified for the "rtcp-xr" attribute associated with a media stream, the receiver of that stream SHOULD send the corresponding XR block. When a collaborative parameter is specified, only the participants indicated by the mode value in the collaborative parameter are concerned. Each such participant that receives an initiator block SHOULD send the corresponding response block. Each such participant MAY also send initiator blocks.6. IANA Considerations
This document defines a new RTCP packet type, the Extended Report (XR) type, within the existing Internet Assigned Numbers Authority (IANA) registry of RTP RTCP Control Packet Types. This document also defines a new IANA registry: the registry of RTCP XR Block Types. Within this new registry, this document defines an initial set of seven block types and describes how the remaining types are to be allocated. Further, this document defines a new SDP attribute, "rtcp-xr", within the existing IANA registry of SDP Parameters. It defines a new IANA registry, the registry of RTCP XR SDP Parameters, and an initial set of six parameters, and describes how additional parameters are to be allocated.6.1. XR Packet Type
The XR packet type defined by this document is registered with the IANA as packet type 207 in the registry of RTP RTCP Control Packet types (PT).6.2. RTCP XR Block Type Registry
This document creates an IANA registry called the RTCP XR Block Type Registry to cover the name space of the Extended Report block type (BT) field specified in Section 3. The BT field contains eight bits, allowing 256 values. The RTCP XR Block Type Registry is to be managed by the IANA according to the Specification Required policy of
RFC 2434 [7]. Future specifications SHOULD attribute block type values in strict numeric order following the values attributed in this document: BT name -- ---- 1 Loss RLE Report Block 2 Duplicate RLE Report Block 3 Packet Receipt Times Report Block 4 Receiver Reference Time Report Block 5 DLRR Report Block 6 Statistics Summary Report Block 7 VoIP Metrics Report Block The BT value 255 is reserved for future extensions. Furthermore, future specifications SHOULD avoid the value 0. Doing so facilitates packet validity checking, since an all-zeros field might commonly be found in an ill-formed packet. Any registration MUST contain the following information: - Contact information of the one doing the registration, including at least name, address, and email. - The format of the block type being registered, consistent with the extended report block format described in Section 3. - A description of what the block type represents and how it shall be interpreted, detailing this information for each of its fields.6.3. The "rtcp-xr" SDP Attribute
The SDP attribute "rtcp-xr" defined by this document is registered with the IANA registry of SDP Parameters as follows: SDP Attribute ("att-field"): Attribute name: rtcp-xr Long form: RTP Control Protocol Extended Report Parameters Type of name: att-field Type of attribute: session and media level Subject to charset: no Purpose: see Section 5 of this document Reference: this document Values: see this document and registrations below
The attribute has an extensible parameter field and therefore a registry for these parameters is required. This document creates an IANA registry called the RTCP XR SDP Parameters Registry. It contains the six parameters defined in Section 5.1: "pkt-loss-rle", "pkt-dup-rle", "pkt-rcpt-times", "stat-summary", "voip-metrics", and "recv-rtt". Additional parameters are to be added to this registry in accordance with the Specification Required policy of RFC 2434 [7]. Any registration MUST contain the following information: - Contact information of the one doing the registration, including at least name, address, and email. - An Augmented Backus-Naur Form (ABNF) [2] definition of the parameter, in accordance with the "format-ext" definition of Section 5.1. - A description of what the parameter represents and how it shall be interpreted, both normally and in Offer/Answer.7. Security Considerations
This document extends the RTCP reporting mechanism. The security considerations that apply to RTCP reports [9, Appendix B] also apply to XR reports. This section details the additional security considerations that apply to the extensions. The extensions introduce heightened confidentiality concerns. Standard RTCP reports contain a limited number of summary statistics. The information contained in XR reports is both more detailed and more extensive (covering a larger number of parameters). The per- packet report blocks and the VoIP Metrics Report Block provide examples. The per-packet information contained in Loss RLE, Duplicate RLE, and Packet Receipt Times Report Blocks facilitates multicast inference of network characteristics (MINC) [11]. Such inference can reveal the gross topology of a multicast distribution tree, as well as parameters, such as the loss rates and delays, along paths between branching points in that tree. Such information might be considered sensitive to autonomous system administrators. The VoIP Metrics Report Block provides information on the quality of ongoing voice calls. Though such information might be carried in an application specific format in standard RTP sessions, making it available in a standard format here makes it more available to potential eavesdroppers.
No new mechanisms are introduced in this document to ensure confidentiality. Encryption procedures, such as those being suggested for a Secure RTCP (SRTCP) [12] at the time that this document was written, can be used when confidentiality is a concern to end hosts. Given that RTCP traffic can be encrypted by the end hosts, autonomous systems must be prepared for the fact that certain aspects of their network topology can be revealed. Any encryption or filtering of XR report blocks entails a loss of monitoring information to third parties. For example, a network that establishes a tunnel to encrypt VoIP Report Blocks denies that information to the service providers traversed by the tunnel. The service providers cannot then monitor or respond to the quality of the VoIP calls that they carry, potentially creating problems for the network's users. As a default, XR packets should not be encrypted or filtered. The extensions also make certain denial of service attacks easier. This is because of the potential to create RTCP packets much larger than average with the per packet reporting capabilities of the Loss RLE, Duplicate RLE, and Timestamp Report Blocks. Because of the automatic bandwidth adjustment mechanisms in RTCP, if some session participants are sending large RTCP packets, all participants will see their RTCP reporting intervals lengthened, meaning they will be able to report less frequently. To limit the effects of large packets, even in the absence of denial of service attacks, applications SHOULD place an upper limit on the size of the XR report blocks they employ. The "thinning" techniques described in Section 4.1 permit the packet-by-packet report blocks to adhere to a predefined size limit.
A. Algorithms
A.1. Sequence Number Interpretation
This is the algorithm suggested by Section 4.1 for keeping track of the sequence numbers from a given sender. It implements the accounting practice required for the generation of Loss RLE Report Blocks. This algorithm keeps track of 16 bit sequence numbers by translating them into a 32 bit sequence number space. The first packet received from a source is considered to have arrived roughly in the middle of that space. Each packet that follows is placed either ahead of or behind the prior one in this 32 bit space, depending upon which choice would place it closer (or, in the event of a tie, which choice would not require a rollover in the 16 bit sequence number). // The reference sequence number is an extended sequence number // that serves as the basis for determining whether a new 16 bit // sequence number comes earlier or later in the 32 bit sequence // space. u_int32 _src_ref_seq; bool _uninitialized_src_ref_seq; // Place seq into a 32-bit sequence number space based upon a // heuristic for its most likely location. u_int32 extend_seq(const u_int16 seq) { u_int32 extended_seq, seq_a, seq_b, diff_a, diff_b; if(_uninitialized_src_ref_seq) { // This is the first sequence number received. Place // it in the middle of the extended sequence number // space. _src_ref_seq = seq | 0x80000000u; _uninitialized_src_ref_seq = false; extended_seq = _src_ref_seq; } else { // Prior sequence numbers have been received. // Propose two candidates for the extended sequence // number: seq_a is without wraparound, seq_b with // wraparound. seq_a = seq | (_src_ref_seq & 0xFFFF0000u); if(_src_ref_seq < seq_a) { seq_b = seq_a - 0x00010000u; diff_a = seq_a - _src_ref_seq;
diff_b = _src_ref_seq - seq_b;
}
else {
seq_b = seq_a + 0x00010000u;
diff_a = _src_ref_seq - seq_a;
diff_b = seq_b - _src_ref_seq;
}
// Choose the closer candidate. If they are equally
// close, the choice is somewhat arbitrary: we choose
// the candidate for which no rollover is necessary.
if(diff_a < diff_b) {
extended_seq = seq_a;
}
else {
extended_seq = seq_b;
}
// Set the reference sequence number to be this most
// recently-received sequence number.
_src_ref_seq = extended_seq;
}
// Return our best guess for a 32-bit sequence number that
// corresponds to the 16-bit number we were given.
return extended_seq;
}
A.2. Example Burst Packet Loss Calculation.
This is an algorithm for measuring the burst characteristics for the
VoIP Metrics Report Block (Section 4.7). The algorithm, which has
been verified against a working implementation for correctness, is
reproduced from ETSI TS 101 329-5 [3]. The algorithm, as described
here, takes precedence over any change that might eventually be made
to the algorithm in future ETSI documents.
This algorithm is event driven and hence extremely computationally
efficient.
Given the following definition of states:
state 1 = received a packet during a gap
state 2 = received a packet during a burst
state 3 = lost a packet during a burst
state 4 = lost an isolated packet during a gap
The "c" variables below correspond to state transition counts, i.e.,
c14 is the transition from state 1 to state 4. It is possible to
infer one of a pair of state transition counts to an accuracy of 1
which is generally sufficient for this application.
"pkt" is the count of packets received since the last packet was
declared lost or discarded, and "lost" is the number of packets lost
within the current burst. "packet_lost" and "packet_discarded" are
Boolean variables that indicate if the event that resulted in this
function being invoked was a lost or discarded packet.
if(packet_lost) {
loss_count++;
}
if(packet_discarded) {
discard_count++;
}
if(!packet_lost && !packet_discarded) {
pkt++;
}
else {
if(pkt >= gmin) {
if(lost == 1) {
c14++;
}
else {
c13++;
}
lost = 1;
c11 += pkt;
}
else {
lost++;
if(pkt == 0) {
c33++;
}
else {
c23++;
c22 += (pkt - 1);
}
}
pkt = 0;
}
At each reporting interval the burst and gap metrics can be
calculated as follows.
// Calculate additional transition counts.
c31 = c13;
c32 = c23;
ctotal = c11 + c14 + c13 + c22 + c23 + c31 + c32 + c33;
// Calculate burst and densities.
p32 = c32 / (c31 + c32 + c33);
if((c22 + c23) < 1) {
p23 = 1;
}
else {
p23 = 1 - c22/(c22 + c23);
}
burst_density = 256 * p23 / (p23 + p32);
gap_density = 256 * c14 / (c11 + c14);
// Calculate burst and gap durations in ms
m = frameDuration_in_ms * framesPerRTPPkt;
gap_length = (c11 + c14 + c13) * m / c13;
burst_length = ctotal * m / c13 - lgap;
/* calculate loss and discard rates */
loss_rate = 256 * loss_count / ctotal;
discard_rate = 256 * discard_count / ctotal;
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Acknowledgments
We thank the following people: Colin Perkins, Steve Casner, and Henning Schulzrinne for their considered guidance; Sue Moon for helping foster collaboration between the authors; Mounir Benzaid for drawing our attention to the reporting needs of MLDA; Dorgham Sisalem and Adam Wolisz for encouraging us to incorporate MLDA block types; and Jose Rey for valuable review of the SDP Signaling section.Contributors
The following people are the authors of this document: Kevin Almeroth, UCSB Ramon Caceres, IBM Research Alan Clark, Telchemy Robert G. Cole, JHU Applied Physics Laboratory Nick Duffield, AT&T Labs-Research Timur Friedman, Paris 6 Kaynam Hedayat, Brix Networks Kamil Sarac, UT Dallas Magnus Westerlund, Ericsson The principal people to contact regarding the individual report blocks described in this document are as follows: sec. report block principal contributors ---- ------------ ---------------------- 4.1 Loss RLE Report Block Friedman, Caceres, Duffield 4.2 Duplicate RLE Report Block Friedman, Caceres, Duffield 4.3 Packet Receipt Times Report Block Friedman, Caceres, Duffield 4.4 Receiver Reference Time Report Block Friedman 4.5 DLRR Report Block Friedman 4.6 Statistics Summary Report Block Almeroth, Sarac 4.7 VoIP Metrics Report Block Clark, Cole, Hedayat The principal person to contact regarding the SDP signaling described in this document is Magnus Westerlund.
References
Normative References
[1] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [2] Crocker, D., Ed. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", RFC 2234, November 1997. [3] ETSI, "Quality of Service (QoS) measurement methodologies", ETSI TS 101 329-5 V1.1.1 (2000-11), November 2000. [4] Handley, M. and V. Jacobson, "SDP: Session Description Protocol", RFC 2327, April 1998. [5] Hovey, R. and S. Bradner, "The Organizations Involved in the IETF Standards Process", BCP 11, RFC 2028, October 1996. [6] ITU-T, "The E-Model, a computational model for use in transmission planning", Recommendation G.107, January 2003. [7] Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA Considerations Section in RFCs", BCP 26, RFC 2434, October 1998. [8] Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with the Session Description Protocol (SDP)", RFC 3264, June 2002. [9] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", RFC 3550, July 2003. [10] TIA/EIA-810-A Transmission Requirements for Narrowband Voice over IP and Voice over PCM Digital Wireline Telephones, December 2000.Informative References
[11] Adams, A., Bu, T., Caceres, R., Duffield, N.G., Friedman, T., Horowitz, J., Lo Presti, F., Moon, S.B., Paxson, V. and D. Towsley, "The Use of End-to-End Multicast Measurements for Characterizing Internal Network Behavior", IEEE Communications Magazine, May 2000. [12] Baugher, McGrew, Oran, Blom, Carrara, Naslund and Norrman, "The Secure Real-time Transport Protocol", Work in Progress.
[13] Caceres, R., Duffield, N.G. and T. Friedman, "Impromptu measurement infrastructures using RTP", Proc. IEEE Infocom 2002. [14] Clark, A.D., "Modeling the Effects of Burst Packet Loss and Recency on Subjective Voice Quality", Proc. IP Telephony Workshop 2001. [15] Handley, M., Perkins, C. and E. Whelan, "Session Announcement Protocol", RFC 2974, October 2000. [16] Reynolds, J., Ed., "Assigned Numbers: RFC 1700 is Replaced by an On-line Database", RFC 3232, January 2002. [17] Schulzrinne, H., Rao, A. and R. Lanphier, "Real Time Streaming Protocol (RTSP)", RFC 2326, April 1998. [18] Sisalem D. and A. Wolisz, "MLDA: A TCP-friendly Congestion Control Framework for Heterogeneous Multicast Environments", Proc. IWQoS 2000.
Authors' Addresses
Kevin Almeroth Department of Computer Science University of California Santa Barbara, CA 93106 USA EMail: almeroth@cs.ucsb.edu Ramon Caceres IBM Research 19 Skyline Drive Hawthorne, NY 10532 USA EMail: caceres@watson.ibm.com Alan Clark Telchemy Incorporated 3360 Martins Farm Road, Suite 200 Suwanee, GA 30024 USA Phone: +1 770 614 6944 Fax: +1 770 614 3951 EMail: alan@telchemy.com Robert G. Cole Johns Hopkins University Applied Physics Laboratory MP2-S170 11100 Johns Hopkins Road Laurel, MD 20723-6099 USA Phone: +1 443 778 6951 EMail: robert.cole@jhuapl.edu Nick Duffield AT&T Labs-Research 180 Park Avenue, P.O. Box 971 Florham Park, NJ 07932-0971 USA Phone: +1 973 360 8726 Fax: +1 973 360 8050 EMail: duffield@research.att.com
Timur Friedman Universite Pierre et Marie Curie (Paris 6) Laboratoire LiP6-CNRS 8, rue du Capitaine Scott 75015 PARIS France Phone: +33 1 44 27 71 06 Fax: +33 1 44 27 74 95 EMail: timur.friedman@lip6.fr Kaynam Hedayat Brix Networks 285 Mill Road Chelmsford, MA 01824 USA Phone: +1 978 367 5600 Fax: +1 978 367 5700 EMail: khedayat@brixnet.com Kamil Sarac Department of Computer Science (ES 4.207) Eric Jonsson School of Engineering & Computer Science University of Texas at Dallas Richardson, TX 75083-0688 USA Phone: +1 972 883 2337 Fax: +1 972 883 2349 EMail: ksarac@utdallas.edu Magnus Westerlund Ericsson Research Ericsson AB SE-164 80 Stockholm Sweden Phone: +46 8 404 82 87 Fax: +46 8 757 55 50 EMail: magnus.westerlund@ericsson.com
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