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                                                          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:


    -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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    -a protocol to govern communication between the Host and  the

     Pathway (PATHWAY ACCESS PROTOCOL).


    -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 (c) 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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                                                    Eric C. Rosen


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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IEN 187                              Bolt Beranek and Newman Inc.
                                                    Eric C. Rosen


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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                                                    Eric C. Rosen


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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                                                    Eric C. Rosen


                           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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