INTERNET-DRAFT Stephan Wenger draft-wenger-avt-rtcp-feedback-00.txt TU Berlin Joerg Ott Universitaet Bremen TZI July 14, 2000 Expires December 2000 RTCP-based Feedback for Predictive Video Coding 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 its working groups. Note that other groups may also distribute working documents as Internet-Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet- Drafts as reference material or to cite them other than as "work in progress." The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. 0. Open Issues 1) Should the draft limit itself to supporting feedback for video only or should it target a more general solution for feedback? At the moment, the draft covers only video. 2) Should the feedback be restricted to point-to-point scenarios or should we support (small group) multicast. At the moment, the draft is designed to scale to (small) group. 3) Feedback traffic explosion is prevented by a) dithering and b) damping. a) somewhat poses constraints on timely transmission of feedback. b) prevents that the encoder can learn about the _severeness_ of a loss problem (e.g. how many receivers have now a bad picture). This prevents adaptive encoder reaction based on the perceived quality of the whole group. At the moment, a) and b) are both to be used to be network friendly. Which mechanisms (besides flooding the network which we want to avoid) are conceivable to support an approach that is able to achieve a better perceived picture quality? Wenger/Ott Expires December 2000 [Page 1] Internet Draft July 14, 2000 4) Is the maximum number of MBs 8191 for SLI sufficient? Yes for MPEG- 1, MPEG-2 and ITU-T H.261, H.263. What about MPEG-4? 5) Should there be a special mode (possibly optimized for point-to-point communication) that allows UMs packets without RR (see section 3)? 6) RPS/NEWPRED also make use of positive acknowledgements. Obviously, this does inherently not scale to multicast. Should there be a point-to-point mode that allows positive ACKs? 7) We have not yet considered the use of layered codecs. When transporting each layer in its own RTP stream, everything should be ok. If not, then we can foresee problems. 8) Section 7 on NEWPRED needs more work (probably based on Fukunaga et. al draft). 9) Further work is needed on maximum group size estimation for using feedback and on more detailed guidelines on calculating the maximum dithering delay for Early RRs (T_dither_max) per UM type. 10) Further investigations are desirable for the Early RR/UM scheduling and damping and the relationship of Early RR/UM scheduling to regular RTCP report scheduling. 1. Abstract 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. Alternatively, where applicable, decoders can inform the encoder through a feedback channel about a loss situation, and the encoder can react accordingly. This approach provides better picture quality and is more efficient with respect to the bandwidth used by the encoder to achieve a given quality. However, using feedback mechanisms is limited to certain application scenarios identified by encoder characteristics, delay constraints, and/or the number of recipients. This document discusses various types of feedback information (called _upstream messages_, UMs) for predictive video coding and defines an RTCP packet format to transmit UMs in an RTP environment. It can be used in conjunction with all payload specifications for predictive video coding schemes currently available for RTP. To reflect the need for very low delay for the transmission of the UMs, which is necessary to make them efficient, the rules for sending receiver reports are enhanced to support Early Receiver Report (Early RRs) and an algorithm is specified that allows Wenger/Ott Expires December 2000 [Page 2] Internet Draft July 14, 2000 for low delay in small multicast groups, but prevents network flooding. 2. Introduction 2.1. 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. 2.2. 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 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. Wenger/Ott Expires December 2000 [Page 3] Internet Draft July 14, 2000 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. 2.3 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 2.2, 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 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. Wenger/Ott Expires December 2000 [Page 4] Internet Draft July 14, 2000 Source coding based mechanisms may react upon the arrival of an upstream 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 Upstream Messages or UMs, are defined in this specification. A UM 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. Most UMs 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 send 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 draft-ietf-avt-rtp-new-06.txt 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 draft-ietf-avt-rtp-new-06.txt. Traffic explosion in such cases in which many receivers identify a picture damage simultaneously is prevented by dithering. As this specification assumes a real-time encoder that has full control over its transmission bit rate, there is no scaling 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, sending fewer pictures per second, lower the quality and bit rate by changing quantization parameters and so forth. The sender is also free to simply ignore feedback messages. Wenger/Ott Expires December 2000 [Page 5] Internet Draft July 14, 2000 Adjusting the tradeoff between the reproduced picture quality of all receivers of a multicast group and the amount of bits spent for encoder-decoder re-synchronization is a very complex task and is not covered in this specification. This document currently 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. 2.4. 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. 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 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. 2.5 Remarks on the size of the multicast group This specification makes an attempt to prevent 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 UMs. This scaling, however, limits the usefulness of this mechanism in multicast groups from a certain size upwards (where the size Wenger/Ott Expires December 2000 [Page 6] Internet Draft July 14, 2000 threshold depends on a number of parameters including loss rate, frame rate). The maximum size of the multicast group is not specified here (which is soft and also depends on application requirements). The authors have done some rough calculations (for which it is too early to present them here in detail) that suggest that feedback is not expected to yield acceptable results for group sizes larger then 10 receivers (often less than five), assuming today's network conditions (RTT, loss rate) and common bit rates. 2.6 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 [xxx] 3. Low delay RTCP Feedback UMs 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. RFC1889bis (i.e. draft revision draft-ietf-avt-rtp-new-06.txt) specifies rules when RTCP receiver reports (RRs) should be sent. This specification modifies those rules in order to allow applications to timely report damaged pictures, since most algorithms that use UMs are very critical to the UM timing. See section 5 and following for a discussion of the impact of delay on the performance of each UM type. The modified algorithm can be outlined as follows: Normally, when no UMs have to be conveyed, RRs are sent following the rules of RFC1889bis. If one or more receivers detect the need for an UM, the receiver first checks whether it has already seen a corresponding UM from any other receiver (which it can do as UMs are transmitted via multicast). If this is the case then the receiver refrains from sending the UM, and continues to follow the regular RR sending schedule. If the receiver has not yet seen a similar UM from any other receiver, it checks whether it has already overdrafted its RTCP bit rate budget before (without waiting for its regularly scheduled RR time). Only if this is not the case, it sends the UM, after waiting a short, random dithering interval period. Note that always a complete RR is sent in addition to the UM, in order to a) follow the rules for compound packets, and b) make sure that a sufficiently large number of RRs from each receiver is transmitted. Considering the overhead for IP and UDP packets, it is believed that these advantages outweigh the disadvantage of preventing RTCP packets that contain only UMs. Wenger/Ott Expires December 2000 [Page 7] Internet Draft July 14, 2000 3.1. Definitions [Note: not all are used in this first revision of the draft.] a) Let the video stream be transmitted at a (roughly) constant frame rate f (in frames per second). This results in an inter-frame time period of tau=1/f if frame are sent in regular intervals. b) For timing considerations, we assume that a single frame is always carried in a single packet. If a frame does not fit into the MTU, then the frame is split across several packets. Gaps are then measured between always the first or always the last packet of a frame. For later considerations on feedback delay, if a frame is split its packets are paced for transmission (rather than sent as a burst) over some time period T_split, this can be modeled as a _constant_ added to the overall transmission delay from the sender to the receiver. c) Let T_rtt be the maximum round trip time as measured by RTCP. Note that this may be asymmetric. d) Let T_jitter be the maximum jitter measured from a sender to a receiver. e) 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 the RFC1889bis draft these are termed tp, tn, respectively). f) Let t_e be the time for which a feedback packet is scheduled. g) Let t_dither_max be the maximum interval for which an RTCP feedback packet may be additionally delayed (to prevent implosions). h) Let T_fd be the delay for the feedback message that a certain packet to return to the sender after. i) Let S be the number of active senders in the RTP session. j) Let N be the current estimate of the number of receivers in the RTP session. 3.2. RTCP Feedback The feedback situation for a packet loss at a receiver is depicted in figure 1 below. At time t0, 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 certain type of feedback information shall be sent back to the sender. Wenger/Ott Expires December 2000 [Page 8] Internet Draft July 14, 2000 To avoid an implosion of immediate feedback packets, the receivers delays transmission of the feedback packet(an Early RTCP RR/FB packets) by a random amount T_fd (with the random number evenly distributed in the interval [0, T_dither_max]. Transmission of the RTCP RR/FB is then scheduled for t_e = t0 + T_fd. The T_dither_max parameter depends on the feedback algorithm used (PLI, SLI, RPSI) and needs to take into account a number of other parameters (such as the estimated round-trip time) to limit the upper bound for the feedback in a way that ensures that the feedback information still makes sense when it reaches the sender. If an RTCP feedback packet is scheduled, the time slot for the next scheduled RTCP RR is updated accordingly to a new t_rr taken from the interval [t_(rr-1) + 2*T_rr, t_e + 2*T_rr] (with T_rr being the newly calculated deterministic RTCP interval. pkt loss detected | | RTCP feedback vXXXXXXXXXXXXXXXXXXXX ) ) |---+--------+-------------+-----+------------| |--------+---------> | | | | ( ( | | t0 te | t_(rr-1) t_rr \_______ ________/ \/ T_dither_max Figure 1: Packet loss and parameters for Early RR scheduling 3.3. Early RR/UM 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 document at the moment; a future revision may contain more detailed guidelines to this end). Then, the following rules apply to transmitting an Upstream Messages (UM) as compound packet with RTCP RR and possibly other information. This compound RTCP packet is referred to as _RTCP RR/UM_. Initially, R sets allow_early=TRUE. Wenger/Ott Expires December 2000 [Page 9] Internet Draft July 14, 2000 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. If R detects a packet loss at time t0 then R should check first whether its next regularly scheduled RTCP RR is within the time bounds for the RTCP UM (t_e + t_dither_max > t_rr). If so, no Early RR is scheduled; instead the UM is appended to the regular RTCP RR. Otherwise, R should check whether it is allowed to transmit an Early RR/FB packet (allow_early==TRUE). If so, R creates a UM unit, calculates t_dither_max and then schedules an early RR/UM packet for t_e = t0 + RND * t_dither_max with the RND function evenly distributed between 0 and 1. If R receives an RR/UM 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. (Note: if the UM is piggybacked onto a regularly scheduled RTCP RR message, this should not affect transmission of the RR; but should the UM then be removed from the compound RR/UM?) Otherwise, when t_e is reached, R creates an RR, appends the UM information, and transmits the RR/UM packet. R then sets allow_early=FALSE and recalculates t_rr += T_rr (possibly t_rr = t_e + 2*T_rr or some value in between; this needs further work). As soon as R sends its next regularly scheduled RTCP RR (at the new t_rr), it sets allow_early=TRUE again. Option: R also starts a timer T_allow (e.g. T_allow=T_rr). If T_rr expires before an Early RR/UM is received from another participant in the RTP session, R sets allow_early=TRUE. If an Early RR/UM is received from another participant before T_allow expires, T_allow is cancelled. If allow_early==FALSE then R calculates t_dither_max and checks the time for the next scheduled RR: if t_rr - t0 < t_dither_max then R creates an FB unit for transmission along with the RR packet at t_rr (see above). Otherwise, R does not send an RTCP RR/UM. Note: A bit in the UM unit is required to indicate whether the transmission occurs as an Early RR/FB or as a regularly scheduled RR/FB packet. This E-bit is to be set accordingly. See section 4 for details. Note: Numerous variations spring to mind on RTCP RR/UM scheduling, dithering, damping, etc. Right now, this is deliberately kept simple for an easy starting point and to provoke further Wenger/Ott Expires December 2000 [Page 10] Internet Draft July 14, 2000 discussions. 3.4. Summary of decision steps Before even considering whether or not to send RTCP UM information an application has to determine whether this mechanism is applicable: 1) An application has to decide whether -- for the current ratio of frame rate with the associated (application-specific) maximum feedback delay and the currently observed round-trip time -- feedback mechanisms can be applied at all. 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. 3) If these tests pass, the application has to follow the rules for transmitting early RTCP RRs or regularly scheduled RTCP RRs with piggybacked UMs. 4. Format of RTCP Feedback messages The general format of an UM is outlined below. Compound packets including UMs are possible. All UMs concerning any given picture of any given receiver MUST be conveyed in a single compound packet, in order to prevent the loss of parts of such a combined message. It SHOULD be avoided to combine different types of UMs for any given picture of any given receiver. 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |V=2| UMT |E| PT=RTCP-Feedb | length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ | Upstream Control Information (UCI) | | +-+-+-+-+-+-+-+-+-+-+-+-+-+ | : padding | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ version (V): 2 bits Identifies the version of RTP, which is the same in RTCP packets as in RTP data packets. Upstream Message Type (UMT): 5 bits Identifies the type of the upstream message. 0: forbidden Wenger/Ott Expires December 2000 [Page 11] Internet Draft July 14, 2000 1: Picture Loss Indication 2: Slice Lost Indication 3: Reference Picture Selection Indication 4-31: reserved Packet Type (PT): 8 bits Constant value (TBD) identifying RTCP Upstream messages. Early Upstream Message (E): 1 bit A bit that, when set, indicates that the UM is sent early, i.e. did not follow the regular schedule for sending RTCP Receiver Reports. Length: 16 bits: Number of bits valid in the UCI field. A zero value indicates that the UCI field is not present (e.g. in case of a Picture Intra Request). SSRC: 32 bits SSRC is the synchronization source identifier for the sender of this packet. Upstream Control Information (UCI): variable Format and semantics of the UCI defer for the various upstream message types. Fragmentation of an upstream message into several UCI fields is prohibited. See the following sections for their definition. 5. Message Type 1: Picture Loss Indication (PLI) 5.1 Semantics With the Picture Loss Indication message a decoder informs the encoder about the loss of one or more full pictures 5.2 Format PLI does not require parameters. Therefore, the length field MUST be 0, and there MUST NOT be Upstream Control Information. 5.3 Timing Rules The timing follows the rules outlined in section 3. In systems that employ both PLI and other UM types it may be advisable to follow the regular RTCP RR timing rules, since PLI is not as delay critical as other UM types. 5.4 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 Wenger/Ott Expires December 2000 [Page 12] Internet Draft July 14, 2000 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 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 upstream message. Hence waiting for the next possible time slot allowed by RFC1889bis RTCP timing rules does not negatively influence system performance. 6. Message Type 2: Slice Lost Indication 6.1 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 upstream 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. 6.2 Format When UMT 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. 6.3 Timing Rules Wenger/Ott Expires December 2000 [Page 13] Internet Draft July 14, 2000 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. Constraints on T_dither_max to be discussed. 6.4 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 upstream 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 UM [TBP.]. While, when those algorithms are used, the timing of the UM 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 UMs. 7. Message Type 3: Reference Picture Selection Indication 7.1 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 Wenger/Ott Expires December 2000 [Page 14] Internet Draft July 14, 2000 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. 7.2 Format When UM 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). 7.3 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. 7.4 Remarks [To Do] 8. 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 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. Wenger/Ott Expires December 2000 [Page 15] Internet Draft July 14, 2000 9. 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. 10. Full Copyright Statement Copyright (C) The Internet Society (1999). 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." 11. Authors' Addresses Stephan Wenger (stewe@cs.tu-berlin.de) TU Berlin Sekr. FR 6-3 Franklinstr. 28-29 D-10587 Berlin Germany Joerg Ott (jo@tzi.uni-bremen.de) Universitaet Bremen TZI MZH 5180 Bibliothekstr. 1 D-28359 Bremen Germany 12. Bibliography: TODO Wenger/Ott Expires December 2000 [Page 16]