Internet DRAFT - draft-wenger-avt-rtcp-feedback
draft-wenger-avt-rtcp-feedback
INTERNET-DRAFT Stephan Wenger
draft-wenger-avt-rtcp-feedback-02.txt TU Berlin
Joerg Ott
Universitaet Bremen TZI
2 March, 2001
Expires September 2001
RTCP-based Feedback: Concepts and Message Timing Rules
Status of this Memo
This document is an Internet-Draft and is in full conformance with all
provisions of Section 10 of RFC 2026. Internet-Drafts are working
documents of the Internet Engineering Task Force (IETF), its areas, and
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Abstract
Real-time media streams are not resilient against packet losses. RTP
[1] provides all the necessary mechanisms to restore ordering and
timing to properly reproduce a media stream at the recipient. RTP
also provides continuous feedback about the overall reception quality
from all receivers -- thereby allowing the sender(s) in the mid-term
(in the order of several seconds to minutes) to adapt their coding
scheme and transmission behavior to the observed network QoS.
However, except for a few payload specific mechanisms [2], RTP makes
no provision for timely feedback that would allow a sender to repair
the media stream immediately: through retransmissions, retro-active
FEC, or media-specific mechanisms such as reference picture
selection.
This document specifies a modification to the algorithm for
scheduling RTCP packets in order to allow occasional timely feedback
to events observed by a receiver (such a lost packets). The message
format for RTCP-based feedback is defined in a companion document
[7].
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1. Introduction
Real-time media streams are not resilient against packet losses. RTP
[1] provides all the necessary mechanisms to restore ordering and
timing present at the sender to properly reproduce a media stream at
a recipient. RTP also provides continuous feedback about the overall
reception quality from all receivers -- thereby allowing the
sender(s) in the mid-term (in the order of several seconds to
minutes) to adapt their coding scheme and transmission behavior to
the observed network QoS. However, except for a few payload specific
mechanisms [2], RTP makes no provision for timely feedback that would
allow a sender to repair the media stream immediately: through
retransmissions, retro-active FEC, or media-specific mechanisms such
as reference picture selection.
Current mechanisms available with RTP to improve error resilience
include audio redundancy coding [3], video redundancy coding [4],
RTP-level FEC [5], and general considerations on more robust media
streams transmission [6]. Particularly in small groups, however,
virtually all kinds of all types of real-time media streams could
benefit from a mechanism that would enable a sender to perform media
stream repair -- including but not limited to audio, video, DTMF, and
text chat streams. In some case of networks with acceptable round-
trip times but scarce bandwidth, occasional retransmissions may be
much preferred over continuous transmission of redundant information.
For example, predictive video coding is not loss resilient. Any loss
of coded data leads to annoying artifacts not only in the reproduced
picture in which the loss occurred, but also in subsequent pictures.
Error resilience can be achieved by spending bits to convey redundant
information using source coding based mechanisms or transport based
mechanisms. This can be done without the use of any feedback between
the decoder(s) and the encoder. Similar consideration apply to
protecting e.g. DTMF (and other tones) carried in an RTP stream [9].
Alternatively, where applicable, receivers can inform the sender
through a feedback channel about a loss situation, and the sender can
react accordingly. This approach provides better media quality and
is more efficient with respect to the bandwidth used by the sender to
achieve a given media quality. However, using feedback mechanisms is
limited to certain application scenarios identified by encoder
characteristics, delay constraints, and/or the number of recipients.
This memo specifies a profile based upon [1] and [10] with enhanced
rules for sending receiver reports to support feedback transmission
reflecting the need for very low delay for conveying feedback, which
is necessary to make them efficient (or workable at all). Immediate
Feedback messages (FB messages) and Early Receiver Reports (Early
RRs) and algorithms are specified that allow for low delay in small
multicast groups, but prevent network flooding in larger ones.
Special consideration is given to point-to-point scenarios.
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In addition, this memo gives some consideration to specific
application scenarios are the respective feedback requirements, at
the moment focusing on predictive video coding.
A companion document [7] discusses various types of general purpose
feedback information (also allowing for extensions specific to
certain media payload) and defines an RTCP packet format to transmit
FBs in an RTP environment. It can be used in conjunction with all
payload specifications for predictive video coding schemes currently
available for RTP.
2. Motivation
2.1 Example: Predictive Video Coding
2.1.1 Video Encoder-decoder synchronicity
Most current video coding schemes for compressed video, such as the
ITU-T H.261 and H.263 and ISO/IEC MPEG[124] employ a mechanism known
as Inter Picture Prediction. Each picture is divided into
macroblocks of uniform size. For each macroblock, one or more
motion vectors may be identified and transmitted. The residual
signal after motion compensation is DCT-transformed, quantized,
entropy coded, and transmitted as well. The encoder reconstructs,
based on this information, a so-called reference picture, which is
used to perform the motion compensation and residual signal coding
steps for the subsequent picture. Since the reference picture is
generated using only such information that is also available at the
decoder, the reference picture is identical to the reconstructed
picture at the decoder. Having identical reference pictures at the
encoder and decoder is referred to as encoder-decoder-synchronicity.
Whenever data is damaged or lost on the way between the encoder and
the decoder, the reconstructed picture at the decoder is no more
identical with the encoder's reference picture -- the encoder-decoder
synchronicity is lost.
Any loss of the encoder-decoder synchronicity results in annoying
artifacts at the decoder. Because the prediction of subsequent
pictures in the decoder is based on a damaged reference picture, the
annoying artifacts are present not only in the picture in which the
loss occurred; they propagate to all subsequent pictures, until,
through source coding based mechanisms, the encoder-decoder
synchronicity is restored. Therefore, the goal of systems employing
predictive video coding in a lossy environment must be to keep the
encoder-decoder synchronicity, or, if this is not possible, to regain
that synchronicity as quickly as possible.
2.1.2. Non-feedback based mechanisms
Avoiding the loss of the encoder-decoder synchronicity corresponds to
avoiding the loss of coded picture data. Such a task can be
performed on the transport layer. In RTP environments, the use of
packet-based FEC is a good example for such a technique. (The use of
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TCP or reliable multicast as the transport for media streams would be
an even better one but is inappropriate for low-delay (interactive)
real-time systems.) FEC schemes, interleaving, and other means for
repairing real-time media streams may also add additional delay and
significant bit rate overhead without being able to guarantee
compensation of virtually all packet losses.
Once the encoder-decoder synchronicity is lost, only source coding
oriented mechanisms can help to regain it. One common way is to send
a non-predictively coded picture (known as Intra picture). Intra
pictures have the disadvantage of being several times bigger than
predictively coded pictures (Inter pictures). Therefore, sending
Intra pictures has negative implications both on the bandwidth and
(in bandwidth limited environments) delay. Another way is to use
Intra macroblock refresh. Here, certain parts of the picture (those
affected by a packet loss) are coded non-predictively in order to
resynchronize the encoder and decoder over time. Intra macroblock
refresh has better delay characteristics then full Intra pictures
because the picture size can be kept constant, but is less efficient
in terms of bit rate/distortion than full Intra pictures. More
sophisticated means such as Reference Picture Selection (RPS) are
also available in modern video coding standards.
Systems not employing feedback channels may use any combination of
the mechanisms described above to add error resilience -- at the cost
of added bit rate and, sometimes, added delay. The number of
additional bits spent for error resilience can be adapted using the
long-term packet loss rate information in the RTCP receiver reports.
But, even when using such adaptive means, it is still likely that
systems spend many more bits then theoretically necessary to achieve
error resilience in order to be on the safe side. Plus, as regular
RTCP feedback is aimed at longer terms, reactivity to sudden losses
is limited. In all practical applications today this means that
fewer bits are available for non redundant picture data, and hence
the overall picture quality suffers.
2.1.3 Feedback based systems
Feedback-based systems try to avoid spending too many bits for
redundant information by informing the encoder about a loss situation
at the decoder(s). The encoder can then react accordingly and spend
redundant bits only when needed possibly only for the part of the
picture that was effected by the loss -- thereby reducing the number
of redundant bits and leaving more bits for useful information. As a
result, a higher reproduced picture quality can generally be expected
when feedback channels are available.
Similar to the observations of section 2.1.2, transport and source
coding based mechanisms can be distinguished that react on loss
situations reported by feedback.
Transport based systems employing feedback react media unaware, by
re-transmitting lost packets. TCP is a good example for a protocol
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following such a scheme. Transport-based feedback in real-time
and/or multicast environments is a complex matter and subject of a
lot of engineering and research in and outside of the IETF. This
specification is not concerned with pure transport-based feedback.
Source coding based mechanisms may react upon the arrival of a
feedback message indicating a loss situation by adding bits that
restore, or at least make an effort to restore, the encoder-decoder
synchronicity. This process has to be performed by a real-time
encoder. However, schemes were reported, that allow the use of
feedback also for non-real-time encoders by storing multiple
representations of the same data (e.g. Inter and Intra coded), and
dynamically switching between those representations.
Several types of feedback messages, called Feedback Messages or FB
messages, can be defined for such a case. An FB message can be as
simple as a Boolean condition, indicating for example the loss of a
full picture (and, therefore, the need of a full Intra picture
transmission). Other feedback messages may contain more complex
information such as information about the damage of a spatial region
of the picture. A special form consists of a message the format and
semantics of which are not known at the transport level, because they
are defined in the video codec standards.
2.2 Feedback Messages
Most FB messages contain negative acknowledge information, indicating
an erroneous situation at the decoder. In others, the nature of the
acknowledge (positive, negative, or both) is part of the feedback
message itself. When used in multicast environments, positive
acknowledge must not be used.
This document assumes that feedback messages are transmitted using
RTCP packets. RTCP messages from the receivers to the sender cannot
be sent at any possible time, in order to prevent traffic explosion
in case of large multicast groups. Instead, the bit rate for all
RTCP messages of all receivers together has to obey a maximum
fraction of the total RTP session bit rate, yielding a very limited
bit rate budget for a single receiver when having a large multicast
group. This, in turn, leads to an increased average delay when the
size of the receiving multicast group grows. (see section 6 of [1]
for details)
This specification defines an algorithm that adheres to the bit rate
limitations for the feedback channel on the long term, but allows
short-term overdrafting for any receiver (but not all of them
simultaneously). Thus, the algorithm allows for better real-time
performance then the one specified in [1]. Traffic explosion in such
cases in which many receivers identify a picture damage
simultaneously is prevented by dithering.
As this specification assumes a sender that has full control over its
transmission bit rate (e.g. a real-time encoder), there is no scaling
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problem on the forward channel. Any reaction to negative feedback
generates additional bits, which have to be conveyed but this is
taken from the sender's total bit rate budget. The encoder can take
this into account by, for example, changing the encoding mode, packet
size, and so forth. The sender is also free to simply ignore
feedback messages. Adjusting the tradeoff between the reproduced
media quality of all receivers of a multicast group and the amount of
additional repair traffic is a media-dependent, very complex task and
is not covered in this specification.
Finally, frequent RTCP-based feedback messages may provide additional
input to the sender(s)'s congestion control algorithms and thus
improve its reactivity towards network congestion.
Feedback messages as well as sender and receiver behavior are to be
specified in separate documents (such as [7]). Such specifications
need to consider that, frequently, packet loss is an indication of
network congestion and thus define mechanisms for media-specific
congestion control in the presence of feedback as defined in this
memo.
2.3. Applications and Relationships to other Standards
This specification is based on RTCP, which implies its use in an RTP
environment. RTP itself is used in a variety of systems such as in
SIP- or H.323-based multimedia conferencing/telephony, SAP-announced
Mbone conferences, and RTSP-based media streaming.
As for the video codecs, there is currently a small set of standards
that are, for the purpose of this discussion, roughly comparable.
Many mechanisms for regaining encoder-decoder synchronicity are
applicable to all video codecs. Others require certain tools (such
as Reference Picture Selection, aka NEWPRED) that are available only
in certain versions of the standards, and/or optional tools whose use
must be negotiated prior to being used.
A few RTP payload specifications such as RFC 2032 [2] already define
a feedback mechanism for some of the coding algorithms considered in
this specification. An application capable of performing both
schemes MUST use the feedback mechanism defined in this
specification, although, for backward compatibility reasons, it MUST
also be capable to conform to the feedback scheme defined in the
respective RTP payload format, if this is required by that payload
format.
Also, audio, DTMF, and text streams could benefit from more immediate
feedback even though the redundancy payload formats work well for
these media.
All kinds of non-interactive media streams (such as RTSP-controlled
media streaming applications) could benefit significantly as without
interactivity there is more time available for media repair.
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2.4 Remarks on the size of the multicast group
This specification prevents traffic explosion on the feedback channel
in a very similar way as RTP does, with the exception of allowing
individual receivers to overdraft their bit rate budget from time to
time. This is necessary in order to allow for low delay, which is
needed by the algorithms reacting to Feedback messages.
This scaling, however, limits the usefulness of this mechanism in
multicast groups from a certain size upwards (where the size
threshold depends on a number of parameters including loss rate,
frame rate, number of packets per frame, and session bandwidth). The
maximum size of the multicast group is soft and also depends on
application requirements and is therefore not specified here.
Considerations on the multicast group sizes will be presented in
section 3.5.
2.5 Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [8]
3. Low delay RTCP Feedback
Two components constitute RTCP-based feedback as described in this
memo:
. Status reports are contained in SR/RR messages and are transmitted
at regular intervals as part of compound RTCP packets (which also
include SDES and possibly other messages); these status reports
provide an overall indication for the recent reception quality of a
media stream. RTP [1] define rules for the transmission of these
status reports.
. Feedback messages as defined in a companion document [7] that
indicate loss or reception of particular pieces of a media stream
(or provide some other form of rather immediate feedback on the
data received). Rules for the transmission of feedback messages
are newly introduced in this memo.
As discussed in [7], RTCP Feedback (FB) messages are just another
RTCP message type. Thus multiple FB messages may be combined in a
single RTCP packet. FB messages may be sent in full compound RTCP
packets along with SR/RR, SDES, and other RTCP messages. Or they may
be transmitted in minimal compound RTCP FB packets (which only
contain the RR/SR and an encryption prefix if necessary to reduce the
message size). RTCP packets that do not contain FB messages are
referred to as non-FB RTCP packets.
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3.1 Algorithm Outline
FB messages are part of the RTCP control streams and are thus subject
to the same bandwidth constraints as other RTCP traffic. This means
in particular that it may not be possible to report a packet loss at
a receiver immediately back to the sender. However, the value of
feedback given to a sender typically decreases over time -- in terms
of the media quality as perceived by the user at the receiving end
and/or the cost required to achieve media stream repair.
RTP [1] specifies rules when compound RTCP packets should be sent.
This specification modifies those rules in order to allow
applications to timely report media loss or reception events, since
most algorithms that use FB messages are very critical to the
feedback timing. See section 5 and following for a discussion of FB
messages and the impact of delay on the performance these FB types.
The modified algorithm can be outlined as follows: Normally, when no
FB messages have to be conveyed, compound RTCP packets are sent
following the rules of RTP [1]. If a receiver detects the need for
an FB message, the receiver first checks whether it has already seen
a corresponding FB message from any other receiver (which it can do
with all FB messages that are transmitted via multicast; for unicast
sessions, there is no such delay). If this is the case then the
receiver refrains from sending the FB message, and continues to
follow the regular RTCP sending schedule. If the receiver has not
yet seen a similar FB message from any other receiver, it checks
whether it has recently exceeded its RTCP bit rate budget to transmit
another FB message (without waiting for its regularly scheduled RTCP
transmission time). Only if this is not the case, it sends the FB
message, after waiting a short, random dithering interval period (in
case of multicast).
FB messages are sent as part of minimal compound RTCP packets . Full
compound RTCP packet are interspersed as per [1] in regular intervals
of at least five seconds.
3.2 Modes of Operation
RTCP-based feedback may operate in one of three modes (figure 1):
a) Immediate feedback mode: the group size is below a certain
threshold (the FB threshold) which gives each receiving party
sufficient bandwidth to transmit the feedback traffic for the
intended purpose. This means, for each receiver there is enough
bandwidth to report each event it is supposed/expected to by means
of a virtually "immediate" Early RTCP packet.
The group size threshold is a function of a number of parameters
including (but not necessarily limited to) the type of feedback
used (e.g. ACK vs. NACK), bandwidth, packet rate, packet loss
probability, media type, codec, and -- again depending on the type
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of FB used -- the (worst case or observed) frequency of events to
report (e.g. frame received, packet lost).
A special case of this is the ACK mode (where positive
acknowledgements are used to confirm reception of data) which is
restricted to point-to-point communications.
b) In Early RTCP mode, the group size and other parameters no longer
allow each receiver to react to each event that would be worth (or
needed) to report. But feedback can still be given sufficiently
often so that it allows the sender to adapt the media stream and
thereby increase the overall reproduced media quality.
c) From some group size upwards, it is no longer useful to provide
feedback from individual receivers at all -- because of the time
scale in which the feedback could be provided and/or because in
large groups the sender(s) have no chance to react to individual
feedback anymore.
As the feedback algorithm described in this memo scales, there is no
need for an agreement on the precise values of the respective
"thresholds" within the group. Hence the borders between all these
modes are fluent.
ACK
feedback
V
:<- - - - NACK feedback - - - ->//
:
: Immediate ||
: Feedback mode ||Early RTCP mode Regular RTCP mode
:<=============>||<=============>//<=================>
: ||
-+---------------||---------------//------------------> group size
2 ||
Application-specific FB Threshold
= f(rate,loss,codec,...)
Figure 1: Modes of operation
The respective thresholds depend on a number of technical parameters
(of the codec, the transport, the feedback used, etc.) but also on
the respective application scenarios. Section 3.5 provides some
useful hints (but no complete precise calculations) on estimating
these thresholds.
3.3 Definitions
a) Let the media stream be transmitted at a (roughly) constant packet
rate f (in packets per second). This results in an average
inter-packet interval of tau=1/f.
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b) Let T_rtt be the maximum round trip time as measured by RTCP
(if available to the receiver). Note that this may be asymmetric.
d) Let t_rr and t_(rr-1) be the time for the next (last) scheduled
RTCP RR transmission calculated prior to reconsideration.
Let T_rr + t_(rr-1) = t_rr. (In RTP [1] these are termed tp, tn,
respectively).
d) Let t_e be the time for which a feedback packet is scheduled.
e) Let t_dither_max be the maximum interval for which an RTCP
feedback packet may be additionally delayed (to prevent
implosions).
f) Let T_fd be the delay for the feedback message that a certain
packet P caused to return to the sender after reception of P.
g) Let S be the number of active senders in the RTP session.
h) Let N be the current estimate of the number of receivers in the
RTP session.
The feedback situation for an event to report at a receiver is
depicted in figure 2 below. At time t0, such an event (e.g. a packet
loss is detected at the receiver. The receiver decides -- based upon
current T_rtt, group size, and other (application-specific)
parameters -- that a feedback message shall be sent back to the
sender.
To avoid an implosion of immediate feedback packets, the receiver
delays transmission of the compound feedback packet by a random
amount T_fd (with the random number evenly distributed in the
interval [0, T_dither_max]. Transmission of the compound RTCP packet
is then scheduled for t_e = t0 + T_fd.
The T_dither_max parameter is chosen based upon the group size, the
RTCP bandwidth constraints, and, if available, the round-trip time.
In addition, the receiver may take into account a number of other
parameters (such as the estimated round-trip time, the type of
feedback to be provided) to possibly extend the upper bound for the
feedback while ensuring that the feedback information still will make
sense when it reaches the sender.
If a compound RTCP feedback packet is scheduled, the time slot for
the next scheduled compound RTCP packet is updated accordingly to a
new t_rr.
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event to
report
detected
|
| RTCP feedback
vXXXXXXXXXXXXXXXXXXXX ) )
|---+--------+-------------+-----+------------| |--------+--------->
| | | | ( ( |
| t0 te |
t_(rr-1) t_rr
\_______ ________/
\/
T_dither_max
Figure 2: Event report and parameters for Early RTCP scheduling
3.4 Early RTCP Algorithm
Assume an active sender S0 (out of S senders) and a number N of
receivers with R being one of these receivers.
Assume further that R has verified that using feedback mechanisms is
reasonable at the current constellation (which is highly application
specific and hence not specified in this memo).
Then, the following rules apply to transmitting a Feedback Messages
as minimal compound RTCP packet:
Initially, R sets allow_early := TRUE.
At a point in time t0, R has transmitted the last RTCP RR packet at
t_(rr-1) and has scheduled the next transmission (prior to
reconsideration) for t_rr.
Now R detects the need to transmit a feedback message (e.g. because a
media "unit" needs to be ACKed or NACKed) at time t0.
R first checks whether there is still a feedback packet waiting for
transmission. If so, the new feedback message is appended to the
packet and the increased RTCP packet size is updated in the RTCP
bandwidth calculation (which may later lead to an adjustment of
t_rr); the schedule for the waiting RTCP feedback packet remains
unchanged.
If no feedback message is already awaiting transmission a new
(minimal) compound RTCP feedback message is created and the interval
T_dither_max is chosen as follows:
i) If the session is a unicast session (group size = 2) then
T_dither_max := 0.
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ii) If the receiver has an RTT estimate to the originator of the
media unit to provide feedback about, then
/ T_rtt/2 if T_rtt/2 > 10ms
T_dither_max := <
\ 10ms otherwise.
iii) If the receiver does not have an RTT estimate to the originator,
then
/ T_rr/2 if T_rr/2 < 100ms
T_dither_max := <
\ 100ms otherwise.
(Note: These values are *still* open to discussion.)
(Note that application-specific feedback considerations may make it
worth while to increase T_dither_max beyond this value.)
Then, R checks whether its next regularly scheduled RTCP packet is
within the time bounds for the RTCP FB (t_e + T_dither_max > t_rr).
If so, no Early RTCP is scheduled; instead the FB message is appended
to the regular RTCP packet and the RTCP bandwidth calculation is
updated to reflect the additional RTCP size. The updated bandwidth
calculation may result in a slightly increased t_rr (=t_rr') but,
even if t_rr' > t_e + T_dither_max, this does not change the updated
transmission time t_rr'.
(Q: if the FB is piggybacked onto a regularly scheduled RTCP RR
message but the same or a superset of the feedback information is
received from another receiver, should the FB then be removed from
the compound RR/FB and its transmission time be revised again from
t_rr' to t_rr as calculated before?)
Otherwise, R MUST check whether it is allowed to transmit an Early
RTCP packet (allow_early == TRUE).
If so, R schedules an Early RTCP packet for t_e = t0 + RND *
T_dither_max with the RND function evenly distributed between 0
and 1.
If R receives an RTCP feedback packet (indicating the same or a
superset of the feedback information R wanted to transmit) before
t_e is reached, the FB information is discarded and the
transmission schedule for the next RR packet is reset to t_rr as
calculated before.
Otherwise, when t_e is reached, R creates an RR, appends the FB
information, and transmits the RTCP packet. R then sets
allow_early := FALSE and recalculates t_rr := t_e + 2*T_rr. As
soon as R sends its next regularly scheduled RTCP RR
(at the new t_rr), it sets allow_early := TRUE again.
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If allow_early == FALSE then R checks the time for the next scheduled
RR: if t_rr - t0 < T_dither_max then R creates an FB message for
transmission along with the RTCP packet at a then slightly modified
t_rr' (see above). Otherwise, R does not send an RTCP feedback
message at all.
In regular RTCP intervals as specified by [1] (i.e. at most every
five seconds), a full compound RTCP packet is sent (which may also
contain a feedback message if one has been created according to the
above rules and scheduled for transmission along the full compound
RTCP message).
The E bit in the message header [7] is used upon reception to detect
whether this RTCP feedback message was sent as Early RTCP or not.
Hence, a feedback message that is sent as an Early RTCP packet MUST
set the E bit in the message header to "1". Feedback messages piggy-
backed on regularly scheduled RTCP packets will MUST set the E bit to
"0".
3.5 Considerations on the Group Size
This section intends to give some brief guidelines to the group sizes
at which the various feedback modes may be used.
3.5.1 ACK mode
The group size MUST be exactly two participants, i.e. point-to-point
communications. Unicast addresses SHOULD be used in the session
description.
For unidirectional as well as bi-directional communication between
two parties, 2.5% of the RTP session bandwidth are available for
feedback. Assuming a ratio of 1:10 for minimal to full compound RTCP
packets, at 64kbit/s, a receiver can report 2.5 events per second
back to the sender, at 256kbit/s 10 events and so forth.
From 768kbit/s upwards, a receiver would be able to acknowledge each
individual frame (not packet!) in a 30 fps video stream.
ACK strategies have to be defined accordingly to work with these
bandwidth limitations.
3.5.2 NACK mode
Negative acknowledgements (or similar types of feedback) have to be
used for all groups larger than two.
Whether or not the use of Immediate or Early RTCP packets should be
considered depends upon a number of parameters including session
bandwidth, codec, special type of feedback, number of senders and
receivers, among many others.
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The crucial parameters -- to which all of the above can be reduced --
is the allowed minimal interval between two RTCP reports and the
number of events that presumably need reporting per time interval.
The minimum interval is derived from the available RTCP bandwidth and
the expected average size of an RTCP packet. The number events to
report e.g. per second may be derived from the packet loss rate and
sender's rate of transmitting packets. From these two values, the
allowable group size for the Immediate feedback mode can be
calculated.
The upper bound for the Early RTCP mode then solely depends on the
acceptable quality degradation, i.e. how many events per time
interval may go unreported.
Example: If a 256kbit/s video with 30 fps is transmitted through a
network with an MTU size of some 1500 bytes, then, in most cases,
each frame would fit in its own packet leading to a packet rate of 30
packets per second. If 5% packet loss occurs in the network (equally
distributed, no inter-dependence between receivers), then each
receiver will have to report 3 packets lost each two seconds.
Assuming a single sender and more then three receivers yields 3.75%
of the RTCP bandwidth allocated to the receivers and thus 9.6kbit/s.
Assuming further a size of 100 bytes for the average compound RTCP
packet allows 12 RTCP packets to be sent per second or 24 in two
seconds. If every receiver needs to report three packets, this
yields a maximum group size of 8 receivers if all loss events shall
be reported. The rules for transmission of immediate RTCP packets
should provide sufficient flexibility for most of this reporting to
occur in a timely fashion.
Extending this example to determine the upper bound for Early RTCP
mode leads to the following considerations: assume that the
underlying coding scheme and the application (as well as the tolerant
users) allow in the order of one loss without repair per two seconds.
Thus the number of packets to be reported by each receiver decreases
to two per two seconds second and increases the group size to 12.
Assuming further that some number of packet losses are correlated,
feedback traffic is further reduced and group sizes of some 15 to 20
can be reasonably well supported using Early RTCP mode.
3.6 Summary of decision steps
3.6.1 General Hints
Before even considering whether or not to send RTCP feedback
information an application has to determine whether this mechanism is
applicable:
1) An application has to decide whether -- for the current ratio of
packet rate with the associated (application-specific) maximum
feedback delay and the currently observed round-trip time (if
available) -- feedback mechanisms can be applied at all.
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This decision may obviously be based upon (and dynamically revised
following) regular RTCP reception statistics.
2) The application has to decide whether -- for a certain observed
error rate, assigned bandwidth, frame rate, and group size -- (and
which) feedback mechanisms can be applied.
Regular RTCP provides valuable input to this step, too.
3) If these tests pass, the application has to follow the rules for
transmitting Early RTCP packets or regularly scheduled RTCP
packets with piggybacked feedback.
3.6.2 Session Description Attributes
A number of additional SDP parameters may be used to describe a
session. These are defined as session level and/or media level
attributes:
3.6.1.1 RTCP Feedback
a=rtcp-fb: {"ack"|"nack"|extension} params
This attribute is used to indicate the feedback (to be) supported by
the sender. "ack" MUST only be used if the media session is allowed
to operate in ACK mode as defined in 3.6.1.2.
It is up to the recipients whether or not they send feedback
information and up to the sender(s) to make use of feedback provided.
3.6.1.2 Unicasting
If an m= line in the SDP describing a session indicates unicast
addresses for a particular media type (and does not operate in multi-
unicast mode with all recipients listed explicitly but still
addressed via unicast), the RTCP feedback MAY operate in ACK feedback
mode.
4. Format of RTCP Feedback messages
The general format of the FB messages are defined in [7].
5. Security Considerations
RTP packets transporting information with the proposed payload for
mat are subject to the security considerations discussed in the RTP
specification [1]. This implies that confidentiality of the media
streams is achieved by encryption.
If the entire stream (extension data and AU data) is to be secured
and all the participants are expected to have the keys to decode the
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entire stream, then the encryption is performed in the usual manner,
and there is no conflict between the two operations (encapsulation
and encryption).
The need for a portion of stream (e.g. extension data) to be
encrypted with a different key, or not to be encrypted, would require
application level signaling protocols to be aware of the usage of
the XT field, and to exchange keys and negotiate their usage on the
media and extension data separately.
6. Acknowledgements
Large parts of the syntax and the text concerned with RPS and NEWPRED
were borrowed from an early I-D from Fukunaga et. al. that was
concerned with MPEG-4 ES packetization.
7. Full Copyright Statement
Copyright (C) The Internet Society (2001). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works.
However, this document itself may not be modified in any way, such as
by removing the copyright notice or references to the Internet Soci-
ety or other Internet organizations, except as needed for the purpose
of developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be fol-
lowed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MER-
CHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE."
8. Authors' Addresses
Stephan Wenger (stewe@cs.tu-berlin.de)
TU Berlin
Sekr. FR 6-3
Franklinstr. 28-29
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D-10587 Berlin
Germany
Joerg Ott (jo@tzi.uni-bremen.de)
Universitaet Bremen TZI
MZH 5180
Bibliothekstr. 1
D-28359 Bremen
Germany
4. Bibliography
[1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP -
A Transport Protocol for Real-time Applications," Internet
Draft, draft-ietf-avt-rtp-new-08.txt, Work in Progress, July
2000.
[2] T. Turletti and C. Huitema, "RTP Payload Format for H.261 Video
Streams, RFC 2032, October 1996.
[3] C. Perkins, I. Kouvelas, O. Hodson, V. Hardman, M. Handley, J.C.
Bolot, A. Vega-Garcia, and S. Fosse-Parisis, "RTP Payload for
Redundant Audio Data," RFC 2198, September 1997.
[4] C. Bormann, L. Cline, G. Deisher, T. Gardos, C. Maciocco, D.
Newell, J. Ott, G. Sullivan, S. Wenger, and C. Zhu, "RTP Payload
Format for the 1998 Version of ITU-T Rec. H.263 Video (H.263+),"
RFC 2429, October 1998.
[5] C. Perkins and O. Hodson, "2354 Options for Repair of Streaming
Media," RFC 2354, June 1998.
[6] J. Rosenberg and H. Schulzrinne, "An RTP Payload Format for
Generic Forward Error Correction,", RFC 2733, December 1999.
[7] S. Fukunaga, N. Sato, K. Yano, A. Miyazaki, K. Hata, R.
Hakenberg, C. Burmeister, "Low Delay RTCP Feedback Format,"
Internet Draft draft-fukunaga-low-delay-rtcp-02.txt, Work in
Progress, February 2001.
[8] S. Bradner, "Key words for use in RFCs to Indicate Requirement
Levels," RFC 2119, March 1997.
[9] H. Schulzrinne and S. Petrack, "RTP Payload for DTMF Digits,
Telephony Tones and Telephony Signals," RFC 2833, May 2000.
[10] H. Schulzrinne and S. Casner, " RTP Profile for Audio and Video
Conferences with Minimal Control," Internet Draft draft-ietf-
avt-profile-new-09.txt, July 2000.
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Appendix A: Considerations On Video
This section of this memo covers feedback messages for a Picture Loss
Indication (PLI), Slice Loss Indication (SLI), and Reference Picture
Selection Indication (RPSI). PLI indicates the loss of a full
picture and roughly corresponds to the Fast Intra Request known from
H.320 systems and from RFC 2032 (H261 packetization). Algorithms
using SLI can be found under the acronym Automatic Repeat Request
(ARQ) in the signal processing literature. Reference Picture
Selection, aka NEWPRED, is available in certain profiles of MPEG-4
(version 2 and later) and as an optional mode in H.263 (version 2 and
later). The packet format specified in this document is open to
extensions so that future feedback mechanisms can easily be
integrated.
All these messages use the payload specific feedback format as
defined in [7], using PT=PSFB and the FMT field to further
distinguish between the three subtypes. These messages are defined
for payload types indicating H.263 and MPEG-4.
Note that the Bit 00 of the first (counting from 1) 32-bit word in
the messages described below is placed in Bit 08 of the fourth
(counting from 1) 32-bit word of the payload type specific feedback
message.
A.1 Message Type 1: Picture Loss Indication (PLI)
A.1.1 Semantics
With the Picture Loss Indication message a decoder informs the
encoder about the loss of one or more full pictures
A.1.2 Format
PLI does not require parameters. Therefore, the length field MUST be
0, and there MUST NOT be Feedback Control Information.
A.1.3 Timing Rules
The timing follows the rules outlined in section 3. In systems that
employ both PLI and other FB types it may be advisable to follow the
regular RTCP RR timing rules, since PLI is not as delay critical as
other FB types.
A.1.4 Remarks
PLI messages typically trigger the sending of full Intra pictures.
Intra Pictures are several times larger then predicted (Inter)
pictures. Their size is independent of the time they are generated.
In most environments, especially when employing bandwidth-limited
links, the use of an Intra picture implies an allowed delay that is a
significant multitude of the typical frame duration. An example: If
the sending frame rate is 10 fps, and an Intra picture is assumed to
be 10 times as big as an Inter picture (not an unrealistic
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assumption, see [] for details), then a full second of latency has to
be accepted. In such an environment there is no need for a
particular short delay in sending the feedback message. Hence
waiting for the next possible time slot allowed by RFC1889bis RTCP
timing rules does not negatively influence system performance.
A.2 Message Type 2: Slice Lost Indication
A.2.1 Semantics
With the Slice Lost Indication a decoder can inform an encoder that
it was unable to decode one, or several consecutive, macroblocks.
The encoder can take appropriate action in order to re-synchronize
encoder and decoder by means of its choice, typically by sending the
lost macroblocks in Intra mode. This feedback message SHALL NOT be
used for video codecs with non-uniform, dynamically changeable
macroblock sizes such as H.263 with enabled Annex Q. In such a case,
an encoder cannot always identify the corrupted spatial region.
A.2.2 Format
When FBT indicates a Slice Lost Indication, then there is one
additional UCI field the content of which is in the following format:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| First | Number | TR |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
First: 13 bits
The macroblock (MB) address of the first lost macroblock. The MB
numbering is done such that the macroblock in the upper left corner
of the picture is considered macroblock number 1 and the number for
each macroblock increases from left to right and then from top to
bottom in raster-scan order (such that if there is a total of N
macroblocks in a picture, the bottom right macroblock is considered
macroblock number N).
Number: 13 bits
The number of lost macroblocks, in scan order as discussed above.
TR: 6 bits
The six least significant bits of the Temporal Reference of the
picture.
A.2.3 Timing Rules
The efficiency of algorithms using the Slice Lost Indication is
reduced greatly when the Indication is not transmitted in a timely
fashion. Motion compensation propagates corrupted pixels that are
not reported as being corrupted. Therefore, the use of the algorithm
discussed in section 3 is highly recommended.
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Constraints on T_dither_max to be discussed.
A.2.4 Remarks
The First field of the UCI defines the first macroblock of a picture
as 1 and not, as one could suspect, as 0. This was done to align
this specification with the comparable mechanism available in H.245.
The maximum number of macroblocks in a picture (2**13 or 8192)
corresponds to the maximum picture sizes of the ITU-T and ISO/IEC
video codecs. If future video codecs offer larger picture sizes
and/or smaller macroblock sizes, then an additional feedback message
has to be defined. The six least significant bits of the Temporal
Reference field are deemed to be sufficient to indicate the picture
in which the loss occurred.
Algorithms were reported that keep track of the regions effected by
motion compensation, in order to allow for a transmission of Intra
macroblocks to all those areas, regardless of the timing of the FB
[TBP.]. While, when those algorithms are used, the timing of the FB
is less critical then without, it has to be observed that those
algorithms correct large parts of the picture and, therefore, have to
transmit many for bits in case of delayed FBs.
A.3 Message Type 3: Reference Picture Selection Indication
A.3.1 Semantics
Modern video coding standards such as MPEG-4 visual version 2 or
H.263 version 2 allow the use of older reference pictures then the
most recent one. Typically, a first-in-first-out queue of reference
pictures is maintained. If an encoder has learned about a loss of
encoder-decoder synchronicity, a known-as-correct reference picture
can be used. As this reference picture is temporally further away
then usual, the resulting predictively coded picture will use more
bits.
Both MPEG-4 and H.263 define a binary format for the _payload_ of an
RPSI message that includes information such as the temporal ID of the
damaged picture and the size of the damaged region. This bit string
is typically small _- a couple of dozen bits -_, of variable length,
and self-contained, i.e. contains all information that is necessary
to perform reference picture selection.
Note that both MPEG-4 and H.263 allow the use of RPSI with positive
feedback information as well. That is, all corrected pictures are
reported. Any form of positive feedback MUST NOT be used when in a
multicast environment (reporting positive feedback about individual
reference pictures at RTCP intervals is not expected to be of much
use anyway). For point-to-point communication, positive feedback MAY
be used but, again, the bit rate budget of RTCP feedback will prevent
the use in most scenarios anyway.
A.3.2 Format
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When FB indicates an RPSI, then the length field is set to the number
of bits of the following bit string that contains the RPS
information. This bit string follows byte aligned in the UCI field.
Bit padding is used to achieve 32-bit word alignment of the UCI
message (and the whole packet).
A.3.3 Timing Rules
RPS is even more critical to delay then algorithms using SLI. This
is due to the fact that the older the RPS message is, the more bits
the encoder has to spend to achieve encoder-decoder synchronicity.
See [TBP.] for some information about the overhead of RPS for certain
bit rate/frame rate/loss rate scenarios.
Therefore, RPS messages should typically be sent as soon as possible,
employing the algorithm of section 3.
Constraints on T_dither_max to be discussed.
A.3.4 Remarks
TBD.
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