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rfc:ien:ien187
                                                        IEN 187
                   ISSUES IN INTERNETTING
               PART 2:  ACCESSING THE INTERNET
                        Eric C. Rosen
                Bolt Beranek and Newman Inc.
                          June 1981

IEN 187 Bolt Beranek and Newman Inc.

                                                  Eric C. Rosen
                   ISSUES IN INTERNETTING
               PART 2:  ACCESSING THE INTERNET

2. Accessing the Internet

   This is the second in a series of papers, the first of which

was IEN 184, that examine some of the issues in designing an

internet. Familiarity with IEN 184 is presupposed. This

particular paper will deal with the issues involved in the design

of internet access protocols and software. The issue of

addressing, however, is left until the next paper in this series.

Part of our technique for exposing and organizing the issues will

be to criticize (sometimes rather severely) the current protocols

and procedures of the Catenet, even though we do not, at the

present time, offer specific alternatives in all cases.

   In IEN 184, section 1.4,  we  outlined  four  steps  in  the

operation of a Network Structure. Let's now look closely at the

first step, viz., how the source Host actually submits a message

to the source Switch. In general, a Host will need to run three

separate protocols to do this:

  1. a protocol to utilize the electrical interface between the
   Host  and  the  initial  component of the Pathway it uses to
   access the source Switch.
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  1. a protocol to govern communication between the Host and the
   Pathway (PATHWAY ACCESS PROTOCOL).
  1. a protocol to govern communication between the Host and the
   source Switch (NETWORK ACCESS PROTOCOL).
   We  can  make  this  point  more  concrete  by  giving  some

examples. Consider the case of an ARPANET host which wants to

access the Catenet via the BBN gateway (which is also a Host on

the ARPANET). Then the ARPANET is the Pathway the host uses to

access the source Switch (the gateway). If the host is a local

or distant host, the electrical interface to the Pathway is the

1822 hardware interface. If it is a VDH host, the electrical

interface is whatever protocol governs the use of the pins on the

modem connectors. If it were an X.25 host, the interface might

be X.21. The PATHWAY ACCESS PROTOCOL is the 1822 protocol, which

governs communication between the host and the first IMP on the

Pathway. The NETWORK ACCESS PROTOCOL in this case would be the

DoD standard Internet Protocol (IP), which governs communication

between the host and the source Switch (gateway).

   If, on the other hand, we consider the case  of  an  ARPANET

host which is communicating with another host on the ARPANET, and

whose data stays purely within the ARPANET, 1822 becomes both the

NETWORK ACCESS PROTOCOL (since the source Switch is now identical

to the source IMP), and the PATHWAY ACCESS PROTOCOL, since the

Pathway is now the 1822 hardware connection.

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   We will have nothing further to  say  about  the  electrical

interface, since that is really just a straightforward hardware

matter. (However, such characteristics of the electrical

interface as error rate, for example, might have to be reflected

in the design of the Pathway Access Protocol.) The design of

both the Pathway Access Protocol and the Network Access Protocol

do raise a large number of interesting issues, and that shall be

the focus of this paper.

   We  believe  it  to  be very unlikely that Host software (or

gateway software) can utilize the internet efficiently unless it

takes the idiosyncrasies of BOTH the Pathway Access Protocol and

the Network Access Protocol into account. A gateway or host

software implementer who spends a great deal of time carefully

building his IP module, but who then writes a "quick and dirty"

1822 module, is likely to find that his inefficient use of 1822

completely sabotages the advantages which his carefully designed

IP is supposed to have. Experience with the ARPANET has shown

many times that poorly constructed host software can create

unnecessary performance problems. It seems, for example, that

many 1822 modules completely ignore the flow control restrictions

of the ARPANET, thereby significantly reducing the throughput

that they can obtain over the ARPANET. We have even encountered

many hosts which cannot properly handle some of the control

messages of the 1822 protocol, which also leads to a very

inefficient use of the ARPANET.

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   It is not difficult to understand why a  host  (or  gateway)

software implementer might overlook the issues having to do with

the proper use of the Pathway Access Protocol. There are a

number of pressures that, if not dealt with properly at a

management level, lead naturally to the neglect of Pathway Access

Protocol issues. An internet implementer might want to

concentrate on the "new stuff", viz., the Network Access

Protocol, IP, and may not be at all interested in the

idiosyncrasies of the older Pathway Access Protocol (1822). He

might be misled, by the belief that the packet-switching networks

underlying the internet should be transparent to it, into

believing that those packet-switching networks can be treated as

simply as if they were circuits. He might also be under pressure

to implement as quickly as possible the necessary functionality

to allow internet access. While this sort of pressure is very

common, the pressure to make the internet PERFORM well (as

opposed to the pressure simply to make it work at all) is

generally not felt until much (sometimes years) later. The

tendency to neglect performance considerations while giving too

much attention to simply obtaining the needed functionality in

the quickest way is also reinforced by such "modern" design

procedures as top-down design, and specification of protocols in

formal languages. While these procedures do have a large number

of advantages, they also serve to obscure performance issues. If

the researchers and designers of protocols, following modern

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design methodologies, do not give adequate consideration to

performance at the time of protocol design, one can hardly expect

the implementers to do so. Yet ARPANET experience has shown

again and again that decisions made at the level of

implementation, apparently too picayune to catch the attention of

the designers, can be important determinants of performance.

Still another reason why protocol software implementers might

tend to disregard the niceties of the Pathway Access Protocol is

the lack of any adequate protocol software certification

procedure. An ARPANET host could be connected to an IMP for

months, transferring large amounts of traffic, without ever

receiving certain 1822 control messages. Then some sort of

change in network conditions could suddenly cause it to receive

that control message once per hour. There really is no way at

present that the implementer could have possibly run tests to

ensure that his software would continue to perform well under the

new circumstances. This problem is somewhat orthogonal to our

main interests, but deserves notice.

   One  of  the  most  important  reasons why protocol software

implementers tend to ignore the details of the Pathway Access

Protocols is the "philosophical" belief that anyone working on

internet software really "ought not" to have to worry about the

details of the underlying networks. We will not attempt to

refute this view, any more than we would attempt to refute the

view of a person who claimed that it "ought not" to rain on his

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day off. We emphasized in IEN 184 that the characteristics of a

Network Structure's Pathways are the main thing that distinguish

one Network Structure from another, and that the problems of

internetting really are just the problems of how to build a

Network Structure with Pathways as ill-behaved as

packet-switching networks. Thus building a successful internet

would seem to be a matter of dealing specifically with the

behavior of the various Pathways, rather than ignoring that

behavior. We assume that that our task is to create an internet

which is robust and which performs well, as opposed to one which

"ought to" perform well but does not. It is true, as we have

said, that within the Network Structure of the Catenet, we want

to regard the ARPANET as a Pathway whose internal structure we do

not have to deal with, but that does NOT mean that we should

regard it as a circuit. Any internet Host or Switch (gateway),

TO PERFORM WELL, will have to have a carefully designed and tuned

Pathway Access Protocol module geared to the characteristics of

the Pathway that it accesses.

   The relationship between the Pathway Access Protocol and the

Network Access Protocol does offer a number of interesting

problems. For one thing, it appears that these protocols do not

fit easily into the OSI Open Systems model. If we are accessing

a single packet-switching network, the Network Access Protocol

appears to be a level 3 protocol in the OSI model, and the

Pathway Access Protocol appears to be a level 2 protocol.

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However, if we are accessing an internet, we still need the level

3 Network Access Protocol, but now the Pathway Access Protocol

also has a level 3 component, in addition to its level 2

component. So the Host is now running two different level 3

protocols, although the Network Access Protocol appears

functionally to be in a layer "above" the level 3 part of the

Pathway Access Protocol. Perhaps the main problem here is that

the OSI model has insufficient generality to capture the

structure of the protocols needed to access an internet like the

Catenet.

   It  is  interesting  to see how some of these considerations

generalize to the case of a Host which needs to access an

internet (call it "B") through a Pathway which is itself an

internet (call it "A"). Then the Host needs a Network Access

Protocol for the internet B, a Network Access Protocol for the

internet A (which is also its Pathway Access Protocol for

internet B), and a Network Access Protocol for the actual network

to which it is directly connected, which is also its Pathway

Access Protocol for internet A. As we create more and more

complicated Network Structures, with internets piled on top of

internets, the Hosts will have a greater and greater protocol

burden placed upon them. Ultimately, we might want to finesse

this problem by removing most of this burden from the Hosts and

putting it in the Switches, and giving the Switches knowledge of

the hierarchical nature of the (internet) Network Structure. For

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example, a Host on the ARPANET might just want to give its data

to some IMP to which it is directly connected, without worrying

at all about whether that data will need to leave the ARPANET and

travel via an internet. The IMP could decide whether that is

necessary, and if so, execute the appropriate protocol to get the

data to some internet Switch at the next highest level of

hierarchy. If the data cannot reach its destination within the

internet at that level, but rather has to go up further in the

hierarchy to another internet, the Switch at the lower level

could make that decision and execute the appropriate protocol.

With a protocol structure like this, we could have an arbitrarily

nested internet, and the Switches at a particular level, as well

as the Hosts (which are at the lowest level), would only have to

know how to access the levels of hierarchy which are immediately

above and/or below them. This procedure would also make the host

software conform more to the OSI model, since only one Network

Access Protocol would be required. However, this sort of

protocol structure, convenient as it might be for the Hosts, does

not eliminate any of the issues about how to most efficiently use

the Pathways of a Network Structure. Rather, it just pushes

those issues up one level, and makes the Switches correspondingly

more complicated. A proper understanding of the issues,

therefore, is independent of what sort of protocol structuring we

design.

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   Having  emphasized  the  need for hosts and gateways to take

account of the details of particular Pathway Access Protocols, we

must point out that this is not always a simple thing to do. If

the Network Structure underlying a Pathway is just a single

network, like the ARPANET, this problem is not so terribly

difficult, since one can expect that there will be available a

lot of experience and information about what a host should do to

access that network efficiently. If, on the other hand, the

Pathway is really an internet itself, the problem is more

difficult, since it is much more difficult to say anything

substantive about its characteristics. This is a point we must

keep in mind as we discuss specific issues in access protocol

design.

   In the remainder of this paper, we will attempt to deal with

a number of issues involved in the design of robust,

high-performance Network and Pathway Access Protocols. We will

not attempt to cover every possible issue here. In particular,

the issue of how to do addressing is important enough to warrant

a paper of its own, and shall be put off until the next paper in

this series. We will attempt throughout to focus on issues which

particularly affect the reliability of the internet configuration

(as perceived by the users), and on issues which affect the

performance of the internet (as perceived by the users).

Wherever possible, we will try to exhibit the way in which the

reliability and performance of a protocol trade off against its

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functionality. If protocol designers concentrate too heavily on

questions of what functionality is desired, as opposed to what

functionality can be provided at a reasonable level of

performance and reliability, they are likely to find out too late

that the protocol gives neither reasonable performance nor

reliability.

2.1 Pathway Up/Down Considerations

   In  general,  a  Host  will be multi-homed to some number of

Switches. In fact, it is easy to imagine a Host which is both

(a) multi-homed to a number of IMPs, within the Network Structure

of the ARPANET (this cannot be done at present, but is planned

for the future), and also (b) multi-homed to a number of gateways

(namely, all the gateways on the ARPANET) within the Network

Structure of the Catenet. Whenever a Host is multi-homed to a

number of Switches in some Network Structure, it has a decision

to make, namely, which of those Switches to use as the source

Switch for some particular data traffic. In order to make this

choice, the very first step a Host will have to take is to

determine which Switches it can reach through operational

Pathways. One thing we can say for sure is that if a Host cannot

reach a particular Switch through any of its possible Pathways,

then it ought not to pick that Switch as the source Switch to

which to send its data. In a case, for example, where the

ARPANET is partitioned, a Host on the ARPANET which needs to send

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internet traffic will want to know which gateways it can reach

through which of its ARPANET interfaces. To make this

determination possible, there must be some sort of "Pathway

Up/Down Protocol", by which the Host determines which of its

potential Pathways to gateways are up and which are down. This

is not to say, of course, that the Hosts have to know which

gateways are up and which are down, but rather, they must know

which gateways they can and cannot reach. Of course, this

situation is quite symmetric. The Switches of a Network

Structure (and in particular, the gateways of an internet) must

be able to determine whether or not they can reach some

particular host at some particular time. Otherwise, the gateway

might send traffic for a certain Host over a network access line

through which there is no path to that Host, thereby causing

unnecessary data loss. Apparently, this problem has occurred

with some frequency in the Catenet; it seems worthwhile to give

it some systematic consideration.

   The design of reliable Pathway Up/down protocols seems  like

something that "ought to be" trivial, but in fact can be quite

difficult. Let's begin by considering the case of an ARPANET

host which simply wants to determine whether it can reach some

IMP to which it is directly connected. The first step for the

host to take (if it is a local or distant host) is to look at the

status of its Ready Line. If the Ready Line to some IMP is not

up, then it is certain that communication with that IMP is not

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possible. If the host is a VDH host, then there is a special

up/down protocol that the host must participate in with the IMP,

and if that fails, the host knows that it cannot communicate with

the IMP. Of course, these situations are symmetric, in that the

IMP has the same need to know whether it can communicate with a

host, and must follow the same procedures to determine whether

this is the case. However, even in these very simple cases,

problems are possible. For example, someone may decide to

interface a host to an IMP via a "clever" front-end which hides

the status of the Ready Line from the host software. If a host

is multi-homed, and has to choose one from among several possible

source IMPs, but cannot "see" the Ready Lines, what would stop it

from sending messages to a dead IMP? Eventually, of course, a

user would notice that his data is not getting through, and would

probably call up the ARPANET Network Control Center to complain

about the unreliability of the network, which, from his

perspective, is mysteriously dropping packets. From the opposite

perspective, one must realize that such a front-end might also

hide the status of the host from the IMP, so that the network has

no way of knowing whether a particular host is currently capable

of communicating with the network. This is especially likely to

happen if the "clever" front-end takes packets from the network

which are destined for a particular host, and then just drops

them if the host is down, with no feedback to either IMP or host.

If a host is multi-homed, but one of its access lines is down,

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this sort of configuration might make it quite difficult for the

network to reach a reasonable decision as to which access line to

use when sending data to that host. The lesson, of course, is

that the status of the Ready Line should never be hidden from the

host software, but it is hard to communicate this lesson to the

designers of host software. Again, the issue is one of

performance vs. functionality. A scheme which hides the status

of the Ready Line from IMP or host may still have the required

(minimum) functionality, but it will just perform poorly under

certain conditions.

   This may seem like a made-up problem  which  probably  would

never occur, but in fact it has occurred. We once had a series

of complaints from a user who claimed that at certain times of

certain days he had been unable to transmit data successfully

over the ARPANET. Upon investigation, we found that during those

times, the user's local IMP had been powered down, due apparently

to a series of local power failures at the user's site. Of

course, the IMP will not transmit data when it is powered down.

But it was somewhat mysterious why we had to inform someone of a

power failure at his own site; surely the host software could

have detected that the IMP was down simply by checking the Ready

Line, and so informed the users. When this user investigated his

own host software (a very old NCP), he found that it would inform

the users that the IMP was down ONLY if the IMP sent the host a

message saying that it was going down. Since the IMP does not

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send a message saying that it is about to lose power, the host

software, which apparently did not check the Ready Line as a

matter of course, did not detect the outage. It looked to the

user, therefore, as though the network had some mysterious and

unreliable way of dropping packets on the floor. It seems that

many hosts presently exist whose networking software is based on

the assumption that the IMPs never go down without warning.

Hosts do sometimes have difficulty determining whether their

Pathway to an IMP is up or down, even when it seems like this

should be totally trivial to determine. Reliable network service

requires, however, that host software and hardware designers do

not hide the status of the IMP from the host, or the status of

the host from the IMP. This will become increasingly important

as more and more hosts become multi-homed.

   Of  course,  this  is  only a first step in a proper up/down

determination. It is not impossible for a Ready Line to be up

but for some problem either in IMP or host to prevent

communications from taking place. So some higher level up/down

protocol is also necessary. Some protocol should be defined by

which Host and Switch can send traffic to each other, and require

the other to respond within a certain time period. A series of

failures to respond would indicate that proper communications is

not possible, at least for the time being. It is important to

note, though, that the need for a higher level up/down protocol

does not obviate the need for the lower level procedure of

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monitoring the Ready Line. If the higher level procedure fails,

but the Ready Line appears to be up, knowledge of both facts is

needed for proper fault isolation and maintenance. Also

important to notice is that if the lower level procedure

indicates that the Pathway is down, the higher level procedure

should not be run. This might not seem important at first

glance, but in practice, it often turns out that attempting to

send traffic to a non-responsive machine results in significant

waste of resources that could be used for something more useful.

   In the more general case, where a Host's Pathway to a source

Switch may include one or more packet-switching networks, it is

far from trivial to determine whether the Switch can be reached

from the Host via the Pathway. Consider, for example, how a

given ARPANET host could determine whether a given Catenet

gateway on the ARPANET can be accessed via some given ARPANET

source IMP. Of course, the first step is to determine whether

communication with that source IMP is possible. Even if it is,

however, the gateway might still be unreachable, since it may be

down, or the network may be partitioned. ("Officially", every

ARPANET Host is supposed to be reachable from any other ARPANET

Host. However, the average connectivity of the ARPANET is only

2.5, which means that only a small number of line or node

failures are needed to induce partitions. Furthermore, a few

ARPANET sites are actually stubs, which means that a single

failure can isolate them from the rest of the ARPANET. As often

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seems to happen in practice, the sites that are stubs seem to be

attached by the least reliable lines, so that partitions are not

infrequent. At any rate, there will probably be networks in the

internet that partition more frequently than the ARPANET does.

Internet protocols must detect and react to network partitions,

instead of simply disregarding them as "too unlikely to worry

about." )

   In  the special case where the Pathway between some Host and

some Switch is the ARPANET, the ARPANET itself can provide

information to the Host telling it whether the Switch is

reachable. If the Switch is not reachable, and a Host attempts

to send an ordinary data packet to it, the ARPANET will inform

the Host whether or not that packet was delivered, and if not,

why not. Unfortunately, the current ARPANET does not provide

this information in response to datagrams. However, we have

already seen the need to provide such information in the case of

logically addressed datagrams (see IEN 183), and plan to

implement a scheme for doing so. An ARPANET Host which is also

an internet Host can implement a low level Pathway up/down

protocol simply by paying attention to the 1822 replies that it

receives from the ARPANET. There are hosts which seem to

disregard these 1822 control messages, and which seem to continue

to send messages for unreachable hosts into the ARPANET. Of

course, this is a senseless waste of resources which can severely

degrade performance. Indeed, it may look to an end-user, or even

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a gateway implementer, as though the ARPANET is throwing away

packets for no reason, when the real problem is that the host

software cannot respond adequately to exceptional conditions

reported to it by the network.

   We  have  spoken  of  the  need for Host and Switch to run a

higher level up/down protocol, to take account of the conditions

when one of them seems reachable to the network, but still will

not perform adequately when another entity attempts to

communicate with it. Switch and Host must run some protocol

together which enables each to validate the proper performance of

the other. The Catenet Monitoring and Control System (CMCC),

currently running on ISIE, runs a protocol of this sort with the

gateways. The CMCC sends a special datagram every minute to each

gateway, and expects to receive an acknowledgment (or echo) for

this special datagram back from the gateway. After three

consecutive minutes of not receiving the echo, the CMCC infers

that the gateway cannot be reached. After receiving a single

echo, the CMCC infers that the gateway can be reached. (Gateways

run a similar protocol with their "neighboring gateways".) A

Pathway up/down protocol which does not rely on the intervening

network to furnish the information would certainly have to

involve some such exchange of packets between the Host and the

Switch, but it would have to be rather more complex than this

one. One of the problems with this protocol is that it is

incapable of detecting outages of less than three minutes. This

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may be suitable for the CMCC's purposes, but is not generally

suitable for a Host which wants to know which source Switch to

send its traffic to. We would not want some Host to spend three

full minutes sending data to a Switch which cannot be reached;

the effect of that could be many thousands of bits of data down

the drain. (Of course, higher level protocols like TCP would

probably recover the lost data eventually through the use of

Host-Host retransmissions, but that involves both a severe drain

on the resources of the Host, which ought to be avoided whenever

possible, and a severe degradation in delay and throughput.)

Another problem with this particular protocol is that it uses

datagrams, which are inherently unreliable, and as a result, the

inference drawn by the CMCC is unreliable. From the fact that

three datagrams fail to get through, it is quite a big jump to

infer that no traffic at all can get through. Another problem is

the periodicity of the test packets. If they get in phase with

something else which may be going on in the network, spurious

results may be produced.

   The  design  of  a  Pathway  up/down  protocol  must also be

sensitive to the fact that some component network of a Pathway

may be passing only certain types of packets and not others. For

example, at times of heavy usage, certain networks may only be

able to handle packets of high priority, and lower priority

packets may either be refused by that net (at its access point),

or, even worse, discarded internally by the net with no feedback.

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The Pathway up/down protocol must be sensitive to this, and will

have to indicate that the Pathway is only "up" to certain classes

of traffic. If a Pathway is really a Network Structure which

will inform its Hosts when it cannot accept certain traffic

types, then this information can be fed back into the up/down

protocol. (Note however that this might be very difficult to do

if the Pathway consists of not a single network, but of an

internet). Alternatively, a Host may have to rely on its higher

level Pathway up/down protocol to determine, for several classes

of traffic, whether the Pathway is up to members of that class.

Apart from the inherent difficulty of doing this, it may be

difficult to map the traffic classes that a given component

network distinguishes into traffic classes that are meaningful to

a Host, or even to the Switches of the internet. Yet we wouldn't

want traffic to be sent into a network which is not accepting

that particular kind of traffic, especially if there are

alternative Pathways which would be willing to accept that

traffic.

   Many of these considerations suggest that the  higher  level

up/down protocols could turn out to be rather intricate and

expensive. Remember that a gateway may have many many hosts

"homed" to it, and must be able to determine, for each and every

one of these hosts, whether communication with it is possible.

Yet it probably is not feasible to suppose that each gateway can

be continuously running an up/down protocol with each potential

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host, and still have time left to handle its ordinary traffic.

This suggests that the primary up/down determination be made from

the low-level protocol, i.e., that the Switches should rely on

the networks underlying the Pathways to inform them whether a

given Host is up or down, and the Hosts should similarly rely on

the networks underlying the Pathways to pass them status

information about the gateways. It would be best if the higher

level up/down protocol only needed to be run intermittently, as a

check on the reliability of the lower level protocol.

Unfortunately, the use of low level up/down protocols is not

always possible. Many networks, unlike the ARPANET, do not even

gather any information about the status of their hosts, and hence

cannot inform a source Host that it is attempting to send data to

a destination Host which is not reachable. (SATNET is an example

of a network that does not pass "destination dead" information.)

In the case where a particular Host-Switch Pathway is itself an

internet, the problem is even worse. Unless the component

networks of that internet can be made to cooperate in obtaining

RELIABLE up/down information and passing it back to the source

Host, it will be very hard for a Host to make any reasonable

determination as to whether a particular Switch is reachable. We

would strongly recommend the incorporation of low level up/down

protocols in ALL component networks of the internet.

   There   is  another  important  problem  in  having  a  Host

determine which of its potential source Switches on the internet

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are up and which are down. In order to run a protocol with the

Switch, or even to query the lower level network about the

Switch, the Host must have some way of identifying the Switch.

It is not so difficult for a Host on the ARPANET to identify the

IMPs that it is directly connected to, since it is quite simple

to devise a protocol by which a Host can send a message down each

of its access lines, asking who is on the other end. It is

rather more difficult for a Host to find out which gateways it is

homed to (i.e., which gateways are on a common network with it).

There is no easy way for an ARPANET Host to find out which other

ARPANET hosts are Catenet gateways. There is no "direct

connection" at which to direct protocol messages. In the current

Catenet, hosts have to know in advance how to identify the

Catenet gateways on their networks (although there are certain

restricted circumstances under which a host can obtain the name

of another gateway from a gateway about which it already knows).

Yet it does not seem like a good idea to require a Host to know,

a priori, which other Hosts on its network are also internet

Switches. This makes it difficult to enable Hosts to take

advantage of newly installed gateways, without making changes by

hand to tables in the Hosts (a procedure which could require

weeks to take effect). There is a rather attractive solution to

this problem. If each component network in the internet can

determine for itself which of its Hosts are also internet

Switches (gateways), then the Switches of that network can

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provide that information to the Hosts. This would require the

existence of a protocol which the gateways run with the Switches

of the individual component networks, by means of which the

gateways declare themselves to be gateways. Each individual

network would also have to have some internal protocol for

disseminating this information to other Hosts, and for keeping

this information up to date. If the network allows GROUP

ADDRESSING, further advantages are possible. The network could

maintain a group address (called, say, "Catenet Gateways") which

varies dynamically as gateways enter and leave the network.

Hosts could find out which gateways are reachable over particular

network access lines by sending some sort of protocol message to

the group address, and waiting to see who replies. Hosts would

then not have to have any a priori knowledge of the gateways on

their home networks.

   One very important though often neglected aspect of  up/down

protocols is the way in which the up/down protocol interacts with

the ability to perform adequate maintenance of the Network

Structure. It is tempting to think that a Pathway up/down

protocol ought to declare a Pathway "down" only if it is totally

dead or otherwise totally unusable. But in fact, a pathway

should be declared down before it becomes totally dead, if its

packet "non-delivery rate" exceeds a certain threshold. (We use

the term "non-delivery rate" where the term "error rate" is more

commonly used. We are trying to emphasize that it is important

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to detect not only errors, in the sense of checksum errors, but

rather any circumstances, including but not limited to checksum

errors, which prevent the proper delivery of packets.) There are

two reasons for this:

   1) The existence  of  a  non-zero  non-delivery  rate  on  a
      Pathway  implies that some packets placed on that Pathway
      will not make it through  to  the  other  end.   In  most
      applications, these packets will have to be retransmitted
      at some higher level of protocol, or else by the end user
      himself  (packetized  speech is one of the few exceptions
      to this).  As the number  of  retransmissions  increases,
      the  delay  also increases, and the throughput decreases.
      So when the non-delivery rate reaches  a  certain  point,
      the  Pathway  should be removed from service, in order to
      improve delay and throughput.  Of  course,  this  assumes
      that  an  alternate  Pathway  is  available  with a lower
      non-delivery  rate.   Also,  other  things  being  equal,
      removing  bandwidth  from  a  Network Structure will also
      tend to increase  delay  and  reduce  throughput,  so  we
      really  want  the up/down protocol to pick out the proper
      cross-over point.
   2) It is often better to fix a Pathway at the first sign  of
      trouble  than  to  wait  for  it  to  fail  totally.  One
      implication of this is that the up/down  protocol  should
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      perform  equally  well  whether  or  not  the  Pathway is
      heavily loaded with traffic.  We would not want to use  a
      protocol  which  made  its determination solely by making
      measurements of user traffic, since that  protocol  would
      not  function  well  during  periods when user traffic is
      very light.  That is,  a  faulty  Pathway  with  no  user
      traffic would not be detected.  Yet if repair work has to
      be  done  on  a  Pathway,  we would most like to find out
      about it during lightly loaded hours, so that a  fix  can
      be  effected with minimal disruption, possibly before the
      heavily loaded hours begin.
   Another  important  characteristic  for  a  Pathway  up/down

protocol to have is the ability to determine the nature of the

Pathway "outage". This is quite important for fault isolation,

but is easy for a host software person to overlook, since he may

not be aware of such issues. If a Host cannot get its packets to

a Switch over a certain Pathway, it will want to regard that

Pathway as down, and will want to use an alternate Pathway. From

the Host perspective, it doesn't care whether the reason it can't

use the Pathway is because of a network partition, or because of

network congestion, or because of some other reason. However, if

the Host personnel want to be able to call up the Pathway

personnel and request that the problem be fixed, it's not enough

to say, "Your network isn't working; call me back when it's

fixed." The more information the Pathway up/down protocol can

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gather, the quicker a fix can be effected. In the case where the

Pathway is the ARPANET, quite a bit of information can be

gathered from proper instrumentation of the 1822 module, and

proper attention by the host software to the 1822 replies; this

will be discussed further in section 2.6.

   The design of the ARPANET's line up/down protocol might be a

good model for the design of a general Pathway up/down protocol.

The design of the ARPANET protocol was based upon a mathematical

analysis of the probabilistic error characteristics of telephone

circuits, and the protocol is intended to bring a line down when

and only when its error rate exceeds a threshold. However, the

error characteristics of Pathways in general (i.e., of

packet-switching networks) are not well understood at all, and

there is no similar mathematical analysis that we can appeal to.

At present, we can offer no ready answer to the question of how a

Host can tell which of several possible source Switches is

reachable, if the Switches are accessed via a network (or

sequence of networks) which will not even inform the Host whether

or not its traffic even gets delivered. This is an important

question which will require further thought, and considerable

experimentation.

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2.2 Choosing a Source Switch

   Once  a  Host  has  determined  which source Switches it can

reach over which of its interfaces, it still has to determine

which one to use for sending some particular packet (unless the

Host is "lucky" enough to find out that only one source Switch is

reachable). Making the proper choice can be quite important,

since the performance which the Host gets may vary greatly

depending upon which source Switch it selects. That is, some

source Switch might be much closer to the destination, in terms

of delay, than another. It then might be quite important to

choose the proper one. To make things a bit more concrete,

consider the case of a Host which is multi-homed (via several

distinct 1822 links) to several ARPANET IMPs, and whose traffic

can be handled entirely within the ARPANET. There are several

things a host might want to take into account in choosing the

best source IMP to use for a particular packet, including:

   1) The loading on the 1822  access  line  to  each  possible
      source IMP.
   2) The distance between each source IMP and the  destination
      Host, for some notion of "distance."
   The  first  of  these  two  quantities is relatively easy to

obtain, since all the Host need do is monitor its own 1822 lines;

it should be possible to devise a monitoring scheme which

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indicates which of the 1822 lines is providing the best service

to its IMP, perhaps simply by measuring the queuing delay

experienced in the Host by messages queued for that line. (Any

such measurement would have to take into account some of the

niceties of the 1822 protocol, though.) Obtaining information

about the second quantity is more difficult. The Host might try

to keep some measurement of round-trip delay (delay until a RFNM

is received) between itself and each destination Host. However,

in order to do this, some traffic for each destination Host would

have to be sent over each access line, so that the delay could be

measured. This means that some traffic has to be sent over a

long delay path, simply in order to determine that that is a long

delay path. A simpler scheme might be for the Host to get delay

information from the IMP. A Host could ask each potential source

IMP what its delay to the destination Host is. By using this

information, plus the information it gathers locally about the

loading of its access lines, the Host could determine which

source IMP provides the shortest path to the destination.

   This would require that we define a protocol by which a Host

can ask the IMPs to which it is homed to provide their delays to

a destination Host. The Host could make these requests

periodically, and then change its selection of source IMPs as

required in order to react to changes in delay. There are a few

subtle protocol issues to be considered here, though. We would

have to make sure that a Host cannot beat a Switch to death by

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constantly asking it what its delays are; probably we would have

to give the Switch the option of not replying to these requests

if it is too busy with other things (like ordinary data traffic).

A bigger problem lies in the assumption that the Switches will

even have this data to provide. The routing algorithm used by

the ARPANET IMPs does, in fact, provide each IMP with a value of

delay, in milliseconds, to each other IMP in the network. There

is no reason why this information could not be fed back to the

hosts on request. Note, however, that while a source IMP knows

its delay to each possible destination IMP, it does not know its

delay to each potential destination HOST over each possible

access line to that Host, since the routing algorithm does not

maintain measurements of delay from an IMP to a locally attached

host. Yet this latter delay might be quite significant. Still,

the information that the ARPANET IMPs could provide to the Hosts

should enable them to make a better choice than they could make

without this information.

   Another  problem  with this idea of having the Switches feed

back delay information to the Hosts is the proper choice of

units. If a Host is going to take the delay information provided

by the network and then add some locally measured delay

information to it, it is important for the Host to know what

units the network is using to measure delay. Yet we also have to

ensure that the network developers and maintainers are free to

change the way in which the network does measurements, and the

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units in which the measurements are taken, WITHOUT NEEDING TO

COORDINATE SUCH CHANGES WITH ALL HOST ADMINISTRATIONS. That is,

we don't want further development of the network, and further

refinements in the way network measurements are done, to be

overly constrained by the fact that the Hosts demand measurements

in a certain unit. We also want to ensure that host software

implementations are not invalidated by a decision to change the

units that the network uses for its internal measurements. So

the protocol would have to enable the Switch to tell the Host

what units it is providing; the Host would then make any

necessary conversions. (Alternatively, the Host could tell the

Switch what units it wants, and the Switch could do the

conversion before sending the information to the Host.)

   In  the  internet  environment,  the   situation   is   more

complicated. An ARPANET Host which is also an internet Host

would have to (a) figure out its delay to each of its source

IMPs, (b) query each source IMP for its delay to each source

gateway, and © query each source gateway about its delay to

each destination. There is no straightforward way to gather the

rest of the needed delay information, however, namely the delay

from the destination gateway to the destination Host. In more

complex Network Structures, with internets nested on top of

internets, this problem becomes increasingly more complex. It

seems that the only really reliable way, and the most

straightforward way, for the source Host to gather information

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about the delays via various source Switches to a destination

Host, is for it to do the measurements itself. This is the

recommended solution. Delay information should also be made

available from the component networks for Hosts which cannot do

this, but it should be understood that those hosts cannot expect

to get as good a quality of service as the hosts which go to more

trouble to do their own measurements.

2.3 Type of Service

   One  very  important  piece  of information that a Host must

specify to the source Switch through the Network Access Protocol

is the "type of service" desired. To quote from the DoD standard

Internet Protocol (IP) specification [1, p. 15], "The Type of

Service is used to indicate the quality of the service desired;

this may be thought of as selecting among Interactive, Bulk, or

Real Time, for example." This seems to make sense, since one

does have the feeling that different types of applications will

fall into different categories, and information about the

categories may help the Switches of the Network Structure through

which the data is moving decide how best to treat it. However,

choosing just the right set of categories of service is quite a

complex matter. For example, both a terminal user of a

time-sharing system, and a user of a query-response system (like

an automated teller) fall under the rubric of "interactive", but

that doesn't mean that the service requirements are the same.

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Both Remote-Job-Entry and File Transfer fall under the rubric of

"bulk", but it is not obvious that they have the same

requirements. Both real-time process control and packetized

voice fall into the category of "Real Time", but the requirements

of these two applications seem to be very different. A very real

issue, which has not yet been given adequate consideration, is

the question of just how many categories of application type

there really should be, and just what the implications of putting

a packet into one of these categories ought to be. As we go on,

we will see a number of problems that arise from failure to

carefully consider this issue.

   It is rather difficult to find examples  of  Network  Access

Protocols which have really useful class-of-service selection

mechanisms. The 1822 protocol allows the user to select from

among two priorities; it allows the choice of single-packet or

multi-packet messages; it allows the choice between "raw packets"

and "controlled packets." It is up to some user (or more

realistically, up to some host software implementer who may have

only a vague and limited understanding of the applications which

his software will serve, and of the network that he is accessing)

to map his application characteristics onto these three choices.

Unfortunately, it is doubtful that there is anyone outside of the

ARPANET group at BBN with any clear understanding of the

implications of making the various choices. The task of making

the optimum choice for some application is further complicated by

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the fact that the effects of making the various choices can be

very dependent on the network load. For example, it is often

possible to get more throughput from single-packet messages than

from multi-packet messages. This will happen if the destination

IMP has several different source Hosts sending multi-packet

messages to it, but is short on buffer space (as many of the

ARPANET IMPs are), and if the multi-packet messages contain only

two or three packets per message. Not only is this sort of thing

very difficult for an arbitrary user to understand (to a naive

network user, it must seem ridiculous), it is also subject to

change without notice. Although users can vary their service

significantly by sending optimum size messages, the principles

governing the "optimum" size are very obscure, and we cannot

really expect users to map their application requirements onto

this network feature in any reasonable manner.

   A  similar  problem  arises with respect to the priority bit

that the 1822 protocol allows. Basically, a priority packet will

get queued ahead of any non-priority packets on the queues for

the inter-IMP links and on the queues for the IMP-Host access

lines. However, priority packets receive no special preference

when competing with non-priority packets for CPU cycles or for

buffer space. Also, there is no notion at all in the ARPANET of

refusing to accept low priority packets because the network is

already too heavily loaded with high priority packets. Although

someone who has carefully studied the ARPANET might be able to

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say what the effect of setting the priority bit is under some

particular set of circumstances, some user who is wondering

whether his application requirements are best served by setting

the priority bit really has no way of answering that question.

The actual effect of the priority bit does not fully correspond

to any intuitive notion of priority that an arbitrary user is

likely to have. Another problem: although it is presently

allowed, it is not really a good idea to let the users choose

whether to set the priority bit or not. Fortunately, most hosts

do not submit packets with the priority bit on. It wouldn't be

terribly surprising, though, if some host software implementer

decided that he would always set the priority bit, in order to

get faster service. Of course, overuse of the priority bit just

means that it will have no effect at all, and that seems to mean

that its use must be controlled in some way, and not simply left

up to each user, as in the 1822 protocol.

   The  IP  offers  even  worse  problems  than  1822  in these

respects. Like 1822, the IP does not really allow the user to

classify his traffic according to application type. Rather, it

forces him to pick one of 5 possible precedence values (from

highest to lowest precedence, whatever that means, exactly), to

pick one of 4 reliability values (from most to least reliable),

to indicate whether he wants his data to be stream data or

datagram data in component networks for which this distinction is

meaningful, to indicate whether he wants high or low speed, and

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to indicate whether speed is more important to him than

reliability is. The idea here, apparently, is that any user can

map his application requirements into certain abstract

properties, and the information which the IP passes from the Host

to the source Switch is supposed to indicate which of these

abstract properties the user needs. At each internet hop, these

abstract properties are supposed to be mapped to particular

properties that are meaningful to the network in question. The

Pathway Access Protocol for that network would then be used to

indicate to the Switches of that component network what

particular properties the data transfer should have within that

network. In fact, the only apparent use of the "type of service"

information in the internet Network Access Protocol (IP) is to

carry information to be passed to the individual Pathway Access

Protocols.

   This  all  sounds  reasonable  enough when considered in the

abstract, but it gives rise to a large number of vexing problems

when we attempt to consider particular ways in which this "type

of service" information is to be used. Empirically, it seems

that few current gateway implementations take any notice of this

information at all. We suggest that the problem is not that the

individual implementers have not had time to write the code to

take account of this information, but rather that it is far from

clear how this information should be handled, or even that this

information is really meaningful. We suggest further that an

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internet user would also have a great deal of difficulty deciding

how to specify the "type of service" information in order to get

a specific quality of service needed by his application.

   Suppose a user needs the  maximum  possible  speed  for  his

application, so he uses IP to indicate that he values speed above

all else. What would the current Catenet do? For concreteness,

suppose there is a choice of sending this user's data either via

a sequence of 4 low-delay terrestrial networks, or through three

satellite networks, each of which contains two satellite hops.

The current implementation of the Catenet would send the data

through the three satellite networks. However, since the user

indicated that he values speed above all else, he will get the

fastest service that each of the satellite networks can provide!

Of course, this may not be what the user will have expected when

he asked for speed, since the fastest service through a satellite

network is not fast. A user may well wonder what the point of

specifying speed is, if his data is going to traverse some

sequence of satellite networks, even if a much faster path is

available. Furthermore, it is not correct to assume, in general,

that a user who values speed will really want the speediest

service through every network. If traffic must go through a

satellite network, it may be important to try to get one-hop

rather than two-hop delay, if this is possible. Yet it may not

be economical to also try to get the speediest service through

all terrestrial networks; the difference between high and low

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speed service through a terrestrial network might be "in the

noise", even when compared to the shortest delay through the

satellite network. It is not impossible, or even unlikely, that

better overall service (or more cost-effective service) can be

achieved by using the fastest possible service through some

networks, but less than the fastest through others. There are

two immediate lessons here. First, the characteristics that a

user specifies in the Network Access Protocol may require some

interaction with routing, since the characteristics he desires

simply cannot be provided, in general, by sending his traffic

through a random series of networks, and then mapping information

he specifies in the Network Access Protocol into information

carried in the individual Pathway Access Protocols. Second, what

a user means intuitively by "speed" just may not map into what

some particular component net means by "speed". Once again, we

see that the basic problem stems from the differing

characteristics of the Pathways in the Network Structure.

   Another  peculiar  feature  of the IP is the mysterious "S/R

bit", which a user is supposed to set to indicate whether he

prefers speed over reliability, or vice versa, should these

conflict. One unsuitable aspect of this is the apparent

assumption that it even makes sense to prefer either speed or

reliability over the other, without specifying more detail. It

is easy to imagine that some user is willing to accept

reliability of less than 100% if he can increase his speed

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somewhat. It is also easy to imagine that a user would be

willing to accept somewhat slower service if it gives him higher

reliability. But there will always be a range that the user

wants to stay within. If his reliability must be moved below a

certain threshold in order to get more speed, he may not want

this, even if he would be willing to say that he prefers speed to

reliability. Similarly, if his delay must go above a certain

threshold to gain more reliability, he may not want this, even

if, when talking in general terms, he says that he needs

reliability more than speed. It really doesn't make any sense at

all to try to map a particular application type into "speed over

reliability" or "reliability over speed", unless ranges and

thresholds are also specified. What this means in practice is

that a user will not be able to make a reasonable choice of how

to set this bit in the IP header; whatever he sets it to is bound

to produce results other than those he expects under some not too

uncommon set of circumstances.

   We  do  not  want to leave unquestioned the tacit assumption

that speed and reliability are opposing virtues, so that

increasing one must be expected to decrease the other. To quote

again from the IP spec, "typically networks invoke more complex

(and delay producing) mechanisms as the need for reliability

increases" [1, p 23]. This reasoning is somewhat superficial.

It may be true that in some networks, the less reliable kinds of

service are speedier, but this is not invariably the case. To

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see this, consider the following (fictitious) network. This

network allows the user to request either "reliable" or

"unreliable" data transfer. Reliable packets are controlled by a

set of protocols, both at the end-end and hop-hop level, which

ensure delivery. Unreliable packets are not under the control of

any such protocols. Furthermore, reliable packets go ahead of

unreliable ones on all queues, in particular, the CPU queue. In

addition, unreliable packets can be flushed from the net at any

time, if some resource they are using (such as buffer space) is

needed for a reliable packet. These latter two measures are

needed to ensure that the net does not become so heavily loaded

with unreliable packets that there is no room for the reliable

ones. (It would not make much sense to advertise a "reliable"

service, and then to allow the unreliable packets to dominate the

network by using most of the network resources. If unreliable

packets could grab most of the resources, leaving the "reliable"

ones to scavenge for the left-over resources, then it would be

virtually inevitable that the service received by the

"unreliable" packets would appear, to the users, to be more

reliable than the service received by the "reliable" packets. To

achieve a true dichotomy between reliable and unreliable service,

the reliable packets must be given priority in all respects over

the unreliable ones. We should also remember, by the way, that

although many protocols combine features of reliability,

sequentiality, error control, and flow control, these are not the

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same, and there is no reason why a network might not offer a

reliable but unsequenced service). This sort of network design

seems quite reasonable, perhaps more reasonable than the design

of any existing network. It would allow for a (presumably

inexpensive) class of service ("unreliable") which would be able

to use only those network resources not needed by the more

reliable (and expensive) class of packets, and which would not

suffer any additional delay due to the presence of the protocols

which would be needed to ensure reliability. In such a network,

unreliable packets might well experience less delay than reliable

ones, WHEN THE NETWORK IS LIGHTLY LOADED; WHEN IT IS HEAVILY

LOADED, HOWEVER, RELIABLE PACKETS WOULD TEND TO EXPERIENCE THE

SMALLER DELAY. If this is the case, it is hard to see how a user

could be expected to make a reasonable choice of IP service

parameters at all. He may know what his needs are, but we can

hardly expect him to know how to map his needs onto particular

aspects of the behavior of a particular network component of an

internet, especially when the behavior determined by that mapping

will vary dynamically with the network loading, and hence with

the time of day.

   Two other peculiarities of the "type of service" feature  of

the IP are worth mentioning. First, there seems to be no notion

of the relation between speed and priority, though in many

networks, the priority of a message is the major determinant of

its speed. (There are, to be sure, networks which attempt to

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treat priority solely as "acceptance class", differentiating it

completely from considerations of speed. However, we know of no

network implementation which has been shown to differentiate

SUCCESSFULLY between these two concepts, and there is reason to

doubt that this differentiation is even possible in principle.)

Second, one of the choices to be made is whether to prefer stream

or datagram service. This is a clear example of something that

is not based on "abstract parameters of quality of service", but

rather on a particular feature of one or two particular networks.

Requesting stream service will NOT do what a user might expect it

to do, namely set up a stream or virtual circuit through the

entire internet. This would require a lengthy connection set-up

procedure, involving reservations of resources in the gateways,

which resources are to be used only for specific connections. If

we are really serious about providing stream service, this is

just as important as obtaining stream service within the

component networks serving as the Pathways of the internet.

Indeed, it is hard to imagine any real use for an internet

"stream service" which treats packets as datagrams during most of

their lifetime in the internet, and then treats them as stream

packets in one or two component networks. It must be remembered

that the sort of stream service provided by a network like SATNET

is only useful to a user if his data appears at the SATNET

interface at fixed periods, synchronized with the scheduling of

the stream slots on the satellite channel. If the data must

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first travel through several datagram networks before reaching

SATNET, IT IS VIRTUALLY IMPOSSIBLE THAT THE DATA WILL ARRIVE AT

SATNET WITH THE PROPER PERIODICITY to allow it to make proper use

of the SATNET stream. Now there are certain specific cases where

it might be possible to provide some sort of stream service, say

if some data is going from a local network through SATNET to

another local network and thence directly to the destination

Host. (Though even in this case, some sort of connection set-up

and reservation of resources in the gateways between SATNET and

the local networks would probably be necessary.) Note, however,

that if a user requests this type of service, he is also

constraining the types of routes his data can travel. If SATNET

is not available, he might not want to use the internet at all at

that time. Or he might be willing to tolerate a less optimal

route ("half a loaf is better than none"), but might not want

"stream service" if the less optimal route has to be used. In no

case can a type of service like "stream" be obtained simply

through the mapping of "type of service" in the internet onto

"type of service" in the component networks.

   We do not want to have a Network Access Protocol  that  will

need to be infinitely expandable, so that the user can indicate

the type of service he wants in each particular network that his

data may eventually travel through. For one thing, as the

internet becomes larger, so that there are more paths between

each possible source and destination, the users will not

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generally know what set of networks their data will travel

through. Since the number of component networks in the internet

may be continually increasing, and since we cannot anticipate in

advance the features that each new network may offer, it does not

really seem reasonable to have to keep adding fields to the IP,

to account for particular characteristics of each new component

network. Yet this seems inevitable with the current approach.

That is, we do not agree with the claim in the IP spec that the

type of service field in the IP indicates "abstract parameters".

Rather, we think the type of service field has been constructed

with certain particular networks in mind, just those networks

which are currently in the Catenet, and that the various service

fields have no meaning whatsoever apart from the particular

"suggested" mappings to protocol features of specific networks

given in the spec. (And since these mappings are only

"suggested", not required, one might wonder whether the type of

service field really has any consistent meaning at all). This

situation is perhaps tolerable in a research environment, where

most of the users of the internet are explicitly concerned with

issues of networking, and willing to try a large number of

experiments to see what sort of service they get. One must

remember, however, that in a truly operational environment, the

average user will not be concerned at all about networking, will

not know anything about networking, will not care about

networking, and will only want the network to appear transparent

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to him. In order for such a user to make successful use of the

type of service field in a Network Access Protocol, the

parameters of the field must be meaningful to him. If they are

only meaningful to network experts, the user will never be able

to figure out how best to set these parameters.

   Rather than providing a type of service specification  which

is nothing but a sort of "linear combination" of the types of

service provided by the component networks, the internet ought to

offer a small, specific number of service types which are

meaningful at the application level. The possible values of

internet service type might be "interactive session,"

"transaction," "file transfer", "packetized speech," and perhaps

a few others. The categories should be simple enough so that the

user can figure out which category his particular application

falls into without needing to know the details of the operation

of the internet. The Switches of the internet should take

responsibility for sending the data on a route which is capable

of providing the requested type of service, and for sending the

data through component networks of the internet in a way which

maximizes the possibility that the type of service requested will

actually be achieved. Of course, in order to do this, we must

first answer a couple of hard questions, such as "Exactly what

characteristics of service do users want and expect for

particular applications?", and "What features must the internet

Switches have, and what features must the component networks

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have, in order to provide service with the necessary

characteristics?" In order to give adequate communications

service in an operational environment, however, these questions

must be given careful consideration by internet designers. To

some extent, these questions are difficult research issues, and

answering them will require doing some systematic experimentation

and instrumentation in the internet. The problem is hard, but

unavoidable. The IP's current approach seems aimed at

side-stepping these issues, since it places the burden entirely

on the user. It tends to give users the illusion that, by

properly specifying the bit fields in the IP header, they can

tune the internet to provide them with the specific type of

service they find most desirable. This is, however, only an

illusion. The perspective taken by the current IP seems to be

not, "How should the internet be designed so as to provide the

needed characteristics of service while providing a simple

interface to the user?", but rather, "Taking the current design

of the internet as a given, how can we give the user the ability

to massage, bend, and twist it so as to get service

characteristics which might be close to what he wants?" The

former perspective seems much more appropriate than the latter.

   Although  we  are  not  at  present  prepared  to  offer  an

alternative to IP, there are several lessons we would like to

draw from this discussion:

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   1) While an internet Network  Access  Protocol  really  does
      need  to  contain  some field which indicates the desired
      type of service in a manner which is abstract  enough  to
      be  mapped  to particular protocol features of particular
      networks, the  proper  specification  of  a  sufficiently
      abstract  set  of  parameters  is  an  open and difficult
      research issue, but one which needs to be studied  if  an
      operational internet configuration is ever to give really
      adequate service to a relatively naive end-user.
   2) Providing the  requested  type  of  service  may  require
      cooperation  from  all  the Switches (perhaps through the
      routing algorithm), and involves more than  just  mapping
      fields  from  the internet Network Access Protocol to the
      particular  access  protocols  used  by   the   component
      networks.   If  the type of service requested by the user
      is to be consistently meaningful, then his  request  must
      be  given  UNIFORM  treatment  by  the internet Switches.
      Different gateways must not  be  allowed   to  treat  the
      request differently.

2.4 Special Features

   The  DoD  Standard  Internet  Protocol  contains a number of

features which, while not strictly necessary in order for a user

to get his data delivered, and distinct from the type of service

field, do affect to some extent the service a user gets from the

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internet. Some of the features are worthy of comment, and that

is the purpose of this section.

2.4.1 Time to Live

   The presence of the "time-to-live" field in the  Catenet  IP

seems like a clear example of something that has no place in an

access protocol. The IP specification [1] has some contradictory

things to say about time-to-live. The user is supposed to set

this field to the number of seconds after which he no longer

cares to have his information delivered, or something like that.

It's far from clear how some user is supposed to make a decision

as to what value to set this to. For one thing, although this

value is supposed to be represented in units of one second [1, p.

24], there does not appear to be any requirement for the gateways

to figure out how many seconds to decrement this value by. The

spec actually says that each gateway should decrement this field

by at least one, even if it has no idea how much time has

actually elapsed [1, p. 40]. Well, a user might ask, is this

field represented in seconds or isn't it? What is the point of

saying in the spec that it is in seconds, if it is not

necessarily in seconds; this will only result in confusion. That

is, any attempt by a user to set this field to a reasonable value

is likely to have unanticipated consequences. Any attempt to

make inferences about internet behavior from the effect that

various settings of the time-to-live field will necessarily be

unreliable.

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   At any rate, unless the Switches  all  keep  a  synchronized

clock, there is no real way for them to determine how long a

packet has been in the network (or internet), as opposed to how

much time it has spent in the Switches, and this difference may

be significant if a packet is sent over several long-haul

networks with long-delay lines but fast Switches. It's hard to

see the point of requiring a user to specify, in the Network

Access Protocol, a value which cannot be assigned any consistent

meaning. (It's not clear what value this information has anyway;

according to the IP spec, "the intention is to cause

undeliverable datagrams to be discarded" [1, p. 24]. But a

reasonable routing algorithm should cause undeliverable datagrams

to be discarded anyway, no matter what value is specified for

time-to-live).

   It  seems  plain  in  any  case  that  over  the years, Host

personnel will begin to tend to set this field to its maximum

value anyway. In most implementations, the setting of this field

will not be left to the end-user, but will be in the code which

implements the IP. Several years from now, no one will remember

the importance of setting this field correctly. Eventually,

someone will discover that the data he sends to a certain place

does not get through, and after months of intensive

investigation, it will turn out that his IP is setting too small

a value in the time-to-live field, and his packets are dying just

before they reach their destination. This will make people tend

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to use the maximum value as a default, reducing the utility of

the information to almost nil. (No one will want to spend the

time re-tuning this value to the optimum as the internet

configuration expands, causing real packet delays to become

longer and longer. In fact, at many Host sites there may not be

anyone who can figure out enough of the Host code to be able to

re-tune this value.)

   Time-to-live, while useful for debugging purposes (perhaps),

has no real place in an operational system, and hence is not

properly part of a Network Access Protocol. If the Switches of a

Network Structure want to perform packet life timing functions,

in a way which is under the control of a single network

administration, and easily modified to reflect changing

realities, that is one thing. It is quite a different thing to

build this into a Host-level protocol, with a contradictory spec,

where it will certainly fall into disuse, or misuse. Protocol

features which are only useful (at best) for network

experimenters and investigators are bound to cause trouble when

invoked at the Host level, as part of a protocol which every Host

must implement, and whose implementers may not fully understand

the implications of what they are doing.

   Some  of  these difficulties have, as their basic cause, the

old implicit model of the internet that we discussed in IEN 185.

The IP conflates protocol features that properly belong to the

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Network Access Protocol with features that properly belong to the

protocol used internally among the Switches. This sort of

conflation, and consequent violation of protocol layering, are

inevitable if the gateways are seen as hosts which patch networks

together, rather than as Switches in an autonomous Network

Structure.

2.4.2 Source Routing

   The  current  IP  has  a  feature known as "source routing,"

which allows each user to specify the sequence of networks that

his internet packet is to travel. We mention this primarily as

an example of something that a Network Access Protocol in a truly

operational environment ought not to have. An acceptable

internet routing algorithm ought to distribute the traffic in

order to achieve some general goal on an internet-wide basis,

such as minimizing delay, maximizing throughput, etc. Any such

routing algorithm is subverted if each user is allowed to specify

his own route. Much of the routing algorithm's ability to

prevent or avoid congestion is also compromised if certain

packets are allowed to follow a route pre-determined by some

user, even if the routing algorithm determines that best service

(either for those packets themselves, or for other packets in the

internet) would be obtained if those packets followed a different

route.

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   To a certain extent, the  presence  of  the  source  routing

option in the IP is probably a result of the rather poor routing

strategy in the present Catenet, and a way of attempting to

obtain better service than the routing algorithm can actually

provide. The long-term solution to this problem would be to

improve the routing algorithm, rather than to subvert it with

something that is basically a kludge. We would claim that the

existence of any application or service that seems to require the

use of source routing is really an indication of some lack or

failure in the design of the internet, and a proper long-term

solution is to improve the situation by making basic

architectural changes in the internet, rather than by grafting on

new kludges.

   Source routing also has its use as an  experimental  device,

allowing tests to be performed which might indicate whether it is

really worthwhile to add some new feature or service to the

internet. (Although the way in which source routing subverts the

basic internet routing algorithm can have disturbing side-effects

on the experimental results, which must be properly controlled

for.) However, we doubt that any truly useful experiments

requiring source routing can be performed by individual users in

isolation. Rather, useful experiments would seem to require the

cooperation and coordination of the participating users as well

as those who are responsible for controlling and maintaining the

internet. So it is not clear that there is any true utility to

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having a source routing option at the level of the Network Access

Protocol, thereby giving each and every user the option of using

it. In an operational environment, this feature should either be

eliminated, or controlled through the use of authorizations,

which would cause gateways to discard source-routed packets which

lack proper authorization.

2.4.3 Fragmentation and Reassembly

   One  of  the  few  problems  which  is really specific to an

internet whose pathways consist of packet-switching networks is

the fact that it is difficult to specify to the user a maximum

packet size to use when giving traffic to the internet. If a

user's traffic is to go through EVERY component packet-switching

network, then the maximum packet size he can use is that of the

component network with the smallest maximum packet size. Yet it

seems unwise to require that no user ever exceed the maximum

packet size of the component network with the smallest maximum

packet size. To do so might lead to very inefficient use of

other component networks which permit larger packet sizes. If a

particular user's traffic does not happen to traverse the

component network with the smallest maximum packet size, the

restriction really does no good, and only leads to inefficiency.

Since, in a large internet, most traffic will probably traverse

only a small subset of the component networks, this is quite

important. In addition, some Hosts with limited resources might

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have a high overhead on a per-packet basis, making it quite

important to allow them to put larger packets into the internet.

   This gives rise to the question of, what should an  internet

Switch do if it must route a packet over a certain Pathway, but

that packet is larger than the maximum size of packets that can

be carried over that Pathway? The solution that has been adopted

in the current Catenet is to allow the internet Switches to

"fragment" the packets into several pieces whenever this is

necessary in order to send the packet over a Pathway with a small

maximum packet size. Each fragment of the original packet is now

treated as an independent datagram, to be delivered to the

destination Host. It is the responsibility of the destination

Host to reassemble the original packet from all the fragments

before passing it up to the next highest protocol layer. (If the

destination happens to have a high per-packet overhead, too bad.)

   The IP has several features whose only purpose is to  enable

this reassembly. These features are extremely general, so that

fragments can be further fragmented, ad infinitum, and correct

reassembly will still be possible. However, it seems that this

feature has not had very much operational testing in the Catenet;

gateway implementers seem to be as reluctant to actually

implement fragmentation as Host implementers are to implement

reassembly. If at least one gateway does do fragmentation, then

if some Host does not do reassembly, it cannot, in general, talk

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to any other Host on the internet. If a source Host knows that a

destination Host does not do reassembly, then it can, through IP,

indicate to the gateways that they ought not to fragment.

However, in that case, any datagrams that are not fragmentable

but which must be transmitted over a Pathway with a smaller

maximum packet size are simply lost in transit.

   It should be noted that the procedure of doing reassembly in

the destination Host violates the precepts of protocol layering

in a basic way. The internet is not transparent to protocol

modules in the Hosts, since a datagram put into the internet by a

protocol module in the source Host might appear at the

destination Host in quite a different form, viz., as a set of

fragments. One might try to avoid this conclusion by claiming

that what we have been calling "the Host software modules" are

really part of a Switch, rather than part of a Host, so that no

transparency is violated. One could also claim that a dog has

five legs, by agreeing to call its tail a leg. But this would no

more make a tail a leg than calling a Host software module "part

of the network" makes it so. One of the main advantages of

properly layered protocols is the ability it provides to change

the network without having to change the Hosts. This is needed

if changes to the network are even to be possible, since any

change that requires Host software to change is, for all

practical purposes, impossible. This suggests that the boundary

of the network be drawn at the boundary where changes are

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possible without coordination among an unlimited number of Host

administrations, and the natural place to draw this boundary is

around the Switches. While the Switches of a Network Structure

can all be under the control of a common administration, the

Hosts cannot. This suggests that any violation of protocol

layering that is as gross as the need to have Hosts do reassembly

is a problem that is to be avoided whenever possible.

   The problems of writing Host-level software to do reassembly

in a reliable manner do not seem to have been fully appreciated.

If a Host's resources (such as buffer space, queuing slots, table

areas, etc.) are very highly utilized, all sorts of performance

sub-optimalities are possible. Without adequate buffer

management (see IEN 182), even lock-ups are possible. One must

remember that reassembly is not a simple matter of sending the

fragments to the next higher level process in proper sequence.

The situation is more complex, since the first fragment of a

datagram cannot be sent up to the next higher protocol level

until all the fragments of that datagram are received. If

buffers are not pre-allocated at the destination Host, then

fragments of some datagrams may need to be discarded to ensure

that there is room to hold all the fragments of some other

datagram; otherwise "reassembly lockup" is possible. If the

internet gateways really did a large amount of fragmentation, so

that Hosts needed to do a large amount of reassembly, this would

almost certainly give rise to a variety of peculiar performance

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problems and phasing effects which could make the recently

discovered "silly window syndrome" look quite benign.

Unfortunately, it is hard to gain an appreciation of these sorts

of problems until one has personally encountered them, at which

point it is often too late to do anything about them.

   Performance   considerations   (as   opposed    simply    to

considerations of functionality) would seem to indicate that

fragmentation and reassembly be avoided whenever possible. Note

that performance problems associated with reassembly might crop

up suddenly at any time in the life of the internet, as some Host

which rarely received fragments in the past suddenly finds itself

bombarded with them, possibly due to a new application. Since

this sort of effect is notoriously difficult to test out in

advance, one would expect potential problems to be lying in wait.

Problems like these tend to crop up at a time when the Host

administration has no one available who understands and can

modify the Host software, which means that such problems can be

very intransigent and difficult to remedy. Of course, problems

in Host networking software are usually blamed on the network

(i.e., on the Switches), which also does not help to speed

problem resolution.

   One way to remove this sort of problem from the Host  domain

is to have the destination Switches themselves do any necessary

reassembly before passing a datagram on to its destination Host.

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This has the advantage that problems which arise will fall under

the domain of the Network administration, which is more likely to

be able to deal with them than are the various Host

administrations. However, this really does not simplify the

situation, or reduce the amount of performance sub-optimalities

that we might be faced with; it just takes the same problems and

puts them somewhere else. ARPANET IMPs do fragmentation (though

only at the source IMP) and reassembly at the destination IMP,

and this has turned out to be quite a tricky and problem-strewn

mechanism. Other approaches should be investigated.

   Of course, one possible way around fragmentation is to adopt

a policy of not routing any packets over Pathways which cannot

handle packets of that size. If there are several possible

routes between source and destination, which have similar

characteristics except for the fact that one of them has a

maximum packet size which is too small, the most efficient means

of handling this problem might just be to avoid using the route

which would require fragmentation. Even if this means taking a

slightly longer route to the destination, the extra delay imposed

during internet transit might be more than compensated for by the

reduction in delay that would be obtained by not forcing the

destination Host to do reassembly. Of course, this scheme

requires interaction with routing, but as long as there are a

small number of possible maximum packet sizes, this scheme is not

difficult to implement (at least, given a reasonable routing

algorithm).

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   Unfortunately, it might be the case that there  just  is  no

route at all to a particular destination, or else no reasonable

route, which does not utilize a Pathway whose maximum packet size

is "too small." In this case, there seems no way around

fragmentation and reassembly. However, a scheme which is worth

considering is that of doing hop-by-hop fragmentation and

reassembly within the internet. That is, rather than having

reassembly done at the destination (Switch or Host), it is

possible to do reassembly at the Switch which is the exit point

from a component network which has an unusually small packet

size. Datagrams would be fragmented upon entry to such networks,

and reassembled upon exit from them, with no burden on either the

destination Switch or the destination Host. The fact that

fragments would never travel more than one hop without reassembly

ameliorates the performance problems somewhat, since the amount

of time a partially reassembled datagram might have to be held

would be less, in general, than if reassembly were done on an

end-end basis.

   A  strategy of doing hop-by-hop reassembly and fragmentation

also allows more efficient use of the internet's Pathways in

certain cases. One problem with the end-end strategy is the

essential "randomness" of its effects. Consider, for example, a

large packet which must traverse several networks with large

maximum packet sizes, and then one network with a small maximum

packet size. The current method of doing fragmentation and

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reassembly allows the packet to remain large throughout the

networks that can handle it, fragmenting it only when it reaches

its final hop. This seems efficient enough, but consider the

case where the FIRST internet hop is the network with the

smallest maximum packet size, and the remaining hops are networks

with large maximum packet sizes. The current strategy then

causes a very inefficient use of the internet, since the packet

must now travel fragmented through ALL the networks, including

the ones which would allow the larger packet size. If some of

these networks impose constraints on a per-packet basis (which

might either be flow control constraints, or monetary constraints

based on per-packet billing), this inefficiency can have a

considerable cost. Hop-by-hop reassembly, on the other hand,

would allow the large packet to be reassembled and to travel

through the remaining networks in the most cost-effective manner.

Such a strategy is most consonant with our general thesis that an

efficient and reliable internet must contain Switches which are

specifically tuned to the characteristics of the individual

Pathways. It also removes the problem from the Host domain,

making the system more consonant with the precepts of protocol

layering.

   There is, unfortunately, one situation in  which  hop-by-hop

fragmentation cannot work. If the Pathway between some

destination Host and the destination Switch has a small maximum

packet size, so that the destination Switch must fragment

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datagrams intended for that Host, then reassembly must be done by

the Host itself, since there is no Switch at the other end of the

Pathway to do the reassembly. This seems to mean that Hosts

whose "home networks" have unusually small maximum packet sizes

will be forced to implement the ability to perform reassembly,

and must tolerate any resultant performance disadvantages.

2.5 Flow Control

   The  topic  of  "flow  control"  or "congestion control" (we

shall be employing these terms rather interchangeably, ignoring

any pedantic distinctions between them) breaks down naturally

into a number of sub-topics. In this section we shall be

concerned with only one such sub-topic, namely, how should the

Switches of the Network Structure enforce flow control

restrictions on the Hosts? We shall not consider here the issue

of how the Switches should do internal flow control, or what

protocols they need to run among themselves to disseminate flow

control information, but only the issue of how the results of any

internal flow control algorithm should be fed back to the hosts.

The IP is a rather unusual Network Access Protocol, in that it

does not have any flow or congestion control features at all.

This makes it very different from most other Network Access

Protocols, such as 1822 or X.25, which do have ways of imposing

controls on the rate at which users can put data into the

network. The IP, on the other hand, is supposed to be a

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"datagram protocol", and therefore (?) is not supposed to impose

any flow or congestion control restrictions on the rate at which

data can be sent into the internet. In this section, we will

discuss whether this is appropriate, and whether the "therefore"

of the previous sentence is really correctly used.

   The  issue  of  how  flow or congestion control restrictions

ought to be passed back to a Host, or more generally, how a

Network Structure ought to enforce its congestion control

restrictions, is a tricky issue. Particularly tricky is the

relation between datagram protocols and flow control. Datagrams

are sometimes known (especially with reference to the ARPANET) as

"uncontrolled packets," which tends to suggest that no flow

control should be applied to them. This way of thinking may be a

holdover from the early days of the ARPANET, when it was quite

lightly loaded. In those days, the flow control which the

ARPANET imposes was much too strict, holding the throughput of

particular connections to an unreasonably low value. Higher

throughput could often be obtained by ignoring the controls, and

just sending as much traffic as necessary for a particular

application. Since the network was lightly loaded, ignoring the

controls did not cause much congestion. Of course, this strategy

breaks down when applied to the more heavily loaded ARPANET of

today. Too much uncontrolled traffic can cause severe

congestion, which reduces throughput for everybody. Therefore

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many people now tend to recognize the need to control the

uncontrolled packets, if we may be forgiven that apparent

contradiction. Clearly, there is some tension here, since it

makes little sense to regard the same traffic as both

"controlled" and "uncontrolled." If a Network Access Protocol is

developed on the assumption that it should be a "datagram

protocol", and hence need not apply any controls to the rate at

which data can be transferred, it will not be an effective medium

for the enforcement of flow control restrictions at the

host-network access point. If congestion begins to become a

problem, so that people gradually begin to realize the importance

of congestion control, they will find that the Network Access

Protocol gives them no way to force the Hosts to restrict their

traffic when that is necessary. The probable result of this

scenario would be to try to develop a scheme to get the

congestion control information to the Hosts in a way that

bypasses the Network Access Protocol. This is our "logical

reconstruction" of the current situation in the Catenet. When

gateways think that there is congestion, they send "source

quench" packets to the Hosts themselves, and the Hosts are

supposed to do something to reduce the congestion. This source

quench mechanism should be recognized for what it is, namely a

protocol which is run between EVERY host and EVERY Switch

(including intermediate Switches, not just source Switches)

within a Network Structure, and which completely bypasses the

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Network Access Protocol (IP). This violates protocol layering in

a very basic way, since proper layering seems to imply that a

source Host should have to run a protocol with a source Switch

only, not with every Switch in the network.

   Of  course,  the fact that some mechanism appears to violate

the constraints of protocol layering is not necessarily a fatal

objection to it. However, given the present state of the art of

flow control techniques, which is quite primitive, flow control

procedures must be designed in a way that permits them to be

easily modified, or even completely changed, as we learn more

about flow control. We must be able to make any sort of changes

to the internal flow control mechanism of a Network Structure

without any need to make changes in Host-level software at the

same time. ARPANET experience indicates quite clearly that

changes which would be technically salutary, but which require

Host software modifications, are virtually impossible to make.

Host personnel cannot justify large expenditures of their own to

make changes for which they perceive no crucial need of their

own, just because network personnel believe the changes would

result in better network service. If we want to be able to

experiment with different internal flow control techniques in the

internet, then we must provide a clean interface between the

internal flow control protocols, and the way in which flow

control information is fed back to the Hosts. We must define a

relatively simple and straightforward interface by which a source

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Switch can enforce flow control restrictions on a Host,

independently of how the source Switch determines just what

restrictions to enforce. The way in which the Switches determine

these restrictions can be changed as we learn more about flow

control, but the Host interface will remain the same.

   It  is  not  clear that the source quench mechanism has been

generally recognized as a new sort of protocol, which bypasses

the usual Network Access Protocol for the internet (IP). One

reason that it may seem strange to dignify this mechanism with

the name of "protocol" is that no one really knows what a source

quench packet really means, and no one really knows what they are

supposed to do when they get one. So generally, they are just

ignored, and the "procedure" of ignoring a control packet seems

like a very degenerate case of a protocol. Further, the source

quench mechanism is a protocol which Host software implementers

seem to feel free to violate with impunity. No implementer could

decide to ignore the protocols governing the form of addresses in

the internet, or he would never be able to send or receive data.

Yet there is no penalty for ignoring source quench packets,

although violating the flow control part of the internetting

protocol seems like something that really ought to be prohibited.

(We have even heard rumors of Host software implementers who have

decided to increase their rate of traffic flow into the internet

upon receiving a source quench packet, on the grounds that if

they are receiving source quench packets, some of their traffic

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is not getting through, and therefore they had better retransmit

their traffic right away.)

   We  have  spoken  of  a  source Switch needing to be able to

ENFORCE flow control restrictions, by which we mean that when a

source Switch determines that a certain source Host ought to

reduce its rate of traffic, the Switch will REFUSE to accept

traffic at a faster rate. Proper flow control can never be

accomplished if we have to rely either on the good will or the

good sense of Host software implementers. (Remember that Host

software implementations will continue for years after the

internet becomes operational, and future implementers may not be

as conversant as current implementers with networking issues).

This means a major change to the IP concept. Yet it seems to

make much more sense to enhance the Catenet Network Access

Protocol to allow for flow control than to try to bypass the

Network Access Protocol entirely by sending control information

directly from intermediate Switches to a Host which is only going

to ignore it.

   We  will  not discuss internal flow control mechanisms here,

except to say that we do not believe at all in "choke packet"

schemes, of which the source quench mechanism is an example.

Eventually, we will propose an internal congestion control scheme

for the internet, but it will not look at all like the source

quench mechanism. (Chapters 5 and 6 of [2] contain some

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interesting discussions of congestion control in general, and of

choke packet schemes in particular.) It appears that some

internet workers are now becoming concerned with the issue of

what to do when source quench packets are received, but this way

of putting the question is somewhat misdirected. When you get

some information, and you still don't know what decision to make

or what action to take, maybe the problem is not so much in the

decision-making process as it is in the information. The proper

question is not, "what should we do when we get source quench

packets?", but rather "what should we get instead of source

quench packets that would provide a clear and meaningful

indication as to what we should do?

   Does  this  mean  that  the internet Network Access Protocol

should not really be a datagram protocol? To some extent, this

is merely a terminological issue. There is no reason why a

protocol cannot enforce congestion or flow control without also

imposing reliability or sequentiality, or any other features that

may unnecessarily add delay or reduce throughput. Whether such a

protocol would be called a "datagram protocol" is a matter of no

import. It is worth noting, though, that the Network Access

Protocol of AUTODIN II (SIP), while officially known as a

datagram protocol, does impose and enforce flow control

restrictions on its hosts.

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   The  only  real  way for a source Switch to enforce its flow

control restrictions on a source Host is simply for the Switch to

REFUSE packets from that Host if the Host is sending too rapidly.

At its simplest, the Switch could simply drop the packets, with

no further action. A somewhat more complex procedure would have

the Switch inform the Host that a packet had been dropped. A yet

more complex procedure would tell the Host when to try again.

Even more complex schemes, like the windowing scheme of X.25, are

also possible. To make any of these work, however, it seems that

a source Switch (gateway) will have to maintain Host-specific

traffic information, which will inevitably place a limit on the

number of Hosts that can be accessing a source Switch

simultaneously. Yet this seems inevitable if we are to take

seriously the need for flow control. At any rate, the need for

flow control really implies the need for the existence of such

limits.

2.6 Pathway Access Protocol Instrumentation

   Fault  isolation  in  an  internet  environment  is  a  very

difficult task, since there are so many components, and so many

ways for each to fail, that a performance problem perceived by

the user may be caused by any of a thousand different scenarios.

Furthermore, by the time the problem becomes evident at the user

level, information as to the cause of the problem may be long

gone. Effective fault isolation in the internet environment will

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require proper instrumentation in ALL internet components,

including the Hosts. We will end this paper with a few remarks

about the sort of instrumentation that Hosts should have, to help

in fault-isolation when there is an apparent network problem. We

have very often found people blaming the ARPANET for lost data,

when in fact the problem is entirely within the host itself. The

main source of this difficulty is that there often is no way for

host personnel to find out what is happening within the host

software. Sometimes host personnel will attempt to deduce the

source of the apparent problem by watching the lights on the IMP

interface blink, and putting that information together with the

folklore that they have heard about the network (which folklore

is rarely true). Our ARPANET experience shows quite clearly that

this sort of fault-isolation procedure just is not useful at all.

What is really needed is a much more complex, objective, and

SYSTEMATIC form of instrumentation, which unfortunately is much

more difficult to do than simply looking at the blinking lights.

   Some  sorts  of essential instrumentation are quite specific

to the sort of Network Access Protocol or Pathway Access Protocol

that is being used. For example, users of the ARPANET often

complain that the IMP is blocking their host for an excessive

amount of time. By itself, this information is not very useful,

since it is only a symptom which can have any of a large number

of causes. In particular, the host itself may be forcing the IMP

to block by attempting to violate ARPANET flow control

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restrictions. One sort of instrumentation which would be useful

for the host to have is a way of keeping track of the total time

it is blocked by the IMP, with the blocking time divided into the

following categories:

   1) Time blocked between messages.
   2) Time blocked between the leader of a message and the data
      of the message.
   3) Time blocked between packets.
   4) Time blocked while  attempting  to  send  a  multi-packet
      message (a subset of 2).
   5) Time blocked during transmission of the data portion of a
      packet.
   6) Time blocked while attempting to transmit a  datagram  (a
      subset of 2).
   While  this information might be very non-trivial for a host

to gather, it does not help us very much in fixing the problem

just to know that "the IMP is blocking" unless we can get a

breakdown like this. In addition, it is useful to have those

categories further broken down by destination Host, in case the

blocking is specific to some particular set of hosts.

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   Additional useful information has to do with the 1822  reply

messages. What percentage of transmitted messages are replied to

with RFNMs? with DEADs? with INCOMPLETEs? This should also be

broken down by destination host. In fact, it would be useful to

keep track of the number of each possible 1822 IMP-host control

message that is received. When problems arise, it may be

possible to correlate this information with the problem symptoms.

   The  basic idea here should be clear -- besides just telling

us that "the network isn't taking packets fast enough", host

personnel should be able to tell us under what conditions the

network is or is not taking packets, and just what "fast enough"

means. If a host is also running an access protocol other than

(or in addition to) 1822, there will be specific measurements

relevant to the operation of that protocol, but in order to say

just what they are, one must be familiar with those particular

protocols. (Again we see the effects of particular Pathway

characteristics, this time on the sort of instrumentation needed

for good fault isolation.) In general, whenever any protocol

module is designed and implemented, the designer AND implementer

(each of whom can contribute from a different but equally

valuable perspective) should try to think of anything the

protocol or the software module which implements it might do

which could hold up traffic flow (e.g., flow control windows

being closed, running out of sequence number space, failing to

get timely acknowledgments, process getting swapped out, etc.),

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and should be able to gather statistics (say, average and maximum

values of the amount of time data transfer is being held up for

each possible cause) which tell us how the protocol module is

performing.

   If  a protocol requires (or allows) retransmissions, rate of

retransmission is a very useful statistic, especially if broken

down by destination host.

   Hosts should be able to supply statistics on the utilization

of host resources. Currently, for example, many hosts cannot

even provide any information about their buffer utilization, or

about the lengths of the various queues which a packet must

traverse when traveling (in either direction) between the host

and the IMP. Yet very high buffer utilization or very long

queues within the host may be a source of performance problems.

When a packet has to go through several protocol modules within a

host (say, from TELNET to TCP to IP to 1822), the host should be

able to supply statistics on average and maximum times it takes

for a packet to get through each of these modules. This can help

in the discovery of unexpected or unanticipated bottlenecks

within the host. (For example, packets may take an unexpectedly

long amount of time to get through a certain module because the

module is often swapped out. This is something that is

especially likely to happen some years after the host software is

initially developed, when no one remembers anymore that the host

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networking software is supposed to have a high priority. This

sort of instrumentation can be quite tricky to get just right,

since one must make sure that there is no period of time that

slips between the time-stamps). The offered and obtained

throughputs through each protocol module are also useful

statistics. In addition, if a host can ever drop packets, it

should keep track of this. It should be able to provide

information as to what percentage of packets to (or from) each

destination host (or source host) were dropped, and this should

be further broken down into categories indicating why the packets

were dropped. (Reasons for hosts' dropping packets will vary

from implementation to implementation).

   Note that this sort of instrumentation  is  much  harder  to

implement if we are using datagram protocols than if we are using

protocols with more control information, because much of this

instrumentation is based on sent or received control information.

The less control information we have, the less we can instrument,

which means that fault-isolation and performance evaluation

become much harder. This seems to be a significant, though not

yet widely-noticed, disadvantage of datagram protocols.

   Host personnel may want to consider having  some  amount  of

instrumentation in removable packages, rather than in permanently

resident code. This ability may be essential for efficiency

reasons if the instrumentation code is either large or slow. In

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that case, it might be necessary to load it in only when a

problem seems evident. Instrumentation should also have the

ability to be turned on and off, so that it is possible to gather

data over particular time windows. This is necessary if the

instrumentation is to be used as part of the evaluation of an

experiment.

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                         REFERENCES

1. "DOD Standard Internet Protocol," COMPUTER COMMUNICATION REVIEW, October 1980, pp. 12-51.

2. "ARPANET Routing Algorithm Improvements," BBN Report No. 4473, August 1980.

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