rfc1046.txt
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described above. This would still be done in a way to not exceed the
allowed service rate of the available bandwidth.
These service rates are suggestions. Some simplifications can be
considered, such as having only two routing classes; low delay, and
other.
Priority
There is the ability to select 8 levels of priority 0-7, in addition
to the class of service selected. To provide this without blocking
the least priority requests, we must give preempted datagrams
frustration points every time a higher priority request cuts in line
in front of it. Thus if a datagram with low priority waits, it will
always get through even when competing against the highest priority
requests. This assumes the TTL (Time-to-Live) field does not expire.
When a datagram with priority arrives at a node, the node will queue
the datagram on the appropriate queue ahead of all datagrams with
lower priority. Each datagram that was preempted gets its priority
raised (locally). The priority data will not bump a lower priority
datagram off its queue, discarding the data. If the queue is full,
the newest data (priority or not) will be discarded. The priority
preemption will preempt only within the class of service queue to
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which the priority data is targeted. A request specifying regular
class of service, will contend on the queue where it is placed, high
throughput or high reliability.
An implementation strategy is to multiply the requested priority by 2
or 4, then store the value in a buffer overhead area. Each time the
datagram is preempted, increment the value by one. Looking at an
example, assume we use a multiplier of 2. A priority 6 buffer will
have an initial local value of 12. A new priority 7 datagram would
have a local value of 14. If 2 priority 7 datagrams arrive,
preempting the priority 6 datagram, its local value is incremented to
14. It can no longer be preempted. After that, it has the same
local value as a priority 7 datagram and will no longer be preempted
within this node. In our example, this means that a priority 0
datagram can be preempted by no more than 14 higher priority
datagrams. The priority is raised only locally in the node. The
datagram could again be preempted in the next node on the route.
Priority queuing changes the effects we were obtaining with the low
delay queuing described above. Once a buffer was queued, the delay
that a datagram would see could be determined. When we accepted low
delay data, we could guarantee a certain maximum delay. With this
addition, if the datagram requesting low delay does not also request
high priority, the guaranteed delay can vary a lot more. It could be
1 up to 28 times as much as without priority queuing.
Discussion and Details
If a low delay queue is for a satellite link (or any high delay
link), the max queue size should be reduced by the number of
datagrams that can be forwarded from the queue during the one way
delay for the link. That is, if the service rate for the low delay
queue is L datagrams per second, the delay added by the high delay
link is D seconds and M is the max delay per node allowed (MGD) in
seconds, then the maximum queue size should be:
Max Queue Size = L ( M - D), M > D
= 0 , M <= D
If the result is negative (M is less than the delay introduced by the
link), then the maximum queue size should be zero because the link
could never provide a delay less than the guaranteed M value. If the
bandwidth is high (as in T1 links), the delay introduced by a
terrestrial link and the terminating equipment could be significant
and greater than the average service time for a single datagram on
the low delay queue. If so, this formula should be used to reduce
the queue size as well. Note that this is reducing the queue size
and is not the same as the allocated bandwidth. Even though the
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queue size is reduced, the chit scheme described below will give low
delay requesters a chance to use the allocated bandwidth.
If a datagram requests multiple classes of service, only one class
can be provided. For example, when both low delay and high
reliability classes are requested, and if the low delay queue is
full, queue the data on the high reliability queue instead. If we
are able to queue the data on the low delay queue, then the datagram
gets part of the high reliability service it also requested, because,
once data is queued, data will not be discarded. However, the
datagram will be routed as a low delay request. The same scheme is
used for any other combinations of service requested. The order of
selection for classes of service when more than one is requested
would be low delay, high throughput, then high reliability. If a
block of datagrams request multiple classes of service, it is quite
possible that datagram reordering will occur. If one queue is full
causing the other queue to be used for some of the data, data will be
forwarded at different service rates. Requesting multiple classes of
service gives the data a better chance of making it through the net
because they have multiple chances of getting on a service queue.
However, the datagrams pay the penalty of possible reordering and
more variability in the one way transmission times.
Besides total buffer consumption, individual class of service queue
sizes should be used to SQ those asking for service except as noted
above.
A request for regular class of service is handled by queuing to the
high reliability or high throughput queues evenly (proportional to
the service rates of queue). The low delay queue should only receive
data with the low delay service type. Its queue is too small to
accept other traffic.
Because of the small queue size for low delay suggested above, it is
difficult for low delay service requests to consume the bandwidth
allocated. To do so, low delay users must keep the small queue
continuously non-empty. This is hard to do with a small queue.
Traffic flow has been shown to be bursty in nature. In order for the
low delay queue to be able to consume the allocated bandwidth, a
count of the various types being forwarded should be kept. The
service rate should increase if the actual percentage falls too low
for the low delay queue. The measure of service rates would have to
be smoothed over time.
While this does sound complicated, a reasonably efficient way can be
described. Every Q seconds, where Q is less than or equal to the
MGD, each class gets N M P chits proportional to their allowed
percentage. Send data for the low delay queue up to the number of
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chits it receives decrementing the chits as datagrams are sent. Next
send from the high reliability queue as many as it has chits for.
Finally, send from the high throughput queue. At this point, each
queue gets N M P chits again. If the low delay queue does not
consume all of its chits, when a low delay datagram arrives, before
chit replenishment, send from the low delay queue immediately. This
provides some smoothing of the actual bandwidth made available for
low delay traffic. If operational experience shows that low delay
requests are experiencing excessive congestion loss but still not
consuming the classes allocated bandwidth, adjustments should be
made. The service rates should be made larger and the queue sizes
adjusted accordingly. This is more important on lower speed links
where the above formula makes the queue small.
What we should see during the Q seconds is that low delay data will
be sent as soon as possible (as long as the volume is below the
allowed percentage). Also, the tendency will be to send all the high
throughput datagrams contiguously. This will give a more regular
measured round trip time for bursts of datagrams. Classes of service
will tend to be grouped together at each intermediate node in the
route. If all of the queues with datagrams have consumed all of
their allocated chits, but one or more classes with empty queues have
unused chits then a percentage of these left over chits should be
carried over. Divide the remaining chit counts by two (with round
down), then add in the refresh chit counts. This allows a 50% carry
over for the next interval. The carry over is self limiting to less
than or equal to the refresh chit count. This prevents excessive
build up. It provides some smoothing of the percentage allocation
over time but will not allow an unused queue to build up chits
indefinitely. No timer is required.
If only a simple subset of the described algorithm is to be
implemented, then low delay queuing would be the best choice. One
should use a small queue. Service the queue with a high service rate
but restrict the bandwidth to a small reasonable percentage of the
available bandwidth. Currently, wide area networks with high traffic
volumes do not provide low delay service unless low delay requests
are able to preempt other traffic.
Applicability
When the output speed and volume match the input speed and volume,
queues don't get large. If the queues never grow large enough to
exceed the guaranteed low delay performance, no queuing algorithm
other than first in, first out, should be used.
The algorithm could be turned on when the main queue size exceeds a
certain threshold. The routing node can periodically check for queue
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build up. This queuing algorithm can be turned on when the maximum
delays will exceed the allowed nodal delay for low delay class of
service. It can also be turned off when queue sizes are no longer a
problem.
Issues
Several issues need to be addressed before type of service queuing as
described should be implemented. What percentage of the bandwidth
should each class of service consume assuming an infinite supply of
each class of service datagrams? What maximum delay (MGD) should be
guaranteed per node for low delay datagrams?
It is possible to provide a more optimal route if the queue sizes for
each class of service are considered in the routing decision. This,
however, adds additional overhead and complexity to each routing
node. This may be an unacceptable additional complexity.
How are we going to limit the use of more desirable classes of
service and higher priorities? The algorithm limits use of the
various classes by restricting queue sizes especially the low delay
queue size. This helps but it seems likely we will want to
instrument the number of datagrams requesting each Type-of-Service
and priority. When a datagram requests multiple classes of service,
increment the instrumentation count once based upon the queue
actually used, selecting, low delay, high throughput, high
reliability, then regular. If instrumentation reveals an excessive
imbalance, Internet operations can give this to administrators to
handle. This instrumentation will show the distribution for types of
service requested by the Internet users. This information can be
used to tune the Internet to service the user demands.
Will the routing algorithms in use today have problems when routing
data with this algorithm? Simulation tests need to be done to model
how the Internet will react. If, for example, an application
requests multiple classes of service, round trip times may fluctuate
significantly. Would TCP have to be more sophisticated in its round
trip time estimator?
An objection to this type of queuing algorithm is that it is making
the routing and queuing more complicated. There is current interest
in high speed packet switches which have very little protocol
overhead when handling/routing packets. This algorithm complicates
not simplifies the protocol. The bandwidth being made available is
increasing. More T1 (1.5 Mbps) and higher speed links are being used
all the time. However, in the history of communications, it seems
that the demand for bandwidth has always exceeded the supply. When
there is wide spread use of optical fiber we may temporarily
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experience a glut of capacity. As soon as 1 gigabit optical fiber
link becomes reasonably priced, new applications will be created to
consume it all. A single full motion high resolution color image
system can consume, as an upper limit, nearly a gigabit per second
channel (30 fps X 24 b/pixel X 1024 X 1024 pixels).
In the study of one gateway, Dave Clark discovered that the per
datagram processing of the IP header constituted about 20% of the
processing time. Much of the time per datagram was spent on
restarting input, starting output and queuing datagrams. He thought
that a small additional amount of processing to support Type-of-
Service would be reasonable. He suggests that even if the code does
slow the gateway down, we need to see if TOS is good for anything, so
this experiment is valuable. To support the new high speed
communications of the near future, Dave wants to see switches which
will run one to two orders of magnitude faster. This can not be done
by trimming a few instructions here or there.
From a practical perspective, the problem this algorithm is trying to
solve is the lack of low delay service through the Internet today.
Implementing only the low delay queuing portion of this algorithm
would allow the Internet to provide a class of service it otherwise
could not provide. Requesters of this class of service would not get
it for free. Low delay class of datagram streams get low delay at
the cost of reliability and throughput.
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