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From: Pekka Savola <pekkas@netcore.fi>
To: Dan Wing <dwing@cisco.com>
Subject: RE: [BEHAVE] Re: CPE equipments and stateful filters
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On Tue, 31 Jul 2007, Dan Wing wrote:
>> Neither of these two
>> forms of traffic could reasonably be described as "unsolicited" in
>> this case.
>
> We have different definitions of 'solicited traffic' and
> 'unsolicited traffic'.  Other than opening a listener on UDP/500
> for IKE (and telling the upstream firewall to please open a
> permission for incoming traffic), the host didn't do anything to
> solicit an incoming IKE packet.  I call that incoming traffic
> 'unsolicited', because the host (and its firewall) are unaware
> of when a client will send a packet to establish a connection.
> An HTTP server or SMTP server fall into this category, for
> example.
>
> This differs from the SIP model and the RTSP 2.0 model.  In
> both SIP and RTSP 2.0, an offer/answer exchange allows the
> server to learn the client desires a new session.  With that
> knowledge, the server can communicate more specific information
> to its firewall about the client's IP address; in this way, the
> firewall doens't need to necessarily permit incoming IKE packets
> from arbitrary remote addresses, unless those are first signaled
> with SIP or RTSP.  This model is also used by HIP and EME, two
> working groups in the IRTF (www.irtf.org).  This is what I would
> consider 'solicited traffic'.

I'm not sure if this distinction matters that much from the security 
perspective.  IMHO the main point is the attack vector and effect. 
Even though SIP and RTSP 2.0 were offer/answer protocols, the main 
point is that there needs to be a service listening (to the whole 
world) on some ports, the same as with IKE.  You're just shifting the 
attack vector from IKE to SIP/RTSP 2.0 which in and of itself doesn't 
help anything; in fact it could make the situation worse, given that 
(AFAIK) the SIP/RTSP 2.0 protocol suite is more complicated than IKE.

The critical question then is, how much (and how complex) code in the 
listening service is required to make a decision whether to drop the 
new session or allow it to be established.  In other words, if a 
vulnerability is found in the listening service software, can it be 
exploited by unauthenticated/unauthorized attackers and can it be used 
for compromise (e.g., remote code execution)?  (I exclude DoS 
vulnerabilities from the interesting ones in this context.)

This also affects protocol design, i.e., the protocols should be 
designed so that the amount and complexity of code needed to drop 
uninteresting connection requests can be minimal.

There have been security vulnerabilities in both IKE and SIP protocol 
suites, but I do not recall outright whether such have enabled 
unauthenticated remote code execution exploits.  If such were to be 
the case, or such vulnerabilities would have a high likelihood of 
succeeding, I'd think that most security administrators would want to 
filter out both IKE and SIP/RTSP 2.0.

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


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From: "Dan Wing" <dwing@cisco.com>
To: "'Pekka Savola'" <pekkas@netcore.fi>
Subject: RE: [BEHAVE] Re: CPE equipments and stateful filters
Date: Tue, 31 Jul 2007 22:33:41 -0700
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Your email and Christian's email are both concerned with end-host
protection.  I am concerned with protecting the access link from
unwanted traffic to save bandwidth and, for battery-powered devices,
to save their batteries.  If the device isn't listening on UDP/500
and would merely throw away or ICMP the incoming packet, there are
advantages with bandwidth-constrained access links and with wireless
devices, to discard the packet in the network (in a firewall device).

-d

> -----Original Message-----
> From: Pekka Savola [mailto:pekkas@netcore.fi] 
> Sent: Tuesday, July 31, 2007 10:05 PM
> To: Dan Wing
> Cc: 'james woodyatt'; 'IPv6 Operations'; 'Behave WG'
> Subject: RE: [BEHAVE] Re: CPE equipments and stateful filters
> 
> On Tue, 31 Jul 2007, Dan Wing wrote:
> >> Neither of these two
> >> forms of traffic could reasonably be described as "unsolicited" in
> >> this case.
> >
> > We have different definitions of 'solicited traffic' and
> > 'unsolicited traffic'.  Other than opening a listener on UDP/500
> > for IKE (and telling the upstream firewall to please open a
> > permission for incoming traffic), the host didn't do anything to
> > solicit an incoming IKE packet.  I call that incoming traffic
> > 'unsolicited', because the host (and its firewall) are unaware
> > of when a client will send a packet to establish a connection.
> > An HTTP server or SMTP server fall into this category, for
> > example.
> >
> > This differs from the SIP model and the RTSP 2.0 model.  In
> > both SIP and RTSP 2.0, an offer/answer exchange allows the
> > server to learn the client desires a new session.  With that
> > knowledge, the server can communicate more specific information
> > to its firewall about the client's IP address; in this way, the
> > firewall doens't need to necessarily permit incoming IKE packets
> > from arbitrary remote addresses, unless those are first signaled
> > with SIP or RTSP.  This model is also used by HIP and EME, two
> > working groups in the IRTF (www.irtf.org).  This is what I would
> > consider 'solicited traffic'.
> 
> I'm not sure if this distinction matters that much from the security 
> perspective.  IMHO the main point is the attack vector and effect. 
> Even though SIP and RTSP 2.0 were offer/answer protocols, the main 
> point is that there needs to be a service listening (to the whole 
> world) on some ports, the same as with IKE.  You're just shifting the 
> attack vector from IKE to SIP/RTSP 2.0 which in and of itself doesn't 
> help anything; in fact it could make the situation worse, given that 
> (AFAIK) the SIP/RTSP 2.0 protocol suite is more complicated than IKE.
> 
> The critical question then is, how much (and how complex) code in the 
> listening service is required to make a decision whether to drop the 
> new session or allow it to be established.  In other words, if a 
> vulnerability is found in the listening service software, can it be 
> exploited by unauthenticated/unauthorized attackers and can 
> it be used 
> for compromise (e.g., remote code execution)?  (I exclude DoS 
> vulnerabilities from the interesting ones in this context.)
> 
> This also affects protocol design, i.e., the protocols should be 
> designed so that the amount and complexity of code needed to drop 
> uninteresting connection requests can be minimal.
> 
> There have been security vulnerabilities in both IKE and SIP protocol 
> suites, but I do not recall outright whether such have enabled 
> unauthenticated remote code execution exploits.  If such were to be 
> the case, or such vulnerabilities would have a high likelihood of 
> succeeding, I'd think that most security administrators would want to 
> filter out both IKE and SIP/RTSP 2.0.
> 
> -- 
> Pekka Savola                 "You each name yourselves king, yet the
> Netcore Oy                    kingdom bleeds."
> Systems. Networks. Security. -- George R.R. Martin: A Clash of Kings


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Subject: RE: [BEHAVE] Re: CPE equipments and stateful filters
Date: Wed, 1 Aug 2007 00:03:36 -0700
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	<001401c7d3fd$8577b530$c5f0200a@amer.cisco.com>
From: Christian Huitema <huitema@windows.microsoft.com>
To: Dan Wing <dwing@cisco.com>, Pekka Savola <pekkas@netcore.fi>
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> From: Dan Wing [mailto:dwing@cisco.com]
> Sent: Tuesday, July 31, 2007 10:34 PM
> To: 'Pekka Savola'
> Cc: 'IPv6 Operations'; 'Behave WG'
> Subject: RE: [BEHAVE] Re: CPE equipments and stateful filters
>=20
> Your email and Christian's email are both concerned with end-host
> protection.  I am concerned with protecting the access link from
> unwanted traffic to save bandwidth and, for battery-powered devices,
> to save their batteries.  If the device isn't listening on UDP/500
> and would merely throw away or ICMP the incoming packet, there are
> advantages with bandwidth-constrained access links and with wireless
> devices, to discard the packet in the network (in a firewall device).

If we followed your reasoning, cell phones should never be allowed to
ring, because doing so might deplete the batteries. They would only be
allowed to make outgoing calls...

There are many other ways to resolve this issue for the small battery
powered devices. For example, they may be programmed to only listen to
local addresses, which would radically solve the issue. They may also
keep their address "hidden", so as to not receive traffic.

There are also some better ways to answer to DOS attacks in a firewall
than blocking traffic outright. If the concern really is the batteries,
then rate limiting comes to mind. For example, if a device receives a
packet and does not respond, or send an ICMP, block the port for some
period.=20

-- Christian Huitema




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On 7/31/07, R=E9mi Denis-Courmont <remi.denis-courmont@nokia.com> wrote:
> At this point, the server really can only do one of three things:
> (1) queue an indefinitely big amount of data,
> (2) stop dequeing data from the TURN client until TCP_B gets "decongested=
",
> (3) drop data for B, and reset/timeout connection with B.

> (2) is flawed, as communication between client and peer A will be unduly
> disrupted. It is an obvious security issue if A and B do not trust each
> others. Even then, it is probably unacceptable: if say, B looses connecti=
vity
> for some reason, the flow to A will be totally disrupted until TCP_B
> connection times out. There are similar problem if connectivity to B is m=
uch
> slower or less reliable than that to A.

I think that 2 is the only possible "correct" behavior, because 1 is
unimplementable and 3 breaks the basic TCP model merely because one
destination has lower throughput than another.  2's problem is that it
restricts both connections to the poorer of the pair (whether
deliberate attack or poor provisioning), but at least they both behave
like valid TCP connections.

There's another possibility of making the protocol more complex by
signalling queueing on the client connection, but I really don't think
we want that.

But unless there's a compelling reason to support it, I think it would
be better if TURN just says one peer per client connection.  I'm not
aware of a compelling reason.  If there is a compelling reason to
support it, I think big red flags and possibly even SHOULD NOT
language should be used to discourage its use except when necessary.

Bruce


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To: "'Christian Huitema'" <huitema@windows.microsoft.com>,
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Subject: RE: [BEHAVE] Re: CPE equipments and stateful filters
Date: Wed, 1 Aug 2007 16:26:18 -0700
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> > From: Dan Wing [mailto:dwing@cisco.com]
> > Sent: Tuesday, July 31, 2007 10:34 PM
> > To: 'Pekka Savola'
> > Cc: 'IPv6 Operations'; 'Behave WG'
> > Subject: RE: [BEHAVE] Re: CPE equipments and stateful filters
> > 
> > Your email and Christian's email are both concerned with end-host
> > protection.  I am concerned with protecting the access link from
> > unwanted traffic to save bandwidth and, for battery-powered devices,
> > to save their batteries.  If the device isn't listening on UDP/500
> > and would merely throw away or ICMP the incoming packet, there are
> > advantages with bandwidth-constrained access links and with wireless
> > devices, to discard the packet in the network (in a 
> firewall device).
> 
> If we followed your reasoning, cell phones should never be allowed to
> ring, because doing so might deplete the batteries. They would only be
> allowed to make outgoing calls...

They would originate and maintain a signaling connection to their
call processing box.  That is what they do today on v4.

> There are many other ways to resolve this issue for the small battery
> powered devices. For example, they may be programmed to only listen to
> local addresses, which would radically solve the issue.

That could work well.

> They may also keep their address "hidden", so as to not receive 
> traffic.

I'm sure what you are referring to here; something akin to RFC3041?

> There are also some better ways to answer to DOS attacks in a
> firewall than blocking traffic outright. If the concern really 
> is the batteries, then rate limiting comes to mind.  For example, 
> if a device receives a packet and does not respond, or send an ICMP, 
> block the port for some period. 

Rate-limiting down to 0 would work fine to protect the battery and
the access link.  Higher, and the device has to wake up and scarce
access link bandwidth is consumed.

-d


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From behave-bounces@ietf.org Thu Aug 02 17:36:42 2007
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Subject: [BEHAVE] Response of firewalls and middleboxes to TCP SYN packets
	that are ECN-Capable?
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In the TCPM working group, my co-authors and I are finishing up a
draft on adding ECN-capability to TCP's SYN/ACK packets
("http://tools.ietf.org/html/draft-ietf-tcpm-ecnsyn").

Dave Thaler gave a report at the March IETF on the problems that
regular TCP data packets can encounter from routers or middleboxes
along the path
("http://www3.ietf.org/proceedings/07mar/slides/tsvarea-3/sld6.htm")

The point of this email is to ask if anyone knows anything about
specific problems that TCP SYN/ACK packets might encounter from the
network if the ECN field in the IP header is marked as ECN-Capable
(ECT) or Congestion Experienced (CE), over and above the problems
that are already known about for TCP data packets.  I am sending
this email to this mailing list because it was suggested by someone
at the TCPM working group last week.

Any feedback that anyone might have would be appreciated.

Many thanks,
- Sally
http://www.icir.org/floyd/



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Subject: RE: [BEHAVE] Re: CPE equipments and stateful filters
Date: Mon, 6 Aug 2007 09:42:17 +0300
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	<70C6EFCDFC8AAD418EF7063CD132D0640547EEF2@WIN-MSG-21.wingroup.windeploy.ntdev.microsoft.com>
From: <Remi.Denis-Courmont@nokia.com>
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>-----Original Message-----
>From: ext Christian Huitema [mailto:huitema@windows.microsoft.com]=20
>Sent: 01 August, 2007 10:04
>
>> From: Dan Wing [mailto:dwing@cisco.com]
>> Sent: Tuesday, July 31, 2007 10:34 PM
>>=20
>> Your email and Christian's email are both concerned with end-host=20
>> protection.  I am concerned with protecting the access link from=20
>> unwanted traffic to save bandwidth and, for battery-powered devices,=20
>> to save their batteries.  If the device isn't listening on=20
>> UDP/500 and would merely throw away or ICMP the incoming packet,
>> there are advantages with bandwidth-constrained access links and with
>> wireless devices, to discard the packet in the network (in a firewall
>> device).

Some low-speed radio links might need to be protected.

However, the challenge with regards to battery consumption is to
not have to send keepalives packet every so often. Quite obviously,
sending requires a lot more power than receiving.

In that sense, the battery would actually last LONGER on a device
with a host firewall (so it does not answer unsolicited packets),
but NO network firewall at all (so that there is no need to send
keepalives).

Explicit firewall signaling would probably provide a little bit
more savings, but "implicit" signaling (hole punching...) is simply
a big NO NO.

However...

>If we followed your reasoning, cell phones should never be=20
>allowed to ring, because doing so might deplete the batteries.=20
>They would only be allowed to make outgoing calls...
>
>There are many other ways to resolve this issue for the small=20
>battery powered devices. For example, they may be programmed=20
>to only listen to local addresses, which would radically solve=20
>the issue. They may also keep their address "hidden", so as to=20
>not receive traffic.
>
>There are also some better ways to answer to DOS attacks in a=20
>firewall than blocking traffic outright. If the concern really=20
>is the batteries, then rate limiting comes to mind. For=20
>example, if a device receives a packet and does not respond,=20
>or send an ICMP, block the port for some period.=20

Whether it is a good or bad business model, it is a matter of
fact that some operators charge their users per-bit or
per-packet, and it seems like IPv6 will not bring a change
here. These guys really NEED stateful firewalls so that their
users do not receive and PAY for packets they did not ask for.
Unless they do not mind a class action from said users.


--=20
R=E9mi Denis-Courmont
Software Engineer
Nokia TP/SP, Helsinki


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From: Pekka Savola <pekkas@netcore.fi>
To: Remi.Denis-Courmont@nokia.com
Subject: RE: [BEHAVE] Re: CPE equipments and stateful filters
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On Mon, 6 Aug 2007, Remi.Denis-Courmont@nokia.com wrote:
> Some low-speed radio links might need to be protected.
...
> Whether it is a good or bad business model, it is a matter of
> fact that some operators charge their users per-bit or
> per-packet, and it seems like IPv6 will not bring a change
> here.

I think this is what the argument boils down to.  I.e., which 
mechanisms are applicable on low-speed radio links vs everywhere else.

I'd argue that because low-speed radio links already need to implement 
lots and lots of other protocol unpleasantry (I'd call it 'junk'), the 
scenario is not very interesting from a generic host perspective.

Maybe the best would be to recognize the (at least) two different 
scenarios and consider applicable solutions separately.

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


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Subject: RE: [BEHAVE] TURN: technical review by September 1
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>-----Original Message-----
>From: ext Dan Wing [mailto:dwing@cisco.com]=20
>Sent: 30 July, 2007 20:27
>To: behave@ietf.org
>Cc: 'Rohan Mahy'; huitema@microsoft.com
>Subject: [BEHAVE] TURN: technical review by September 1
>
>In Chicago at IETF-69 we explained the urgency around=20
>rfc3489bis ("STUNbis") and TURN.  ICE needs to be published at=20
>the same time as rfc3489bis and TURN because ICE normatively=20
>references those documents.  Rfc3489bis underwent substantial=20
>revision between -06 and -08, and I just WGLC'd rfc3489bis-08=20
>a few minutes ago.  However, TURN is not yet ready to be WGLC'd.

Frankly, I do not understand why ICE normatively depends on TURN.
I could use SOCKS, UPnP, NAT-PMP, you-name-it... to generate
my candidates, yet ICE does not depend on these.

Of course, STUNbis is a dependency due to embracing and extending
the STUN messages format. But TURN... ??? In fact, while I understand
it might be late to "fix" this, I do not understand why ICE
specifically mention TURN all over the place in the details, while it
mentions tons of other potential non-normative ways to obtain
candidates in its introduction, but this is a minor almost-editorial
issue. So why is TURN not an informative reference?


However... I am sorry to say that, but I think it would be wrong to
Declare TURN "ready" before we=B4ve had the chance to examine its
Interactions with ICE-TCP, and that=B4s not something I can put within
5 weeks (counting from the end of last IETF). I mean, ICE-TCP is no
easy task in itself, and it would clearly be a pita when TURN turns
out to work fine with UDP and be broken with TCP (I would really like
we do not reproduce the mistakes from MSRP this time).

An alternative is to split TURN into UDP/UDP, first and the other
combinations separately, but I would not like this at all if I were
one of the editor. That=B4s what=B4s been done with ICE though, and I
expect that=B4s what implementors do - Does anyone know of a ICE-TCP
stack out yet at all?

Regards,

--=20
R=E9mi Denis-Courmont
Software Engineer
Nokia TP/SP, Helsinki=20


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Subject: RE: [BEHAVE] TURN: technical review by September 1
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>-----Original Message-----
>From: lowekamp@gmail.com [mailto:lowekamp@gmail.com]
>
>> (2) stop dequeing data from the TURN client until TCP_B gets=20
>> "decongested",
>> (3) drop data for B, and reset/timeout connection with B.
>
>> (2) is flawed, as communication between client and peer A will be=20
>> unduly disrupted. It is an obvious security issue if A and B do not=20
>> trust each others. Even then, it is probably unacceptable: if say, B=20
>> looses connectivity for some reason, the flow to A will be totally=20
>> disrupted until TCP_B connection times out. There are=20
>> similar problem if connectivity to B is much slower or less reliable
>> than that to A.

>I think that 2 is the only possible "correct" behavior,=20
>because 1 is unimplementable and 3 breaks the basic TCP model=20
>merely because one destination has lower throughput than=20
>another.  2's problem is that it restricts both connections to=20
>the poorer of the pair (whether deliberate attack or poor=20
>provisioning), but at least they both behave like valid TCP=20
>connections.

(2) is hardly more correct than (3). B could bring its TCP window
to zero for as long as the TCP timeout, then accept a few bytes,
and go back to zero, and so on...

And then, there=B4s the issue that it allows a single TURN client to
exhaust the server resources more easily.

>There's another possibility of making the protocol more=20
>complex by signalling queueing on the client connection, but I=20
>really don't think we want that.

Agreed. That would be the proper solution, but that would also
be overly complicated. At the end, we=B4d redefine conveying datagrams
on top of TCP, so you can do TCP over TCP :-(

>But unless there's a compelling reason to support it, I think=20
>it would be better if TURN just says one peer per client=20
>connection.  I'm not aware of a compelling reason.  If there=20
>is a compelling reason to support it, I think big red flags=20
>and possibly even SHOULD NOT language should be used to=20
>discourage its use except when necessary.

Personnaly, I tend to agree with Alfred and Evgeniy:
Multiple simultaneously-active connections should not be supported
at all. In fact, the send&data indications format is not quite
Suited for connection-oriented transports.

By the way, does ICE-TCP requires the TURN server to accept
multiple connections? If so, how does this fit with the requirements
to only allow one connection in order not to "bypass security"?


--=20
R=E9mi Denis-Courmont
Software Engineer
Nokia TP/SP, Helsinki=20


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From behave-bounces@ietf.org Mon Aug 06 10:36:56 2007
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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [BEHAVE] TURN: technical review by September 1
Date: Mon, 6 Aug 2007 07:36:46 -0700
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Hi Remi,

I seem to remember that gathering TURN candidates specifically was a =20
SHOULD in a previous version of ICE.  Now it just says: "Agents =20
SHOULD obtain relayed candidates..." but doesn't specify a specific =20
method.  Even so, it seems that normatively describing how to use at =20
least one method with ICE is quite important.

A related issue is that since SIP calls can fork, ICE needs to work =20
in this environment, so any relay protocol needs to have =20
deterministic behavior in this environment. I personally would not =20
have done TURN if we did not need support for forking.  Without =20
forking we could have reused another relay protocol (like SOCKS).

And as much as I would like to wish otherwise, I suspect you are =20
right that we don't know how well TURN TCP really works without an =20
early implementation ICE TCP.

thanks,
-rohan


On Aug 6, 2007, at 2:32 AM, <Remi.Denis-Courmont@nokia.com> wrote:
>> -----Original Message-----
>> From: ext Dan Wing [mailto:dwing@cisco.com]
>> Sent: 30 July, 2007 20:27
>> To: behave@ietf.org
>> Cc: 'Rohan Mahy'; huitema@microsoft.com
>> Subject: [BEHAVE] TURN: technical review by September 1
>>
>> In Chicago at IETF-69 we explained the urgency around
>> rfc3489bis ("STUNbis") and TURN.  ICE needs to be published at
>> the same time as rfc3489bis and TURN because ICE normatively
>> references those documents.  Rfc3489bis underwent substantial
>> revision between -06 and -08, and I just WGLC'd rfc3489bis-08
>> a few minutes ago.  However, TURN is not yet ready to be WGLC'd.
>
> Frankly, I do not understand why ICE normatively depends on TURN.
> I could use SOCKS, UPnP, NAT-PMP, you-name-it... to generate
> my candidates, yet ICE does not depend on these.
>
> Of course, STUNbis is a dependency due to embracing and extending
> the STUN messages format. But TURN... ??? In fact, while I understand
> it might be late to "fix" this, I do not understand why ICE
> specifically mention TURN all over the place in the details, while it
> mentions tons of other potential non-normative ways to obtain
> candidates in its introduction, but this is a minor almost-editorial
> issue. So why is TURN not an informative reference?
>
>
> However... I am sorry to say that, but I think it would be wrong to
> Declare TURN "ready" before we=B4ve had the chance to examine its
> Interactions with ICE-TCP, and that=B4s not something I can put within
> 5 weeks (counting from the end of last IETF). I mean, ICE-TCP is no
> easy task in itself, and it would clearly be a pita when TURN turns
> out to work fine with UDP and be broken with TCP (I would really like
> we do not reproduce the mistakes from MSRP this time).
>
> An alternative is to split TURN into UDP/UDP, first and the other
> combinations separately, but I would not like this at all if I were
> one of the editor. That=B4s what=B4s been done with ICE though, and I
> expect that=B4s what implementors do - Does anyone know of a ICE-TCP
> stack out yet at all?
>
> Regards,
>
> --=20
> R=E9mi Denis-Courmont
> Software Engineer
> Nokia TP/SP, Helsinki



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From behave-bounces@ietf.org Mon Aug 06 10:38:30 2007
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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [BEHAVE] TURN: technical review by September 1
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Hi Remi,

I think (2) is the normal, default handling for TCP to TCP relays.  =20
Yes, B could make its window very, very small, but TCP A is still no =20
worse off than a direct connection to the peer would have been.  I =20
also don't see how this exhausts server resources.  The server does =20
not have to do any work while it is not reading. It just needs to =20
keep connection state for the two connections.

The TURN server could still set some minimum responsiveness or =20
throughput metric after which it employs (3), but (2) is still the =20
best default IMO.

thanks,
-rohan



On Aug 6, 2007, at 2:54 AM, <Remi.Denis-Courmont@nokia.com> wrote:
>> -----Original Message-----
>> From: lowekamp@gmail.com [mailto:lowekamp@gmail.com]
>>
>>> (2) stop dequeing data from the TURN client until TCP_B gets
>>> "decongested",
>>> (3) drop data for B, and reset/timeout connection with B.
>>
>>> (2) is flawed, as communication between client and peer A will be
>>> unduly disrupted. It is an obvious security issue if A and B do not
>>> trust each others. Even then, it is probably unacceptable: if say, B
>>> looses connectivity for some reason, the flow to A will be totally
>>> disrupted until TCP_B connection times out. There are
>>> similar problem if connectivity to B is much slower or less reliable
>>> than that to A.
>
>> I think that 2 is the only possible "correct" behavior,
>> because 1 is unimplementable and 3 breaks the basic TCP model
>> merely because one destination has lower throughput than
>> another.  2's problem is that it restricts both connections to
>> the poorer of the pair (whether deliberate attack or poor
>> provisioning), but at least they both behave like valid TCP
>> connections.
>
> (2) is hardly more correct than (3). B could bring its TCP window
> to zero for as long as the TCP timeout, then accept a few bytes,
> and go back to zero, and so on...
>
> And then, there=B4s the issue that it allows a single TURN client to
> exhaust the server resources more easily.
>
>> There's another possibility of making the protocol more
>> complex by signalling queueing on the client connection, but I
>> really don't think we want that.
>
> Agreed. That would be the proper solution, but that would also
> be overly complicated. At the end, we=B4d redefine conveying datagrams
> on top of TCP, so you can do TCP over TCP :-(
>
>> But unless there's a compelling reason to support it, I think
>> it would be better if TURN just says one peer per client
>> connection.  I'm not aware of a compelling reason.  If there
>> is a compelling reason to support it, I think big red flags
>> and possibly even SHOULD NOT language should be used to
>> discourage its use except when necessary.
>
> Personnaly, I tend to agree with Alfred and Evgeniy:
> Multiple simultaneously-active connections should not be supported
> at all. In fact, the send&data indications format is not quite
> Suited for connection-oriented transports.
>
> By the way, does ICE-TCP requires the TURN server to accept
> multiple connections? If so, how does this fit with the requirements
> to only allow one connection in order not to "bypass security"?
>
>
> --=20
> R=E9mi Denis-Courmont
> Software Engineer
> Nokia TP/SP, Helsinki



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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 Rohan Mahy" <rohan@ekabal.com>
Subject: Re: [BEHAVE] TURN: technical review by September 1
Date: Tue, 7 Aug 2007 09:37:42 +0300
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On Monday 06 August 2007 17:36:46 ext Rohan Mahy wrote:
> A related issue is that since SIP calls can fork, ICE needs to work
> in this environment, so any relay protocol needs to have
> deterministic behavior in this environment. I personally would not
> have done TURN if we did not need support for forking.  Without
> forking we could have reused another relay protocol (like SOCKS).

After a bit of thinking, I doubt forking really requires multiplexing=20
TCP-over-TCP. The client can simply reconnect to the TURN server and reques=
t=20
the same port number as was already allocated using REQUESTED-PORT-PROPS,=20
REQUESTED-IP and REQUESTED-TRANSPORT, and preserve 1:1 TCP connection=20
relaying.

As such, I see no reasons for allowing 1:N (TCP|TLS)-to-TCP.

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


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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [BEHAVE] TURN: technical review by September 1
Date: Tue, 7 Aug 2007 07:03:02 -0700
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Hi R=E9mi,

I have to think about this suggestion a bit more.  Today, sending a =20
new Allocate request with an existing allocated address and port =20
causes a move.  We would need to add a new attribute at least.

I'm not sure this is really any simpler than supporting and (briefly) =20=

using 1:N TCP connections.  I don't think it is a practical problem =20
to limit the speed of the client to TURN server connection to the =20
speed of the slowest of the server to peer connections, but this =20
would be good to model.

thanks,
-rohan

On Aug 6, 2007, at 11:37 PM, R=E9mi Denis-Courmont wrote:
> On Monday 06 August 2007 17:36:46 ext Rohan Mahy wrote:
>> A related issue is that since SIP calls can fork, ICE needs to work
>> in this environment, so any relay protocol needs to have
>> deterministic behavior in this environment. I personally would not
>> have done TURN if we did not need support for forking.  Without
>> forking we could have reused another relay protocol (like SOCKS).
>
> After a bit of thinking, I doubt forking really requires multiplexing
> TCP-over-TCP. The client can simply reconnect to the TURN server =20
> and request
> the same port number as was already allocated using REQUESTED-PORT-=20
> PROPS,
> REQUESTED-IP and REQUESTED-TRANSPORT, and preserve 1:1 TCP connection
> relaying.
>
> As such, I see no reasons for allowing 1:N (TCP|TLS)-to-TCP.
>
> --=20
> R=E9mi Denis-Courmont



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[ Please direct responses to BEHAVE mailing list, NOT MMUSIC! ]

	Hello,

If anyone is implementing an ICE or something-using-STUN stack, I have put=
=20
sample FINGERPRINT and MESSAGE-INTEGRITY test vectors. I might have messed=
=20
them up, since I have not had the chance to test with third parties.

Pending processing by the secretariat, you can find them here:
http://www.remlab.net/files/ietf/draft-denis-behave-rfc3489bis-test-vectors=
=2D00.txt

Regards,

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


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Subject: Re: [BEHAVE] STUN fingerprint and integrity tests
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R=E9mi Denis-Courmont wrote:
> [ Please direct responses to BEHAVE mailing list, NOT MMUSIC! ]
>=20
> 	Hello,
>=20
> If anyone is implementing an ICE or something-using-STUN stack, I have =
put=20
> sample FINGERPRINT and MESSAGE-INTEGRITY test vectors. I might have mes=
sed=20
> them up, since I have not had the chance to test with third parties.

It seems that you did not adjust the header length before calculating
the MESSAGE-INTEGRITY for the request message (see last paragraph of
section 14.4 of rfc3489bis-08).


--=20
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [RFC1855-compliant space for rent ]
Work: marc@8x8.com            [                                 ]
[                                                               ]


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Subject: Re: [BEHAVE] STUN fingerprint and integrity tests
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You must be right, the SHA1s were computed according to -06.

I will fix them tomorrow. thanks for the catch.

Remi Denis-Courmont

--- message original ---
De : "ext Marc Petit-Huguenin" <petithug@acm.org>
Objet : Re: [BEHAVE] STUN fingerprint and integrity tests
Date : 7 ao=C3=BBt 2007
Heure : 18:55:38

R=C3=A9mi Denis-Courmont wrote:
> [ Please direct responses to BEHAVE mailing list, NOT MMUSIC! ]
>=20
> 	Hello,
>=20
> If anyone is implementing an ICE or something-using-STUN stack, I have =
put=20
> sample FINGERPRINT and MESSAGE-INTEGRITY test vectors. I might have =
messed=20
> them up, since I have not had the chance to test with third parties.

It seems that you did not adjust the header length before calculating
the MESSAGE-INTEGRITY for the request message (see last paragraph of
section 14.4 of rfc3489bis-08).


--=20
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [RFC1855-compliant space for rent ]
Work: marc@8x8.com            [                                 ]
[                                                               ]


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From: =?utf-8?q?R=C3=A9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
Organization: Nokia TP-SP-SWD
To: behave@ietf.org
Subject: Re: [BEHAVE] STUN fingerprint and integrity tests
Date: Wed, 8 Aug 2007 12:03:19 +0300
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> I will fix them tomorrow. thanks for the catch.

Alright, here are the hopefully fixed vectors:

static const char req[] =3D
"\x00\x01\x00\x44"
"\x21\x12\xa4\x42\xb7\xe7\xa7\x01\xbc\x34\xd6\x86\xfa\x87\xdf\xae"
"\x00\x24\x00\x04" "\x6e\x00\x01\xff"
"\x80\x29\x00\x08" "\x93\x2f\xf9\xb1\x51\x26\x3b\x36\x00\x06\x00"
 "\x09\x65\x76\x74\x6a\x3a\x68\x36\x76\x59\x20\x20\x20"
"\x00\x08\x00\x14" "\x62\x4e\xeb\xdc\x3c\xc9\x2d\xd8\x4b\x74\xbf\x85"
  "\xd1\xc0\xf5\xde\x36\x87\xbd\x33"
"\x80\x28\x00\x04" "\xad\x8a\x85\xff";

static const char resp[] =3D
"\x01\x01\x00\x3c"
"\x21\x12\xa4\x42\xb7\xe7\xa7\x01\xbc\x34\xd6\x86\xfa\x87\xdf\xae"
"\x80\x22\x00\x0b" "\x74\x65\x73\x74\x20\x76\x65\x63\x74\x6f\x72\x20"
"\x00\x20\x00\x08" "\x00\x01\xa1\x47\x5e\x12\xa4\x43"
"\x00\x08\x00\x14" "\xab\x4e\x53\x29\x61\x00\x08\x4c\x89\xf2\x7c\x69"
 "\x30\x33\x5c\xa3\x58\x14\xea\x90"
"\x80\x28\x00\x04" "\xae\x25\x8d\xf2";

Do they look OK to you?

=2D-=20
R=C3=A9mi 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, "ext Sally Floyd" <sallyfloyd@mac.com>
Subject: Re: [BEHAVE] Response of firewalls and middleboxes to TCP SYN packets
	that are ECN-Capable?
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On Friday 03 August 2007 00:36:32 ext Sally Floyd wrote:
> In the TCPM working group, my co-authors and I are finishing up a
> draft on adding ECN-capability to TCP's SYN/ACK packets
> ("http://tools.ietf.org/html/draft-ietf-tcpm-ecnsyn").
>
> Dave Thaler gave a report at the March IETF on the problems that
> regular TCP data packets can encounter from routers or middleboxes
> along the path
> ("http://www3.ietf.org/proceedings/07mar/slides/tsvarea-3/sld6.htm")
>
> The point of this email is to ask if anyone knows anything about
> specific problems that TCP SYN/ACK packets might encounter from the
> network if the ECN field in the IP header is marked as ECN-Capable
> (ECT) or Congestion Experienced (CE), over and above the problems
> that are already known about for TCP data packets.  I am sending
> this email to this mailing list because it was suggested by someone
> at the TCPM working group last week.

I remember having to disable ECN on my desktop PC to be able to reach some=
=20
websites at all. I am not aware of any specific additionnal problem with=20
SYN/ACK. I would expect that be the result of broken firewall/router=20
implementations that discard any packet with unknown IP header flags,=20
regardless of the transport header.

That's pure guessing though.

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


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From behave-bounces@ietf.org Wed Aug 08 12:17:08 2007
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Subject: Re: [BEHAVE] STUN fingerprint and integrity tests
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R=C3=A9mi Denis-Courmont wrote:
>> I will fix them tomorrow. thanks for the catch.
>=20
> Alright, here are the hopefully fixed vectors:
>=20
> static const char req[] =3D
> "\x00\x01\x00\x44"
> "\x21\x12\xa4\x42\xb7\xe7\xa7\x01\xbc\x34\xd6\x86\xfa\x87\xdf\xae"
> "\x00\x24\x00\x04" "\x6e\x00\x01\xff"
> "\x80\x29\x00\x08" "\x93\x2f\xf9\xb1\x51\x26\x3b\x36\x00\x06\x00"
>  "\x09\x65\x76\x74\x6a\x3a\x68\x36\x76\x59\x20\x20\x20"
> "\x00\x08\x00\x14" "\x62\x4e\xeb\xdc\x3c\xc9\x2d\xd8\x4b\x74\xbf\x85"
>   "\xd1\xc0\xf5\xde\x36\x87\xbd\x33"
> "\x80\x28\x00\x04" "\xad\x8a\x85\xff";
>=20
> static const char resp[] =3D
> "\x01\x01\x00\x3c"
> "\x21\x12\xa4\x42\xb7\xe7\xa7\x01\xbc\x34\xd6\x86\xfa\x87\xdf\xae"
> "\x80\x22\x00\x0b" "\x74\x65\x73\x74\x20\x76\x65\x63\x74\x6f\x72\x20"
> "\x00\x20\x00\x08" "\x00\x01\xa1\x47\x5e\x12\xa4\x43"
> "\x00\x08\x00\x14" "\xab\x4e\x53\x29\x61\x00\x08\x4c\x89\xf2\x7c\x69"
>  "\x30\x33\x5c\xa3\x58\x14\xea\x90"
> "\x80\x28\x00\x04" "\xae\x25\x8d\xf2";
>=20
> Do they look OK to you?
>=20

Yes, I added them to my unit tests.

Thanks.

--=20
Marc Petit-Huguenin           [                                 ]
Home: marc@petit-huguenin.org [RFC1855-compliant space for rent ]
Work: marc@8x8.com            [                                 ]
[                                                               ]


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From: =?iso-8859-1?q?R=E9mi_Denis-Courmont?= <rdenis@simphalempin.com>
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To: v6ops WG <v6ops@ops.ietf.org>
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Le vendredi 3 ao=FBt 2007, Nathan Ward a =E9crit :
> >  1) There are proprietary Teredo extensions (AFAIK, at least one
> > that allows working with symmetric NATs).  Do these depend on
> > additional code the servers?  If so, one server is not necessarily
> > equivalent to another, and I could understand why MSFT would be
> > reticent to use anycast.  If so, one also might need to reconsider
> > whether an anycast prefix makes sense when servers may provide
> > different feature sets.
>
> I'm not aware of these extensions, can anyone provide info?

=46rom http://www.microsoft.com/technet/network/ipv6/teredo.mspx :

  Symmetric NATs
  (...)
  Teredo in Windows Vista and Windows Server 2008 can work between
  Teredo clients if only one Teredo client is behind one or more
  symmetric NATs. For example, Teredo in Windows Vista and Windows
  Server 2008 will work if one of the peers is behind a symmetric NAT
  and the other is behind a cone or restricted NAT.

As far as I can tell, symmetric NAT detection is already part of the=20
base spec, so it is not unlikely that this unspecified extension does=20
not involve any change to the servers.


What's weirdest to me, as a regular of the BEHAVE WG, is support for=20
symmetric-to-restricted traversal. Symmetric-to-cone is "easy", but=20
symmetric-to-restricted, hmm?! At least, ICE cannot do that.

=2D-=20
R=E9mi Denis-Courmont
http://www.remlab.net/

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RĂ©mi Denis-Courmont wrote:
>> I will fix them tomorrow. thanks for the catch.
> 
> Alright, here are the hopefully fixed vectors:
> 
> static const char req[] =
> "\x00\x01\x00\x44"
> "\x21\x12\xa4\x42\xb7\xe7\xa7\x01\xbc\x34\xd6\x86\xfa\x87\xdf\xae"
> "\x00\x24\x00\x04" "\x6e\x00\x01\xff"
> "\x80\x29\x00\x08" "\x93\x2f\xf9\xb1\x51\x26\x3b\x36\x00\x06\x00"
>  "\x09\x65\x76\x74\x6a\x3a\x68\x36\x76\x59\x20\x20\x20"
> "\x00\x08\x00\x14" "\x62\x4e\xeb\xdc\x3c\xc9\x2d\xd8\x4b\x74\xbf\x85"
>   "\xd1\xc0\xf5\xde\x36\x87\xbd\x33"
> "\x80\x28\x00\x04" "\xad\x8a\x85\xff";
> 
> static const char resp[] =
> "\x01\x01\x00\x3c"
> "\x21\x12\xa4\x42\xb7\xe7\xa7\x01\xbc\x34\xd6\x86\xfa\x87\xdf\xae"
> "\x80\x22\x00\x0b" "\x74\x65\x73\x74\x20\x76\x65\x63\x74\x6f\x72\x20"
> "\x00\x20\x00\x08" "\x00\x01\xa1\x47\x5e\x12\xa4\x43"
> "\x00\x08\x00\x14" "\xab\x4e\x53\x29\x61\x00\x08\x4c\x89\xf2\x7c\x69"
>  "\x30\x33\x5c\xa3\x58\x14\xea\x90"
> "\x80\x28\x00\x04" "\xae\x25\x8d\xf2";
> 
> Do they look OK to you?
> 


Remi, many thanks for doing this. It looks ok after I fixed some bugs in my
code. Is the 64-bytes HMAC-SHA1 padding removed from rfc3489-bis, btw?

could I ask that the USERNAME padding is changed from 0x20 to 0x00 ?


/alfred


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> From http://www.microsoft.com/technet/network/ipv6/teredo.mspx :
> 
>   Symmetric NATs
>   (...)
>   Teredo in Windows Vista and Windows Server 2008 can work between
>   Teredo clients if only one Teredo client is behind one or more
>   symmetric NATs. For example, Teredo in Windows Vista and Windows
>   Server 2008 will work if one of the peers is behind a symmetric NAT
>   and the other is behind a cone or restricted NAT.
> 
> As far as I can tell, symmetric NAT detection is already part of the 
> base spec, so it is not unlikely that this unspecified extension does 
> not involve any change to the servers.
> 
>
> What's weirdest to me, as a regular of the BEHAVE WG, is support for 
> symmetric-to-restricted traversal. Symmetric-to-cone is "easy", but 
> symmetric-to-restricted, hmm?! At least, ICE cannot do that.

Using RFC3489 terms, ICE can successfully establish a direct peer-to-
peer UDP session between two hosts where one is behind a 'symmetric' 
NAT and another is behind a 'restricted cone' NAT (but you're right
that ICE cannot establish that connection if it's a 'port restricted 
cone').

RFC4787's terminology, although longer than what most people use and
are familiar with, does a much cleaner job of separating the port 
assignment (which is part of the problem; this is the major problem
with 'symmetric' NATs) from the filtering (which is the other part 
of the problem).

-d


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To: "ext Alfred E. Heggestad" <aeh@db.org>
Subject: Re: [BEHAVE] STUN fingerprint and integrity tests
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On Thursday 09 August 2007 01:23:36 ext Alfred E. Heggestad wrote:
> Remi, many thanks for doing this. It looks ok after I fixed some bugs in =
my
> code. Is the 64-bytes HMAC-SHA1 padding removed from rfc3489-bis, btw?

Yes. That text has been removed a bunch of revisions ago. You'd have to ask=
=20
the editors why, but it was simply a "bad" idea - HMAC-SHA1 already pads=20
internally anyway.

> could I ask that the USERNAME padding is changed from 0x20 to 0x00 ?

rfc3489bis-08 says attribute padding can be anything. I am purposedly using=
=20
white spaces instead of nuls to root out incorrect C implementations. I=20
should perhaps have used 0x41 to make it more obvious :)

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


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From: =?iso-8859-1?q?R=E9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
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To: "ext Rohan Mahy" <rohan@ekabal.com>, behave@ietf.org
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	Hello,

On Tuesday 07 August 2007 17:03:02 ext Rohan Mahy wrote:
> I have to think about this suggestion a bit more.  Today, sending a
> new Allocate request with an existing allocated address and port
> causes a move.  We would need to add a new attribute at least.

While I do understand the point of moving a UDP flow, I doubt moving a TCP=
=20
session makes any sense. If the TCP connection breaks, the server and the=20
server cannot really know up to where the other side has successfully=20
received data - all they know is that it's somewhere between the highest=20
received acknowledgement sequence and the highest sent sequence. But within=
=20
the window, it's unknown by design.

In theory, both sides could replicate "stream offsets" above the socket lay=
er,=20
and exchange them when moving a connection, but I think nobody wants to go=
=20
into that. And we'd have to replicate the TCP window buffers as well to=20
support retransmission.

As such, I really think "moving" should be restricted to UDP server-to-peer=
=20
flows. For TCP server-to-peer flows, if the client-to-server session breaks=
,=20
the server must break the server-to-peer connection.

> I'm not sure this is really any simpler than supporting and (briefly)
> using 1:N TCP connections.  I don't think it is a practical problem
> to limit the speed of the client to TURN server connection to the
> speed of the slowest of the server to peer connections, but this
> would be good to model.

Say I want to send a 2gigs movie to a forked MSRP session. Should the faste=
st=20
receiver wait for the slowest one? As a user, I definitely expect the faste=
st=20
to not to have to wait.

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


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From: "Alfred E. Heggestad" <aeh@db.org>
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RĂ©mi Denis-Courmont wrote:
> On Thursday 09 August 2007 01:23:36 ext Alfred E. Heggestad wrote:
>> Remi, many thanks for doing this. It looks ok after I fixed some bugs in my
>> code. Is the 64-bytes HMAC-SHA1 padding removed from rfc3489-bis, btw?
> 
> Yes. That text has been removed a bunch of revisions ago. You'd have to ask 
> the editors why, but it was simply a "bad" idea - HMAC-SHA1 already pads 
> internally anyway.
> 

ok thanks, that makes the implementation much simpler.

with the current MESSAGE-INTEGRITY calculation, it means that the receiver must
modify the incoming data before calculating HMAC-SHA1, in case the FINGERPRINT
attribute is present after M-I. it also means that the STUN receiver must parse
the whole message first, to check if the FINGERPRINT attribute is present, and
the subtract 8 from the STUN Header length field.

rfc3489bis-08 says that before calculating the M-I, a dummy M-I must be inserted
into the message (to get the correct length). would it not be better to include
the FINGERPRINT as a "dummy" attribute also? If that was the case, the length
fields needs not modification at all.


>> could I ask that the USERNAME padding is changed from 0x20 to 0x00 ?
> 
> rfc3489bis-08 says attribute padding can be anything. I am purposedly using 
> white spaces instead of nuls to root out incorrect C implementations. I 
> should perhaps have used 0x41 to make it more obvious :)
> 

okay. btw, could you add that the mapped address is 127.0.0.1:32853 in your
response vector?



/alfred


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From: Brian Korver <briank@networkresonance.com>
Subject: Re: [BEHAVE] STUN fingerprint and integrity tests
Date: Wed, 8 Aug 2007 12:12:20 -0700
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These look good.

-briank
briank@networkresonance.com



On Aug 8, 2007, at 2:03 AM, R=E9mi Denis-Courmont wrote:
>> I will fix them tomorrow. thanks for the catch.
>
> Alright, here are the hopefully fixed vectors:
>
> static const char req[] =3D
> "\x00\x01\x00\x44"
> "\x21\x12\xa4\x42\xb7\xe7\xa7\x01\xbc\x34\xd6\x86\xfa\x87\xdf\xae"
> "\x00\x24\x00\x04" "\x6e\x00\x01\xff"
> "\x80\x29\x00\x08" "\x93\x2f\xf9\xb1\x51\x26\x3b\x36\x00\x06\x00"
>  "\x09\x65\x76\x74\x6a\x3a\x68\x36\x76\x59\x20\x20\x20"
> "\x00\x08\x00\x14" "\x62\x4e\xeb\xdc\x3c\xc9\x2d\xd8\x4b\x74\xbf\x85"
>   "\xd1\xc0\xf5\xde\x36\x87\xbd\x33"
> "\x80\x28\x00\x04" "\xad\x8a\x85\xff";
>
> static const char resp[] =3D
> "\x01\x01\x00\x3c"
> "\x21\x12\xa4\x42\xb7\xe7\xa7\x01\xbc\x34\xd6\x86\xfa\x87\xdf\xae"
> "\x80\x22\x00\x0b" "\x74\x65\x73\x74\x20\x76\x65\x63\x74\x6f\x72\x20"
> "\x00\x20\x00\x08" "\x00\x01\xa1\x47\x5e\x12\xa4\x43"
> "\x00\x08\x00\x14" "\xab\x4e\x53\x29\x61\x00\x08\x4c\x89\xf2\x7c\x69"
>  "\x30\x33\x5c\xa3\x58\x14\xea\x90"
> "\x80\x28\x00\x04" "\xae\x25\x8d\xf2";
>
> Do they look OK to you?
>
> --=20
> R=E9mi Denis-Courmont
>
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave









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From behave-bounces@ietf.org Fri Aug 10 01:54:34 2007
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From: =?utf-8?q?R=C3=A9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
Organization: Nokia TP-SP-SWD
To: "ext Alfred E. Heggestad" <aeh@db.org>
Subject: Re: [BEHAVE] STUN fingerprint and integrity tests
Date: Fri, 10 Aug 2007 08:54:09 +0300
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On Thursday 09 August 2007 20:12:44 you wrote:
> with the current MESSAGE-INTEGRITY calculation, it means that the receiver
> must modify the incoming data before calculating HMAC-SHA1, in case the
> FINGERPRINT attribute is present after M-I. it also means that the STUN
> receiver must parse the whole message first, to check if the FINGERPRINT
> attribute is present, and the subtract 8 from the STUN Header length fiel=
d.
>
> rfc3489bis-08 says that before calculating the M-I, a dummy M-I must be
> inserted into the message (to get the correct length). would it not be
> better to include the FINGERPRINT as a "dummy" attribute also? If that was
> the case, the length fields needs not modification at all.

Yes. The current -07/-08 text is optimized for sending, while the -06 was=20
implicitly optimized for receiving. In practice, -07/-08 makes sending bare=
ly=20
easier, and receiving much more intricate.

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


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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [BEHAVE] TURN: technical review by September 1
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On Aug 9, 2007, at 4:01 AM, R=E9mi Denis-Courmont wrote:
>> I'm not sure this is really any simpler than supporting and (briefly)
>> using 1:N TCP connections.  I don't think it is a practical problem
>> to limit the speed of the client to TURN server connection to the
>> speed of the slowest of the server to peer connections, but this
>> would be good to model.
>
> Say I want to send a 2gigs movie to a forked MSRP session. Should =20
> the fastest
> receiver wait for the slowest one? As a user, I definitely expect =20
> the fastest
> to not to have to wait.

Yes, I still think it is reasonable for the fastest receiver to wait =20
for the slowest one.  This only happens during the period where you =20
are forking.  You probably won't start sending a 2gig movie until =20
you've resolved which instance you are sending it to, and therefore =20
selected a single fork by then.

thanks,
-rohan




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From: Cullen Jennings <fluffy@cisco.com>
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Subject: [BEHAVE] Minor concern about one of the milestones 
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One of our milestones is

Submit standards-track document that describes how an application can  
determine the type of NAT it is behind

I think it is very hard for an application to reliably do this except  
under environments where all the traffic to the NAT is controlled.  
(Basically this was the general limitation of RFC 3489). I hope that  
the document for this milestone is more an approach to characterizing  
and testing behavior of NATs and is not a general suggestion that an  
application might be able to figure out what type of NAT it is behind  
then modify the applications behavior based on the type of NAT is is  
behind. I seem to recall the old milestone was more along the lines  
of testing but I can not find a copy of it.

Cullen <with my individual hat on>


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From behave-bounces@ietf.org Mon Aug 13 00:33:05 2007
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Remi -

> I remember having to disable ECN on my desktop PC to be able to reach =20=

> some
> websites at all. I am not aware of any specific additionnal problem =20=

> with
> SYN/ACK. I would expect that be the result of broken firewall/router
> implementations that discard any packet with unknown IP header flags,
> regardless of the transport header.

Many thanks for the feedback.  I will add that to the draft.

I have an old paper, "Measuring Interactions Between Transport
Protocols and Middleboxes" ("http://www.icir.org/tbit/", Figure 1)
that shows that there are many paths where the TCP SYN exchange
doesn't complete if the SYN packet contains a known or unknown IP
Option...

- Sally
http://www.icir.org/floyd/



------------------------------------------------------------------------=20=

------------------------------------------
On Aug 8, 2007, at 6:52 AM, R=E9mi Denis-Courmont wrote:

> On Friday 03 August 2007 00:36:32 ext Sally Floyd wrote:
>> In the TCPM working group, my co-authors and I are finishing up a
>> draft on adding ECN-capability to TCP's SYN/ACK packets
>> ("http://tools.ietf.org/html/draft-ietf-tcpm-ecnsyn").
>>
>> Dave Thaler gave a report at the March IETF on the problems that
>> regular TCP data packets can encounter from routers or middleboxes
>> along the path
>> ("http://www3.ietf.org/proceedings/07mar/slides/tsvarea-3/sld6.htm")
>>
>> The point of this email is to ask if anyone knows anything about
>> specific problems that TCP SYN/ACK packets might encounter from the
>> network if the ECN field in the IP header is marked as ECN-Capable
>> (ECT) or Congestion Experienced (CE), over and above the problems
>> that are already known about for TCP data packets.  I am sending
>> this email to this mailing list because it was suggested by someone
>> at the TCPM working group last week.
>
> I remember having to disable ECN on my desktop PC to be able to reach =20=

> some
> websites at all. I am not aware of any specific additionnal problem =20=

> with
> SYN/ACK. I would expect that be the result of broken firewall/router
> implementations that discard any packet with unknown IP header flags,
> regardless of the transport header.
>
> That's pure guessing though.
>
> --=20
> R=E9mi Denis-Courmont
>



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On Thursday 09 August 2007 20:12:44 ext Alfred E. Heggestad wrote:
> okay. btw, could you add that the mapped address is 127.0.0.1:32853 in yo=
ur
> response vector?

Hmm, yes... but... whoops, I SHOULD have used example IP addresses range...=
=20
Looks like I'll have to update the vectors yet again :-(

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


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From: =?iso-8859-1?q?R=E9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
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Subject: Re: [BEHAVE] TURN: technical review by September 1
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On Friday 10 August 2007 17:14:58 ext Rohan Mahy wrote:
> On Aug 9, 2007, at 4:01 AM, R=E9mi Denis-Courmont wrote:
> >> I'm not sure this is really any simpler than supporting and (briefly)
> >> using 1:N TCP connections.  I don't think it is a practical problem
> >> to limit the speed of the client to TURN server connection to the
> >> speed of the slowest of the server to peer connections, but this
> >> would be good to model.
> >
> > Say I want to send a 2gigs movie to a forked MSRP session. Should
> > the fastest
> > receiver wait for the slowest one? As a user, I definitely expect
> > the fastest
> > to not to have to wait.
>
> Yes, I still think it is reasonable for the fastest receiver to wait
> for the slowest one.  This only happens during the period where you
> are forking.  You probably won't start sending a 2gig movie until
> you've resolved which instance you are sending it to, and therefore
> selected a single fork by then.

I do not quite agree. To recap, there are lots of issues with 1:N TURN/TCP=
=20
sessions.


=46irst, there is no TCP protocol that supports forking as of now. Contrary=
 to=20
my earlier assumption (my mistake), MSRP does not support forking at all.=20
Quoting draft-ietf-simple-message-sessions-19 =A75.4:

   (...) The
   node MUST also check to make sure the session is not already in use
   on another connection.  If the session is already in use, it MUST
   reject the request with a 506 response.

      If it were legal to have multiple connections associated with the
      same session, a security problem would exist.  If the initial SEND
      request is not protected, an eavesdropper might learn the URI, and
      use it to insert messages into the session via a different
      connection.


Second, I am still doubtful how practical it would be to "resolve" a fork w=
hen=20
the slowest peer is very slow.


Third, there is no way for the client to request resetting a TCP session. O=
ne=20
simple way would have been to close the TURN connection, but that only work=
s=20
if we have 1:1 connection, and disallow "moving" TCP connections.


=46orth, I may have missed something, but how will support for multiple TCP=
 not=20
contradict the security requirement NOT to allow running servers?


=46ifth, and loosely related, "moving" TCP connections is not generally=20
possible. It might work for TCP-framed RTP. But it does not work with=20
any "normally TCP" protocol, such as TLS or MSRP, as these cannot cope with=
=20
reordering or packet loss.


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


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From: =?iso-8859-1?q?R=E9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
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	Hello,

It currently seems that REQUESTED-IP is supposed to not be "obfuscated" whi=
le=20
RELAY-ADDRESS is. I expect both options to contain the same IP address... i=
s=20
there a reason for this apparent inconsistency?

Regards,

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


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> -----Original Message-----
> From: Brian Haberman [mailto:brian@innovationslab.net] 
> Sent: Thursday, June 07, 2007 6:16 AM
> To: Dan Wing
> Cc: 'Marshall Eubanks'; mboned@ietf.org; 'Alvaro Fernandez'; 
> magma@ietf.org; behave@ietf.org; 'Marshall Eubanks'; Toerless Eckert
> Subject: Re: [BEHAVE] Re: [magma] RE: [MBONED] FW: 
> WGLC:draft-ietf-behave-multicast-06.txt
....
> 
> 
> 
> Dan Wing wrote:
> >> The NAPT knows what is SSM. Couldn't it also do a Multicast Group  
> >> Translation for SSM groups ? Why
> >> is this more difficult than, say, port translation ?
> > 
> > It may be fairly easy -- except for rewriting the 
> > advertisements in SAP,
> > HTTP, HTTPS (a big problem there), email, SIP, RTSP, ....
> > 
> > And NAPTs don't do that kind of rewriting today; 
> > draft-ietf-behave-multicast
> > is only a BCP and trying to describe what a NAT should best 
> > be doing as of
> > its date of publication.
> > 
> > 
> > If we want NAPTs to do this, we would want a different 
> > document that was
> > standards-track and that document would define how we 
> > wanted NAPTs to
> > perform this function.  However, I don't see a successful 
> > path if such an
> > operation would require the NAPT to have an ALG for SAP, 
> > HTTP, SIP, and RTSP
> > and requires the NAT also rewrite those packets.  We'd need 
> > something else.
> > ALGs don't work well for a variety of reasons.
> 
> Given that, I would suggest that the document state that
> administrators of the network behind the NAPT assign sub-ranges
> of the SSM space to their multicast sources.  This could be done
> manually or *possibly* with the tools developed by the MALLOC
> WG.  For the ma & pa type residential networks, I doubt you will
> see multiple machines behind a NAPT sourcing SSM.  For more
> corporate types, the network admin can control the allocation of
> the SSM range.

I moved text discussing what applications should do into an appendix,
as the document's BCP scope is really around NATs (rather than 
applications that need to handle NATs).  Based on your comment above,
and some other input, the text now has the following paragraph.  I
believe this captures your suggestion.

  "If multiple SSM sources on the inside of a NAT choose the same
   multicast group address, those sources are uniquely identifiable
   because their IP addresses are unique.  However, if their multicast
   traffic is NATted and sent on the NAT's public interface, the traffic
   from those individual sources is no longer uniquely identifiable.
   This will cause problems for multicast receivers which will see an
   intermixing of traffic from those sources.  Resolution of this issue
   is left for future study.  In the meantime, applications that source
   SSM multicast traffic are encouraged to allow the user to modify the
   multicast SSM address so that users can avoid this problem if that
   application is placed behind a NAT."

The text doesn't quite go so far to say that a range has to be assigned a
priori.  But let me know if it is acceptable.

I have not submitted -09, but a preview is here if anyone wants to see the
full context:
http://svn.resiprocate.org/rep/ietf-drafts/behave/draft-ietf-behave-multicast-
09.txt

-d


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> -----Original Message-----
> From: Prashant Jhingran [mailto:prashantj@huawei.com] 
> Sent: Friday, June 22, 2007 3:01 AM
> To: 'Dan Wing'; behave@ietf.org; magma@ietf.org
> Cc: 'Toerless Eckert'
> Subject: RE: [magma] multicast senders behind an IGMP Proxy
> 
> Hi,
> 
> I would like to suggest following methods to  resolve this
> issue:
> 
> - TTL scoping can be used to control the distribution of
> multicast traffic with the objective of easing stress on
> scarce resources viz. bandwidth.

Toerless and I went back and forth on this suggestion, and
finally decided on the following text in the appendix of
multicast-09 (which we're about to publish):

   A multicast source that wants its traffic to not traverse a router
   (e.g., leave a home network) may find it useful to send traffic with
   IP TTL=1.  Both ASM and SSM sources may find this useful.

-d

> - Another way (and probably a preferred choice) would be to
> use administratively scoped IP multicast [rfc2365]. 
>
> Moreover, these methods can (to a certain extent) help in
> achieving privacy.
> 
> PS: "sending audio to all of your IP-enabled speakers
> distributed throughout your house" seems to be very
> ambitious usage of multicast :-) 
> 
> Regards,
> Prashant Jhingran
> 
> Huawei Technologies
> +91-80-41117676 Ext -7175
> +91-9448927814
> 
> "Sing, dance, serve, meditate and celebrate" 
>  - www.artofliving.org  
> 
> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com] 
> Sent: Friday, June 22, 2007 5:48 AM
> To: behave@ietf.org; magma@ietf.org
> Cc: Toerless Eckert
> Subject: [magma] multicast senders behind an IGMP Proxy
> 
> During WGLC, there was clear consensus from MAGMA and MBONED
> that supporting
> multicast senders behind the IGMP Proxy device was
> desirable.  However,
> while wrapping up draft-ietf-behave-multicast-07, Toerless
> pointed out a
> problem with multicast senders behind an IGMP Proxy device
> (any IGMP Proxy
> device -- not just an IGMP Proxy in a NAT).
> 
> The problem is that any multicast flows generated behind the
> IGMP proxy will
> be forwarded up the access link towards the Internet -- even
> for flows that
> you want to keep within your network.  Concrete examples of
> generating
> multicast within a home network include:
> 
>  * sending audio to all of your IP-enabled speakers
> distributed 
>    throughout your house,  
>  * IP-enabled intercom system, 
>  * HD video to multiple telvision sets (kitchen and living
> room).
> 
> In each case, you "could" do unicast, but especially for HD
> one might easily
> envision not having sufficient bandwidth or not wanting to
> waste the
> bandwidth (which might be used for other things).
> 
> Does anyone have suggestions on how this might be resolved?
> 
> -d
> 
> 
> _______________________________________________
> magma mailing list
> magma@ietf.org
> https://www1.ietf.org/mailman/listinfo/magma
> 


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From behave-bounces@ietf.org Tue Aug 14 07:46:27 2007
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From: =?iso-8859-1?q?R=E9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
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To: behave@ietf.org, ext Dan Wing <dwing@cisco.com>, jdrosen@cisco.com,
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Subject: [BEHAVE] rfc3489bis-08 last minute comment: 401 still problematic
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	Hello,

It was recently agreed that incorrect usernames and incorrect message=20
integrity hashes would be treated identically by STUN servers, which is to=
=20
say, yield a 401 error. The idea is that an attacker cannot probe for valid=
=20
usernames separately from the shared secret.

However, at the moment, this is still possible because stale nonce=20
verification is done AFTER username validation, but BEFORE message integrit=
y=20
validation. IMHO, it should be done after both, otherwise a bad username=20
yields 401, but a bad message integrity along with a bad nonce yields 438.

Therefore, I would suggest replacing that (=A710.2.2)
"
   o  If the NONCE is no longer valid, the server MUST generate an error
      response with an error code of 438 (Stale Nonce).  This response
      MUST include a NONCE and REALM attribute.

   o  Using the password associated with the username in the USERNAME
      attribute, compute the value for the message-integrity as
      described in Section 14.4.  If the resulting value does not match
      the contents of the MESSAGE-INTEGRITY attribute, the server MUST
      reject the request with an error response.  This response MUST use
      an error code of 401 (Unauthorized).  It MUST include a REALM and
      NONCE attribute.
"
with this:
"
   o  Using the password associated with the username in the USERNAME
      attribute, compute the value for the message-integrity as
      described in Section 14.4.  If the resulting value does not match
      the contents of the MESSAGE-INTEGRITY attribute, the server MUST
      reject the request with an error response.  This response MUST use
      an error code of 401 (Unauthorized).  It MUST include a REALM and
      NONCE attribute.

   o  Otherwise, if the NONCE is no longer valid, the server MUST
      generate an error response with an error code of 438 (Stale Nonce).
      This response MUST include a NONCE and REALM attribute.
"

Best regards,

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


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Subject: [BEHAVE] Re: rfc3489bis-08 last minute comment: 401 still
	problematic
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Well, this is a dual-edged sword.

Though you are correct, that such a change would stop a user from being 
able to probe for usernames separately from passwords, it does allow an 
attacker to cause the server to compute message integrity for any 
request, as long as the attacker knows a valid username. With the 
current ordering, they need to know the current username and have a 
valid nonce. Having a valid nonce helps eliminate the impact of DDOS 
attacks. With your change, if an attacker can guess a username, they can 
have a botnet army each send requests with valid usernames and invalid 
nonces and MI, and cause the server to do lots of hashing without return 
routability.

I think the current text is the lesser of two evils in this.

-Jonathan R.

Rémi Denis-Courmont wrote:
> 	Hello,
> 
> It was recently agreed that incorrect usernames and incorrect message 
> integrity hashes would be treated identically by STUN servers, which is to 
> say, yield a 401 error. The idea is that an attacker cannot probe for valid 
> usernames separately from the shared secret.
> 
> However, at the moment, this is still possible because stale nonce 
> verification is done AFTER username validation, but BEFORE message integrity 
> validation. IMHO, it should be done after both, otherwise a bad username 
> yields 401, but a bad message integrity along with a bad nonce yields 438.
> 
> Therefore, I would suggest replacing that (§10.2.2)
> "
>    o  If the NONCE is no longer valid, the server MUST generate an error
>       response with an error code of 438 (Stale Nonce).  This response
>       MUST include a NONCE and REALM attribute.
> 
>    o  Using the password associated with the username in the USERNAME
>       attribute, compute the value for the message-integrity as
>       described in Section 14.4.  If the resulting value does not match
>       the contents of the MESSAGE-INTEGRITY attribute, the server MUST
>       reject the request with an error response.  This response MUST use
>       an error code of 401 (Unauthorized).  It MUST include a REALM and
>       NONCE attribute.
> "
> with this:
> "
>    o  Using the password associated with the username in the USERNAME
>       attribute, compute the value for the message-integrity as
>       described in Section 14.4.  If the resulting value does not match
>       the contents of the MESSAGE-INTEGRITY attribute, the server MUST
>       reject the request with an error response.  This response MUST use
>       an error code of 401 (Unauthorized).  It MUST include a REALM and
>       NONCE attribute.
> 
>    o  Otherwise, if the NONCE is no longer valid, the server MUST
>       generate an error response with an error code of 438 (Stale Nonce).
>       This response MUST include a NONCE and REALM attribute.
> "
> 
> Best regards,
> 

-- 
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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On Tuesday 14 August 2007 16:00:18 ext Jonathan Rosenberg wrote:
> Well, this is a dual-edged sword.
>
> Though you are correct, that such a change would stop a user from being
> able to probe for usernames separately from passwords, it does allow an
> attacker to cause the server to compute message integrity for any
> request, as long as the attacker knows a valid username. With the
> current ordering, they need to know the current username and have a
> valid nonce. Having a valid nonce helps eliminate the impact of DDOS
> attacks. With your change, if an attacker can guess a username, they can
> have a botnet army each send requests with valid usernames and invalid
> nonces and MI, and cause the server to do lots of hashing without return
> routability.

True. Then perhaps NONCE should be checked before both USERNAME and M-I. It=
=20
cannot depend on the username anyway (since the server must return a valid=
=20
NONCE to an unauthenticated request).

No?

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


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An important point of clarification here.

Though FINGERPRINT is currently the only attribute beyond the MI, there 
could potentially be others in the future so you can't depend on it 
being the only one. So inserting a dummy FINGERPRINT won't help.

A different idea which might help is that the hash is computed by 
*starting* with the attributes, going through all of the attributes up 
to but not including M-I, and then including the STUN message header as 
the *last* attribute, with the length instead set as equal to the length 
of the data that the hash is computed over.

This would mean that, you could do the parse and integrity computation 
simultaneously.

-Jonathan R.

Alfred E. Heggestad wrote:
> RĂ©mi Denis-Courmont wrote:
>> On Thursday 09 August 2007 01:23:36 ext Alfred E. Heggestad wrote:
>>> Remi, many thanks for doing this. It looks ok after I fixed some bugs 
>>> in my
>>> code. Is the 64-bytes HMAC-SHA1 padding removed from rfc3489-bis, btw?
>>
>> Yes. That text has been removed a bunch of revisions ago. You'd have 
>> to ask the editors why, but it was simply a "bad" idea - HMAC-SHA1 
>> already pads internally anyway.
>>
> 
> ok thanks, that makes the implementation much simpler.
> 
> with the current MESSAGE-INTEGRITY calculation, it means that the 
> receiver must
> modify the incoming data before calculating HMAC-SHA1, in case the 
> FINGERPRINT
> attribute is present after M-I. it also means that the STUN receiver 
> must parse
> the whole message first, to check if the FINGERPRINT attribute is 
> present, and
> the subtract 8 from the STUN Header length field.
> 
> rfc3489bis-08 says that before calculating the M-I, a dummy M-I must be 
> inserted
> into the message (to get the correct length). would it not be better to 
> include
> the FINGERPRINT as a "dummy" attribute also? If that was the case, the 
> length
> fields needs not modification at all.
> 
> 
>>> could I ask that the USERNAME padding is changed from 0x20 to 0x00 ?
>>
>> rfc3489bis-08 says attribute padding can be anything. I am purposedly 
>> using white spaces instead of nuls to root out incorrect C 
>> implementations. I should perhaps have used 0x41 to make it more 
>> obvious :)
>>
> 
> okay. btw, could you add that the mapped address is 127.0.0.1:32853 in your
> response vector?
> 
> 
> 
> /alfred
> 
> 
> _______________________________________________
> 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
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I think there was consensus in the room in chicago to make this either 
informational or experimental.

-Jonathan R.

Cullen Jennings wrote:
> 
> One of our milestones is
> 
> Submit standards-track document that describes how an application can 
> determine the type of NAT it is behind
> 
> I think it is very hard for an application to reliably do this except 
> under environments where all the traffic to the NAT is controlled. 
> (Basically this was the general limitation of RFC 3489). I hope that the 
> document for this milestone is more an approach to characterizing and 
> testing behavior of NATs and is not a general suggestion that an 
> application might be able to figure out what type of NAT it is behind 
> then modify the applications behavior based on the type of NAT is is 
> behind. I seem to recall the old milestone was more along the lines of 
> testing but I can not find a copy of it.
> 
> Cullen <with my individual hat on>
> 
> 
> _______________________________________________
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> 

-- 
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
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OK, here is the text now:

<section title="Receiving a Request">

<t>
   After the server has done the basic processing of a request, it
   performs the checks listed below in the order specified:
</t>

<list style="symbols">
   <t>If the message does not contain a MESSAGE-INTEGRITY attribute,
     the server MUST
     generate an error response with an error code of
     401 (Unauthorized). This response MUST include a REALM value. It is
     RECOMMENDED that the REALM value be the domain name of
     the provider of the STUN server. The response MUST
     include a NONCE, selected by the server.
   </t>

   <t>
     If the message contains a MESSAGE-INTEGRITY
     attribute, but is missing the USERNAME, REALM or
     NONCE attributes, the server MUST generate an error
     response with an error code of 400 (Bad Request).
   </t>

   <t>
     If the NONCE is no longer valid, the server MUST
     generate an error response with an error code of 438
     (Stale Nonce). This response MUST include a NONCE and REALM
     attribute.
   </t>

   <t>
     If the username in the USERNAME attribute is not valid, the server
     MUST generate an error response with an error code of
     401 (Unauthorized). This response MUST include a REALM value. It is
     RECOMMENDED that the REALM value be the domain name of
     the provider of the STUN server. The response MUST
     include a NONCE, selected by the server.
   </t>

   <t>Using the password associated with the username in
     the USERNAME attribute,
     compute the value for the message-integrity as
     described in
     <xref target="message_integrity"/>. If the
     resulting value does not match the contents of the
     MESSAGE-INTEGRITY attribute, the server MUST
     reject the request with an error response. This
     response MUST use an error code of 401 (Unauthorized). It MUST 
include a REALM
     and NONCE attribute.
   </t>
</list>


<t>If these checks pass, the server continues to process the
   request or indication. Any response generated by the server
   MUST include the MESSAGE-INTEGRITY attribute, computed
   using the username and password utilized to authenticate
   the request. The REALM, NONCE, and USERNAME attributes SHOULD NOT
   be included. </t>

</section>



That work?

-Jonathan R.

Rémi Denis-Courmont wrote:
> On Tuesday 14 August 2007 16:00:18 ext Jonathan Rosenberg wrote:
>> Well, this is a dual-edged sword.
>>
>> Though you are correct, that such a change would stop a user from being
>> able to probe for usernames separately from passwords, it does allow an
>> attacker to cause the server to compute message integrity for any
>> request, as long as the attacker knows a valid username. With the
>> current ordering, they need to know the current username and have a
>> valid nonce. Having a valid nonce helps eliminate the impact of DDOS
>> attacks. With your change, if an attacker can guess a username, they can
>> have a botnet army each send requests with valid usernames and invalid
>> nonces and MI, and cause the server to do lots of hashing without return
>> routability.
> 
> True. Then perhaps NONCE should be checked before both USERNAME and M-I. It 
> cannot depend on the username anyway (since the server must return a valid 
> NONCE to an unauthenticated request).
> 
> No?
> 

-- 
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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On Tuesday 14 August 2007 16:44:31 ext Jonathan Rosenberg wrote:
> That work?

Looks good, thanks.

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


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WGLC finished for draft-ietf-behave-nat-icmp-04.txt with no comments.  I just
requested publication of the document.  The PROTO writeup is below.

-d

-----

PROTO (RFC4858) writeup for:  draft-ietf-behave-nat-icmp-04
August 14, 2007
written by:  Dan Wing, dwing@cisco.com

   (1.a)  Who is the Document Shepherd for this document?

Dan Wing, dwing@cisco.com


          Has the
          Document Shepherd personally reviewed this version of the
          document and, in particular, does he or she believe this
          version is ready for forwarding to the IESG for publication?

Yes.

   (1.b)  Has the document had adequate review both from key WG members
          and from key non-WG members?  Does the Document Shepherd have
          any concerns about the depth or breadth of the reviews that
          have been performed?

This document has been reviewed extensively within BEHAVE, TCPM (Joe
Touch), and authors of draft-ietf-tcpm-icmp-attacks and
RFC4963.



   (1.c)  Does the Document Shepherd have concerns that the document
          needs more review from a particular or broader perspective,
          e.g., security, operational complexity, someone familiar with
          AAA, internationalization, or XML?

No.


   (1.d)  Does the Document Shepherd have any specific concerns or
          issues with this document that the Responsible Area Director
          and/or the IESG should be aware of?  For example, perhaps he
          or she is uncomfortable with certain parts of the document, or
          has concerns whether there really is a need for it.  In any
          event, if the WG has discussed those issues and has indicated
          that it still wishes to advance the document, detail those
          concerns here.  Has an IPR disclosure related to this document
          been filed?  If so, please include a reference to the
          disclosure and summarize the WG discussion and conclusion on
          this issue.

ICMP payload validation (section 4.1 in the document) encourages NATs
to validate ICMP packets.  The document shepherd is concerned this may
cause difficulties with ICMP extensions that the NAT is unaware of.


   (1.e)  How solid is the WG consensus behind this document?  Does it
          represent the strong concurrence of a few individuals, with
          others being silent, or does the WG as a whole understand and
          agree with it?

It represents a relatively solid WG consensus.


   (1.f)  Has anyone threatened an appeal or otherwise indicated extreme
          discontent?  If so, please summarize the areas of conflict in
          separate email messages to the Responsible Area Director.  (It
          should be in a separate email because this questionnaire is
          entered into the ID Tracker.)

No such concerns have been raised.



   (1.g)  Has the Document Shepherd personally verified that the
          document satisfies all ID nits?  (See
          http://www.ietf.org/ID-Checklist.html and
          http://tools.ietf.org/tools/idnits/.)  Boilerplate checks are
          not enough; this check needs to be thorough.  Has the document
          met all formal review criteria it needs to, such as the MIB
          Doctor, media type, and URI type reviews?  If the document
          does not already indicate its intended status at the top of
          the first page, please indicate the intended status here.

All nits are satisfied.

Document is intended to be BCP.


   (1.h)  Has the document split its references into normative and
          informative?  Are there normative references to documents that
          are not ready for advancement or are otherwise in an unclear
          state?  If such normative references exist, what is the
          strategy for their completion?  Are there normative references
          that are downward references, as described in [RFC3967]?  If
          so, list these downward references to support the Area
          Director in the Last Call procedure for them [RFC3967].

All references have been double-checked and appear accurate.


   (1.i)  Has the Document Shepherd verified that the document's IANA
          Considerations section exists and is consistent with the body
          of the document?  If the document specifies protocol
          extensions, are reservations requested in appropriate IANA
          registries?  Are the IANA registries clearly identified?  If
          the document creates a new registry, does it define the
          proposed initial contents of the registry and an allocation
          procedure for future registrations?  Does it suggest a
          reasonable name for the new registry?  See [RFC2434].  If the
          document describes an Expert Review process, has the Document
          Shepherd conferred with the Responsible Area Director so that
          the IESG can appoint the needed Expert during IESG Evaluation?

There are no IANA considerations for this document.


   (1.j)  Has the Document Shepherd verified that sections of the
          document that are written in a formal language, such as XML
          code, BNF rules, MIB definitions, etc., validate correctly in
          an automated checker?

Document contains no formal language.


   (1.k)  The IESG approval announcement includes a Document
          Announcement Write-Up.  Please provide such a Document
          Announcement Write-Up.  Recent examples can be found in the
          "Action" announcements for approved documents.  The approval
          announcement contains the following sections:

          Technical Summary
             Relevant content can frequently be found in the abstract
             and/or introduction of the document.  If not, this may be
             an indication that there are deficiencies in the abstract
             or introduction.


This document specifies the behavioral properties required of the
Network Address Translator (NAT) devices in conjunction with the
ICMP protocol. 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.


          Working Group Summary
             Was there anything in the WG process that is worth noting?
             For example, was there controversy about particular points
             or were there decisions where the consensus was
             particularly rough?

No significant controversy.


          Document Quality
             Are there existing implementations of the protocol?  Have a
             significant number of vendors indicated their plan to
             implement the specification?

This is a BCP document, and describes current practice in existing NATs,
and describes some improvements to current practice.


             Are there any reviewers that
             merit special mention as having done a thorough review,
             e.g., one that resulted in important changes or a
             conclusion that the document had no substantive issues?

They have been acknowledged in the acknowledgements section.

             If
             there was a MIB Doctor, Media Type, or other Expert Review,
             what was its course (briefly)?  In the case of a Media Type
             Review, on what date was the request posted?

          Personnel
             Who is the Document Shepherd for this document?  Who is the
             Responsible Area Director?  If the document requires IANA
             experts(s), insert 'The IANA Expert(s) for the registries
             in this document are <TO BE ADDED BY THE AD>.'

Document shepherd:  Dan Wing, dwing@cisco.com
Responsible AD:  Magnus Westerlund, magnus.westerlund@ericsson.com


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I expect to finish the BEHAVE minutes today, but the relevant part
of the minutes from this discussion in Chicago is below. 

I'd like to make sure the mailing list agrees with removing 
draft-ietf-behave-behavior-discovery and its associated milestone
from standards track.  This means the document status (and 
associated milestone) would change to Experimental or change
to Informational.

  (Per the minutes, below, Bruce and Derek will be preparing
   text and we will decide, as a separate action, if it should
   be Informational or Experimental.)

If you have objections to making draft-ietf-behave-behavior-discovery
a non-standards-track document, please reply to this email.

-d

-----

excerpt from BEHAVE's IETF-69 minutes (not yet published):

...
    * Current draft is proposed standard.  Bruce suggests this
      should be demoted to Experimental.  It's important to have
      a document that explains what was broken about RFC3489,
      and why it was broken.  Still some value in diagnosing how
      NATs work.

    comments on Experimental:

    Remi: The only protocol I'm aware of that does this is
    Teredo, which has its own mechanism, so it's not affected by
    this proposed change.

    Philip:  is anyone using behavior-discovery?

    Bruce:  if anyone needs this functionality, they're doing
    RFC3489.  Bruce knows of 3 people using this, and what
    they're doing is trying to determine if they need a relay
    server.  In some cases, where they know who they will be
    communicating to, this works fine.  In other cases, they
    aren't using it appropriately.  I want to strenghen the
    applicability statement to  make this clearer.

    Philip:  The one interesting use, which could be hurt by
    making this Experimental, is to diagnose problems in running
    ICE.  p2p-sip might be hurt by this lack of diagnostics.

    Bruce:  p2p is one possible application that could benefit
    from behavior-discovery -- with enough tests, you could
    perhaps determine if you could be a superpeer.  However, such
    a determination would be experimental at this point.

    Philip:  How quickly do we need to decide if this should be
    experimental.

    Jonathan:  I think the change to Experimental is appropriate,
    and anything using this is unreliable.  It's the difference
    between working 95% and
    99.99% of the time.

    Magnus:  There are grounds for defining this as an experiment
    and deciding how well this works before moving forward.  "Do
    we have an experiment". It seems we do have one.

    Bruce:  Today this is useful for diagnostics and
    instrumentation.  We do not have a clear view on how this can
    be useful for changing application behavior.

    Francois Audet:  This should be Informational, not
    Experimental.  People have used techniques like this for a
    long time, even before STUN, to detect bad NATs.  The fear is
    people will look at behavior-discovery and say "this is
    easier than ICE, let's use it", but making it Experimental
    won't necessarily accomplish that goal.  It certainly doesn't
    belong in standards-track.

    Dan asked for hum on standards-track:  a quiet, yet strong
    consensus to demote away from standards-track.

    Bruce:  we'll adjust the applicability statement and bring
    that to the list.
...






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ok - I certainly understand making it informational but on  
experimental, I would ask what is the experiment? If the experiment  
is "can an application with a single IP behind a NAT reliably predict  
the NAT behavior?", well then I think we already know the answer to  
this experiment. Anyways, I was not in the room so sorry for missing  
the discussion around this.

Cullen

On Aug 14, 2007, at 6:24 AM, Jonathan Rosenberg wrote:

> I think there was consensus in the room in chicago to make this  
> either informational or experimental.
>
> -Jonathan R.
>
> Cullen Jennings wrote:
>> One of our milestones is
>> Submit standards-track document that describes how an application  
>> can determine the type of NAT it is behind
>> I think it is very hard for an application to reliably do this  
>> except under environments where all the traffic to the NAT is  
>> controlled. (Basically this was the general limitation of RFC  
>> 3489). I hope that the document for this milestone is more an  
>> approach to characterizing and testing behavior of NATs and is not  
>> a general suggestion that an application might be able to figure  
>> out what type of NAT it is behind then modify the applications  
>> behavior based on the type of NAT is is behind. I seem to recall  
>> the old milestone was more along the lines of testing but I can  
>> not find a copy of it.
>> Cullen <with my individual hat on>
>> _______________________________________________
>> 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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Jonathan Rosenberg wrote:
> OK, here is the text now:
> 
> <section title="Receiving a Request">
> 
> <t>
>   After the server has done the basic processing of a request, it
>   performs the checks listed below in the order specified:
> </t>
> 
> <list style="symbols">
>   <t>If the message does not contain a MESSAGE-INTEGRITY attribute,
>     the server MUST
>     generate an error response with an error code of
>     401 (Unauthorized). This response MUST include a REALM value. It is
>     RECOMMENDED that the REALM value be the domain name of
>     the provider of the STUN server. The response MUST
>     include a NONCE, selected by the server.
>   </t>
> 
>   <t>
>     If the message contains a MESSAGE-INTEGRITY
>     attribute, but is missing the USERNAME, REALM or
>     NONCE attributes, the server MUST generate an error
>     response with an error code of 400 (Bad Request).
>   </t>
> 
>   <t>
>     If the NONCE is no longer valid, the server MUST
>     generate an error response with an error code of 438
>     (Stale Nonce). This response MUST include a NONCE and REALM
>     attribute.
>   </t>
> 
>   <t>
>     If the username in the USERNAME attribute is not valid, the server
>     MUST generate an error response with an error code of
>     401 (Unauthorized). This response MUST include a REALM value. It is
>     RECOMMENDED that the REALM value be the domain name of
>     the provider of the STUN server. The response MUST
>     include a NONCE, selected by the server.
>   </t>
> 
>   <t>Using the password associated with the username in
>     the USERNAME attribute,
>     compute the value for the message-integrity as
>     described in
>     <xref target="message_integrity"/>. If the
>     resulting value does not match the contents of the
>     MESSAGE-INTEGRITY attribute, the server MUST
>     reject the request with an error response. This
>     response MUST use an error code of 401 (Unauthorized). It MUST
> include a REALM
>     and NONCE attribute.
>   </t>
> </list>
> 
> 
> <t>If these checks pass, the server continues to process the
>   request or indication. Any response generated by the server

Per 10.2, the long-term credential mechanism cannot be used to protect
indications.

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


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FYI, the second WGLC for draft-ietf-behave-p2p-state-03.txt completed with no
comments received during the WGLC period.  I have requested publication of
draft-ietf-behave-p2p-state-03.txt.  PROTO writeup is below.

-d

-----

PROTO (RFC4858) writeup for:  draft-ietf-behave-p2p-state-03.txt
August 14, 2007
written by:  Dan Wing, dwing@cisco.com

   (1.a)  Who is the Document Shepherd for this document?

Dan Wing, dwing@cisco.com

          Has the
          Document Shepherd personally reviewed this version of the
          document and, in particular, does he or she believe this
          version is ready for forwarding to the IESG for publication?

Yes.


   (1.b)  Has the document had adequate review both from key WG members
          and from key non-WG members?  Does the Document Shepherd have
          any concerns about the depth or breadth of the reviews that
          have been performed?

Yes, it has been reviewed by some members of the BEHAVE working group and some
members of SIP-P2P.  The document shepherd believes the document accurately
describes today's state-of-the-art for NAT traversal.


   (1.c)  Does the Document Shepherd have concerns that the document
          needs more review from a particular or broader perspective,
          e.g., security, operational complexity, someone familiar with
          AAA, internationalization, or XML?

No concerns.


   (1.d)  Does the Document Shepherd have any specific concerns or
          issues with this document that the Responsible Area Director
          and/or the IESG should be aware of?  For example, perhaps he
          or she is uncomfortable with certain parts of the document, or
          has concerns whether there really is a need for it.  In any
          event, if the WG has discussed those issues and has indicated
          that it still wishes to advance the document, detail those
          concerns here.  Has an IPR disclosure related to this document
          been filed?  If so, please include a reference to the
          disclosure and summarize the WG discussion and conclusion on
          this issue.

This document was WGLC'd previously, and all substantive comments
were integrated into the document.  During this second WGLC, there
were no additional comments.


   (1.e)  How solid is the WG consensus behind this document?  Does it
          represent the strong concurrence of a few individuals, with
          others being silent, or does the WG as a whole understand and
          agree with it?

Most of the working group appears satisified with the document as
being useful to explain how applications are traversing NATs today.



   (1.f)  Has anyone threatened an appeal or otherwise indicated extreme
          discontent?  If so, please summarize the areas of conflict in
          separate email messages to the Responsible Area Director.  (It
          should be in a separate email because this questionnaire is
          entered into the ID Tracker.)

There have been no such concerns.


   (1.g)  Has the Document Shepherd personally verified that the
          document satisfies all ID nits?  (See
          http://www.ietf.org/ID-Checklist.html and
          http://tools.ietf.org/tools/idnits/.)

The only remaining nits are to older versions of the drafts (-03
instead of -04).


          Boilerplate checks are
          not enough; this check needs to be thorough.  Has the document
          met all formal review criteria it needs to, such as the MIB
          Doctor, media type, and URI type reviews?

Yes; none are required.


          If the document
          does not already indicate its intended status at the top of
          the first page, please indicate the intended status here.

Informational.


   (1.h)  Has the document split its references into normative and
          informative?

Yes.


          Are there normative references to documents that
          are not ready for advancement or are otherwise in an unclear
          state?

It contains a reference to RFC2766, which has recently been declared
Historic; however, the document is Informational so this isn't a
significant concern.


          If such normative references exist, what is the
          strategy for their completion?  Are there normative references
          that are downward references, as described in [RFC3967]?  If
          so, list these downward references to support the Area
          Director in the Last Call procedure for them [RFC3967].

The document is Informational, so this isn't a concern.


   (1.i)  Has the Document Shepherd verified that the document's IANA
          Considerations section exists and is consistent with the body
          of the document?  If the document specifies protocol
          extensions, are reservations requested in appropriate IANA
          registries?  Are the IANA registries clearly identified?  If
          the document creates a new registry, does it define the
          proposed initial contents of the registry and an allocation
          procedure for future registrations?  Does it suggest a
          reasonable name for the new registry?  See [RFC2434].  If the
          document describes an Expert Review process, has the Document
          Shepherd conferred with the Responsible Area Director so that
          the IESG can appoint the needed Expert during IESG Evaluation?


Document is Informational and has no IANA considerations.


   (1.j)  Has the Document Shepherd verified that sections of the
          document that are written in a formal language, such as XML
          code, BNF rules, MIB definitions, etc., validate correctly in
          an automated checker?

There is no formal language in the document.


   (1.k)  The IESG approval announcement includes a Document
          Announcement Write-Up.  Please provide such a Document
          Announcement Write-Up.  Recent examples can be found in the
          "Action" announcements for approved documents.  The approval
          announcement contains the following sections:

          Technical Summary
             Relevant content can frequently be found in the abstract
             and/or introduction of the document.  If not, this may be
             an indication that there are deficiencies in the abstract
             or introduction.


This memo documents the various methods known to be in use by
applications to establish direct UDP or TCP communication in the
presence of Network Address Translators (NATs).


          Working Group Summary
             Was there anything in the WG process that is worth noting?
             For example, was there controversy about particular points
             or were there decisions where the consensus was
             particularly rough?

Agreeing on RFC3489 terminology (which is well known) versus
RFC4787 terminology (which is more accurate but longer) involved some
controversy.  Consensus was to use RFC4787 terminology.


          Document Quality
             Are there existing implementations of the protocol?  Have a
             significant number of vendors indicated their plan to
             implement the specification?

Yes, in a manner of speaking -- the document explains how existing
software traverses NATs.

             Are there any reviewers that
             merit special mention as having done a thorough review,
             e.g., one that resulted in important changes or a
             conclusion that the document had no substantive issues?

They have been acknowledged in the Acknowledgements section.

             If
             there was a MIB Doctor, Media Type, or other Expert Review,
             what was its course (briefly)?  In the case of a Media Type
             Review, on what date was the request posted?

The document is Informational and only describes the state-of-the-art
for traversal of NATs for UDP and TCP.

          Personnel
             Who is the Document Shepherd for this document?  Who is the
             Responsible Area Director?  If the document requires IANA
             experts(s), insert 'The IANA Expert(s) for the registries
             in this document are <TO BE ADDED BY THE AD>.'

Document shepherd:  Dan Wing, dwing@cisco.com
Responsible AD:  Magnus Westerlund, magnus.westerlund@ericsson.com


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From: "Dan Wing" <dwing@cisco.com>
To: "'Cullen Jennings'" <fluffy@cisco.com>,
	"'Jonathan Rosenberg'" <jdrosen@cisco.com>
Subject: RE: [BEHAVE] Minor concern about one of the milestones
Date: Tue, 14 Aug 2007 23:01:53 -0700
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When Derek and Bruce finish their applicability text, we may know
better if this is, in fact, something we can declare as an Experiment
(hopefully with exit criteria), or if this would be Informational.

(That is, as long as we have consensus around removing the document
from standards-track!)

-d


> -----Original Message-----
> From: Cullen Jennings [mailto:fluffy@cisco.com] 
> Sent: Tuesday, August 14, 2007 11:46 AM
> To: Jonathan Rosenberg
> Cc: Behave WG
> Subject: Re: [BEHAVE] Minor concern about one of the milestones
> 
> 
> ok - I certainly understand making it informational but on  
> experimental, I would ask what is the experiment? If the experiment  
> is "can an application with a single IP behind a NAT reliably 
> predict  
> the NAT behavior?", well then I think we already know the answer to  
> this experiment. Anyways, I was not in the room so sorry for missing  
> the discussion around this.
> 
> Cullen
> 
> On Aug 14, 2007, at 6:24 AM, Jonathan Rosenberg wrote:
> 
> > I think there was consensus in the room in chicago to make this  
> > either informational or experimental.
> >
> > -Jonathan R.
> >
> > Cullen Jennings wrote:
> >> One of our milestones is
> >> Submit standards-track document that describes how an application  
> >> can determine the type of NAT it is behind
> >> I think it is very hard for an application to reliably do this  
> >> except under environments where all the traffic to the NAT is  
> >> controlled. (Basically this was the general limitation of RFC  
> >> 3489). I hope that the document for this milestone is more an  
> >> approach to characterizing and testing behavior of NATs 
> and is not  
> >> a general suggestion that an application might be able to figure  
> >> out what type of NAT it is behind then modify the applications  
> >> behavior based on the type of NAT is is behind. I seem to recall  
> >> the old milestone was more along the lines of testing but I can  
> >> not find a copy of it.
> >> Cullen <with my individual hat on>
> >> _______________________________________________
> >> 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
> 
> 
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave


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On Tuesday 14 August 2007 16:24:03 ext Jonathan Rosenberg wrote:
> An important point of clarification here.
>
> Though FINGERPRINT is currently the only attribute beyond the MI, there
> could potentially be others in the future so you can't depend on it
> being the only one. So inserting a dummy FINGERPRINT won't help.

Ok, that's right.

With this in mind, I think the same considerations need to be made for the=
=20
=46INGERPRINT attribute: if we ever had something past it, it will need to =
hash=20
the message length the way M-I does.

Otherwise, anything that adds optional attributes after FINGERPRINT will be=
=20
undeployable for not being backward compatible (-> Uho, STUN NAT control...=
).

Regards,

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


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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)	: T. Eckert, D. Wing
	Filename	: draft-ietf-behave-multicast-09.txt
	Pages		: 16
	Date		: 2007-8-15
	
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.

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I am starting the 3rd WGLC on draft-ietf-behave-multicast-09, "IP Multicast
Requirements for a Network Address (and port) Translator NAT".  The WGLC ends
on September 3.  The previous WGLCs were for -06 and -03 (which also went to
IESG).


Draft-ietf-behave-multicast-09 integrates all of the comments received to
date.  Significant changes from -08 include:

* Consideration of Gorry Fairhurst's comments in Chicago about a multicast
source inside of a multi-homed network; unfortunately the best I could
describe to deal with that problem was the new requirement 5-a:

           "a:  If a network is multihomed, the NATs or the network
                configuration MUST ensure that duplicate instances of
                the multicast data traffic does not appear on the public
                network.  This can be accomplished by network design (an
                access control list) or a protocol between the NATs
                (e.g., the NAT with the higher IP address transmits
                multicast data towards the public network and other NATs
                drop it)."

* a NAT with multiple public interfaces SHOULD pair for multicast in the same
way it pairs for unicast (new requirement 4-c).

* The "application considerations" section has been moved to an appendix, as
it seems outside the scope of a document describing the behavior for NATs.


Differences between -09 and -08:
 
http://tools.ietf.org/rfcdiff?url2=http://tools.ietf.org/id/draft-ietf-behave-
multicast-09.txt
  http://tinyurl.com/yqej5l

Differences between -09 and -06:
 
http://tinyurl.com/create.php?url=http://tools.ietf.org/rfcdiff?url1=http://to
ols.ietf.org/id/draft-ietf-behave-multicast-06.txt&url2=http://tools.ietf.org/
id/draft-ietf-behave-multicast-09.txt
  http://tinyurl.com/yulw5t


Please reply with any comments.
-d



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From: "Dan Wing" <dwing@cisco.com>
To: "'Behave WG'" <behave@ietf.org>
Date: Wed, 15 Aug 2007 17:52:51 -0700
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I have posted the draft minutes from BEHAVE's IETF-69 meeting in Chicago to:
http://www3.ietf.org/proceedings/07jul/minutes/behave.txt

Thanks to all the presenters!


Please reply with any changes to the minutes. 


Audio of the session is here:
http://limestone.uoregon.edu/ftp/pub/videolab/media/ietf69/ietf69-ch5-thurs-af
noon-plenary.mp3

-d


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Date: Wed, 15 Aug 2007 20:28:17 -0700 (PDT)
From: Pyda Srisuresh <srisuresh@yahoo.com>
Subject: Re: [Sip] RE: [BEHAVE] Re: ICE deployment data before LC for RFC
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Magnus,

I was out of town and just got back. Just catching up on the e-mails. Please
see my responses inline below.

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

> Suresh,
> 
> I would propose that you drop MMUSIC mailing listif responding to this 
> as you appear to want to discuss the removal of a BEHAVE milestone, 
> rather than ICE and its way forward. But I will leave it there to let 
> the MMUSIC people see my response.
> 
[suresh] I am not opposed to ICE moving forward. Please see below for my
objection to ICE.

> It seems most of the issues in this discussion is between, what I at 
> least thought was a specific way of performing the hole punching rather 
> than the general concept that I acknowledge has existed since the 
> creation of NATs. I will drop these items an respond to few of them.

[suresh] Sorry, the hole punchign techniques havent been there since the
conception of NATs. They came about  principally in the context of P2P
applications, much aftet the original NATs.
> 
> Pyda Srisuresh skrev:
> > --- Magnus Westerlund <magnus.westerlund@ericsson.com> wrote:
> >>
> >> There also has been discussion about if there should be multiple 
> >> solutions etc. But, to my knowledge the consensus also there has been 
> >> that one solution is the way to go.
> >>
> > [suresh] I have not been part of the discussion pertaining to the merit of
> > multiple solutions vs single soultions. This may have been discussed in the
> > context of SIP and NAT traversal. So, I cant comment on this.
> 
> Yes, that was in the context of ICE. As my email was about that.
> 
> > 
> >> So Suresh, you and everyone else as participants in the IETF is 
> >> responsible for what you call HOLE PUNCHING TECHNIQUES are not an WG 
> >> item. 
> > 
> > [suresh] Magnus - I dont know what you are talking about. How do you mean I
> am
> > responsible for why the HOLE PUNCHING TECHNIQUE draft is not a WG item. The
> > ford-behave-app draft has been around for many years since the inception of
> > BEHAVE. Last I checked, The application milestone is no longer on BEHVAE
> > charter. I certainly am not responsible for removign the milestone.
> 
> This was written as reaction on how work gets done in the IETF. One or 
> more proponents goes out and sell their idea so that enough people are 
> interested to participate in the work on that idea. In cases the 
> proponents are not succesful in getting traction for the idea it rarely 
> goes anywhere. In the case of the ford-behave-app draft there has been 
> no support for keeping the milestone or more important, do the work.
> 

[suresh] I personally know of several applications using the TCP/UDP HOLE
PUNCHING techniques directly (not using ICE). Irrespective of whether the IETF
endorses design guidelines using HOLE PUNCHING techniques or not, this work
will continue. It is ironic to me why the WG chose not to take up design
guidelines using HOLE PUNCHING techniques as WG item. I cant explain it.  

> 
> > [suresh] Magnus - I have not been part of the consensus for ICE. However, I
> do
> > know ICE adapts the HOLE PUNCHING TECHNIQUES and the BCP recommendations
> > identified in ford-behave-app draft. Yet, neither of the ICE drafts (UDP or
> the
> > TCP version) refer the ford-behave-app draft or prior work on HOLE PUNCHING
> as
> > normative reference. This iS WRONG. 
> > 
> > I believe, the ICE drafts should refer the prior work (kegel work on UDP
> hole
> > punching, USENIX paper for TCP hole punching and ford-behave-app draft for
> > application traversal design) on which the ICE work is based.
> > 
> 
> I think you should review your sources here. The first version of 
> ford-beahve-app was submitted february 2005. At that point STUN (RFC 
> 3489 had been published in almost 2 years (March 2003). The first 
> individual version of ICE was submitted to IETF february 2003. I am not 
> going to untangle this, but if you are going down this path: Please have 
> you facts right.
> 
[suresh] Below is the chronology of the sources I am refering to. You can see
for yourself what I am talking about. Nothing complex to untagle.

a) http://alumnus.caltech.edu/~dank/peer-nat.html (Dec 98)- First documented
reference to UDP hole punching technique. 

b) draft-ford-midcom-nat-p2p-00.txt (Sept 03) - First known reference to TCP
hole punching technique and P2P application design guidelines

c) Peer-to-peer communication across NATs USENIX '05 paper (04/2005) - This is
derived from draft-ford-midcom-nat-p2p draft. Considerign that
draft-ford-midcom-nat-p2p draft expired, this is the first formal documented
reference to TCP hole punching

d) draft-ford-behave-app-05.txt (March '07) - This is the currently active
application design guidelines draft, derived from draft-ford-midcom-nat-p2p
draft.

e) draft-rosenberg-sipping-ice-00 (Feb 03) - First rev of ICE over UDP draft.
This uses UDP hole punching technique. The ICE draft currently in MMUSIC WG has
evolved and folds in a number of application design guidelines in
ford-midcom-nat-p2p and ford-behave-app drafts, besides the UDP HOLE PUNCHING
technique.

f) draft-ietf-mmusic-ice-tcp-00 (Mar 06) - First rev of ICE over TCP draft.
This uses TCP hole punching technique and folds in a number of application
design guidelines in ford-midcom-nat-p2p and ford-behave-app drafts.

I  believe, there should be acknowledgement where the acknowledgement is due.
The ICE drafts use the HOLE PUNCHING technique and adapt guidelines specified
in ford-behave-app draft. But none of a), c) or e)  are referred by the ICE
drafts (ICE and ICE-TCP).

> I am not denying that work commonly builds up on others. And that it is 
> common curtesy to reference such work. However, that is commonly done by 
> an informative reference. Normative references are for things that you 
> absolutely need to include to build interoperable implementations. 
> Please do review IESG statement on references:
> 
>
http://www.ietf.org/IESG/STATEMENTS/iesg-statement-normative-informative-references.txt
> 

[suresh] Thanks for the pointer. The description is subject to individual
interpretation. For example, it says "Normative references specify documents
that must be read to understand or implement the technology in the new RFC". If
you describe the technique used inside a draft, does it make the technique
non-normative? I dont want to get into the nitty grittys on this.

I will be fine, if the ICE drafts referred the docs I mentioned even as
informational references.

> 
> 
> > On the one hand, IETF (IESG) has been adamant about the standard level of
> the
> > normative references for a proposes standard. Yet, it is ironic that when
> HOLE
> > PUNCHING  is normative to the design of ICE, the IETF is not allowing the
> HOLE
> > PUNCHING work to be standardized. 
> 
> Once more a missunderstanding about what it means to be a normative 
> reference in an RFC.
> 
[suresh] Please see my comment above.

> > 
> > Why is the IETF not allowing HOLE PUNCHING TECHNIQUES and P2P application
> > traversal identified in draft-ford-behave-app draft to move forward as a
> BCP WG
> > item?
> > 
> 
> As stated before. There is a clear lack of interest in this document. I 
> wished there wasn't. But the fact is that no-one except you have really 
> argued for it, not even your co-authors. We can't publish a document 
> where no-one reads and reviews it and stil claim that their exist 
> consensus behind it.

[suresh] Magnus - My co-authors are not arguing for it because they have given
up they can get anywhere with arguing. It is just that I didnt. 

regards,
suresh 

> 
> Regards
> 
> 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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Subject: [BEHAVE] Re: [MBONED] WGLC for draft-ietf-behave-multicast-09
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Dan Wing wrote:
> I am starting the 3rd WGLC on draft-ietf-behave-multicast-09, "IP Multicast
> Requirements for a Network Address (and port) Translator NAT".  The WGLC ends
> on September 3.  The previous WGLCs were for -06 and -03 (which also went to
> IESG).
[...]

Here are my thoughts upon reading this draft.

In section 3 background it says:

   This specifically means that IGMP protocol packets sent by the NAT
   device will always use IP address of the interface (inside or
   outside) to which they are sent, but because those packets are
           ^^^^^
   logically "sourced" and not "forwarded", NAT does not have any impact
   into this.

I might be confused here, but shouldn't it be "_from_ which they are
sent"?

In 4.1.1:

   REQ-1:   For IP multicast packets that are forward to a host(s) on
                                              ^^^^^^^^^
forwarded

My only substantial comment is regarding 4.3 and RTP.

I think it should be noted that there might be SSRC collisions that
are not detected since RTP packets from different hosts behind a NAT
might appear to be from the same. One could then note that CNAME
(defined in RFC 3550) might be of help. One interesting thing here
is that hosts might use their IP address to construct the CNAME. This
means that their private addresses may be exposed. I don't know how
serious a problem that is though... However, it is not unlikely that
two hosts behind two different NATs have the same private IP
address. If they both source RTP, they may pick the same CNAME which
is a problem.

RFC 3550 says in 6.5.1:

   applications MAY provide a means to configure a unique CNAME, but the
   burden is on the translator to translate CNAMEs from private
   addresses to public addresses if necessary to keep private addresses
   from being exposed.

Doing this however, could mean that two multicast sources behind the
same NAT end up sending RTP with the same CNAME, so I don't know how
good that is.

Anyhow, I believe this draft should note some of the issues with CNAME.

Finally, one tiny editorial comment. At least I would write
"Source-Specific Multicast" rather than "source specific multicast", and
"Any Source Multicast" rather than "any source multicast". At least I
find this easier to read.

Stig


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Remi - any comment on the other part of my mail, on the alternative hash 
computation?

-Jonathan R.

RĂ©mi Denis-Courmont wrote:
> On Tuesday 14 August 2007 16:24:03 ext Jonathan Rosenberg wrote:
>> An important point of clarification here.
>>
>> Though FINGERPRINT is currently the only attribute beyond the MI, there
>> could potentially be others in the future so you can't depend on it
>> being the only one. So inserting a dummy FINGERPRINT won't help.
> 
> Ok, that's right.
> 
> With this in mind, I think the same considerations need to be made for the 
> FINGERPRINT attribute: if we ever had something past it, it will need to hash 
> the message length the way M-I does.
> 
> Otherwise, anything that adds optional attributes after FINGERPRINT will be 
> undeployable for not being backward compatible (-> Uho, STUN NAT control...).
> 
> Regards,
> 

-- 
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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Fixed, thanks.

-Jonathan R.

Marc Petit-Huguenin wrote:
> Jonathan Rosenberg wrote:
>> OK, here is the text now:
>>
>> <section title="Receiving a Request">
>>
>> <t>
>>   After the server has done the basic processing of a request, it
>>   performs the checks listed below in the order specified:
>> </t>
>>
>> <list style="symbols">
>>   <t>If the message does not contain a MESSAGE-INTEGRITY attribute,
>>     the server MUST
>>     generate an error response with an error code of
>>     401 (Unauthorized). This response MUST include a REALM value. It is
>>     RECOMMENDED that the REALM value be the domain name of
>>     the provider of the STUN server. The response MUST
>>     include a NONCE, selected by the server.
>>   </t>
>>
>>   <t>
>>     If the message contains a MESSAGE-INTEGRITY
>>     attribute, but is missing the USERNAME, REALM or
>>     NONCE attributes, the server MUST generate an error
>>     response with an error code of 400 (Bad Request).
>>   </t>
>>
>>   <t>
>>     If the NONCE is no longer valid, the server MUST
>>     generate an error response with an error code of 438
>>     (Stale Nonce). This response MUST include a NONCE and REALM
>>     attribute.
>>   </t>
>>
>>   <t>
>>     If the username in the USERNAME attribute is not valid, the server
>>     MUST generate an error response with an error code of
>>     401 (Unauthorized). This response MUST include a REALM value. It is
>>     RECOMMENDED that the REALM value be the domain name of
>>     the provider of the STUN server. The response MUST
>>     include a NONCE, selected by the server.
>>   </t>
>>
>>   <t>Using the password associated with the username in
>>     the USERNAME attribute,
>>     compute the value for the message-integrity as
>>     described in
>>     <xref target="message_integrity"/>. If the
>>     resulting value does not match the contents of the
>>     MESSAGE-INTEGRITY attribute, the server MUST
>>     reject the request with an error response. This
>>     response MUST use an error code of 401 (Unauthorized). It MUST
>> include a REALM
>>     and NONCE attribute.
>>   </t>
>> </list>
>>
>>
>> <t>If these checks pass, the server continues to process the
>>   request or indication. Any response generated by the server
> 
> Per 10.2, the long-term credential mechanism cannot be used to protect
> indications.
> 

-- 
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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Actually, forget it. I withdraw this idea. Its a big change with 
questionable value, late in the game.

-Jonathan R.

Jonathan Rosenberg wrote:
> Remi - any comment on the other part of my mail, on the alternative hash 
> computation?
> 
> -Jonathan R.
> 
> RĂ©mi Denis-Courmont wrote:
>> On Tuesday 14 August 2007 16:24:03 ext Jonathan Rosenberg wrote:
>>> An important point of clarification here.
>>>
>>> Though FINGERPRINT is currently the only attribute beyond the MI, there
>>> could potentially be others in the future so you can't depend on it
>>> being the only one. So inserting a dummy FINGERPRINT won't help.
>>
>> Ok, that's right.
>>
>> With this in mind, I think the same considerations need to be made for 
>> the FINGERPRINT attribute: if we ever had something past it, it will 
>> need to hash the message length the way M-I does.
>>
>> Otherwise, anything that adds optional attributes after FINGERPRINT 
>> will be undeployable for not being backward compatible (-> Uho, STUN 
>> NAT control...).
>>
>> Regards,
>>
> 

-- 
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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Stig Venaas wrote:

> Dan Wing wrote:
> > I am starting the 3rd WGLC on 
> > draft-ietf-behave-multicast-09, "IP Multicast
> > Requirements for a Network Address (and port) Translator 
> > NAT".  The WGLC ends
> > on September 3.  The previous WGLCs were for -06 and -03 
> > (which also went to
> > IESG).
> [...]
> 
> Here are my thoughts upon reading this draft.
> 
> In section 3 background it says:
> 
>    This specifically means that IGMP protocol packets sent by the NAT
>    device will always use IP address of the interface (inside or
>    outside) to which they are sent, but because those packets are
>            ^^^^^
>    logically "sourced" and not "forwarded", NAT does not have 
>    any impact into this.
> 
> I might be confused here, but shouldn't it be "_from_ which they are
> sent"?
> 
> In 4.1.1:
> 
>    REQ-1:   For IP multicast packets that are forward to a host(s) on
>                                               ^^^^^^^^^
> forwarded

Thanks, I made those adjustments in -10, which I expect will
be sent to IESG.

> My only substantial comment is regarding 4.3 and RTP.
> 
> I think it should be noted that there might be SSRC collisions that
> are not detected since RTP packets from different hosts behind a NAT
> might appear to be from the same. One could then note that CNAME
> (defined in RFC 3550) might be of help. One interesting thing here
> is that hosts might use their IP address to construct the CNAME. This
> means that their private addresses may be exposed. I don't know how
> serious a problem that is though... However, it is not unlikely that
> two hosts behind two different NATs have the same private IP
> address. If they both source RTP, they may pick the same CNAME which
> is a problem.
> 
> RFC 3550 says in 6.5.1:
> 
>    applications MAY provide a means to configure a unique 
> CNAME, but the
>    burden is on the translator to translate CNAMEs from private
>    addresses to public addresses if necessary to keep private 
> addresses
>    from being exposed.
> 
> Doing this however, could mean that two multicast sources behind the
> same NAT end up sending RTP with the same CNAME, so I don't know how
> good that is.
>
> Anyhow, I believe this draft should note some of the issues 
> with CNAME.

I agreed.  

Would the following additional text in the non-normative
Application Considerations appendix be sufficient:

   As many NATs use the same private address space (e.g.,
   192.168.0.0/16, [RFC1918]), RTP stacks are encouraged to generate
   CNAMEs properly (see end of Section 6.5.1 of [RFC3550].)

Please let me know if that is suitable.

> Finally, one tiny editorial comment. At least I would write
> "Source-Specific Multicast" rather than "source specific 
> multicast", and
> "Any Source Multicast" rather than "any source multicast". At least I
> find this easier to read.

Done.  That does help readability.

Thanks for your review.

-d


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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-09.txt
	Pages           : 47
	Date            : 2007-08-17

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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Hi,
=20
Just another comment:
=20
If the sender behind the NAT uses Ethernet and maps de lower 28 bits IP =
multicast address ( first 4 bits are always 1110) to 23 bits in the =
multicast MAC address, =BFwhich 5 high order bits uses the NAT to send =
the IP multicast packets?=20
=20
Considering that SSM range is 232/8. How the NAT knows if the paquet is =
within the SSM range?
=20
Regards
=20
Alvaro

________________________________

De: Dan Wing [mailto:dwing@cisco.com]
Enviado el: lun 13/08/2007 19:30
Para: 'Brian Haberman'
CC: 'Marshall Eubanks'; mboned@ietf.org; Alvaro Fernandez; =
magma@ietf.org; behave@ietf.org; 'Marshall Eubanks'; 'Toerless Eckert'
Asunto: RE: [BEHAVE] Re: [magma] RE: [MBONED] FW: =
WGLC:draft-ietf-behave-multicast-06.txt



> -----Original Message-----
> From: Brian Haberman [mailto:brian@innovationslab.net]
> Sent: Thursday, June 07, 2007 6:16 AM
> To: Dan Wing
> Cc: 'Marshall Eubanks'; mboned@ietf.org; 'Alvaro Fernandez';
> magma@ietf.org; behave@ietf.org; 'Marshall Eubanks'; Toerless Eckert
> Subject: Re: [BEHAVE] Re: [magma] RE: [MBONED] FW:
> WGLC:draft-ietf-behave-multicast-06.txt
....
>
>
>
> Dan Wing wrote:
> >> The NAPT knows what is SSM. Couldn't it also do a Multicast Group=20
> >> Translation for SSM groups ? Why
> >> is this more difficult than, say, port translation ?
> >
> > It may be fairly easy -- except for rewriting the
> > advertisements in SAP,
> > HTTP, HTTPS (a big problem there), email, SIP, RTSP, ....
> >
> > And NAPTs don't do that kind of rewriting today;
> > draft-ietf-behave-multicast
> > is only a BCP and trying to describe what a NAT should best
> > be doing as of
> > its date of publication.
> >
> >
> > If we want NAPTs to do this, we would want a different
> > document that was
> > standards-track and that document would define how we
> > wanted NAPTs to
> > perform this function.  However, I don't see a successful
> > path if such an
> > operation would require the NAPT to have an ALG for SAP,
> > HTTP, SIP, and RTSP
> > and requires the NAT also rewrite those packets.  We'd need
> > something else.
> > ALGs don't work well for a variety of reasons.
>
> Given that, I would suggest that the document state that
> administrators of the network behind the NAPT assign sub-ranges
> of the SSM space to their multicast sources.  This could be done
> manually or *possibly* with the tools developed by the MALLOC
> WG.  For the ma & pa type residential networks, I doubt you will
> see multiple machines behind a NAPT sourcing SSM.  For more
> corporate types, the network admin can control the allocation of
> the SSM range.

I moved text discussing what applications should do into an appendix,
as the document's BCP scope is really around NATs (rather than
applications that need to handle NATs).  Based on your comment above,
and some other input, the text now has the following paragraph.  I
believe this captures your suggestion.

  "If multiple SSM sources on the inside of a NAT choose the same
   multicast group address, those sources are uniquely identifiable
   because their IP addresses are unique.  However, if their multicast
   traffic is NATted and sent on the NAT's public interface, the traffic
   from those individual sources is no longer uniquely identifiable.
   This will cause problems for multicast receivers which will see an
   intermixing of traffic from those sources.  Resolution of this issue
   is left for future study.  In the meantime, applications that source
   SSM multicast traffic are encouraged to allow the user to modify the
   multicast SSM address so that users can avoid this problem if that
   application is placed behind a NAT."

The text doesn't quite go so far to say that a range has to be assigned =
a
priori.  But let me know if it is acceptable.

I have not submitted -09, but a preview is here if anyone wants to see =
the
full context:
http://svn.resiprocate.org/rep/ietf-drafts/behave/draft-ietf-behave-multi=
cast-
09.txt

-d



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<HTML dir=3Dltr><HEAD><TITLE>RE: [BEHAVE] Re: [magma] RE: [MBONED] FW: =
WGLC:draft-ietf-behave-multicast-06.txt</TITLE>=0A=
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dunicode">=0A=
<META content=3D"MSHTML 6.00.6000.16481" name=3DGENERATOR></HEAD>=0A=
<BODY>=0A=
<DIV id=3DidOWAReplyText53736 dir=3Dltr>=0A=
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 =
size=3D2>Hi,</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Just another =
comment:</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>If the sender behind the NAT =
uses Ethernet and maps de lower 28 bits IP multicast address ( first =
4&nbsp;bits are always 1110) to 23 bits in the multicast MAC address, =
=BFwhich 5 high order bits uses the NAT to send the IP multicast =
packets? </FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Considering&nbsp;that SSM =
range is 232/8. How the NAT knows if the paquet is within the SSM =
range?</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Regards</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Alvaro</FONT></DIV></DIV>=0A=
<DIV dir=3Dltr><BR>=0A=
<HR tabIndex=3D-1>=0A=
<FONT face=3DTahoma size=3D2><B>De:</B> Dan Wing =
[mailto:dwing@cisco.com]<BR><B>Enviado el:</B> lun 13/08/2007 =
19:30<BR><B>Para:</B> 'Brian Haberman'<BR><B>CC:</B> 'Marshall Eubanks'; =
mboned@ietf.org; Alvaro Fernandez; magma@ietf.org; behave@ietf.org; =
'Marshall Eubanks'; 'Toerless Eckert'<BR><B>Asunto:</B> RE: [BEHAVE] Re: =
[magma] RE: [MBONED] FW: =
WGLC:draft-ietf-behave-multicast-06.txt<BR></FONT><BR></DIV>=0A=
<DIV>=0A=
<P><FONT size=3D2>&gt; -----Original Message-----<BR>&gt; From: Brian =
Haberman [<A =
href=3D"mailto:brian@innovationslab.net">mailto:brian@innovationslab.net<=
/A>]<BR>&gt; Sent: Thursday, June 07, 2007 6:16 AM<BR>&gt; To: Dan =
Wing<BR>&gt; Cc: 'Marshall Eubanks'; mboned@ietf.org; 'Alvaro =
Fernandez';<BR>&gt; magma@ietf.org; behave@ietf.org; 'Marshall Eubanks'; =
Toerless Eckert<BR>&gt; Subject: Re: [BEHAVE] Re: [magma] RE: [MBONED] =
FW:<BR>&gt; =
WGLC:draft-ietf-behave-multicast-06.txt<BR>....<BR>&gt;<BR>&gt;<BR>&gt;<B=
R>&gt; Dan Wing wrote:<BR>&gt; &gt;&gt; The NAPT knows what is SSM. =
Couldn't it also do a Multicast Group&nbsp;<BR>&gt; &gt;&gt; Translation =
for SSM groups ? Why<BR>&gt; &gt;&gt; is this more difficult than, say, =
port translation ?<BR>&gt; &gt;<BR>&gt; &gt; It may be fairly easy -- =
except for rewriting the<BR>&gt; &gt; advertisements in SAP,<BR>&gt; =
&gt; HTTP, HTTPS (a big problem there), email, SIP, RTSP, ....<BR>&gt; =
&gt;<BR>&gt; &gt; And NAPTs don't do that kind of rewriting =
today;<BR>&gt; &gt; draft-ietf-behave-multicast<BR>&gt; &gt; is only a =
BCP and trying to describe what a NAT should best<BR>&gt; &gt; be doing =
as of<BR>&gt; &gt; its date of publication.<BR>&gt; &gt;<BR>&gt; =
&gt;<BR>&gt; &gt; If we want NAPTs to do this, we would want a =
different<BR>&gt; &gt; document that was<BR>&gt; &gt; standards-track =
and that document would define how we<BR>&gt; &gt; wanted NAPTs =
to<BR>&gt; &gt; perform this function.&nbsp; However, I don't see a =
successful<BR>&gt; &gt; path if such an<BR>&gt; &gt; operation would =
require the NAPT to have an ALG for SAP,<BR>&gt; &gt; HTTP, SIP, and =
RTSP<BR>&gt; &gt; and requires the NAT also rewrite those packets.&nbsp; =
We'd need<BR>&gt; &gt; something else.<BR>&gt; &gt; ALGs don't work well =
for a variety of reasons.<BR>&gt;<BR>&gt; Given that, I would suggest =
that the document state that<BR>&gt; administrators of the network =
behind the NAPT assign sub-ranges<BR>&gt; of the SSM space to their =
multicast sources.&nbsp; This could be done<BR>&gt; manually or =
*possibly* with the tools developed by the MALLOC<BR>&gt; WG.&nbsp; For =
the ma &amp; pa type residential networks, I doubt you will<BR>&gt; see =
multiple machines behind a NAPT sourcing SSM.&nbsp; For more<BR>&gt; =
corporate types, the network admin can control the allocation of<BR>&gt; =
the SSM range.<BR><BR>I moved text discussing what applications should =
do into an appendix,<BR>as the document's BCP scope is really around =
NATs (rather than<BR>applications that need to handle NATs).&nbsp; Based =
on your comment above,<BR>and some other input, the text now has the =
following paragraph.&nbsp; I<BR>believe this captures your =
suggestion.<BR><BR>&nbsp; "If multiple SSM sources on the inside of a =
NAT choose the same<BR>&nbsp;&nbsp; multicast group address, those =
sources are uniquely identifiable<BR>&nbsp;&nbsp; because their IP =
addresses are unique.&nbsp; However, if their multicast<BR>&nbsp;&nbsp; =
traffic is NATted and sent on the NAT's public interface, the =
traffic<BR>&nbsp;&nbsp; from those individual sources is no longer =
uniquely identifiable.<BR>&nbsp;&nbsp; This will cause problems for =
multicast receivers which will see an<BR>&nbsp;&nbsp; intermixing of =
traffic from those sources.&nbsp; Resolution of this =
issue<BR>&nbsp;&nbsp; is left for future study.&nbsp; In the meantime, =
applications that source<BR>&nbsp;&nbsp; SSM multicast traffic are =
encouraged to allow the user to modify the<BR>&nbsp;&nbsp; multicast SSM =
address so that users can avoid this problem if that<BR>&nbsp;&nbsp; =
application is placed behind a NAT."<BR><BR>The text doesn't quite go so =
far to say that a range has to be assigned a<BR>priori.&nbsp; But let me =
know if it is acceptable.<BR><BR>I have not submitted -09, but a preview =
is here if anyone wants to see the<BR>full context:<BR><A =
href=3D"http://svn.resiprocate.org/rep/ietf-drafts/behave/draft-ietf-beha=
ve-multicast-">http://svn.resiprocate.org/rep/ietf-drafts/behave/draft-ie=
tf-behave-multicast-</A><BR>09.txt<BR><BR>-d<BR></FONT></P></DIV></BODY><=
/HTML>
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From: "Alvaro Fernandez" <Alvaro@soportemv.com>
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	"Brian Haberman" <brian@innovationslab.net>
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My mistake. The info is in the IP header.
Alvaro

________________________________

De: Alvaro Fernandez [mailto:Alvaro@soportemv.com]
Enviado el: dom 19/08/2007 17:11
Para: Dan Wing; Brian Haberman
CC: behave@ietf.org; Marshall Eubanks; mboned@ietf.org; magma@ietf.org
Asunto: RE: [BEHAVE] Re: [magma] RE: [MBONED] =
FW:WGLC:draft-ietf-behave-multicast-06.txt


Hi,
=20
Just another comment:
=20
If the sender behind the NAT uses Ethernet and maps de lower 28 bits IP =
multicast address ( first 4 bits are always 1110) to 23 bits in the =
multicast MAC address, =BFwhich 5 high order bits uses the NAT to send =
the IP multicast packets?=20
=20
Considering that SSM range is 232/8. How the NAT knows if the paquet is =
within the SSM range?
=20
Regards
=20
Alvaro

________________________________

De: Dan Wing [mailto:dwing@cisco.com]
Enviado el: lun 13/08/2007 19:30
Para: 'Brian Haberman'
CC: 'Marshall Eubanks'; mboned@ietf.org; Alvaro Fernandez; =
magma@ietf.org; behave@ietf.org; 'Marshall Eubanks'; 'Toerless Eckert'
Asunto: RE: [BEHAVE] Re: [magma] RE: [MBONED] FW: =
WGLC:draft-ietf-behave-multicast-06.txt



> -----Original Message-----
> From: Brian Haberman [mailto:brian@innovationslab.net]
> Sent: Thursday, June 07, 2007 6:16 AM
> To: Dan Wing
> Cc: 'Marshall Eubanks'; mboned@ietf.org; 'Alvaro Fernandez';
> magma@ietf.org; behave@ietf.org; 'Marshall Eubanks'; Toerless Eckert
> Subject: Re: [BEHAVE] Re: [magma] RE: [MBONED] FW:
> WGLC:draft-ietf-behave-multicast-06.txt
....
>
>
>
> Dan Wing wrote:
> >> The NAPT knows what is SSM. Couldn't it also do a Multicast Group=20
> >> Translation for SSM groups ? Why
> >> is this more difficult than, say, port translation ?
> >
> > It may be fairly easy -- except for rewriting the
> > advertisements in SAP,
> > HTTP, HTTPS (a big problem there), email, SIP, RTSP, ....
> >
> > And NAPTs don't do that kind of rewriting today;
> > draft-ietf-behave-multicast
> > is only a BCP and trying to describe what a NAT should best
> > be doing as of
> > its date of publication.
> >
> >
> > If we want NAPTs to do this, we would want a different
> > document that was
> > standards-track and that document would define how we
> > wanted NAPTs to
> > perform this function.  However, I don't see a successful
> > path if such an
> > operation would require the NAPT to have an ALG for SAP,
> > HTTP, SIP, and RTSP
> > and requires the NAT also rewrite those packets.  We'd need
> > something else.
> > ALGs don't work well for a variety of reasons.
>
> Given that, I would suggest that the document state that
> administrators of the network behind the NAPT assign sub-ranges
> of the SSM space to their multicast sources.  This could be done
> manually or *possibly* with the tools developed by the MALLOC
> WG.  For the ma & pa type residential networks, I doubt you will
> see multiple machines behind a NAPT sourcing SSM.  For more
> corporate types, the network admin can control the allocation of
> the SSM range.

I moved text discussing what applications should do into an appendix,
as the document's BCP scope is really around NATs (rather than
applications that need to handle NATs).  Based on your comment above,
and some other input, the text now has the following paragraph.  I
believe this captures your suggestion.

  "If multiple SSM sources on the inside of a NAT choose the same
   multicast group address, those sources are uniquely identifiable
   because their IP addresses are unique.  However, if their multicast
   traffic is NATted and sent on the NAT's public interface, the traffic
   from those individual sources is no longer uniquely identifiable.
   This will cause problems for multicast receivers which will see an
   intermixing of traffic from those sources.  Resolution of this issue
   is left for future study.  In the meantime, applications that source
   SSM multicast traffic are encouraged to allow the user to modify the
   multicast SSM address so that users can avoid this problem if that
   application is placed behind a NAT."

The text doesn't quite go so far to say that a range has to be assigned =
a
priori.  But let me know if it is acceptable.

I have not submitted -09, but a preview is here if anyone wants to see =
the
full context:
http://svn.resiprocate.org/rep/ietf-drafts/behave/draft-ietf-behave-multi=
cast-
09.txt

-d



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<HTML dir=3Dltr><HEAD><TITLE>RE: [BEHAVE] Re: [magma] RE: [MBONED] FW: =
WGLC:draft-ietf-behave-multicast-06.txt</TITLE>=0A=
<META http-equiv=3DContent-Type content=3D"text/html; charset=3Dunicode">=0A=
<META content=3D"MSHTML 6.00.6000.16481" name=3DGENERATOR></HEAD>=0A=
<BODY>=0A=
<DIV id=3DidOWAReplyText42993 dir=3Dltr>=0A=
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 size=3D2>My mistake. =
The info is in the IP header.</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Alvaro</FONT></DIV></DIV>=0A=
<DIV dir=3Dltr><BR>=0A=
<HR tabIndex=3D-1>=0A=
<FONT face=3DTahoma size=3D2><B>De:</B> Alvaro Fernandez =
[mailto:Alvaro@soportemv.com]<BR><B>Enviado el:</B> dom 19/08/2007 =
17:11<BR><B>Para:</B> Dan Wing; Brian Haberman<BR><B>CC:</B> =
behave@ietf.org; Marshall Eubanks; mboned@ietf.org; =
magma@ietf.org<BR><B>Asunto:</B> RE: [BEHAVE] Re: [magma] RE: [MBONED] =
FW:WGLC:draft-ietf-behave-multicast-06.txt<BR></FONT><BR></DIV>=0A=
<DIV dir=3Dltr>=0A=
<DIV id=3DidOWAReplyText53736 dir=3Dltr>=0A=
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 =
size=3D2>Hi,</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Just another =
comment:</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>If the sender behind the NAT =
uses Ethernet and maps de lower 28 bits IP multicast address ( first =
4&nbsp;bits are always 1110) to 23 bits in the multicast MAC address, =
=BFwhich 5 high order bits uses the NAT to send the IP multicast =
packets? </FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Considering&nbsp;that SSM =
range is 232/8. How the NAT knows if the paquet is within the SSM =
range?</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Regards</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Alvaro</FONT></DIV></DIV>=0A=
<DIV dir=3Dltr><BR>=0A=
<HR tabIndex=3D-1>=0A=
<FONT face=3DTahoma size=3D2><B>De:</B> Dan Wing =
[mailto:dwing@cisco.com]<BR><B>Enviado el:</B> lun 13/08/2007 =
19:30<BR><B>Para:</B> 'Brian Haberman'<BR><B>CC:</B> 'Marshall Eubanks'; =
mboned@ietf.org; Alvaro Fernandez; magma@ietf.org; behave@ietf.org; =
'Marshall Eubanks'; 'Toerless Eckert'<BR><B>Asunto:</B> RE: [BEHAVE] Re: =
[magma] RE: [MBONED] FW: =
WGLC:draft-ietf-behave-multicast-06.txt<BR></FONT><BR></DIV>=0A=
<DIV>=0A=
<P><FONT size=3D2>&gt; -----Original Message-----<BR>&gt; From: Brian =
Haberman [<A =
href=3D"mailto:brian@innovationslab.net">mailto:brian@innovationslab.net<=
/A>]<BR>&gt; Sent: Thursday, June 07, 2007 6:16 AM<BR>&gt; To: Dan =
Wing<BR>&gt; Cc: 'Marshall Eubanks'; mboned@ietf.org; 'Alvaro =
Fernandez';<BR>&gt; magma@ietf.org; behave@ietf.org; 'Marshall Eubanks'; =
Toerless Eckert<BR>&gt; Subject: Re: [BEHAVE] Re: [magma] RE: [MBONED] =
FW:<BR>&gt; =
WGLC:draft-ietf-behave-multicast-06.txt<BR>....<BR>&gt;<BR>&gt;<BR>&gt;<B=
R>&gt; Dan Wing wrote:<BR>&gt; &gt;&gt; The NAPT knows what is SSM. =
Couldn't it also do a Multicast Group&nbsp;<BR>&gt; &gt;&gt; Translation =
for SSM groups ? Why<BR>&gt; &gt;&gt; is this more difficult than, say, =
port translation ?<BR>&gt; &gt;<BR>&gt; &gt; It may be fairly easy -- =
except for rewriting the<BR>&gt; &gt; advertisements in SAP,<BR>&gt; =
&gt; HTTP, HTTPS (a big problem there), email, SIP, RTSP, ....<BR>&gt; =
&gt;<BR>&gt; &gt; And NAPTs don't do that kind of rewriting =
today;<BR>&gt; &gt; draft-ietf-behave-multicast<BR>&gt; &gt; is only a =
BCP and trying to describe what a NAT should best<BR>&gt; &gt; be doing =
as of<BR>&gt; &gt; its date of publication.<BR>&gt; &gt;<BR>&gt; =
&gt;<BR>&gt; &gt; If we want NAPTs to do this, we would want a =
different<BR>&gt; &gt; document that was<BR>&gt; &gt; standards-track =
and that document would define how we<BR>&gt; &gt; wanted NAPTs =
to<BR>&gt; &gt; perform this function.&nbsp; However, I don't see a =
successful<BR>&gt; &gt; path if such an<BR>&gt; &gt; operation would =
require the NAPT to have an ALG for SAP,<BR>&gt; &gt; HTTP, SIP, and =
RTSP<BR>&gt; &gt; and requires the NAT also rewrite those packets.&nbsp; =
We'd need<BR>&gt; &gt; something else.<BR>&gt; &gt; ALGs don't work well =
for a variety of reasons.<BR>&gt;<BR>&gt; Given that, I would suggest =
that the document state that<BR>&gt; administrators of the network =
behind the NAPT assign sub-ranges<BR>&gt; of the SSM space to their =
multicast sources.&nbsp; This could be done<BR>&gt; manually or =
*possibly* with the tools developed by the MALLOC<BR>&gt; WG.&nbsp; For =
the ma &amp; pa type residential networks, I doubt you will<BR>&gt; see =
multiple machines behind a NAPT sourcing SSM.&nbsp; For more<BR>&gt; =
corporate types, the network admin can control the allocation of<BR>&gt; =
the SSM range.<BR><BR>I moved text discussing what applications should =
do into an appendix,<BR>as the document's BCP scope is really around =
NATs (rather than<BR>applications that need to handle NATs).&nbsp; Based =
on your comment above,<BR>and some other input, the text now has the =
following paragraph.&nbsp; I<BR>believe this captures your =
suggestion.<BR><BR>&nbsp; "If multiple SSM sources on the inside of a =
NAT choose the same<BR>&nbsp;&nbsp; multicast group address, those =
sources are uniquely identifiable<BR>&nbsp;&nbsp; because their IP =
addresses are unique.&nbsp; However, if their multicast<BR>&nbsp;&nbsp; =
traffic is NATted and sent on the NAT's public interface, the =
traffic<BR>&nbsp;&nbsp; from those individual sources is no longer =
uniquely identifiable.<BR>&nbsp;&nbsp; This will cause problems for =
multicast receivers which will see an<BR>&nbsp;&nbsp; intermixing of =
traffic from those sources.&nbsp; Resolution of this =
issue<BR>&nbsp;&nbsp; is left for future study.&nbsp; In the meantime, =
applications that source<BR>&nbsp;&nbsp; SSM multicast traffic are =
encouraged to allow the user to modify the<BR>&nbsp;&nbsp; multicast SSM =
address so that users can avoid this problem if that<BR>&nbsp;&nbsp; =
application is placed behind a NAT."<BR><BR>The text doesn't quite go so =
far to say that a range has to be assigned a<BR>priori.&nbsp; But let me =
know if it is acceptable.<BR><BR>I have not submitted -09, but a preview =
is here if anyone wants to see the<BR>full context:<BR><A =
href=3D"http://svn.resiprocate.org/rep/ietf-drafts/behave/draft-ietf-beha=
ve-multicast-">http://svn.resiprocate.org/rep/ietf-drafts/behave/draft-ie=
tf-behave-multicast-</A><BR>09.txt<BR><BR>-d<BR></FONT></P></DIV></DIV></=
BODY></HTML>
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From behave-bounces@ietf.org Mon Aug 20 02:33:46 2007
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From: =?utf-8?q?R=C3=A9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
Organization: Nokia TP-SP-SWD
To: behave@ietf.org, ext Jonathan Rosenberg <jdrosen@cisco.com>
Subject: Re: [BEHAVE] I-D Action:draft-ietf-behave-rfc3489bis-09.txt
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On Friday 17 August 2007 18:20:01 ext behave-bounces@ietf.org wrote:
> 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.

Page 31: As far as I can tell as a non native speaker, "then the converting=
=20
the result to network byte order" does not look grammatically correct.

Otherwise, I am fine with this new revision document.

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


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From: =?iso-8859-1?q?R=E9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
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Subject: [BEHAVE] STUN/ICE: Username length inconstency
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	Hello,

It seems there is a minor error in the maximum username length between ICE-=
17=20
and RFC3489bis-09.

STUN says USERNAME are at most 127 Unicode code points. There is no way a I=
CE=20
user fragment of 1024 bytes could fit in there. I would assume the ICE ufra=
g=20
should be limited to (127-1)/2 =3D 63 characters, so that the resulting STU=
N=20
username is always within the 127 code points limits, including one code=20
point for the semi-colon.

In any case, an ICE implemention that fails to obey this limit would fail t=
o=20
interoperate with another instance of itself, which is pretty lame :)
Also, it ensures the ICE connectivity check packets size remain within the=
=20
minimum IPv4 MTU.

I hope, that I either misunderstood something, or that this is a minor enou=
gh=20
change that it can still be done this late in the process.

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


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From behave-bounces@ietf.org Mon Aug 20 10:04:14 2007
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From: "BREARD Gael RD-SIRP-ISS" <gael.breard@orange-ftgroup.com>
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Hi,
=20
(My apology if this topic has already been addressed.)
=20
STUN seems to only allow authentication based on a shared secret.=20
(And especially TURN requires it)
=20
There is some scenario we do not have shared secret but a PKI =
architecture.
=20
Why do we not allow the use of TLS mutual authentication for STUN (and =
so authorize it for TURN) ?
=20
Moreover, when using TLS with "server only authentication", is it really =
useful to challenge each request ?
=20
Ga=EBl


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From behave-bounces@ietf.org Mon Aug 20 11:02:11 2007
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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 BREARD Gael RD-SIRP-ISS" <gael.breard@orange-ftgroup.com>
Subject: Re: [BEHAVE] TLS mutual authentication for STUN
Date: Mon, 20 Aug 2007 18:01:43 +0300
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On Monday 20 August 2007 17:03:30 ext BREARD Gael RD-SIRP-ISS wrote:
> STUN seems to only allow authentication based on a shared secret.
> (And especially TURN requires it)

STUN defines three authentication profiles:
=2D long-term client credentials
=2D short-term client credentials
=2D no client credentials

STUN is only a framework... You can specify a usage that does not require=20
STUN-level client authentication, and then run it on top of TLS and use=20
TLS-level authentication. Or, you can use STUN-level authentication, and yo=
u=20
have to specify how the credentials are obtained (e.g. ICE gets them from t=
he=20
SDP).

TURN is currently out of sync in how it obtains credentials, so it's diffic=
ult=20
to comment on whether it does or does not allow something.

> There is some scenario we do not have shared secret but a PKI architectur=
e.
>
> Why do we not allow the use of TLS mutual authentication for STUN (and so
> authorize it for TURN) ?

STUN recommends that the client authenticates the server when TLS is used.
This is pretty much common sense, otherwise you might as well not use TLS.

It does not say the server cannot authenticate the client. Of course, the s=
pec=20
does not require that, since most people use login/password pairs rather th=
an=20
client-side certificates.

> Moreover, when using TLS with "server only authentication", is it really
> useful to challenge each request ?

If there are no client-side certificates, the server needs to authenticate =
the=20
client with *something*.

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


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From behave-bounces@ietf.org Mon Aug 20 11:27:17 2007
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Subject: Re: [BEHAVE] TLS mutual authentication for STUN
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Rémi Denis-Courmont a écrit :
> TURN is currently out of sync in how it obtains credentials, so it's difficult 
> to comment on whether it does or does not allow something.  
Ok, my concern was specifically about this sentence (turn-04 §6.1.2):

Allocate requests MUST be authenticated, and
   furthermore, MUST be authenticated using either a shared secret known
   between the client and server, or a short term password derived from
   it.

which seems to insist on the "shared secret" necessity.

I only think it would be a good idea to allow (in TURN draft) no "STUN 
authentication" for allocate requests sent over TLS with mutual 
authentication.


>> Moreover, when using TLS with "server only authentication", is it really
>> useful to challenge each request ?    
> If there are no client-side certificates, the server needs to authenticate the 
> client with *something*. 
I agree, but it needs to be done (in principle) only *one* time in all 
the TLS session (not at each request).
(Though I understand, it's simpler to do the same than in TCP case.)

Gaël


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Subject: [BEHAVE] Re: STUN/ICE: Username length inconstency
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This is a good catch. I recently introduced the limit in STUN based on 
last call comments, forgetting that we had ICE use cases that could 
result in values longer than 127. I had picked 127 thinking about human 
readable usernames. My bad. I think the limit in STUN should be 
increased to match ICE, not the other way around.

-Jonathan R.

Rémi Denis-Courmont wrote:
> 	Hello,
> 
> It seems there is a minor error in the maximum username length between ICE-17 
> and RFC3489bis-09.
> 
> STUN says USERNAME are at most 127 Unicode code points. There is no way a ICE 
> user fragment of 1024 bytes could fit in there. I would assume the ICE ufrag 
> should be limited to (127-1)/2 = 63 characters, so that the resulting STUN 
> username is always within the 127 code points limits, including one code 
> point for the semi-colon.
> 
> In any case, an ICE implemention that fails to obey this limit would fail to 
> interoperate with another instance of itself, which is pretty lame :)
> Also, it ensures the ICE connectivity check packets size remain within the 
> minimum IPv4 MTU.
> 
> I hope, that I either misunderstood something, or that this is a minor enough 
> change that it can still be done this late in the process.
> 

-- 
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: =?iso-8859-1?q?R=E9mi_Denis-Courmont?= <remi.denis-courmont@nokia.com>
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To: "ext BREARD Gael MAPS-ISS" <gael.breard@orange-ftgroup.com>
Subject: Re: [BEHAVE] TLS mutual authentication for STUN
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On Monday 20 August 2007 18:27:02 ext BREARD Gael MAPS-ISS wrote:
> R=E9mi Denis-Courmont a =E9crit :
> > TURN is currently out of sync in how it obtains credentials, so it's
> > difficult to comment on whether it does or does not allow something.
>
> Ok, my concern was specifically about this sentence (turn-04 =A76.1.2):
>
> Allocate requests MUST be authenticated, and
>    furthermore, MUST be authenticated using either a shared secret known
>    between the client and server, or a short term password derived from
>    it.
>
> which seems to insist on the "shared secret" necessity.

You need a shared secret to secure UDP or TCP STUN requests, and it would b=
e=20
pretty suboptimal to use TLS for bulk relayed traffic, *except* when you ha=
ve=20
to use TLS because HTTP CONNECT is your only way out. In fact, RTP over TLS=
=20
very much sucks and is really the fallback of fallback cases.

If you want to encrypt the bulk content, you'd better do it end-to-end (wit=
h=20
SRTP or DTLS), not end-to-relay.


Hence, if you want to use a PKI for authentication, what you would want is =
the=20
infamous Shared Secret Request. It's been purposedly removed because nobody=
=20
ever implemented it. And with the implementor hat, if it were ever to be=20
reintroduced, I would rather it could be done over DTLS, than need a TCP=20
connection to the relay.

=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: "ext Jonathan Rosenberg" <jdrosen@cisco.com>
Date: Tue, 21 Aug 2007 09:26:44 +0300
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Subject: [BEHAVE] Re: STUN/ICE: Username length inconstency
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On Tuesday 21 August 2007 01:16:39 ext Jonathan Rosenberg wrote:
> This is a good catch. I recently introduced the limit in STUN based on
> last call comments, forgetting that we had ICE use cases that could
> result in values longer than 127. I had picked 127 thinking about human
> readable usernames. My bad. I think the limit in STUN should be
> increased to match ICE, not the other way around.

Hmm, I am not sure 1023 character long username are that great really. For =
a=20
start, after concatenation (2047 characters), you'll get connectivity check=
s=20
that exceed the typical Internet MTU. That brings either problem:
=2D NATs will not handle fragmented packets, or
=2D ICE implementation will refuse to send at all (packet too big).

Second, I wonder if if makes sense to have a higher entropy for the usernam=
e=20
(12,000 bits here!) than we have for the message integrity (160 bits=20
currently) or the shared secret password. But that's something for security=
=20
gurus to judge. Even if we switched to SHA512, we'd still have more than=20
enough username entropy with just 63 characters from each sides (756 bits).

Third, we do not want to drain the entropy pool of ICE-capable device too=20
badly. There are better stuff to dedicate entropy to that ICE connectivity=
=20
checks, such as generating TLS parameters.

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


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From: "=?ISO-8859-1?Q?Ga=EBl_Br=E9ard?=" <gael.breard@orange-ftgroup.com>
To: "=?ISO-8859-1?Q?R=E9mi_Denis-Courmont?=" <remi.denis-courmont@nokia.com>
Subject: Re: [BEHAVE] TLS mutual authentication for STUN
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2007/8/21, R=E9mi Denis-Courmont <remi.denis-courmont@nokia.com>:
>
>
> You need a shared secret to secure UDP or TCP STUN requests, and it would
> be
> pretty suboptimal to use TLS for bulk relayed traffic, *except* when you
> have
> to use TLS because HTTP CONNECT is your only way out. In fact, RTP over
> TLS
> very much sucks and is really the fallback of fallback cases.


I agree.  (note: with some proxy/fw you do not need TLS after HTTP CONNECT)

If you want to encrypt the bulk content, you'd better do it end-to-end (wit=
h
> SRTP or DTLS), not end-to-relay.
>
>
> Hence, if you want to use a PKI for authentication, what you would want i=
s
> the
> infamous Shared Secret Request.


yes...

It's been purposedly removed because nobody
> ever implemented it. And with the implementor hat, if it were ever to be
> reintroduced, I would rather it could be done over DTLS, than need a TCP
> connection to the relay.


except when  HTTP CONNECT is your only way out ;) ...

Thanks for your answers.

Ga=EBl

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<br><br><div><span class=3D"gmail_quote">2007/8/21, R=E9mi Denis-Courmont &=
lt;<a href=3D"mailto:remi.denis-courmont@nokia.com">remi.denis-courmont@nok=
ia.com</a>&gt;:</span><blockquote class=3D"gmail_quote" style=3D"border-lef=
t: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1=
ex;">
<br>You need a shared secret to secure UDP or TCP STUN requests, and it wou=
ld be<br>pretty suboptimal to use TLS for bulk relayed traffic, *except* wh=
en you have<br>to use TLS because HTTP CONNECT is your only way out. In fac=
t, RTP over TLS
<br>very much sucks and is really the fallback of fallback cases.</blockquo=
te><div><br>I agree.&nbsp; (note: with some proxy/fw you do not need TLS af=
ter HTTP CONNECT)<br></div><br><blockquote class=3D"gmail_quote" style=3D"b=
order-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; paddin=
g-left: 1ex;">
If you want to encrypt the bulk content, you&#39;d better do it end-to-end =
(with<br>SRTP or DTLS), not end-to-relay.<br><br><br>Hence, if you want to =
use a PKI for authentication, what you would want is the<br>infamous Shared=
 Secret Request.=20
</blockquote><div><br>yes... <br></div><br><blockquote class=3D"gmail_quote=
" style=3D"border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0=
.8ex; padding-left: 1ex;">It&#39;s been purposedly removed because nobody<b=
r>
ever implemented it. And with the implementor hat, if it were ever to be<br=
>reintroduced, I would rather it could be done over DTLS, than need a TCP<b=
r>connection to the relay.</blockquote><div><br>except when&nbsp; HTTP CONN=
ECT is your only way out ;) ...
<br><br>Thanks for your answers.<br><br>Ga=EBl<br></div></div><br>

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From behave-bounces@ietf.org Fri Aug 24 08:32:50 2007
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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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Subject: [BEHAVE] TURN Set Active Destination problem (again)
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	Hello,

About Set Active Destination over UDP, TURN-04 says:

   If the active destination is already set and the request was
   received over UDP, the Set Active Destination request is rejected
   with a 439 Active Destination Already Set error response.  This
   prevents the race condition described in the previous section.

If however, the successful response to a Set Active Destination request got=
=20
lost, the request is retransmitted. But then, following the current rule, t=
he=20
server returns 439. Well, in theory, it must return a success response agai=
n,=20
as it should see that the transaction ID is the same as a previous request,=
=20
and apply the idempotency principle, but that's a impractical.

May I suggest that the server would return a successful response whenever i=
t=20
receives a Set Active Destination request where the REMOTE-ADDRESS is=20
identical to the currently active destination?

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


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This document is looking very good - few big items

I don't understand when a server would send a status indication with  
the state as LISTEN. Can we remove this value?

I'm a bit lost on auth stuff - probably need separate thread on that.  
3489bis removed a way to get short term secrets but turn seems to  
assume they are still there. More importantly, we need to have some  
sort of security on the connection status indication bet since this  
can not be challenged, we can only use short term credentials.  
However, it's not clear how to get one. I suspect we may need a  
conference call some time on STUN/TURN credentials. It's not clear  
that the right message have enough in them to make replay attacks not  
be a problem for short term credentials usage.

What happens when relay receives more than it can send. My initial  
thought on this is 1) when receiving UDP packets, discard them. 2)  
when receiving TCP, stop reading from the incoming TCP buffer until  
their is space to send. If there are multiple incoming TCP  
connection, order they get serviced in is not specified and clients  
can't make any assumptions about it.

If a client ends up with two remote TCP connection (on the same  
allocation), it would be nice to have a way to close one of them but  
not the other.

The connection status indication probably needs the remote address  
attribute so that one can figure out what is going on when there are  
two TCP connection to same allocation.

Section 6.1.2.1 .. the idea that a move can be signaled by using the  
same credential would work for short term credentials but not long  
term ones as more than one client can share the same long term. I  
find this "move" idea fraught with peril to implement - particularly  
in the TCP case, I wonder if we really need it. I have a hard time  
imaging an TCP connection with active data flowing not getting out of  
sync during a move.

The redirect to a different server when overloaded seems like a bad  
idea. Imagine I have two servers A and B that are both overloaded -  
seems like I could have endless redirects. Having the client know  
about two servers via DNS then try one and if it is busy try the  
other makes more sense to me.

It might make sense to suggest that after a TCP allocation happened,  
the TURN server using BSD sockets would not want to start listening  
on it until their was at least one valid permission set on it.



Trivial stuff after this ....

Be nice to have an example showing TCP simultaneous open.

Section 4.1 - 2nd para - you suggest that there are cases where TLS  
will get thorough a firewall and TCP will not - this is true but so  
is the converse so don't want to imply one way or other is better for  
firewall traversal.

Figure 1 - would make more sense to me with s/STUN/TURN

Section 6.1.2.1 - It is confusing how the client would know what IP  
address the server supports  for relaying when the client wants to  
use the REQUEST-IP attribute. If you added some text explaining that  
this was used when allocating a second port after first port was  
allocated, it would be easier to understand.


In
   The server SHOULD only allocate ports in the range 1024-65535.  This
    is one of several ways to prohibit relayed transport addresses from
    being used to attempt to run standard services.
This does not seem to be a good way (or the way used in the draft) to  
accomplish this so I would just delete the second sentence.





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On Jul 19, 2007, at 6:05 AM, Philip Matthews wrote:

>
> I will just note that Saikat implemented port prediction in his TCP  
> study
> and found it worked pretty well.
>
> http://www.guha.cc/saikat/pub/imc05-tcpnat.pdf
>
> However, I think his setup was more of
> a test setup rather than a production setup.

I ddi some testing on port prediction awhile back and it worked out  
very poorly for me. Port prediction did not really work at all when  
you have thousands of users behind a single service provider NAT. The  
rate new flows on the NAT is just too fast. It also runs into serious  
problems when you starting having a single client behind a NAT but  
several things happen at the same time on the client. You might think  
the applications are not correlated but they turned out not to be -  
right when one was trying to set up nat pinholes at the start of the  
call was when - even another application - decided to do something  
else that was triggered by the call. 
  


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Hi,

A couple of comments, but I don't see that we are going anywhere
productive.

First of all, I can agree with it being regretable that there aren't
energy in the WG to document what to consider when designing solutions.
But, I can't force people (much) to do things. There need to be
sufficient support. It is obvious that people will design applications
and protocols that will do hole punching and not use ICE and STUN.

Regarding the references, as you presented them earlier, it didn't
appear that they where prior art to the IETF work. Although, the common
courtesy is to reference where one have gotten the ideas, I think this
is not a simple case to find who is first with a idea. There also
usually a chain of refinements. Putting that history into a standards
document, is bloating it and mostly irrelevant to the one actually
implementing something.

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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Cullen Jennings wrote:
> 

[snip]

> 
> The redirect to a different server when overloaded seems like a bad
> idea. Imagine I have two servers A and B that are both overloaded -
> seems like I could have endless redirects. Having the client know about
> two servers via DNS then try one and if it is busy try the other makes
> more sense to me.
> 

I agree that using the redirect in this case will not really help.  My
proposal was to send back a domain name instead of an IP address, and to
use it in the client in replacement of the current domain used to find
the SRV records, but by remembering all the IP addresses already tried
and excluding them from the results of the subsequents SRV queries.


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



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Ah, I see what you are thinking but seems like it would be better to  
just start out with the SRV record and never need the redirect.

On Aug 29, 2007, at 8:56 AM, Marc Petit-Huguenin wrote:

> Cullen Jennings wrote:
>>
>
> [snip]
>
>>
>> The redirect to a different server when overloaded seems like a bad
>> idea. Imagine I have two servers A and B that are both overloaded -
>> seems like I could have endless redirects. Having the client know  
>> about
>> two servers via DNS then try one and if it is busy try the other  
>> makes
>> more sense to me.
>>
>
> I agree that using the redirect in this case will not really help.  My
> proposal was to send back a domain name instead of an IP address,  
> and to
> use it in the client in replacement of the current domain used to find
> the SRV records, but by remembering all the IP addresses already tried
> and excluding them from the results of the subsequents SRV queries.
>
>
> -- 
> Marc Petit-Huguenin           [                                 ]
> Home: marc@petit-huguenin.org [RFC1855-compliant space for rent ]
> Work: marc@8x8.com            [                                 ]
> [                                                               ]


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From: Marc Petit-Huguenin <petithug@acm.org>
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Subject: Re: [BEHAVE] Comments on draft-ietf-behave-turn-04
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Cullen Jennings wrote:
> 
> Ah, I see what you are thinking but seems like it would be better to
> just start out with the SRV record and never need the redirect.

Yes, but the redirect permits to do things a little bit more
dynamically.  Here's an example:

Let's say we have 3 data centers, one in London,  one in Santa Clara, CA
and another in Hong Kong.  The initial SRV lookup for the example.com
domain returns this:

SRV 0 1 3478 redirect.london.example.com.
SRV 0 1 3478 redirect.santa-clara.example.com.
SRV 0 1 3478 redirect.hong-kong.example.com.


All RR have the same priority, so one of them is randomly selected.  The
client sends the request to the server, the server uses a geoip lookup
to detect the closer data center to the client, and sends a redirect
with the domain hong-kong.example.com.  The client then do a SRV query
for hong-kong.example.com, with the following result:

SRV 0 1 3478 turn1.hong-kong.example.com.
SRV 0 1 3478 turn2.hong-kong.example.com.
SRV 1 1 3478 turn1.santa-clara.example.com.
SRV 1 1 3478 turn2.santa-clara.example.com.
SRV 1 1 3478 turn1.hong-kong.example.com.
SRV 1 1 3478 turn2.hong-kong.example.com.


The TURN servers in the Hong Kong data center will be tried first, then
the TURN servers in the other data centers if they are unreachable.

Note that this is just an example, there is better ways to achieve the
same result.

> 
> On Aug 29, 2007, at 8:56 AM, Marc Petit-Huguenin wrote:
> 
>> Cullen Jennings wrote:
>>>
>>
>> [snip]
>>
>>>
>>> The redirect to a different server when overloaded seems like a bad
>>> idea. Imagine I have two servers A and B that are both overloaded -
>>> seems like I could have endless redirects. Having the client know about
>>> two servers via DNS then try one and if it is busy try the other makes
>>> more sense to me.
>>>
>>
>> I agree that using the redirect in this case will not really help.  My
>> proposal was to send back a domain name instead of an IP address, and to
>> use it in the client in replacement of the current domain used to find
>> the SRV records, but by remembering all the IP addresses already tried
>> and excluding them from the results of the subsequents SRV queries.
>>

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


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Comments on draft-ietf-behave-turn-04

Technical Review
by Alfred E. Heggestad <aeh@db.org>
30. Aug 2007




General:

- Did we consider the scenario where the NAT between the TURN client and the
   TURN server reboots? In such a case both TURN client and server will still
   send and receive data (such as RTP), and if NAT in between reboots, a new
   NAT mapping will be created. The TURN server will not be able to update the
   5-tuple for the "flow" and the result is no RTP data send/received. If we
   can use some kind of unique "Session ID" for the TURN allocation, it will
   still be possible to handle the cases where the NAT reboots. On Allocate
   Request the client could choose a unique 64-bit session ID, which will be
   used in all future communications with the server (for that session).
   Alternatively, the TURN server could keep track of all session IDs, and
   allocate it for the client (returned in Allocate Response). All sub-
   sequent TURN messages from the client will then contain this 64-bit
   session ID, and the TURN server can lookup the allocation (Session) without
   looking at the Source IP+port. Now, this will probably also introduce some
   new security issues..

- Why is the FINGERPRINT attribute not mentioned at all in TURN ?

- There are still references to "Shared Secret request", in section 6.1.1.1
   and 6.1.1.2. This is removed in rfc3489bis-08 and should also be removed
   in TURN

- The BANDWIDTH parameter is a bit problematic, since the TURN allocation
   is normally done before an SDP Offer is sent. If a user-agent is sending
   an SDP offer with audio codecs G.711 (64 kbps) and G.729 (8 kbps) it will
   use the highest bandwidth (64 kbps) to the TURN server. When the call is
   established with G.729, it should be possible to send an update to the
   TURN server, telling it that the bandwidth need changed to 8 kbps




 > Section 4
 >
 >   When the TURN client to server communication is over a datagram
 >   protocol (UDP), any datagram received from the active peer that has
 >   the STUN magic cookie is wrapped in a Data Indication.  Likewise any
 >   datagram sent by the client that has the STUN magic cookie and is
 >   intended for the active peer is wrapped in a Send Indication.  This
 >   wrapping prevents the STUN relay server from inappropriately
 >   interpreting end-to-end data.

Does this mean that the TURN client/server has to scan the whole datagram
looking for the STUN magic cookie? I think not, and I would assume that
it only need to check the 20 first bytes of the datagram.




 > Section 6.2
 >
 >   Response with an appropriate error response code, such as a 431
 >   (Integrity Failure) or 436 (Unknown User).

error codes 431 and 436 are no longer in rfc3489bis-09




 > 6.5.  Connection Status Indication
 >
 >    When the TURN to peer leg is TCP, the TURN client needs to be aware
 >    of the status of these TCP connections.  The TURN extension defines
 >    application states for a TCP connection as follows: LISTEN,
 >    ESTABLISHED, CLOSED.  Consequently, the TURN server sends a

Probably a new state should be defined, if the TCP connection failed
(host unreachable, port unreachable)




 > 7.9.  CONNECT_STAT
 >
 >  This attribute is used by...

It should be mentioned which TURN Requests/Responses/Indications this
attribute is valid for.




 >  17.2.  Informative References
 >

Should be updated:


[1]  draft-ietf-behave-rfc3489bis-06 -> draft-ietf-behave-rfc3489bis-09

[8]  draft-ietf-mmusic-ice-16        -> draft-ietf-mmusic-ice-16

[9]  draft-ietf-behave-nat-udp-08    -> RFC 4787




------------------------------------------------------------------------------
NITS


General:

There is a mix of "Allocate request" and "Allocate Request" in the
document. This term should be consistantly used, and either stick
to one of them.


 > Abstract
 >
 >  ..Relays around NAT (TURN), is useful for elements behind NATs whose
 >    mapping behavior is address and port dependent.  The extension..
                          ^^^^^^^^^^^^^^^^^^^^^^^^^^

TURN is also useful for elements behind "Address-Dependent Mapping" NATs,
right? Ideally this should be written "..whose mapping behavior is NOT
Endpoint-Independent Mapping.."

Terminology from RFC 4787:
     -  Endpoint-Independent Mapping:
     -  Address-Dependent Mapping:
     -  Address and Port-Dependent Mapping:




 > 6.1.1.1.  Initial Requests
 >
 >   ...
 >   adajacent pair, even and odd respectively, for compatibility with a
     ^^^^^^^^^

typo, should be "adjacent"




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TCP
------
I'm unclear on exactly how the TCP Connect request is supposed to  
work.  I noticed that there is nothing in TURN-04 that says how an  
allocation enters LISTEN, although previous versions stated that an  
allocation would enter LISTEN at the time of allocation.

If the intent is that simultaneous open is the sole means by which  
allocations can connect to each other, I have a difficult time  
believing this is feasible.  In order for that to work as true  
simultaneous open, the SYNs must cross on the wire. Assuming a BSD- 
ish implementation, if they don't cross on the wire, you have the  
possibility that a SYN will be received on a socket that is still in  
the CLOSED pseudo-state.    On Mac OSX, the SYN received in CLOSED  
(socket is bound but not listen()ing ) is silently absorbed and both  
sides will retransmit until something that looks similar to  
simultaneous open happens.  On Linux, when SYN is received in CLOSED  
(as before--bound but not listening), an RST is transmitted, and the  
connection attempt is terminated.   Unless there is some means  
outside of TURN  to guarantee that Connect requests are sent such  
that SYNs literally cross on the wire (so that both receivers are in  
SYN_SENT and not CLOSED), I don't see how you can expect that  
mechanism to work well.  Of course, there are things one might do to  
suppress the RST so SYN retransmissions might occur, but an RST to a  
SYN received in CLOSED doesn't sound wrong to me.

Therefore, it looks better to establish a good way to manage whether  
an allocation is LISTENing, or that it is actively connecting, and  
that it is not doing both simultaneously.  Previously on the BEHAVE  
list, a way to do both was brought up, but, while interesting, sounds  
like an exploit and forces an implementation into the dark nether  
regions of platform-specific behavior.  The fact that  the socket API  
doesn't allow a bind() if a socket is already listen()ing is  
something TURN should respect.  Besides, most clients that get  
retrofitted with TURN probably want to treat TURN as a shim in their  
transports to the greatest extent possible, and won't be interested  
in building a bunch of special behavior around TURN's peculiar LISTEN- 
while-Connect-request capability even if you did manage to solve that  
issue.  Those implementations would simply grab a new port when they  
want to make a new connection--just like how everyone does today.

Also, multiplexing traffic across multiple TCP connections within a  
single allocation has been brought up on the list before.  I agree  
that it would be powerful and desirable to do that...if you could do  
it properly.  It's the "properly" part that scares me.  I think there  
is a class of use cases that this mechanism could support well.  Such  
cases would be where only one of the connections is sending "lots" of  
traffic, and the other(s) are sending "little".   (Ignore, for the  
time being, how the TURN server would distinguish this subset of  
possibilities and allow them while suppressing the others a priori.)  
This presumes that we have a good way to adjudicate BANDWIDTH control.

 From the spec, it sounds like BANDWIDTH is meant to cap the traffic  
on the allocation.  Is this for the sum off all permissions, and in  
both directions?  Exactly what the TURN server does to constrain this  
traffic hasn't been addressed.  Discarding chunks of data in TCP  
could cause a painful loss of framing for higher level protocols,  
such as SIP.   The only other options that are obvious are to  
throttle reading from the client and all of the allocation's peer  
connections and let TCP manage the congestion, or tear down with  
RST.   Exactly how you divide the bandwidth between reading each of  
the connections involved needs a lot of discussion.  It would be  
terrible if the media stream starved the signaling stream, and TURN  
can't tell the difference.

Therefore, I would be greatly relieved if multiple TCP connections on  
a single allocation were not supported.


Set Active Destination
----------------------------------
For UDP, it would be reasonable to suggest that the client start  
using Send indications shortly before (perhaps as long as one  
second?) sending a Set Active Destination request to clear the  
current active destination.  This would help the case that some of  
the packets sent prior to the SAD are processed by the server after  
the SAD is processed, due to packet ordering.


Thanks,

Brian Hibbard
Estacado Systems



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