GENWiki

Premier IT Outsourcing and Support Services within the UK

User Tools

Site Tools


rfc:rfc1016

Network Working Group W. Prue Request for Comments: 1016 J. Postel

                                                                   ISI
                                                             July 1987
           Something a Host Could Do with Source Quench:
             The Source Quench Introduced Delay (SQuID)

Status of this Memo

 This memo is intended to explore the issue of what a host could do
 with a source quench.  The proposal is for each source host IP module
 to introduce some delay between datagrams sent to the same
 destination host.  This is an "crazy idea paper" and discussion is
 essential.  Distribution of this memo is unlimited.

Introduction

 A gateway may discard Internet datagrams if it does not have the
 buffer space needed to queue the datagrams for output to the next
 network on the route to the destination network.  If a gateway
 discards a datagram, it may send a source quench message to the
 Internet source host of the datagram.  A destination host may also
 send a source quench message if datagrams arrive too fast to be
 processed.  The source quench message is a request to the host to cut
 back the rate at which it is sending traffic to the Internet
 destination.  The gateway may send a source quench message for every
 message that it discards.  On receipt of a source quench message, the
 source host should cut back the rate at which it is sending traffic
 to the specified destination until it no longer receives source
 quench messages from the gateway.  The source host can then gradually
 increase the rate at which it sends traffic to the destination until
 it again receives source quench messages [1,2].
 The gateway or host may send the source quench message when it
 approaches its capacity limit rather than waiting until the capacity
 is exceeded.  This means that the data datagram which triggered the
 source quench message may be delivered.

The SQuID Concept

 Suppose the IP module at the datagram source has a queue of datagrams
 to send, and the IP module has a parameter "D".  D is the introduced
 delay between sending datagrams from the queue to the network.  That
 is, when the IP module discovers a datagram waiting to be sent to the
 network, it sends it to the network then waits time D before even
 looking at the datagram queue again.  Normally, the value of D is

Prue & Postel [Page 1] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 zero.
 Imagine that when a source quench is received (or any other signal is
 received that the host should slow down its transmissions to the
 network), the value of D is increased.  As time goes by, the value of
 D is decreased.

The SQuID Algorithm

        on increase event:
             D <-- maximum (D + K, I)
                                      (where K = .020 second,
                                             I = .075 second)
        on decrease event:
             D <-- maximum (D - J, 0)
                                      (where J = .001 second)
 An increase event is receipt of one or more source quenches in a
 event period E, (where E is 2.000 seconds).
 A decrease event is when S time has passed since D was decreased and
 there is a datagram to send (where S is 1.000 seconds).
 A cache of D's is kept for the last M hosts communicated with.
 Note that when no datagrams are sent to a destination for some time
 the D for that destination is not decreased, but, if a destination is
 not used for a long time that D for that destination may fall out of
 the cache.

Possible Refinements

 Keep a separate outgoing queue of datagrams for each destination
 host, local subnet, or network.
 Keep the cache of D's per network or local subnet, instead of per
 host.
 "I" could be based upon the basic speed of the slowest intervening
 network (see Appendix A).
 "D" could be limited to never go below "I" if the above refinement
 were implemented.
 "S" could be based upon the round trip time.

Prue & Postel [Page 2] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 "D" could be adjusted datagram by datagram based upon the length of
 the datagrams.  Wait longer after a long datagram.
 The delay algorithm could be implemented such that if a source
 doesn't send a datagram when it is next allowed (the introduced delay
 interval) or for N such intervals that the source gets a credit for
 one and only one free (no delay) datagram.

Implementation Ideas

 Since IP does not normally keep much state information about things,
 we want the default or idle IP to have no state about these D values.
 Since the default D value is zero, let us propose that the IP will
 keep a list of only those destinations with non zero D's.
 When the IP wants to send a datagram, it searches the D-list to see
 if the destination is noted.  If it is not, the D value is zero, so
 the IP sends the datagram at once.  If the destination is listed, the
 IP must wait D time indicated before sending that particular
 datagram.  It could look at a datagram addressed to a different
 destination, and possibly send it in the mean time.
 When the IP receives a source quench, it checks to see if the
 destination in the datagram that caused the source quench is on the
 list.  If so, it adds K to the D value.  If not, it appends the
 destination to the list with the D value set to "I".

A Closer Look At the Problem

 Some implementations of IP send one SQ for every N datagrams they
 discard (for example, N=20) so the SQ messages will not make the
 congestion problem much worse [3].  In such situations any of the
 sources of the 20 datagrams may get the SQ not necessarily the one
 causing the most traffic.  However if a host continues to send
 datagrams at a high rate it has a high probability of receiving a SQ
 message sooner or later.  It is much like a speeder on a highway.
 Not all speeders get speeding tickets but the ones who speed most
 often or most excessively are most likely to be ticketed.  In this
 case they will get a ticket and their car may be destroyed.
 With memory becoming so inexpensive many IP nodes put an artificially
 low limit on the size of their queues so that through node delay will
 not be excessive [4].  For example, if one megabyte of data is
 buffered to be sent over a 56 kb/s line the last datagram will wait
 over 2 minutes before being sent.
 One problem with SQ is that the IP or ICMP specification does not
 have a well defined event to indicate receipt of SQ to higher level

Prue & Postel [Page 3] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 protocols.  Therefore many TCP implementations do not get notified
 about SQ events and thus do not react to SQ.  TCP is not the only
 source of IP datagrams either.  Other protocols should also respond
 to SQ events in some appropriate way.  TCP and other protocols at
 that level should do something about a source quench, however,
 discussion of their behavior is beyond the scope of this memo.  Note
 that implementation of SQ processing at one level of protocol should
 not interfere with the behavior of higher level protocols.  This
 however, is difficult to do.
 For protocols using IP which are trying to transfer large amounts of
 data the data flow is most typically very bursty.  TCP for example,
 might send 5-10 segments into a window of 5-10 K bytes then wait for
 the acknowledgment of the data which opens the window again.  NETBLT
 as defined by RFC-998 is a rate based protocol which has parameters
 for burst size and burst rate.
 One purpose of the bursts is to allow the source computer to generate
 several datagrams at once to provide more efficient scheduling.  An
 other reason is to keep the network busy accepting data to maximize
 effective throughput in spite of a potentially large network round
 trip delay.  To send a datagram then wait for an acknowledgment is a
 simple but not efficient protocol on a large wide area network.
 The reasons for efficiencies obtained at the source node by
 generating many datagrams at once are not as applicable in an
 intermediate IP node.  Since each datagram is potentially from a
 different node they must all be treated individually.  Datagrams
 received in a burst may also overload the queue of an intermediate
 node losing datagrams and causing SQs to be generated.  If the queue
 is near a threshold and a burst comes, possibly all of the datagrams
 will be lost.  When datagrams arrive evenly spaced, less datagrams
 are likely to be lost because the inter-arrival time allows the queue
 a little time to empty out.  Therefore datagrams spaced with some
 delay between them may be better for intermediate IP nodes.
 Congestion is most likely to occur at IP nodes which are gateways
 between a slower network and a faster one.  The congestion will be in
 the send queue from the slow network to the fast network.  An SQ
 being returned to the sender will return on the faster network.  (See
 diagram below.)

A Gateway Source Quench Concept

 In order for the SQuID algorithm to work we rely upon the gateways to
 send SQs to us to tell us how we are doing.  Because the loss of a
 single datagram affects data flow so much (see lost datagram
 discussion in Observed Results below) it would be much better for the

Prue & Postel [Page 4] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 source IP node if it got a warning before datagrams were discarded.
 We propose gateway IP nodes start SQing before the node is flooded at
 a level we call SQ Keep (SQK) but forward the datagram.  If the queue
 level reaches a critical level, SQ Toss level (SQT), the gateway
 should toss datagrams to resolve the problem unless the datagram is
 an ICMP message.  Even ICMP messages will be tossed if the MaxQ level
 is reached.  Once the gateway starts sending SQs it should continue
 to do so until the queue level goes below a low water mark level
 (SQLW) as shown below.  This is analogous to methods some operating
 systems use to handle memory space management.
 The gateway should try to send SQ to as many of the contributors of
 the congestion as possible but only once per contributor per second
 or two.
 Source Quench Queue Levels
       +--------------+ MaxQ level
       |              |> datagrams tossed & SQed (but not ICMP msgs.)
       +--------------+ SQT level (95%)
       |              |\
       |              | > datagrams SQed but forwarded
       |              |/
       +--------------+ SQK level (70%)
       |              |\
       |              | \ datagrams SQed but forwarded if SQK level
       |              | / exceeded & SQLW or lower not yet reached
       |              |/
       +--------------+ SQLW level (50%)
       |              |\
       |              | \
       |              |  \
       |              |   \ datagrams forwarded
       |              |   /
       |              |  /
       |              | /
       |              |/
       +--------------+

Description of the Test Model

 We needed some way of testing our algorithm and its various
 parameters.  It was important to check the interaction between IP
 with the SQuID algorithm and TCP.  We also wanted to try various
 combinations of retransmission strategy and source quench strategy
 which required control of the entire test network.  We therefore
 decided to build an Internet model.

Prue & Postel [Page 5] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 Using this example configuration for illustration:

_ LAN _ WAN _ LAN _

1 2 3 4
TCP/IP —10 Mb/s– IP —56 kb/s– IP —10 Mb/s–TCP/IP
_| |_ _| |_
 A program was written in C which created queues and structures to put
 on the queues representing datagrams carrying data, acknowledgments
 and SQs.  The program moved datagrams from one queue to the next
 based upon rules defined below
 A client fed the TCP in node 1 data at the rate it would accept.  The
 TCP function in node 1 would chop the data up into fixed 512 byte
 datagrams for transmission to the IP in node 1.  When the datagrams
 were given to IP for transmission, a timestamp was put on it and a
 copy of it was put on a TCP ack-wait queue (data sent but not yet
 acknowledged).  In particular TCP assumed that once it handed data to
 IP, the data was sent immediately for purposes of retransmission
 timeouts even though our algorithm has IP add delay before
 transmission.
 Each IP node had one queue in each direction (left and right).  For
 each IP in the model IP would forward datagrams at the rate of the
 communications line going to the next node.  Thus the fifth datagram
 on IP 2's queue going right would take 5 X 73 msec or 365 msec before
 it would appear at the end of IP 3's queue.  The time to process each
 datagram was considered to be less than the time it took for the data
 to be sent over the 56 kb/s lines and therefore done during those
 transmission times and not included in the model.  For the LAN
 communications this is not the case but since they were not at the
 bottleneck of the path this processing time was ignored.  However
 because LAN communications are typically shared band width, the LAN
 band width available to each IP instance was considered to be 1 Mb/s,
 a crude approximation.
 When the data arrived at node 4 the data was immediately given to the
 TCP receive function which validated the sequence number.  If the
 datagram was in sequence the datagram was turned into an ack datagram
 and sent back to the source.  An ack datagram carries no data and
 will move the right edge of the window, the window size past the just
 acked data sequence number.  The ack datagram is assumed to be 1/8 of
 the length of a data datagram and thus can be transmitted from one
 node to the next 8 times faster.  If the sequence number is less than
 expected (a retransmission due to a missed ack) then it too is turned
 into an ack.  A larger sequence number datagram is queued
 indefinitely until the missing datagrams are received.

Prue & Postel [Page 6] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 We also modeled the gateway source quench algorithm.  When a datagram
 was put on an IP queue the number on the queue was compared to an SQ
 keep level (SQK).  If it was greater, an SQ was generated and
 returned to the sender. If it was larger than the SQ toss (SQT) level
 it was also discarded.  Once SQs were generated they would continue
 to be sent until the queue level went below SQ Low Water (SQLW) level
 which was below the original SQK level.  These percentages were
 modifiable as were many parameters.  An SQ could be lost if it
 exceeded the maximum queue size (MaxQ), but a source quench was never
 sent about tossing a source quench.
 Upon each transition from one node to the next, the datagram was
 vulnerable to datagram loss due to errors.  The loss rate could be
 set as M losses out of N datagrams sent, thus the model allowed for
 multi-datagram loss bursts or single datagram losses.  We used a
 single datagram loss rate of 1 lost datagram per 300 datagrams sent
 for much of our testing.  While this may seem low for Internet
 simulation, remember it does not include losses due to congestion.
 Some network parameters we used were a maximum queue length of 15
 datagrams per IP direction left and right.  We started sending SQ if
 the queue was 70% full, SQK level, tossed data datagrams, but not SQ
 datagrams, if 95% of the queue was reached, SQT level, and stopped
 SQing when a 50% SQLW level was reached (see above).
 We ignored additional SQs for 2 seconds after receipt of one SQ.
 This was done because some Internet nodes only send one SQ for every
 20 datagrams they discard even though our model sent SQs for every
 datagram discarded.  Other IP node may send one SQ per discarded
 packet. The SQuID algorithm needed a way to handle both types of SQ
 generation.  We therefore treated one or a burst of SQs as a single
 event and incremented our D by a larger amount than would be
 appropriate for responding individually to the multiple SQs of the
 verbose nodes.
 The simulation did not do any fragmenting of datagrams.  Silly window
 syndrome was avoided.  The model did not implement nor simulate the
 TTL (time-to-live) function.
 The model allowed for a flexible topology definition with many TCP
 source/destination pairs on host IP nodes or gateway IP nodes with
 various windows allowed.  An IP node could have any number of TCPs
 assigned to it.  Each line could have an individually set speed.  Any
 TCP could send to any other TCP.  The routing from one location to
 another was fixed.  Therefore datagrams did not arrive out of
 sequence.  However, datagrams arrived in ascending order, but not
 consecutively, on a regular basis because of datagram losses.
 Datagrams going "left" through a node did not affect the queue size,

Prue & Postel [Page 7] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 or SQ chances, of data going "right" through the node.
 The TCP retransmission timer algorithm used an Alpha of .15 and a
 Beta of 1.5.  The test was run without the benefit of the more
 sophisticated retransmission timer algorithm proposed by Van Jacobson
 [5].
 The program would display either the queue sizes of the various IP
 nodes and the TCP under test as time passed or do a crude plot of
 various parameters of interest including SRTT, perceived round trip
 time, throughput, and the critical queue size.
 As we observed the effects of various algorithms for responding to SQ
 we adapted our model to better react to SQ.  Initial tests showed if
 we incremented slowly and decremented quickly we observed
 oscillations around the correct value but more of the time was spent
 over driving the network, thus losing datagrams, than at a value
 which helped the congestion situation.
 A significant problem is the delay between when some intermediate
 node starts dropping datagrams and sending source quenches to the
 time when the source quenches arrive at the source host and can begin
 to effect the behavior at the data source.  Because of this and the
 possibility that a IP might send only one SQ for each 20 datagrams
 lost, we decided that the increase in D per source quench should be
 substantial (for example, D should increase by 20 msec for every
 source quench), and the decrease with time should be very slow (for
 example, D should decrease 1 msec every second).  Note that this is
 the opposite behavior than suggested in an early draft by one of the
 authors.
 However, when many source quenches are received (for example, when a
 source quench is received for every datagram dropped) in a short time
 period the D value is increased excessively.  To prevent D from
 growing too large, we decided to ignore subsequent source quenches
 for a time (for example, 2 seconds) once we had increased D.
 Tests were run with only one TCP sending data to learn as much as
 possible how an unperturbed session might run.  Other test runs would
 introduce and eliminate competing traffic dynamically between other
 TCP instances on the various nodes to see how the algorithms reacted
 to changes in network load.  A potential flaw in the model is that
 the defined TCPs with open windows always tried to forward data.
 Their clients feeding them data never paused to think what they were
 going to type nor got swapped out in favor of other applications nor
 turned the session around logically to listen to the other end for
 more user commands.  In other words all of the simulated TCP sessions
 were doing file transfers.

Prue & Postel [Page 8] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 The model was defined to allow many mixes of competing algorithms for
 responding to SQ.  It allowed comparing effective throughput between
 TCPs with small windows and large windows and those whose IP would
 introduce inter-datagram delays and those who totally ignored SQ.  It
 also allowed comparisons with various inter-datagram increment
 amounts and decrement amounts.  Because of the number of possible
 configurations and parameter combinations only a few combinations of
 parameters were tested. It is hoped they were the most appropriate
 ones upon which to concentrate.

Observed Results

 All of our algorithms oscillate, some worse than others.
 If we put in just the right amount of introduced delay we seem to get
 the best throughput.  But finding the right amount is not easy.
 Throughput is adversely affected, heavily, by a single lost datagram
 at least for the short time.  Examine what happens when a window is
 35 datagrams wide with an average round trip delay of 2500 msec using
 512 byte datagrams when a single datagram is lost at the beginning.
 Thirty five datagrams are given by TCP to IP and a timer is started
 on the first datagram.  Since the first datagram is missing, the
 receiving TCP will not sent an acknowledgment but will buffer all 34
 of the out-of-sequence datagrams.  After 1.5 X 2500 msec, or 3750
 msec, have elapsed the datagram times out and is resent.  It arrives
 and is acked, along with the other 34, 2500 msec later.  Before the
 lost datagram we might have been sending at the average rate a 56
 kb/s line could accept, about one every 75 msec.  After loss of the
 datagram we send at the rate of one in 6250 msec over 83 times
 slower.
 If the lost datagram in the above example is other than the first
 datagram the situation becomes the same when all of the datagrams
 before the lost datagram are acknowledged.  The example holds true
 then for any single lost datagram in the window.
 When SQ doesn't always cause datagram loss the sender continues to
 send too fast (queue size oscillates a lot).  It is important for the
 SQ to cause feed-back into the sending system as soon as possible,
 therefore when the source host IP receives an SQ it must make
 adjustments to the send rate for the datagrams still on the send
 queue not just datagrams IP is requested to send after the SQ.
 Through network delay goes up as the network queue lengths go up.
 Window size affect the chance of getting SQed.  Look at our model
 above using a queue level of 15 for node 2 before SQs are generated

Prue & Postel [Page 9] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 and a window size of 20 datagrams.  We assumed that we could send
 data over the LAN at a sustained average rate of 1 Mb/s or about 18
 times as fast as over the WAN.  When TCP sends a burst of 20
 datagrams to node 1 they make it to node 2 in 81 msec.  The
 transition time from node 2 to node 3 is 73 msec, therefore, in 81
 msec, only one datagram is forwarded to node 3.  Thus the 17th, 18th,
 19th, and 20th datagram are lost every time we send a whole window.
 More are lost when the queue is not empty.  If a sequence of acks
 come back in response to the sent data, the acks tend to return at
 the rate at which data can traverse the net thus pacing new send data
 by opening the window at the rate which the network can accept it.
 However as soon as one datagram is lost all of the subsequent acks
 are deferred and batched until receipt of the missing data block
 which acks all of the datagrams and opens the window to 20 again.
 This causes the max queue size to be exceeded again.
 If we assume a window smaller than the max queue size in the
 bottleneck node, any time we send a window's worth of data, it is
 done independently of the current size of the queue.  The larger the
 send window, the larger a percentage of the stressed queue we send.
 If we send 50% of the stressed queue size any time that queue is more
 than 50% we threaten to overflow the queue.  Evenly spaced single
 datagram bursts have the least chance of overflowing the queue since
 they represent the minimum percentage of the max queue one may send.
 When a big window opens up (that is, a missing datagram at the head
 of a 40 datagram send queue gets retransmitted and acked), the
 perceived round trip time for datagrams subsequently sent hits a
 minimum value then goes up linearly.  The SRTT goes down then back up
 in a nice smooth curve.  This is caused by the fact IP will not add
 delay if the queue is empty and IP has not sent any datagrams to the
 destination for our introduced delay time.  But as many datagrams are
 added to the IP pre-staged send queue in bursts all have the same
 send time as far as TCP is concerned.  IP will delay each datagram on
 the head of the queue by the introduced delay amount.  The first may
 be undelayed as just described but all of the others are delayed by
 their ordinal number on the queue times the introduced delay amount.
 It seems as though in a race between a TCP session which delays
 sending to IP and one who does not, the delayer will get better
 throughput because less datagrams are lost.  The send window may also
 be increased to keep the pipeline full.  If however the non delayer
 uses windowing to reduce the chance of SQ datagram loss his
 throughput may possibly be better because no fair queuing algorithm
 is in place.
 If gateways send SQs early and don't toss data until its critical and
 keep sending SQs until a low water mark is hit, effective throughput

Prue & Postel [Page 10] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 seems to go up.
 At the startup of our tests throughput was very high, then dropped
 off quickly as the last of the window got clobbered.  Our model
 should have used a slow start up algorithm to minimize the startup
 shock.  However the learning curve to estimate the proper value for D
 was probably quicker.
 A large part of the perceived RTT is due to the delay getting off the
 TCP2IP (TCP transitional) queue when we used large windows.  If IP
 would invoke some back-pressure to TCP in a real implementation this
 can be significantly reduced.  Reducing the window would do this for
 us at the expense of throughput.
 After an SQ burst which tosses datagrams the sender gets in a mode
 where TCP may only send one or two datagrams per RTT until the queued
 but not acked segments fall into sequence and are acked.  This
 assumes only the head of the retransmission queue is retransmitted on
 a timeout.  We can send one datagram upon timeout.  When the ack for
 the retransmission is received the window opens allowing sending a
 second.  We then wait for the next lost datagram to time out.
 If we stop sending data for a while but allow D to be decreased, our
 algorithm causes the introduced delay to dwindle away.  We would thus
 go through a new startup learning curve and network oscillation
 sequence.
 One thing not observed often was TCP timing out a segment before the
 source IP even sent the datagram the first time.  As discussed above
 the first datagram on the queue of a large burst is delayed minimally
 and succeeding datagrams have linearly increasing delays.  The
 smoothed round trip delay algorithm has a chance to adapt to the
 perceived increasing round trip times.

Unstructured Thoughts and Comments

 The further down a route a datagram traverses before being clobbered
 the greater the waste of network resources.  SQs which do not destroy
 the datagram referred to are better than ones that do if return path
 resources are available.
 Any fix must be implementable piecemeal.  A fix can not be installed
 in all or most nodes at one time.  The SQuID algorithm fulfills this
 requirement.  It could be implemented, installed in one location, and
 used effectively.
 If it can be shown that by using the new algorithm effective
 throughput can be increased over implementations which do not

Prue & Postel [Page 11] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 implement it that may well be effective impetus to get vendors to
 implement it.
 Once a source host has an established average minimum inter-datagram
 delay to a destination (see Appendix A), this information should be
 stored across system restarts.  This value might be used each time
 data is sent to the given host as a minimum inter-datagram delay
 value.
 Window closing algorithms reduce the average inter-datagram delay and
 the burst size but do not affect the minimum inter-datagram spacing
 by TCP.
 Currently an IP gateway node can know if it is in a critical path
 because its queues stay high or keep building up.  Its optimum queue
 size is one because it always has something to do and the through
 node delay is at a minimum.  It is very important that the gateway at
 the critical path not so discourage data flow that its queue size
 drops to zero.  If the gateway tosses datagrams this stops data flow
 for TCP for a while (as described in Observed Results above).  This
 argues for the gateway algorithm described above which SQs but does
 not toss datagrams unless necessary.  Optimally we should try to have
 a queue size somewhat larger than 1 but less than say 50% of the max
 queue size.  Large queues lead to large delay.
 TCP's SRTT is made artificially large by introducing delay at IP but
 the perceived round trip time variance is probably smaller allowing a
 smaller Beta value for the timeout value.
 So that a decrease timer is not needed for the "D" decrease function,
 upon the next sent datagram to a delayed destination just decrease
 the delay by the amount of time since we last did this divided by the
 decrease timer interval.  An alternate algorithm would be to decrease
 it by only one decrease unit amount if more than the timer interval
 has gone by.  This eliminates the problem caused by the delay, "D",
 dwindling away if we stop sending for a while.  The longer we send
 using this "D", the more likely it is that it is too large a delay
 and the more we should decrease it.
 It is better for the network and the sender for our introduced delay
 to be a little on the high side.  It minimizes the chances of getting
 a datagram clobbered by sending it into a congested gateway.  A lost
 datagram scenario described above showed that one lost datagram can
 reduce our effective delay by one to two orders of magnitude
 temporarily.  Also if the delay is a little high, the net is less
 stressed and the queues get smaller, reducing through network delay.
 The RTT experienced at a given time verses the minimum RTT possible

Prue & Postel [Page 12] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 for the given route does give a good measure of congestion.  If we
 ever get congestion control working RTT may have little to do with
 the amount of congestion.  Effective throughput when compared with
 the possible throughput (or some other measure) is the only real
 measure of congestion.
 Slow startup of TCP is a good thing and should be encouraged as an
 additional mechanism for alleviating network overload.
 The network dynamics tends to bunch datagrams.  If we properly space
 data instead of bunching it like windowing techniques to control
 overflow of queues then greater throughput is accomplished because
 the absolute rate we can send is pacing our sending not the RTT.  We
 eliminate "stochastic bunching" [6].
 The longer the RTT the more network resources the data takes to
 traverse the net.
 Should "fair queuing" say that a longer route data transfer should
 get less band width than a shorter one (since it consumes more of the
 net)?  Being fair locally on each node may be unfair overall to
 datagrams traversing many nodes.
 If we solve congestion problems today, we will start loading up the
 net with more data tomorrow.  When this causes congestion in a year
 will that type of congestion be harder to solve than todays or is it
 not our problem?  John Nagle suggests "In a large net, we may well
 try to force congestion out to the fringes and keep the interior of
 the net uncongested by controlling entry to the net.  The IMP-based
 systems work that way, or at least used to.  This has the effect of
 concentrating congestion at the entrance to the long-haul system.
 That's where we want it; the Source Quench / congestion window / fair
 queuing set of strategies are able to handle congestion at the LAN to
 WAN bottleneck [7].  Our algorithm should try to push the network
 congestion out to the extremities and keep the interior network
 congestion free.
 Use of the algorithm is aesthetically appealing because the data is
 sitting in our local queue instead of consuming resources inside the
 net.  We give data to the network only when it is ready to accept it.
 An averaged minimum inter-datagram arrival value will give a measure
 of the network bottleneck speed at the receiver.  If the receiver
 does not defer or batch together acks the same would be learned from
 the inter-datagram arrival time of the acks.  A problem is that IP
 doesn't have knowledge of the datagram contents.  However IP does
 know from which host a datagram comes.

Prue & Postel [Page 13] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 If SQuID limits the size of its pre-net buffering properly (causes
 back-pressure to TCP) then artificially high RTT measurements would
 not occur.
 TCP might, in the future, get a way to query IP for the current
 introduced delay, D, for a given destination and if the value is too
 excessive abort or not start a session.
 With the new algorithm TCP could have an arbitrarily large window to
 send into without fear of over running queue sizes in intermediate
 nodes (not that any TCP ever considered having this fear before).
 Thus it could have a window size which would allow it to always be
 sending; keeping the pipe full and seldom getting in the stop-and-
 wait mode of sending.  This presupposes that the local IP is able to
 cause some sort of back pressure so that the local IPs queues are not
 over run.  TCP would still be operating in the burst mode of sending
 but the local IP would be sending a datagram for the TCP as often as
 the network could accept it keeping the data flow continuous though
 potentially slow.
 Experience implementing protocols suggests avoiding timers in
 protocols whenever possible.  IP, as currently defined, does not use
 timers. The SQuID algorithm uses two at the IP level.  A way to
 eliminate the introduced delay decrease timer is to decrease the D
 value when we check the send queue for data to send.  We would
 decrease "D" by one "J" unit if "S" time, or more, has elapsed.  The
 other timer is not so easily eliminated.  If the IP implementation is
 periodically awakened to check for work to do, it could check the
 time stamps of the head of the queues to see if any datagrams have
 waited long enough.  This would mean we would necessarily wait too
 long before sending, but it may not be too large of a variance from
 our desired rates.  The additional delay would help congestion and
 reduce our chances of SQ.  Another option is setting a real timer
 which is say 25-50% too large and hope that our periodic work in IP
 will allow us to notice a datagram is delayed enough, and send it.
 Upon sending the datagram we would cancel the timer, avoiding the
 timer interrupt and environment swap.  In other implementations the
 communications interface or the link level protocol may be able to
 provide the timing needed without interrupts to the main processor.
 Background tasks like some file transfers could query IP for the
 latest delay characteristics for a given destination to determine if
 it is appropriate to consume network resources now or if it would be
 better to wait until later.
 We should consider what would happen if IP, using the SQuID
 algorithm, and TCP both introduced delay between the datagrams.  If
 TCPs delay was greater than IP's, then when IP got the datagrams it

Prue & Postel [Page 14] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 would forward them immediately as described in the algorithm above.
 This is because when the IP send queue is empty and it has been at
 least as long as IP wants the higher level protocol, TCP, gets one
 free (no delay) send.  Note also that IP will be decreasing the
 amount of delay it wants introduced because of the successful
 transmissions without SQs.  This would affect other protocols who
 might also send to the same destination.  If TCP sends too quickly
 then IP will protect the network from its indiscretion as described
 in the basic algorithm however TCPs round trip time estimates will be
 much closer to reality.  A lost datagram will thus be detected more
 quickly.  If TCP also uses windowing to limit its sending rate, it
 might, because of its success with a smaller window, increase the
 window size increasing TCPs efficiency.
 As this algorithm is implemented everywhere, the SQ Keep (SQK) and SQ
 Low Water (SQLW) queue level percentages should be dropped to reduce
 queue sizes and thus the through net delay.  The percentage we lower
 SQK and SQLW to should be adjusted based upon the percentage of time
 the queue is empty.  The more the queue is empty the more likely it
 is that the node is discouraging data flow too much.  The more time
 the queue is not empty but not too full, the more likely it is the
 node is not excessively discouraging data flow.  How uniform the
 queue size is, is a measure of how well the network citizens are
 behaved.
 As the congestion is pushed to the sources, gateways which are
 bottlenecks can more easily detect someone not playing by the rules
 (sending datagrams in bursts) and deal with the offender.

Prue & Postel [Page 15] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

Appendix A – Determination of the Value for the Parameter "I"

 To get to the correct value for the delay needed quickly, when an
 event occurred and the currently used delay was minimal, the
 transmission time for an average sized datagram across the slowest
 communications link was used for a first value.  How a real IP node
 is to guess this value is discussed below.  In our example the
 transition between node 2 and node 3 is the bottleneck. Using the 56
 kb/s line, a 512 byte datagram would take 73 msec with no queuing or
 processing time is considered.  This value is defined to be the
 minimum inter-datagram arrival time (MIAT).  Assuming a perfect
 network, ignoring factors other than transmission speed, this is the
 minimum time one could expect between receipt of datagrams at the
 destination, because of the slowed data rate through the bottleneck.
 This won't hold true if the datagrams do not follow the same path.
 The MIAT, minimum inter-datagram arrival time, may be guessed at by
 comparing the arrival timestamps of consecutive datagrams from a
 source of data.  Each value to be considered needs to be adjusted up
 or down based upon the size between the second datagram received and
 the typical datagram size.  More simply stated, a datagram which is
 half the size of the average datagram can be transmitted across a
 line in half the time.  Therefore, double its IAT before considering
 it for an MIAT.  If the timestamp of a datagrams is taken based upon
 an event caused by the start of the datagram arriving, not the
 completion of the datagram arriving, then the first datagram's size
 is the limiting length and should be used to adjust its IAT.  In
 order to keep the algorithm simple, arrival times for short datagram
 could be ignored as could IATs which were orders of magnitude too
 large (see below).
 Once a minimal value is found based upon looking for small values
 over a minute or more, the value might be time averaged with a value
 kept like TCP's SRTT in order to reduce the effects of a false MIAT.
 We could assume the limiting facility would be a 9.6 kb/s line, a
 56-64 kb/s line, or a 1.5 Mb/s line.  These facilities would provide
 a MIAT of 427 msec, 73-64 msec, or 3 msec respectively, for a
 datagram 512 bytes long.  These are almost orders of magnitude in
 differences.  If the MIAT a node measures is not in this range but
 close, the value it is closest to may be used for its MIAT from that
 source.
 One of the good things about this measurement is that it is an
 entirely passive measurement.  No additional traffic is needed to
 measure it.  If a source is not sending data continuously then the
 longer measured values won't be validated as minimal values.  The
 assumption is that at least sometimes the source will send multiple
 datagrams at a time.

Prue & Postel [Page 16] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

 The MIAT measurement is totally unaffected by satellite delay
 characteristics of any intervening facilities.  The chance of getting
 a valid minimal reading is affected by the number of nodes traversed,
 but the value measured if it is valid will not be affected by the
 number of nodes traversed.  Stated another way, when a pair of
 datagrams traverse from one node to the next the datagrams are
 susceptible to being separated by a datagram from another source.
 Both of these factors affect SRTT. The value obtained requires no
 topological knowledge of the route.
 A potential problem with the measurement is, it will not be the
 proper value for some forms of alternate routes.  If a T1 link is the
 bottleneck route some times and other times it is a 56 kb/s link our
 first guess for inter-datagram delay would be too small for the 56
 kb/s line route.  Another problem is that if one datagram goes via
 one route and the next goes via another, their inter-datagram arrival
 difference could lead to a small false measurement.  If intervening
 networks fragment datagrams then the measured IAT between segments
 could be misleading.  A solution to this problem is to ignore
 fragmented datagram IATs.
 This number represents the minimum inter-datagram delay the sending
 IP should use to send to us, the measuring site, for the given
 datagram size.  If we assume that the return path will be through the
 same facilities or the same type, then as described above this value
 can be the first guess for inter-datagram introduced delay, "D" (in
 the algorithm).  It represents the "I" parameter.
 These MIATs may be cached on a host, subnet, or network basis.  The
 last "n" hosts MIATs could be kept.  For infrequent destinations an
 entry per destination network would be applicable to many
 destinations.  If the local net is in fact a subnet, then the other
 local subnet MIATs could be kept.
 If a good algorithm is found for MIAT, comparing it to the average
 IAT (during data transfer) would give a good measure of the amount of
 network traffic load.  Since IP has no idea when the source of data
 is sending as fast as possible, to get a valid average, upper layer
 protocols would have to figure this out for themselves.  IP could
 however provide an interface to get the current MIAT for a given
 destination.

Prue & Postel [Page 17] RFC 1016 Source Quench Introduced Delay – SQuID July 1987

References

 [1]  Postel, Jon, "Internet Protocol - DARPA Internet Program
 Protocol Specification", RFC 791, ISI, September 1981.
 [2]  Postel, Jon, "Internet Control Message Protocol - DARPA Internet
 Program Protocol Specification", RFC 792, ISI, September 1981.
 [3]  Karels, M., "Re: Source Quench", electronic mail message to J.
 Postel and INENG-INTEREST, Tue, 24 Feb 87.
 [4] Nagle, John B., "On Packet Switches With Infinite Storage", RFC
 970, FACC Palo Alto, December 1985.
 [5] Jacobson, Van, "Re: interpacket arrival variance and mean",
 electronic mail message to TCP-IP,  Mon, 15 Jun 87 06:08:01 PDT
 [6] Jacobson, Van, "Re: Appropriate measures of gateway performance"
 electronic mail message to J. Noel Chiappa  and INENG-INTEREST, Sun,
 22 Mar 87 15:04:44 PST.
 [7] Nagle, John B., "Source quench, and congestion generally",
 electronic mail message to B. Braden and INENG-INTEREST, Thu, 5 Mar
 87 11:08:49 PST.
 [8] Nagle, John B., "Congestion Control in IP/TCP Internetworks", RFC
 896, FACC Palo Alto, 6 January 1984.

Prue & Postel [Page 18]

/data/webs/external/dokuwiki/data/pages/rfc/rfc1016.txt · Last modified: 1987/07/18 19:49 by 127.0.0.1

Donate Powered by PHP Valid HTML5 Valid CSS Driven by DokuWiki