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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Behavior Engineering for Hindrance Avoidance Working Group of the IETF.


	Title           : IP Multicast Requirements for a Network Address (and port) Translator (NAT)
	Author(s)       : D. Wing, T. Eckert
	Filename        : draft-ietf-behave-multicast-11.txt
	Pages           : 15
	Date            : 2007-10-04

This document specifies requirements for a Network Address (and port)
Translator (NAT) that supports Any Source IP Multicast or Source-
Specific IP Multicast.  An IP multicast-capable NAT device that
adheres to the requirements of this document can optimize the
operation of IP multicast applications that are generally unaware of
IP multicast NAT devices.

A URL for this Internet-Draft is:
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A new version of I-D, draft-ietf-behave-multicast-11.txt has been successfuly submitted by Dan Wing and posted to the IETF repository.

Filename:	 draft-ietf-behave-multicast
Revision:	 11
Title:		 IP Multicast Requirements for a Network Address (and port) Translator (NAT)
Creation_date:	 2007-10-04
WG ID:		 behave
Number_of_pages: 15

Abstract:
This document specifies requirements for a Network Address (and port)
Translator (NAT) that supports Any Source IP Multicast or Source-
Specific IP Multicast.  An IP multicast-capable NAT device that
adheres to the requirements of this document can optimize the
operation of IP multicast applications that are generally unaware of
IP multicast NAT devices.
                                                                                  


The IETF Secretariat.




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From behave-bounces@ietf.org Fri Oct 05 19:04:46 2007
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From: "Dan Wing" <dwing@cisco.com>
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Our milestones have been updated -- most of the changes are slipped dates.
The behavior-discovery document is now also Experimental, per the consensus
reached on the mailing list.

Document status is at http://www.employees.org/behave/document-status.html.
In summary:

Documents wrapping up:
  * draft-ietf-behave-multicast-11, in IESG processing
  * draft-ietf-behave-nat-icmp-05, submitted to IESG
  * draft-ietf-behave-p2p-state-04, submitted to IESG

Active documents:
  * draft-ietf-behave-rfc3489bis-10, awaiting revision based on AD comments
  * draft-ietf-behave-turn-04, waiting for architectural rewrite based on
    http://www1.ietf.org/mail-archive/web/behave/current/msg02742.html
  * draft-ietf-behave-turn-ipv6-03, waiting on TURN
  * draft-ietf-behave-nat-behavior-discovery-01, now Experimental


New milestone dates:

http://www.ietf.org/html.charters/behave-charter.html 

...
Done	  	Submit BCP that defines unicast UDP behavioral requirements
for NATs to IESG
Done	  	Submit a BCP that defines TCP behavioral requireents for NATs
to IESG
Done	  	Submit IPv6 relay protocol to IESG
Done	  	Submit a BCP that defines ICMP behavioral requirements for
NATs to IESG
Done	  	Submit informational that discusses current NAT traversal
techniques used by applications
Done	  	Submit BCP that defines multicast UDP
Oct 2007	  	Submit revision of RFC 3489 to IESG behavioral
requirements for NATs to IESG
Dec 2007	  	Submit standards-track IPv6 relay protocol
Dec 2007	  	Submit standards-track relay protocol
Jan 2008	  	Submit experimental document that describes how an
application can determine the type of NAT it is behind
Jan 2008	  	Close WG or recharter
...

-d


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From behave-bounces@ietf.org Mon Oct 08 11:10:11 2007
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Subject: [BEHAVE] BOF request under consideration: SAFE
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Hi,

We ADs have received a request for a BOF in Vancouver: SAFE -
Self-Address Fixing Evolution. See description below. I would appreciate
any feedback on this. For public discussion please use the SAFE mailing
list.

Post: safe@ietf.org
Subscribe: https://www1.ietf.org/mailman/listinfo/safe

Draft:
http://www.ietf.org/internet-drafts/draft-wing-behave-nat-control-stun-usage-03.txt


SAFE - Self-Address Fixing Evolution
------------------------------------

Chairs:
  TBD


ICE and its companion protocol STUN have been successfully deployed on
the Internet for NAT traversal.  ICE and STUN have several characteristics
which contribute to their success:

  1. incremental deployment.  ICE and STUN are functional without any
     modifications to existing NATs.
  2. nested NATs.  ICE and STUN work when there are multiple NATs
     between a host and the Internet.
  3. topology unaware.  ICE and STUN are not configured with
     information about NATs, firewalls, or their locations -- only
     with the IP address of a server on the Internet.
  4. simple security model.  If a host behind a NAT is allowed to send
     a packet across the NAT, it is allowed to receive a response.
  5. works on routed networks, which allows operation in both
     enterprise networks and home networks.

Other NAT traversal protocols do not share these characteristics,
which hinders their widespread deployment.  Specifically,

  * incremental deployment is not possible with MIDCOM, NSIS-NSLP,
    UPnP, or Bonjour.  With all of these protocols, both the NAT
    and the endpoint have to support the same protocol.
  * nested NATs are not possible with UPnP or Bonjour.
  * topology awareness is required of MIDCOM.
  * security must be established between the controlling entity
    and the NAT for MIDCOM and NSIS-NSLP.
  * Both UPnP and Bonjour use broadcast packets which don't work
    well on routed networks.

However, a drawback of ICE/STUN is its chatty keepalive traffic,
which is a result of STUN not knowing the binding lifetime of its
on-path NATs.  This chattiness causes a burden on servers and
consumes network bandwidth, which is especially critical on wireless
networks.  It is desirable to reduce this chattiness while still
retaining the important characteristics of STUN and ICE.

This BoF is intended to discuss one proposed technique,
draft-wing-behave-nat-control-stun-usage, which nearly eliminates
STUN's keepalive chatter and still preserves the desirable
characteristics of STUN/ICE.

The purpose of this BoF is to create a working group for this effort.


Agenda:
  Introduction, Agenda ....................................  5
  Summary of existing NAT traversal techniques ............ 40
   (UPnP, NAT-PMP/Bonjour, MIDCOM techniques, NSIS-NSLP, ICE)
  draft-wing-behave-nat-control-stun-usage ................ 40
  Q&A ..................................................... 25
                                                    ----------
                                                    total: 110

Cheers

Magnus Westerlund

IETF Transport Area Director & TSVWG Chair
----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM/M
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------


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Date: Mon, 08 Oct 2007 18:08:22 -0400
From: Jonathan Rosenberg <jdrosen@cisco.com>
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Subject: [BEHAVE] TURN status and direction
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Almost a month back, I sent a note to the behave list about a possible 
change in TURN to try to simplify things:

http://www1.ietf.org/mail-archive/web/behave/current/msg02742.html

the design team has been meeting 1-2 times a week since then, working 
through this proposal, and also working through the details on TCP and 
forking. Our original plan was to have a revised TURN available by 
October 1, but given the issues we were tackling, this has proven to be 
impossible. We're aiming for a draft now prior to the IETF.

So, until then, I wanted to summarize our current thinking on changes in 
TURN.

As in my previous note, we still think its a good idea to add a small 
shim to UDP that is always used. This would be a 1-byte shim, which 
signals a channel number. Channel 0 is reserved for TURN traffic between 
  the client and the TURN server. THe other channels are for data to a 
specific remote destination (IP address and port). Except for channel 0, 
the channel numbers in each direction are independent. The Send and Data 
indications remain, allowing the client and TURN server respectively to 
send TURN-encapsulated data in each direction. However, these messages 
will now carry a channel attribute that is used to establish a channel. 
So, for example, when the client begins sending data through the TURN 
server to a particular remote destination, it will use the Send 
indication. It picks an unallocated channel, and includes that in the 
Send indication. The TURN server will, in response to a Send indication, 
send an empty Data indication back to the client, echoing the channel 
number. It will also store a mapping that binds this channel to the 
remote destination (IP and port) in the Send. The empty Data indication 
allows the client to know that the channel allocation has succeeded. 
Once the client has received this empty Data indication, it knows it is 
free to send data on that allocated channel. The client can then send 
UDP packets for that remote destination by just sending a UDP packet to 
the TURN server with the 8-byte header including the allocated channel.

For data from the TURN server back to the client, it works just the 
same. The TURN server includes a channel number in its Data indications, 
and the client acknowledges this with an empty Send indication echoing 
the channel number back. When the TURN server receives data from a 
remote peer (assuming there is a permission), it sends it on the 
appropriate data channel if one has been set up, else sends it 
unencapsulated which has the side-effect of allocating a channel.

We debated many variations on this technique; the leading alternative 
was to use a separate transaction for channel allocation, rather than 
piggybacking on the data and send indications. This works less 
effectively with ICE (sends more packets and increases delay), and 
requires a STUN server side transaction model in clients, which is not 
required here. So we chose the approach above.

The second issue we debated extensively was in response to Matt Mathis' 
comments, about the case of two remote tcp connections being multiplexed 
into a single tcp connection from client to TURN server. This is 
something that WILL happen; there are several use cases for it. The most 
likely one is a call from an ICE agent to a non-ICE peer, where the 
media stream is actually something like MSRP, and the request forks to 
two remote agents. This case will cause the two remote agents to each 
open a tcp connection to the default candidate in the SDP - usually a 
TURN candidate. Typically only one of these connections survives in the 
long term; depending on which remote agent answers. But during the early 
media period both remain.

We worked through some approaches for causing multiple tcp connections 
to get formed from client to turn server when this happens. In the end, 
they were all really complicated. We felt it was better to simplify turn 
than add features to deal with this case. So instead our proposal is to 
make clear in the spec that, should this case happen (multiple tcp 
connections to remote agents), and more than a trivial amount of data is 
to be sent (and we'll quantify that), the agent MUST do a re-invite (or 
equivalent for non-SIP cases) to get a new turn candidate and thus 
separate connections for one of the remote peers. This is a purposeful 
limitation in TURN - its only meant to be used with a single remote 
correspondent at a time.

We are now beginning the process of integrating these changes into the 
TURN draft, while working through the other issues raised in the review 
period. Comments or questions on the above proposals are welcome.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From: Lars Eggert <lars.eggert@nokia.com>
Subject: Re: [BEHAVE] TURN status and direction
Date: Wed, 10 Oct 2007 09:03:42 -0500
To: ext Jonathan Rosenberg <jdrosen@cisco.com>
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Hi,

On 2007-10-8, at 17:08, ext Jonathan Rosenberg wrote:
> We worked through some approaches for causing multiple tcp  
> connections to get formed from client to turn server when this  
> happens. In the end, they were all really complicated.

these approaches are complicated because multiplexing different TCP  
connections onto a single one *is* a complex operation.

> We felt it was better to simplify turn than add features to deal  
> with this case. So instead our proposal is to make clear in the  
> spec that, should this case happen (multiple tcp connections to  
> remote agents), and more than a trivial amount of data is to be  
> sent (and we'll quantify that), the agent MUST do a re-invite (or  
> equivalent for non-SIP cases) to get a new turn candidate and thus  
> separate connections for one of the remote peers. This is a  
> purposeful limitation in TURN - its only meant to be used with a  
> single remote correspondent at a time.

I'm not sure I fully understand this proposal based on this short  
description, but I can see some reasons why this could be  
problematic. First, the TURN server cannot enforce how an application  
uses TCP. Second, the ability to re-invite may not be present in many  
TCP-using applications.

If TURN needs the ability to multiplex TCP connections, it needs to  
specify a mechanism that works for *all* uses of TCP, and not just a  
convenient subset.

Lars
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inline.

Lars Eggert wrote:

>> We felt it was better to simplify turn than add features to deal with 
>> this case. So instead our proposal is to make clear in the spec that, 
>> should this case happen (multiple tcp connections to remote agents), 
>> and more than a trivial amount of data is to be sent (and we'll 
>> quantify that), the agent MUST do a re-invite (or equivalent for 
>> non-SIP cases) to get a new turn candidate and thus separate 
>> connections for one of the remote peers. This is a purposeful 
>> limitation in TURN - its only meant to be used with a single remote 
>> correspondent at a time.
> 
> I'm not sure I fully understand this proposal based on this short 
> description, but I can see some reasons why this could be problematic. 
> First, the TURN server cannot enforce how an application uses TCP. 
> Second, the ability to re-invite may not be present in many TCP-using 
> applications.
> 
> If TURN needs the ability to multiplex TCP connections, it needs to 
> specify a mechanism that works for *all* uses of TCP, and not just a 
> convenient subset.

TURN has always assumed that there was some kind of rendezvous protocol 
working with it. The entire permission model is built off of the idea 
that a client knows the IP addresses of its remote peers, through that 
rendezvous mechanism.

So, what we have here is just another requirement on the rendezvous 
protocol. We require that, it either return a single address for a 
remote peer, or if it ends up with multiple, it have the ability to 
provide an updated IP address and port to the remote peer.

This requirement is readily met by SIP and also seems to be OK for other 
things you can think of that might use TURN. The design team couldn't 
think of cases where this limitation was a problem.

So I am reluctant to add functionality into the TURN layer when we can 
address the problem in the rendezvous protocol. Doing something in one 
place is better than two.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From behave-bounces@ietf.org Wed Oct 10 13:45:52 2007
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(Sorry I didn't pipe up last month when you sent the original message 
and asked for comments. My only excuse is that I wasn't paying attention.)

Jonathan Rosenberg wrote:
 > As in my previous note, we still think its a good idea to add a small
 > shim to UDP that is always used. This would be a 1-byte shim, which
 > signals a channel number.

As an implementor of a TURN server based on the previous drafts, I have 
a couple concerns with this change in the design that I didn't see 
addressed in this mail or your earlier message on the subject: Adding a 
single byte at the beginning of every relayed packet will 1) necessitate 
recalculation of the UDP checksum, as well as 2) potentially causing 
alignment problems on architectures that care about that (for relays 
that want to do something with the payload on its way through, eg 
collect statistics).

Have you considered 1) making the channel number byte a negotiated 
option, so that simplistic uses can enjoy lower overhead on the relay, 
and/or 2) using 2 bytes for the channel number when it is there, to make 
recomputing the checksum simpler (it could be adjusted rather than 
recomputed from scratch), or 4 bytes, to also alleviate alignment problems?

The current model of 0 modifications to the payload (once the active 
destination is set) certainly makes for a more efficient relay process 
in all of the cases where there is a single destination per allocation. 
Although I understand the need for adding the channel functionality (and 
cleaning out the "Set Active Destination" stuff), it would be sad to 
lose the current model's post-setup elegance even when the extra 
functionality isn't required.


 > The client can then send
 > UDP packets for that remote destination by just sending a UDP packet
 > to the TURN server with the 8-byte header including the allocated
 > channel.

I think I missed something here. Which 8 byte header? Did you mean 8 *bit*?



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Le Wednesday 10 October 2007 20:44:11 ext Laine Stump, vous avez =E9crit=A0:
> Have you considered 1) making the channel number byte a negotiated
> option, so that simplistic uses can enjoy lower overhead on the relay,
> and/or 2) using 2 bytes for the channel number when it is there, to make
> recomputing the checksum simpler (it could be adjusted rather than
> recomputed from scratch), or 4 bytes, to also alleviate alignment problem=
s?

I wouldn't mind using 2 bytes instead of 1 if that really matters. In any=20
case, MTU limits tend to be multiple of 8 bytes, so adding two rather than=
=20
one bytes will make no practical difference with regard to packets size.

As regard alignment, I think the most important thing is that, the=20
(mis)alignment MUST be always the same, so that it can be compensated easil=
y=20
(as with the Ethernet 14-bytes header), rather than have it retain some kin=
d=20
of alignment. In fact, if the payload is RTP, the needed alignment would be=
 4=20
bytes, but what if, as Lars pointed out, is not RTP? For the reference, som=
ey=20
IETF protocols use multiple of 8-bytes boundaries (IPv6 and ICMPv6 comes to=
=20
mind), and who knows what the requirement for the upper-layer protocol will=
=20
be??

> The current model of 0 modifications to the payload (once the active
> destination is set) certainly makes for a more efficient relay process
> in all of the cases where there is a single destination per allocation.

Bah, modern IP stacks (both in kernel- and user-land) support vectorized I/=
O=20
to work around this.

> Although I understand the need for adding the channel functionality (and
> cleaning out the "Set Active Destination" stuff), it would be sad to
> lose the current model's post-setup elegance even when the extra
> functionality isn't required.

I'd say the opposite. I'd rather have some unique and harmonized way of doi=
ng=20
things, that is simple to implement, easy to demultiplex and has predictibl=
e=20
alignment properties.

=2D-=20
R=E9mi Denis-Courmont


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From behave-bounces@ietf.org Thu Oct 11 07:53:41 2007
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Subject: Re: [BEHAVE] TURN status and direction
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-Original Message From: "R=E9mi Denis-Courmont"=20
<remi.denis-courmont@nokia.com>

> As regard alignment, I think the most important thing is that, the
> (mis)alignment MUST be always the same, so that it can be compensated=20
> easily
> (as with the Ethernet 14-bytes header), rather than have it retain some=
=20
> kind
> of alignment. In fact, if the payload is RTP, the needed alignment woul=
d=20
> be 4
> bytes, but what if, as Lars pointed out, is not RTP? For the reference,=
=20
> somey
> IETF protocols use multiple of 8-bytes boundaries (IPv6 and ICMPv6 come=
s=20
> to
> mind), and who knows what the requirement for the upper-layer protocol=20
> will
> be??

Isn't the issue more a case of what the various processors will accomodat=
e=20
as word alignment rather than what the protocols natural alignment is?

My understanding is that x86 is quite happy to do 32-bit word accesses th=
at=20
are not aligned to 32-word boundaries.  However, it will do two rather th=
an=20
one bus accesses to do it (sub-optimal, but not a disaster).  In the past=
,=20
processors such as PowePC and Sparc couldn't cope at all with doing=20
non-aligned accesses (more of a disaster!), although I think this situati=
on=20
is now improved.  Problems may occur on embedded processors such a ARM/AR=
C,=20
and DSP cores that has code optimized to exploit RTP's alignment to 32-bi=
t=20
boundaries.

To me, having a 4 byte header seems safest, and least likely to cause=20
problems.

> > The current model of 0 modifications to the payload (once the active
> > destination is set) certainly makes for a more efficient relay proces=
s
> > in all of the cases where there is a single destination per allocatio=
n.
>
> Bah, modern IP stacks (both in kernel- and user-land) support vectorize=
d=20
> I/O
> to work around this.

I've not heard of such stacks (or is it something like Windows I/O=20
completion ports?), but can they vectorize with non-aligned boundaries?

Pete.
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Pete Cordell
Codalogic
XML Schema to C++ data binding
 http://www.codalogic.com/
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D




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From: =?iso-8859-1?q?R=E9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
Organization: Nokia TP-SP-SWD
To: "ext Pete Cordell" <pete.cordell@tech-know-ware.com>
Subject: Re: [BEHAVE] TURN status and direction
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Le Thursday 11 October 2007 14:52:31 ext Pete Cordell, vous avez =E9crit=A0:
> Isn't the issue more a case of what the various processors will accomodate
> as word alignment rather than what the protocols natural alignment is?
(...)
> To me, having a 4 byte header seems safest, and least likely to cause
> problems.

It does not matter what the processors want if the protocol does not abide =
by=20
the processor wishes in the first place. If we have a non-RTP protocol that=
=20
includes 64-bits quantity (e.g. doing IPv6 tunneling through a TURN relay),=
=20
we'd need a 8-bytes alignment to make 64-bits processors happy/happier. Doe=
s=20
that mean the TURN header should be a multiple of 8 bytes?

What if I have a 128-bits processor and a protocol with 128-bits quantities=
?=20
TURN can be used with any upper-layer protocols... It seems a lot more safe=
=20
to have a *fixed-size* *always* *included* header. That's how Ethernet=20
drivers cope with the 14 bytes Etherne II header; receive the packet with a=
=20
two bytes initial offset from the "natural" boundary (14+2=3D16) so the net=
work=20
header is well-aligned. However, for this to work, the header size must be=
=20
known in advance.

> > Bah, modern IP stacks (both in kernel- and user-land) support vectorized
> > I/O to work around this.
>
> I've not heard of such stacks (or is it something like Windows I/O
> completion ports?), but can they vectorize with non-aligned boundaries?

I meant, scatter-gather I/O arrays: struct iovec with POSIX sendmsg/recvmsg=
,=20
or WSABUF with Windows WSASend/WSARecv.

=2D-=20
R=E9mi Denis-Courmont


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From behave-bounces@ietf.org Thu Oct 11 09:58:44 2007
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From: "Pete Cordell" <pete.cordell@tech-know-ware.com>
To: =?iso-8859-1?Q?R=E9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
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Subject: Re: [BEHAVE] TURN status and direction
Date: Thu, 11 Oct 2007 14:57:46 +0100
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- Original Message From: "R=E9mi Denis-Courmont"

> > Isn't the issue more a case of what the various processors will=20
> > accomodate
> > as word alignment rather than what the protocols natural alignment is=
?
> (...)
> > To me, having a 4 byte header seems safest, and least likely to cause
> > problems.
>
> It does not matter what the processors want if the protocol does not ab=
ide=20
> by
> the processor wishes in the first place. ...

If I wasn't convinced by the utility of the scatter-gather I/O you mentio=
n=20
below I would dispute that.  Fortunately I can save myself (and yourself)=
=20
the effort!

> ... It seems a lot more safe
> to have a *fixed-size* *always* *included* header.

I  agree that a fixed sized, always present (or at least, predictably=20
present) header is important.

> > > Bah, modern IP stacks (both in kernel- and user-land) support=20
> > > vectorized
> > > I/O to work around this.
> >
> > I've not heard of such stacks (or is it something like Windows I/O
> > completion ports?), but can they vectorize with non-aligned boundarie=
s?
>
> I meant, scatter-gather I/O arrays: struct iovec with POSIX=20
> sendmsg/recvmsg,
> or WSABUF with Windows WSASend/WSARecv.

So, on a 64-bit machine that's handling a 64-bit aligned protocol, I take=
 it=20
you mean doing something like:

    char stun_channel[1];    // Could be just a char, but hey...
    int64_t protocol_buffer[LOTS];

    struct iovec recv_iovec[2] =3D {
        { (caddr_t)stun_channel, 1 },
        { (caddr_t)protocol_buffer, sizeof(protocol_buffer) } };

    int res =3D recvmsg( ???, recv_iovec, 2, ??? );

    if( stun_channel[0] !=3D 0 )
        digest_protocol( protocol_buffer );

If there are no issues with this approach, then I don't have any alignmen=
t=20
concerns.

Cheers,

Pete.
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D
Pete Cordell
Codalogic
XML Schema to C++ data binding
 http://www.codalogic.com/
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D




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From behave-bounces@ietf.org Thu Oct 11 11:30:26 2007
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Subject: Re: [BEHAVE] TURN status and direction
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(A preface - our implementation tries to keep relayed packets in the 
kernel, and in the original buffers the ethernet driver put them in, as 
much as possible (for UDP->UDP anyway), so recvmsg() scatter-gather 
tricks won't help. On the other hand, for those who are copying every 
packet up to an application, even recomputing the checksum is a 
non-issue (you're doing it anyway), and you probably don't care about 
the other small differences in performance either (they're dwarfed in 
comparison to just copying from kernel to application, and context 
switching back and forth), which is why you likely think I'm 
nit-picking. At the application level, I have 0 problems with the 
proposed protocol changes, too.)

Rémi Denis-Courmont wrote:
> What if I have a 128-bits processor and a protocol with 128-bits quantities? 
> TURN can be used with any upper-layer protocols... It seems a lot more safe 
> to have a *fixed-size* *always* *included* header.

Your point about data other than RTP having broader alignment 
requirements is well taken (of course, I'd wager that the great majority 
  of alignment needs would be met by 32 bits, a much higher percentage 
than with 1 byte headers anyway).

But alignment for the sake of the hardware architecture is secondary for 
me. Since I'm working on an x86 architecture, I'm (selfishly) more 
concerned about spending the time to redo the entire checksum rather 
than just adjusting it* or leaving it alone (hardware assist makes that 
a moot point, but isn't always available), and moving the bytes around 
(or allocating auxiliary buffers) rather than not.


* It's a simple matter to adjust the checksum of data that has had an 
even number of bytes prepended (or any number of bytes appended), but 
I'm not aware of a simple algorithm to adjust the checksum when an odd 
number of bytes has been prepended - instead the entire checksum must be 
recomputed.

> That's how Ethernet 
> drivers cope with the 14 bytes Ethernet II header; receive the packet with a 
> two bytes initial offset from the "natural" boundary (14+2=16) so the network 
> header is well-aligned. However, for this to work, the header size must be 
> known in advance.

At the level that matters to those of us trying to do an in-kernel, 
0-copy implementation (ie the ethernet driver) even if you make the 
assumption that every packet will be IPv4, you still can't make the 
assumption that every packet will be a TURN "1-byte-channel + data" 
encapsulated packet (since other traffic for other protocols can be 
arrving on the same interface). Unless you dedicate an interface to 
receiving only the encapsulated TURN data, your stated goal of "fixed 
size, always included header" can't be met at this level, and you're 
doomed to either reduced performance due to non-aligned memory accesses, 
or copying the data to align it. (Again, alignment is only a problem 
when the relay wants to do something with the data, rather than just 
blindly passing it along, so it's not as important as the checksum 
problem, or the mere presence of any amount of extra data).

As far as predictability is concerned - since any processing taking 
place will have to be based on the address:port tuple flows anyway, as 
long as the format of the packet is predictable/identical for all 
packets in the same flow, code paths for the different formats 
(encapsulated, unencapsulated) can be made available and the code can be 
just as streamlined for each possibility. There would of course be some 
extra code to write, and a bit of setup at the beginning of the session, 
but that happens once, compared to thousands of relayed packets per session.

>>> Bah, modern IP stacks (both in kernel- and user-land) support vectorized
>>> I/O to work around this.
>> I've not heard of such stacks (or is it something like Windows I/O
>> completion ports?), but can they vectorize with non-aligned boundaries?
> 
> I meant, scatter-gather I/O arrays: struct iovec with POSIX sendmsg/recvmsg, 
> or WSABUF with Windows WSASend/WSARecv.

Even in the kernel, when adding the encapsulation header at write time, 
yes - this works reasonably well, but different semantics (eg, a list of 
packet buffers) may require that you spend the time to acquire a second 
buffer, initialize it, attach it to the list, and free it once you're 
finished (actually, if the buffer has some headroom/tailroom built in, 
it may be more efficient to just move the ethernet/IP/UDP headers down 
by one byte, or the payload up by one byte and pop the channel in the 
middle). When reading encapsulated packets, I don't see how it could be 
any help at all (as described above).

I know that each of these things (recomputing the entire checksum, 
allocating and freeing a 2nd buffer to add in the encapsulation byte, 
accessing large words at unaligned addresses) doesn't take much by 
itself, but added together and repeated for every single media (or 
whatever) packet they can make a difference, especially when there's so 
little else that needs to happen to get the packet through. I'd rather 
pay a bit extra up-front in development time and session setup time.



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From behave-bounces@ietf.org Thu Oct 11 19:06:16 2007
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Subject: Re: [BEHAVE] TURN status and direction
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On Oct 10, 2007, at 10:44 AM, Laine Stump wrote:
> (Sorry I didn't pipe up last month when you sent the original  
> message and asked for comments. My only excuse is that I wasn't  
> paying attention.)
>
> Jonathan Rosenberg wrote:
> > As in my previous note, we still think its a good idea to add a  
> small
> > shim to UDP that is always used. This would be a 1-byte shim, which
> > signals a channel number.
>
> As an implementor of a TURN server based on the previous drafts, I  
> have a couple concerns with this change in the design that I didn't  
> see addressed in this mail or your earlier message on the subject:  
> Adding a single byte at the beginning of every relayed packet will  
> 1) necessitate recalculation of the UDP checksum, as well as 2)  
> potentially causing alignment problems on architectures that care  
> about that (for relays that want to do something with the payload  
> on its way through, eg collect statistics).
>
> Have you considered 1) making the channel number byte a negotiated  
> option, so that simplistic uses can enjoy lower overhead on the  
> relay, and/or 2) using 2 bytes for the channel number when it is  
> there, to make recomputing the checksum simpler (it could be  
> adjusted rather than recomputed from scratch), or 4 bytes, to also  
> alleviate alignment problems?

I'm not following your reasoning for making the channel number being  
a negotiated option.  The TURN server has to re-compute the UDP  
checksum on relayed data because the address 4-tuple in the IP header  
changes, right?

I don't see the point of going to a 2 byte channel number tho'.   
Either the TURN server is relaying the packet or processing it  
locally (channel=0).  In the former case (which is the common case),  
alignment doesn't matter but bumping up against MTU limits does.  In  
the later case, alignment may be required (or at least desired), but  
these events don't occur often and so an extra memcpy for the  
purposes of alignment is a small price to pay for optimizing the  
common case.



>
> The current model of 0 modifications to the payload (once the  
> active destination is set) certainly makes for a more efficient  
> relay process in all of the cases where there is a single  
> destination per allocation. Although I understand the need for  
> adding the channel functionality (and cleaning out the "Set Active  
> Destination" stuff), it would be sad to lose the current model's  
> post-setup elegance even when the extra functionality isn't required.
>
>
> > The client can then send
> > UDP packets for that remote destination by just sending a UDP packet
> > to the TURN server with the 8-byte header including the allocated
> > channel.
>
> I think I missed something here. Which 8 byte header? Did you mean  
> 8 *bit*?
>
>
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave

-brian
briank@briank.com





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Thanks for the review Magnus. I've updated the document based on yor 
comments. Responses below.

Magnus Westerlund wrote:
> Hi,
> 
> I have performed a combined AD-review / shepherding review and have some
> comments. I think some of these will require an updated version before
> proceeding. I hope this can be accomplished quickly to avoid to much delay.
> 
> 
> 1. Section 7.2.1:
> It is not obvious why the number 7900 ms is the end of listening for
> responses. How did one arrive on that number?

Well, its basically the same value we had from rfc3489. But its a 
reasonable number. The final request is sent at 6300ms, so this gives us 
1.6 seconds to wait for a response to that before giving up. In case 
this last packet does get through, we want to wait at least as long as a 
big RTT; 1.6 seconds provides lots of margin.

I added the note in parens to the following text:

    <t>Retransmissions continue until a response is received, or until
    a total of 7 requests have been sent.  If, after the last request,
    a duration equal to 16 times the RTO has passed without a response
    (providing ample time to get a response if only this final request
    actually succeeds),

> 
> 2. Section 12.2:
> "   A STUN server can detect when a given Binding Request message was
>    sent from an RFC 3489 [RFC3489] client by the absence of the correct
>    value in the Magic Cookie field.  When the server detects an RFC 3489
>    client, it SHOULD copy the value seen in the Magic Cookie field in
>    the Binding Request to the Magic Cookie field in the Binding Response
>    message, and insert a MAPPED-ADDRESS attribute instead of an XOR-
>    MAPPED-ADDRESS attribute."
> 
> I think this text should have some warning about what functionality
> issues regarding MUXing STUN with other protocols may have due to this
> backwards compatibility issue. If I understand things correctly, having
> 3489 compatibility means that demuxer only can use the two first bits
> and two last of the first word to separate STUN from normal data packets?

Per the rest of this thread, I added your proposed text verbatim to this 
section.

> 
> 3. Section 14.4:
> 
> "   The key used as input to HMAC is the password."
> 
> Section 10.2:
> 
>    "For STUN servers used in conjunction with SIP servers, it is
>    desirable to use the same credentials for authentication to the SIP
>    server and STUN server.  Typically, SIP systems utilizing SIP's
>    digest authentication mechanism do not actually store the password in
>    the database.  Rather, they store a value called H(A1), which is
>    computed as:
> 
>                 H(A1) = MD5(username ":" realm ":" password)
> 
>    If a system wishes to utilize this credential, the STUN password
>    would be computed by taking the user-entered username and password,
>    and using H(A1) as the STUN password.  It is RECOMMENDED that clients
>    utilize this construction for the STUN password."
> 
> Isn't these two in contradiction. In addition aren't they a potential
> for interoperability issues between client and server. This as the
> actual key used in the HMAC must be the same to allow correct integrity
> verification. So, how is it inteded to work? Is it something the usage
> needs to define?

I had intended that it each deployment could make a decision on this one 
way or another. But upon further consideration I agree this is a recipe 
for interop problems. Even if its not SIP there is no reason why you 
cannot use the H(A1) as the stun password. I've aligned the text around 
this, and moved the rules on computing the HMAC using H(A1) into the 
section on hmac computation itself, so that it is mandatory.


> 
> 4. Section 15.1.2:
> 
> "  A rogue client may use a STUN server as a reflector, sending it
>    requests with a falsified source IP address and port.  In such a
>    case, the response would be delivered to that source IP and port.
>    There is no amplification with this attack, and it is mitigated by
>    ingress source address filtering."
> 
> I think there is missing a clarification on what type of amplication
> that is not possible with STUN. To my understanding there can't be any
> packet rate amplification in this attack. However some percent of packet
> size amplification seems possible due to NONCE and other error
> indication added by the server. Maybe a clarification that there is no
> amplification in the number of packets is warrented.

Text now reads:

	  <t>
	    A rogue client may use a STUN server as a reflector,
	    sending it requests with a falsified source IP address and
	    port. In such a case, the response would be delivered to
	    that source IP and port. There is no amplification of the
	    number of packets with this attack (the STUN server sends
	    one packet for each packet sent by the client), though
	    there is a small increase in the amount of data, since
	    STUN responses are typically larger than requests. This
	    attack is mitigated by ingress source address filtering.  </t>


> 
> 5. Section 15.2:
> 
> "   Most of the attacks in this section revolve around an attacker
>    modifying the reflexive address learned by a STUN client through a
>    Binding Request/Binding Response transaction.  Since the usage of the
>    reflexive address is a function of the usage, the applicability and
>    remediation of these attacks is usage-specific.  In common
>    situations, modification of the reflexive address by an on-path
>    attacker is easy to do.  Consider, for example, the common situation
>    where STUN is run directly over UDP.  In this case, an on-path
>    attacker can modify the source IP address of the Binding Request
>    before it arrives at the STUN server.  The STUN server will then
>    return this IP address in the XOR-MAPPED-ADDRESS attribute to the
>    client.  Protecting against this attack by using a message-integrity
>    check is impossible, since a message-integrity value cannot cover the
>    source IP address, since the intervening NAT must be able to modify
>    this value.  Instead, one solution to preventing the attacks listed
>    below is for the client to verify the reflexive address learned, as
>    is done in ICE [I-D.ietf-mmusic-ice].  Other usages may use other
>    means to prevent these attacks."
> 
> I think there are a serious thinking error that affects most of the
> attack variants in this description. If on on-path attacker changes the
> source IP-address of the packet the packet will not be routed back to
> the client issuing the request. Thus they are looking like lost packets.
> To me the only attack being really viable is if the attacker changes the
> source address to itself and then forwards the packet to the client
> (target of the attack). This is of course assuming that the attacker is
> not sitting very closely to the server and can see all/most traffic to
> and from the server?

Actually thats exactly the assumption; you can rewrite the source IP to 
point to anyone that you'd naturally be on-path for a message from the 
STUN server to that IP. As you say, this is more valuable when the 
attacker is closest to the server, less so as it gets closer to the 
client. I could swear there was text on this somewhere at some point but 
its clearly not here now. I will mention this and then discuss how that 
limitation applies more in 15.2.1 and 15.2.2.


> 
> Can we be a bit clearer on when the attacks are possible. Thus I think
> that 15.2.1 and 15.2.2 are more difficult to achieve (assuming integrity
> checks between client and server) than 15.2.3 and 15.2.4.
> 
> 6. Section 15.3:
> 
> "   This specification uses SHA-1 for computation of the message
>    integrity. "
> 
> Shouldn't there be a HMAC before SHA-1 here?

Fixed.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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A new version of I-D, draft-ietf-behave-rfc3489bis-11.txt has been successfuly submitted by Jonathan Rosenberg and posted to the IETF repository.

Filename:	 draft-ietf-behave-rfc3489bis
Revision:	 11
Title:		 Session Traversal Utilities for (NAT) (STUN)
Creation_date:	 2007-10-11
WG ID:		 behave
Number_of_pages: 44

Abstract:
Session Traversal Utilities for NAT (STUN) is a protocol that serves
as a tool for other protocols in dealing with NAT traversal.  It can
be used by an endpoint to determine the IP address and port allocated
to it by a NAT.  It can also be used to check connectivity between
two endpoints, and as a keep-alive protocol to maintain NAT bindings.
STUN works with many existing NATs, and does not require any special
behavior from them.

STUN is not a NAT traversal solution by itself.  Rather, it is a tool
to be used in the context of a NAT traversal solution.  This is an
important change from the previous version of this specification (RFC
3489), which presented STUN as a complete solution.

This document obsoletes RFC 3489.
                                                                                  


The IETF Secretariat.




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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Behavior Engineering for Hindrance Avoidance Working Group of the IETF.


	Title           : Session Traversal Utilities for (NAT) (STUN)
	Author(s)       : J. Rosenberg, et al.
	Filename        : draft-ietf-behave-rfc3489bis-11.txt
	Pages           : 44
	Date            : 2007-10-11

Session Traversal Utilities for NAT (STUN) is a protocol that serves
as a tool for other protocols in dealing with NAT traversal.  It can
be used by an endpoint to determine the IP address and port allocated
to it by a NAT.  It can also be used to check connectivity between
two endpoints, and as a keep-alive protocol to maintain NAT bindings.
STUN works with many existing NATs, and does not require any special
behavior from them.

STUN is not a NAT traversal solution by itself.  Rather, it is a tool
to be used in the context of a NAT traversal solution.  This is an
important change from the previous version of this specification (RFC
3489), which presented STUN as a complete solution.

This document obsoletes RFC 3489.

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From: =?iso-8859-1?q?R=E9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
Organization: Nokia TP-SP-SWD
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	Laine,

As I already noted, real-life MTUs tend to be multiple of 8, and are pretty=
=20
much always multiple of 4 bytes. As such, I have no problem using a two byt=
e=20
header, or even 4 bytes, if it is deemed that NAT-like O(1) IP&UDP checksum=
=20
adjustment is a nice thing to have.

In fact, one big advantage with the 4 bytes header may be to harmonize the =
UDP=20
and TCP framing of TURN, i.e. two bytes channel and two bytes length. That=
=20
being noted, I am not an author to the TURN I-D, nor am I attending the DT.

=2D-=20
R=E9mi Denis-Courmont


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From: =?iso-8859-1?q?R=E9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
Organization: Nokia TP-SP-SWD
To: behave@ietf.org
Subject: Re: [BEHAVE] TURN status and direction
Date: Fri, 12 Oct 2007 09:55:03 +0300
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Le Friday 12 October 2007 02:05:04 ext Brian Korver, vous avez =E9crit=A0:
> I don't see the point of going to a 2 byte channel number tho'.
> Either the TURN server is relaying the packet or processing it
> locally (channel=3D0).  In the former case (which is the common case),
> alignment doesn't matter but bumping up against MTU limits does.  In
> the later case, alignment may be required (or at least desired), but
> these events don't occur often and so an extra memcpy for the
> purposes of alignment is a small price to pay for optimizing the
> common case.

The point of 2 bytes rather than 1 byte channel ID is that you can "fix" th=
e=20
outer UDP checksum with a single substraction. Many NATs device "fix" the U=
DP=20
checksum by simply substracting and adding the modified port numbers, rathe=
r=20
than checksumming the entire UDP payload.

Of course, this type of optimization is only valid for raw (in-kernel)=20
implementations, where the checksum field can be accessed.

=2D-=20
R=E9mi Denis-Courmont


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The IESG has received a request from the Behavior Engineering for 
Hindrance Avoidance WG (behave) to consider the following document:

- 'Session Traversal Utilities for (NAT) (STUN) '
   <draft-ietf-behave-rfc3489bis-11.txt> as a Proposed Standard

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The file can be obtained via
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From: Brian Korver <briank@briank.com>
Subject: Re: [BEHAVE] TURN status and direction
Date: Fri, 12 Oct 2007 11:27:46 -0700
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On Oct 11, 2007, at 11:55 PM, R=E9mi Denis-Courmont wrote
> Le Friday 12 October 2007 02:05:04 ext Brian Korver, vous avez =E9crit =
:
>> I don't see the point of going to a 2 byte channel number tho'.
>> Either the TURN server is relaying the packet or processing it
>> locally (channel=3D0).  In the former case (which is the common =
case),
>> alignment doesn't matter but bumping up against MTU limits does.  In
>> the later case, alignment may be required (or at least desired), but
>> these events don't occur often and so an extra memcpy for the
>> purposes of alignment is a small price to pay for optimizing the
>> common case.
>
> The point of 2 bytes rather than 1 byte channel ID is that you can =20
> "fix" the
> outer UDP checksum with a single substraction. Many NATs device =20
> "fix" the UDP
> checksum by simply substracting and adding the modified port =20
> numbers, rather
> than checksumming the entire UDP payload.
>
> Of course, this type of optimization is only valid for raw (in-kernel)
> implementations, where the checksum field can be accessed.

We're optimizing for in-kernel NAT implementations?  Hmm, I still
don't see the real point to the 2 byte channel number.


>
> --=20
> R=E9mi Denis-Courmont
>
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave

-brian
briank@briank.com





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From: Brian Korver <briank@briank.com>
Subject: Re: [BEHAVE] TURN status and direction
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On Oct 11, 2007, at 11:52 PM, R=E9mi Denis-Courmont wrote:
> 	Laine,
>
> As I already noted, real-life MTUs tend to be multiple of 8, and =20
> are pretty
> much always multiple of 4 bytes. As such, I have no problem using a =20=

> two byte
> header, or even 4 bytes, if it is deemed that NAT-like O(1) IP&UDP =20
> checksum
> adjustment is a nice thing to have.

I have been assuming that packet sizes are essentially random, but =20
perhaps
that's not a correct assumption.  Specifically, I assume that increasing
the packet size by 2 bytes instead of 1 byte will make it more likely =20=

that
the packet will then exceed the MTU.  Exceeding the MTU seems like an
especially bad thing for TURN to do.


>
> In fact, one big advantage with the 4 bytes header may be to =20
> harmonize the UDP
> and TCP framing of TURN, i.e. two bytes channel and two bytes =20
> length. That
> being noted, I am not an author to the TURN I-D, nor am I attending =20=

> the DT.
>
> --=20
> R=E9mi Denis-Courmont
>
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave

-brian
briank@briank.com





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The design team had a call today to discuss all of this. Our thinking 
was this:

1. its always a good thing to support real implementations; if folks 
want to do kernel level implementations and they get a measurable 
benefit by using an even number of bytes for the overhead, thats a good 
thing

2. There is a big jump in complexity once we make usage of the channel 
mechanism optional. In fact the overall protocol becomes more complex 
than if we had left it out. A big thing to remember is that, with ICE, 
TURN ends up being the 'mechanism of last resort', and if the rest of 
behaves deliverables are implemented, the need for turn goes away. So 
given that it is a protocol meant to address a corner case that goes 
away, I think you need to drive the complexity way down, even at the 
expense of a few more bytes. So I'd rather ALWAYS use it.

3. While you can work your way around the word aligmnent issues and 
still have a decent implementation that performs well, its likely this 
will be troubling for some implementors.

4. the unification of the framing between UDP and TCP is attractive.


So I'd propose that the framing for UDP also use the 32 bit framing that 
TCP uses. This would include a 16 bit length that is always set to zero 
for UDP. This also makes it predictable so it can be compressed out 
should someone have real heartache about it (a ROHC extension or 
something could handle that I suspect).

-Jonathan R.

Brian Korver wrote:
> On Oct 11, 2007, at 11:55 PM, Rémi Denis-Courmont wrote
>> Le Friday 12 October 2007 02:05:04 ext Brian Korver, vous avez écrit :
>>> I don't see the point of going to a 2 byte channel number tho'.
>>> Either the TURN server is relaying the packet or processing it
>>> locally (channel=0).  In the former case (which is the common case),
>>> alignment doesn't matter but bumping up against MTU limits does.  In
>>> the later case, alignment may be required (or at least desired), but
>>> these events don't occur often and so an extra memcpy for the
>>> purposes of alignment is a small price to pay for optimizing the
>>> common case.
>>
>> The point of 2 bytes rather than 1 byte channel ID is that you can 
>> "fix" the
>> outer UDP checksum with a single substraction. Many NATs device "fix" 
>> the UDP
>> checksum by simply substracting and adding the modified port numbers, 
>> rather
>> than checksumming the entire UDP payload.
>>
>> Of course, this type of optimization is only valid for raw (in-kernel)
>> implementations, where the checksum field can be accessed.
> 
> We're optimizing for in-kernel NAT implementations?  Hmm, I still
> don't see the real point to the 2 byte channel number.
> 
> 
>>
>> -- 
>> Rémi Denis-Courmont
>>
>>
>> _______________________________________________
>> Behave mailing list
>> Behave@ietf.org
>> https://www1.ietf.org/mailman/listinfo/behave
> 
> -brian
> briank@briank.com
> 
> 
> 
> 
> 
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [BEHAVE] TURN status and direction
Date: Mon, 15 Oct 2007 11:44:26 -0700
To: Jonathan Rosenberg <jdrosen@cisco.com>
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I don't know what I think of this change - not sure I really care, =20
but I don't view the reasons as very strong.

On Oct 12, 2007, at 2:41 PM, Jonathan Rosenberg wrote:

> The design team had a call today to discuss all of this. Our =20
> thinking was this:
>
> 1. its always a good thing to support real implementations; if =20
> folks want to do kernel level implementations and they get a =20
> measurable benefit by using an even number of bytes for the =20
> overhead, thats a good thing

I don't believe this would offer any performance improvement in the =20
kernel or not - any modern processor I am aware of can copy and =20
checksum the data just as fast as coping the data.

>
> 2. There is a big jump in complexity once we make usage of the =20
> channel mechanism optional. In fact the overall protocol becomes =20
> more complex than if we had left it out. A big thing to remember is =20=

> that, with ICE, TURN ends up being the 'mechanism of last resort', =20
> and if the rest of behaves deliverables are implemented, the need =20
> for turn goes away. So given that it is a protocol meant to address =20=

> a corner case that goes away, I think you need to drive the =20
> complexity way down, even at the expense of a few more bytes. So =20
> I'd rather ALWAYS use it.

Agree with the goal of making the protocol less complicated but I =20
really have a hard time believing the inclusion of a length field =20
that is always set to zero for UDP somehow makes the protocol =20
substantially less complex.

>
> 3. While you can work your way around the word aligmnent issues and =20=

> still have a decent implementation that performs well, its likely =20
> this will be troubling for some implementors.

Again, I don't believe this at all. Dealing with a TCP stream is =20
going to more or less make you deal with misaligned stuff anyways. We =20=

are talking a handful of lines of code.

>
> 4. the unification of the framing between UDP and TCP is attractive.

Personally I view this as amazingly minimal since the code for =20
reading them will still be fairly separate due to the difference in =20
handling connected TCP sockets and most likely a single file =20
descriptor for the UDP implementations.

>
>
> So I'd propose that the framing for UDP also use the 32 bit framing =20=

> that TCP uses. This would include a 16 bit length that is always =20
> set to zero for UDP. This also makes it predictable so it can be =20
> compressed out should someone have real heartache about it (a ROHC =20
> extension or something could handle that I suspect).

I don't know that I really have an opinion on if this is a good =20
change or not but I just find the reasons sort of dubious.


>
> -Jonathan R.
>
> Brian Korver wrote:
>> On Oct 11, 2007, at 11:55 PM, R=E9mi Denis-Courmont wrote
>>> Le Friday 12 October 2007 02:05:04 ext Brian Korver, vous avez =20
>>> =E9crit :
>>>> I don't see the point of going to a 2 byte channel number tho'.
>>>> Either the TURN server is relaying the packet or processing it
>>>> locally (channel=3D0).  In the former case (which is the common =20
>>>> case),
>>>> alignment doesn't matter but bumping up against MTU limits =20
>>>> does.  In
>>>> the later case, alignment may be required (or at least desired), =20=

>>>> but
>>>> these events don't occur often and so an extra memcpy for the
>>>> purposes of alignment is a small price to pay for optimizing the
>>>> common case.
>>>
>>> The point of 2 bytes rather than 1 byte channel ID is that you =20
>>> can "fix" the
>>> outer UDP checksum with a single substraction. Many NATs device =20
>>> "fix" the UDP
>>> checksum by simply substracting and adding the modified port =20
>>> numbers, rather
>>> than checksumming the entire UDP payload.
>>>
>>> Of course, this type of optimization is only valid for raw (in-=20
>>> kernel)
>>> implementations, where the checksum field can be accessed.
>> We're optimizing for in-kernel NAT implementations?  Hmm, I still
>> don't see the real point to the 2 byte channel number.
>>>
>>> --=20
>>> R=E9mi Denis-Courmont
>>>
>>>
>>> _______________________________________________
>>> Behave mailing list
>>> Behave@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/behave
>> -brian
>> briank@briank.com
>> _______________________________________________
>> Behave mailing list
>> Behave@ietf.org
>> https://www1.ietf.org/mailman/listinfo/behave
>
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ =20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave


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From: Eric Rescorla <ekr@networkresonance.com>
To: Cullen Jennings <fluffy@cisco.com>
Subject: Re: [BEHAVE] TURN status and direction
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At Mon, 15 Oct 2007 11:44:26 -0700,
Cullen Jennings wrote:
> 
> 
> I don't know what I think of this change - not sure I really care,  
> but I don't view the reasons as very strong.
> 
> On Oct 12, 2007, at 2:41 PM, Jonathan Rosenberg wrote:
> 
> > The design team had a call today to discuss all of this. Our  
> > thinking was this:
> >
> > 1. its always a good thing to support real implementations; if  
> > folks want to do kernel level implementations and they get a  
> > measurable benefit by using an even number of bytes for the  
> > overhead, thats a good thing
> 
> I don't believe this would offer any performance improvement in the  
> kernel or not - any modern processor I am aware of can copy and  
> checksum the data just as fast as coping the data.

I tend to agree with Cullen here. Do we have some actual measurements?


> > 2. There is a big jump in complexity once we make usage of the  
> > channel mechanism optional. In fact the overall protocol becomes  
> > more complex than if we had left it out. A big thing to remember is  
> > that, with ICE, TURN ends up being the 'mechanism of last resort',  
> > and if the rest of behaves deliverables are implemented, the need  
> > for turn goes away. So given that it is a protocol meant to address  
> > a corner case that goes away, I think you need to drive the  
> > complexity way down, even at the expense of a few more bytes. So  
> > I'd rather ALWAYS use it.
> 
> Agree with the goal of making the protocol less complicated but I  
> really have a hard time believing the inclusion of a length field  
> that is always set to zero for UDP somehow makes the protocol  
> substantially less complex.

To the extent to which we care about on-the-wire packet size, it
seems like adding 2 bytes to avoid an if statement in the code
seems unwise.

-Ekr


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A new version of I-D, draft-ietf-behave-p2p-state-05.txt has been successfuly submitted by Pyda Srisuresh and posted to the IETF repository.

Filename:	 draft-ietf-behave-p2p-state
Revision:	 05
Title:		 State of Peer-to-Peer(P2P) Communication Across Network Address Translators(NATs)
Creation_date:	 2007-10-15
WG ID:		 behave
Number_of_pages: 29

Abstract:
This memo documents the various methods known to be in use by
applications to establish direct communication in the presence
of Network Address Translators (NATs) at the current time. Although
this memo is intended to be mainly descriptive, the Security 
Considerations section makes some purely advisory recommendations
about how to deal with security vulnerabilities the applications
could inadvertently create when using the methods described. This
memo covers NAT traversal approaches used by both TCP and UDP
based applications. This memo is not an endorsement of the methods
described, but merely an attempt to capture them in a document.
                                                                                  


The IETF Secretariat.




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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Behavior Engineering for Hindrance Avoidance Working Group of the IETF.


	Title           : State of Peer-to-Peer(P2P) Communication Across Network Address Translators(NATs)
	Author(s)       : P. Srisuresh, et al.
	Filename        : draft-ietf-behave-p2p-state-05.txt
	Pages           : 29
	Date            : 2007-10-16

This memo documents the various methods known to be in use by
applications to establish direct communication in the presence
of Network Address Translators (NATs) at the current time. Although
this memo is intended to be mainly descriptive, the Security 
Considerations section makes some purely advisory recommendations
about how to deal with security vulnerabilities the applications
could inadvertently create when using the methods described. This
memo covers NAT traversal approaches used by both TCP and UDP
based applications. This memo is not an endorsement of the methods
described, but merely an attempt to capture them in a document.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-behave-p2p-state-05.txt

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Behavior Engineering for Hindrance Avoidance Working Group of the IETF.


	Title           : NAT Behavioral Requirements for ICMP protocol
	Author(s)       : P. Srisuresh, et al.
	Filename        : draft-ietf-behave-nat-icmp-06.txt
	Pages           : 22
	Date            : 2007-10-16

This document specifies the behavioral properties required of the 
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Internet Control Message Protocol (ICMP). The objective of this
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A new version of I-D, draft-ietf-behave-nat-icmp-06.txt has been successfuly submitted by Pyda Srisuresh and posted to the IETF repository.

Filename:	 draft-ietf-behave-nat-icmp
Revision:	 06
Title:		 NAT Behavioral Requirements for ICMP protocol
Creation_date:	 2007-10-15
WG ID:		 behave
Number_of_pages: 22

Abstract:
This document specifies the behavioral properties required of the 
Network Address Translator (NAT) devices in conjunction with the
Internet Control Message Protocol (ICMP). The objective of this
memo is to make NAT devices more predictable and compatible with
diverse application protocols that traverse the devices. Companion
documents provide behavioral recommendations specific to TCP, UDP
and other protocols.
                                                                                  


The IETF Secretariat.




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The TURN editing team is anxious to start making changes to TURN. The 
big thing we need to get group consensus on is this: Is there general 
consensus around adding some kind of shim (we can debate 8 vs. 32 bits) 
for UDP, for the purposes of adding a channel concept so we can remove 
the active destination functionality from the spec?

Please weigh in.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From: "Derek MacDonald" <derek@counterpath.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>
Subject: Re: [BEHAVE] General consensus on a UDP shim for TURN
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I prefer the channel approach over the active destination approach mainly
for aesthetic reasons. AD always seemed like a bit of a hack and required
lots of jumping through hoops when forking happened, but it wasn't too hard
to hide from the user of our turn library.

That said, I think we need to keep the shim as light as possible; I have
seen deployments where there is very little headroom in the UDP packets--~20
bytes.  It's possible to optimize down to th bone when you have prediictable
uris(numeric, 10 digit) and record routes(IP addrs) in your network. 8 bits
sounds good:)
-Derek


On 10/16/07, Jonathan Rosenberg <jdrosen@cisco.com> wrote:
>
> The TURN editing team is anxious to start making changes to TURN. The
> big thing we need to get group consensus on is this: Is there general
> consensus around adding some kind of shim (we can debate 8 vs. 32 bits)
> for UDP, for the purposes of adding a channel concept so we can remove
> the active destination functionality from the spec?
>
> Please weigh in.
>
> Thanks,
> Jonathan R.
> --
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
>

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I prefer the channel approach over the active destination approach mainly for aesthetic reasons. AD always seemed like a bit of a hack and required lots of jumping through hoops when forking happened, but it wasn&#39;t too hard to hide from the user of our turn library.
<br><br>That said, I think we need to keep the shim as light as possible; I have seen deployments where there is very little headroom in the UDP packets--~20 bytes.&nbsp; It&#39;s possible to optimize down to th bone when you have prediictable uris(numeric, 10 digit) and record routes(IP addrs) in your network. 8 bits sounds good:)
<br>-Derek<br><br><br><div><span class="gmail_quote">On 10/16/07, <b class="gmail_sendername">Jonathan Rosenberg</b> &lt;<a href="mailto:jdrosen@cisco.com">jdrosen@cisco.com</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
The TURN editing team is anxious to start making changes to TURN. The<br>big thing we need to get group consensus on is this: Is there general<br>consensus around adding some kind of shim (we can debate 8 vs. 32 bits)<br>
for UDP, for the purposes of adding a channel concept so we can remove<br>the active destination functionality from the spec?<br><br>Please weigh in.<br><br>Thanks,<br>Jonathan R.<br>--<br>Jonathan D. Rosenberg, Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 600 Lanidex Plaza
<br>Cisco Fellow&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Parsippany, NJ 07054-2711<br>Cisco Systems<br><a href="mailto:jdrosen@cisco.com">jdrosen@cisco.com</a>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;FAX:&nbsp;&nbsp; (973) 952-5050<br><a href="http://www.jdrosen.net">
http://www.jdrosen.net</a>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) 952-5000<br><a href="http://www.cisco.com">http://www.cisco.com</a><br><br><br>_______________________________________________<br>Behave mailing list<br><a href="mailto:Behave@ietf.org">
Behave@ietf.org</a><br><a href="https://www1.ietf.org/mailman/listinfo/behave">https://www1.ietf.org/mailman/listinfo/behave</a><br></blockquote></div><br>

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Jonathan Rosenberg wrote:
> The TURN editing team is anxious to start making changes to TURN. The
> big thing we need to get group consensus on is this: Is there general
> consensus around adding some kind of shim (we can debate 8 vs. 32 bits)
> for UDP, for the purposes of adding a channel concept so we can remove
> the active destination functionality from the spec?

I like the channel proposal.  This is simpler to implement than the
active destination stuff and also remove the problem of having to choose
what destination is the active one.  I already wrote some basic code
implementing this, and didn't find any issue yet.

I support using 16 bits for the channel #, and no length field.  I fell
better with 65535 channels available in each direction than 255, knowing
that channels # cannot be reused.


-- 
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [RFC1855-compliant space for rent ]
Work: marc@8x8.com            [                                 ]
[                                                               ]


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From behave-bounces@ietf.org Thu Oct 18 02:08:53 2007
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Le Tuesday 16 October 2007 23:17:48 ext Jonathan Rosenberg, vous avez =E9cr=
it=A0:
> The TURN editing team is anxious to start making changes to TURN. The
> big thing we need to get group consensus on is this: Is there general
> consensus around adding some kind of shim (we can debate 8 vs. 32 bits)
> for UDP, for the purposes of adding a channel concept so we can remove
> the active destination functionality from the spec?

I tend to prefer fixed size "overhead" (whatever the size), but I won't whi=
ne=20
if the SAD approach is maintained.

But I really do NOT want to have both options in the protocol.

=2D-=20
R=E9mi Denis-Courmont


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I requested a review of STUNbis from the DNS directorate and here is the
review done by Peter Koch. Authors please respond to the comments he
provide.

Cheers

Magnus Westerlund

IETF Transport Area Director & TSVWG Chair
----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM/M
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
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From: Peter Koch <pk@DENIC.DE>
To: Jari Arkko <jari.arkko@piuha.net>
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	Magnus Westerlund <magnus.westerlund@ericsson.com>
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On Fri, Oct 12, 2007 at 03:59:52PM +0300, Jari Arkko wrote:

> Magnus Westerlund wrote:

> > STUN (draft-ietf-behave-rfc3489bis-11) is using DNS to locate STUN
> > services for the desired protocol (UDP or TCP). Could you please have
> > the DNS directorate review what is written on this usage.

I have reviewed this draft for the DNS Directorate and therefore focused
on Section 9 "DNS Discovery of a Server" and the security considerations.
There are several issues with the use of SRV RRs that need to be addressed.

> 9.  DNS Discovery of a Server
> 
>    This section describes an optional procedure for STUN that allows a
>    client to use DNS to determine the IP address and port of a server.
>    A STUN usage must describe if and when this extension is used.  To
>    use this procedure, the client must have a domain name and a service
>    name; the usage must also describe how the client obtains these.

NIT: since it's not about the client's name but the servers: "the client
     must _know_ ..."; otherwise easy to mis-read.

>    When a client wishes to locate a STUN server in the public Internet
>    that accepts Binding Request/Response transactions, the SRV service
>    name is "stun".  STUN usages MAY define additional DNS SRV service
>    names.

This is in violation of RFC 2782. The port name assigned by IANA is
"nat-stun-port", so this should be the one used in SRV handling. Same for
the definition of additional service names.

>    The domain name is resolved to a transport address using the SRV
>    procedures specified in [RFC2782].  The DNS SRV service name is the
>    service name provided as input to this procedure.  The protocol in
>    the SRV lookup is the transport protocol the client will run STUN
>    over: "udp" for UDP, "tcp" for TCP, and "tls" for TLS-over-TCP.  If,

The introduction of "tls" as another string for the RFC 2782 "Proto" field
also violates that RFC, since it's not defined as an IP protocol in the
"protocol-numbers" registry.

>    in the future, additional SRV records are defined for TLS over other
>    transport protocols, those will need to utilize an SRV transport
>    token of the form "tls-foo" for transport protocol "foo".

Even more so, the "Proto" parameter isn't application specific, so the
draft under consideration is just the wrong place to make this reservation.

>    The procedures of RFC 2782 are followed to determine the server to
>    contact.  RFC 2782 spells out the details of how a set of SRV records
>    are sorted and then tried.  However, RFC2782 only states that the
>    client should "try to connect to the (protocol, address, service)"
>    without giving any details on what happens in the event of failure.

I'm not sure I understand this concern. Since RFC 2782's section on
"Usage rules" says that one should iterate over the sorted (according to
Priority and Weight) list until a "connection" succeeds, isn't that all
that's needed?

>    When following these procedures, if the STUN transaction times out
>    without receipt of a response, the client SHOULD retry the request to
>    the next server in the list of servers from the DNS SRV response.

Well, _ordered_ list as of RFC 2782.

>    Such a retry is only possible for request/response transmissions,
>    since indication transactions generate no response or timeout.
> 
>    The default port for STUN requests is 3478, for both TCP and UDP.
>    Administrators SHOULD use this port in their SRV records for UDP and
>    TCP, but MAY use others.  There is no default port for STUN over TLS,

I'd emphasize here that the administrator _of the STUN server_ is encouraged
to use port 3478 and then, subsequently, to use this port in the DNS,
not vice versa. I.e., the DNS SRV RR should reflect which port is actually
listened on.

>    however a STUN server SHOULD use a port number for TLS different from
>    3478 so that the server can determine whether the first message it
>    will receive after the TCP connection is set up, is a STUN message or
>    a TLS message.
> 
>    If no SRV records were found, the client performs an A or AAAA record
>    lookup of the domain name.  The result will be a list of IP

The change here is to allow AAAA in addition to A, the original fallback
mechanism already envisioned in RFC 2782.  However, it's time to move
forward and instead of extending a bad idea to v6 it should be terminated and
no "fallback" given.  Especially because this is not a legacy service that
existed before SRV was invented and RFC 3489 already included the SRV usage,
there's no need to address "old" v6 clients agnostic of SRV.  Operational
laziness shouldn't be encouraged.
At the very least, the fallback MUST be excluded for explicit "stop" SRV
RRs (those including the "." target).

>    addresses, each of which can be contacted at the default port using
>    UDP or TCP, independent of the STUN usage.  For usages that require
>    TLS, lack of SRV records is equivalent to a failure of the
>    transaction, since the request or indication MUST NOT be sent unless
>    SRV records provided a transport address specifically for TLS.

In the Security Considerations section the usual DNS caveats and threats
should be mentioned, especially since SRV provides a level of indirection
that can't be protected by use of TLS (you might end up to securely talk
to the perfectly authenticated - albeit wrong - server).
Also, even though the mechanism of obtaining the server's domain name
is not specified in the draft, I'd suggest to mention the risk of hardcoding
the domain name in the application software because of the chance to
"loose" a domain name (by accidental expiry, legal action and so on).

-Peter

PS: Unrelated to DNS, aren't there privacy concerns worth mentioning
    because of the opportunity to monitor application usage?

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Date: Mon, 22 Oct 2007 13:41:17 -0400
To: Magnus Westerlund <magnus.westerlund@ericsson.com>,
	Peter Koch <pk@DENIC.DE>
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Subject: [BEHAVE] Re: [Fwd: Re: DNS directorate review of
	draft-ietf-behave-rfc3489bis-11]
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I have given an initial response to some (but no all) of these points  
inline.

- Philip

On 19-Oct-07, at 04:59 , Magnus Westerlund wrote:

> I requested a review of STUNbis from the DNS directorate and here  
> is the
> review done by Peter Koch. Authors please respond to the comments he
> provide.
>
> Cheers
>
> Magnus Westerlund
>
> IETF Transport Area Director & TSVWG Chair
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM/M
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>
> From: Peter Koch <pk@DENIC.DE>
> Date: October 18, 2007 17:59:00  EDT (CA)
> To: Jari Arkko <jari.arkko@piuha.net>
> Cc: dns directorate <dns-dir@ops.ietf.org>, Magnus Westerlund  
> <magnus.westerlund@ericsson.com>
> Subject: Re: DNS directorate review of draft-ietf-behave-rfc3489bis-11
>
>
> On Fri, Oct 12, 2007 at 03:59:52PM +0300, Jari Arkko wrote:
>
>> Magnus Westerlund wrote:
>
>>> STUN (draft-ietf-behave-rfc3489bis-11) is using DNS to locate STUN
>>> services for the desired protocol (UDP or TCP). Could you please  
>>> have
>>> the DNS directorate review what is written on this usage.
>
> I have reviewed this draft for the DNS Directorate and therefore  
> focused
> on Section 9 "DNS Discovery of a Server" and the security  
> considerations.
> There are several issues with the use of SRV RRs that need to be  
> addressed.
>
>> 9.  DNS Discovery of a Server
>>
>>    This section describes an optional procedure for STUN that  
>> allows a
>>    client to use DNS to determine the IP address and port of a  
>> server.
>>    A STUN usage must describe if and when this extension is used.  To
>>    use this procedure, the client must have a domain name and a  
>> service
>>    name; the usage must also describe how the client obtains these.
>
> NIT: since it's not about the client's name but the servers: "the  
> client
>      must _know_ ..."; otherwise easy to mis-read.

Yes. This would improve readability.

>
>>    When a client wishes to locate a STUN server in the public  
>> Internet
>>    that accepts Binding Request/Response transactions, the SRV  
>> service
>>    name is "stun".  STUN usages MAY define additional DNS SRV service
>>    names.
>
> This is in violation of RFC 2782. The port name assigned by IANA is
> "nat-stun-port", so this should be the one used in SRV handling.  
> Same for
> the definition of additional service names.

The use of "stun" for this service was introduced in RFC 3489, and I  
believe that
there is already some deployment today.
As I see it, the problem is that IANA assigned a name of "nat-stun- 
port" for
the assigned port number for STUN, rather than simply "stun".
So my initial thought is to see if IANA has any objections to changing
the name of this port to "stun".


>
>>    The domain name is resolved to a transport address using the SRV
>>    procedures specified in [RFC2782].  The DNS SRV service name is  
>> the
>>    service name provided as input to this procedure.  The protocol in
>>    the SRV lookup is the transport protocol the client will run STUN
>>    over: "udp" for UDP, "tcp" for TCP, and "tls" for TLS-over- 
>> TCP.  If,
>
> The introduction of "tls" as another string for the RFC 2782  
> "Proto" field
> also violates that RFC, since it's not defined as an IP protocol in  
> the
> "protocol-numbers" registry.

How do other protocols handle this issue (wanting to run over both TCP
and TLS over TCP)?

>
>>    The procedures of RFC 2782 are followed to determine the server to
>>    contact.  RFC 2782 spells out the details of how a set of SRV  
>> records
>>    are sorted and then tried.  However, RFC2782 only states that the
>>    client should "try to connect to the (protocol, address, service)"
>>    without giving any details on what happens in the event of  
>> failure.
>
> I'm not sure I understand this concern. Since RFC 2782's section on
> "Usage rules" says that one should iterate over the sorted  
> (according to
> Priority and Weight) list until a "connection" succeeds, isn't that  
> all
> that's needed?

This paragraph was added before I came on board as an author.
However, I believe the concern is as follows. RCC 2782 says:

                 for each address record found, try to connect to the
                (protocol, address, service).

This doesn't say what to do if the attempt fails. Since there are at  
least
conceivably a couple of ways this might be handled in STUN, it seems  
desirable
to be explicit. Thus the document says:
    When following these procedures, if the STUN transaction times out
    without receipt of a response, the client SHOULD retry the  
request to
    the next server in the list of servers from the DNS SRV response.
    Such a retry is only possible for request/response transmissions,
    since indication transactions generate no response or timeout.


>
>>    When following these procedures, if the STUN transaction times out
>>    without receipt of a response, the client SHOULD retry the  
>> request to
>>    the next server in the list of servers from the DNS SRV response.
>
> Well, _ordered_ list as of RFC 2782.
I think that you are suggesting that this be reworded as
"the next server in the order defined by RFC 2782".
This seems good.

>>
>>    The default port for STUN requests is 3478, for both TCP and UDP.
>>    Administrators SHOULD use this port in their SRV records for  
>> UDP and
>>    TCP, but MAY use others.  There is no default port for STUN  
>> over TLS,
>
> I'd emphasize here that the administrator _of the STUN server_ is  
> encouraged
> to use port 3478 and then, subsequently, to use this port in the DNS,
> not vice versa. I.e., the DNS SRV RR should reflect which port is  
> actually
> listened on.

Yes. Good point.


>
>>    however a STUN server SHOULD use a port number for TLS  
>> different from
>>    3478 so that the server can determine whether the first message it
>>    will receive after the TCP connection is set up, is a STUN  
>> message or
>>    a TLS message.
>>
>>    If no SRV records were found, the client performs an A or AAAA  
>> record
>>    lookup of the domain name.  The result will be a list of IP
>
> The change here is to allow AAAA in addition to A, the original  
> fallback
> mechanism already envisioned in RFC 2782.  However, it's time to move
> forward and instead of extending a bad idea to v6 it should be  
> terminated and
> no "fallback" given.  Especially because this is not a legacy  
> service that
> existed before SRV was invented and RFC 3489 already included the  
> SRV usage,
> there's no need to address "old" v6 clients agnostic of SRV.   
> Operational
> laziness shouldn't be encouraged.
> At the very least, the fallback MUST be excluded for explicit  
> "stop" SRV
> RRs (those including the "." target).
>
>>    addresses, each of which can be contacted at the default port  
>> using
>>    UDP or TCP, independent of the STUN usage.  For usages that  
>> require
>>    TLS, lack of SRV records is equivalent to a failure of the
>>    transaction, since the request or indication MUST NOT be sent  
>> unless
>>    SRV records provided a transport address specifically for TLS.
>
> In the Security Considerations section the usual DNS caveats and  
> threats
> should be mentioned, especially since SRV provides a level of  
> indirection
> that can't be protected by use of TLS (you might end up to securely  
> talk
> to the perfectly authenticated - albeit wrong - server).
> Also, even though the mechanism of obtaining the server's domain name
> is not specified in the draft, I'd suggest to mention the risk of  
> hardcoding
> the domain name in the application software because of the chance to
> "loose" a domain name (by accidental expiry, legal action and so on).
>
> -Peter
>
> PS: Unrelated to DNS, aren't there privacy concerns worth mentioning
>     because of the opportunity to monitor application usage?
>
>



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From behave-bounces@ietf.org Mon Oct 22 21:35:53 2007
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From: Philip Matthews <philip_matthews@magma.ca>
Date: Mon, 22 Oct 2007 21:35:27 -0400
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Subject: [BEHAVE] Would anyone be negatively affected by this change?
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It was recently pointed out to us by Peter Koch
(in a message forwarded to this list by Magnus)
that
     http://www.iana.org/assignments/port-numbers
assigns a service name of "nat-stun-port" to port 3478,
which is the well-known port assigned to STUN.
As Peter points out, this introduces an inconsistency
with RFC 2782.

One way to correct this inconsistency is to change the
service name from "nat-stun-port" to just "stun".

The authors of draft-ietf-behave-rfc3489bis would like to know
if anyone would be negatively affected by this change if we were
to go ahead with it. Does anyone currently use "nat-stun-port"
as a service name for STUN?

- Philip





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Subject: [BEHAVE] TURN : How can a client ask a TCP disconnection ?
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One simple question about TURN :

Imagine the case where a TURN client has allocated a TCP address (a TCP
server) on a TURN server.
Several remote peers have established a connection on this allocated port.

And now, the client wants to disconnect one of them (and only one).
I did not found any request to do that.


--=20
Ga=EBl Br=E9ard

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<br>One simple question about TURN :<br><br>Imagine the case where a TURN c=
lient has allocated a TCP address (a TCP server) on a TURN server.<br>Sever=
al remote peers have established a connection on this allocated port.<br>
<br>And now, the client wants to disconnect one of them (and only one).<br>=
I did not found any request to do that. <br><br><br>-- <br>Ga=EBl Br=E9ard

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	ext =?utf-8?q?Ga=C3=ABl_Br=C3=A9ard?= <gael.breard@gmail.com>
Subject: Re: [BEHAVE] TURN : How can a client ask a TCP disconnection ?
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Le Friday 26 October 2007 10:38:50 ext Ga=C3=ABl Br=C3=A9ard, vous avez =C3=
=A9crit=C2=A0:
> One simple question about TURN :
>
> Imagine the case where a TURN client has allocated a TCP address (a TCP
> server) on a TURN server.
> Several remote peers have established a connection on this allocated port.
>
> And now, the client wants to disconnect one of them (and only one).
> I did not found any request to do that.

It used to be so that you terminated the TCP connection. Multiple TCP=20
connections per TURN session was then added, but remains buggy to this day.

As far as I understand, the current public draft is way out-of-sync with wh=
ere=20
the WG is heading anyway.

=2D-=20
R=C3=A9mi Denis-Courmont


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Subject: [BEHAVE] Re: [MMUSIC] Looking for UAs supporting ICE
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Hi Livio,

Sorry for the late response.

My employer, Eyeball Networks, has a user-agent and other NAT-traversal
solutions that support recent drafts of STUN, TURN and ICE.  We have three
products of interest:
 
- Eyeball Messenger: A soft-client or user-agent that provides voice, video
and IM features with NAT traversal using STUN, TURN and ICE. 

- AnyFirewall Engine: Industry's leading firewall and NAT traversal SDK
offering a comprehensive implementation of STUN, TURN and ICE (along with
optional support for UPnP gateways and HTTP-tunneling), and

- AnyFirewall Server: A carrier-grade STUN and TURN server ready for mass
deployment.

Eyeball white-paper titled, "NAT Traversal using ICE for VoIP
and Internet Communications"  provides a high level overview of how
NAT-traversal using ICE works, and how Eyeball solutions can help.
(http://www.eyeball.com/technology/whitepapers/NAT_Traversal_Using_ICE.pdf) 

Hope this helps.  Please let me know if you need additional information.

Regards,
Raihan


-------------------

   Date: Thu, 23 Aug 2007 16:45:23 +0200
>> From: Livio Torrero <livio.torrero@polito.it>
>> Subject: [MMUSIC] Looking for UAs supporting ICE
>> To: MMUSIC <mmusic@ietf.org>
>> Message-ID: <46CD9D83.6090305@polito.it>
>> Content-Type: text/plain; charset=ISO-8859-15; format=flowed
>>
>> I'm currently investigating NAT traversal techniques for SIP User
>>  Agents
>> and I've read all the ICE-related drafts. I'm very interested in it and
>>
>> I would like to try a UA supporting ICE. Could you please suggest me if
>>
>> and where I can download a free UA supporting ICE? If it is not
>> possible, can you please suggest me commercial UAs supporting it? I've
>> just tried Counterpath X-lite but it seems to me that the ICE version
>> included is out of date. Is there any chance to have an UA supporting a
>>
>> more recent version?
>>
>> Best regards,
>> Livio Torrero



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From behave-bounces@ietf.org Fri Oct 26 14:43:59 2007
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From: "David Barrett" <dbarrett@quinthar.com>
To: "'Raihan'" <raihan@eyeball.com>,
	<livio.torrero@polito.it>
Subject: RE: [BEHAVE] Re: [MMUSIC] Looking for UAs supporting ICE
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That's great!  Do you have any test results showing the effectiveness of the
NAT penetration technology?  Specifically, what is the success rate of
direct peer connectivity (non-TURN), and how long does it take to set up a
connection?

-david

> -----Original Message-----
> From: Raihan [mailto:raihan@eyeball.com]
> Sent: Friday, October 26, 2007 10:36 AM
> To: livio.torrero@polito.it
> Cc: behave@ietf.org; mmusic@ietf.org
> Subject: [BEHAVE] Re: [MMUSIC] Looking for UAs supporting ICE
> 
> Hi Livio,
> 
> Sorry for the late response.
> 
> My employer, Eyeball Networks, has a user-agent and other NAT-traversal
> solutions that support recent drafts of STUN, TURN and ICE.  We have three
> products of interest:
> 
> - Eyeball Messenger: A soft-client or user-agent that provides voice,
> video
> and IM features with NAT traversal using STUN, TURN and ICE.
> 
> - AnyFirewall Engine: Industry's leading firewall and NAT traversal SDK
> offering a comprehensive implementation of STUN, TURN and ICE (along with
> optional support for UPnP gateways and HTTP-tunneling), and
> 
> - AnyFirewall Server: A carrier-grade STUN and TURN server ready for mass
> deployment.
> 
> Eyeball white-paper titled, "NAT Traversal using ICE for VoIP
> and Internet Communications"  provides a high level overview of how
> NAT-traversal using ICE works, and how Eyeball solutions can help.
> (http://www.eyeball.com/technology/whitepapers/NAT_Traversal_Using_ICE.pdf
> )
> 
> Hope this helps.  Please let me know if you need additional information.
> 
> Regards,
> Raihan
> 
> 
> -------------------
> 
>    Date: Thu, 23 Aug 2007 16:45:23 +0200
> >> From: Livio Torrero <livio.torrero@polito.it>
> >> Subject: [MMUSIC] Looking for UAs supporting ICE
> >> To: MMUSIC <mmusic@ietf.org>
> >> Message-ID: <46CD9D83.6090305@polito.it>
> >> Content-Type: text/plain; charset=ISO-8859-15; format=flowed
> >>
> >> I'm currently investigating NAT traversal techniques for SIP User
> >>  Agents
> >> and I've read all the ICE-related drafts. I'm very interested in it and
> >>
> >> I would like to try a UA supporting ICE. Could you please suggest me if
> >>
> >> and where I can download a free UA supporting ICE? If it is not
> >> possible, can you please suggest me commercial UAs supporting it? I've
> >> just tried Counterpath X-lite but it seems to me that the ICE version
> >> included is out of date. Is there any chance to have an UA supporting a
> >>
> >> more recent version?
> >>
> >> Best regards,
> >> Livio Torrero
> 
> 
> 
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave



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Behave,

I think the discuss from Lars below needs WG discussion. I like to here
opinions about this.

Cheers

Magnus Westerlund

IETF Transport Area Director & TSVWG Chair
----------------------------------------------------------------------
Multimedia Technologies, Ericsson Research EAB/TVM/M
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
----------------------------------------------------------------------

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	FILETIME=[E9915330:01C818E2]

Discuss:
Section 4.1., paragraph 4:
>    REQ-3:  When an ICMP Error packet is received, the NAT device
>    SHOULD do the following.
>    a) If the IP checksum of the embedded packet fails to validate, drop
>    the Error packet

  DISCUSS: I disagree. (And I apologize for not catching this during my
  earlier review.) The reason that (part of) the packet that caused an
  ICMP message to be generated is included is for diagnostic purposes,
  so the end hosts can determine what went wrong and determine whether
  they need to react in some way. This can't happen if NATs drop these
  packets. Also, the reason why some end system OSs drop ICMP messages
  with invalid checksums is to protect against some attacks. NATs,
  however, are not firewalls. It is inappropriate for this document to
  specify NAT mechanisms that implement a specific security policy.
  There is also no IETF standard that recommends that end hosts drop
  ICMP packets with payloads that have invalid checksums, which is
  another reason why this document should not recommend that NATs do
  this. (And, as an aside, the effectiveness of this "security" measure
  is questionable, because if I can't spoof the complete packet with a
  valid checksum, I'd just include a partial packet in the payload, for
  which you can't validate the checksum, but that you must still act
  on.)

Comment:
Section 240, paragraph 0:
>    ICMP Query session timer MUST not expire in less than 60 seconds. We

  s/MUST not/MUST NOT/

Section 240, paragraph 1:
>    RECOMMEND however that the administrator(s) be allowed to configure

  RECOMMEND is not an RFC2119 term. You need to rephrase to say
  RECOMMENDED, MUST or REQUIRED. (Or use lowercase here, because REQ-2
  has the correct 2119 phrasing.)





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Cc: Peter Koch <pk@DENIC.DE>, Behave WG <behave@ietf.org>,
	Dan Wing <dwing@cisco.com>
Subject: [BEHAVE] Re: [Fwd: Re: DNS directorate review of
	draft-ietf-behave-rfc3489bis-11]
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Peter,

Thanks for the review. Some additional comments and responses inline:


Philip Matthews wrote:
> I have given an initial response to some (but no all) of these points 
> inline.
> 
> - Philip
> 
> On 19-Oct-07, at 04:59 , Magnus Westerlund wrote:
> 
>> I requested a review of STUNbis from the DNS directorate and here is the
>> review done by Peter Koch. Authors please respond to the comments he
>> provide.
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> IETF Transport Area Director & TSVWG Chair
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVM/M
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone +46 8 4048287
>> Torshamsgatan 23           | Fax   +46 8 7575550
>> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>>
>> From: Peter Koch <pk@DENIC.DE>
>> Date: October 18, 2007 17:59:00  EDT (CA)
>> To: Jari Arkko <jari.arkko@piuha.net>
>> Cc: dns directorate <dns-dir@ops.ietf.org>, Magnus Westerlund 
>> <magnus.westerlund@ericsson.com>
>> Subject: Re: DNS directorate review of draft-ietf-behave-rfc3489bis-11
>>
>>
>> On Fri, Oct 12, 2007 at 03:59:52PM +0300, Jari Arkko wrote:
>>
>>> Magnus Westerlund wrote:
>>
>>>> STUN (draft-ietf-behave-rfc3489bis-11) is using DNS to locate STUN
>>>> services for the desired protocol (UDP or TCP). Could you please have
>>>> the DNS directorate review what is written on this usage.
>>
>> I have reviewed this draft for the DNS Directorate and therefore focused
>> on Section 9 "DNS Discovery of a Server" and the security considerations.
>> There are several issues with the use of SRV RRs that need to be 
>> addressed.
>>
>>> 9.  DNS Discovery of a Server
>>>
>>>    This section describes an optional procedure for STUN that allows a
>>>    client to use DNS to determine the IP address and port of a server.
>>>    A STUN usage must describe if and when this extension is used.  To
>>>    use this procedure, the client must have a domain name and a service
>>>    name; the usage must also describe how the client obtains these.
>>
>> NIT: since it's not about the client's name but the servers: "the client
>>      must _know_ ..."; otherwise easy to mis-read.
> 
> Yes. This would improve readability.

fixed.


> 
>>
>>>    When a client wishes to locate a STUN server in the public Internet
>>>    that accepts Binding Request/Response transactions, the SRV service
>>>    name is "stun".  STUN usages MAY define additional DNS SRV service
>>>    names.
>>
>> This is in violation of RFC 2782. The port name assigned by IANA is
>> "nat-stun-port", so this should be the one used in SRV handling. Same for
>> the definition of additional service names.
> 
> The use of "stun" for this service was introduced in RFC 3489, and I 
> believe that
> there is already some deployment today.
> As I see it, the problem is that IANA assigned a name of "nat-stun-port" 
> for
> the assigned port number for STUN, rather than simply "stun".
> So my initial thought is to see if IANA has any objections to changing
> the name of this port to "stun".

I agree. Changing this to 'nat-stun-port' at this time would break 
backwards compatibility with rfc3489; something we have been trying to 
retain, at least for deployed functionality (including this).

> 
> 
>>
>>>    The domain name is resolved to a transport address using the SRV
>>>    procedures specified in [RFC2782].  The DNS SRV service name is the
>>>    service name provided as input to this procedure.  The protocol in
>>>    the SRV lookup is the transport protocol the client will run STUN
>>>    over: "udp" for UDP, "tcp" for TCP, and "tls" for TLS-over-TCP.  If,
>>
>> The introduction of "tls" as another string for the RFC 2782 "Proto" 
>> field
>> also violates that RFC, since it's not defined as an IP protocol in the
>> "protocol-numbers" registry.
> 
> How do other protocols handle this issue (wanting to run over both TCP
> and TLS over TCP)?

The only other solutions I am aware of for this, would be to add NAPTR 
support as a layer on top, or add something akin to STARTTLS to stun, 
both of which I'd rather avoid. Hopefully there is a simpler alternative 
you can suggest.

> 
>>
>>>    The procedures of RFC 2782 are followed to determine the server to
>>>    contact.  RFC 2782 spells out the details of how a set of SRV records
>>>    are sorted and then tried.  However, RFC2782 only states that the
>>>    client should "try to connect to the (protocol, address, service)"
>>>    without giving any details on what happens in the event of failure.
>>
>> I'm not sure I understand this concern. Since RFC 2782's section on
>> "Usage rules" says that one should iterate over the sorted (according to
>> Priority and Weight) list until a "connection" succeeds, isn't that all
>> that's needed?
> 
> This paragraph was added before I came on board as an author.
> However, I believe the concern is as follows. RCC 2782 says:
> 
>                 for each address record found, try to connect to the
>                (protocol, address, service).
> 
> This doesn't say what to do if the attempt fails. Since there are at least
> conceivably a couple of ways this might be handled in STUN, it seems 
> desirable
> to be explicit. Thus the document says:
>    When following these procedures, if the STUN transaction times out
>    without receipt of a response, the client SHOULD retry the request to
>    the next server in the list of servers from the DNS SRV response.
>    Such a retry is only possible for request/response transmissions,
>    since indication transactions generate no response or timeout.

Right. This trend, of clarifying failure treatment, was something 
started in RFC 3263 and has carried into most of the DNS behaviors 
described in other RAI-related protocols.

> 
> 
>>
>>>    When following these procedures, if the STUN transaction times out
>>>    without receipt of a response, the client SHOULD retry the request to
>>>    the next server in the list of servers from the DNS SRV response.
>>
>> Well, _ordered_ list as of RFC 2782.
> I think that you are suggesting that this be reworded as
> "the next server in the order defined by RFC 2782".
> This seems good.

fixed.

> 
>>>
>>>    The default port for STUN requests is 3478, for both TCP and UDP.
>>>    Administrators SHOULD use this port in their SRV records for UDP and
>>>    TCP, but MAY use others.  There is no default port for STUN over TLS,
>>
>> I'd emphasize here that the administrator _of the STUN server_ is 
>> encouraged
>> to use port 3478 and then, subsequently, to use this port in the DNS,
>> not vice versa. I.e., the DNS SRV RR should reflect which port is 
>> actually
>> listened on.
> 
> Yes. Good point.

Added a sentence saying this.

> 
> 
>>
>>>    however a STUN server SHOULD use a port number for TLS different from
>>>    3478 so that the server can determine whether the first message it
>>>    will receive after the TCP connection is set up, is a STUN message or
>>>    a TLS message.
>>>
>>>    If no SRV records were found, the client performs an A or AAAA record
>>>    lookup of the domain name.  The result will be a list of IP
>>
>> The change here is to allow AAAA in addition to A, the original fallback
>> mechanism already envisioned in RFC 2782.  However, it's time to move
>> forward and instead of extending a bad idea to v6 it should be 
>> terminated and
>> no "fallback" given.  Especially because this is not a legacy service 
>> that
>> existed before SRV was invented and RFC 3489 already included the SRV 
>> usage,
>> there's no need to address "old" v6 clients agnostic of SRV.  Operational
>> laziness shouldn't be encouraged.
>> At the very least, the fallback MUST be excluded for explicit "stop" SRV
>> RRs (those including the "." target).

Its not so much about backwards compatibility as domains that simply 
elect not to deploy SRV. Unfortunately we've seen lots of cases where 
folks just deploy A records and don't use SRV at all. The fallback makes 
the client maximally interopreable.

>>
>>>    addresses, each of which can be contacted at the default port using
>>>    UDP or TCP, independent of the STUN usage.  For usages that require
>>>    TLS, lack of SRV records is equivalent to a failure of the
>>>    transaction, since the request or indication MUST NOT be sent unless
>>>    SRV records provided a transport address specifically for TLS.
>>
>> In the Security Considerations section the usual DNS caveats and threats
>> should be mentioned, especially since SRV provides a level of indirection
>> that can't be protected by use of TLS (you might end up to securely talk
>> to the perfectly authenticated - albeit wrong - server).
>> Also, even though the mechanism of obtaining the server's domain name
>> is not specified in the draft, I'd suggest to mention the risk of 
>> hardcoding
>> the domain name in the application software because of the chance to
>> "loose" a domain name (by accidental expiry, legal action and so on).

In fact the text used to describe these standard DNS attacks. However 
the draft was restructured and broken into this core document and into 
usages. As the security considerations section discusses, usages 
actually need to do the analysis to see what attacks apply. For example 
one stun usage doesn't use the DNS mechanism at all and so the attack 
doesnt apply. ICE does use it, and the attack you mention is described 
there.

I mentioned that the domain name should not be hard coded.

>>
>> -Peter
>>
>> PS: Unrelated to DNS, aren't there privacy concerns worth mentioning
>>     because of the opportunity to monitor application usage?

STUN attacks whereby an attacker can intercept traffic to monitor it, 
are described as part of the usage. ICE disucsses this attack for example.

Thanks,
Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From: Carlos Pignataro <cpignata@cisco.com>
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Subject: Re: [BEHAVE] [Fwd: DISCUSS and COMMENT: draft-ietf-behave-nat-icmp]
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Magnus, Lars,

Please find my 2=C2=A2 in a couple of comments inline.

On 10/29/2007 5:06 AM, Magnus Westerlund said the following:
> Behave,
>=20
> I think the discuss from Lars below needs WG discussion. I like to here
> opinions about this.
>=20
> Cheers
>=20
> Magnus Westerlund
>=20
> IETF Transport Area Director & TSVWG Chair
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM/M
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
>=20
>=20
> -----------------------------------------------------------------------=
-
>=20
> Subject:
> DISCUSS and COMMENT: draft-ietf-behave-nat-icmp
> From:
> Lars Eggert <lars.eggert@nokia.com>
> Date:
> Sat, 27 Oct 2007 17:46:45 -0400
> To:
> iesg@ietf.org
>=20
> To:
> iesg@ietf.org
> CC:
> behave-chairs@tools.ietf.org, draft-ietf-behave-nat-icmp@tools.ietf.org
>=20
>=20
> Discuss:
> Section 4.1., paragraph 4:
>>    REQ-3:  When an ICMP Error packet is received, the NAT device
>>    SHOULD do the following.
>>    a) If the IP checksum of the embedded packet fails to validate, dro=
p
>>    the Error packet
>=20
>   DISCUSS: I disagree. (And I apologize for not catching this during my
>   earlier review.) The reason that (part of) the packet that caused an
>   ICMP message to be generated is included is for diagnostic purposes,
>   so the end hosts can determine what went wrong and determine whether
>   they need to react in some way.

This requirement is about the IP Header Checksum of the embedded packet,
and ICMPs should not be generated when the IP Header Checksum fails:

For hosts: RFC1122:
   3.2  PROTOCOL WALK-THROUGH
      3.2.1 Internet Protocol -- IP
         3.2.1.2  Checksum: RFC-791 Section 3.1
...
            A host MUST verify the IP header checksum on every received
            datagram and silently discard every datagram that has a bad
            checksum.

For routers: RFC1812:
5.2.2 IP Header Validation
...
   that the header is meaningful.  If the packet fails any of the
   following tests, it MUST be silently discarded, and the error SHOULD
   be logged.
...
   (2) The IP checksum must be correct.

Therefore, I don't disagree with this requirement.
If the IP Header Checksum is incorrect, and the NAT fixes-up the
embedded IP Address, how should it update the checksum?

> This can't happen if NATs drop these
>   packets. Also, the reason why some end system OSs drop ICMP messages
>   with invalid checksums is to protect against some attacks. NATs,
>   however, are not firewalls. It is inappropriate for this document to
>   specify NAT mechanisms that implement a specific security policy.
>   There is also no IETF standard that recommends that end hosts drop
>   ICMP packets with payloads that have invalid checksums, which is
>   another reason why this document should not recommend that NATs do
>   this. (And, as an aside, the effectiveness of this "security" measure
>   is questionable, because if I can't spoof the complete packet with a
>   valid checksum, I'd just include a partial packet in the payload, for
>   which you can't validate the checksum, but that you must still act
>   on.)

The REQ-3 a) quoted in the discuss refers to the IP Header Checksum of
the embedded packet. However this comment seems to be referring to REQ-3
d), i.e.:

   d) If the embedded packet contains the entire transport segment,
   and the transport protocol of the embedded packet requires the
   recipient to validate the transport checksum, and the checksum
   fails to validate, drop the Error packet.

In that case, I think a MAY for REQ-3 d) is likely more appropriate (to
leave some room for implementing it if desired), although another option
is  removing it as a requirement. I agree that the "protection"
effectiveness is questionable, but in any case it's a mechanism that I'd
think would kick in in very limited cases. I cannot think of a general
harmful scenario that would result for dropping those, but maybe
something that might not get widely implemented either. There seem to be
soem corner case scenarios. E.g.: this requirement may possibly mask a
bug in a host or NAT setting bad tcp checksums that trigger the
generation of ICMPs before the destination TCP gets to process the
transport checksum (e.g., unreachables, time exceeded, that also embed
the complete segment), and those ICMPs would not reach the sending host.
However, it those packets did reach the destination application, the
source would not receive a hint of what the problem is either.

Thanks,

--Carlos.

>=20
> Comment:
> Section 240, paragraph 0:
>>    ICMP Query session timer MUST not expire in less than 60 seconds. W=
e
>=20
>   s/MUST not/MUST NOT/
>=20
> Section 240, paragraph 1:
>>    RECOMMEND however that the administrator(s) be allowed to configure
>=20
>   RECOMMEND is not an RFC2119 term. You need to rephrase to say
>   RECOMMENDED, MUST or REQUIRED. (Or use lowercase here, because REQ-2
>   has the correct 2119 phrasing.)
>=20
>=20
>=20
>=20
>=20
>=20
> -----------------------------------------------------------------------=
-
>=20
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave

--=20
--Carlos Pignataro.
Escalation RTP - cisco Systems


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From: Pyda Srisuresh <srisuresh@yahoo.com>
Subject: Re: [BEHAVE] [Fwd: DISCUSS and COMMENT: draft-ietf-behave-nat-icmp]
To: Magnus Westerlund <magnus.westerlund@ericsson.com>,
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Lars and Magnus, 

Please see my responses inline.

regards,
suresh

--- Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:

> Behave,
> 
> I think the discuss from Lars below needs WG discussion. I like to here
> opinions about this.
> 
> Cheers
> 
> Magnus Westerlund
> 
> IETF Transport Area Director & TSVWG Chair
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVM/M
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------
> > To: iesg@ietf.org
> From: Lars Eggert <lars.eggert@nokia.com>
> Date: Sat, 27 Oct 2007 17:46:45 -0400
> CC: behave-chairs@tools.ietf.org, draft-ietf-behave-nat-icmp@tools.ietf.org
> Subject: DISCUSS and COMMENT: draft-ietf-behave-nat-icmp 
> 
> Discuss:
> Section 4.1., paragraph 4:
> >    REQ-3:  When an ICMP Error packet is received, the NAT device
> >    SHOULD do the following.
> >    a) If the IP checksum of the embedded packet fails to validate, drop
> >    the Error packet
> 
>   DISCUSS: I disagree. (And I apologize for not catching this during my
>   earlier review.) The reason that (part of) the packet that caused an
>   ICMP message to be generated is included is for diagnostic purposes,
>   so the end hosts can determine what went wrong and determine whether
>   they need to react in some way. This can't happen if NATs drop these
>   packets. 

[suresh] As Carlos responded on this e-mail thread, ICMP error messages should
not be generated by endhosts or router when the IP Header Checksum fails.
Packets with bad IP header checksum MUST be silently dropped by endhosts and
routers. The recommendation here is also to drop an ICMP error message if the
embedded IP Header Checksum is BAD.

>            Also, the reason why some end system OSs drop ICMP messages
>   with invalid checksums is to protect against some attacks. NATs,
>   however, are not firewalls. It is inappropriate for this document to
>   specify NAT mechanisms that implement a specific security policy.
>   There is also no IETF standard that recommends that end hosts drop
>   ICMP packets with payloads that have invalid checksums, which is
>   another reason why this document should not recommend that NATs do
>   this.

[Suresh] I Assume, you are refering to Req-3d here. We had extensive discussion
on this in the WG and in the end agreed to make this a SHOULD. Here are some
reasons I recall.

End-hosts should not be generating IP packets with BAD transport checksum.
Further, ICMP error messages are not generated due to BAD transport checksum in
the IP packet. So, the likely case where an ICMP error packet contains a BAD
embedded transport checksum is when it is spoofed. Also, many new
implemenations now check against BAD embedded trasnport checksum in ICMP error
messages (when the payload contains  requsite bytes to compute the checksum).
So, we decided to make this a SHOULD requirement to mitigate malformed ICMP
errors from reaching endhosts.



>        (And, as an aside, the effectiveness of this "security" measure
>   is questionable, because if I can't spoof the complete packet with a
>   valid checksum, I'd just include a partial packet in the payload, for
>   which you can't validate the checksum, but that you must still act
>   on.)

[Suresh] Right, checksum validation merely mitigates end hosts from malformed
ICMP Error attacks, doesnt eliminate it. The security consideratiosn section
also has a discussion on this.
 
> 
> Comment:
> Section 240, paragraph 0:
> >    ICMP Query session timer MUST not expire in less than 60 seconds. We
> 
>   s/MUST not/MUST NOT/
> 
[suresh] Will do.

> Section 240, paragraph 1:
> >    RECOMMEND however that the administrator(s) be allowed to configure
> 
>   RECOMMEND is not an RFC2119 term. You need to rephrase to say
>   RECOMMENDED, MUST or REQUIRED. (Or use lowercase here, because REQ-2
>   has the correct 2119 phrasing.)
> 
[suresh] OK. I will reword the sentence as follows. Thanks.

    It is RECOMMENDED however that the administrator(s) be allowed to
    configure the timer.

regards,
suresh

> 
> 
> 
> > _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
> 





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From behave-bounces@ietf.org Tue Oct 30 12:36:07 2007
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From: Lars Eggert <lars.eggert@nokia.com>
Subject: Re: [BEHAVE] [Fwd: DISCUSS and COMMENT: draft-ietf-behave-nat-icmp]
Date: Tue, 30 Oct 2007 18:34:24 +0200
To: ext Carlos Pignataro <cpignata@cisco.com>
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Hi,

you are right that I had been mixing two issues in my DISCUSS,. i.e.,  
what NATs should do when the IP checksum of the embedded packet is  
incorrect, and what they should do when the transport checksum is  
incorrect. Let's discuss these separately.

On 2007-10-29, at 17:13, ext Carlos Pignataro wrote:
>> Discuss:
>> Section 4.1., paragraph 4:
>>>    REQ-3:  When an ICMP Error packet is received, the NAT device
>>>    SHOULD do the following.
>>>    a) If the IP checksum of the embedded packet fails to  
>>> validate, drop
>>>    the Error packet
>>
>>   DISCUSS: I disagree. (And I apologize for not catching this  
>> during my
>>   earlier review.) The reason that (part of) the packet that  
>> caused an
>>   ICMP message to be generated is included is for diagnostic  
>> purposes,
>>   so the end hosts can determine what went wrong and determine  
>> whether
>>   they need to react in some way.
>
> This requirement is about the IP Header Checksum of the embedded  
> packet,
> and ICMPs should not be generated when the IP Header Checksum fails:

I agree that ICMPs should not be generated in response to receiving a  
packet with an invalid checksum.

But REQ-3 is about acting on the IP and transport checksums of  
packets that are embedded in the payload of an ICMP packet. RFCs 1122  
and 1812 don't apply to this.

> If the IP Header Checksum is incorrect, and the NAT fixes-up the
> embedded IP Address, how should it update the checksum?

Dropping them is one option.

Leave the checksum incorrect but still translating the addresses is  
another, and one that leaves the end host in the loop. It leaves the  
end host the choice on what to do, whereas dropping the packet  
removes that choice.

(Not translating the embedded IP addresses is a third option, but I'm  
not sure how useful that is.)

>> This can't happen if NATs drop these
>>   packets. Also, the reason why some end system OSs drop ICMP  
>> messages
>>   with invalid checksums is to protect against some attacks. NATs,
>>   however, are not firewalls. It is inappropriate for this  
>> document to
>>   specify NAT mechanisms that implement a specific security policy.
>>   There is also no IETF standard that recommends that end hosts drop
>>   ICMP packets with payloads that have invalid checksums, which is
>>   another reason why this document should not recommend that NATs do
>>   this. (And, as an aside, the effectiveness of this "security"  
>> measure
>>   is questionable, because if I can't spoof the complete packet  
>> with a
>>   valid checksum, I'd just include a partial packet in the  
>> payload, for
>>   which you can't validate the checksum, but that you must still act
>>   on.)
>
> The REQ-3 a) quoted in the discuss refers to the IP Header Checksum of
> the embedded packet. However this comment seems to be referring to  
> REQ-3
> d), i.e.:
>
>    d) If the embedded packet contains the entire transport segment,
>    and the transport protocol of the embedded packet requires the
>    recipient to validate the transport checksum, and the checksum
>    fails to validate, drop the Error packet.
>
> In that case, I think a MAY for REQ-3 d) is likely more appropriate  
> (to
> leave some room for implementing it if desired),

A MAY is not appropriate. No standards track RFC allows this (MAY) or  
recommends this (SHOULD/MUST) for end systems. So even a MAY for NATs  
goes beyond our current recommendations for end systems.

> although another option
> is  removing it as a requirement.

This would be my preferred resolution.

> I agree that the "protection"
> effectiveness is questionable, but in any case it's a mechanism  
> that I'd
> think would kick in in very limited cases.

So if you agree that the effectiveness is questionable and it may not  
kick in very often either, why burden NAT implementations with this  
functionality?

Lars


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From: Lars Eggert <lars.eggert@nokia.com>
Subject: Re: [BEHAVE] [Fwd: DISCUSS and COMMENT: draft-ietf-behave-nat-icmp]
Date: Tue, 30 Oct 2007 18:38:58 +0200
To: ext Pyda Srisuresh <srisuresh@yahoo.com>
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On 2007-10-30, at 17:46, ext Pyda Srisuresh wrote:
> [suresh] As Carlos responded on this e-mail thread, ICMP error  
> messages should
> not be generated by endhosts or router when the IP Header Checksum  
> fails.
> Packets with bad IP header checksum MUST be silently dropped by  
> endhosts and
> routers. The recommendation here is also to drop an ICMP error  
> message if the
> embedded IP Header Checksum is BAD.

As I said in my earlier reply to Carlos, the 1122 and 1812  
recommendations aren't about IP headers embedded in ICMP payloads.

> [Suresh] I Assume, you are refering to Req-3d here. We had  
> extensive discussion
> on this in the WG and in the end agreed to make this a SHOULD. Here  
> are some
> reasons I recall.
>
> End-hosts should not be generating IP packets with BAD transport  
> checksum.
> Further, ICMP error messages are not generated due to BAD transport  
> checksum in
> the IP packet. So, the likely case where an ICMP error packet  
> contains a BAD
> embedded transport checksum is when it is spoofed.

No. The likely case is flaky hardware that introduces bit errors.

> Also, many new
> implemenations now check against BAD embedded trasnport checksum in  
> ICMP error
> messages (when the payload contains  requsite bytes to compute the  
> checksum).
> So, we decided to make this a SHOULD requirement to mitigate  
> malformed ICMP
> errors from reaching endhosts.

NATs are not firewalls. It is not their task to protect the end  
systems from anything.

Lars


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On 10/30/2007 12:34 PM, Lars Eggert said the following:
> Hi,
> 
> you are right that I had been mixing two issues in my DISCUSS,. i.e.,  
> what NATs should do when the IP checksum of the embedded packet is  
> incorrect, and what they should do when the transport checksum is  
> incorrect. Let's discuss these separately.
> 
> On 2007-10-29, at 17:13, ext Carlos Pignataro wrote:
>>> Discuss:
>>> Section 4.1., paragraph 4:
>>>>    REQ-3:  When an ICMP Error packet is received, the NAT device
>>>>    SHOULD do the following.
>>>>    a) If the IP checksum of the embedded packet fails to  
>>>> validate, drop
>>>>    the Error packet
>>>   DISCUSS: I disagree. (And I apologize for not catching this  
>>> during my
>>>   earlier review.) The reason that (part of) the packet that  
>>> caused an
>>>   ICMP message to be generated is included is for diagnostic  
>>> purposes,
>>>   so the end hosts can determine what went wrong and determine  
>>> whether
>>>   they need to react in some way.
>> This requirement is about the IP Header Checksum of the embedded  
>> packet,
>> and ICMPs should not be generated when the IP Header Checksum fails:
> 
> I agree that ICMPs should not be generated in response to receiving a  
> packet with an invalid checksum.
> 
> But REQ-3 is about acting on the IP and transport checksums of  
> packets that are embedded in the payload of an ICMP packet. RFCs 1122  
> and 1812 don't apply to this.

IMHO, the question is: under which legitimate circumstances can an IP
datagram with invalid Internet Header Checksum generate an ICMP error,
such that it gets embedded in the ICMP? RFC1122 and 1812 do apply in the
processing of a packet with invalid checksum, and if they need to be
silently dropped, how can they get embedded into ICMPs? So the citations
are in the ICMP-eliciting path. The only option (besides bugs) appears
to be that the IP Hdr checksum got corrupted after the IP Hdr got
embedded on the icmp and it's on its way back, in which case the ICMP
checksum would fail and the "corrupted" header won't be processed by the
originating host anyway, correct?

> 
>> If the IP Header Checksum is incorrect, and the NAT fixes-up the
>> embedded IP Address, how should it update the checksum?
> 
> Dropping them is one option.
> 
> Leave the checksum incorrect but still translating the addresses is  
> another, and one that leaves the end host in the loop. It leaves the  
> end host the choice on what to do, whereas dropping the packet  
> removes that choice.

It's not clear to me that there's a lot of meaningful information that
the host would want to be in the loop for in these cases (see the
previous para).

> 
> (Not translating the embedded IP addresses is a third option, but I'm  
> not sure how useful that is.)

But this one could potentially expose cross-realm addresses.

> 
>>> This can't happen if NATs drop these
>>>   packets. Also, the reason why some end system OSs drop ICMP  
>>> messages
>>>   with invalid checksums is to protect against some attacks. NATs,
>>>   however, are not firewalls. It is inappropriate for this  
>>> document to
>>>   specify NAT mechanisms that implement a specific security policy.
>>>   There is also no IETF standard that recommends that end hosts drop
>>>   ICMP packets with payloads that have invalid checksums, which is
>>>   another reason why this document should not recommend that NATs do
>>>   this. (And, as an aside, the effectiveness of this "security"  
>>> measure
>>>   is questionable, because if I can't spoof the complete packet  
>>> with a
>>>   valid checksum, I'd just include a partial packet in the  
>>> payload, for
>>>   which you can't validate the checksum, but that you must still act
>>>   on.)
>> The REQ-3 a) quoted in the discuss refers to the IP Header Checksum of
>> the embedded packet. However this comment seems to be referring to  
>> REQ-3
>> d), i.e.:
>>
>>    d) If the embedded packet contains the entire transport segment,
>>    and the transport protocol of the embedded packet requires the
>>    recipient to validate the transport checksum, and the checksum
>>    fails to validate, drop the Error packet.
>>
>> In that case, I think a MAY for REQ-3 d) is likely more appropriate  
>> (to
>> leave some room for implementing it if desired),
> 
> A MAY is not appropriate. No standards track RFC allows this (MAY) or  
> recommends this (SHOULD/MUST) for end systems. So even a MAY for NATs  
> goes beyond our current recommendations for end systems.

lowercase "might"? :) See below.

> 
>> although another option
>> is  removing it as a requirement.
> 
> This would be my preferred resolution.
> 
>> I agree that the "protection"
>> effectiveness is questionable, but in any case it's a mechanism  
>> that I'd
>> think would kick in in very limited cases.
> 
> So if you agree that the effectiveness is questionable and it may not  
> kick in very often either, why burden NAT implementations with this  
> functionality?

Because it appears the WG found consensus around leaving it as something
potentially (or under some cases) useful, and it appears to be something
that a "strict policy" might want, I think that leaving room for
implementing it (while not requiring) is of value.

Thanks,

--Carlos.

> 
> Lars
> 

-- 
--Carlos Pignataro.
Escalation RTP - cisco Systems


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From: Lars Eggert <lars.eggert@nokia.com>
Subject: Re: [BEHAVE] [Fwd: DISCUSS and COMMENT: draft-ietf-behave-nat-icmp]
Date: Wed, 31 Oct 2007 10:30:31 +0200
To: ext Carlos Pignataro <cpignata@cisco.com>
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On 2007-10-30, at 19:29, ext Carlos Pignataro wrote:
> On 10/30/2007 12:34 PM, Lars Eggert said the following:
>> I agree that ICMPs should not be generated in response to receiving a
>> packet with an invalid checksum.
>>
>> But REQ-3 is about acting on the IP and transport checksums of
>> packets that are embedded in the payload of an ICMP packet. RFCs 1122
>> and 1812 don't apply to this.
>
> IMHO, the question is: under which legitimate circumstances can an IP
> datagram with invalid Internet Header Checksum generate an ICMP error,
> such that it gets embedded in the ICMP? RFC1122 and 1812 do apply  
> in the
> processing of a packet with invalid checksum, and if they need to be
> silently dropped, how can they get embedded into ICMPs?

You assume that an ICMP message that contains an embedded IP packet  
with an invalid checksum was generated somewhere in response to  
receiving this IP packet with an invalid header checksum.

It is equally possible that the ICMP message was generated based on  
an IP packet with a valid checksum, but then got corrupted during  
transmission.

> So the citations
> are in the ICMP-eliciting path. The only option (besides bugs) appears
> to be that the IP Hdr checksum got corrupted after the IP Hdr got
> embedded on the icmp and it's on its way back, in which case the ICMP
> checksum would fail and the "corrupted" header won't be processed  
> by the
> originating host anyway, correct?

The ICMP checksum only covers the ICMP header and not the payload.  
(Unless I'm seriously misreading RFC792.)

>> Leave the checksum incorrect but still translating the addresses is
>> another, and one that leaves the end host in the loop. It leaves the
>> end host the choice on what to do, whereas dropping the packet
>> removes that choice.
>
> It's not clear to me that there's a lot of meaningful information that
> the host would want to be in the loop for in these cases (see the
> previous para).

It's not obvious to me that hosts wouldn't. Not dropping these  
packets leaves them the option to choose.

>> (Not translating the embedded IP addresses is a third option, but I'm
>> not sure how useful that is.)
>
> But this one could potentially expose cross-realm addresses.

Right, that's why it's not useful.

Let me try to make my point with a different argument: What would a  
plain RFC1812 router do, if it was in the path instead of a NAT? A  
router would simply forward the ICMP packet, irrespective of whether  
the IP checksum of a packet embedded on an ICMP message was valid or  
not. A NAT should do exactly the same, in my opinion, i.e., it should  
forward the packet whenever it can. So if it has a binding that  
allows it to translate the embedded IP header, it should always do  
so, even if the embedded checksum is invalid.

>>>> The REQ-3 a) quoted in the discuss refers to the IP Header  
>>>> Checksum of
>>> the embedded packet. However this comment seems to be referring to
>>> REQ-3
>>> d), i.e.:
>>>
>>>    d) If the embedded packet contains the entire transport segment,
>>>    and the transport protocol of the embedded packet requires the
>>>    recipient to validate the transport checksum, and the checksum
>>>    fails to validate, drop the Error packet.
>>>
>>> In that case, I think a MAY for REQ-3 d) is likely more appropriate
>>> (to leave some room for implementing it if desired),
>>
>> A MAY is not appropriate. No standards track RFC allows this (MAY) or
>> recommends this (SHOULD/MUST) for end systems. So even a MAY for NATs
>> goes beyond our current recommendations for end systems.
>
> lowercase "might"? :) See below.

I wouldn't even want to allow that.

>>> although another option
>>> is  removing it as a requirement.
>>
>> This would be my preferred resolution.
>>
>>> I agree that the "protection"
>>> effectiveness is questionable, but in any case it's a mechanism
>>> that I'd
>>> think would kick in in very limited cases.
>>
>> So if you agree that the effectiveness is questionable and it may not
>> kick in very often either, why burden NAT implementations with this
>> functionality?
>
> Because it appears the WG found consensus around leaving it as  
> something
> potentially (or under some cases) useful, and it appears to be  
> something
> that a "strict policy" might want, I think that leaving room for
> implementing it (while not requiring) is of value.

BEHAVE isn't chartered to design security mechanisms ("strict  
policy".) I'm not saying that that shouldn't be done, but BEHAVE  
under the current charter isn't the WG to do it.

Lars


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From behave-bounces@ietf.org Wed Oct 31 09:07:43 2007
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From: Philip Matthews <philip_matthews@magma.ca>
Subject: Re: [BEHAVE] TURN : How can a client ask a TCP disconnection ?
Date: Wed, 31 Oct 2007 09:06:49 -0400
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On 26-Oct-07, at 03:53 , R=E9mi Denis-Courmont wrote:

> Le Friday 26 October 2007 10:38:50 ext Ga=EBl Br=E9ard, vous avez =
=E9crit :
>> One simple question about TURN :
>>
>> Imagine the case where a TURN client has allocated a TCP address =20
>> (a TCP
>> server) on a TURN server.
>> Several remote peers have established a connection on this =20
>> allocated port.
>>
>> And now, the client wants to disconnect one of them (and only one).
>> I did not found any request to do that.
>
> It used to be so that you terminated the TCP connection. Multiple TCP
> connections per TURN session was then added, but remains buggy to =20
> this day.
>
> As far as I understand, the current public draft is way out-of-sync =20=

> with where
> the WG is heading anyway.
>
> --=20
> R=E9mi Denis-Courmont

Ga=EBl:

Thanks for bring this up.
Remi is right in that TURN is undergoing major changes right now. =20
There is a
design team that talks by conference call twice a week. We were =20
hoping to have
a draft out by Oct 1, but that proved too ambitious with all the =20
concerns that
people had, and so we are now just trying to get something out before =20=

the
next meeting. The problem is that people are now starting to look at =20
ICE/STUN/TURN
as a general NAT traversal solution, and solutions that work well for =20=

SIP are
not always general enough.
For example, for SIP one can say that in most cases, there is only one
remote destination that one exchanges media with, and in the few =20
cases this is
not true, it is reasonable to require the client to obtain a second =20
TURN address.
However, this may or may not be a reasonable assumption in the =20
general case.

So if you could wait just a few more weeks ...

- Philip



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From: Pyda Srisuresh <srisuresh@yahoo.com>
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--- Lars Eggert <lars.eggert@nokia.com> wrote:

> On 2007-10-30, at 19:29, ext Carlos Pignataro wrote:
> > On 10/30/2007 12:34 PM, Lars Eggert said the following:
> >> I agree that ICMPs should not be generated in response to receiving a
> >> packet with an invalid checksum.
> >>
> >> But REQ-3 is about acting on the IP and transport checksums of
> >> packets that are embedded in the payload of an ICMP packet. RFCs 1122
> >> and 1812 don't apply to this.
> >
> > IMHO, the question is: under which legitimate circumstances can an IP
> > datagram with invalid Internet Header Checksum generate an ICMP error,
> > such that it gets embedded in the ICMP? RFC1122 and 1812 do apply  
> > in the
> > processing of a packet with invalid checksum, and if they need to be
> > silently dropped, how can they get embedded into ICMPs?
> 
> You assume that an ICMP message that contains an embedded IP packet  
> with an invalid checksum was generated somewhere in response to  
> receiving this IP packet with an invalid header checksum.
> 
> It is equally possible that the ICMP message was generated based on  
> an IP packet with a valid checksum, but then got corrupted during  
> transmission.

[suresh] We all agree that IP packets with BAD IP header checksum are silently
dropped by end-hosts and routers. ICMP errors are not generated by end-hosts or
routers in response to IP packets with bad IP header checksums. 

In that case, how could the embedded IP header checksum have been corrupted?
You say, one possibility is corruption during transmission. Corruption of which
specific bits in the IP header? It could have been transmission corruption on
any part of the IP header - Src/Dest IP addresses, TTL, Length, Checksum etc.
We cannot assume that every other part of the IP header may be OK, but checksum
alone may have gotten corrupted. In other words, if the IP header checksum is
corrupted, the entire IP header is suspect.  That is the gist of recommendation
for REQ-3a. 

Unlike a router that does not examine the ICMP error payload, NAT examines the
embedded payload for session match and modify IP addresses and fixes checksum
before forwarding the IP packet to the intended recipient. In order for a NAT
to make the mods, NAT needs to start with a good base. Modifying an IP packet,
fully knowing that it's header is suspect, could result in bombarding the wrong
end-host with the ICMP error message.

It woudl be derelict to not drop suspect IP packets for a device that examines
the payload and detects checksum failure. I do believe, the current
recommendation of SHOULD as Best Current Practice is the right thing to do. 

> 
> > So the citations
> > are in the ICMP-eliciting path. The only option (besides bugs) appears
> > to be that the IP Hdr checksum got corrupted after the IP Hdr got
> > embedded on the icmp and it's on its way back, in which case the ICMP
> > checksum would fail and the "corrupted" header won't be processed  
> > by the
> > originating host anyway, correct?
> 
> The ICMP checksum only covers the ICMP header and not the payload.  
> (Unless I'm seriously misreading RFC792.)

[suresh] I do believe, the ICMP checksum covers the entire ICMP message
includign the payload.
> 
> >> Leave the checksum incorrect but still translating the addresses is
> >> another, and one that leaves the end host in the loop. It leaves the
> >> end host the choice on what to do, whereas dropping the packet
> >> removes that choice.
> >
> > It's not clear to me that there's a lot of meaningful information that
> > the host would want to be in the loop for in these cases (see the
> > previous para).
> 
> It's not obvious to me that hosts wouldn't. Not dropping these  
> packets leaves them the option to choose.
> 
> >> (Not translating the embedded IP addresses is a third option, but I'm
> >> not sure how useful that is.)
> >
> > But this one could potentially expose cross-realm addresses.
> 
> Right, that's why it's not useful.
> 
> Let me try to make my point with a different argument: What would a  
> plain RFC1812 router do, if it was in the path instead of a NAT? A  
> router would simply forward the ICMP packet, irrespective of whether  
> the IP checksum of a packet embedded on an ICMP message was valid or  
> not. A NAT should do exactly the same, in my opinion, i.e., it should  
> forward the packet whenever it can. So if it has a binding that  
> allows it to translate the embedded IP header, it should always do  
> so, even if the embedded checksum is invalid.
> 

[suresh] Please see my comment above. NAT does a combination of router and
end-host functionality. Unlike a router, it is a requirement for NAT to examine
the ICMP error payload in order to determine where to forward the ICMP error
packet. If the enbedded IP header is suspect, NAT cannot do its job right.

> >>>> The REQ-3 a) quoted in the discuss refers to the IP Header  
> >>>> Checksum of
> >>> the embedded packet. However this comment seems to be referring to
> >>> REQ-3
> >>> d), i.e.:
> >>>
> >>>    d) If the embedded packet contains the entire transport segment,
> >>>    and the transport protocol of the embedded packet requires the
> >>>    recipient to validate the transport checksum, and the checksum
> >>>    fails to validate, drop the Error packet.
> >>>
> >>> In that case, I think a MAY for REQ-3 d) is likely more appropriate
> >>> (to leave some room for implementing it if desired),
> >>
> >> A MAY is not appropriate. No standards track RFC allows this (MAY) or
> >> recommends this (SHOULD/MUST) for end systems. So even a MAY for NATs
> >> goes beyond our current recommendations for end systems.
> >
> > lowercase "might"? :) See below.
> 
> I wouldn't even want to allow that.
> 
> >>> although another option
> >>> is  removing it as a requirement.
> >>
> >> This would be my preferred resolution.
> >>
> >>> I agree that the "protection"
> >>> effectiveness is questionable, but in any case it's a mechanism
> >>> that I'd
> >>> think would kick in in very limited cases.
> >>
> >> So if you agree that the effectiveness is questionable and it may not
> >> kick in very often either, why burden NAT implementations with this
> >> functionality?
> >
> > Because it appears the WG found consensus around leaving it as  
> > something
> > potentially (or under some cases) useful, and it appears to be  
> > something
> > that a "strict policy" might want, I think that leaving room for
> > implementing it (while not requiring) is of value.
> 
> BEHAVE isn't chartered to design security mechanisms ("strict  
> policy".) I'm not saying that that shouldn't be done, but BEHAVE  
> under the current charter isn't the WG to do it.
>

[suresh] Lars - This is not one of designing a security mechanism within BCP.
This is a NAT functionality requirement. NAT needs to examine the IP and
transport headers in order to make decisions concerning translation and
forwarding. When the IP and transport checksums are BAD, the corresponding IP
and transport headers are suspect. When the IP and trasport headers are
suspect, NAT does not have a good base to start from. I hope, you agree, it is
not advisible to do lookups or translations knowing fully well the base headers
are suspect. Hence, the recommendations REQ-3a, 3d. Thanks.

regards,
suresh

> Lars
> 
> > _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
> 





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Date: Wed, 31 Oct 2007 11:09:29 -0400
From: Carlos Pignataro <cpignata@cisco.com>
Organization: cisco Systems, Inc.
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Subject: Re: [BEHAVE] [Fwd: DISCUSS and COMMENT: draft-ietf-behave-nat-icmp]
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Lars,

Thanks for the follow-up, please see some comments inline (separating
Req3-a from Req3-d)

On 10/31/2007 4:30 AM, Lars Eggert said the following:
> On 2007-10-30, at 19:29, ext Carlos Pignataro wrote:
>> On 10/30/2007 12:34 PM, Lars Eggert said the following:
>>> I agree that ICMPs should not be generated in response to receiving a
>>> packet with an invalid checksum.
>>>
>>> But REQ-3 is about acting on the IP and transport checksums of
>>> packets that are embedded in the payload of an ICMP packet. RFCs 1122
>>> and 1812 don't apply to this.
>> IMHO, the question is: under which legitimate circumstances can an IP
>> datagram with invalid Internet Header Checksum generate an ICMP error,
>> such that it gets embedded in the ICMP? RFC1122 and 1812 do apply  
>> in the
>> processing of a packet with invalid checksum, and if they need to be
>> silently dropped, how can they get embedded into ICMPs?
> 
> You assume that an ICMP message that contains an embedded IP packet  
> with an invalid checksum was generated somewhere in response to  
> receiving this IP packet with an invalid header checksum.
> 
> It is equally possible that the ICMP message was generated based on  
> an IP packet with a valid checksum, but then got corrupted during  
> transmission.

Yes, as noted below.

> 
>> So the citations
>> are in the ICMP-eliciting path. The only option (besides bugs) appears
>> to be that the IP Hdr checksum got corrupted after the IP Hdr got
>> embedded on the icmp and it's on its way back, in which case the ICMP
>> checksum would fail and the "corrupted" header won't be processed  
>> by the
>> originating host anyway, correct?
> 
> The ICMP checksum only covers the ICMP header and not the payload.  
> (Unless I'm seriously misreading RFC792.)

No, the ICMP checksum covers the complete ICMP message starting from the
ICMP Type. It does not include a pseudo-header (unlike ICMPv6). For ICMP
errors, the checksum includes the embedded packet. As such, the host
receiving the ICMP would not get to processing the "corrupted" header.

> 
>>> Leave the checksum incorrect but still translating the addresses is
>>> another, and one that leaves the end host in the loop. It leaves the
>>> end host the choice on what to do, whereas dropping the packet
>>> removes that choice.
>> It's not clear to me that there's a lot of meaningful information that
>> the host would want to be in the loop for in these cases (see the
>> previous para).
> 
> It's not obvious to me that hosts wouldn't. Not dropping these  
> packets leaves them the option to choose.

It is not clear to me under which conditions it would, and that the host
would actually get to processing the embedded info.

> 
>>> (Not translating the embedded IP addresses is a third option, but I'm
>>> not sure how useful that is.)
>> But this one could potentially expose cross-realm addresses.
> 
> Right, that's why it's not useful.
> 
> Let me try to make my point with a different argument: What would a  
> plain RFC1812 router do, if it was in the path instead of a NAT? A  
> router would simply forward the ICMP packet, irrespective of whether  
> the IP checksum of a packet embedded on an ICMP message was valid or  
> not. A NAT should do exactly the same, in my opinion, i.e., it should  
> forward the packet whenever it can. So if it has a binding that  
> allows it to translate the embedded IP header, it should always do  
> so, even if the embedded checksum is invalid.

I don't think that this comparison is valid, because an RFC1812 router
does not examine and modify the ICMP payload to perform its function
(and as was noted on this list, from the outside network's perspective a
NAT is more host-like). However, one additional viewpoint, if the
embedded IP Header checksum is invalid, and a binding is found, do you
think the NAT should trust and act upon that binding (given that the
embedded IP addresses, IP protocol numbers can be corrupt)? Do you think
it should recalculate a correct checksum? IMHO, a SHOULD in Req3-a is
useful.

> 
>>>>> The REQ-3 a) quoted in the discuss refers to the IP Header  
>>>>> Checksum of
>>>> the embedded packet. However this comment seems to be referring to
>>>> REQ-3
>>>> d), i.e.:
>>>>
>>>>    d) If the embedded packet contains the entire transport segment,
>>>>    and the transport protocol of the embedded packet requires the
>>>>    recipient to validate the transport checksum, and the checksum
>>>>    fails to validate, drop the Error packet.
>>>>
>>>> In that case, I think a MAY for REQ-3 d) is likely more appropriate
>>>> (to leave some room for implementing it if desired),
>>> A MAY is not appropriate. No standards track RFC allows this (MAY) or
>>> recommends this (SHOULD/MUST) for end systems. So even a MAY for NATs
>>> goes beyond our current recommendations for end systems.
>> lowercase "might"? :) See below.
> 
> I wouldn't even want to allow that.
> 
>>>> although another option
>>>> is  removing it as a requirement.
>>> This would be my preferred resolution.
>>>
>>>> I agree that the "protection"
>>>> effectiveness is questionable, but in any case it's a mechanism
>>>> that I'd
>>>> think would kick in in very limited cases.
>>> So if you agree that the effectiveness is questionable and it may not
>>> kick in very often either, why burden NAT implementations with this
>>> functionality?
>> Because it appears the WG found consensus around leaving it as  
>> something
>> potentially (or under some cases) useful, and it appears to be  
>> something
>> that a "strict policy" might want, I think that leaving room for
>> implementing it (while not requiring) is of value.
> 
> BEHAVE isn't chartered to design security mechanisms ("strict  
> policy".) I'm not saying that that shouldn't be done, but BEHAVE  
> under the current charter isn't the WG to do it.

I was thinking in terms of "reliability" rather than "security". If the
embedded L4 checksum is incorrect, the L4 ports or pseudo-headers may be
incorrect, resulting in a mis-binding. I'd think that the NAT could not
reliably perform its function (or could mis-associate). OTOH, I agree
that there may be cases in which a host may want to see these. IMHO, the
text of Req3-d is useful, and it should be a recommendation; the level
of recommendation IMHO could be either MAY or SHOULD (though SHOULD may
be a bit too strong here).

Thanks,

--Carlos.

> 
> Lars
> 

-- 
--Carlos Pignataro.
Escalation RTP - cisco Systems


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