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Network Working Group                                         C. BormannRequest for Comments: 2689                       Universitaet Bremen TZICategory: Informational                                   September 1999Providing Integrated Services over Low-bitrate LinksStatus of this Memo   This memo provides information for the Internet community.  It does   not specify an Internet standard of any kind.  Distribution of this   memo is unlimited.Copyright Notice   Copyright (C) The Internet Society (1999).  All Rights Reserved.Abstract   This document describes an architecture for providing integrated   services over low-bitrate links, such as modem lines, ISDN B-   channels, and sub-T1 links.  It covers only the lower parts of the   Internet Multimedia Conferencing Architecture [1]; additional   components required for application services such as Internet   Telephony (e.g., a session initiation protocol) are outside the scope   of this document.  The main components of the architecture are: a   real-time encapsulation format for asynchronous and synchronous low-   bitrate links, a header compression architecture optimized for real-   time flows, elements of negotiation protocols used between routers   (or between hosts and routers), and announcement protocols used by   applications to allow this negotiation to take place.1.  Introduction   As an extension to the "best-effort" services the Internet is well-   known for, additional types of services ("integrated services") that   support the transport of real-time multimedia information are being   developed for, and deployed in the Internet.  Important elements of   this development are:   -  parameters for forwarding mechanisms that are appropriate for      real-time information [11,12],   -  a setup protocol that allows establishing special forwarding      treatment for real-time information flows (RSVP [4]),   -  a transport protocol for real-time information (RTP/RTCP [6]).Bormann                      Informational                      [Page 1]

RFC 2689       Integrated Services over Low-bitrate Links September 1999   In addition to these elements at the network and transport levels of   the Internet Multimedia Conferencing Architecture [1], further   components are required to define application services such as   Internet Telephony, e.g., protocols for session initiation and   control.  These components are outside the scope of this document.   Up to now, the newly developed services could not (or only very   inefficiently) be used over forwarding paths that include low-bitrate   links such as 14.4, 33.6, and 56 kbit/s modems, 56 and 64 kbit/s ISDN   B-channels, or even sub-T1 links.  The encapsulation formats used on   these links are not appropriate for the simultaneous transport of   arbitrary data and real-time information that has to meet stringent   delay requirements.  Transmission of a 1500 byte packet on a 28.8   kbit/s modem link makes this link unavailable for the transmission of   real-time information for about 400 ms.  This adds a worst-case delay   that causes real-time applications to operate with round-trip delays   on the order of at least a second -- unacceptable for real-time   conversation.  In addition, the header overhead associated with the   protocol stacks used is prohibitive on low-bitrate links, where   compression down to a few dozen bytes per real-time information   packet is often desirable.  E.g., the overhead of at least 44   (4+20+8+12) bytes for HDLC/PPP, IP, UDP, and RTP completely   overshadows typical audio payloads such as the 19.75 bytes needed for   a G.723.1 ACELP audio frame -- a 14.4 kbit/s link is completely   consumed by this header overhead alone at 40 real-time frames per   second total (i.e., at 25 ms packetization delay for one stream or 50   ms for two streams, with no space left for data, yet).  While the   header overhead can be reduced by combining several real-time   information frames into one packet, this increases the delay incurred   while filling that packet and further detracts from the goal of   real-time transfer of multi-media information over the Internet.   This document describes an approach for addressing these problems.   The main components of the architecture are:   -  a real-time encapsulation format for asynchronous and synchronous      low-bitrate links,   -  a header compression architecture optimized for real-time flows,   -  elements of negotiation protocols used between routers (or between      hosts and routers), and   -  announcement protocols used by applications to allow this      negotiation to take place.Bormann                      Informational                      [Page 2]

RFC 2689       Integrated Services over Low-bitrate Links September 19992.  Design Considerations   The main design goal for an architecture that addresses real-time   multimedia flows over low-bitrate links is that of minimizing the   end-to-end delay.  More specifically, the worst case delay (after   removing possible outliers, which are equivalent to packet losses   from an application point of view) is what determines the playout   points selected by the applications and thus the delay actually   perceived by the user.   In addition, any such architecture should obviously undertake every   attempt to maximize the bandwidth actually available to media data;   overheads must be minimized.   An important component of the integrated services architecture is the   provision of reservations for real-time flows.  One of the problems   that systems on low-bitrate links (routers or hosts) face when   performing admission control for such reservations is that they must   translate the bandwidth requested in the reservation to the one   actually consumed on the link.  Methods such as data compression   and/or header compression can reduce the requirements on the link,   but admission control can only make use of the reduced requirements   in its calculations if it has enough information about the data   stream to know how effective the compression will be.  One goal of   the architecture therefore is to provide the integrated services   admission control with this information.  A beneficial side effect   may be to allow the systems to perform better compression than would   be possible without this information.  This may make it worthwhile to   provide this information even when it is not intended to make a   reservation for a real-time flow.3.  The Need for a Concerted Approach   Many technical approaches come to mind for addressing these problems,   in particular a new form of low-delay encapsulation to address delay   and header compression methods to address overhead.  This section   shows that these techniques should be combined to solve the problem.3.1.  Real-Time Encapsulation   The purpose of defining a real-time link-layer encapsulation protocol   is to be able to introduce newly arrived real-time packets into the   link-layer data stream without having to wait for the currently   transmitted (possibly large) packet to end.  Obviously, a real-time   encapsulation must be part of any complete solution as the problem of   delays induced by large frames on the link can only be solved on this   layer.Bormann                      Informational                      [Page 3]

RFC 2689       Integrated Services over Low-bitrate Links September 1999   To be able to switch to a real-time packet quickly in an interface   driver, it is first necessary to identify packets that belong to   real-time flows.  This can be done using a heuristic approach (e.g.,   favor the transmission of highly periodic flows of small packets   transported in IP/UDP, or use the IP precedence fields in a specific   way defined within an organization).  Preferably, one also could make   use of a protocol defined for identifying flows that require special   treatment, i.e. RSVP.  Of the two service types defined for use with   RSVP now, the guaranteed service will only be available in certain   environments; for this and various other reasons, the service type   chosen for many adaptive audio/video applications will most likely be   the controlled-load service.  Controlled-load does not provide   control parameters for target delay; thus it does not unambiguously   identify those packet streams that would benefit most from being   transported in a real-time encapsulation format.  This calls for a   way to provide additional parameters in integrated services flow   setup protocols to control the real-time encapsulation.   Real-time encapsulation is not sufficient on its own, however: Even   if the relevant flows can be appropriately identified for real-time   treatment, most applications simply cannot operate properly on low-   bitrate links with the header overhead implied by the combination of   HDLC/PPP, IP, UDP, and RTP, i.e. they absolutely require header   compression.3.2.  Header Compression   Header compression can be performed in a variety of elements and at a   variety of levels in the protocol architecture.  As many vendors of   Internet Telephony products for PCs ship applications, the approach   that is most obvious to them is to reduce overhead by performing   header compression at the application level, i.e. above transport   protocols such as UDP (or actually by using a non-standard,   efficiently coded header in the first place).   Generally, header compression operates by installing state at both   ends of a path that allows the receiving end to reconstruct   information omitted at the sending end.  Many good techniques for   header compression (RFC 1144, [2]) operate on the assumption that the   path will not reorder the frames generated.  This assumption does not   hold for end-to-end compression; therefore additional overhead is   required for resequencing state changes and for compressed packets   making use of these state changes.   Assume that a very good application level header compression solution   for RTP flows could be able to save 11 out of the 12 bytes of an RTP   header [3].  Even this perfect solution only reduces the total header   overhead by 1/4.  It would have to be deployed in all applications,Bormann                      Informational                      [Page 4]

RFC 2689       Integrated Services over Low-bitrate Links September 1999   even those that operate on systems that are attached to higher-   bitrate links.   Because of this limited effectiveness, the AVT group that is   responsible for RTP within the IETF has decided to not further pursue   application level header compression.   For router and IP stack vendors, the obvious approach is to define   header compression that can be negotiated between peer routers.   Advanced header compression techniques now being defined in the IETF   [2] certainly can relieve the link from significant parts of the   IP/UDP overhead (i.e., most of 28 of the 44 bytes mentioned above).   One of the design principles of the new IP header compression   developed in conjunction with IPv6 is that it stops at layers the   semantics of which cannot be inferred from information in lower layer   (outer) headers.  Therefore, this header compression technique alone   cannot compress the data that is contained within UDP packets.   Any additional header compression technique runs into a problem: If   it assumes specific application semantics (i.e., those of RTP and a   payload data format) based on heuristics, it runs the risk of being   triggered falsely and (e.g. in case of packet loss) reconstructing   packets that are catastrophically incorrect for the application   actually being used.  A header compression technique that can be   operated based on heuristics but does not cause incorrect   decompression even if the heuristics failed is described in [7]; a   companion document describes the mapping of this technique to PPP   [10].   With all of these techniques, the total IP/UDP/RTP header overhead   for an audio stream can be reduced to two bytes per packet.  This   technology need only be deployed at bottleneck links; high-speed   links can transfer the real-time streams without routers or switches   expending CPU cycles to perform header compression.4.  Principles of Real-Time Encapsulation for Low-Bitrate Links   The main design goal for a real-time encapsulation is to minimize the   delay incurred by real-time packets that become available for sending   while a long data packet is being sent.  To achieve this, the   encapsulation must be able to either abort or suspend the transfer of   the long data packet.  As an additional goal is to minimize the   overhead required for the transmission of packets from periodic   flows, this strongly argues for being able to suspend a packet, i.e.   segment it into parts between which the real-time packets can be   transferred.Bormann                      Informational                      [Page 5]

RFC 2689       Integrated Services over Low-bitrate Links September 19994.1.  Using existing IP fragmentation   Transmitting only part of a packet, to allow higher-priority traffic   to intervene and then resuming its transmission later on, is a kind   of fragmentation.  Fragmentation is an existing functionality of the   IP layer: An IPv4 header already contains fields that allow a large   IP datagram to be fragmented into small parts.  A sender's "real-time   PPP" implementation might simply indicate a small MTU to its IP stack   and thus cause all larger datagrams to be fragmented down to a size   that allows the access delay goals to be met (this assumes that the   IP stack is able to priority-tag fragments, or that the PPP   implementation is able to correlate the fragments to the initial one   that carries the information relevant for prioritizing, or that only   initial fragments can be high-priority).  (Also, a PPP implementation   can negotiate down the MTU of its peer, causing the peer to fragment   to a small size, which might be considered a crude form of   negotiating an access delay goal with the peer system -- if that   system supports priority queueing at the fragment level.)   Unfortunately, a full, 20 byte IP header is needed for each fragment   (larger when IP options are used).  This limits the minimum size of   fragments that can be used without too much overhead.  (Also, the   size of non-final fragments must be a multiple of 8 bytes, further   limiting the choice.)  With path MTU discovery, IP level   fragmentation causes TCP implementations to use small MSSs -- this   further increases the per-packet overhead to 40 bytes per fragment.   In any case, fragmentation at the IP level persists on the path   further down to the datagram receiver, increasing the transmission   overheads and router load throughout the network.  With its high   overhead and the adverse effect on the Internet, IP level   fragmentation can only be a stop-gap mechanism when no other   fragmentation protocol is available in the peer implementation.4.2.  Link-Layer Mechanisms   Cell-oriented multiplexing techniques such as ATM that introduce   regular points where cells from a different packet can be   interpolated are too inefficient for low-bitrate links; also, they   are not supported by chips used to support the link layer in low-   bitrate routers and host interfaces.Bormann                      Informational                      [Page 6]

RFC 2689       Integrated Services over Low-bitrate Links September 1999   Instead, the real-time encapsulation should as far as possible make   use of the capabilities of the chips that have been deployed.  On   synchronous lines, these chips support HDLC framing; on asynchronous   lines, an asynchronous variant of HDLC that usually is implemented in   software is being used.  Both variants of HDLC provide a delimiting   mechanism to indicate the end of a frame over the link.  The obvious   solution to the segmentation problem is to combine this mechanism   with an indication of whether the delimiter terminates or suspends   the current packet.   This indication could be in an octet appended to each frame   information field; however, seven out of eight bits of the octet   would be wasted.  Instead, the bit could be carried at the start of   the next frame in conjunction with multiplexing information (PPP   protocol identifier etc.) that will be required here anyway.  Since   the real-time flows will in general be periodic, this multiplexing   information could convey (part of) the compressed form of the header   for the packet.  If packets from the real-time flow generally are of   constant length (or have a defined maximum length that is often   used), the continuation of the suspended packet could be immediately   attached to it, without expending a further frame delimiter, i.e.,   the interpolation of the real-time packet would then have zero   overhead.  Since packets from low-delay real-time flows generally   will not require the ability to be further suspended, the   continuation bit could be reserved for the non-real-time packet   stream.   One real-time encapsulation format with these (and other) functions   is described in ITU-T H.223 [13], the multiplex used by the H.324   modem-based videophone standard [14].  It was investigated whether   compatibility could be achieved with this specification, which will   be used in future videophone-enabled (H.324 capable) modems.   However, since the multiplexing capabilities of H.223 are limited to   15 schedules (definitions of sequences of packet types that can be   identified in a multiplex header), for general Internet usage a   superset or a more general encapsulation would have been required.   Also, a PPP-style negotiation protocol was needed instead of using   (and necessarily extending) ITU-T H.245 [15] for setting the   parameters of the multiplex.  In the PPP context, the interactions   with the encapsulations for data compression and link layer   encryption needed to be defined (including operation in the presence   of padding).  But most important, H.223 requires synchronous HDLC   chips that can be configured to send frames without an attached CRC,   which is not possible with all chips deployed in commercially   available routers; so complete compatibility was unachievable.Bormann                      Informational                      [Page 7]

RFC 2689       Integrated Services over Low-bitrate Links September 1999   Instead of adopting H.223, it was decided to pursue an approach that   is oriented towards compatibility both with existing hardware and   existing software (in particular PPP) implementations.  The next   subsection groups these implementations according to their   capabilities.4.3.  Implementation models   This section introduces a number of terms for types of   implementations that are likely to emerge.  It is important to have   these different implementation models in mind as there is no single   approach that fits all models best.4.3.1.  Sender types   There are two fundamental approaches to real-time transmission on   low-bitrate links:   Sender type 1      The PPP real-time framing implementation is able to control the      transmission of each byte being transmitted with some known,      bounded delay (e.g., due to FIFOs).  For example, this is      generally true of PC host implementations, which directly access      serial interface chips byte by byte or by filling a very small      FIFO.  For type 1 senders, a suspend/resume type approach will be      typically used: When a long frame is to be sent, the attempt is to      send it undivided; only if higher priority packets come up during      the transmission will the lower-priority long frame be suspended      and later resumed.  This approach allows the minimum variation in      access delay for high-priority packets; also, fragmentation      overhead is only incurred when actually needed.   Sender type 2      With type 2 senders, the interface between the PPP real-time      framing implementation and the transmission hardware is not in      terms of streams of bytes, but in terms of frames, e.g., in the      form of multiple (prioritized) send queues directly supported by      hardware.  This is often true of router systems for synchronous      links, in particular those that have to support a large number of      low-bitrate links.  As type 2 senders have no way to suspend a      frame once it has been handed down for transmission, they      typically will use a queues-of-fragments approach, where long      packets are always split into units that are small enough to      maintain the access delay goals for higher-priority traffic.      There is a trade-off between the variation in access delay      resulting from a large fragment size and the overhead that is      incurred for every long packet by choosing a small fragment size.Bormann                      Informational                      [Page 8]

RFC 2689       Integrated Services over Low-bitrate Links September 19994.3.2.  Receiver types   Although the actual work of formulating transmission streams for   real-time applications is performed at the sender, the ability of the   receiver to immediately make use of the information received depends   on its characteristics:   Receiver type 1      Type 1 receivers have full control over the stream of bytes      received within PPP frames, i.e., bytes received are available      immediately to the PPP real-time framing implementation (with some      known, bounded delay e.g. due to FIFOs etc.).   Receiver type 2      With type 2 receivers, the PPP real-time framing implementation      only gets hold of a frame when it has been received completely,      i.e., the final flag has been processed (typically by some HDLC      chip that directly fills a memory buffer).4.4.  Conclusion   As a result of the diversity in capabilities of current   implementations, there are now two specifications for real-time   encapsulation: One, the multi-class extension to the PPP multi-link   protocol, is providing the solution for the queues-of-fragments   approach by extending the single-stream PPP multi-link protocol by   multiple classes [8].  The other encapsulation, PPP in a real-time   oriented HDLC-like framing, builds on this specification end extends   it by a way to dynamically delimit multiple fragments within one HDLC   frame [9], providing the solution for the suspend/resume type   approach.5.  Principles of Header Compression for Real-Time Flows   A good baseline for a discussion about header compression is in the   new IP header compression specification that was designed in   conjunction with the development of IPv6 [2].  The techniques used   there can reduce the 28 bytes of IPv4/UDP header to about 6 bytes   (depending on the number of concurrent streams); with the remaining 4   bytes of HDLC/PPP overhead and 12 bytes for RTP the total header   overhead can be about halved but still exceeds the size of a G.723.1   ACELP frame.  Note that, in contrast to IP header compression, the   environment discussed here assumes the existence of a full-duplex PPP   link and thus can rely on negotiation where IP header compression   requires repeated transmission of the same information.  (The use of   the architecture of the present document with link layer multicasting   has not yet been examined.)Bormann                      Informational                      [Page 9]

RFC 2689       Integrated Services over Low-bitrate Links September 1999   Additional design effort was required for RTP header compression.   Applying the concepts of IP header compression, of the (at least) 12   bytes in an RTP header, 7 bytes (timestamp, sequence, and marker bit)   would qualify as RANDOM; DELTA encoding cannot generally be used   without further information since the lower layer header does not   unambiguously identify the semantics and there is no TCP checksum   that can be relied on to detect incorrect decompression.  Only a more   semantics-oriented approach can provide better compression (just asRFC 1144 can provide very good compression of TCP headers by making   use of semantic knowledge of TCP and its checksumming method).   For RTP packets, differential encoding of the sequence number and   timestamps is an efficient approach for certain cases of payload data   formats.  E.g., speech flows generally have sequence numbers and   timestamp fields that increase by 1 and by the frame size in   timestamp units, resp.; the CRTP (compressed RTP) specification makes   use of this relationship by encoding these fields only when the   second order difference is non-zero [7].6.  Announcement Protocols Used by Applications   As argued, the compressor can operate best if it can make use of   information that clearly identifies real-time streams and provides   information about the payload data format in use.   If these systems are routers, this consent must be installed as   router state; if these systems are hosts, it must be known to their   networking kernels.  Sources of real-time information flows are   already describing characteristics of these flows to their kernels   and to the routers in the form of TSpecs in RSVP PATH messages [4].   Since these messages make use of the router alert option, they are   seen by all routers on the path; path state about the packet stream   is normally installed at each of these routers that implement RSVP.   Additional RSVP objects could be defined that are included in PATH   messages by those applications that desire good performance over low-   bitrate links; these objects would be coded to be ignored by routers   that are not interested in them (class number 11bbbbbb as defined in   [4], section 3.10).   Note that the path state is available in the routers even when no   reservation is made; this allows informed compression of best-effort   traffic.  It is not quite clear, though, how path state could be torn   down quickly when a source ceases to transmit.Bormann                      Informational                     [Page 10]

RFC 2689       Integrated Services over Low-bitrate Links September 19997.  Elements of Hop-By-Hop Negotiation Protocols   The IP header compression specification attempts to account for   simplex and multicast links by providing information about the   compressed streams only in the forward direction.  E.g., a full   IP/UDP header must be sent after F_MAX_TIME (currently 3 seconds),   which is a negligible total overhead (e.g. one full header every 150   G.723.1 packets), but must be considered carefully in scheduling the   real-time transmissions.  Both simplex and multicast links are not   prevailing in the low-bitrate environment (although multicast   functionality may become more important with wireless systems); in   this document, we therefore assume full-duplex capability.   As compression techniques will improve, a negotiation between the two   peers on the link would provide the best flexibility in   implementation complexity and potential for extensibility.  The peer   routers/hosts can decide which real-time packet streams are to be   compressed, which header fields are not to be sent at all, which   multiplexing information should be used on the link, and how the   remaining header fields should be encoded.  PPP, a well-tried suite   of negotiation protocols, is already used on most of the low-bitrate   links and seems to provide the obvious approach.  Cooperation from   PPP is also needed to negotiate the use of real-time encapsulations   between systems that are not configured to automatically do so.   Therefore, PPP options that can be negotiated at the link setup (LCP)   phase are included in [8], [9], and [10].8.  Security Considerations   Header compression protocols that make use of assumptions about   application protocols need to be carefully analyzed whether it is   possible to subvert other applications by maliciously or   inadvertently enabling their use.   It is generally not possible to do significant hop-by-hop header   compression on encrypted streams.  With certain security policies, it   may be possible to run an encrypted tunnel to a network access server   that does header compression on the decapsulated packets and sends   them over an encrypted link encapsulation; see also the short mention   of interactions between real-time encapsulation and encryption insection 4 above.  If the security requirements permit, a special RTP   payload data format that encrypts only the data may preferably be   used.Bormann                      Informational                     [Page 11]

RFC 2689       Integrated Services over Low-bitrate Links September 19999.  References    [1]  Handley, M., Crowcroft, J., Bormann, C. and J. Ott, "The         Internet Multimedia Conferencing Architecture", Work in         Progress.    [2]  Degermark, M., Nordgren, B. and S. Pink, "IP Header         Compression",RFC 2507, February 1999.    [3]  Scott Petrack, Ed Ellesson, "Framework for C/RTP: Compressed         RTP Using Adaptive Differential Header Compression",         contribution to the mailing list rem-conf@es.net, February         1996.    [4]  Braden, R., Zhang, L., Berson, S., Herzog, S. and S. Jamin,         "Resource ReSerVation Protocol (RSVP) -- Version 1 Functional         Specification",RFC 2205, September 1997.    [5]  Sklower, K., Lloyd, B., McGregor, G., Carr, D. and T.         Coradetti, "The PPP Multilink Protocol (MP)",RFC 1990, August         1996.    [6]  Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,         "RTP: A Transport Protocol for Real-Time Applications",RFC1889, January 1996.    [7]  Casner, S. and V. Jacobson, "Compressing IP/UDP/RTP Headers for         Low-Speed Serial Links",RFC 2508, February 1999.    [8]  Bormann, C., "The Multi-Class Extension to Multi-Link PPP",RFC2686, September 1999.    [9]  Bormann, C., "PPP in a Real-time Oriented HDLC-like Framing",RFC 2687, September 1999.   [10]  Engan, M., Casner, S. and C. Bormann, "IP Header Compression         over PPP",RFC 2509, February 1999.   [11]  Wroclawski, J.,   "Specification of the Controlled-Load Network         Element Service",RFC 2211, September 1997.   [12]  Shenker, S., Partridge, C. and R. Guerin.  "Specification of         Guaranteed Quality of Service",RFC 2212, September 1997.Bormann                      Informational                     [Page 12]

RFC 2689       Integrated Services over Low-bitrate Links September 1999   [13]  ITU-T Recommendation H.223, "Multiplexing protocol for low bit         rate multimedia communication", International Telecommunication         Union, Telecommunication Standardization Sector (ITU-T), March         1996.   [14]  ITU-T Recommendation H.324, "Terminal for low bit rate         multimedia communication", International Telecommunication         Union, Telecommunication Standardization Sector (ITU-T), March         1996.   [15]  ITU-T Recommendation H.245, "Control protocol for multimedia         communication", International Telecommunication Union,         Telecommunication Standardization Sector (ITU-T), March 1996.10.  Author's Address   Carsten Bormann   Universitaet Bremen FB3 TZI   Postfach 330440   D-28334 Bremen, GERMANY   Phone: +49.421.218-7024   Fax:   +49.421.218-7000   EMail: cabo@tzi.orgAcknowledgements   Much of the early discussion that led to this document was done with   Scott Petrack and Cary Fitzgerald.  Steve Casner, Mikael Degermark,   Steve Jackowski, Dave Oran, the other members of the ISSLL subgroup   on low bitrate links (ISSLOW), and in particular the ISSLL WG co-   chairs Eric Crawley and John Wroclawski have helped in making this   architecture a reality.Bormann                      Informational                     [Page 13]

RFC 2689       Integrated Services over Low-bitrate Links September 1999Full 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.Acknowledgement   Funding for the RFC Editor function is currently provided by the   Internet Society.Bormann                      Informational                     [Page 14]