Audio/Video Transport Working Group Stephen Casner Internet Draft Packet Design November 17, 2000 Van Jacobson Expires June 2001 Packet Design draft-ietf-avt-crtp-enhance-01.txt Tmima Koren Bruce Thompson Dan Wing Patrick Ruddy Alex Tweedly Cisco Systems John Geevarghese Telseon Compressing IP/UDP/RTP Headers for Low-Speed Serial Links 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. Internet Drafts may be updated, replaced, or obsolete by other documents at any time. It is not appropriate 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.txt This draft is being submitted as a possible work item to the IETF Audio/Video Transport working group. To subscribe to the mailing list send a message to rem-conf-request@es.net with the line "subscribe" in the body of the message. Archives are available from: ftp://ftp.es.net/pub/mail-archive/rem-conf Copyright Notice Copyright (C) The Internet Society (1999-2000). All Rights Reserved. Abstract This document describes a method for compressing the headers of IP/UDP/RTP datagrams to reduce overhead on low-speed serial links. In many cases, all three headers can be compressed to 2-4 bytes. Comments are solicited and should be addressed to the working group mailing list rem-conf@es.net and/or the author(s). 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. 1. Introduction Since the Real-time Transport Protocol was published as an RFC [1], there has been growing interest in using RTP as one step to achieve interoperability among different implementations of network audio/video applications. However, there is also concern that the 12-byte RTP header is too large an overhead for 20-byte payloads when operating over low speed lines such as dial-up modems at 14.4 or 28.8 kb/s. (Some existing applications operating in this environment use an application-specific protocol with a header of a few bytes that has reduced functionality relative to RTP.) Header size may be reduced through compression techniques as has been done with great success for TCP [2]. In this case, compression might be applied to the RTP header alone, on an end-to-end basis, or to the combination of IP, UDP and RTP headers on a link-by-link basis. Compressing the 40 bytes of combined headers together provides substantially more gain than compressing 12 bytes of RTP header alone because the resulting size is approximately the same (2-4 bytes) in either case. Compressing on a link-by-link basis also provides better performance because the delay and loss rate are lower. Therefore, the method defined here is for combined compression of IP, UDP and RTP headers on a link-by-link basis. This document defines a compression scheme that may be used with IPv4, IPv6 or packets encapsulated with more than one IP header, though the initial focus is on IPv4. The IP/UDP/RTP compression defined here is intended to fit within the more general compression framework specified in [3] for use with both IPv6 and IPv4. That framework defines TCP and non-TCP as two classes of transport above IP. This specification creates IP/UDP/RTP as a third class extracted from the non-TCP class. 2. Assumptions and Tradeoffs The goal of this compression scheme is to reduce the IP/UDP/RTP headers to two bytes for most packets in the case where no UDP checksums are being sent, or four bytes with checksums. It is motivated primarily by the specific problem of sending audio and video over 14.4 and 28.8 dialup modems. These links tend to provide full-duplex communication, so the protocol takes advantage of that fact, though the protocol may also be used with reduced performance on simplex links. This compression scheme performs best on local links with low round-trip-time. This specification does not address segmentation and preemption of large packets to reduce the delay across the slow link experienced by small real-time packets, except to identify in Section 4 some interactions between segmentation and compression that may occur. Segmentation schemes may be defined separately and used in conjunction with the compression defined here. It should be noted that implementation simplicity is an important factor to consider in evaluating a compression scheme. Communications servers may need to support compression over perhaps as many as 100 dial-up modem lines using a single processor. Therefore, it may be appropriate to make some simplifications in the design at the expense of generality, or to produce a flexible design that is general but can be subsetted for simplicity. Higher compression gain might be achieved by communicating more complex models for the changing header fields from the compressor to the decompressor, but that complexity is deemed unnecessary. The next sections discuss some of the tradeoffs listed here. 2.1. Simplex vs. Full Duplex In the absence of other constraints, a compression scheme that worked over simplex links would be preferred over one that did not. However, operation over a simplex link requires periodic refreshes with an uncompressed packet header to restore compression state in case of error. If an explicit error signal can be returned instead, the delay to recovery may be shortened substantially. The overhead in the no-error case is also reduced. To gain these performance improvements, this specification includes an explicit error indication sent on the reverse path. On a simplex link, it would be possible to use a periodic refresh instead. Whenever the decompressor detected an error in a particular packet stream, it would simply discard all packets in that stream until an uncompressed header was received for that stream, and then resume decompression. The penalty would be the potentially large number of packets discarded. The periodic refresh method described in Section 3.3 of [3] applies to IP/UDP/RTP compression on simplex links or links with high delay as well as to other non-TCP packet streams. 2.2. Links with high bit error rate and long round trip delay IP/UDP/RTP header compression may be used in scenarios where a compressed link could extend over a long physical distance and involve multiple layer-2 switching points. An example of such a link is RTP transport over ATM AAL5, where the "link" would actually traverse through multiple layer-2 switching points on the path from the CRTP transmitter (compressor) to the CRTP receiver (decompressor). Another example is a wireless link. Such links may experience significant packet loss and/or long round trip delays. Contexts get invalidated due to packet loss, but the CRTP error recovery mechanism using CONTEXT_STATE messages is not efficient due to the long round trip delay. In scenarios such as this, it is desirable to minimize context invalidation. A set of enhancements is defined for error prevention and recovery. The enhancements make CRTP more robust and resilient to packet loss, which in turn will reduce context invalidation. 2.3. Segmentation and Layering Delay induced by the time required to send a large packet over the slow link is not a problem for one-way audio, for example, because the receiver can adapt to the variance in delay. However, for interactive conversations, minimizing the end-to-end delay is critical. Segmentation of large, non-real-time packets to allow small real-time packets to be transmitted between segments can reduce the delay. This specification deals only with compression and assumes segmentation, if included, will be handled as a separate layer. It would be inappropriate to integrate segmentation and compression in such a way that the compression could not be used by itself in situations where segmentation was deemed unnecessary or impractical. Similarly, one would like to avoid any requirements for a reservation protocol. The compression scheme can be applied locally on the two ends of a link independent of any other mechanisms except for the requirements that the link layer provide some packet type codes, a packet length indication, and good error detection. Conversely, separately compressing the IP/UDP and RTP layers loses too much of the compression gain that is possible by treating them together. Crossing these protocol layer boundaries is appropriate because the same function is being applied across all layers. 3. The Compression Algorithm The compression algorithm defined in this document draws heavily upon the design of TCP/IP header compression as described in RFC 1144 [2]. Readers are referred to that RFC for more information on the underlying motivations and general principles of header compression. 3.1. The basic idea In TCP header compression, the first factor-of-two reduction in data rate comes from the observation that half of the bytes in the IP and TCP headers remain constant over the life of the connection. After sending the uncompressed header once, these fields may be elided from the compressed headers that follow. The remaining compression comes from differential coding on the changing fields to reduce their size, and from eliminating the changing fields entirely for common cases by calculating the changes from the length of the packet. This length is indicated by the link-level protocol. For RTP header compression, some of the same techniques may be applied. However, the big gain comes from the observation that although several fields change in every packet, the difference from packet to packet is often constant and therefore the second-order difference is zero. By maintaining both the uncompressed header and the first-order differences in the session state shared between the compressor and decompressor, all that must be communicated is an indication that the second-order difference was zero. In that case, the decompressor can reconstruct the original header without any loss of information simply by adding the first-order differences to the saved uncompressed header as each compressed packet is received. Just as TCP/IP header compression maintains shared state for multiple simultaneous TCP connections, this IP/UDP/RTP compression SHOULD maintain state for multiple session contexts. A session context is defined by the combination of the IP source and destination addresses, the UDP source and destination ports, and the RTP SSRC field. A compressor implementation might use a hash function on these fields to index a table of stored session contexts. The compressed packet carries a small integer, called the session context identifier or CID, to indicate in which session context that packet should be interpreted. The decompressor can use the CID to index its table of stored session contexts directly. Because the RTP compression is lossless, it may be applied to any UDP traffic that benefits from it. Most likely, the only packets that will benefit are RTP packets, but it is acceptable to use heuristics to determine whether or not the packet is an RTP packet because no harm is done if the heuristic gives the wrong answer. This does require executing the compression algorithm for all UDP packets, or at least those with even port numbers (see section 3.4). Most compressor implementations will need to maintain a "negative cache" of packet streams that have failed to compress as RTP packets for some number of attempts in order to avoid further attempts. Failing to compress means that some fields in the potential RTP header that are expected to remain constant most of the time, such as the payload type field, keep changing. Even if the other such fields remain constant, a packet stream with a constantly changing SSRC field SHOULD be entered in the negative cache to avoid consuming all of the available session contexts. The negative cache is indexed by the source and destination IP address and UDP port pairs but not the RTP SSRC field since the latter may be changing. When RTP compression fails, the IP and UDP headers MAY still be compressed. Fragmented IP Packets that are not initial fragments and packets that are not long enough to contain a complete UDP header MUST NOT be sent as FULL_HEADER packets. Furthermore, packets that do not additionally contain at least 12 bytes of UDP data MUST NOT be used to establish RTP context. If such a packet is sent as a FULL_HEADER packet, it MAY be followed by COMPRESSED_UDP packets but MUST NOT be followed by COMPRESSED_RTP packets. 3.2. Header Compression for RTP Data Packets In the IPv4 header, only the total length, packet ID, and header check-sum fields will normally change. The total length is redundant with the length provided by the link layer, and since this compression scheme must depend upon the link layer to provide good error detection (e.g., PPP's CRC [4]), the header checksum may also be elided. This leaves only the packet ID, which, assuming no IP fragmentation, would not need to be communicated. However, in order to maintain lossless compression, changes in the packet ID will be transmitted. The packet ID usually increments by one or a small number for each packet. (Some systems increment the ID with the bytes swapped, which results in slightly less compression.) In the IPv6 base header, there is no packet ID nor header checksum and only the payload length field changes. In the UDP header, the length field is redundant with the IP total length field and the length indicated by the link layer. The UDP check-sum field will be a constant zero if the source elects not to generate UDP checksums. Otherwise, the checksum must be communicated intact in order to preserve the lossless compression. Maintaining end-to-end error detection for applications that require it is an important principle. In the RTP header, the SSRC identifier is constant in a given context since that is part of what identifies the particular context. For most packets, only the sequence number and the timestamp will change from packet to packet. If packets are not lost or misordered upstream from the compressor, the sequence number will increment by one for each packet. For audio packets of constant duration, the timestamp will increment by the number of sample periods conveyed in each packet. For video, the timestamp will change on the first packet of each frame, but then stay constant for any additional packets in the frame. If each video frame occupies only one packet, but the video frames are generated at a constant rate, then again the change in the timestamp from frame to frame is constant. Note that in each of these cases the second-order difference of the sequence number and timestamp fields is zero, so the next packet header can be constructed from the previous packet header by adding the first-order differences for these fields that are stored in the session context along with the previous uncompressed header. When the second-order difference is not zero, the magnitude of the change is usually much smaller than the full number of bits in the field, so the size can be reduced by encoding the new first-order difference and transmitting it rather than the absolute value. The M bit will be set on the first packet of an audio talkspurt and the last packet of a video frame. If it were treated as a constant field such that each change required sending the full RTP header, this would reduce the compression significantly. Therefore, one bit in the compressed header will carry the M bit explicitly. If the packets are flowing through an RTP mixer, most commonly for audio, then the CSRC list and CC count will also change. However, the CSRC list will typically remain constant during a talkspurt or longer, so it need be sent only when it changes. 3.3. The protocol The compression protocol must maintain a collection of shared information in a consistent state between the compressor and decompressor. There is a separate session context for each IP/UDP/RTP packet stream, as defined by a particular combination of the IP source and destination addresses, UDP source and destination ports, and the RTP SSRC field. The number of session contexts to be maintained MAY be negotiated between the compressor and decompressor. Each context is identified by an 8- or 16-bit identifier, depending upon the number of contexts negotiated, so the maximum number is 65536. Both uncompressed and compressed packets MUST carry the context ID and a 4-bit sequence number used to detect packet loss between the compressor and decompressor. Each context has its own separate sequence number space so that a single packet loss need only invalidate one context. The shared information in each context consists of the following items: o The full IP, UDP and RTP headers, possibly including a CSRC list, for the last packet sent by the compressor or reconstructed by the decompressor. o The first-order difference for the IPv4 ID field, initialized to 1 whenever an uncompressed IP header for this context is received and updated each time a delta IPv4 ID field is received in a compressed packet. o The first-order difference for the RTP timestamp field, initialized to 0 whenever an uncompressed packet for this context is received and updated each time a delta RTP timestamp field is received in a compressed packet. o The last value of the 4-bit sequence number, which is used to detect packet loss between the compressor and decompressor. o The current generation number for non-differential coding of UDP packets with IPv6 (see [3]). For IPv4, the generation number may be set to zero if the COMPRESSED_NON_TCP packet type, defined below, is never used. o A context-specific delta encoding table (see section 3.3.4) may optionally be negotiated for each context. In order to communicate packets in the various uncompressed and compressed forms, this protocol depends upon the link layer being able to provide an indication of four new packet formats in addition to the normal IPv4 and IPv6 packet formats: FULL_HEADER - communicates the uncompressed IP header plus any following headers and data to establish the uncompressed header state in the decompressor for a particular context. The FULL- HEADER packet also carries the 8- or 16-bit session context identifier and the 4-bit sequence number to establish synchronization between the compressor and decompressor. The format is shown in section 3.3.1. COMPRESSED_UDP - communicates the IP and UDP headers compressed to 6 or fewer bytes (often 2 if UDP checksums are disabled), followed by any subsequent headers (possibly RTP) in uncompressed form, plus data. This packet type is used when there are differences in the usually constant fields of the (potential) RTP header. The RTP header includes a potentially changed value of the SSRC field, so this packet may redefine the session context. The format is shown in section 3.3.3. COMPRESSED_RTP - indicates that the RTP header is compressed along with the IP and UDP headers. The size of this header may still be just two bytes, or more if differences must be communicated. This packet type is used when the second-order difference (at least in the usually constant fields) is zero. It includes delta encodings for those fields that have changed by other than the expected amount to establish the first-order differences after an uncompressed RTP header is sent and whenever they change. The format is shown in section 3.3.2. CONTEXT_STATE - indicates a special packet sent from the decompressor to the compressor to communicate a list of context IDs for which synchronization has or may have been lost. This packet is only sent across the point-to-point link so it requires no IP header. The format is shown in section 3.3.5. When this compression scheme is used with IPv6 as part of the general header compression framework specified in [3], another packet type MAY be used: COMPRESSED_NON_TCP - communicates the compressed IP and UDP headers as defined in [3] without differential encoding. If it were used for IPv4, it would require one or two bytes more than the COMPRESSED_UDP form listed above in order to carry the IPv4 ID field. For IPv6, there is no ID field and this non-differential compression is more resilient to packet loss. Assignments of numeric codes for these packet formats in the Point- to-Point Protocol [4] are to be made by the Internet Assigned Numbers Authority. 3.3.1. FULL_HEADER (uncompressed) packet format The definition of the FULL_HEADER packet given here is intended to be the consistent with the definition given in [3]. Full details on design choices are given there. The format of the FULL_HEADER packet is the same as that of the original packet. In the IPv4 case, this is usually an IP header, followed by a UDP header and UDP payload that may be an RTP header and its payload. However, the FULL_HEADER packet may also carry IP encapsulated packets, in which case there would be two IP headers followed by UDP and possibly RTP. Or in the case of IPv6, the packet may be built of some combination of IPv6 and IPv4 headers. Each successive header is indicated by the type field of the previous header, as usual. The FULL_HEADER packet differs from the corresponding normal IPv4 or IPv6 packet in that it must also carry the compression context ID and the 4-bit sequence number. In order to avoid expanding the size of the header, these values are inserted into length fields in the IP and UDP headers since the actual length may be inferred from the length provided by the link layer. Two 16-bit length fields are needed; these are taken from the first two available headers in the packet. That is, for an IPv4/UDP packet, the first length field is the total length field of the IPv4 header, and the second is the length field of the UDP header. For an IPv4 encapsulated packet, the first length field would come from the total length field of the first IP header, and the second length field would come from the total length field of the second IP header. As specified in Sections 5.3.2 of [3], the position of the context ID (CID) and 4-bit sequence number varies depending upon whether 8- or 16-bit context IDs have been selected, as shown in the following diagram (16 bits wide, with the most-significant bit is to the left): For 8-bit context ID: +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |0|1| Generation| CID | First length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 0 | seq | Second length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ For 16-bit context ID: +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |1|1| Generation| 0 | seq | First length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | CID | Second length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ The first bit in the first length field indicates the length of the CID. The length of the CID MUST either be constant for all contexts or two additional distinct packet types MUST be provided to separately indicate COMPRESSED_UDP and COMPRESSED_RTP packet formats with 8- and 16-bit CIDs. The second bit in the first length field is 1 to indicate that the 4-bit sequence number is present, as is always the case for this IP/UDP/RTP compression scheme. The generation field is used with IPv6 for COMPRESSED_NON_TCP packets as described in [3]. For IPv4-only implementations that do not use COMPRESSED_NON_TCP packets, the compressor SHOULD set the generation value to zero. For consistent operation between IPv4 and IPv6, the generation value is stored in the context when it is received by the decompressor, and the most recent value is returned in the CONTEXT_STATE packet. When a FULL_HEADER packet is received, the complete set of headers is stored into the context selected by the context ID. The 4-bit sequence number is also stored in the context, thereby resynchronizing the decompressor to the compressor. When COMPRESSED_NON_TCP packets are used, the 4-bit sequence number is inserted into the "Data Field" of that packet and the D bit is set as described in Section 6 of [3]. When a COMPRESSED_NON_TCP packet is received, the generation number is compared to the value stored in the context. If they are not the same, the context is not up to date and MUST be refreshed by a FULL_HEADER packet. If the generation does match, then the compressed IP and UDP header information, the 4-bit sequence number, and the (potential) RTP header are all stored into the saved context. The amount of memory required to store the context will vary depending upon how many encapsulating headers are included in the FULL_HEADER packet. The compressor and decompressor MAY negotiate a maximum header size. 3.3.2. COMPRESSED_RTP packet format When the second-order difference of the RTP header from packet to packet is zero, the decompressor can reconstruct a packet simply by adding the stored first-order differences to the stored uncompressed header representing the previous packet. All that need be communicated is the session context identifier and a small sequence number (not related to the RTP sequence number) to maintain synchronization and detect packet loss between the compressor and decompressor. If the second-order difference of the RTP header is not zero for some fields, the new first-order difference for just those fields is communicated using a compact encoding. The new first-order difference values are added to the corresponding fields in the uncompressed header in the decompressor's session context, and are also stored explicitly in the context to be added to the corresponding fields again on each subsequent packet in which the second-order difference is zero. Each time the first-order difference changes, it is transmitted and stored in the context. In practice, the only fields for which it is useful to store the first-order difference are the IPv4 ID field and the RTP timestamp. For the RTP sequence number field, the usual increment is 1. If the sequence number changes by other than 1, the difference must be communicated but does not set the expected difference for the next packet. Instead, the expected first-order difference remains fixed at 1 so that the difference need not be explicitly communicated on the next packet assuming it is in order. For the RTP timestamp, when a FULL_HEADER, COMPRESSED_NON_TCP or COMPRESSED_UDP packet is sent to refresh the RTP state, the stored first-order difference is initialized to zero. If the timestamp is the same on the next packet (e.g., same video frame), then the second-order difference is zero. Otherwise, the difference between the timestamps of the two packets is transmitted as the new first- order difference to be added to the timestamp in the uncompressed header stored in the decompressor's context and also stored as the first-order difference in that context. Each time the first-order difference changes on subsequent packets, that difference is again transmitted and used to update the context. Similarly, since the IPv4 ID field frequently increments by one, the first-order difference for that field is initialized to one when the state is refreshed by a FULL_HEADER packet, or when a COMPRESSED_NON_TCP packet is sent since it carries the ID field in uncompressed form. Thereafter, whenever the first-order difference changes, it is transmitted and stored in the context. A bit mask will be used to indicate which fields have changed by other than the expected difference. In addition to the small link sequence number, the list of items to be conditionally communicated in the compressed IP/UDP/RTP header is as follows: I = IPv4 packet ID (always 0 if no IPv4 header) U = UDP checksum M = RTP marker bit S = RTP sequence number T = RTP timestamp L = RTP CSRC count and list If 4 bits are needed for the link sequence number to get a reasonable probability of loss detection, there are too few bits remaining to assign one bit to each of these items and still fit them all into a single byte to go along with the context ID. It is not necessary to explicitly carry the U bit to indicate the presence of the UDP checksum because a source will typically include check-sums on all packets of a session or none of them. When the session state is initialized with an uncompressed header, if there is a nonzero checksum present, an unencoded 16-bit checksum will be inserted into the compressed header in all subsequent packets until this setting is changed by sending another uncompressed packet. Of the remaining items, the L bit for the CSRC count and list may be the one least frequently used. Rather than dedicating a bit in the mask to indicate CSRC change, an unusual combination of the other bits may be used instead. This bit combination is denoted MSTI. If all four of the bits for the IP packet ID, RTP marker bit, RTP sequence number and RTP timestamp are set, this is a special case indicating an extended form of the compressed RTP header will follow. That header will include an additional byte containing the real values of the four bits plus the CC count. The CSRC list, of length indicated by the CC count, will be included just as it appears in the uncompressed RTP header. The other fields of the RTP header (version, P bit, X bit, payload type and SSRC identifier) are assumed to remain relatively constant. In particular, the SSRC identifier is defined to be constant for a given context because it is one of the factors selecting the context. If any of the other fields change, the uncompressed RTP header MUST sent as described in Section 3.3.3. The following diagram shows the compressed IP/UDP/RTP header with dotted lines indicating fields that are conditionally present. The most significant bit is numbered 0. Multi-byte fields are sent in network byte order (most significant byte first). The delta fields are often a single byte as shown but may be two or three bytes depending upon the delta value as explained in Section 3.3.4. 0 1 2 3 4 5 6 7 +...............................+ : msb of session context ID : (if 16-bit CID) +-------------------------------+ | lsb of session context ID | +---+---+---+---+---+---+---+---+ | M | S | T | I | link sequence | +---+---+---+---+---+---+---+---+ : : + UDP checksum + (if nonzero in context) : : +...............................+ : : + "RANDOM" fields + (if encapsulated) : : +...............................+ : M'| S'| T'| I'| CC : (if MSTI = 1111) +...............................+ : delta IPv4 ID : (if I or I' = 1) +...............................+ : delta RTP sequence : (if S or S' = 1) +...............................+ : delta RTP timestamp : (if T or T' = 1) +...............................+ : : : CSRC list : (if MSTI = 1111 : : and CC nonzero) : : +...............................+ : : : RTP header extension : (if X set in context) : : : : +-------------------------------+ | | | RTP data | / / / / | | +-------------------------------+ : padding : (if P set in context) +...............................+ When more than one IPv4 header is present in the context as initialized by the FULL_HEADER packet, then the IP ID fields of encapsulating headers MUST be sent as absolute values as described in [3]. These fields are identified as "RANDOM" fields. They are inserted into the COMPRESSED_RTP packet in the same order as they appear in the original headers, immediately following the UDP checksum if present or the MSTI byte if not, as shown in the diagram. Only if an IPv4 packet immediately precedes the UDP header will the IP ID of that header be sent differentially, i.e., potentially with no bits if the second difference is zero, or as a delta IPv4 ID field if not. If there is not an IPv4 header immediately preceding the UDP header, then the I bit MUST be 0 and no delta IPv4 ID field will be present. 3.3.3. COMPRESSED_UDP packet format If there is a change in any of the fields of the RTP header that are normally constant (such as the payload type field), then an uncompressed RTP header MUST be sent. If the IP and UDP headers do not also require updating, this RTP header MAY be carried in a COMPRESSED_UDP packet rather than a FULL_HEADER packet. The COMPRESSED_UDP packet has the same format as the COMPRESSED_RTP packet except that the M, S and T bits are always 0 and the corresponding delta fields are never included: 0 1 2 3 4 5 6 7 +...............................+ : msb of session context ID : (if 16-bit CID) +-------------------------------+ | lsb of session context ID | +---+---+---+---+---+---+---+---+ | 0 | 0 | 0 | I | link sequence | +---+---+---+---+---+---+---+---+ : : + UDP checksum + (if nonzero in context) : : +...............................+ : : + "RANDOM" fields + (if encapsulated) : : +...............................+ : delta IPv4 ID : (if I = 1) +-------------------------------+ | UDP data | : (uncompressed RTP header) : Note that this constitutes a form of IP/UDP header compression different from COMPRESSED_NON_TCP packet type defined in [3]. The motivation is to allow reaching the target of two bytes when UDP checksums are disabled, as IPv4 allows. The protocol in [3] does not use differential coding for UDP packets, so in the IPv4 case, two bytes of IP ID, and two bytes of UDP checksum if nonzero, would always be transmitted in addition to two bytes of compression prefix. For IPv6, the COMPRESSED_NON_TCP packet type MAY be used instead. 3.3.4. Encoding of differences The delta fields in the COMPRESSED_RTP and COMPRESSED_UDP packets are encoded with a variable-length mapping for compactness of the more commonly-used values. A default encoding is specified below, but it is RECOMMENDED that implementations use a table-driven delta encoder and decoder to allow negotiation of a table specific for each session if appropriate, possibly even an optimal Huffman encoding. Encodings based on sequential interpretation of the bit stream, of which this default table and Huffman encoding are examples, allow a reasonable table size and may result in an execution speed faster than a non- table-driven implementation with explicit tests for ranges of values. The default delta encoding is specified in the following table. This encoding was designed to efficiently encode the small changes that may occur in the IP ID and in RTP sequence number when packets are lost upstream from the compressor, yet still handling most audio and video deltas in two bytes. The column on the left is the decimal value to be encoded, and the column on the right is the resulting sequence of bytes shown in hexadecimal and in the order in which they are transmitted (network byte order). The first and last values in each contiguous range are shown, with ellipses in between: Decimal Hex -16384 C0 00 00 : : -129 C0 3F 7F -128 80 00 : : -1 80 7F 0 00 : : 127 7F 128 80 80 : : 16383 BF FF 16384 C0 40 00 : : 4194303 FF FF FF For positive values, a change of zero through 127 is represented directly in one byte. If the most significant two bits of the byte are 10 or 11, this signals an extension to a two- or three-byte value, respectively. The least significant six bits of the first byte are combined, in decreasing order of significance, with the next one or two bytes to form a 14- or 22-bit value. Negative deltas may occur when packets are misordered or in the intentionally out-of-order RTP timestamps on MPEG video [5]. These events are less likely, so a smaller range of negative values is encoded using otherwise redundant portions of the positive part of the table. A change in the RTP timestamp value less than -16384 or greater than 4194303 forces the RTP header to be sent uncompressed using a FULL_HEADER, COMPRESSED_NON_TCP or COMPRESSED_UDP packet type. The IP ID and RTP sequence number fields are only 16 bits, so negative deltas for those fields SHOULD be masked to 16 bits and then encoded (as large positive 16-bit numbers). 3.3.5. Error Recovery Whenever the 4-bit sequence number for a particular context increments by other than 1, except when set by a FULL_HEADER or COMPRESSED_NON_TCP packet, the decompressor MUST invalidate that context and send a CONTEXT_STATE packet back to the compressor indicating that the context has been invalidated. All packets for the invalid context MUST be discarded until a FULL_HEADER or COMPRESSED_NON_TCP packet is received for that context to re- establish consistent state (unless the "twice" algorithm is used as described later in this section). Since multiple compressed packets may arrive in the interim, the decompressor SHOULD NOT retransmit the CONTEXT_STATE packet for every compressed packet received, but instead SHOULD limit the rate of retransmission to avoid flooding the reverse channel. When an error occurs on the link, the link layer will usually discard the packet that was damaged (if any), but may provide an indication of the error. Some time may elapse before another packet is delivered for the same context, and then that packet would have to be discarded by the decompressor when it is observed to be out of sequence, resulting in at least two packets lost. To allow faster recovery if the link does provide an explicit error indication, the decompressor MAY optionally send an advisory CONTEXT_STATE packet listing the last valid sequence number and generation number for one or more recently active contexts (not necessarily all). For a given context, if the compressor has sent no compressed packet with a higher sequence number, and if the generation number matches the current generation, no corrective action is required. Otherwise, the compressor MAY choose to mark the context invalid so that the next packet is sent in FULL_HEADER or COMPRESSED_NON_TCP mode (FULL_HEADER is required if the generation doesn't match). However, note that if the link round-trip-time is large compared to the inter-packet spacing, there may be several packets from multiple contexts in flight across the link, increasing the probability that the sequence numbers will already have advanced when the CONTEXT_STATE packet is received by the compressor. The result could be that some contexts are invalidated unnecessarily, causing extra bandwidth to be consumed. The format of the CONTEXT_STATE packet is shown in the following diagrams. The first byte is a type code to allow the CONTEXT_STATE packet type to be shared by multiple compression schemes within the general compression framework specified in [3]. The contents of the remainder of the packet depends upon the compression scheme. For the IP/UDP/RTP compression scheme specified here, the remainder of the CONTEXT_STATE packet is structured as a list of blocks to allow the state for multiple contexts to be indicated, preceded by a one-byte count of the number of blocks. Two type code values are used for the IP/UDP/RTP compression scheme. The value 1 indicates that 8-bit session context IDs are being used: 0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ | 1 = IP/UDP/RTP with 8-bit CID | +---+---+---+---+---+---+---+---+ | context count | +---+---+---+---+---+---+---+---+ +---+---+---+---+---+---+---+---+ | session context ID | +---+---+---+---+---+---+---+---+ | I | 0 | 0 | 0 | sequence | +---+---+---+---+---+---+---+---+ | 0 | 0 | generation | +---+---+---+---+---+---+---+---+ ... +---+---+---+---+---+---+---+---+ | session context ID | +---+---+---+---+---+---+---+---+ | I | 0 | 0 | 0 | sequence | +---+---+---+---+---+---+---+---+ | 0 | 0 | generation | +---+---+---+---+---+---+---+---+ The value 2 indicates that 16-bit session context IDs are being used. The session context ID is sent in network byte order (most significant byte first): 0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ | 2 = IP/UDP/RTP with 16-bit CID| +---+---+---+---+---+---+---+---+ | context count | +---+---+---+---+---+---+---+---+ +---+---+---+---+---+---+---+---+ | | + session context ID + | | +---+---+---+---+---+---+---+---+ | I | 0 | 0 | 0 | sequence | +---+---+---+---+---+---+---+---+ | 0 | 0 | generation | +---+---+---+---+---+---+---+---+ ... +---+---+---+---+---+---+---+---+ | | + session context ID + | | +---+---+---+---+---+---+---+---+ | I | 0 | 0 | 0 | sequence | +---+---+---+---+---+---+---+---+ | 0 | 0 | generation | +---+---+---+---+---+---+---+---+ The bit labeled "I" is set to one for contexts that have been marked invalid and require a FULL_HEADER of COMPRESSED_NON_TCP packet to be transmitted. If the I bit is zero, the context state is advisory. The I bit is set to zero to indicate advisory context state that MAY be sent following a link error indication. Since the CONTEXT_STATE packet itself may be lost, retransmission of one or more blocks is allowed. It is expected that retransmission will be triggered only by receipt of another packet, but if the line is near idle, retransmission MAY be triggered by a relatively long timer (on the order of 1 second). If a CONTEXT_STATE block for a given context is retransmitted, it may cross paths with the FULL_HEADER or COMPRESSED_NON_TCP packet intended to refresh that context. In that case, the compressor MAY choose to ignore the error indication. In the case where UDP checksums are being transmitted, the decompressor MAY attempt to use the "twice" algorithm described in section 10.1 of [3]. In this algorithm, the delta is applied more than once on the assumption that the delta may have been the same on the missing packet(s) and the one subsequently received. Success is indicated by a checksum match. For the scheme defined here, the difference in the 4- bit sequence number tells number of times the delta must be applied. Note, however, that there is a nontrivial risk of an incorrect positive indication. It may be advisable to request a FULL_HEADER or COMPRESSED_NON_TCP packet even if the "twice" algorithm succeeds. Some errors may not be detected, for example if 16 packets are lost in a row and the link level does not provide an error indication. In that case, the decompressor will generate packets that are not valid. If UDP checksums are being transmitted, the receiver will probably detect the invalid packets and discard them, but the receiver does not have any means to signal the decompressor. Therefore, it is RECOMMENDED that the decompressor verify the UDP checksum periodically, perhaps one out of 16 packets. If an error is detected, the decompressor would invalidate the context and signal the compressor with a CONTEXT_STATE packet. 3.4. Compression of RTCP Control Packets By relying on the RTP convention that data is carried on an even port number and the corresponding RTCP packets are carried on the next higher (odd) port number, one could tailor separate compression schemes to be applied to RTP and RTCP packets. For RTCP, the compression could apply not only to the header but also the "data", that is, the contents of the different packet types. The numbers in Sender Report (SR) and Receiver Report (RR) RTCP packets would not compress well, but the text information in the Source Description (SDES) packets could be compressed down to a bit mask indicating each item that was present but compressed out (for timing purposes on the SDES NOTE item and to allow the end system to measure the average RTCP packet size for the interval calculation). However, in the compression scheme defined here, no compression will be done on the RTCP headers and "data" for several reasons (though compression SHOULD still be applied to the IP and UDP headers). Since the RTP protocol specification suggests that the RTCP packet interval be scaled so that the aggregate RTCP bandwidth used by all participants in a session will be no more than 5% of the session bandwidth, there is not much to be gained from RTCP compression. Compressing out the SDES items would require a significant increase in the shared state that must be stored for each context ID. And, in order to allow compression when SDES information for several sources was sent through an RTP "mixer", it would be necessary to maintain a separate RTCP session context for each SSRC identifier. In a session with more than 255 participants, this would cause perfect thrashing of the context cache even when only one participant was sending data. Even though RTCP is not compressed, the fraction of the total bandwidth occupied by RTCP packets on the compressed link remains no more than 5% in most cases, assuming that the RTCP packets are sent as COMPRESSED_UDP packets. Given that the uncompressed RTCP traffic consumes no more than 5% of the total session bandwidth, then for a typical RTCP packet length of 90 bytes, the portion of the compressed bandwidth used by RTCP will be no more than 5% if the size of the payload in RTP data packets is at least 108 bytes. If the size of the RTP data payload is smaller, the fraction will increase, but is still less than 7% for a payload size of 37 bytes. For large data payloads, the compressed RTCP fraction is less than the uncompressed RTCP fraction (for example, 4% at 1000 bytes). 3.5. Compression of non-RTP UDP Packets As described earlier, the COMPRESSED_UDP packet MAY be used to compress UDP packets that don't carry RTP. Whatever data follows the UDP header is unlikely to have some constant values in the bits that correspond to usually constant fields in the RTP header. In particular, the SSRC field would likely change. Therefore, it is necessary to keep track of the non-RTP UDP packet streams to avoid using up all the context slots as the "SSRC field" changes (since that field is part of what identifies a particular RTP context). Those streams may each be given a context, but the encoder would set a flag in the context to indicate that the changing SSRC field should be ignored and COMPRESSED_UDP packets should always be sent instead of COMPRESSED_RTP packets. 4. Enhancements for links with high bit error rate and long round trip delay 4.1 The negative cache stream flag Certain streams, known or suspected to not be RTP, can be placed in a "negative cache" at the compressor, so only the IP and UDP headers are compressed. It is beneficial to notify the decompressor that the compressed stream is in the negative cache: for such streams the context is shorter - there is no need to include the RTP header, and all RTP-related calculations can be avoided. In this enhancement, a new flag bit "N" is added to the FULL_HEADER packet that initializes a context at the decompressor. The bit occupied by the new flag was previously always set to zero. If the N flag is set to 1, this indicates that no COMPRESSED_RTP packets will be transmitted in this context. This flag is only an optimization and the decompressor may choose to ignore it. Format of the FULL_HEADER length fields with the negative cache flag: For 8-bit context ID: +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |0|1| Generation| CID | First length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 0 |N| seq | Second length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ N=1: negative cache stream For 16-bit context ID: +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |1|1| Generation| 0 |N| seq | First length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ N=1: negative cache stream +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | CID | Second length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 4.2 Reject a new compressed stream In a point to point link the two nodes can agree on the number of compressed sessions they are prepared to support for this link. In an end-to-end scheme a host may have compressed sessions with many hosts and eventually may run out of resources. When the end-to-end tunnel is negotiated, the number of contexts needed may not be predictable. This enhancement allows the negotiated number of contexts to be larger than could be accommodated if many tunnels are established. Then, as context resources are consumed, an attempt to set up a new context may be rejected. The compressor initiates a compression of a stream by sending a FULL_HEADER packet. Currently if the decompressor has insufficient resources to decompress the new stream, it can send a CONTEXT_STATE packet to invalidate the newly compressed stream. The compressor does not know the reason for the invalidation: usually this happens when the decompressor gets out of synchronization due to packet loss. The compressor will most likely reattempt to compress this stream by sending another FULL_HEADER. This enhancement specifies that the decompressor may reject the compression of a stream by sending a REJECT message to the compressor. A REJECT message tells the compressor to stop compressing this stream. The REJECT message is a CONTEXT_STATE message with an additional flag: Type code = 1 :CONTEXT_STATE for 8-bit CID streams Type code = 2 : CONTEXT_STATE for16-bit CID streams Here is the format of CONTEXT_STATE packets with REJECT flags: 0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ |Type code=1: CS, 8-bit CID | +---+---+---+---+---+---+---+---+ | context count | +---+---+---+---+---+---+---+---+ | session context ID | +---+---+---+---+---+---+---+---+ | 1 |R=1| 0 | 0 | sequence | R is the REJECT flag +---+---+---+---+---+---+---+---+ | 0 | 0 | generation | +---+---+---+---+---+---+---+---+ . . . +---+---+---+---+---+---+---+---+ | session context ID | +---+---+---+---+---+---+---+---+ | 1 |R=1| 0 | 0 | sequence | R is the REJECT flag +---+---+---+---+---+---+---+---+ | 0 | 0 | generation | +---+---+---+---+---+---+---+---+ 0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ |Type code=2: CS, 16-bit CID| +---+---+---+---+---+---+---+---+ | context count | +---+---+---+---+---+---+---+---+ | | + session context ID + | | +---+---+---+---+---+---+---+---+ | 1 |R=1| 0 | 0 | sequence | R is the REJECT flag +---+---+---+---+---+---+---+---+ | 0 | 0 | generation | +---+---+---+---+---+---+---+---+ . . . +---+---+---+---+---+---+---+---+ | | + session context ID + | | +---+---+---+---+---+---+---+---+ | 1 |R=1| 0 | 0 | sequence | R is the REJECT flag +---+---+---+---+---+---+---+---+ | 0 | 0 | generation | +---+---+---+---+---+---+---+---+ The session CID, sequence and generation are taken from the FULL_HEADER. The compressor may decide to wait for a while before attempting to compress additional streams destined to the rejecting host. 4.3 Including IP ID in the UDP checksum A UDP checksum can be used by the decompressor to verify validity of the packet it reconstructed, especially when the 'twice' algorithm is used. When the 'twice' algorithm was defined in RFC 2507 and subsequently incorporated into RFC 2508, the fact that the IP ID field is not included in the checksum was overlooked. Since the IP ID field is conveyed with a delta value, accurate reconstruction of the IP ID field cannot be verified using the current specifications. This enhancement modifies the function of the UDP checksum to include the IP ID value, but only between the compressor and decompressor. That is, when a UDP checksum is present (nonzero), the compressor will 1's complement subtract the IP ID value from the UDP checksum before compression and the decompressor will 1's complement add the IP ID value to the UDP checksum after any validation operations and before delivering the packet further downstream. 4.4 CRTP Headers Checksum When a UDP checksum is not present (has value zero) in a stream, the compressor MAY replace it with a 16-bit headers checksum (HDRCKSUM). The HDRCKSUM can be used to validate the IP ID and all the headers in the reconstructed packet. Hence it can be used by the decompressor to validate reconstructed packets when 'twice' is used, and to validate every 16'th packet as recommended in RFC2508, Section 3.3.5. A new flag in the FULL_HEADER packet, as specified below, indicates when set that all COMPRESSED_UDP and COMPRESSED_RTP packets sent in that context will have HDRCKSUM inserted. If a packet in the same stream subsequently arrives at the compressor with a UDP checksum present, then a new FULL_HEADER packet must be sent with the flag cleared to re-establish the context. The HDRCKSUM is calculated in the same way as a UDP checksum, but includes only the pseudo-IP header (as defined for UDP), the IP ID (as in Section 2.3), the UDP header and for COMPRESSED_RTP packets, the fixed part of the RTP header (first 12 bytes). The extended part of the RTP header and the RTP data will not be included in the HDRCKSUM. The HDRCKSUM is placed in the COMPRESSED_UDP or COMPRESSED_RTP packets where a UDP checksum would have been. The decompressor MUST zero out the UDP checksum field in the reconstructed packets. The HDRCKSUM does not validate the RTP data. If the link layer is configured to deliver packets without checking for errors, errors in the RTP data will not be detected. Over such links, the compressor SHOULD add the HDRCKSUM if a UDP checksum is not present, and the decompressor SHOULD validate each reconstructed packet to make sure that at least the headers are correct. This ensures that the packet will be delivered to the right destination. If only HDRCKSUM is available, the RTP data will be delivered even if it includes errors. This might be a desirable feature for applications that can tolerate errors in the RTP data. Same holds for the extended part of the RTP header. Here is the format of the FULL_HEADER length fields with the new flag that indicates that a header checksum will be added in COMPRESSED_UDP and COMPRESSED_RTP packets: For 8-bit context ID: +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |0|1| Generation| CID | First length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | 0 |C|N| seq | Second length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ C=1: HDRCKSUM will be added For 16-bit context ID: +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |1|1| Generation| 0 |C|N| seq | First length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ C=1: HDRCKSUM will be added +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | CID | Second length field +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ 4.5 Enhancement to COMPRESSED_UDP packet format (CU*) The COMPRESSED_UDP packet includes the whole RTP header, so it can restore all RTP-related parameters at the decompressor. It is also specified to reset the delta RTP timestamp to zero and the delta RTP sequence number to zero.It can also convey a new value for the delta IP ID. It is possible to accommodate some packet loss between the compressor and decompressor using the "twice" algorithm in RFC 2508, but this requires reliably communicating the absolute values and the deltas for the differential fields. The reliability of communication of the absolute values in the RTP header can be increased by sending a COMPRESSED_UDP packet repeatedly, but this resets the delta timestamp. RFC 2508 describes the format of COMPRESSED_UDP as being the same as COMPRESSED_RTP except that the M, S and T bits are always 0 and the corresponding delta fields are never included. This enhancement changes that specification to say that the T bit may be nonzero to indicate that the RTP timestamp delta is included explicitly rather than being reset to zero. Sometimes it is necessary to change just a few fields of the RTP header. A second part of this enhancement adds more flag bits to the COMPRESSED_UDP packet to select individual uncompressed fields of the RTP header to be included in the packet. Since there are flag bits to indicate inclusion of both delta values and absolute values, the flag nomenclature is changed. The original S,T,I bits which indicate the inclusion of deltas are renamed dS, dT, dI, and the inclusion of absolute values is indicated by S,T,I. The M bit is absolute as before. The format of the flags/sequence byte for the original COMPRESSED_UDP packet is shown here for reference: +---+---+---+---+---+---+---+---+ | 0 | 0 | 0 |dI | link sequence | +---+---+---+---+---+---+---+---+ The new definition of the flags/sequence byte plus an extension flags byte is as follows, where the new F flag indicates the inclusion of the extension flags byte: +---+---+---+---+---+---+---+---+ | F | I |dT |dI | link sequence | +---+---+---+---+---+---+---+---+ : M : S : T :pt : CC : (if F = 1) +...+...+...+...+...............+ dI = delta IP ID dT = delta RTP timestamp I = absolute IP ID F = additional flags byte M = marker bit S = absolute RTP sequence number T = absolute RTP timestamp pt = RTP payload type CC = number of CSRC identifiers Some short notations: FH FULL_HEADER CR COMPRESSED_RTP CR+ COMPRESSED_RTP with delta fields CU COMPRESSED_UDP CU* enhanced COMPRESSED_UDP When F=0, there is only one flags byte, and the only available flags are: dI, dT and I. In this case the packet includes the full RTP header If dT=0, the decompressor sets deltaT to 0 If dI=0, the decompressor sets deltaI to 1 Some example packet formats will illustrate the use of the new flags. First, a 'traditional' COMPRESSED_UDP with full RTP header, when F=0: 0 1 2 3 4 5 6 7 +...............................+ : msb of session context ID : (if 16-bit CID) +-------------------------------+ | lsb of session context ID | +---+---+---+---+---+---+---+---+ |F=0| I |dT |dI | link sequence | +---+---+---+---+---+---+---+---+ : : + UDP checksum + (if nonzero in context) : : +...............................+ : : + "RANDOM" fields + (if encapsulated) : : +...............................+ : delta IPv4 ID : (if dI = 1) +...............................+ : delta RTP timestamp : (if dT = 1) +...............................+ : : + IPv4 ID + (if I = 1) : : +...............................+ | UDP data | : (uncompressed RTP header) : When F=1, there is an additional flags byte and the available flags are: dI, dT, I, M, S, T, pt, CC. In this case the packet does not include the full RTP header, but includes selected fields from the RTP header as specified by the flags. The delta values of the context are not reset even if they are not specified in the packet: If dT=0, the decompressor KEEPS THE CURRENT deltaT (and DOES NOT set the deltaT to 0) If dI=0, the decompressor KEEPS THE CURRENT deltaI (and DOES NOT set the the deltaI to 1) A CU* packet is similar in contents and behavior to a CR packet, but it has more flag bits, some of which correspond to absolute values for RTP header fields. COMPRESSED_UDP with individual RTP fields, when F=1 : 0 1 2 3 4 5 6 7 +...............................+ : msb of session context ID : (if 16-bit CID) +-------------------------------+ | lsb of session context ID | +---+---+---+---+---+---+---+---+ |F=1| I |dT |dI | link sequence | +---+---+---+---+---+---+---+---+ : M : S : T :pt : CC : +...+...+...+...+...............+ : : + UDP checksum + (if nonzero in context) : : +...............................+ : : : "RANDOM" fields : (if encapsulated) : : +...............................+ : delta IPv4 ID : (if dI = 1) +...............................+ : delta RTP timestamp : (if dT = 1) +...............................+ : : + IPv4 ID + (if I = 1) : : +...............................+ : : + RTP sequence number + (if S = 1) : : +...............................+ : : + + : : + RTP timestamp + (if T = 1) : : + + : : +...............................+ : RTP payload type : (if pt = 1) +...............................+ : : : CSRC list : (if CC > 0) : : +...............................+ : : : RTP header extension : (if X set in context) : : +-------------------------------+ | | / RTP data / / / | | +-------------------------------+ : padding : (if P set in context) +...............................+ Usage for the CU* packet: It is useful for the compressor to periodically refresh the state of the decompressor to avoid having the decompressor send CONTEXT_STATE messages in the case of unrecoverable packet loss. Using the flags F=0 I dI dT, this CU* packet refreshes all the context parameters. When compression is done over a lossy link with a long round trip delay, we want to minimize context invalidation. If the delta values are changing frequently, the context might get invalidated often. In such cases the compressor may choose to include absolute values in the CRTP packets instead of delta values, using CU* packets with the flags: F=1, and any of S, T, I as necessary. 4.6 Acknowledgement packet (ACK packet) The ACK packet will be sent from decompressor to compressor to indicate receipt of a compressed packet with the ACK'd RTP sequence number. It's a CONTEXT_STATE packet with type codes 4 and 5. The ACK packet is to be used in a separately negotiated mode of operation as described in the next section. Type code = 4 : ACK a packet of a context with 8-bit CID Type code = 5 : ACK a packet of a context with 16-bit CID The format for the ACK packet is: 0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ | Type code=4: ACK, 8-bit CID | +---+---+---+---+---+---+---+---+ | context count | +---+---+---+---+---+---+---+---+ | session context ID | +---+---+---+---+---+---+---+---+ | | + RTP sequence number + | | +---+---+---+---+---+---+---+---+ . . . +---+---+---+---+---+---+---+---+ | session context ID | +---+---+---+---+---+---+---+---+ | | + RTP sequence number + | | +---+---+---+---+---+---+---+---+ 0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+ | Type code=5: ACK, 16-bit CID | +---+---+---+---+---+---+---+---+ | context count | +---+---+---+---+---+---+---+---+ | | + session context ID + | | +---+---+---+---+---+---+---+---+ | | + RTP sequence number + | | +---+---+---+---+---+---+---+---+ . . . +---+---+---+---+---+---+---+---+ | | + session context ID + | | +---+---+---+---+---+---+---+---+ | | + RTP sequence number + | | +---+---+---+---+---+---+---+---+ 4.6.1 CRTP operation in ACK mode This mode of operation is optional and must be negotiated per link. 4.6.1.1 Description of the ACK mode The ACK mode is a mode of operation in which the compressor and decompressor continuously verify that their context states are synchronized. The compressor repeatedly notifies the decompressor about changes in the context state, until the decompressor acknowledges reception of the changes by sending ACK packets to the decompressor. This effort of synchronizing the context states helps minimize context invalidation. The context state shared between the compressor and decompressor includes all the fields of the uncompressed headers and the first order differences (delta fields) of the fields that change by a constant value from packet to packet. Each field follows its known change pattern: either stays constant or is incremented by its corresponding delta field. Fields that follow their change pattern are compressed. They are reconstructed by the decompresor from the context state at the decompressor. Correct decompression of a packet depends on whether the context state at the compressor when the packet is compressed and sent is identical to the context state at the decompressor when that packet is received and decompressed. When a field changes in a way that is different from its change pattern, the compressor assigns a new value to the field, and stores it in the context state at the compressor side. The decompressor must be informed about the change so that it can update the state on its side to match the state at the compressor. The compressor notifies the decompressor about such changes by including information about the changed field in the compressed packet. (for example if dT was assigned a new value, the compressor can send a CR+ packet that includes dT). The context is not synchronized until the decompressor receives the packet that includes the changed field and updates its state accordingly. The decompressor indicates reception of the change by sending an ACK packet to the compressor. The ACK packet includes the RTP sequence number of the packet that it is ACK'ing, so the compressor can identify which packet is ACK'd. The compressor can't assume that the decompressor received the change until the ACK packet is received. Depending on the round trip delay of the link, the compressor might have to send a few more packets before the ACK from the decompressor arrives. In this case the compressor must repeat the change in all subsequent packets. Reception of the ACK is an indication that the decompressor updated its context with the changed value. Now that their contexts are synchronized again, the compressor can stop including the changed field in the compressed packets. The decompressor must be able to recognize the repeat packets (the packets that repeat the same change and were sent while the compressor was waiting for the ACK packet). Those repeat packets don't require an ACK. If in the process of changing some fields additional changes are required, the compressor will switch to send packets that include all changes. The decompressor must ACK one of the packets that include all the changes. The compressor and decompressor must be in full agreement about which packets must be ACK'd: packets that include changes are larger in size, and if they are not ACK'd, the changes are repeated in all subsequent packets, and bandwidth is wasted. Let's summarize which packets require an ACK: 1. A Packet that assigns a value to any context state field must be ACK'd. This includes FH and CU packets because they initialize fields in the context state. 2. Repeat packets don't require an ACK How are repeat packets identified? A packet is considered to be a repeat packet if: 1. It updates the same fields as the previous packet 2. Each field is updated by a value that is equal to the one assigned to this field in the previous packet plus the corresponding delta for this field, when applicable. 4.6.1.2 The Random IP ID The IP ID change pattern is to be incremented by dI. In some implementations, the IP ID counter is shared across multiple streams, so as a result of the varying mix of packets the increment for any particular stream is not constant. When compressing such a stream, the compressor must include in each packet either dI or I. It is recommended to include I rather then dI because a loss of a packet that includes a new delta value dI will invalidate the context. According to the rules set above, each packet will have to be ACK'd. To correct this we'll define a new change pattern for the IP ID: random value. The IP ID assumes this change pattern when dI is set to be 0. We add a rule to the ACK rules: 3. When the value of dI is 0, packets that update only the IP ID field don't require an ACK. And add to rule 2 of the repeat packet rules: 2. Each field is updated by a value that is equal to the one assigned to this field in the previous packet plus the corresponding delta for this field, when applicable. An exception to this rule is the IP ID field: if the value of dI is 0, the IP ID may be assigned any value. 4.6.1.3 Implementation hints when using the ACK scheme 1. When a delta field is updated, add the matching absolute field too (dT and T, dI and I). Loss of a packet that updates only the delta value can easily cause context invalidation. 2. Set dI=0 when the IP ID is changing randomly, and include I in all packets. 3. If you ACK'd a packet, but the number of repeat packets exceed your estimate, ACK again (your previous ACK was probably lost) Here is an example to demonstrate the usage of the ACK scheme. In this stream the audio codec sends a sample every 10 msec The first talk spurt is 1 second long. Then there are 2 seconds silence, then another talk spurt. When there is no loss on the link, we can use the following sequence: (The deltaID is not constant so we send deltaID in each packet) seq# Time pkt type 1 10 FH 2 20 CR+ dI dT=10 3 30 CR+ dI 4 40 CR+ dI ... 100 1000 CR+ dI 101 3010 CR+ dI dT=2010 102 3020 CR+ dI dT=10 103 3030 CR+ dI 104 3040 CR+ dI ... In the above sequence if a packet is lost, we cannot recover ('twice' will not work due to the random IP ID) and the context must be invalidated. Here is the same sequence using the ACK scheme(CU* is the enhanced CU): seq# Time pkt type flags 1 10 FH FH must be ACK'd 2 20 FH repeat ACK 1 3 30 CU* 1 I dT dI M 0 T 0 I T=30 dT=10 dI=0 dI,dT changed (packet 3 was lost) (I and T sent too) 4 40 CU* 1 I dT dI M 0 T 0 I T=40 dT=10 dI=0 repeat 5 50 CU* 1 I dT dI M 0 T 0 I T=50 dT=10 dI=0 repeat 6 60 CU* 1 I dT dI M 0 T 0 I T=60 dT=10 dI=0 repeat ACK 4 == got new dI=0 and dT=10 at T=40. dI was set to 0, so I does not require an ACK. No need to ACK 5 and 6: repeat packets 7 70 CU* 1 I 0 0 M 0 0 0 I 8 80 CU* 1 I 0 0 M 0 0 0 I ... 100 1000 CU* 1 I 0 0 M 0 0 0 I 101 3010 CU* 1 I 0 0 M 0 T 0 I T=3010 T changed, keep deltas! 102 3020 CU* 1 I 0 0 M 0 T 0 I T=3020 repeat ACK 101 == got new T at sequence 101 No need to ACK packet 102 because 3010 + dT = 3020 If 101 is lost, 102 will be ACK'd 103 3030 CU* 1 I 0 0 M 0 0 0 I 104 3040 CU* 1 I 0 0 M 0 0 0 I ... The same sequences, when delta IP ID is constant: seq# Time pkt type 1 10 FH 2 20 CR+ dI dT=10 3 30 CR 4 40 CR ... 100 1000 CR 101 3010 CR+ dT=2010 102 3020 CR+ dT=10 103 3030 CR 104 3040 CR ... seq# Time pkt type flags 1 10 FH FH must be ACK'd 2 20 FH repeat ACK 1 3 30 CU* 1 I dT dI M 0 T 0 I dI T=30 dT=10 dI,dT changed (packet 3 was lost) (I and T sent too) 4 40 CU* 1 I dT dI M 0 T 0 I dI T=40 dT=10 repeat 5 50 CU* 1 I dT dI M 0 T 0 I dI T=50 dT=10 repeat 6 60 CU* 1 I dT dI M 0 T 0 I dI T=60 dT=10 repeat ACK 4 == got new dI and dT=10 at T=40. No need to ACK 5 and 6: no changes 7 70 CR 8 80 CR ... 100 1000 CR 101 3010 CU* 1 0 0 0 M 0 T 0 T=3010 T changed, keep deltas! 102 3020 CU* 1 0 0 0 M 0 T 0 T=3020 repeat ACK 101 == got new T at sequence 101 No need to ACK packet 102 because 3010 + dT = 3020 If 101 is lost, 102 will be ACK'd 103 3030 CR 104 3040 CR ... 4.7 Accept a new compressed stream Lack of any feedback from the decompressor might indicate either that everything goes well (the stream is decompressed successfully), or that nothing goes well (the link is down, the decompressor is not functioning, the decompressor is out of sync but there is no back channel). The compressor initiates a compression of a stream by sending a FULL_HEADER packet. This enhancement specifies that the decompressor SHOULD acknowledge the reception of the FULL_HEADER packet by sending an ACK packet (see 2.6) with the sequence number of the FULL_HEADER packet. This reassures the compressor that the new compressed stream is received and decompressed. It also indicates that a back channel exists. 4.8 CRTP operation in 'N' mode This scheme is similar to the ACK scheme in that the compressor tries to keep the decompressor in sync by sending changes multiple times. The 'N' is a number that represents the quality of the link between the hosts, and it means that the probability of more than 'N' adjacent packets getting lost on this link is small. For every change in a base value or a delta value, if the compressor includes the change in N+1 consecutive packets, there is a very good chance that the compressor and decompressor can stay in sync using the 'twice' algorithm. CONTEXT_STATE packets should also be repeated N+1 times (using the same sequence number). It is up to the implementation to find a scheme to derive an appropriate N for a link. This scheme may be used at any time and does not require negotiation. Here is the same example in 'N' mode, when N=2 and deltaID is constant: seq# Time pkt type flags 1 10 FH 2 20 FH repeat constant fields 3 30 FH repeat constant fields 4 40 CU* 1 I dT dI M 0 T 0 I dI T=40 dT=10 5 50 CU* 1 I dT dI M 0 T 0 I dI T=50 dT=10 repeat delta 6 60 CU* 1 I dT dI M 0 T 0 I dI T=60 dT=10 repeat delta 7 70 CR 8 80 CR ... 100 1000 CR 101 3010 CU* 1 0 0 0 M 0 T 0 T=3010 T changed, keep deltas! 102 3020 CU* 1 0 0 0 M 0 T 0 T=3020 repeat updated T 103 3030 CU* 1 0 0 0 M 0 T 0 T=3030 repeat updated T 104 3040 CR 105 3050 CR ... 4.9 Select mode of operation As link conditions change, it might be necessary to change the mode of operation from N mode to ACK mode and vice versa. Mode changes are initiated by the compressor and acknowledged by the decompressor. When acknowledging a request to move to N mode, the decompressor may suggest a different N to use (for example based on loss patterns seen by the decompressor). The decompressor may send multiple acknowledgement packets to one request packet, e.g. send an ACK-Nmode packet when the decompressor suggests to change the N that is currently being used. The compressor determines the N to be used, and it MAY be different than the value of the parameter N in the REQ-Nmode and ACK-Nmode packets exchanged between the compressor and decompressor. If the compressor receives an ACK packet that does not match the current mode, the compressor SHOULD send another REQ packet to set the right mode. Hence the compressor can use any of the ACK packets to verify the current mode: if the ACK does not match the current mode, the compressor will send a REQ to set the mode. 0 1 2 3 4 5 6 7 +---+---+---+---+---+---+---+---+---+---+ | Type code=6: Select operation mode | +---+---+---+---+---+---+---+---+---+---+ | Opcode | +---+---+---+---+---+---+---+---+---+---+ : Parameter : +.......................................+ Opcode Mnemonic Description Parameter -------------------------------------------------------------------------------- 1 REQ-Nmode Request to move to N mode N to be used 2 ACK-Nmode Acknowledge move to N mode N to be used 3 REQ-ACKmode Request to move to ACK mode none 4 ACK-ACKmode Acknowledge move to ACK mode none 4.10 Negotiating usage of enhanced-CRTP and ACK scheme RFC 2509 [IPCPHP] specifies how the use of CRTP is negotiated on PPP links using the IP Compression Protocol option of IPCP: IPCP option 2: IP compression protocol protocol 0x61 indicates RFC 2507 header compression sub-option 1 enables use of COMPRESSED_RTP, COMPRESSED_UDP and CONTEXT_STATE as specified in RFC 2508 For the enhancements defined in this document, two new sub-options are added: sub-option 2 (length=2) : enables use of CRTP with enhancements 4.1 - 4.5 and 4.7 (all except ACK mode) sub-option 3 (length=2) : enables use of CRTP with enhancements 4.1 - 4.7 (ACK scheme) 5. Interaction With Segmentation A segmentation scheme may be used in conjunction with RTP header compression to allow small, real-time packets to interrupt large, presumably non-real-time packets in order to reduce delay. It is assumed that the large packets bypass the compressor and decompressor since the interleaving would modify the sequencing of packets at the decompressor and cause the appearance of errors. Header compression should be less important for large packets since the overhead ratio is smaller. If some packets from an RTP session context are selected for segmentation (perhaps based on size) and some are not, there is a possibility of re-ordering. This would reduce the compression efficiency because the large packets would appear as lost packets in the sequence space. However, this should not cause more serious problems because the RTP sequence numbers should be reconstructed correctly and will allow the application to correct the ordering. Link errors detected by the segmentation scheme using its own sequencing information MAY be indicated to the compressor with an advisory CONTEXT_STATE message just as for link errors detected by the link layer itself. The context ID byte is placed first in the COMPRESSED_RTP header so that this byte MAY be shared with the segmentation layer if such sharing is feasible and has been negotiated. Since the compressor may assign context ID values arbitrarily, the value can be set to match the context identifier from the segmentation layer. 6. Negotiating Compression The use of IP/UDP/RTP compression over a particular link is a function of the link-layer protocol. It is expected that such negotiation will be defined separately for PPP [4], for example. The following items MAY be negotiated: o The size of the context ID. o The maximum size of the stack of headers in the context. o A context-specific table for decoding of delta values. o Using CRTP enhancements. 7. Acknowledgments Several people have contributed to the design of this compression scheme and related problems. Scott Petrack initiated discussion of RTP header compression in the AVT working group at Los Angeles in March, 1996. Carsten Bormann has developed an overall architecture for compression in combination with traffic control across a low- speed link, and made several specific contributions to the scheme described here. David Oran independently developed a note based on similar ideas, and suggested the use of PPP Multilink protocol for segmentation. Mikael Degermark has contributed advice on integration of this compression scheme with the IPv6 compression framework. 8. References: [1] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson, "RTP: A Transport Protocol for real-time applications", RFC 1889, January 1996. [2] Jacobson, V., "TCP/IP Compression for Low-Speed Serial Links", RFC 1144, February 1990. [3] Degermark, M., Nordgren, B. and S. Pink, "Header Compression for IPv6", RFC 2507, February 1999. [4] Simpson, W., "The Point-to-Point Protocol (PPP)", STD 51, RFC 1661, July 1994. [5] Hoffman, D., Fernando, G., Goyal, V. and M. Civanlar, "RTP Payload Format for MPEG1/MPEG2 Video", RFC 2250, January 1998. 9. Security Considerations Because encryption eliminates the redundancy that this compression scheme tries to exploit, there is some inducement to forego encryption in order to achieve operation over a low-bandwidth link. However, for those cases where encryption of data and not headers is satisfactory, RTP does specify an alternative encryption method in which only the RTP payload is encrypted and the headers are left in the clear. That would allow compression to still be applied. A malfunctioning or malicious compressor could cause the decompressor to reconstitute packets that do not match the original packets but still have valid IP, UDP and RTP headers and possibly even valid UDP check-sums. Such corruption may be detected with end-to-end authentication and integrity mechanisms which will not be affected by the compression. Constant portions of authentication headers will be compressed as described in [3]. No authentication is performed on the CONTEXT_STATE control packet sent by this protocol. An attacker with access to the link between the decompressor and compressor could inject false CONTEXT_STATE packets and cause compression efficiency to be reduced, probably resulting in congestion on the link. However, an attacker with access to the link could also disrupt the traffic in many other ways. A potential denial-of-service threat exists when using compression techniques that have non-uniform receiver-end computational load. The attacker can inject pathological datagrams into the stream which are complex to decompress and cause the receiver to be overloaded and degrading processing of other streams. However, this compression does not exhibit any significant non-uniformity. A security review of this protocol found no additional security considerations. 10. Authors' Addresses Stephen L. Casner Packet Design, Inc. 66 Willow Place Menlo Park, CA 94025 United States of America Email: casner@acm.org Van Jacobson Packet Design, Inc. 66 Willow Place Menlo Park, CA 94025 United States of America EMail: van@packetdesign.com Tmima Koren Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 United States of America Email: tmima@cisco.com John Geevarghese Telseon Inc, 480 S. California Palo-Alto, CA-94306 Email: geevjohn@hotmail.com Bruce Thompson Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 United States of America Email: brucet@cisco.com Dan Wing Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 United States of America Email: dwing@cisco.com Patrick Ruddy Cisco Systems, Inc. 3rd Floor, 96 Commercial Street Edinburgh EH6 6LX Scotland Email: pruddy@cisco.com Alex Tweedly Cisco Systems, Inc. 3 The Square, Stockley Park Uxbridge, Middlesex UB11 1BN United Kingdom Email: agt@cisco.com 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 Society 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 followed, 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 MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Datatracker
draft-ietf-avt-crtp-enhance-01
Internet-Draft
