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From: Iljitsch van Beijnum <iljitsch@muada.com>
Subject: Re: [RAM] Number of DFZ routers - radical improvement of BGP unlikely
Date: Thu, 6 Sep 2007 18:46:36 +0200
To: Robin Whittle <rw@firstpr.com.au>
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[Sorry to again reply many weeks later]

On 7-jul-2007, at 3:42, Robin Whittle wrote:

> I can't imagine any
> backwards-compatible upgrade or incrementally deployable
> replacement for BGP could achieve the radical improvements I was
> hoping for.

> Maybe someone else will have some great ideas for dramatically
> souping up BGP.  BGP solves some very difficult problems.  I can't
> think of any drastic improvements.

That's an interesting challenge. As the saying goes: given enough  
thrust, pigs fly just fine. The question is whether it would be  
easier to replace it. So far, I can't think of anyone else other than  
myself speaking out in favor of that, or even entertaining the  
question seriously.  :-)

Obviously a new protocol would have to be able to interact with BGP  
and be not much worse at talking BGP than a native speaker.

_______________________________________________
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From ram-bounces@iab.org Thu Sep 06 21:47:56 2007
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Subject: Re: [RAM] Number of DFZ routers - radical improvement of BGP unlikely
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Date: Thu,  6 Sep 2007 21:47:44 -0400 (EDT)
From: jnc@mercury.lcs.mit.edu (Noel Chiappa)
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    > From: Iljitsch van Beijnum <iljitsch@muada.com>

    > The question is whether it would be easier to replace it. So far, I
    > can't think of anyone else other than myself speaking out in favor of
    > that, or even entertaining the question seriously. :-)

Really? :-)

    > Obviously a new protocol would have to be able to interact with BGP and
    > be not much worse at talking BGP than a native speaker.

Well, "interact with BGP" and "talk BGP" sound like two very different things
to me - unless by "talk BGP" you mean "emulate a BGP speaker".

The thing is that if you have a system with a fundamentally different model
of the world (e.g. map-based, instead of route-table based), the interface
between it and BGP is necessarily going to be something of a kludge, and any
emulation is perforce going to be be something less than stellar.

	Noel

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From ram-bounces@iab.org Fri Sep 07 07:27:00 2007
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From: Iljitsch van Beijnum <iljitsch@muada.com>
Subject: Re: [RAM] Number of DFZ routers - radical improvement of BGP unlikely
Date: Fri, 7 Sep 2007 13:25:15 +0200
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On 7-sep-2007, at 3:47, Noel Chiappa wrote:

>> From: Iljitsch van Beijnum <iljitsch@muada.com>
>
>> The question is whether it would be easier to replace it. So far, I
>> can't think of anyone else other than myself speaking out in favor of
>> that, or even entertaining the question seriously. :-)

> Really? :-)

Dangers of caching: information from long ago may disappear unless  
it's refreshed periodically.  (-:

>> Obviously a new protocol would have to be able to interact with  
>> BGP and
>> be not much worse at talking BGP than a native speaker.

> Well, "interact with BGP" and "talk BGP" sound like two very  
> different things
> to me - unless by "talk BGP" you mean "emulate a BGP speaker".

Yes, they are different things: there is the protocol, which would be  
more or less independent from BGP, and the implementation, which  
would almost certainly need to be able to talk both the new protocol  
and talk to BGP speakers. Whether this counts as emulating... don't  
know.

> The thing is that if you have a system with a fundamentally  
> different model
> of the world (e.g. map-based, instead of route-table based), the  
> interface
> between it and BGP is necessarily going to be something of a  
> kludge, and any
> emulation is perforce going to be be something less than stellar.

At the very least a new protocol would have to support the situation  
where the new protocol is spoken in the middle and BGP both on the  
left and on the right. Obviously the BGP speakers may at that point  
also interact through regular BGP, e.g.:

BGP A -- New -- BGP B
       \       /
         BGP C

(Where "new" may be one or more ASes running the new protocol.) So  
the new protocol must be able to emulate BGP processing so well that  
it allows A and B to make (almost) identical decisions as in the case  
where there was no new protocol. For instance, AS paths must be at  
least retained and probably be added to as information travels  
through the new protocol. It gets even more interesting when two  
clouds speaking the new protocol are separated by a BGP cloud.

I don't think an internet-wide map based protocol is feasible, unless  
aggregation is so draconian that the protocol itself becomes largely  
irrelevant and operators will mostly be dealing with traffic  
engineering exceptions rather than the protocol itself. A system  
where there is a map of the "center" of the network (as seen from a  
given vantage point) where mapping to/from prefix tables happens at  
the edges of the map would be very interesting.

_______________________________________________
RAM mailing list
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From ram-bounces@iab.org Fri Sep 07 09:07:09 2007
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Date: Fri, 7 Sep 2007 09:06:44 EDT
Subject: Re: [RAM] Number of DFZ routers - radical improvement of BGP unlikely
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=20
I am pretty in line  with Iljitsch. I also think that it is =20
feasonable/reasonable to have the new design in the middle. At least for an=20=
 introduction=20
period.
When I would talk about 1-2 K FIB entries, I mean of course 1-2 K =20
links,mainly loose links, the more remote, the looser,  however precisest in=
 length. I=20
would also call the  database "map-based". But because this is also a routin=
g=20
table, the  alternative should rather be called "prefix-based".
=20
Same remarks about the current stage.
We have had a very long discussion about the dual nature id/loc.  We never=20
had a discussion about strict/loose links. I mean  algorithmically  determin=
ed=20
loose links.
We have adopted the term "stretch" but none of the current models =20
(LISP,lvip,..) has anything to do with it. It takes  map-based models, as  t=
o rate them=20
by stretch values, doesn't it ? (I would claim stretch=3D1 for my  map-based=
=20
concept of course :-)
=20
Another fundamental point of discussion is:=20
Shall a) the IP-header  serve the model or b) shall the model serve  the=20
IP-header ?
We have seen that a) is possible (TOS converted to DSCP). Why not again  her=
e=20
!?=20
=20
In summary, it is too early to start with the finish.
=20
Heiner =20
=20
In einer eMail vom 07.09.2007 13:27:10 Westeurop=E4ische Normalzeit schreibt=
 =20
iljitsch@muada.com:

On  7-sep-2007, at 3:47, Noel Chiappa wrote:

>> From: Iljitsch van  Beijnum <iljitsch@muada.com>
>
>> The question is whether  it would be easier to replace it. So far, I
>> can't think of anyone  else other than myself speaking out in favor of
>> that, or even  entertaining the question seriously. :-)

> Really?  :-)

Dangers of caching: information from long ago may disappear  unless =20
it's refreshed periodically.  (-:

>>  Obviously a new protocol would have to be able to interact with  =20
>> BGP and
>> be not much worse at talking BGP than a  native speaker.

> Well, "interact with BGP" and "talk BGP" sound  like two very =20
> different things
> to me - unless by "talk  BGP" you mean "emulate a BGP speaker".

Yes, they are different things:  there is the protocol, which would be =20
more or less independent from  BGP, and the implementation, which =20
would almost certainly need to be  able to talk both the new protocol =20
and talk to BGP speakers. Whether  this counts as emulating... don't =20
know.

> The thing is  that if you have a system with a fundamentally =20
> different  model
> of the world (e.g. map-based, instead of route-table based),  the =20
> interface
> between it and BGP is necessarily going  to be something of a =20
> kludge, and any
> emulation is  perforce going to be be something less than stellar.

At the very least  a new protocol would have to support the situation =20
where the new  protocol is spoken in the middle and BGP both on the =20
left and on the  right. Obviously the BGP speakers may at that point =20
also interact  through regular BGP, e.g.:

BGP A -- New -- BGP B
\       /
BGP C

(Where "new" may be one or more ASes running the new  protocol.) So =20
the new protocol must be able to emulate BGP  processing so well that =20
it allows A and B to make (almost) identical  decisions as in the case =20
where there was no new protocol. For  instance, AS paths must be at =20
least retained and probably be added  to as information travels =20
through the new protocol. It gets even  more interesting when two =20
clouds speaking the new protocol are  separated by a BGP cloud.

I don't think an internet-wide map based  protocol is feasible, unless =20
aggregation is so draconian that the  protocol itself becomes largely =20
irrelevant and operators will mostly  be dealing with traffic =20
engineering exceptions rather than the  protocol itself. A system =20
where there is a map of the "center" of  the network (as seen from a =20
given vantage point) where mapping  to/from prefix tables happens at =20
the edges of the map would be very  interesting.

_______________________________________________
RAM  mailing  list
RAM@iab.org
https://www1.ietf.org/mailman/listinfo/ram







  =20

-------------------------------1189170404
Content-Type: text/html; charset="ISO-8859-1"
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<HTML><HEAD>
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<BODY id=3Drole_body style=3D"FONT-SIZE: 10pt; COLOR: #000000; FONT-FAMILY:=20=
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bottomMargin=3D7 leftMargin=3D7 topMargin=3D7 rightMargin=3D7><FONT id=3Drol=
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face=3DArial color=3D#000000 size=3D2>
<DIV>
<DIV>I am&nbsp;pretty&nbsp;in line &nbsp;with Iljitsch. I also think that it=
 is=20
feasonable/reasonable to have the new design in the middle. At least for an=20
introduction period.</DIV>
<DIV>When I would talk about 1-2 K FIB entries, I mean of course 1-2 K=20
links,mainly loose links,&nbsp;the more remote, the looser,=20
however&nbsp;precisest in length.&nbsp;I would also call the=20
database&nbsp;"map-based". But because this is also a routing table, the=20
alternative should rather be called "prefix-based".</DIV>
<DIV>&nbsp;</DIV>
<DIV>Same remarks about the current stage.</DIV>
<DIV>We have&nbsp;had a very long discussion about the dual nature&nbsp;id/l=
oc.=20
We never had a discussion about strict/loose links. I mean&nbsp; algorithmic=
ally=20
determined loose links.</DIV>
<DIV>We have adopted the term "stretch" but none of the current models=20
(LISP,lvip,..) has anything to do with it. It takes&nbsp; map-based models,=20=
as=20
to rate them by stretch values, doesn't it ? (I would claim stretch=3D1 for=20=
my=20
map-based concept of course :-)</DIV>
<DIV>&nbsp;</DIV>
<DIV>Another fundamental point of discussion is: </DIV>
<DIV>Shall a) the IP-header &nbsp;serve the model or b) shall the model serv=
e=20
the IP-header ?</DIV>
<DIV>We have seen that a) is possible (TOS converted to DSCP). Why not again=
=20
here !? </DIV>
<DIV>&nbsp;</DIV>
<DIV>In summary, it is too early&nbsp;to start with the finish.</DIV>
<DIV>&nbsp;</DIV>
<DIV>Heiner&nbsp; </DIV>
<DIV>&nbsp;</DIV>
<DIV>In einer eMail vom 07.09.2007 13:27:10 Westeurop=E4ische Normalzeit sch=
reibt=20
iljitsch@muada.com:</DIV>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: blue 2px solid"><=
FONT=20
  style=3D"BACKGROUND-COLOR: transparent" face=3DArial color=3D#000000 size=
=3D2>On=20
  7-sep-2007, at 3:47, Noel Chiappa wrote:<BR><BR>&gt;&gt; From: Iljitsch va=
n=20
  Beijnum &lt;iljitsch@muada.com&gt;<BR>&gt;<BR>&gt;&gt; The question is whe=
ther=20
  it would be easier to replace it. So far, I<BR>&gt;&gt; can't think of any=
one=20
  else other than myself speaking out in favor of<BR>&gt;&gt; that, or even=20
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  :-)<BR><BR>Dangers of caching: information from long ago may disappear=20
  unless&nbsp; <BR>it's refreshed periodically.&nbsp; (-:<BR><BR>&gt;&gt;=20
  Obviously a new protocol would have to be able to interact with&nbsp;=20
  <BR>&gt;&gt; BGP and<BR>&gt;&gt; be not much worse at talking BGP than a=20
  native speaker.<BR><BR>&gt; Well, "interact with BGP" and "talk BGP" sound=
=20
  like two very&nbsp; <BR>&gt; different things<BR>&gt; to me - unless by "t=
alk=20
  BGP" you mean "emulate a BGP speaker".<BR><BR>Yes, they are different thin=
gs:=20
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rom=20
  BGP, and the implementation, which&nbsp; <BR>would almost certainly need t=
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  this counts as emulating... don't&nbsp; <BR>know.<BR><BR>&gt; The thing is=
=20
  that if you have a system with a fundamentally&nbsp; <BR>&gt; different=20
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=20
  the&nbsp; <BR>&gt; interface<BR>&gt; between it and BGP is necessarily goi=
ng=20
  to be something of a&nbsp; <BR>&gt; kludge, and any<BR>&gt; emulation is=20
  perforce going to be be something less than stellar.<BR><BR>At the very le=
ast=20
  a new protocol would have to support the situation&nbsp; <BR>where the new=
=20
  protocol is spoken in the middle and BGP both on the&nbsp; <BR>left and on=
 the=20
  right. Obviously the BGP speakers may at that point&nbsp; <BR>also interac=
t=20
  through regular BGP, e.g.:<BR><BR>BGP A -- New -- BGP B<BR>&nbsp; &nbsp;=20
  &nbsp;&nbsp; \&nbsp; &nbsp; &nbsp;&nbsp; /<BR>&nbsp; &nbsp; &nbsp;=20
  &nbsp;&nbsp; BGP C<BR><BR>(Where "new" may be one or more ASes running the=
 new=20
  protocol.) So&nbsp; <BR>the new protocol must be able to emulate BGP=20
  processing so well that&nbsp; <BR>it allows A and B to make (almost) ident=
ical=20
  decisions as in the case&nbsp; <BR>where there was no new protocol. For=20
  instance, AS paths must be at&nbsp; <BR>least retained and probably be add=
ed=20
  to as information travels&nbsp; <BR>through the new protocol. It gets even=
=20
  more interesting when two&nbsp; <BR>clouds speaking the new protocol are=20
  separated by a BGP cloud.<BR><BR>I don't think an internet-wide map based=20
  protocol is feasible, unless&nbsp; <BR>aggregation is so draconian that th=
e=20
  protocol itself becomes largely&nbsp; <BR>irrelevant and operators will mo=
stly=20
  be dealing with traffic&nbsp; <BR>engineering exceptions rather than the=20
  protocol itself. A system&nbsp; <BR>where there is a map of the "center" o=
f=20
  the network (as seen from a&nbsp; <BR>given vantage point) where mapping=20
  to/from prefix tables happens at&nbsp; <BR>the edges of the map would be v=
ery=20
  interesting.<BR><BR>_______________________________________________<BR>RAM=
=20
  mailing=20
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From ram-bounces@iab.org Fri Sep 07 09:12:27 2007
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From: Iljitsch van Beijnum <iljitsch@muada.com>
Subject: Re: [RAM] The mapping problem: rendezvous points?
Date: Fri, 7 Sep 2007 15:10:59 +0200
To: David Conrad <drc@virtualized.org>
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[sorry, replying to an even older message... but I think this is  
important]

On 9-mei-2007, at 20:07, David Conrad wrote:

> If an application works today, I'm having trouble understanding how  
> the introduction of additional O(10-100ms) latencies in the event  
> of a cache miss (which will likely only occur at the first packet  
> in a packet train) is going to have significant impact.  In those  
> odd cases that there is an impact, how much work would it be to  
> modify the application operation to be more tolerant of network  
> delays?

I'm assuming an encapsulation device close to the source host that  
serves more than a single host, and maybe even a great number of hosts.

If nothing much is happening and caches are empty, and a host then  
starts a session towards some remote destination, in a pull model,  
the encapsulating device must first look up a mapping so it can  
perform the required encapsulation. So the first packet must be  
dropped. At the start of a session, that would be the first TCP SYN  
(or equivalent packet for other transport protocols). At this point,  
it doesn't matter all that much how fast the mapping information  
becomes available, because the application has to wait for a  
retransmission of the first packet. Obviously it doesn't help if the  
mapping system is so slow that the second packet is also dropped. But  
I think 10 - 100 ms is extremely optimistic, DNS is about an order of  
magnitude slower than that in practice and it takes ~ 350 ms round  
trip across the world.

So the difference here is that unlike with the DNS, where you can  
proceed as soon as there is an answer, you have to time out here, and  
both doing that too fast and too slow increases latency.

But that's only when the sky is blue and the sun is shining. When an  
encapsulation device suddenly gets a lot of traffic that has been  
going for a while, so it's happening at high speeds, but it somehow  
doesn't have any mapping state (mapping state timed out, rerouting  
between encapsulation devices, maybe even a reboot), very many  
packets will be lost while a mapping lookup is done. Also, if a large  
number of flows require mapping requests, it's entirely possible that  
these must be rate limited, increasing delays even further.

I guess it would be possible for the encapsulation device to send a  
message back to the source host that indicates that the packet was  
dropped so it should be retransmitted. I'm guessing TCP will do that  
for ICMP "packet too big" messages today.

However, I would prefer "initial non-optimizedness" over "initial  
loss". If there is some place packets for destinations that aren't  
mapped yet can be sent to so they arrive anyway, even if this path is  
slower, this takes a lot of pressure off of the encapsulation  
devices, making the downsides of pull and caching less problematic:  
mapping requests can be done at the rate that best suit the  
encapsulation device rather than as fast as possible to avoid  
application impact.

I suppose there are some applications that will try to judge network  
characteristics based on the first few packets, and in such a  
situation, they would come to unnecessarily conservative conclusions.  
However, this could be addressed by changing these applications  
rather than restricting the internet architecture.

The issue of determining rendezvous locations that Marshall brought  
up is simple enough: have each rendezvous point handle a very long  
prefix, such as 96.0.0.0/6. This of course sucks if you have 97.0.0.1  
and you're in South Africa while the relevant rendezvous point for  
96.0.0.0/6 is in Japan, but since the detour only happens for the  
first few packets this shouldn't be a huge issue.

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From ram-bounces@iab.org Sun Sep 09 22:17:29 2007
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    > From: Iljitsch van Beijnum <iljitsch@muada.com>

    > If nothing much is happening and caches are empty, and a host then
    > starts a session towards some remote destination, in a pull model, the
    > encapsulating device must first look up a mapping so it can perform the
    > required encapsulation. So the first packet must be dropped.

Well, is that latter really necessary? I know it's more complex to hold onto
the packet until the mapping comes back, but if dropping that first packet
causes problems, we could write that code without changing anything else in
the system; i.e. it's an optimization we can add later, invisibly to the
rest of the system, if it turns out we need it.


    > If there is some place packets for destinations that aren't mapped yet
    > can be sent to so they arrive anyway, even if this path is slower, this
    > takes a lot of pressure off of the encapsulation devices,

Interesting idea. Let me ponder that...

    > The issue of determining rendezvous locations .. is simple enough: have
    > each rendezvous point handle a very long prefix, such as 96/6.

How about piggybacking the user's data packet on the request for the
EID->RLOC binding? When the request gets to whereever that binding is
available, it's sent on to the ETR, while the mapping is sent back to the
requestor? That avoids all the extra mechanism/configuration of rendezvous
points.

	Noel

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From ram-bounces@iab.org Sun Sep 09 22:26:24 2007
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    > From: Iljitsch van Beijnum <iljitsch@muada.com>

    > At the very least a new protocol would have to support the situation
    > where the new protocol is spoken in the middle and BGP both on the left
    > and on the right.

Right, and if the fundamental concepts of the new design are very different
from those of BGP (e.g. it exchages maps, rather than routing table entries),
that translation, forwards and backwards, could be rather tricky.

    > It gets even more interesting when two clouds speaking the new protocol
    > are separated by a BGP cloud.

Now you're either i) limiting the new system to having the capabilities of
BGP (in which case, you lose a lot of the point of doing something new), or
ii) you have to accept that two such discontiguous regions of New Stuff
simply don't get all the benefits of New Stuff until they somehow manage to
connect directly.


    > I don't think an internet-wide map based protocol is feasible

We will have to agree to differ... :-)

    > A system where there is a map of the "center" of the network (as seen
    > from a given vantage point) where mapping to/from prefix tables
    > happens at the edges of the map would be very interesting.

It happens that map-based system are ideally suited to building systems where
eveyone has a different view of the network...

	Noel

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From ram-bounces@iab.org Mon Sep 10 03:44:22 2007
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=20

-----Original Message-----
From: Noel Chiappa [mailto:jnc@mercury.lcs.mit.edu]=20
Sent: 10 September 2007 03:16
To: ram@iab.org
Cc: jnc@mercury.lcs.mit.edu
Subject: Re: [RAM] The mapping problem: rendezvous points?

    > From: Iljitsch van Beijnum <iljitsch@muada.com>

    > If nothing much is happening and caches are empty, and a host then
    > starts a session towards some remote destination, in a pull model,
the
    > encapsulating device must first look up a mapping so it can
perform the
    > required encapsulation. So the first packet must be dropped.

Well, is that latter really necessary? I know it's more complex to hold
onto the packet until the mapping comes back, but if dropping that first
packet causes problems, we could write that code without changing
anything else in the system; i.e. it's an optimization we can add later,
invisibly to the rest of the system, if it turns out we need it.


[alb] If the pull takes a while, someone could send lots of packets to
dud destinations, and if you hold all those packets waiting to find the
mapping you have DoS potential?

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=20
In einer eMail vom 10.09.2007 04:26:40 Westeurop=E4ische Normalzeit schreibt=
 =20
jnc@mercury.lcs.mit.edu:

> It gets even more interesting when two clouds speaking the new  protocol
> are separated by a BGP  cloud.




that would be l'art pour l'art, but makes no sense - IMHO
=20
Heiner



  =20

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; charset=3DISO-8859-1">
<META content=3D"MSHTML 6.00.2900.3157" name=3DGENERATOR></HEAD>
<BODY id=3Drole_body style=3D"FONT-SIZE: 10pt; COLOR: #000000; FONT-FAMILY:=20=
Arial"=20
bottomMargin=3D7 leftMargin=3D7 topMargin=3D7 rightMargin=3D7><FONT id=3Drol=
e_document=20
face=3DArial color=3D#000000 size=3D2>
<DIV>
<DIV>In einer eMail vom 10.09.2007 04:26:40 Westeurop=E4ische Normalzeit sch=
reibt=20
jnc@mercury.lcs.mit.edu:</DIV>
<BLOCKQUOTE=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: blue 2px solid"><=
FONT=20
  style=3D"BACKGROUND-COLOR: transparent" face=3DArial color=3D#000000 size=
=3D2>&nbsp;=20
  &nbsp; &gt; It gets even more interesting when two clouds speaking the new=
=20
  protocol<BR>&nbsp; &nbsp; &gt; are separated by a BGP=20
cloud.<BR><BR></FONT></BLOCKQUOTE></DIV>
<DIV></DIV>
<DIV>that would be l'art pour l'art, but makes no sense - IMHO</DIV>
<DIV>&nbsp;</DIV>
<DIV>Heiner</DIV></FONT>   </BODY></HTML>

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From: Iljitsch van Beijnum <iljitsch@muada.com>
Subject: Re: [RAM] Tunneling overheads and fragmentation
Date: Mon, 10 Sep 2007 15:03:50 +0200
To: Robin Whittle <rw@firstpr.com.au>
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[caught up, yay!]

On 21-jul-2007, at 16:40, Robin Whittle wrote:

> Even then, I fear that in order to preserve both reachability and
> efficiency (and any reachability problems which arise from
> fragmentation), that hosts in all networks, including non-upgraded
> networks, will need to adopt a somewhat lower MTU setting - for
> all the packets they send.

That would assume that there is something that doesn't support the  
"regular" MTU (in theory, there is no such thing, in practice: 1500  
bytes) between the place the packets are encapsulated and the place  
the packets are decapsulated.

I'm still operating under the assumption that both those places are  
in ISP networks. Now obviously there are lots of places in ISP  
networks that only support 1500-byte packets, but what would be a  
better decision here: push out a reduced packet size EVERYWHERE which  
we probably won't be able to raise any time soon, or require ISPs to  
either:

1. Implement path MTU discovery correctly, or:
2. Make sure that all encapsulated packets travel over paths that  
support at least 1500 byte + encapsulation sized packets

> I wonder to what extent every possible application respects the
> operating system's MTU.

The two are unrelated. Applications talk to transport protocols. The  
most popular one, TCP, breaks large chunks of data into smaller  
segments and coalesces multiple small chunks into larger segments.  
UDP simply adds its header and hands over the packet to the IP layer.  
The IP layer will fragment packets that are too large to be sent out  
over the interface of choice and/or the packet's destination. The  
only time there are problems (except of course from firewalls that  
don't like fragmentation) is when the transport protocol or  
application doesn't want fragmentation (DF=1) but the packet is  
larger than the interface MTU. Not sure what happens then, except  
that there is no way that a packet larger than the interface MTU is  
sent in one piece.

> I wonder if there are any widely
> used applications, such as games, P2P programs etc. which are
> hard-coded to assume a certain MTU which is close to, or right at,
> the limit of what can safely be sent across most of the Net.

Mostly video streaming, although that's all quickly moving to TCP  
these days. A common packet size here is 1450 bytes but I think  
that's excluding 28 bytes IPv4 + UDP so the packets are really 1478  
bytes.

I also remember seeing fragmented TCP traffic coming from DSL users.  
Obviously some link didn't do 1500 bytes in the middle but rather  
than use PMTUD the devices involved fragmented. I think DF wasn't set  
in this case, but it could have been cleared by a router or other box  
somewhere along the way. (I once implemented that myself when I got  
dial-up service delivered over a tunnel that only supported a 576- 
byte MTU.)

> Perhaps, in an optimistic scenario, recognising that some packets
> are to be encapsulated by ITRs, an ISP network could set up its
> DHCP system or whatever it is which gives DSL modems their
> parameters, to reduce the MTU and MSS settings sufficiently that
> the ITR-applied encapsulation still results in packets which will
> not be fragmented in most transit and border routers.

There are several MTU-related DHCP options but I'm not sure how  
widely they are supported. Note by the way that ATM which is used for  
both ADSL and DOCSIS (cable) supports MTUs much larger than 1500 bytes.

By the way, I'm currently working on this:

http://www.ietf.org/internet-drafts/draft-van-beijnum-multi-mtu-01.txt

It doesn't directly address this issue but it allows for systems with  
different MTUs to coexist on the same subnet so it becomes a lot  
easier to deploy jumboframes.

> This really needs to be done for all hosts in all networks - not
> just hosts in networks which have been upgraded with ITRs and ETRs
> etc.

Oh joy.

Doing path MTU discovery is probably easier, and note that if one  
host in a TCP session has a reduced MTU, it will let the other know  
during the three-way handshake so the unencumbered host won't send  
packets that are too large.

> Even if the host had its own ITFH
> (Ingress Tunnel Function in Host) function, I don't see how the
> operating system could tell application programs that there is one
> MTU and MSS setting for packets going to some addresses and
> another setting for packets going to other addresses.

Not a problem, path MTU discovery already does this today.

However, I see issues with hosts doing their own en/decapsulation:  
that way, the locators are exposed to hosts and they are at risk for  
becoming just as unrenumberable as IP addresses today.

The only way to make sure that you can renumber easily is if there  
are no firewall rules looking at locators. And the only way that will  
happen is if the relationship between location and identity is strong  
enough that spoofing it is not a viable attack vector. Concretely:  
today the routing system is unspoofable enough that people filter on  
IP addresses. The routing system is run by service providers who  
generally aren't in the business of attacking people. But if a  
locator mapping system is also open to end-users, it may be possible  
for attacks to use this path and people won't be happy to filter on  
just identifiers, just like they aren't happy to filter on just DNS  
names today. So either the ISPs must run it or there must be a heavy  
layer of magic security dust.

> UDP encapsulation, as used by LISP and I think eFIT-APT, involves
> 20 bytes for the IP header, 8 bytes for the UDP header and some
> number of bytes, such as 4, for extra stuff

What exactly was the reason for UDP encapsulation again? I think Dino  
said something about load balancing and firewalling. I'm not buying  
that: you can load balance on the destination IP address and anyone  
running firewalls in the middle of the routing system is best served  
with a single deny any any rule.

> This is where IPv6's long addresses and headers become really
> ugly. There would be 40 bytes for IP-in-IP and 52 for basic UDP
> encapsulation.

We really need bigger packets to offset the ever-increasing overhead.


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From ram-bounces@iab.org Mon Sep 10 09:15:43 2007
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From: Iljitsch van Beijnum <iljitsch@muada.com>
Subject: Re: [RAM] The mapping problem: rendezvous points?
Date: Mon, 10 Sep 2007 15:14:11 +0200
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On 10-sep-2007, at 4:16, Noel Chiappa wrote:

>> If nothing much is happening and caches are empty, and a host then
>> starts a session towards some remote destination, in a pull model,  
>> the
>> encapsulating device must first look up a mapping so it can  
>> perform the
>> required encapsulation. So the first packet must be dropped.

> Well, is that latter really necessary?

"Routers are not in the storage business."

But if they were, there would be a limit to what they can store, and  
I think it's likely this limit is smaller than the maximum number of  
packets that the router can receive in the maximum time it can take  
until the mapping process completes.

So I'd say that drops will happen, even if some storage is added as  
an optimization.

>> If there is some place packets for destinations that aren't mapped  
>> yet
>> can be sent to so they arrive anyway, even if this path is slower,  
>> this
>> takes a lot of pressure off of the encapsulation devices,

> Interesting idea. Let me ponder that...

>> The issue of determining rendezvous locations .. is simple enough:  
>> have
>> each rendezvous point handle a very long prefix, such as 96/6.

> How about piggybacking the user's data packet on the request for the
> EID->RLOC binding?

Wouldn't that be a specific implementation of the general idea of  
sending packets for which there is no mapping to a location that  
knows how to deliver them?

> When the request gets to whereever that binding is
> available, it's sent on to the ETR, while the mapping is sent back  
> to the
> requestor? That avoids all the extra mechanism/configuration of  
> rendezvous
> points.

It's a tradeoff between the complexity of having an additional system  
or the complexity of having multiple functions in one system. I'd say  
that it would be good to design things such that it's possible to do  
it in separate rendezvous points and then let the implementers figure  
out whether they want to do it that way or not.

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From: Dino Farinacci <dino@cisco.com>
Subject: Re: [RAM] The mapping problem: rendezvous points?
Date: Mon, 10 Sep 2007 13:07:51 -0700
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>> From: Iljitsch van Beijnum <iljitsch@muada.com>
>
>> If nothing much is happening and caches are empty, and a host then
>> starts a session towards some remote destination, in a pull model,
> the
>> encapsulating device must first look up a mapping so it can
> perform the
>> required encapsulation. So the first packet must be dropped.
>
> Well, is that latter really necessary? I know it's more complex to  
> hold
> onto the packet until the mapping comes back, but if dropping that  
> first
> packet causes problems, we could write that code without changing
> anything else in the system; i.e. it's an optimization we can add  
> later,
> invisibly to the rest of the system, if it turns out we need it.

This is the reason why the data-triggered based Map-Reply in LISP  
takes a packet that does not have a mapping cached and copies the  
inner destination address to the outer header's destination address  
field.

And hence LISP 1.5 allows no drop of packets if an EID-prefix  
namespace is routed on another topology where you only route on IDs  
(where the Internet topology we know of today routes only on locator  
prefixes).

Dino


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From ram-bounces@iab.org Mon Sep 10 19:07:29 2007
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Subject: Re: [RAM] The mapping problem: rendezvous points?
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    > From: Dino Farinacci <dino@cisco.com>

    > if an EID-prefix namespace is routed on another topology where you only
    > route on IDs (where the Internet topology we know of today routes only
    > on locator prefixes)

Err, minor terminological nit: one my definitions of "locator" is 'names used
by the path-selection system'. So if you have a system which is doing
"rout[ing]" (i.e. doing next-hop selection along the path), then the things
it is using as the name of the destination are, *by definition*, locators...
even if you call them EIDs!

	Noel

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Subject: Re: [RAM] The mapping problem: rendezvous points?
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>     > From: Dino Farinacci <dino@cisco.com>
>
>     > if an EID-prefix namespace is routed on another topology  
> where you only
>     > route on IDs (where the Internet topology we know of today  
> routes only
>     > on locator prefixes)
>
> Err, minor terminological nit: one my definitions of "locator" is  
> 'names used
> by the path-selection system'. So if you have a system which is doing
> "rout[ing]" (i.e. doing next-hop selection along the path), then  
> the things
> it is using as the name of the destination are, *by definition*,  
> locators...
> even if you call them EIDs!

I agree with your definition, but in LISP 1 and 1.5, when you copy  
the inner DA to the outer DA, we call the inner DA an EID because it  
is assigned to a host. And we use that EID as a locator in the IGP.  
When it is copied to the outer DA for map resolution, I guess it is  
also a locator outside of the site.

But the semantic of this outer DA, in lieu of map resolution comes  
out of the EID-prefix allocation space and not an ISPs space.

So I know this can be confusing or controversial, but when I look at  
an address, I call it from where it was *allocated* from and not how  
it is used.

Dino

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From ram-bounces@iab.org Tue Sep 11 15:42:59 2007
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Originally, Noel said:
> Err, minor terminological nit: one my definitions of "locator" is
> 'names used by the path-selection system'. So if you have a system
> which is doing "rout[ing]" (i.e. doing next-hop selection along the
> path), then the things it is using as the name of the destination are,
> *by definition*, locators... even if you call them EIDs!

Later on, Dino said:
% I agree with your definition, but in LISP 1 and 1.5, when you copy the
% inner DA to the outer DA, we call the inner DA an EID because it is
% assigned to a host. And we use that EID as a locator in the IGP.
% When it is copied to the outer DA for map resolution, I guess it
% is also a locator outside of the site.

The Internet community has a well understood name for objects that
are used sometimes as identifiers and sometimes for routing/forwarding.
Normally, these objects with mixed semantics are called "addresses",
not "EIDs" and not "Locators".

I really think that people would have an easier time understanding
the various proposals (not just LISP) if we really would move
to crisper terminology usage.

Cheers,

Ran


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From ram-bounces@iab.org Tue Sep 11 17:03:27 2007
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From: Iljitsch van Beijnum <iljitsch@muada.com>
Subject: Re: [RRG] Re: [RAM] Tunneling overheads and fragmentation
Date: Tue, 11 Sep 2007 23:01:44 +0200
To: Brian E Carpenter <brian.e.carpenter@gmail.com>
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[back on RAM because I believe that's where the LISP folks hang out  
and this is relevant for them, besides, people may think I'm a  
researcher...]

On 11-sep-2007, at 22:05, Brian E Carpenter wrote:

>>> I'm still operating under the assumption that both those places  
>>> are in
>>> ISP networks. Now obviously there are lots of places in ISP networks
>>> that only support 1500-byte packets,

> Can somebody provide evidence for this statement?

Minira#conf t
Enter configuration commands, one per line.  End with CNTL/Z.
Minira(config)#interface fastEthernet 1/0
Minira(config-if)#ip mtu ?
   <68-1500>  MTU (bytes)

Granted, this is a fairly old box in a small network, but I don't see  
anyone seriously claiming that ALL ISP networks support packets  
larger than 1500 bytes on ALL their internal links (and also on inter- 
ISP links).

That said, it may very well be feasible to engineer a "jack up"  
solution such that the encapsulated = larger packets only see parts  
of networks that support more than 1500 bytes with a level of effort  
that is within reason. (Let me put it this way: what's easier to  
support: a million prefix in the routing table or a 1600 byte MTU?)

>>> but what would be a better decision
>>> here: push out a reduced packet size EVERYWHERE which we probably  
>>> won't
>>> be able to raise any time soon, or require ISPs to either:

>>> 1. Implement path MTU discovery correctly, or:
>>> 2. Make sure that all encapsulated packets travel over paths that
>>> support at least 1500 byte + encapsulation sized packets

>> All these options look impossible or ugly to me.

> The first one has been eluding us for years, but the network
> still works. What's the evidence on actual deployment? (Also
> see below.)

Please note that this is about the network between *TRs, which is a  
much simpler universe than the host-to-host universe where weird  
OSes, NAT and firewalls get in the way. (Again, assuming that end- 
users don't get to run en/decapsulation boxes themselves.)

> The second one sounds like something that is in the ISPs'
> enlightened self interest, in which case it will happen.

Right. However, that would probably still be a deployment problem  
because we may have to wait for upgrades to happen. Elsewhere, I was  
more or less dragged into a discussion about tunnel MTU issues.  
Believe me when I tell you that  this is a minefield. However, the  
good thing with a LISP-like solution is that we get to design  
everything on all ends (ITR, ETR, mapping) so it's not too hard to  
implement fragmentation at the encapsulation layer. The issue with  
IPv4 encapsulation is that the ID space is too small to support  
decent packet rates. With something like LISP, we can implement an ID  
field there and severely tighten the reassembly window (maybe even go  
so far that fragments from one source must arrive in-order) so this  
works much better. The mapping system could distribute MTU/MRU  
information if that's helpful.

[...]

> Hence RFC 4821, which does need to get deployed.

RFC 4821 (PMTUD without ICMP) is great for TCP, but it's not  
reasonably implementable for most UDP-based protocols/applications.  
It also suffers from the problem that you don't know if your  
corresponent supports it so it requires a leap of faith that things  
will work out.

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Subject: Re: [RRG] Re: [RAM] Tunneling overheads and fragmentation
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> Granted, this is a fairly old box in a small network, but I don't  
> see anyone seriously claiming that ALL ISP networks support packets  
> larger than 1500 bytes on ALL their internal links (and also on  
> inter-ISP links).

I did a survey about a month ago and it is true. I will yield to  
those folks who responded to me to protect their privacy. ;-)

But the main gist was:

o We are going to 9K MTUs on all our internal links.
o Where we don't have 9K MTUs, we use 4470.
o Virtually no one runs ISP links at 1500.

Dino

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From: Iljitsch van Beijnum <iljitsch@muada.com>
Subject: Re: [RRG] Re: [RAM] Tunneling overheads and fragmentation
Date: Tue, 11 Sep 2007 23:49:55 +0200
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On 11-sep-2007, at 23:41, Dino Farinacci wrote:

>> Granted, this is a fairly old box in a small network, but I don't  
>> see anyone seriously claiming that ALL ISP networks support  
>> packets larger than 1500 bytes on ALL their internal links (and  
>> also on inter-ISP links).

> I did a survey about a month ago and it is true. I will yield to  
> those folks who responded to me to protect their privacy. ;-)

> But the main gist was:

> o We are going to 9K MTUs on all our internal links.
> o Where we don't have 9K MTUs, we use 4470.
> o Virtually no one runs ISP links at 1500.

That's good news, because then we can apparently treat anyone using  
1500 as an exception that can be dealt with operationally rather than  
something that must be addressed in the protocol.

But what about inter-ISP links? I'm assuming this isn't a big issue  
for high capacity private peering, but here in Europe a lot of  
peering happens over exchanges, which obviously use equipment that  
can easily handle larger packets, but so many people are on a big fat  
shared subnet that it's a given at someone will bring down the lowest  
common denominator to 1500.

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From: Dino Farinacci <dino@cisco.com>
Subject: Re: [RRG] Re: [RAM] Tunneling overheads and fragmentation
Date: Tue, 11 Sep 2007 14:53:50 -0700
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>> Granted, this is a fairly old box in a small network, but I don't  
>> see anyone seriously claiming that ALL ISP networks support  
>> packets larger than 1500 bytes on ALL their internal links (and  
>> also on inter-ISP links).
>
> I did a survey about a month ago and it is true. I will yield to  
> those folks who responded to me to protect their privacy. ;-)
>
> But the main gist was:
>
> o We are going to 9K MTUs on all our internal links.
> o Where we don't have 9K MTUs, we use 4470.
> o Virtually no one runs ISP links at 1500.

And hence why I put this paragraph (the second one below) in section  
5 of draft-farinacci-lisp-03.txt to reflect Brian Carpenter's comment  
on the subject:

-----

    Since additional tunnel headers are prepended, the packet becomes
    larger and in theory can exceed the MTU of any link traversed from
    the ITR to the ETR.  It is recommended, in IPv4 that packets do not
    get fragmented as they are encapsulated by the ITR.  Instead, the
    packet is dropped and an ICMP Too Big message is returned to the
    source.

    In practice, this is not really a problem.  Hosts typically do not
    originate IP packets larger than 1500 bytes.  And second, a survey
    has been taken (from a list of ISPs, see Acknowledgement section)
    where nearly all ISP link MTUs are either 4470 bytes or support
    Ethernet jumbo frames of 9000 bytes.  Therefore, we don't anticipate
    any problems with prepending additional headers.

-----

Dino

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From: Dino Farinacci <dino@cisco.com>
Subject: Re: [RRG] Re: [RAM] Tunneling overheads and fragmentation
Date: Tue, 11 Sep 2007 14:54:46 -0700
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> But what about inter-ISP links? I'm assuming this isn't a big issue  
> for high capacity private peering, but here in Europe a lot of  
> peering happens over exchanges, which obviously use equipment that  
> can easily handle larger packets, but so many people are on a big  
> fat shared subnet that it's a given at someone will bring down the  
> lowest common denominator to 1500.

All GigE and higher. We are okay.

Dino

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On Tue, 11 Sep 2007, Dino Farinacci wrote:
>> But what about inter-ISP links? I'm assuming this isn't a big issue for 
>> high capacity private peering, but here in Europe a lot of peering happens 
>> over exchanges, which obviously use equipment that can easily handle larger 
>> packets, but so many people are on a big fat shared subnet that it's a 
>> given at someone will bring down the lowest common denominator to 1500.
>
> All GigE and higher. We are okay.

While it may be technically OK, it's nowhere near operationally 
simple, whether or not you count that as "OK".

Many/most inter-ISP links are through GE-based IXes.  I believe most 
of those use a common VLAN for all the peers.  So in order to upgrade 
from MTU 1500, all the ISPs connecting to the VLAN need to update 
basically at once to MTU 9000 or big packets get blackholed to 
non-updated peers.  Another alternative is having a separate "big-MTU 
VLAN", but moving operators from one VLAN to another introduces IP 
address changes, resulting in BGP flaps and rather big operational 
cost.  I believe these big-MTU VLANs haven't been very popular to date 
due to lack of demand (= you'll need to connect to both VLANs in any 
case, and there's rather little benefit _to date_ of supporting 1500+ 
inter-ISP MTUs).

-- 
Pekka Savola                 "You each name yourselves king, yet the
Netcore Oy                    kingdom bleeds."
Systems. Networks. Security. -- George R.R. Martin: A Clash of Kings

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From ram-bounces@iab.org Wed Sep 12 10:25:00 2007
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Dino's notes all sound correct, with one exception.

The exception is a detail and does not alter his main point,
which is that by using Jumbo Ethernet, tunnelling
of 1518 byte Ethernet frames ought not create any
fragmentation issues.  However, just to be crystal clear
to anyone who isn't tracking Ethernet closely, here
is the summary analysis.


Correction:
	Just as "4K" MTU really means 4470 bytes, Jumbo Ethernet
	sizes of "9K" really mean an IP MTU of 9180 bytes.  This
	means the link MTU for Jumbo Ethernet is really:
            (9180 + sizeof(Ethernet header) + sizeof(VLAN tag))

	Implementers who want to be more widely deployable will
	have a "9K" link MTU that allows for at least 2 VLAN tags
	(considering the widespread use of Q-in-Q encapsulation)
	or 1 VLAN tag plus 2 Ethernet MAC headers (for MAC-in-MAC
	encapsulation).   Both Q-in-Q and MAC-in-MAC encapsulations
	are commonly used in Metro Service Provider deployments
	of Ethernet (e.g. in Japan, Korea, Sweden, and in the US).

History:
	The Jumbo Ethernet 9K frame size was demanded by Ethernet
	users who were transitioning away from ATM/AAL5.  So they
	wanted Jumbo Ethernet to support existing IP packet lengths
	for IP/ATM AAL5 [see RFC-1626] *without* any risk of IP
	fragmentation.

Caveats:
	Jumbo Ethernet is explicitly not conformant to IEEE 802.3
	standards.  So all implementations that I'm aware of have
	a default MTU (i.e. out of the box) that is IEEE 802.3
	compliant.  Normally, users must explicitly configure the
	larger MTU size on each interface before it is enabled.

	1 GigE standards permit deployment with CSMA/CD, although
	I am not personally aware of any such deployments.  A GigE
	CSMA/CD deployment is likely not to work properly with
	an Ethernet frame size above that permitted by IEEE 802.3
	standards.  Switched Ethernet should be fine, however,
	and all actual 1 GigE deployments that I am aware of are
	switched and do not use CSMA/CD.

Yours,

Ran
rja@extremenetworks.com




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On 12-sep-2007, at 16:24, RJ Atkinson wrote:

> Dino's notes all sound correct, with one exception.

I'd like to support Pekka's point about internet exchanges, though.

> The exception is a detail and does not alter his main point,
> which is that by using Jumbo Ethernet, tunnelling
> of 1518 byte Ethernet frames ought not create any
> fragmentation issues.  However, just to be crystal clear
> to anyone who isn't tracking Ethernet closely, here
> is the summary analysis.

Please note that ethernet frame size / MTU may or may not include the  
Ethernet_II header (which doesn't conform to IEEE 802.3) and if it  
does, it may or may not include the frame check sequence. I'll use  
the IP MTU excluding ethernet headers unless otherwise specified.

> Correction:
> 	Just as "4K" MTU really means 4470 bytes, Jumbo Ethernet
> 	sizes of "9K" really mean an IP MTU of 9180 bytes.  This
> 	means the link MTU for Jumbo Ethernet is really:
>            (9180 + sizeof(Ethernet header) + sizeof(VLAN tag))

I think most people take it to mean 9000 bytes. This is the only real  
jumboframe IP MTU size I've seen implemented in hosts. However,  
routers and especially switches come with many different maximum  
frame sizes. I've looked over some product literature and compiled  
the following list: 1508, 1530, 1536, 1546, 1998, 2000, 2018, 4464,  
4470, 8092, 8192, 9000, 9176, 9180, 9216, 17976, 64000 and 65280.

> Caveats:
> 	Jumbo Ethernet is explicitly not conformant to IEEE 802.3
> 	standards.  So all implementations that I'm aware of have
> 	a default MTU (i.e. out of the box) that is IEEE 802.3
> 	compliant.  Normally, users must explicitly configure the
> 	larger MTU size on each interface before it is enabled.

> 	1 GigE standards permit deployment with CSMA/CD, although
> 	I am not personally aware of any such deployments.  A GigE
> 	CSMA/CD deployment is likely not to work properly with
> 	an Ethernet frame size above that permitted by IEEE 802.3
> 	standards.  Switched Ethernet should be fine, however,
> 	and all actual 1 GigE deployments that I am aware of are
> 	switched and do not use CSMA/CD.

When gigabit ethernet was introduced the _minimum_ frame size (not IP  
MTU size) was (sort of) increased from 64 to 512 bytes when using  
CSMA/CD to make sure a frame would fill the entire collision domain  
even at the faster bit rate. However, the _maximum_ frame size was  
never increased because it would break old stuff. Lots of gigabit  
ethernet equipment supports larger frames, but few 10 or 100 Mbps  
implementations do. So having a larger frame size for GigE would make  
interoperation with older ethernet equipment a problem.

Although the issue comes up in the IEEE 802.3 working group every few  
years, no progress has been made here because the IEEE protocols  
simply don't have the hooks that make maximum frame size negotiation  
between ethernet systems possible. However, this is much easier at  
the IP level (especially IPv6), so I'm proposing that the IETF pick  
up the slack. See the discussion about my multi-mtu draft on the  
internet area list:

http://www1.ietf.org/mail-archive/web/int-area/current/index.html

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From: RJ Atkinson <rja@extremenetworks.com>
Subject: Re: [RAM] Tunnelling & GigE jumbo frame size
Date: Wed, 12 Sep 2007 13:42:58 -0400
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On  12 Sep 2007, at 12:42, Iljitsch van Beijnum wrote:
> On 12-sep-2007, at 16:24, RJ Atkinson wrote:
>> Correction:
>> 	Just as "4K" MTU really means 4470 bytes, Jumbo Ethernet
>> 	sizes of "9K" really mean an IP MTU of 9180 bytes.  This
>> 	means the link MTU for Jumbo Ethernet is really:
>>            (9180 + sizeof(Ethernet header) + sizeof(VLAN tag))
>
> I think most people take it to mean 9000 bytes.

I strongly disagree -- speaking as someone who spends a lot of time
talking with A LOT of different Ethernet end users around the world,
including users with deployments on all continents.  The origin
for the requirement came from super-computing sites with ATM
deployments, and they drove all the vendors in the same direction
(and vendors that didn't comply with their definition lost
potential business as a result; economics tends to win in such cases).

> This is the only real jumboframe IP MTU size I've seen implemented
> in hosts.

You should look around more broadly then.

I've seen the jumbo MTU that I outlined in multiple versions of UNIX
and in other kinds of systems.  Moreover, the jumbo MTU is primarily
deployed by service providers and super-computer centers, and is
not often used by ordinary end systems (e.g. one's home computer).

I'm not current with MTU support in current Windows, but earlier
versions of Windows did support more than 9000 bytes on Gigabit
Ethernet.

> However, routers and especially switches come with many different  
> maximum frame sizes. I've looked over some product literature and  
> compiled the following list: 1508, 1530, 1536, 1546, 1998, 2000,  
> 2018, 4464, 4470, 8092, 8192, 9000, 9176, 9180, 9216, 17976, 64000  
> and 65280.

As there is no standard, the maximum for a particular device will vary.
Most networking devices make this parameter *configurable*, below
some maximum value.  Varying maxima was not the question on the table,
however.

The question on the table is *narrowly* "what is meant by 9K MTU ?".
I stand by the contents of my original note.

Ran


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From: Iljitsch van Beijnum <iljitsch@muada.com>
Subject: Re: [RAM] Tunnelling & GigE jumbo frame size
Date: Wed, 12 Sep 2007 20:13:24 +0200
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On 12-sep-2007, at 19:42, RJ Atkinson wrote:

>>> 	Just as "4K" MTU really means 4470 bytes, Jumbo Ethernet
>>> 	sizes of "9K" really mean an IP MTU of 9180 bytes.  This
>>> 	means the link MTU for Jumbo Ethernet is really:
>>>            (9180 + sizeof(Ethernet header) + sizeof(VLAN tag))

>> I think most people take it to mean 9000 bytes.

> I strongly disagree [...]

>> This is the only real jumboframe IP MTU size I've seen implemented
>> in hosts.

> You should look around more broadly then.

(Oh wait: I once saw a 10 Gbit ethernet card in a couple of PowerMac  
G5s that did 16384 bytes.)

> I've seen the jumbo MTU that I outlined in multiple versions of UNIX
> and in other kinds of systems.  Moreover, the jumbo MTU is primarily
> deployed by service providers and super-computer centers, and is
> not often used by ordinary end systems (e.g. one's home computer).

Let me make several points at once. This is my computer at home, a  
Mac laptop:

$ sudo ifconfig en0 mtu 9180
Password:
$ ifconfig en0
en0: flags=8863<UP,BROADCAST,SMART,RUNNING,SIMPLEX,MULTICAST> mtu 1500
         supported media: autoselect 10baseT/UTP <half-duplex>  
10baseT/UTP <full-duplex> 10baseT/UTP <full-duplex,hw-loopback>  
10baseT/UTP <full-duplex,flow-control> 100baseTX <half-duplex>  
100baseTX <full-duplex> 100baseTX <full-duplex,hw-loopback> 100baseTX  
<full-duplex,flow-control> 1000baseT <full-duplex> 1000baseT <full- 
duplex,hw-loopback> 1000baseT <full-duplex,flow-control> none
$ sudo ifconfig en0 mtu 9000
$ ifconfig en0
en0: flags=8863<UP,BROADCAST,SMART,RUNNING,SIMPLEX,MULTICAST> mtu 9000
$ ifconfig lo0
lo0: flags=8049<UP,LOOPBACK,RUNNING,MULTICAST> mtu 16384

In other words: it can do 9000 bytes but not 9180 and the default MTU  
on the loopback interface is 16834 so obviously 9000 is not a maximum  
imposed by the OS. And apparantly, it can't do half duplex for  
gigabit ethernet.

>> However, routers and especially switches come with many different  
>> maximum frame sizes. I've looked over some product literature and  
>> compiled the following list: 1508, 1530, 1536, 1546, 1998, 2000,  
>> 2018, 4464, 4470, 8092, 8192, 9000, 9176, 9180, 9216, 17976, 64000  
>> and 65280.

> As there is no standard, the maximum for a particular device will  
> vary.
> Most networking devices make this parameter *configurable*, below
> some maximum value.  Varying maxima was not the question on the table,
> however.

> The question on the table is *narrowly* "what is meant by 9K MTU ?".
> I stand by the contents of my original note.

Well obviously we shouldn't be saying "9k MTU" because it is too  
imprecise. Also, it's irrelevant unless we're trying to come up with  
a unified MTU size, which is something I am very much against, as it  
doesn't solve the problem but only moves the goalposts. We need to  
move to a situation where the capability to use larger packets is  
used where it exists and to the degree that it exists.

If we can increase the average packet size used on the internet by  
transmitting the same amount of data in fewer packets and routers/ 
switches are built such that they mostly use power when they are  
actually doing work, this will help with the power/heat issue.

But for the purposes of a jack up solution we just need a few dozen  
extra bytes to accommodate the new headers without triggering the  
breakage that ensues when 1500-byte packets can't be forwarded  
without fragmentation.

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From: Dino Farinacci <dino@cisco.com>
Subject: Re: [RAM] Tunnelling & GigE jumbo frame size
Date: Wed, 12 Sep 2007 11:22:00 -0700
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> Well obviously we shouldn't be saying "9k MTU" because it is too  
> imprecise. Also, it's irrelevant unless we're trying to come up  
> with a unified MTU size, which is something I am very much against,  
> as it doesn't solve the problem but only moves the goalposts. We  
> need to move to a situation where the capability to use larger  
> packets is used where it exists and to the degree that it exists.

I will change the reference in the LISP spec from "jumbo frames of  
9000 bytes" to "jumbo frames of 9180 bytes". Will be reflected in the  
next ID update.

Dino

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From: RJ Atkinson <rja@extremenetworks.com>
Subject: Re: [RAM] Tunnelling & GigE jumbo frame size
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On  12 Sep 2007, at 14:13, Iljitsch van Beijnum wrote:
> Let me make several points at once. This is my computer at home,
> a Mac laptop:

(A bunch of irrelevant stuff deleted here.  What a Mac
can or can't do isn't representative of anything beyond Apple,
certainly not of the entire world.)

> We need to move to a situation where the capability to use larger
> packets is used where it exists and to the degree that it exists.

I know of a LOT of metro SPs and ISPs and IXs that already support
the 9180++ MTU that I outlined before -- in their operational network
right now.  I think NDAs do not permit me to enumerate the list
that I know about personally...it isn't a short list in any event.

> But for the purposes of a jack up solution we just need a few dozen  
> extra bytes to accommodate the new headers without triggering the  
> breakage that ensues when 1500-byte packets can't be forwarded  
> without fragmentation.

If you believe the above, why have you been wasting everyone's time
just now ?

Ran


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From: Iljitsch van Beijnum <iljitsch@muada.com>
Subject: Re: [RAM] Tunnelling & GigE jumbo frame size
Date: Wed, 12 Sep 2007 21:35:39 +0200
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On 12-sep-2007, at 20:34, RJ Atkinson wrote:

>> We need to move to a situation where the capability to use larger
>> packets is used where it exists and to the degree that it exists.

> I know of a LOT of metro SPs and ISPs and IXs that already support
> the 9180++ MTU that I outlined before

Yes, but the way IP currently works means that ALL equipment in an IP  
subnet must share the same MTU. This is not a problem in itself if  
you have a homogonous ISP network (but the fact that you need to  
enable the larger size by hand is even there) but that requirement  
makes it extremely hard to deploy larger packet sizes in many other  
environments.

>> But for the purposes of a jack up solution we just need a few  
>> dozen extra bytes to accommodate the new headers without  
>> triggering the breakage that ensues when 1500-byte packets can't  
>> be forwarded without fragmentation.

> If you believe the above, why have you been wasting everyone's time
> just now ?

Does anyone believe anything else??

I guess I should read the latest LISP draft as apparently there  
specific a jumboframe size is mentioned, and I have no idea for what  
purpose.

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From: Dino Farinacci <dino@cisco.com>
Subject: Re: [RAM] Tunnelling & GigE jumbo frame size
Date: Wed, 12 Sep 2007 12:47:58 -0700
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> I guess I should read the latest LISP draft as apparently there  
> specific a jumboframe size is mentioned, and I have no idea for  
> what purpose.

To summarize the issue we are trying to put to rest:

o All router-to-router links on the downstream side of an ITR use  
either 4470 or
   9180 byte MTUs.
o We don't care about the links inside a site because a packet is not  
LISP
   encapsulated there (in the typical case of a CE deployment of LISP).
o Hosts can originate 1500 byte sized packets.
o ITRs and TE-ITRs can prepend IPv4 and/or IPv6 headers without  
worrying about
   doing PMTU discovery or fragmentation.

So even when ~80 bytes of IPv6 is used for encapsulation, we have  
enough breathing room to avoid fragmentation (in the event we have 2  
headers, one for purposes of Loc/ID separation and the other for ISP- 
based TE).

Dino

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From ram-bounces@iab.org Wed Sep 12 16:03:45 2007
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Date: Wed, 12 Sep 2007 22:02:47 +0200
From: Gert Doering <gert@space.net>
To: Dino Farinacci <dino@cisco.com>
Subject: Re: [RAM] Tunnelling & GigE jumbo frame size
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Hi,

On Wed, Sep 12, 2007 at 12:47:58PM -0700, Dino Farinacci wrote:
> To summarize the issue we are trying to put to rest:
> 
> o All router-to-router links on the downstream side of an ITR use  
> either 4470 or 9180 byte MTUs.

Where is this assumption coming from?  As long as there are FastE links,
this is not going to happen.

Gert Doering
        -- NetMaster
-- 
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Joseph-Dollinger-Bogen 14          Aufsichtsratsvors.: A. Grundner-Culemann
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From ram-bounces@iab.org Wed Sep 12 16:07:49 2007
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From: RJ Atkinson <rja@extremenetworks.com>
Subject: Re: [RAM] Tunnelling & GigE jumbo frame size
Date: Wed, 12 Sep 2007 16:07:41 -0400
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On  12 Sep 2007, at 16:02, Gert Doering wrote:
> On Wed, Sep 12, 2007 at 12:47:58PM -0700, Dino Farinacci wrote:
>> To summarize the issue we are trying to put to rest:
>>
>> o All router-to-router links on the downstream side of an ITR use
>> either 4470 or 9180 byte MTUs.
>
> Where is this assumption coming from?
> As long as there are FastE links, this is not going to happen.

Are you trying to say that you currently have deployed
router-to-router links inside an ISP/IX with less than
4470 bytes of IP MTU ?

Btw, at least some 100baseTX/FX switches/routers can
support the 9180 IP MTU.  So it is not necessarily
the case that 100baseTX/FX implies a 1518 byte link MTU.

Ran


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Subject: Re: [RRG] Re: [RAM] Tunneling overheads and fragmentation
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Hi,

I seem to have missed that e-mail.  Sorry for replying not in sequence.

On Tue, Sep 11, 2007 at 02:53:50PM -0700, Dino Farinacci wrote:
> >I did a survey about a month ago and it is true. I will yield to  
> >those folks who responded to me to protect their privacy. ;-)

We haven't been asked in this survey - and our data looks quite different.

> >o We are going to 9K MTUs on all our internal links.
> >o Where we don't have 9K MTUs, we use 4470.
> >o Virtually no one runs ISP links at 1500.

In nearly all locations we have devices that still have Fast Ethernet 
links (because upgrading to GigE is expensive and the amount of
packes flowing through the link doesn't require it).

While we don't run 1500 on links where MPLS is used, most Cisco gear
in use cannot go over 1520...1530 on FastEthernet ports.  This limits
the GigE machines to 1530 as well, as things really break (today) if 
you share a layer2 segment between machines with different MTUs.

So there is no way our network could run on a MTU of 4470 or even 
higher any time soon.

[..]
>    In practice, this is not really a problem.  Hosts typically do not
>    originate IP packets larger than 1500 bytes.  And second, a survey
>    has been taken (from a list of ISPs, see Acknowledgement section)
>    where nearly all ISP link MTUs are either 4470 bytes or support
>    Ethernet jumbo frames of 9000 bytes.  Therefore, we don't anticipate
>    any problems with prepending additional headers.

This is handwaving, and not a good basis for research work.

Gert Doering
        -- NetMaster
-- 
Total number of prefixes smaller than registry allocations:  122119

SpaceNet AG                        Vorstand: Sebastian v. Bomhard
Joseph-Dollinger-Bogen 14          Aufsichtsratsvors.: A. Grundner-Culemann
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Date: Wed, 12 Sep 2007 22:11:15 +0200
From: Gert Doering <gert@space.net>
To: Dino Farinacci <dino@cisco.com>
Subject: Re: [RRG] Re: [RAM] Tunneling overheads and fragmentation
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Hi,

On Tue, Sep 11, 2007 at 02:54:46PM -0700, Dino Farinacci wrote:
> >But what about inter-ISP links? I'm assuming this isn't a big issue  
> >for high capacity private peering, but here in Europe a lot of  
> >peering happens over exchanges, which obviously use equipment that  
> >can easily handle larger packets, but so many people are on a big  
> >fat shared subnet that it's a given at someone will bring down the  
> >lowest common denominator to 1500.
> 
> All GigE and higher. We are okay.

You might be.  We're not.  

At the DECIX, currently the 3rd biggest exchange in Europe, about one 
third of the members (all sharing a common L2 network!) are still 
connected with 100 Mbit/s.

Which means "1500 for all of them".

Gert Doering
        -- NetMaster
-- 
Total number of prefixes smaller than registry allocations:  122119

SpaceNet AG                        Vorstand: Sebastian v. Bomhard
Joseph-Dollinger-Bogen 14          Aufsichtsratsvors.: A. Grundner-Culemann
D-80807 Muenchen                   HRB: 136055 (AG Muenchen)
Tel: +49 (89) 32356-444            USt-IdNr.: DE813185279

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From ram-bounces@iab.org Wed Sep 12 16:14:56 2007
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Date: Wed, 12 Sep 2007 22:14:52 +0200
From: Gert Doering <gert@space.net>
To: RJ Atkinson <rja@extremenetworks.com>
Subject: Re: [RAM] Tunnelling & GigE jumbo frame size
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Hi,

On Wed, Sep 12, 2007 at 04:07:41PM -0400, RJ Atkinson wrote:
> >>o All router-to-router links on the downstream side of an ITR use
> >>either 4470 or 9180 byte MTUs.
> >
> >Where is this assumption coming from?
> >As long as there are FastE links, this is not going to happen.
>=20
> Are you trying to say that you currently have deployed
> router-to-router links inside an ISP/IX with less than
> 4470 bytes of IP MTU ?

We have no single ethernet link in our network that has a MTU *above* 1530
(there are some that could be cranked up, but there was no need yet)

I have a large number of links that have a MTU of *exactly* 1500 (because
there is equipment that cannot handle 1530).

All exchange points that we're connected to run the fabric at 1500 byte
MTU, because they have members that have equipment that cannot handle
more.  There are *some* IXPs that have two different LANs, one with
1500 and one with "Jumbo", but that's not very widespread yet.

So: the answer is "yes".

> Btw, at least some 100baseTX/FX switches/routers can
> support the 9180 IP MTU.  So it is not necessarily
> the case that 100baseTX/FX implies a 1518 byte link MTU.

Switches are easy.  Routers are not.

What good is having a Jumbo-capable switch if there is one peer on the
shared fabric that can only receive 1500-byte (+ Ethernet header) packets?

Gert Doering
        -- NetMaster
--=20
Total number of prefixes smaller than registry allocations:  122119

SpaceNet AG                        Vorstand: Sebastian v. Bomhard
Joseph-Dollinger-Bogen 14          Aufsichtsratsvors.: A. Grundner-Culemann
D-80807 Muenchen                   HRB: 136055 (AG Muenchen)
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From: Dino Farinacci <dino@cisco.com>
Subject: Re: [RRG] Re: [RAM] Tunneling overheads and fragmentation
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> So there is no way our network could run on a MTU of 4470 or even
> higher any time soon.

Then you don't deploy ITRs or you deploy fragmenting-ITRs.

> [..]
>>    In practice, this is not really a problem.  Hosts typically do not
>>    originate IP packets larger than 1500 bytes.  And second, a survey
>>    has been taken (from a list of ISPs, see Acknowledgement section)
>>    where nearly all ISP link MTUs are either 4470 bytes or support
>>    Ethernet jumbo frames of 9000 bytes.  Therefore, we don't  
>> anticipate
>>    any problems with prepending additional headers.
>
> This is handwaving, and not a good basis for research work.

You are right, we are trying to make a decision and are doing  
engineering now past the research stage.   ;-)

Dino



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