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📄 rfc998.txt

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Network Working Group                                     David D. ClarkRequest for Comments:  998                               Mark L. LambertObsoletes:  RFC 969                                          Lixia Zhang                                                                     MIT                                                              March 1987                 NETBLT: A Bulk Data Transfer Protocol1. Status   This document is a description of, and a specification for, the   NETBLT protocol.  It is a revision of the specification published in   NIC RFC-969.  The protocol has been revised after extensive research   into NETBLT's performance over long-delay, high-bandwidth satellite   channels.  Most of the changes in the protocol specification have to   do with the computation and use of data timers in a multiple   buffering data transfer model.   This document is published for discussion and comment, and does not   constitute a standard.  The proposal may change and certain parts of   the protocol have not yet been specified; implementation of this   document is therefore not advised.2. Introduction   NETBLT (NETwork BLock Transfer) is a transport level protocol   intended for the rapid transfer of a large quantity of data between   computers.  It provides a transfer that is reliable and flow   controlled, and is designed to provide maximum throughput over a wide   variety of networks.  Although NETBLT currently runs on top of the   Internet Protocol (IP), it should be able to operate on top of any   datagram protocol similar in function to IP.   NETBLT's motivation is to achieve higher throughput than other   protocols might offer.  The protocol achieves this goal by trying to   minimize the effect of several network-related problems: network   congestion, delays over satellite links, and packet loss.   Its transmission rate-control algorithms deal well with network   congestion; its multiple-buffering capability allows high throughput   over long-delay satellite channels, and its various   timeout/retransmit algorithms minimize the effect of packet loss   during a transfer.  Most importantly, NETBLT's features give it good   performance over long-delay channels without impairing performance   over high-speed LANs.Clark, Lambert, & Zhang                                         [Page 1]RFC 998                                                       March 1987   The protocol works by opening a connection between two "clients" (the   "sender" and the "receiver"), transferring the data in a series of   large data aggregates called "buffers", and then closing the   connection.  Because the amount of data to be transferred can be very   large, the client is not required to provide at once all the data to   the protocol module.  Instead, the data is provided by the client in   buffers.  The NETBLT layer transfers each buffer as a sequence of   packets; since each buffer is composed of a large number of packets,   the per-buffer interaction between NETBLT and its client is far more   efficient than a per-packet interaction would be.   In its simplest form, a NETBLT transfer works as follows:  the   sending client loads a buffer of data and calls down to the NETBLT   layer to transfer it.  The NETBLT layer breaks the buffer up into   packets and sends these packets across the network in Internet   datagrams.  The receiving NETBLT layer loads these packets into a   matching buffer provided by the receiving client.  When the last   packet in the buffer has arrived, the receiving NETBLT checks to see   that all packets in that buffer have been correctly received.  If   some packets are missing, the receiving NETBLT requests that they be   resent.  When the buffer has been completely transmitted, the   receiving client is notified by its NETBLT layer.  The receiving   client disposes of the buffer and provides a new buffer to receive   more data.  The receiving NETBLT notifies the sender that the new   buffer is ready, and the sender prepares and sends the next buffer in   the same manner.  This continues until all the data has been sent; at   that time the sender notifies the receiver that the transmission has   been completed.  The connection is then closed.   As described above, the NETBLT protocol is "lock-step".  Action halts   after a buffer is transmitted, and begins again after confirmation is   received from the receiver of data.  NETBLT provides for multiple   buffering, a transfer model in which the sending NETBLT can transmit   new buffers while earlier buffers are waiting for confirmation from   the receiving NETBLT.  Multiple buffering makes packet flow   essentially continuous and markedly improves performance.   The remainder of this document describes NETBLT in detail.  The next   sections describe the philosophy behind a number of protocol   features:  packetization, flow control, transfer reliability, and   connection management. The final sections describe NETBLT's packet   formats.3. Buffers and Packets   NETBLT is designed to permit transfer of a very large amounts of data   between two clients.  During connection setup the sending NETBLT can   inform the receiving NETBLT of the transfer size; the maximum   transfer length is 2**32 bytes.  This limit should permit any   practical application.  The transfer size parameter is for the use of   the receiving client; the receiving NETBLT makes no use of it.  AClark, Lambert, & Zhang                                         [Page 2]RFC 998                                                       March 1987   NETBLT receiver accepts data until told by the sender that the   transfer is complete.   The data to be sent must be broken up into buffers by the client.   Each buffer must be the same size, save for the last buffer.  During   connection setup, the sending and receiving NETBLTs negotiate the   buffer size, based on limits provided by the clients.  Buffer sizes   are in bytes only; the client is responsible for placing data in   buffers on byte boundaries.   NETBLT has been designed and should be implemented to work with   buffers of any size.  The only fundamental limitation on buffer size   should be the amount of memory available to the client.  Buffers   should be as large as possible since this minimizes the number of   buffer transmissions and therefore improves performance.   NETBLT is designed to require a minimum amount of memory, allowing   the client to allocate as much memory as possible for buffer storage.   In particular, NETBLT does not keep buffer copies for retransmission   purposes.  Instead, data to be retransmitted is recopied directly   from the client buffer.  This means that the client cannot release   buffer storage piece by piece as the buffer is sent, but this has not   been a problem in preliminary NETBLT implementations.   Buffers are broken down by the NETBLT layer into sequences of DATA   packets.  As with the buffer size, the DATA packet size is negotiated   between the sending and receiving NETBLTs during connection setup.   Unlike buffer size, DATA packet size is visible only to the NETBLT   layer.   All DATA packets save the last packet in a buffer must be the same   size.  Packets should be as large as possible, since NETBLT's   performance is directly related to packet size.  At the same time,   the packets should not be so large as to cause internetwork   fragmentation, since this normally causes performance degradation.   All buffers save the last buffer must be the same size; the last   buffer can be any size required to complete the transfer.  Since the   receiving NETBLT does not know the transfer size in advance, it needs   some way of identifying the last packet in each buffer.  For this   reason, the last packet of every buffer is not a DATA packet but   rather an LDATA packet.  DATA and LDATA packets are identical save   for the packet type.4. Flow Control   NETBLT uses two strategies for flow control, one internal and one at   the client level.   The sending and receiving NETBLTs transmit data in buffers; client   flow control is therefore at a buffer level.  Before a buffer can beClark, Lambert, & Zhang                                         [Page 3]RFC 998                                                       March 1987   transmitted, NETBLT confirms that both clients have set up matching   buffers, that one is ready to send data, and that the other is ready   to receive data.  Either client can therefore control the flow of   data by not providing a new buffer.  Clients cannot stop a buffer   transfer once it is in progress.   Since buffers can be quite large, there has to be another method for   flow control that is used during a buffer transfer.  The NETBLT layer   provides this form of flow control.   There are several flow control problems that could arise while a   buffer is being transmitted.  If the sending NETBLT is transferring   data faster than the receiving NETBLT can process it, the receiver's   ability to buffer unprocessed packets could be overflowed, causing   packet loss.  Similarly, a slow gateway or intermediate network could   cause packets to collect and overflow network packet buffer space.   Packets will then be lost within the network.  This problem is   particularly acute for NETBLT because NETBLT buffers will generally   be quite large, and therefore composed of many packets.   A traditional solution to packet flow control is a window system, in   which the sending end is permitted to send only a certain number of   packets at a time.  Unfortunately, flow control using windows tends   to result in low throughput.  Windows must be kept small in order to   avoid overflowing hosts and gateways, and cannot easily be updated,   since an end-to-end exchange is required for each window change.   To permit high throughput over a variety of networks and gateways,   NETBLT uses a novel flow control method: rate control.  The   transmission rate is negotiated by the sending and receiving NETBLTs   during connection setup and after each buffer transmission.  The   sender uses timers, rather than messages from the receiver, to   maintain the negotiated rate.   In its simplest form, rate control specifies a minimum time period   per packet transmission.  This can cause performance problems for   several reasons.  First, the transmission time for a single packet is   very small, frequently smaller than the granularity of the timing   mechanism.  Also, the overhead required to maintain timing mechanisms   on a per packet basis is relatively high and lowers performance.   The solution is to control the transmission rate of groups of   packets, rather than single packets.  The sender transmits a burst of   packets over a negotiated time interval, then sends another burst.   In this way, the overhead decreases by a factor of the burst size,   and the per-burst transmission time is long enough that timing   mechanisms will work properly.  NETBLT's rate control therefore has   two parts, a burst size and a burst rate, with (burst size)/(burst   rate) equal to the average transmission time per packet.Clark, Lambert, & Zhang                                         [Page 4]RFC 998                                                       March 1987   The burst size and burst rate should be based not only on the packet   transmission and processing speed which each end can handle, but also   on the capacities of any intermediate gateways or networks.   Following are some intuitive values for packet size, buffer size,   burst size, and burst rate.   Packet sizes can be as small as 128 bytes.  Performance with packets   this small is almost always bad, because of the high per-packet   processing overhead.  Even the default Internet Protocol packet size   of 576 bytes is barely big enough for adequate performance.  Most   networks do not support packet sizes much larger than one or two   thousand bytes, and packets of this size can also get fragmented when   traveling over intermediate networks, lowering performance.   The size of a NETBLT buffer is limited only by the amount of memory   available to a client.  Theoretically, buffers of 100 Kbytes or more   are possible.  This would mean the transmission of 50 to 100 packets   per buffer.   The burst size and burst rate are obviously very machine dependent.   There is a certain amount of transmission overhead in the sending and   receiving machines associated with maintaining timers and scheduling   processes.  This overhead can be minimized by sending packets in   large bursts.  There are also limitations imposed on the burst size   by the number of available packet buffers in the operating system   kernel. On most modern operating systems, a burst size of between   five and ten packets should reduce the overhead to an acceptable   level.  A preliminary NETBLT implementation for the IBM PC/AT sends   packets in bursts of five.  It could send more, but is limited by the   available memory.   The burst rate is in part determined by the granularity of the   sender's timing mechanism, and in part by the processing speed of the   receiver and any intermediate gateways.  It is also directly related   to the burst size.  Burst rates from 20 to 45 milliseconds per 5-   packet burst have been tried on the IBM PC/AT and Symbolics 3600   NETBLT implementations with good results within a single local-area   network.  This value clearly depends on the network bandwidth and   packet buffering available.   All NETBLT flow control parameters (packet size, buffer size, burst   size, and burst rate) are negotiated during connection setup.  The   negotiation process is the same for all parameters.  The client   initiating the connection (the active end) proposes and sends a set   of values for each parameter in its connection request.  The other   client (the passive end) compares these values with the highest-   performance values it can support.  The passive end can then modify   any of the parameters, but only by making them more restrictive.  The   modified parameters are then sent back to the active end in its   response message.Clark, Lambert, & Zhang                                         [Page 5]RFC 998                                                       March 1987   The burst size and burst rate can also be re-negotiated after each   buffer transmission to adjust the transfer rate according to the   performance observed from transferring the previous buffer.  The   receiving end sends burst size and burst rate values in its OK   messages (described later).  The sender compares these values with   the values it can support.  Again, it may then modify any of the   parameters, but only by making them more restrictive.  The modified   parameters are then communicated to the receiver in a NULL-ACK   packet, described later.   Obviously each of the parameters depend on many factors -- gateway   and host processing speeds, available memory, timer granularity --   some of which cannot be checked by either client.  Each client must   therefore try to make as best a guess as it can, tuning for

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