Internet Info 1997 December/Internet_Info_CD-ROM_Walnut_Creek_December_1997.iso/drafts/draft_ietf_i/draft-ietf-issll-isslow-01.txt

Text File | 1997-03-26 | 30KB | 639 lines

INTERNET-DRAFT                                           Carsten Bormann
Expires: September 1997                              Universitaet Bremen
                                                              March 1997


          Providing integrated services over low-bitrate links
                     draft-ietf-issll-isslow-01.txt


Status of this memo

   This document is an Internet-Draft.  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
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   To learn the current status of any Internet-Draft, please check the
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   ftp.isi.edu (US West Coast).

   Distribution of this document is unlimited.

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.

   This document is a product of the IETF ISSLL working group.
   Comments are solicited and should be addressed to the working group's
   mailing list at [e-mail] and/or the author.







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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 (in the intserv working group of the IETF)

   -    a setup protocol that allows establishing special forwarding
        treatment for real-time information flows (RSVP [4], in the rsvp
        working group of the IETF)

   -    a transport protocol for real-time information (RTP/RTCP [6], in
        the avt working group of the IETF).

   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 and 28.8 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.  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 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 (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:


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   -    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.


2.  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 any single one of these techniques is not sufficient to
   solve the problem.

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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 in 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.

   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).  Preferably, one also could make use of the
   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 be the
   controlled-load service.  Controlled-load does not provide control
   parameters for target delay; this makes it very hard to 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 are not possible 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 internet
   telephony vendors 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[1].

   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
_________________________
  [1] or actually by using a non-standard, efficiently coded head-
er in the first place.


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   hold for end-to-end compression; therefore additional overhead is
   required for resequencing state changes and 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,
   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., of 28 of the 44 bytes mentioned above).

   One of the design principles of the header compression contemplated
   for IPv6 is that it stops at layers the semantics of which cannot be
   inferred from information in lower layer (outer) headers.  Therefore,
   the original IPv6 header compression 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].

   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.  An additional goal is to minimize the overhead
   required for the transmission of packets from periodic flows.  This

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   strongly argues for being able to suspend a packet, i.e. segment it
   into parts between which the real-time packets can be transferred.

   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.

   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, the multiplex used by the H.324
   videophone standard.  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 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

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   was unachievable.

   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.


4.1.  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.

4.1.1.  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.



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4.1.2.  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).


4.2.  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
   IPv6 header compression draft [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 IPv6 header compression, the environment
   discussed here assumes the existence of a full-duplex PPP link and
   thus can rely on negotiation where IPv6 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.)

   Additional design effort was required for RTP header compression.
   Applying the concepts of IPv6 header compression, of the (at least)
   12 bytes in an RTP header, 7 bytes (timestamp, sequence, and marker

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   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 as
   RFC 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).

   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
   teared down quickly when a source ceases to transmit.


7.  Elements of Hop-By-Hop Negotiation Protocols


   The IPv6 header compression draft 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

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   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], and [9] (and need to be put into [7], FIX
   ME).


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 in
   section 4 above.  If the security requirements permit, a special RTP
   payload data format that encrypts only the data may preferably be
   used.

9.  Author's Address


   Carsten Bormann
   Universitaet Bremen FB3 TZI
   Postfach 330440
   D-28334 Bremen, GERMANY
   [e-mail]
   phone +49.421.218-7024
   fax +49.421.218-7000


Acknowledgements

   Much of the early discussion that led to this document was done with
   Scott Petrack of IBM Israel and Cary Fitzgerald of Cisco Systems.

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   Steve Casner, Mikael Degermark, Steve Jackowski, Dave Oran, and the
   other members of the ISSLL subgroup on low bitrate links (ISSLOW)
   have helped making this architecture a reality.

10.  References


   [1]  Mark Handley, Jon Crowcroft, Carsten Bormann, ``The Internet
        Multimedia Conferencing Architecture,'' Internet-Draft draft-
        ietf-mmusic-confarch-00.txt, Work in Progress, February 1996.

   [2]  M. Degermark, B. Nordgren, S. Pink, ``Header Compression for
        IPv6,'' Internet-Draft draft-degermark-ipv6-hc-02.txt, Work in
        Progress, November 1996.

   [3]  Scott Petrack, Ed Ellesson, ``Framework for C/RTP: Compressed
        RTP Using Adaptive Differential Header Compression'',
        contribution to the mailing list [e-mail], February 1996.

   [4]  R. Braden, Ed., L. Zhang, S. Berson, S. Herzog, S. Jamin,
        Resource ReSerVation Protocol (RSVP) -- Version 1 Functional
        Specification, Internet-Draft draft-ietf-rsvp-spec-14.txt, Work
        in Progress, November 1996.

   [5]  K. Sklower, B. Lloyd, G. McGregor, D. Carr, T. Coradetti, ``The
        PPP Multilink Protocol (MP)'', RFC 1990, August 1996 (obsoletes
        RFC1717).

   [6]  H. Schulzrinne, S. Casner, R. Frederick & V. Jacobson, ``RTP: A
        Transport Protocol for Real-Time Applications'', RFC 1889,
        January 1996.

   [7]  S. Casner, V. Jacobson, ``Compressing IP/UDP/RTP Headers for
        Low-Speed Serial Links'', Internet-Draft draft-ietf-avt-
        crtp-01.txt, Work in Progress, November 1996.

   [8]  C. Bormann, ``The Multi-Class Extension to Multi-Link PPP'',
        internet Draft draft-ietf-issll-isslow-mcml-01.txt, Work in
        Progress, March 1997.

   [9]  C. Bormann, ``PPP in a real-time oriented HDLC-like framing,
        internet Draft draft-ietf-issll-isslow-rtf-00.txt, Work in
        Progress, March 1997.











Bormann                                                        [Page 11]