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From: Colin Perkins <csp@csperkins.org>
Date: Fri, 1 Jun 2007 18:15:23 +0100
To: IETF AVT WG <avt@ietf.org>,
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Subject: [BEHAVE] Moving draft-marjou-behave-app-rtp-keepalive to AVT
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There has been some private discussion on adopting the draft on  
Application Mechanisms for maintaining alive NAT mappings associated  
with RTP flows (draft-marjou-behave-app-rtp-keepalive-01.txt) as an  
AVT working group draft. This has previously been discussed in AVT,  
BEHAVE and MMUSIC. Opinions on whether this draft should become a  
working group draft, and on the appropriate home for it, are  
solicited by 8 June 2007.

Colin (as AVT co-chair)


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From: "Dan Wing" <dwing@cisco.com>
To: "'Colin Perkins'" <csp@csperkins.org>, "'IETF AVT WG'" <avt@ietf.org>,
	<behave@ietf.org>, "'mmusic'" <mmusic@ietf.org>
Date: Fri, 1 Jun 2007 11:09:16 -0700
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Subject: [BEHAVE] RE: Moving draft-marjou-behave-app-rtp-keepalive to AVT
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> There has been some private discussion on adopting the draft on  
> Application Mechanisms for maintaining alive NAT mappings associated  
> with RTP flows (draft-marjou-behave-app-rtp-keepalive-01.txt) as an  
> AVT working group draft. This has previously been discussed in AVT,  
> BEHAVE and MMUSIC. Opinions on whether this draft should become a  
> working group draft, and on the appropriate home for it, are  
> solicited by 8 June 2007.
> 
> Colin (as AVT co-chair)

We need a document that consolidates NAT keepalive mechanisms for
UDP-based protocols, most notably RTP (and, to a lesser extent,
SIP).  Currently this is spread in various documents and there
is no consolidated guidance to implementtors on the strength
or weakness of various approaches.  Section 4 of Xavier's 
document shows the current state of the art -- 7 different
keepalive techniques are in various levels of active use.

Most of the keepalive techniques apply to RTP, and thus AVT seems
the best home.  There are a few exceptions which don't fit into
AVT, such as the keepalive technique used by SIP Outbound 
(draft-ietf-sip-outbound), but I don't see the exceptions as
significant enough to resist making this an AVT WG item.  If we 
had an "RAIWG" (akin to TSVWG), it might fit best there, but we 
don't have one.  It doesn't quite fit into BEHAVE, as NATs only
care to see a UDP packet is occasionally sent to the peer and 
NATs don't care what sort of UDP packet is sent; however, the
receiving peer may very much care if it's a STUN packet, a
0-byte packet, or the 5 other types of packets described in
Xavier's document.

I support adding a milestone to AVT for keepalives, and adopting 
that document to meet that milestone.

-d


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From behave-bounces@ietf.org Fri Jun 01 15:12:16 2007
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From: "Dan Wing" <dwing@cisco.com>
To: <Markus.Isomaki@nokia.com>, <magnus.westerlund@ericsson.com>
Subject: RE: [BEHAVE] NAT Control STUN Usage
Date: Fri, 1 Jun 2007 12:12:10 -0700
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> I think STUN NAT control would be very valuable.
> 
> Explicit NAT/FW control in general is important especially 
> for wireless devices, where keepalives (especially for UDP) 
> are really bad for battery consumption. This is not just a 
> nice-to-have optimization, but I believe that in practice 
> you could not use Internet-based VoIP/IM services over e.g. 
> HSDPA or WiMAX, if you had to do the keepalives in the 
> current way.  
> 
> What makes STUN interesting compared to e.g. NSIS is its 
> incremental deployment path. STUN is already supported in 
> many SIP and XMPP clients, and the popularity is increasing. 
> An existing STUN client implementation can be relatively 
> easily extended to support NAT control as well (small
> thing compared to e.g. full ICE support). STUN NAT control 
> would then very well fit into the process how STUN is 
> otherwise used, so the cost would be quite small even if 
> there were no STUN-capable NATs on client's path. STUN-
> capable NATs could then be gradually deployed to improve 
> the situation. Frankly, I don't see a similar deployment 
> potential for NSIS.
> 
> In comparison to UPnP STUN has certain technical benefits 
> like the ability to deal with nested NATs. Also, UPnP is 
> quite limited to local area use and would not solve the wide-
> area wireless access issue I mentioned in the beginning.
> 
> To have best possible value, I think STUN NAT control should 
> also have a way to deal with firewalls without NATs, i.e. there 
> would have to be a way to discover if there are STUN capable 
> firewalls on the path. 

Thanks for your note of support.

I added a capability to discover firewalls in the -02 document
via a 'tagging' mechanism, which is similar to how NSIS/TIST/NLS
find their on-path NSIS/TIST/NLS-aware devices, yet still provides
the incremental deployment of STUN and STUN Control.  It's
in section 5.3 of the -02 version of the document (which appears
to not have yet been posted), available at:
http://svn.resiprocate.org/rep/ietf-drafts/dwing/draft-wing-behave-nat-contr
ol-stun-usage-02.html#anchor7

> Of course, the more we add to the protocol the less obvious 
> the incremental deployment argument becomes.
> 
> So, in summary: This is a pressing problem. None of the existing
> solutions look promising for getting deployed. STUN NAT control has
> certain aspects that would make the deployment more 
> incremental and thus easier. Thus, I am very supportive for the 
> approval of the BoF.

Thanks again,
-d


> Regards,
> 	Markus 
> 
>  
> 
> >-----Original Message-----
> >From: ext Magnus Westerlund [mailto:magnus.westerlund@ericsson.com] 
> >Sent: 30 May, 2007 16:57
> >To: Magnus Westerlund
> >Cc: behave@ietf.org; Philip Matthews; Dan Wing
> >Subject: Re: [BEHAVE] NAT Control STUN Usage
> >
> >Magnus Westerlund skrev:
> >> Philip Matthews skrev:
> >>> Sorry to be coming into this conversation late.
> >>> I am just catching up on various mailing lists after 
> >finishing my drafts.
> >>>
> >>> BEHAVE is chartered to work on STUN.
> >>> STUN requires STUN servers to work, and defined how they behave.
> >>> Why shouldn't we recommend placing some of the servers 
> into NATs as 
> >>> this draft does?
> >>>
> >>> This draft is trying to address the problem of finding the 
> >allocated 
> >>> mappings for nested NAT cases. I think this problem is an 
> important 
> >>> one to solve, because nested NATs are becoming more and 
> more common.
> >>>
> >>> I think BEHAVE is a good place to work on a solution to 
> >this problem, 
> >>> and I think the proposed solution is reasonable.
> >>>
> >>> I think it would be silly to NOT solve this problem just 
> because of 
> >>> charter issues.
> >>>
> >> 
> >> The problem is that large parts of this problem already has been 
> >> solved in the numerous NAT control protocols. Even in IETF 
> >we have two 
> >> different WG currently chartered with solving NAT control issues, 
> >> MIDCOM and NSIS. Going in this new direction for BEHAVE needs IETF 
> >> wide consensus and also good arguments why previous 
> efforts are not 
> >> solving the problem. So if you are interested in this 
> issue, please 
> >> get together and start discussing having a BOF.
> >> 
> >
> >Hi,
> >
> >Dan has requested a BOF on this topic. But I still haven't 
> >seen much debate why we would need something other than what 
> >is already under development within the IETF. I would 
> >appreciate some discussion and clarification before making any 
> >decision if there should be an BOF or not.
> >
> >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
> >-------------------------------------------------------------
> ---------
> >
> >
> >_______________________________________________
> >Behave mailing list
> >Behave@ietf.org
> >https://www1.ietf.org/mailman/listinfo/behave
> >
> 
> 
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
> 


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From behave-bounces@ietf.org Fri Jun 01 15:18:03 2007
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From: "Dan Wing" <dwing@cisco.com>
To: "'Melinda Shore'" <mshore@cisco.com>,
	"'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
Subject: RE: [BEHAVE] NAT Control STUN Usage
Date: Fri, 1 Jun 2007 12:17:58 -0700
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Melinda Shore [mailto:mshore@cisco.com] wrote:

> On 5/30/07 9:57 AM, "Magnus Westerlund" 
> <magnus.westerlund@ericsson.com> wrote:
> > Dan has requested a BOF on this topic. But I still haven't seen much
> > debate why we would need something other than what is already under
> > development within the IETF. I would appreciate some discussion and
> > clarification before making any decision if there should be 
> > an BOF or not.
> 
> It seems dubious to me that yet another on-path signaling protocol is
> needed, or particularly that yet another on-path NAT signaling 
> protocol is needed, when the problems that limit the deployability 
> of any on-path NAT signaling protocol in the first place remain 
> unsolved.
> 
> I initially brought the on-path firewall/NAT signaling work 
> to the IETF lo, those many years ago because it solves some hard 
> problems around topology and ordering.  Unfortunately, however, 
> it requires that at least one end of a "path" not be NATted, and 
> that either or both peers know that it's not NATted. 

ICE resolves that specific problem by using multiple candidates.

> The solutions that have been proposed (when they've been proposed 
> - there's been a lot of hand-waving) are often worse than the 
> problem, involving throwing new elements into the network
> or sending out an effectively unbounded number of messages to 
> do discovery.

I believe you are, here, referring to ICE's use of candidates,
which I know you find unappealing.  

ICE does bound the number of messages it sends, and it does
prioritize the discovery messages in order to quickly find a
useable candidate address.

> The on-path stuff is extremely appealing but I think there are some
> difficult problems that need to be solved before it can be 
> deployed, and it seems to me that it ought to be deployable 
> before developing a redundant protocol.

-d


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From: "Dan Wing" <dwing@cisco.com>
To: "'Melinda Shore'" <mshore@cisco.com>,
	"'Philip Matthews'" <philip_matthews@magma.ca>
Subject: RE: [BEHAVE] NAT Control STUN Usage
Date: Fri, 1 Jun 2007 13:07:03 -0700
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Melinda Shore [mailto:mshore@cisco.com] wrote:

> On 5/30/07 5:32 PM, "Philip Matthews" 
> <philip_matthews@magma.ca> wrote:
> > I would be interested in hearing what you see as the 
> > deployment problems.
> 
> Sure:
> 
> You need to have something that can turn the message around and
> provide a reply,

That would be a STUN server, in the case of today's STUN, and also
in the case of draft-wing-behave-nat-control-stun-usage.

> or else you need to receive responses back from
> the middleboxes. 


> The first option is unreliable

It is no more unreliable than other UDP-based protocols (e.g., DNS).

> and the second
> option leads to the possibilities of message proliferation and
> message expansion (since you can no longer rely on packet rewriting
> as a way of propagating topological information).

Yep, and that's only necessary if the discovery process includes firewall
discovery.  It wasn't until -02 that firewall discovery was in
draft-wing-behave-nat-control-stun-usage; prior to that the document only
did NAT discovery which doesn't lead to expansion of the response message to
discover on-path NATs.

I'm not sure what you're referring to with 'message proliferation' -- is
that in relation to discovery or control of multiple middleboxes, all of
which need to be discovered and controlled separately causing proliferation
of messages to each of them, or referring to something else?

> You need to have some way to secure messages when you don't
> know in advance who you'll be sending messages to/receiving
> messages from,

That is the substantial difference between
draft-wing-behave-nat-control-stun-usage and previous attempts in this space
(e.g., UPnP, MIDCOM, NSIS).

With draft-wing-behave-nat-control-stun-usage, the security model is easier
because you're using the same host port ('socket') to open the NAT binding
as you're using to discover on-path NATs as you're using to request the NAT
to extend its binding timeout.  Most other approaches (e.g., MIDCOM and
NSIS) do need a more involved security model because they need to create
some association between the host and its NAT and what the host wants the
NAT to do.  To another extreme, other approaches (e.g., UPnP and NAT-PMP
(Bonjour)) allow any internal host to request and open any NAT binding it
wants -- for other hosts or for other ports on the same host.

This makes STUN, and STUN Control, unique -- in order to open a binding for
a port, you need to send a packet from that port, and in order to adjust its
binding duration you need to send a packet from that same port and be able
to respond to the NAT's nonce challange.  This means to adjust someone's NAT
binding using draft-wing-behave-nat-control-stun-usage  the attacker needs
to be on-path between the host and its NAT -- such an attacker is able to
already cause much harm to existing protocols such as DNS and TCP which
cannot adequately defend themselves from that same on-path attacker.

> and you'll be doing it in situations where you're
> trying to limit packet size (avoid UDP fragmentation) 

draft-wing-behave-nat-control-stun-usage only increases packet size with the
new 'tagging' feature, added in -02 to support Markus' request for discovery
of on-path firewalls.  This discovery adds 12 bytes to the STUN Response
packet for every firewall that wants to be separately controlled using STUN
Control.  The typical STUN Response packet is approximately 150 bytes
(including IP and UDP headers).  I think we're fine -- we can have about 30
on-path firewalls before exceeding 512 bytes.  (I chose 512 bytes because
that's what DNS chose and it seems reasonable.)

> and limit computational load (public key operations in a 
> middlebox).  We found that group keys are an excellent solution 
> for use within a single security domain, but are obviously 
> unsuitable for use in situations in which you might be crossing 
> arbitrary administrative boundaries and can't know in advance 
> who they'll be.

STUN Control has no need for such heavy-handed security because
each port on the host can only adjust its own bindings on the 
NAT.

Crossing administrative boundaries isn't possible with typical
NAPTs because a packet has to originate from behind the NAPT
in order to create the binding on the NAPT.  Anyway, handling
authorization across domains isn't related to what 
draft-wing-behave-nat-control-stun-usage attempts to address 
because a host can only control NAT bindings that it, itself
created with its own UDP port.  As the host didn't create NAPT
bindings on public interfaces of another administrative domain's
NAT, it cannot control them.  The other administrative domain 
is responsible for its own policy -- just as today they are 
responsible for whatever their NAT traversal scheme is (if any)
and for keeping their NAT bindings alive (if any), their own
HTTP proxies, and their own firewall policies and whatever else
they like to run.  

> One end or the other end has to have a public address/presence.
> If neither end has a public address or presence, that presence
> has to be outsourced to some other device.  That often means
> dropping yet another box (a relay, for example) into both ends
> of the network.  The placement of the relay has to be topologically
> appropriate to the endpoints or else the relay ends up being
> a sort of SBC.
>
> If you do go with a relay, you have to have some way to associate
> it with the endpoint and you have to have some way to publicize the
> association.  The way that's typically handled right now is through
> telephony signaling.  In fact, this whole thing is looking an
> awful lot like a VoIP network, and it's not actually very useful
> for other applications.  (Presumably that's the reason for an
> interest in using STUN rather than an actual signaling protocol,
> as well).
>
> It's doable, but it's not doable in a general way and it's really
> going to work only in very constrained environments.  It's not an
> 80/20 solution, but rather a 20/80 one.

Your points in the above two paragraphs are related to ICE and
if ICE determines (through its connectivity checks) that it needs a 
relay (a TURN server) or if ICE discovers (through its connectivity
checks) if can establish connectivity without a relay using IP 
addresses learned via STUN.  This is interesting but not germane 
to draft-wing-behave-nat-control-stun-usage.

Further, the existing deployments of ICE, on the Internet, have
demonstrated it works 100% of the time.  

I do hope the vendors that have shared that information with me 
privately are able to comment to that effect publicly in this 
thread.

> And, by the way, STUN would probably need substantial changes
> to the protocol machinery in order to handle state maintenance.
> Packetization is really not that interesting (a little interesting,
> but not a lot) - what matters is how the protocol behaves.  "NAT
> control" installs state in middleboxes - you'll need to be able
> to handle errors like transmission failures, device reboots,
> routing changes, and so on.  You'll also need to be able to 
> delete the state you install. 

Merely sending a UDP or TCP packet across a NAT creates state --
the NAT binding.  All draft-wing-behave-nat-control-stun-usage
does is query the NAT for its binding duration and provide a way
to adjust that duration -- and that query and control is only 
for the same IP address and same UDP port that created that 
initial state on the NAT.

> By the time you're done tweaking it so
> that it will be a useful state maintenance tool you'll have a
> full-blown signaling protocol, only a quirky one that's not
> very general and probably isn't very different from other signaling
> protocols aside from packetization.  (FWIW, a few years ago I
> wrote up a "STUN-based Signaling" specification that uses the NLS
> state machine but STUN packetization, and there should still be
> copies floating around ID archives - it solves a political problem
> but not a technical one and was rightly ignored).

draft-wing-behave-nat-control-stun-usage avoids this complication
by using the same IP address and UDP port on the host that created
the NAT binding.

> I could accept (and do accept) the argument that NSIS is not a
> good choice for situations in which the substantial set up costs
> can't be amortized well, but that's not the argument that's being
> presented here.
> 
> Anyway, the broader point is that we've already done on-path
> signaling six ways from Sunday and the hard problems association
> with NAT signaling remain unsolved.  I have a hard time seeing why
> it's a good idea to create yet another protocol while continuing to
> avoid dealing with the problems associated with the architecture.

-d


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From: "Dan Wing" <dwing@cisco.com>
To: "'Melinda Shore'" <mshore@cisco.com>,
	"'Philip Matthews'" <philip_matthews@magma.ca>
Subject: RE: [BEHAVE] NAT Control STUN Usage
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Melinda Shore [mailto:mshore@cisco.com] wrote:

> On 5/31/07 2:12 PM, "Philip Matthews" 
> <philip_matthews@magma.ca> wrote:
> > If I understand you correctly, STUN currently takes the 
> > first approach
> > by requiring a STUN server somewhere in the network. And 
> > you consider
> > this unreliable because there is nothing that forces 
> > someone to provide
> > a STUN server?  Do I understand you correctly?
> 
> It forces the STUN server to be 1) addressable (not so much
> of an issue), and 2) in a topologically-appropriate location.
> Problem 2) has led to the development of ICE, about which the
> less is said the better.
> 
> > Some, but not all, of them are mitigated by the Shared 
> > Secret exchange, but in practice not many people seem to 
> > implement this.
> 
> Well, on the one hand the exchange is of really limited utility
> since it doesn't prove identity, but on the other hand it's the
> only protection (other than obscurity) against an attacker.
> Explicit signaling is always going to be more secure, *if*
> you can authenticate the NAT.

RFC3489's shared secret exchange only provides protection against an
attacker who is on-path during subsequent UDP communications with your STUN
server.  The protection prevents that attacker from spoofing STUN responses;
the attacker could spoof those responses in order to:

  1. cause you to insert a victim's IP address into your subsequent SDP (in
order to use you as a launchpad for a DoS attack against a victim), or to
  2. give you an IP address they control so that you use it in your
subsequent SDP (in order that they can guarantee they'll be one of the
candidate IP addresses and increase the chance they'll be on-path for your
subsequent communication)

(1) was a significant attack when we had only RFC3489 (STUN).  With the
advent of ICE and ICE's connectivity checks, only 6 packets would be sent
towards that victim machine (if the victim isn't listening for STUN
packets); less if the victim is listening for STUN and sends a STUN error
response.

(2) the attacker was already on-path between the STUN client and its STUN
server (in order to spoof the STUN response), so it's possible -- depending
on where the attacker is located -- that they might not be on path for the
natural communication with your peer.  An attacker located next to the STUN
server might be interested in this attack, for example; but an attacker in
the same coffeeshop (on the same 802.11 network, for example) would not need
to perform this active attack because they will still be on-path for your
subsequent communication with your intended peer.

(2) is also thwarted by using SRTP.

> > All he is proposing (as far as
> > I know),
> > is to  discover the binding that a NAT has installed for a 
> > particular
> > local host and port,
> > and then either query the NAT for the timeout value associated with
> > the binding, or suggest a value to the NAT.
> > He is NOT proposing to use STUN to signal the connection, and the
> > only other
> > involvement of STUN will be as part of ICE to select a path for the
> > media.
> 
> Look, STUN was "just" going to be used until midcom was finished.  And
> then it was "just" to be used in cases where a NAT cannot be modified.
> And then BEHAVE was "just" going to define NAT behavior so that STUN
> would work across it.  And now it's "just" putting a STUN server
> inside a NAT so that a NAT is basically a midcom device.  

draft-wing-behave-nat-control-stun-usage is outside of BEHAVE's current
charter, which is why a BoF has been requested to discuss if and how
draft-wing-behave-nat-control-stun-usage should be standardized in IETF.

> Take another look at the draft - it's *definitely* defining on-path 
> signaling to NATs,

Yes, it is.

> and it's doing it without referencing existing on-path signaling
> protocols.

draft-wing-behave-nat-control-stun-usage's NAT discovery and especially its
NAT Control technique works differently than previous NAT control
techniques.  All other techniques have involved a separate control channel
(SNMP for MIDCOM; UPnP invented its own XML-based protocol), whereas
draft-wing-behave-nat-control-stun-usage uses the same UDP port that
originally opened the NAT binding in order to control that very same NAT
binding's timeout.

There is overlap with draft-wing-behave-nat-control-stun-usage-02's new
"tagging" discovery technique with NSIS, TIST, and NLS; the new "tagging"
discovery technique is used to find firewalls (not NATs).  I included a
reference to NSIS, TIST, and NLS when I added that new discovery technique.


-d


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To: "'Pyda Srisuresh'" <srisuresh@yahoo.com>,
	"'Cullen Jennings'" <fluffy@cisco.com>, "'Behave WG'" <behave@ietf.org>
Subject: RE: [BEHAVE] ICMP type 3, code 13, one more time 
Date: Fri, 1 Jun 2007 16:29:45 -0700
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Yes, it appears to be the WG's consensus that ICMP type 3, error code 13 is
the way to go forward.

-d
 

> -----Original Message-----
> From: Pyda Srisuresh [mailto:srisuresh@yahoo.com] 
> Sent: Sunday, May 27, 2007 12:10 AM
> To: Dan Wing; 'Cullen Jennings'; 'Behave WG'
> Subject: RE: [BEHAVE] ICMP type 3, code 13, one more time 
> 
> Dan,
> 
> From the e-mails below, it seems, the use of ICMP type 3, 
> error code 13 for
> REQ-8 has WG consensus. If my read is incorrect, please summarize the
> consensus. I would appreciate that.
> 
> This is the last of the outstanding comments on the draft, 
> AFAIK. Once I have
> the closure on this, I will be happy to get the next rev of 
> the draft. I
> believe, the next rev of the draft should be ready for a WGLC. 
> 
> Thanks.
> 
> regards,
> suresh 
> 
> 
> --- Dan Wing <dwing@cisco.com> wrote:
> 
> >  
> > 
> > > -----Original Message-----
> > > From: Cullen Jennings [mailto:fluffy@cisco.com] 
> > > Sent: Thursday, May 03, 2007 3:07 PM
> > > To: Behave WG
> > > Subject: [BEHAVE] ICMP type 3, code 13, one more time 
> > > 
> > > 
> > > I was having lunch today with Srisuresh and we came to an 
> > > interesting  
> > > inside on how to resolve this. The important thing is not 
> if a NAT  
> > > thinks it's a router, host, or power tool. The important 
> thing is  
> > > what does an applications that receives the ICMP error does. One  
> > > observation is that it might be that you want error codes 
> > > sent to the inside to be different than one sent to the outside. 
> > 
> > When the NAT can't create a new mapping, the ICMP error 
> would only be
> > sent to the inside.  Afterall, it's not possible to even create a 
> > new mapping from the outside.
> > 
> > > The other  
> > > observation is that if 9,10,13 are all soft errors - do 
> we want a  
> > > soft or hard error here?
> > 
> > A NAT that runs out of resources (say, runs out of UDP mappings) 
> > seems a transient event on par with a routing error such as 
> > 'network unreachable' (ICMP type=3, code=0), which is classified 
> > as a soft error.
> > 
> > (as individual.)
> > 
> > -d
> > 
> > 
> > _______________________________________________
> > Behave mailing list
> > Behave@ietf.org
> > https://www1.ietf.org/mailman/listinfo/behave
> > 
> 


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From: Lars Eggert <lars.eggert@nokia.com>
Subject: Re: [BEHAVE] RE: Moving draft-marjou-behave-app-rtp-keepalive to AVT
Date: Mon, 4 Jun 2007 10:38:39 +0300
To: ext Dan Wing <dwing@cisco.com>
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On 2007-6-1, at 21:09, ext Dan Wing wrote:
> We need a document that consolidates NAT keepalive mechanisms for
> UDP-based protocols, most notably RTP (and, to a lesser extent,
> SIP).  Currently this is spread in various documents and there
> is no consolidated guidance to implementtors on the strength
> or weakness of various approaches.

I'm wondering if it'd make sense to add this guidance as a section in  
http://tools.ietf.org/html/draft-ietf-tsvwg-udp-guidelines?

Lars



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Subject: [BEHAVE] Re: [AVT] RE: Moving draft-marjou-behave-app-rtp-keepalive
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I think that the AVT WG has the right people to review and recommend  
way(s) to do keep alive for RTP flows - so if we are doing this work  
anywhere, AVT  would be my preferred WG to do it in. Now that Behave  
has published, RFC 4787, I think it is easy for the AVT group to  
ensure that whatever the recommendation is,  it will work with NATs  
that are compliant with BEHAVE. This helps resolve the issues of  
making sure that enough NAT people are involved with the discussion.


On Jun 1, 2007, at 11:09 AM, Dan Wing wrote:

>> There has been some private discussion on adopting the draft on
>> Application Mechanisms for maintaining alive NAT mappings associated
>> with RTP flows (draft-marjou-behave-app-rtp-keepalive-01.txt) as an
>> AVT working group draft. This has previously been discussed in AVT,
>> BEHAVE and MMUSIC. Opinions on whether this draft should become a
>> working group draft, and on the appropriate home for it, are
>> solicited by 8 June 2007.
>>
>> Colin (as AVT co-chair)
>
> We need a document that consolidates NAT keepalive mechanisms for
> UDP-based protocols, most notably RTP (and, to a lesser extent,
> SIP).  Currently this is spread in various documents and there
> is no consolidated guidance to implementtors on the strength
> or weakness of various approaches.  Section 4 of Xavier's
> document shows the current state of the art -- 7 different
> keepalive techniques are in various levels of active use.
>
> Most of the keepalive techniques apply to RTP, and thus AVT seems
> the best home.  There are a few exceptions which don't fit into
> AVT, such as the keepalive technique used by SIP Outbound
> (draft-ietf-sip-outbound), but I don't see the exceptions as
> significant enough to resist making this an AVT WG item.  If we
> had an "RAIWG" (akin to TSVWG), it might fit best there, but we
> don't have one.  It doesn't quite fit into BEHAVE, as NATs only
> care to see a UDP packet is occasionally sent to the peer and
> NATs don't care what sort of UDP packet is sent; however, the
> receiving peer may very much care if it's a STUN packet, a
> 0-byte packet, or the 5 other types of packets described in
> Xavier's document.
>
> I support adding a milestone to AVT for keepalives, and adopting
> that document to meet that milestone.
>
> -d
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt


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To: "'Lars Eggert'" <lars.eggert@nokia.com>
Subject: RE: [BEHAVE] RE: Moving draft-marjou-behave-app-rtp-keepalive to AVT
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> On 2007-6-1, at 21:09, ext Dan Wing wrote:
> > We need a document that consolidates NAT keepalive mechanisms for
> > UDP-based protocols, most notably RTP (and, to a lesser extent,
> > SIP).  Currently this is spread in various documents and there
> > is no consolidated guidance to implementtors on the strength
> > or weakness of various approaches.
> 
> I'm wondering if it'd make sense to add this guidance as a 
> section in  
> http://tools.ietf.org/html/draft-ietf-tsvwg-udp-guidelines?

I agree "send periodic UDP traffic" would be good in that
document.  REQ-6 of draft-ford-behave-app-05 may provide the 
beginnings of some useful text (CC'ing the authors of that 
document).

As with all applications on today's Internet, an RTP 
application needs to be robust and able to deal with whatever 
is sent to it, so perhaps you're right -- an application-
specific keepalive document may well be overly specific and
we should, rather, document what any UDP-based application
should do to prevent middleboxes (such as NATs) from 
clobbering a flow.

-d


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The BEHAVE working group concluded its WGLC of
draft-ietf-behave-multicast-06 with no comments received.  I would like
MBONED and MAGMA to please take a look at this draft prior to its submission
to IESG.

Please send replies to behave@ietf.org.  Thanks!

-Dan Wing
 chair, BEHAVE working group

> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com] 
> Sent: Thursday, May 17, 2007 2:47 PM
> To: 'behave@ietf.org'
> Subject: WGLC: draft-ietf-behave-multicast-06.txt
> 
> The working group last call is starting now for "Multicast 
> Requirements for a Network Address Port Translator (NAPT)", 
> http://www.ietf.org/internet-drafts/draft-ietf-behave-multicas
> t-06.txt: 
> 
>    This document specifies requirements for a Network Address
>    Translator (NAT) and Network Address and Port Translator (NAPT)
>    that supports any-source multicast or single-source multicast.  A
>    multicast-capable NAPT device that adheres to the requirements of
>    this document can optimize the operation of multicast
>    applications that are generally unaware of multicast NAPT
>    devices.
> 
> This WGLC will finish on Thursday, May 31 (two weeks from today).
> 
> If this WGLC is successful, I will ask for review by the 
> MAGMA working group and mboned@lists.uoregon.edu prior to 
> submitting to IESG.
> 
> -d
> 


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Date: Tue, 5 Jun 2007 09:22:47 +0200
From: "Xavier Marjou" <xavier.marjou@orange-ftgroup.com>
To: "Lars Eggert" <lars.eggert@nokia.com>
Subject: Re: [BEHAVE] RE: Moving draft-marjou-behave-app-rtp-keepalive to AVT
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	Tom Taylor <tom.taylor@rogers.com>, Joerg Ott <jo@acm.org>,
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If there was a consensus about using a very easy RTP-keepalive
mechanism that could even be re-used for other UDP protocols than RTP
itslef (eg: an empty UDP packet), I think it could be integrated in
the tsvwg-udp-guidelines document. However, there is no consensus on
such a mechanism and some of the possible mechanisms really need AVT
people expertise. So I would also be in favor of moving the draft to
AVT.

Xavier

On 6/4/07, Lars Eggert <lars.eggert@nokia.com> wrote:
> On 2007-6-1, at 21:09, ext Dan Wing wrote:
> > We need a document that consolidates NAT keepalive mechanisms for
> > UDP-based protocols, most notably RTP (and, to a lesser extent,
> > SIP).  Currently this is spread in various documents and there
> > is no consolidated guidance to implementtors on the strength
> > or weakness of various approaches.
>
> I'm wondering if it'd make sense to add this guidance as a section in
> http://tools.ietf.org/html/draft-ietf-tsvwg-udp-guidelines?
>
> Lars
>
>
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
>
>
>


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Thread-Topic: Moving draft-marjou-behave-app-rtp-keepalive to AVT
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Hi,

I think that defining the keepalive for RTP that will not break RTP is
an important work and can be done in AVT.=20
I read the draft and it looks like a good start. I support as having it
an AVT working group draft


Roni Even=20

> -----Original Message-----
> From: Colin Perkins [mailto:csp@csperkins.org]
> Sent: Friday, June 01, 2007 8:15 PM
> To: IETF AVT WG; behave@ietf.org; mmusic
> Cc: Even, Roni; Tom Taylor; Dan Wing ((dwing)); Cullen Jennings; Joerg
> Ott; Xavier Marjou; Jean-Francois Mule; Magnus Westerlund
> Subject: Moving draft-marjou-behave-app-rtp-keepalive to AVT
>=20
> There has been some private discussion on adopting the draft on
> Application Mechanisms for maintaining alive NAT mappings associated
> with RTP flows (draft-marjou-behave-app-rtp-keepalive-01.txt) as an
> AVT working group draft. This has previously been discussed in AVT,
> BEHAVE and MMUSIC. Opinions on whether this draft should become a
> working group draft, and on the appropriate home for it, are
> solicited by 8 June 2007.
>=20
> Colin (as AVT co-chair)


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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [BEHAVE] RE: Moving draft-marjou-behave-app-rtp-keepalive to AVT
Date: Tue, 5 Jun 2007 16:05:29 +0100
To: Lars Eggert <lars.eggert@nokia.com>,
 Dan Wing <dwing@cisco.com>
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On 4 Jun 2007, at 21:54, Dan Wing wrote:
>> On 2007-6-1, at 21:09, ext Dan Wing wrote:
>>> We need a document that consolidates NAT keepalive mechanisms for
>>> UDP-based protocols, most notably RTP (and, to a lesser extent,
>>> SIP).  Currently this is spread in various documents and there
>>> is no consolidated guidance to implementtors on the strength
>>> or weakness of various approaches.
>>
>> I'm wondering if it'd make sense to add this guidance as a
>> section in
>> http://tools.ietf.org/html/draft-ietf-tsvwg-udp-guidelines?
>
> I agree "send periodic UDP traffic" would be good in that
> document.  REQ-6 of draft-ford-behave-app-05 may provide the
> beginnings of some useful text (CC'ing the authors of that
> document).
>
> As with all applications on today's Internet, an RTP
> application needs to be robust and able to deal with whatever
> is sent to it, so perhaps you're right -- an application-
> specific keepalive document may well be overly specific and
> we should, rather, document what any UDP-based application
> should do to prevent middleboxes (such as NATs) from
> clobbering a flow.

The udp-guidelines draft should probably outline what sort of keep  
alive mechanism is needed, but the implementation is likely protocol  
specific, and should be done by the group designing that protocol, I  
think.

-- 
Colin Perkins
http://csperkins.org/




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From: Lars Eggert <lars.eggert@nokia.com>
Subject: Re: [BEHAVE] RE: Moving draft-marjou-behave-app-rtp-keepalive to AVT
Date: Tue, 5 Jun 2007 18:10:19 +0300
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On 2007-6-5, at 18:05, ext Colin Perkins wrote:
> The udp-guidelines draft should probably outline what sort of keep  
> alive mechanism is needed, but the implementation is likely  
> protocol specific, and should be done by the group designing that  
> protocol, I think.

Makes sense to me.

Lars



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> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]=20
> Sent: Monday, June 04, 2007 7:38 PM
> To: mboned@ietf.org; magma@ietf.org
> Cc: 'Behave WG'
> Subject: [MBONED] FW: WGLC: draft-ietf-behave-multicast-06.txt
>=20
> The BEHAVE working group concluded its WGLC of
> draft-ietf-behave-multicast-06 with no comments received.  I=20
> would like
> MBONED and MAGMA to please take a look at this draft prior to=20
> its submission
> to IESG.
>=20
> Please send replies to behave@ietf.org.  Thanks!

http://www.ietf.org/internet-drafts/draft-ietf-behave-multicast-06.txt:=20

What if the NAPT or NAT device just passed IGMP queries and reports
through, changing only the source address of the reports, and let the
multicast router outside do all the hard work? I'm wondering whether the
IGMP aggregation function is mandatory in NAT/NATP.

Bert


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From: Bharat Joshi <bharat_joshi@infosys.com>
To: behave@ietf.org
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Subject: [BEHAVE] Re: [MBONED] FW: WGLC: draft-ietf-behave-multicast-06.txt
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Hi Dan,

    I have couple of questions and suggestions:

> The BEHAVE working group concluded its WGLC of
> draft-ietf-behave-multicast-06 with no comments received.  I would like
> MBONED and MAGMA to please take a look at this draft prior to its submission
> to IESG.
> 
> Please send replies to behave@ietf.org.  Thanks!

* In section 4.2, why is it that a NAPT device implementing this draft
needs to aggregate IGMP message only for IGMPv3? I think if it is
implementing 'RFC4605', it will anyway do the aggregation.

* In case of IGMPv3, why is it that aggregation is a MUST? Also I think
when one of the hosts behind the NAPT generates LEAVE and another
generates JOIN simultaneously, the session will be UP till the
group-timer timesout and as JOIN is also received, it will be updated
and so session will always be UP. Am I mixing the session created in
NAPT with the state in upstream router?

* Typo "lesaving the session" -> "leaving the session"

* In section 4.4, though complete Multicast Range is 224/4 but out of
this range 232/8 is used for Source Specific Multicast or Single-Source
Multicast. So is the following statement syntactically correct

"Any-sosurce multicast (ASM) uses the IP addresses in the 224.0.0.0 -
239.255.255.255 range [IANA-ALLOC]."

What I meant is that it looks to be incorrect to call this range as ASM
range.

* In section 4.4,

     a.  This UDP mapping SHOULD be destroyed when the host leaves
         that host group.

  How do we find out that host has left the group? Please correct me if
I am wrong. We are talking about a host behind the NAPT device
generating Multicast data traffic [UDP traffic]. A multicast host does
not need to join the multicast group [By sending an IGMP join message]
to send multicast traffic to that group. So how does NAPT finds out when
it can destroy this mapping and so this mapping will have to timeout
before it can be removed.

Thanks & Regards,
Bharat

PS: I am not subscribed to 'behave' mailing list so please reply-all or
include my id in your reply.

> -Dan Wing
>  chair, BEHAVE working group
> 
> > -----Original Message-----
> > From: Dan Wing [mailto:dwing@cisco.com] 
> > Sent: Thursday, May 17, 2007 2:47 PM
> > To: 'behave@ietf.org'
> > Subject: WGLC: draft-ietf-behave-multicast-06.txt
> > 
> > The working group last call is starting now for "Multicast 
> > Requirements for a Network Address Port Translator (NAPT)", 
> > http://www.ietf.org/internet-drafts/draft-ietf-behave-multicas
> > t-06.txt: 
> > 
> >    This document specifies requirements for a Network Address
> >    Translator (NAT) and Network Address and Port Translator (NAPT)
> >    that supports any-source multicast or single-source multicast.  A
> >    multicast-capable NAPT device that adheres to the requirements of
> >    this document can optimize the operation of multicast
> >    applications that are generally unaware of multicast NAPT
> >    devices.
> > 
> > This WGLC will finish on Thursday, May 31 (two weeks from today).
> > 
> > If this WGLC is successful, I will ask for review by the 
> > MAGMA working group and mboned@lists.uoregon.edu prior to 
> > submitting to IESG.
> > 
> > -d
> > 
> 
> _______________________________________________
> MBONED mailing list
> MBONED@ietf.org
> https://www1.ietf.org/mailman/listinfo/mboned


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From behave-bounces@ietf.org Tue Jun 05 13:27:21 2007
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Date: Tue, 05 Jun 2007 14:10:54 +0530
From: Prashant Jhingran <prashantj@huawei.com>
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Subject: [BEHAVE] RE: [magma] FW: WGLC: draft-ietf-behave-multicast-06.txt
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Hi,

I have following comments on this draft:

Comment 1:
------------
Since NAPT is not applicable to IPv6 [RFC2460] then no need
to mention MLD along with  IGMP in the draft.

Comment 2:
------------
In Section 4.2, aggregation is made optional while in
section 4.3 its mandatory.
However the reason given for failure of not implementing 4.3
is also applicable  when 4.2 is not implemented. Moreover,
an IGMP device should perform aggregation for optimization
purposes.
Therefore REQ-2 & REQ-3 should be combined and aggregation
should be made  mandatory for all versions of IGMP.

Comment 3:
------------
On Page 4 of the draft:
"based on forwarding state established by the IGMP/MLP proxy
routing."

It should be Should be "IGMP/MLD" instead of typo "IGMP/MLP"


Regards,
Prashant Jhingran

Huawei Technologies
+91-80-41117676 Ext -5129
Mobile:+91-9448927814

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-----Original Message-----
From: Dan Wing [mailto:dwing@cisco.com] 
Sent: Tuesday, June 05, 2007 5:08 AM
To: mboned@ietf.org; magma@ietf.org
Cc: 'Behave WG'
Subject: [magma] FW: WGLC:
draft-ietf-behave-multicast-06.txt

The BEHAVE working group concluded its WGLC of
draft-ietf-behave-multicast-06 with no comments received.  I
would like
MBONED and MAGMA to please take a look at this draft prior
to its submission
to IESG.

Please send replies to behave@ietf.org.  Thanks!

-Dan Wing
 chair, BEHAVE working group

> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com] 
> Sent: Thursday, May 17, 2007 2:47 PM
> To: 'behave@ietf.org'
> Subject: WGLC: draft-ietf-behave-multicast-06.txt
> 
> The working group last call is starting now for "Multicast

> Requirements for a Network Address Port Translator
(NAPT)", 
>
http://www.ietf.org/internet-drafts/draft-ietf-behave-multic
as
> t-06.txt: 
> 
>    This document specifies requirements for a Network
Address
>    Translator (NAT) and Network Address and Port
Translator (NAPT)
>    that supports any-source multicast or single-source
multicast.  A
>    multicast-capable NAPT device that adheres to the
requirements of
>    this document can optimize the operation of multicast
>    applications that are generally unaware of multicast
NAPT
>    devices.
> 
> This WGLC will finish on Thursday, May 31 (two weeks from
today).
> 
> If this WGLC is successful, I will ask for review by the 
> MAGMA working group and mboned@lists.uoregon.edu prior to 
> submitting to IESG.
> 
> -d
> 

_______________________________________________
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magma@ietf.org
https://www1.ietf.org/mailman/listinfo/magma




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> http://www.ietf.org/internet-drafts/draft-ietf-behave-multicas
> t-06.txt: 
> 
> What if the NAPT or NAT device just passed IGMP queries and reports
> through, changing only the source address of the reports, and let the
> multicast router outside do all the hard work? I'm wondering 
> whether the IGMP aggregation function is mandatory in NAT/NATP.

For IGMPv3, you would have the problem described in the document, namely:

   Failure to do this aggregation will cause undesired temporary
   blackholing of multicast traffic.  For example, consider two hosts
   behind the same NAPT.  If one host is joining a session at the same
   time another is lesaving the session, and the NAPT merely relays the
   join and leave upstream, the session will be terminated and the join
   and leave announcements do not comply with section 5 of [RFC3376].

Based on Prashant Jhingran's comment,
<http://www1.ietf.org/mail-archive/web/behave/current/msg02373.html>, I'm
trying to determine if this 
same problem exists for IGMPv1 and IGMPv2.  If so, we will need to require
aggregating all flavors of IGMP to avoid that problem.

-d


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> -----Original Message-----
> From: Bharat Joshi [mailto:bharat_joshi@infosys.com] 
> Sent: Monday, June 04, 2007 9:48 PM
> To: behave@ietf.org
> Subject: [BEHAVE] Re: [MBONED] FW: WGLC: 
> draft-ietf-behave-multicast-06.txt
> 
> 
> Hi Dan,
> 
>     I have couple of questions and suggestions:

Thanks for your comments!

> > The BEHAVE working group concluded its WGLC of
> > draft-ietf-behave-multicast-06 with no comments received.  
> > I would like
> > MBONED and MAGMA to please take a look at this draft prior 
> . to its submission
> > to IESG.
> > 
> > Please send replies to behave@ietf.org.  Thanks!
> 
> * In section 4.2, why is it that a NAPT device implementing this draft
> needs to aggregate IGMP message only for IGMPv3? I think if it is
> implementing 'RFC4605', it will anyway do the aggregation.

It's my understanding that things will work -- somewhat sub-optimally --
with IGMPv1 or IGMPv2 without the NAT doing aggregation of IGMPv1/v2
messages and blindly just forwarding them upstream.

If my understanding is incorrect, please let me know (it may well be;
multicast is not my expertise).

> * In case of IGMPv3, why is it that aggregation is a MUST? 
> Also I think
> when one of the hosts behind the NAPT generates LEAVE and another
> generates JOIN simultaneously, the session will be UP till the
> group-timer timesout and as JOIN is also received, it will be updated
> and so session will always be UP. Am I mixing the session created in
> NAPT with the state in upstream router?

Slightly after that requirement in the document is this explanation:

    >
    >  Failure to do this aggregation will cause undesired
    >  temporary blackholing of multicast traffic.  For example,
    >  consider two hosts behind the same NAPT. If one host is
    >  joining a session at the same time another is leaving the
    >  session, and the NAPT merely relays the join and leave
    >  upstream, the session will be terminated and the join and
    >  leave announcements do not comply with section 5 of <xref
    >  target="RFC3376"></xref>.
    >

> * Typo "lesaving the session" -> "leaving the session"

Thanks, fixed.

> * In section 4.4, though complete Multicast Range is 224/4 but out of
> this range 232/8 is used for Source Specific Multicast or 
> Single-Source
> Multicast. So is the following statement syntactically correct
> 
> "Any-sosurce multicast (ASM) uses the IP addresses in the 224.0.0.0 -
> 239.255.255.255 range [IANA-ALLOC]."
> 
> What I meant is that it looks to be incorrect to call this 
> range as ASM range.

You're right.  I pulled this from the Terminology section of RFC3569 
("An Overview of Source-Specific Multicast (SSM)").  I have adjusted
my document so that it describes the ASM space as 224/8 through
231/8 and 233/8 through 239/8 (with a hole for 224/8).

> * In section 4.4,
> 
>      a.  This UDP mapping SHOULD be destroyed when the host leaves
>          that host group.
> 
>   How do we find out that host has left the group? Please 
> correct me if I am wrong.
> We are talking about a host behind the NAPT device
> generating Multicast data traffic [UDP traffic]. A multicast host does
> not need to join the multicast group [By sending an IGMP join message]
> to send multicast traffic to that group. So how does NAPT 
> finds out when
> it can destroy this mapping and so this mapping will have to timeout
> before it can be removed.

Ah, good point which I hadn't considered.  I have removed that 
requirement.

> Thanks & Regards,
> Bharat
> 
> PS: I am not subscribed to 'behave' mailing list so please 
> reply-all or include my id in your reply.

-d

> > -Dan Wing
> >  chair, BEHAVE working group
> > 
> > > -----Original Message-----
> > > From: Dan Wing [mailto:dwing@cisco.com] 
> > > Sent: Thursday, May 17, 2007 2:47 PM
> > > To: 'behave@ietf.org'
> > > Subject: WGLC: draft-ietf-behave-multicast-06.txt
> > > 
> > > The working group last call is starting now for "Multicast 
> > > Requirements for a Network Address Port Translator (NAPT)", 
> > > http://www.ietf.org/internet-drafts/draft-ietf-behave-multicas
> > > t-06.txt: 
> > > 
> > >    This document specifies requirements for a Network Address
> > >    Translator (NAT) and Network Address and Port Translator (NAPT)
> > >    that supports any-source multicast or single-source 
> multicast.  A
> > >    multicast-capable NAPT device that adheres to the 
> requirements of
> > >    this document can optimize the operation of multicast
> > >    applications that are generally unaware of multicast NAPT
> > >    devices.
> > > 
> > > This WGLC will finish on Thursday, May 31 (two weeks from today).
> > > 
> > > If this WGLC is successful, I will ask for review by the 
> > > MAGMA working group and mboned@lists.uoregon.edu prior to 
> > > submitting to IESG.
> > > 
> > > -d
> > > 
> > 
> > _______________________________________________
> > MBONED mailing list
> > MBONED@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mboned
> 
> 
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> -----Original Message-----
> From: Prashant Jhingran [mailto:prashantj@huawei.com] 
> Sent: Tuesday, June 05, 2007 1:41 AM
> To: behave@ietf.org
> Cc: dwing@cisco.com
> Subject: RE: [magma] FW: WGLC: draft-ietf-behave-multicast-06.txt
> 
>  
> Hi,
> 
> I have following comments on this draft:
> 
> Comment 1:
> ------------
> Since NAPT is not applicable to IPv6 [RFC2460] then no need
> to mention MLD along with  IGMP in the draft.

Removed.  Thanks.

> Comment 2:
> ------------
> In Section 4.2, aggregation is made optional while in
> section 4.3 its mandatory.
> However the reason given for failure of not implementing 4.3
> is also applicable  when 4.2 is not implemented.

I'll have to dig into that deeper.  I was under the impression
the same problem (loss of a multicast stream when different 
listners behind the same proxy left and joined) didn't occur 
with IGMPv1 or v2.

> Moreover,
> an IGMP device should perform aggregation for optimization
> purposes.

Agreed, and I'll add a suggestion to that effect.  (Of course,
if we find that IGMPv1-3 all require aggregation in order to
avoid missing portions of multicast streams during leave/join
by different hosts behind the NAT, I'll drop the suggestion
as it'll become a hard MUST-level requirement.)

> Therefore REQ-2 & REQ-3 should be combined and aggregation
> should be made  mandatory for all versions of IGMP.
>
> Comment 3:
> ------------
> On Page 4 of the draft:
> "based on forwarding state established by the IGMP/MLP proxy
> routing."
> 
> It should be Should be "IGMP/MLD" instead of typo "IGMP/MLP"

I fixed the typo by deleting MLD.  

Thanks for your review.   There is another thread, CC'd to 
magma and mboned, where the aggregation issue was also brought
up; I'll probably consolidate the conversation about that point
in that thread.

-d


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Subject: RE: [BEHAVE] Re: [MBONED] FW: WGLC: 
	draft-ietf-behave-multicast-06.txt
From: Bharat Joshi <bharat_joshi@infosys.com>
To: Dan Wing <dwing@cisco.com>
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Hi Dan,

    Please see my replies in line.

Thanks,
Bharat

> > * In section 4.2, why is it that a NAPT device implementing this draft
> > needs to aggregate IGMP message only for IGMPv3? I think if it is
> > implementing 'RFC4605', it will anyway do the aggregation.
> 
> It's my understanding that things will work -- somewhat sub-optimally --
> with IGMPv1 or IGMPv2 without the NAT doing aggregation of IGMPv1/v2
> messages and blindly just forwarding them upstream.
> 
> If my understanding is incorrect, please let me know (it may well be;
> multicast is not my expertise).
> 

Things should work even if we do not aggregate IGMPv3 message also. Why
we do aggregation is that we do not want to send multiple messages
upstream? Because upstream would forward multicast data traffic if he
knows that there is atleast one host interested in a group.

Aggregating IGMPv3 is a little more complex than V1 and V2.

I just wanted to point out that if an NAPT is implementing RFC 4605 than
it should be aggregating all versions of IGMP. So why do you want to
specifically mention the case for IGMPv3?

> > * In case of IGMPv3, why is it that aggregation is a MUST? 
> > Also I think
> > when one of the hosts behind the NAPT generates LEAVE and another
> > generates JOIN simultaneously, the session will be UP till the
> > group-timer timesout and as JOIN is also received, it will be updated
> > and so session will always be UP. Am I mixing the session created in
> > NAPT with the state in upstream router?
> 
> Slightly after that requirement in the document is this explanation:
> 
>     >
>     >  Failure to do this aggregation will cause undesired
>     >  temporary blackholing of multicast traffic.  For example,
>     >  consider two hosts behind the same NAPT. If one host is
>     >  joining a session at the same time another is leaving the
>     >  session, and the NAPT merely relays the join and leave
>     >  upstream, the session will be terminated and the join and
>     >  leave announcements do not comply with section 5 of <xref
>     >  target="RFC3376"></xref>.
>     >

Thats exactly why I raised this question. How does this creates an
issue? This can happen in a host as well. Lets say application just
started and send a join and immediately after this because of some
failure, it goes down and send a leave. I am not sure what the problem
is.

> > * In section 4.4, though complete Multicast Range is 224/4 but out of
> > this range 232/8 is used for Source Specific Multicast or 
> > Single-Source
> > Multicast. So is the following statement syntactically correct
> > 
> > "Any-sosurce multicast (ASM) uses the IP addresses in the 224.0.0.0 -
> > 239.255.255.255 range [IANA-ALLOC]."
> > 
> > What I meant is that it looks to be incorrect to call this 
> > range as ASM range.
> 
> You're right.  I pulled this from the Terminology section of RFC3569 
> ("An Overview of Source-Specific Multicast (SSM)").  I have adjusted
> my document so that it describes the ASM space as 224/8 through
> 231/8 and 233/8 through 239/8 (with a hole for 224/8).
> 

I think this may not be completely correct. This is because 239/8 is
available to administrator and they can use this for SSM within a given
domain.

> > * In section 4.4,
> > 
> >      a.  This UDP mapping SHOULD be destroyed when the host leaves
> >          that host group.
> > 
> >   How do we find out that host has left the group? Please 
> > correct me if I am wrong.
> > We are talking about a host behind the NAPT device
> > generating Multicast data traffic [UDP traffic]. A multicast host does
> > not need to join the multicast group [By sending an IGMP join message]
> > to send multicast traffic to that group. So how does NAPT 
> > finds out when
> > it can destroy this mapping and so this mapping will have to timeout
> > before it can be removed.
> 
> Ah, good point which I hadn't considered.  I have removed that 
> requirement.
> 

Ok. So now how do you remove that mapping? I guess only once that times
out. Right?


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From: "Manfredi, Albert E" <albert.e.manfredi@boeing.com>
To: "Brian Haberman" <brian@innovationslab.net>, "Dan Wing" <dwing@cisco.com>
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> -----Original Message-----
> From: Brian Haberman [mailto:brian@innovationslab.net]=20

> The main issue being that on the upstream interface of the NAPT, the
> same source address will be used for the Report and Done message.  So
> depending on the order of those messages, traffic may stop temporarily
> (until the router sends another Query).

Except that, I thought, the multicast router has to send group-specific
or group-and-source-specific queries before it stops forwarding
multicast packets, in IGMPv2 (the former only) and v3. This is supposed
to ensure that there are no members left in the group, before the
multicast router stops forwarding the packets.

So I'm not sure why there would be this gap in multicasts if everything
is working as it should.

Not that I have anything against aggregation. Just that it seems
unnecessary. It would be nice if the NAT or NAPT device could be as
transparent as possible, I think.

Bert


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From: "Manfredi, Albert E" <albert.e.manfredi@boeing.com>
To: "Marshall Eubanks" <tme@multicasttech.com>
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> -----Original Message-----
> From: Marshall Eubanks [mailto:tme@multicasttech.com]=20

> On Jun 6, 2007, at 1:01 PM, Brian Haberman wrote:

> > But it could be supported by having an administrator inside the NAPT
> > assign different groups to all the SSM senders.  That way, (S,G)
> > collisions will not occur outside the NAPT.
> >
> > I don't see an easy, automated solution.
>=20
> 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 ?

Port translation is only important within the NAPT device itself.
Whereas in multicast, the problem is how to advertize the address of the
group, and how to make sure this address remain stable. I think manual
configuration is probably a good bet, if not the only way.

Bert


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Dan Wing wrote:
>> http://www.ietf.org/internet-drafts/draft-ietf-behave-multicas
>> t-06.txt: 
>>
>> What if the NAPT or NAT device just passed IGMP queries and reports
>> through, changing only the source address of the reports, and let the
>> multicast router outside do all the hard work? I'm wondering 
>> whether the IGMP aggregation function is mandatory in NAT/NATP.
> 
> For IGMPv3, you would have the problem described in the document, namely:
> 
>    Failure to do this aggregation will cause undesired temporary
>    blackholing of multicast traffic.  For example, consider two hosts
>    behind the same NAPT.  If one host is joining a session at the same
>    time another is lesaving the session, and the NAPT merely relays the
>    join and leave upstream, the session will be terminated and the join
>    and leave announcements do not comply with section 5 of [RFC3376].

The main issue being that on the upstream interface of the NAPT, the
same source address will be used for the Report and Done message.  So
depending on the order of those messages, traffic may stop temporarily
(until the router sends another Query).

> 
> Based on Prashant Jhingran's comment,
> <http://www1.ietf.org/mail-archive/web/behave/current/msg02373.html>, I'm
> trying to determine if this 
> same problem exists for IGMPv1 and IGMPv2.  If so, we will need to require
> aggregating all flavors of IGMP to avoid that problem.

I would favor aggregation for all flavors of IGMP.  The implementations
of RFC 4605 that I have seen do this.

Regards,
Brian


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Marshall Eubanks wrote:

>>>>>
>>>>>
>>>>> In the BEHAVE draft, REQ-4, if there is a host on the inside interface
>>>>> sending UDP packets to a multicast group, the NAPT can change the
>>>> source
>>>>> address and the port but not the multicast group address. In this
>>>> way, I
>>>>> think the outside routers can not differentiate the two channels
>>>> because
>>>>> they have the same source (the public interface) and the same
>>>>> multicast
>>>>> group address.
>>>>>
>>>>
>>>> Is it possible for the NAPT to have multiple external addresses to use
>>>> as source addresses?
>>>>
>>>
>>> Why would it need this ? ASM - (*,G) : SSM (S,G) ?
>>> Or are you worried about 2 different SSM sources using the same G.
>>>
>>
>> Yes, that is the scenario Alvaro described.  Two senders using the same
>> G both behind the same NAPT.
>>
> 
> This has to be supported IMHO.
> 

But it could be supported by having an administrator inside the NAPT
assign different groups to all the SSM senders.  That way, (S,G)
collisions will not occur outside the NAPT.

I don't see an easy, automated solution.

Regards,
Brian


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Brian:

=20

My e-mail was not an answer to Albert, sorry for that,=20

=20

It is an idea regarding the BEHAVE Internet Draft and how the multicast =
NAPT can allow outside multicast routers to uniquely identify different =
multicast SSM channels (S,G) coming from inside interfaces in the case =
two inside interfaces use the same multicast group.

=20

In the BEHAVE draft, REQ-4, if there is a host on the inside interface =
sending UDP packets to a multicast group, the NAPT can change the source =
address and the port but not the multicast group address. In this way, I =
think the outside routers can not differentiate the two channels because =
they have the same source (the public interface) and the same multicast =
group address.

=20

Allowing the NAPT to change the multicast group address in the case two =
sender behind the NAPT use the same group address in a similar way the =
NATP changes a UDP port is a possible solution. There are other =
possibilities.

=20

Regards


Alvaro


________________________________

De: Brian Haberman [mailto:brian@innovationslab.net]
Enviado el: mi=E9 06/06/2007 16:06
Para: Alvaro Fernandez
CC: Manfredi, Albert E; behave@ietf.org; mboned@ietf.org; magma@ietf.org
Asunto: Re: [magma] RE: [MBONED] FW: WGLC: =
draft-ietf-behave-multicast-06.txt



Alvaro,
     I am not sure what problem this solves.  Alfred's comments had to
do with receivers behind the NAPT sending Report and Done messages that
had to be aggregated at the NAPT in order to avoid having the external
router stop the multicast stream.  What you describe *appears* to put
the senders behind the NAPT.  Could you elaborate on what your concern =
is?

Regards,
Brian


Alvaro Fernandez wrote:
> Hi all.
>
>=20
>
> If you have two sources from two different inside interfaces sending
> multicast data to the outside interface and both sources use the same
> multicast address (G1) then outside routers, using for example PIM-SM,
> will consider that there is only one multicast channel and will send =
all
> the traffic. Also hosts receiving all the traffic need to separate it.
>
>=20
>
> One solution is to reserve an IP address range inside SSM range ( for
> example 232.0.0.0 to 232.0.255.255) and let the NAPT change, not only
> the source address of the sender, but also the multicast address of =
the
> sender to be unique in this range. In this way outside routers ( and
> receivers ) will consider traffic as coming from different channels =
(IP
> public ,G2) and ( IP public, G3) with G2 and G3 inside the said range.
>
>=20
>
> Best Regards
>
>=20
>
> Alvaro
>
>
> =
------------------------------------------------------------------------
> *De:* Manfredi, Albert E [mailto:albert.e.manfredi@boeing.com]
> *Enviado el:* mar 05/06/2007 16:02
> *Para:* behave@ietf.org; mboned@ietf.org; magma@ietf.org
> *Asunto:* [magma] RE: [MBONED] FW: WGLC: =
draft-ietf-behave-multicast-06.txt
>
>> -----Original Message-----
>> From: Dan Wing [mailto:dwing@cisco.com]
>> Sent: Monday, June 04, 2007 7:38 PM
>> To: mboned@ietf.org; magma@ietf.org
>> Cc: 'Behave WG'
>> Subject: [MBONED] FW: WGLC: draft-ietf-behave-multicast-06.txt
>>
>> The BEHAVE working group concluded its WGLC of
>> draft-ietf-behave-multicast-06 with no comments received.  I
>> would like
>> MBONED and MAGMA to please take a look at this draft prior to
>> its submission
>> to IESG.
>>
>> Please send replies to behave@ietf.org.  Thanks!
>
> =
http://www.ietf.org/internet-drafts/draft-ietf-behave-multicast-06.txt:
>
> What if the NAPT or NAT device just passed IGMP queries and reports
> through, changing only the source address of the reports, and let the
> multicast router outside do all the hard work? I'm wondering whether =
the
> IGMP aggregation function is mandatory in NAT/NATP.
>
> Bert
>
> _______________________________________________
> magma mailing list
> magma@ietf.org
> https://www1.ietf.org/mailman/listinfo/magma
>
>
> =
------------------------------------------------------------------------
>
> _______________________________________________
> magma mailing list
> magma@ietf.org
> https://www1.ietf.org/mailman/listinfo/magma



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<HTML dir=3Dltr><HEAD><TITLE>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 6.00.6000.16441" name=3DGENERATOR></HEAD>=0A=
<BODY>=0A=
<DIV id=3DidOWAReplyText17236 dir=3Dltr>=0A=
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 size=3D2>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"FONT-SIZE: 10pt; COLOR: black; FONT-FAMILY: Arial; =
mso-ansi-language: EN-GB">Brian:</SPAN><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><?xml:namespace prefix =3D o ns =3D =
"urn:schemas-microsoft-com:office:office" /><o:p></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT size=3D3><FONT face=3D"Times =
New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB">My e-mail&nbsp;was not an answer to Albert, sorry for that, =
</SPAN><SPAN lang=3DEN-GB style=3D"mso-ansi-language: =
EN-GB"><o:p></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT size=3D3><FONT face=3D"Times =
New Roman">&nbsp;<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB">It is an idea regarding the BEHAVE Internet Draft and how the =
multicast NAPT can allow outside multicast routers to uniquely identify =
different multicast SSM channels (S,G) coming from inside interfaces in =
the case two inside interfaces use the same multicast group.</SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB"><o:p>&nbsp;</o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB">In the BEHAVE draft, REQ-4, if there is a host on the inside =
interface sending UDP packets to a multicast group, the NAPT can change =
the source address and the port but not the multicast group address. In =
this way, I think the outside routers can not differentiate the two =
channels because they have the same source (the public interface) and =
the same multicast group address.<o:p></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB"><o:p>&nbsp;</o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB">Allowing the NAPT to change the multicast group address in the =
case two sender behind the NAPT use the same group address in a similar =
way&nbsp;the NATP&nbsp;changes a UDP port is a possible solution. There =
are other possibilities.<o:p></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB"><o:p>&nbsp;</o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB">Regards<o:p></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"FONT-SIZE: 10pt; FONT-FAMILY: Arial; mso-ansi-language: =
EN-GB"><BR>Alvaro<o:p></o:p></SPAN></P></FONT></DIV></DIV>=0A=
<DIV dir=3Dltr><BR>=0A=
<HR tabIndex=3D-1>=0A=
<FONT face=3DTahoma size=3D2><B>De:</B> Brian Haberman =
[mailto:brian@innovationslab.net]<BR><B>Enviado el:</B> mi=E9 06/06/2007 =
16:06<BR><B>Para:</B> Alvaro Fernandez<BR><B>CC:</B> Manfredi, Albert E; =
behave@ietf.org; mboned@ietf.org; magma@ietf.org<BR><B>Asunto:</B> Re: =
[magma] RE: [MBONED] FW: WGLC: =
draft-ietf-behave-multicast-06.txt<BR></FONT><BR></DIV>=0A=
<DIV>=0A=
<P><FONT size=3D2>Alvaro,<BR>&nbsp;&nbsp;&nbsp;&nbsp; I am not sure what =
problem this solves.&nbsp; Alfred's comments had to<BR>do with receivers =
behind the NAPT sending Report and Done messages that<BR>had to be =
aggregated at the NAPT in order to avoid having the external<BR>router =
stop the multicast stream.&nbsp; What you describe *appears* to =
put<BR>the senders behind the NAPT.&nbsp; Could you elaborate on what =
your concern is?<BR><BR>Regards,<BR>Brian<BR><BR><BR>Alvaro Fernandez =
wrote:<BR>&gt; Hi all.<BR>&gt;<BR>&gt;&nbsp;<BR>&gt;<BR>&gt; If you have =
two sources from two different inside interfaces sending<BR>&gt; =
multicast data to the outside interface and both sources use the =
same<BR>&gt; multicast address (G1) then outside routers, using for =
example PIM-SM,<BR>&gt; will consider that there is only one multicast =
channel and will send all<BR>&gt; the traffic. Also hosts receiving all =
the traffic need to separate it.<BR>&gt;<BR>&gt;&nbsp;<BR>&gt;<BR>&gt; =
One solution is to reserve an IP address range inside SSM range ( =
for<BR>&gt; example 232.0.0.0 to 232.0.255.255) and let the NAPT change, =
not only<BR>&gt; the source address of the sender, but also the =
multicast address of the<BR>&gt; sender to be unique in this range. In =
this way outside routers ( and<BR>&gt; receivers ) will consider traffic =
as coming from different channels (IP<BR>&gt; public ,G2) and ( IP =
public, G3) with G2 and G3 inside the said =
range.<BR>&gt;<BR>&gt;&nbsp;<BR>&gt;<BR>&gt; Best =
Regards<BR>&gt;<BR>&gt;&nbsp;<BR>&gt;<BR>&gt; =
Alvaro<BR>&gt;<BR>&gt;<BR>&gt; =
------------------------------------------------------------------------<=
BR>&gt; *De:* Manfredi, Albert E [<A =
href=3D"mailto:albert.e.manfredi@boeing.com">mailto:albert.e.manfredi@boe=
ing.com</A>]<BR>&gt; *Enviado el:* mar 05/06/2007 16:02<BR>&gt; *Para:* =
behave@ietf.org; mboned@ietf.org; magma@ietf.org<BR>&gt; *Asunto:* =
[magma] RE: [MBONED] FW: WGLC: =
draft-ietf-behave-multicast-06.txt<BR>&gt;<BR>&gt;&gt; -----Original =
Message-----<BR>&gt;&gt; From: Dan Wing [<A =
href=3D"mailto:dwing@cisco.com">mailto:dwing@cisco.com</A>]<BR>&gt;&gt; =
Sent: Monday, June 04, 2007 7:38 PM<BR>&gt;&gt; To: mboned@ietf.org; =
magma@ietf.org<BR>&gt;&gt; Cc: 'Behave WG'<BR>&gt;&gt; Subject: [MBONED] =
FW: WGLC: draft-ietf-behave-multicast-06.txt<BR>&gt;&gt;<BR>&gt;&gt; The =
BEHAVE working group concluded its WGLC of<BR>&gt;&gt; =
draft-ietf-behave-multicast-06 with no comments received.&nbsp; =
I<BR>&gt;&gt; would like<BR>&gt;&gt; MBONED and MAGMA to please take a =
look at this draft prior to<BR>&gt;&gt; its submission<BR>&gt;&gt; to =
IESG.<BR>&gt;&gt;<BR>&gt;&gt; Please send replies to =
behave@ietf.org.&nbsp; Thanks!<BR>&gt;<BR>&gt; =
http://www.ietf.org/internet-drafts/draft-ietf-behave-multicast-06.txt:<B=
R>&gt;<BR>&gt; What if the NAPT or NAT device just passed IGMP queries =
and reports<BR>&gt; through, changing only the source address of the =
reports, and let the<BR>&gt; multicast router outside do all the hard =
work? I'm wondering whether the<BR>&gt; IGMP aggregation function is =
mandatory in NAT/NATP.<BR>&gt;<BR>&gt; Bert<BR>&gt;<BR>&gt; =
_______________________________________________<BR>&gt; magma mailing =
list<BR>&gt; magma@ietf.org<BR>&gt; <A =
href=3D"https://www1.ietf.org/mailman/listinfo/magma">https://www1.ietf.o=
rg/mailman/listinfo/magma</A><BR>&gt;<BR>&gt;<BR>&gt; =
------------------------------------------------------------------------<=
BR>&gt;<BR>&gt; _______________________________________________<BR>&gt; =
magma mailing list<BR>&gt; magma@ietf.org<BR>&gt; <A =
href=3D"https://www1.ietf.org/mailman/listinfo/magma">https://www1.ietf.o=
rg/mailman/listinfo/magma</A><BR></FONT></P></DIV></BODY></HTML>
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--===============0661617142==--





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Date: Wed, 6 Jun 2007 14:51:47 -0400
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Subject: [BEHAVE] Re: [magma] RE: [MBONED] FW: WGLC:
	draft-ietf-behave-multicast-06.txt
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On Jun 6, 2007, at 1:01 PM, Brian Haberman wrote:

>
>
> Marshall Eubanks wrote:
>
>>>>>>
>>>>>>
>>>>>> In the BEHAVE draft, REQ-4, if there is a host on the inside  
>>>>>> interface
>>>>>> sending UDP packets to a multicast group, the NAPT can change the
>>>>> source
>>>>>> address and the port but not the multicast group address. In this
>>>>> way, I
>>>>>> think the outside routers can not differentiate the two channels
>>>>> because
>>>>>> they have the same source (the public interface) and the same
>>>>>> multicast
>>>>>> group address.
>>>>>>
>>>>>
>>>>> Is it possible for the NAPT to have multiple external addresses  
>>>>> to use
>>>>> as source addresses?
>>>>>
>>>>
>>>> Why would it need this ? ASM - (*,G) : SSM (S,G) ?
>>>> Or are you worried about 2 different SSM sources using the same G.
>>>>
>>>
>>> Yes, that is the scenario Alvaro described.  Two senders using  
>>> the same
>>> G both behind the same NAPT.
>>>
>>
>> This has to be supported IMHO.
>>
>
> But it could be supported by having an administrator inside the NAPT
> assign different groups to all the SSM senders.  That way, (S,G)
> collisions will not occur outside the NAPT.
>
> I don't see an easy, automated solution.

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 ?

Marshall


>
> Regards,
> Brian



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Marshall Eubanks wrote:
> Hello;
> 
> On 6/6/07, Brian Haberman <brian@innovationslab.net> wrote:
>>
>>
>> Alvaro Fernandez wrote:
>> > Brian:
>> >
>> >
>> >
>> > My e-mail was not an answer to Albert, sorry for that,
>> >
>> >
>> >
>> > It is an idea regarding the BEHAVE Internet Draft and how the multicast
>> > NAPT can allow outside multicast routers to uniquely identify different
>> > multicast SSM channels (S,G) coming from inside interfaces in the case
>> > two inside interfaces use the same multicast group.
>> >
>>
>> How prevalent do people expect the scenario of a SSM sender being
>> located behind a NAPT?
>>
> 
> I would expect that this has the potential of being fairly common.
> 
> - Enterprises use NATs a lot. (I am not going to get into the question
> as to whether they should...)
> 
> - Enterprises use multicast video
> 
> - This video needs to go to remote offices, which are commonly behind
> their own NATs.

I assume you mean without utilizing a VPN connection?

> 
> If that multicast moves to SSM then they will need NAT traversal.
> 
> 
>> >
>> >
>> > In the BEHAVE draft, REQ-4, if there is a host on the inside interface
>> > sending UDP packets to a multicast group, the NAPT can change the
>> source
>> > address and the port but not the multicast group address. In this
>> way, I
>> > think the outside routers can not differentiate the two channels
>> because
>> > they have the same source (the public interface) and the same multicast
>> > group address.
>> >
>>
>> Is it possible for the NAPT to have multiple external addresses to use
>> as source addresses?
>>
> 
> Why would it need this ? ASM - (*,G) : SSM (S,G) ?
> Or are you worried about 2 different SSM sources using the same G.
> 

Yes, that is the scenario Alvaro described.  Two senders using the same
G both behind the same NAPT.

Regards,
Brian


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From behave-bounces@ietf.org Wed Jun 06 18:03:26 2007
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From: "Alvaro Fernandez" <Alvaro@soportemv.com>
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	<mboned@ietf.org>, <magma@ietf.org>
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Subject: [BEHAVE] RE: [magma] RE: [MBONED] FW: WGLC:
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Hi all.

=20

If you have two sources from two different inside interfaces sending =
multicast data to the outside interface and both sources use the same =
multicast address (G1) then outside routers, using for example PIM-SM, =
will consider that there is only one multicast channel and will send all =
the traffic. Also hosts receiving all the traffic need to separate it.

=20

One solution is to reserve an IP address range inside SSM range ( for =
example 232.0.0.0 to 232.0.255.255) and let the NAPT change, not only =
the source address of the sender, but also the multicast address of the =
sender to be unique in this range. In this way outside routers ( and =
receivers ) will consider traffic as coming from different channels (IP =
public ,G2) and ( IP public, G3) with G2 and G3 inside the said range.

=20

Best Regards

=20

Alvaro=20


________________________________

De: Manfredi, Albert E [mailto:albert.e.manfredi@boeing.com]
Enviado el: mar 05/06/2007 16:02
Para: behave@ietf.org; mboned@ietf.org; magma@ietf.org
Asunto: [magma] RE: [MBONED] FW: WGLC: =
draft-ietf-behave-multicast-06.txt



> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
> Sent: Monday, June 04, 2007 7:38 PM
> To: mboned@ietf.org; magma@ietf.org
> Cc: 'Behave WG'
> Subject: [MBONED] FW: WGLC: draft-ietf-behave-multicast-06.txt
>
> The BEHAVE working group concluded its WGLC of
> draft-ietf-behave-multicast-06 with no comments received.  I
> would like
> MBONED and MAGMA to please take a look at this draft prior to
> its submission
> to IESG.
>
> Please send replies to behave@ietf.org.  Thanks!

http://www.ietf.org/internet-drafts/draft-ietf-behave-multicast-06.txt:

What if the NAPT or NAT device just passed IGMP queries and reports
through, changing only the source address of the reports, and let the
multicast router outside do all the hard work? I'm wondering whether the
IGMP aggregation function is mandatory in NAT/NATP.

Bert

_______________________________________________
magma mailing list
magma@ietf.org
https://www1.ietf.org/mailman/listinfo/magma



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<HTML dir=3Dltr><HEAD><TITLE>[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 6.00.6000.16441" name=3DGENERATOR></HEAD>=0A=
<BODY>=0A=
<DIV id=3DidOWAReplyText10136 dir=3Dltr>=0A=
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 size=3D2>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT size=3D3><FONT face=3D"Times =
New Roman"><?xml:namespace prefix =3D o ns =3D =
"urn:schemas-microsoft-com:office:office" /><o:p>Hi =
all.</o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><o:p><FONT face=3D"Times New Roman" =
size=3D3>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT size=3D3><FONT face=3D"Times =
New Roman">If you have two sources from two different inside interfaces =
sending multicast data to the outside interface and both sources use the =
same multicast address (G1) then outside routers, using for example =
PIM-SM, will consider that there is only one multicast channel and will =
send all the traffic. Also hosts receiving all the traffic need to =
separate it.<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><o:p><FONT face=3D"Times New Roman" =
size=3D3>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT size=3D3><FONT face=3D"Times =
New Roman">One solution is to reserve an IP address range inside SSM =
range ( for example 232.0.0.0 to 232.0.255.255) and let the NAPT change, =
not only the source address of the sender, but also the multicast =
address of the sender to be unique in this range. In this way outside =
routers ( and receivers ) will consider traffic as coming from different =
channels (IP public ,G2) and ( IP public, G3) with G2 and G3 inside the =
said range.<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><o:p><FONT face=3D"Times New Roman" =
size=3D3>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT size=3D3><FONT face=3D"Times =
New Roman">Best Regards<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><o:p><FONT face=3D"Times New Roman" =
size=3D3>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT size=3D3><FONT face=3D"Times =
New Roman">Alvaro <o:p></o:p></FONT></FONT></SPAN></P></FONT></DIV></DIV>=0A=
<DIV dir=3Dltr><BR>=0A=
<HR tabIndex=3D-1>=0A=
<FONT face=3DTahoma size=3D2><B>De:</B> Manfredi, Albert E =
[mailto:albert.e.manfredi@boeing.com]<BR><B>Enviado el:</B> mar =
05/06/2007 16:02<BR><B>Para:</B> behave@ietf.org; mboned@ietf.org; =
magma@ietf.org<BR><B>Asunto:</B> [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: Dan Wing =
[<A href=3D"mailto:dwing@cisco.com">mailto:dwing@cisco.com</A>]<BR>&gt; =
Sent: Monday, June 04, 2007 7:38 PM<BR>&gt; To: mboned@ietf.org; =
magma@ietf.org<BR>&gt; Cc: 'Behave WG'<BR>&gt; Subject: [MBONED] FW: =
WGLC: draft-ietf-behave-multicast-06.txt<BR>&gt;<BR>&gt; The BEHAVE =
working group concluded its WGLC of<BR>&gt; =
draft-ietf-behave-multicast-06 with no comments received.&nbsp; =
I<BR>&gt; would like<BR>&gt; MBONED and MAGMA to please take a look at =
this draft prior to<BR>&gt; its submission<BR>&gt; to =
IESG.<BR>&gt;<BR>&gt; Please send replies to behave@ietf.org.&nbsp; =
Thanks!<BR><BR>http://www.ietf.org/internet-drafts/draft-ietf-behave-mult=
icast-06.txt:<BR><BR>What if the NAPT or NAT device just passed IGMP =
queries and reports<BR>through, changing only the source address of the =
reports, and let the<BR>multicast router outside do all the hard work? =
I'm wondering whether the<BR>IGMP aggregation function is mandatory in =
NAT/NATP.<BR><BR>Bert<BR><BR>____________________________________________=
___<BR>magma mailing list<BR>magma@ietf.org<BR><A =
href=3D"https://www1.ietf.org/mailman/listinfo/magma">https://www1.ietf.o=
rg/mailman/listinfo/magma</A><BR></FONT></P></DIV></BODY></HTML>
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_______________________________________________
Behave mailing list
Behave@ietf.org
https://www1.ietf.org/mailman/listinfo/behave

--===============1158650788==--





From behave-bounces@ietf.org Wed Jun 06 18:03:34 2007
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Subject: [BEHAVE] Re: [magma] RE: [MBONED] FW: WGLC:
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Alvaro,
     I am not sure what problem this solves.  Alfred's comments had to
do with receivers behind the NAPT sending Report and Done messages that
had to be aggregated at the NAPT in order to avoid having the external
router stop the multicast stream.  What you describe *appears* to put
the senders behind the NAPT.  Could you elaborate on what your concern is?

Regards,
Brian


Alvaro Fernandez wrote:
> Hi all.
> 
>  
> 
> If you have two sources from two different inside interfaces sending
> multicast data to the outside interface and both sources use the same
> multicast address (G1) then outside routers, using for example PIM-SM,
> will consider that there is only one multicast channel and will send all
> the traffic. Also hosts receiving all the traffic need to separate it.
> 
>  
> 
> One solution is to reserve an IP address range inside SSM range ( for
> example 232.0.0.0 to 232.0.255.255) and let the NAPT change, not only
> the source address of the sender, but also the multicast address of the
> sender to be unique in this range. In this way outside routers ( and
> receivers ) will consider traffic as coming from different channels (IP
> public ,G2) and ( IP public, G3) with G2 and G3 inside the said range.
> 
>  
> 
> Best Regards
> 
>  
> 
> Alvaro
> 
> 
> ------------------------------------------------------------------------
> *De:* Manfredi, Albert E [mailto:albert.e.manfredi@boeing.com]
> *Enviado el:* mar 05/06/2007 16:02
> *Para:* behave@ietf.org; mboned@ietf.org; magma@ietf.org
> *Asunto:* [magma] RE: [MBONED] FW: WGLC: draft-ietf-behave-multicast-06.txt
> 
>> -----Original Message-----
>> From: Dan Wing [mailto:dwing@cisco.com]
>> Sent: Monday, June 04, 2007 7:38 PM
>> To: mboned@ietf.org; magma@ietf.org
>> Cc: 'Behave WG'
>> Subject: [MBONED] FW: WGLC: draft-ietf-behave-multicast-06.txt
>>
>> The BEHAVE working group concluded its WGLC of
>> draft-ietf-behave-multicast-06 with no comments received.  I
>> would like
>> MBONED and MAGMA to please take a look at this draft prior to
>> its submission
>> to IESG.
>>
>> Please send replies to behave@ietf.org.  Thanks!
> 
> http://www.ietf.org/internet-drafts/draft-ietf-behave-multicast-06.txt:
> 
> What if the NAPT or NAT device just passed IGMP queries and reports
> through, changing only the source address of the reports, and let the
> multicast router outside do all the hard work? I'm wondering whether the
> IGMP aggregation function is mandatory in NAT/NATP.
> 
> Bert
> 
> _______________________________________________
> magma mailing list
> magma@ietf.org
> https://www1.ietf.org/mailman/listinfo/magma
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> magma mailing list
> magma@ietf.org
> https://www1.ietf.org/mailman/listinfo/magma


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Subject: [BEHAVE] Re: [magma] RE: [MBONED] FW: WGLC:
	draft-ietf-behave-multicast-06.txt
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Brian Haberman wrote:
> Marshall Eubanks wrote:
>
>   
>>>>>> In the BEHAVE draft, REQ-4, if there is a host on the inside interface
>>>>>> sending UDP packets to a multicast group, the NAPT can change the
>>>>>>             
>>>>> source
>>>>>           
>>>>>> address and the port but not the multicast group address. In this
>>>>>>             
>>>>> way, I
>>>>>           
>>>>>> think the outside routers can not differentiate the two channels
>>>>>>             
>>>>> because
>>>>>           
>>>>>> they have the same source (the public interface) and the same
>>>>>> multicast
>>>>>> group address.
>>>>>>
>>>>>>             
>>>>> Is it possible for the NAPT to have multiple external addresses to use
>>>>> as source addresses?
>>>>>
>>>>>           
>>>> Why would it need this ? ASM - (*,G) : SSM (S,G) ?
>>>> Or are you worried about 2 different SSM sources using the same G.
>>>>
>>>>         
>>> Yes, that is the scenario Alvaro described.  Two senders using the same
>>> G both behind the same NAPT.
>>>
>>>       
>> This has to be supported IMHO.
>>
>>     
>
> But it could be supported by having an administrator inside the NAPT
> assign different groups to all the SSM senders.  That way, (S,G)
> collisions will not occur outside the NAPT.
>
>   
In any case, receivers have to learn the public (S,G), so part of the 
problem is how receivers learn channel addresses.
If channels are advertised by sources using SAP for example, the NAT box 
could translate S and G (SAP adrvertisement) on the fly
 (I dont know if this would be hard) so that they dont collide.
If channels are advertised by other means (say a web site) it seems that 
the translation in the Nat box and the advertisement
have to be closely coordinated. One way would be to use a special 
multicast address coding (for example use a hash or a suffix of the 
private source address as a part of the group address,
so that they never collide. Obviously this works only if all private 
addresses behind the NAT are distinct)

cheers
Jean-Jacques


> I don't see an easy, automated solution.
>
> Regards,
> Brian
>
> _______________________________________________
> MBONED mailing list
> MBONED@ietf.org
> https://www1.ietf.org/mailman/listinfo/mboned
>   



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Subject: [BEHAVE] Re: [magma] RE: [MBONED] FW: WGLC:
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Alvaro Fernandez wrote:
> Brian:
> 
>  
> 
> My e-mail was not an answer to Albert, sorry for that,
> 
>  
> 
> It is an idea regarding the BEHAVE Internet Draft and how the multicast
> NAPT can allow outside multicast routers to uniquely identify different
> multicast SSM channels (S,G) coming from inside interfaces in the case
> two inside interfaces use the same multicast group.
> 

How prevalent do people expect the scenario of a SSM sender being
located behind a NAPT?

>  
> 
> In the BEHAVE draft, REQ-4, if there is a host on the inside interface
> sending UDP packets to a multicast group, the NAPT can change the source
> address and the port but not the multicast group address. In this way, I
> think the outside routers can not differentiate the two channels because
> they have the same source (the public interface) and the same multicast
> group address.
> 

Is it possible for the NAPT to have multiple external addresses to use
as source addresses?

>  
> 
> Allowing the NAPT to change the multicast group address in the case two
> sender behind the NAPT use the same group address in a similar way the
> NATP changes a UDP port is a possible solution. There are other
> possibilities.

>From the global perspective, how does a receiver outside the NAPT know
the (S,G) combination to join?

Regards,
Brian



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> > It is an idea regarding the BEHAVE Internet Draft and how 
> > the multicast
> > NAPT can allow outside multicast routers to uniquely 
> > identify different
> > multicast SSM channels (S,G) coming from inside interfaces 
> > in the case
> > two inside interfaces use the same multicast group.
> 
> How prevalent do people expect the scenario of a SSM sender being
> located behind a NAPT?

Not very prevalent with today's broadcast model for SSM where
the broadcaster is the video service provider.

However, we don't want to encourage proliferation of NATs which
are compliant with a BCP that do not allow SSM senders behind
those NATs.  With today's bandwidth to the home we might 
consider someone running a 'radio station' from behind a NAT.
With tomorrow's bandwidth, we might considering someone doing
video sharing to their family from behind a NAT.  Hard to say;
I would like to do this once if we know how to do it today.

> > In the BEHAVE draft, REQ-4, if there is a host on the 
> > inside interface
> > sending UDP packets to a multicast group, the NAPT can 
> > change the source
> > address and the port but not the multicast group address. 
> > In this way, I
> > think the outside routers can not differentiate the two 
> > channels because
> > they have the same source (the public interface) and the 
> > same multicast
> > group address.
> > 
> 
> Is it possible for the NAPT to have multiple external addresses to use
> as source addresses?

Some NAPTs do that, yes -- larger ones used by service providers
and by enterprises.  Your traditional residential-class NAPT doesn't
do that, though.

> > Allowing the NAPT to change the multicast group address in 
> > the case two
> > sender behind the NAPT use the same group address in a 
> > similar way the
> > NATP changes a UDP port is a possible solution. There are other
> > possibilities.
> 
> From the global perspective, how does a receiver outside the 
> NAPT know the (S,G) combination to join?

-d


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


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Jean-Jacques Pansiot <Jean-Jacques.Pansiot@crc.u-strasbg.fr> wrote:

...
> > But it could be supported by having an administrator inside the NAPT
> > assign different groups to all the SSM senders.  That way, (S,G)
> > collisions will not occur outside the NAPT.
>
> In any case, receivers have to learn the public (S,G), so part of the 
> problem is how receivers learn channel addresses.
> If channels are advertised by sources using SAP for example, 
> the NAT box could translate S and G (SAP adrvertisement) on the fly
> (I dont know if this would be hard) so that they dont collide.
>
> If channels are advertised by other means (say a web site) it 
> seems that the translation in the Nat box and the advertisement
> have to be closely coordinated. 

Such rewriting of the destination IP address, which creates a different
multicast realm, is complicated.  It's out of scope in the current document,
on purpose; it is fraught with problems (imagine if the session announcement
arrived via email, for example, instead of via SAP -- the NAT couldn't
rewrite it.  Similarly if it was communicated via HTTPS (instead of HTTP)).
The document currently says this in the introduction:

   As with normal NAPT operation for unicast flows, multicast packets
   received from the outside interface and forwarded to the inside
   interface do not have their source IP addresses changed.  Such
   multicast packets do not need to have their destination IP address
   changed (unless the NAT device wishes to establish septate multicast
   domains, but this is not the typical operation).

If this needs to be strengthened, please let me know.  Perhaps we need to
strongly discourage such an operation with requirements.

-d


> One way would be to use a special 
> multicast address coding (for example use a hash or a suffix of the 
> private source address as a part of the group address,
> so that they never collide. Obviously this works only if all private 
> addresses behind the NAT are distinct)
> 
> cheers
> Jean-Jacques
> 
> 
> > I don't see an easy, automated solution.
> >
> > Regards,
> > Brian
> >
> > _______________________________________________
> > MBONED mailing list
> > MBONED@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mboned
> >   
> 
> 
> 
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave


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> 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.

-d


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From: "Dan Wing" <dwing@cisco.com>
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Subject: [BEHAVE] BoF request: STUN Control Usage
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As individual contributor:

I would like to summarize my reasons for calling this BoF, and the
value I see STUN Control can bring.  Hopefully this summary will tease
out futher objections or further agreement for this BoF and discussion
of draft-wing-behave-nat-control-stun-usage itself.

 1. ICE works.  This has been well demonstrated by its successful
    deployment on the Internet by millions of endpoints.  It works
    without explicit support by subscriber's NATs or ISP's NATs.

 2. A drawback of ICE (and STUN) is the chattiness of keepalive
    messages (primarily with SIP servers [draft-ietf-sip-outbound]) 
    and the chattiness of binding discovery messages for each 
    call (primarily due to ICE).

STUN Control adds an optimization to STUN's existing areas of
applicability by allowing an end host to negotiate and extend NAT
binding lifetimes.  This improves the performance of ICE and 
sip-outbound and any other protocols making use of STUN.

In our draft, we don't compare STUN Control to other protocols, so
here are some points:

* STUN Control is inherently limited in the scope of control it
provides.  It only allows for extension of binding lifetimes, and only
allows such extensions for the binding associated with the source
IP/port the STUN Control request is coming from.  Because of its
fundamentally limited scope of control, the NAT can operate with a
trivial authorization policy - a control message coming on its
internal interface is always authorized to manage the binding lifetime
for the private address and port the message comes from.  This is, in
fact, exactly the same authorization policies used by NATs today to
authorize the creation and termination of bindings.  STUN Control
merely extends that authorization to make the termination more 
controlled.  The trivial scope of control and authorization policy 
means that authentication, credentials, shared secrets, and so on, 
are not required.  The need for such security measures in other, 
richer control protocols makes them hard to deploy.  STUN Control 
is meant to be trivially deployable to solve a small but important 
problem.

* STUN Control works with nested NATs; UPnP does not.

* STUN Control is incrementally deployable.  Other mechanisms are not
(they require modifying end hosts and NATs, or else they fail or
they fall back to other mechanisms).  STUN Control is an
optimization to ICE and sip-outbound.  If a host or a NAT doesn't
support STUN Control, ICE and sip-outbound continue to work --
albeit with more frequent keepalive messages and binding discovery
messages sent to the STUN server (which is what they do today 
without STUN Control).

Please send comments to behave@ietf.org.

-d

-----

(Below is my original BoF proposal.  Chairs are still TBD.  Since
my original request, it has been suggested that one hour isn't 
sufficient, as we need to discuss deployment barriers to existing 
techniques, and optimizations that would be useful for sip-outbound 
and ICE).


Description:
ICE and its companion protocol STUN are emerging as a popular
mechanism to traverse NATs without requiring deployment of new
software in existing NATs.  ICE has been specified for SIP, extended
to Jabber/XMPP, and will work with RTSP.  ICE is incrementally
deployable, providing operation over any sort of NAT and any sort of
firewall, without requiring upgrades to those NATs and firewalls.

A successful Firewall or NAT control protocols needs to also be
incrementally deployable so that applications can request enhanced
services from firewalls and from NATs to permit certain flows towards
a host and to the host's applications.  Incremental deployment allows
hosts to deploy the protocol without requiring NATs or firewalls in
front of the host to be upgraded to enjoy basic operation, but after
upgrading to enjoy enhanced operation.

Draft-wing-behave-nat-control-stun-usage describes how STUN could be
used to communicate with both NATs and firewalls to enhance the
operation of those NATs and firewalls to allow certain flows or treat
them differently.

This BoF is intended to discuss one proposed technique,
draft-wing-behave-nat-control-stun-usage as an incrementally-deployable
NAT control mechanism and firewall control mechanism, and to determine
if there is sufficient community interest to move this work forward
in the IETF.

Agenda:
  Agenda bash .............................................  5
  Summary of existing NAT traversal techniques ............ 20
   (UPnP, NAT-PMP/Bonjour, MIDCOM techniques, NSIS GIST, ICE)
  STUN Control ............................................ 30
                                                     ---------
                                                     total: 55


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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.

Regards,
Brian


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------_=_NextPart_001_01C7A924.8AB055B7
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Brian / Dan:

=20

"Unicast" TCP/UDO sessions are uniquely identified by the tuple

=20

 ( source IP address, source TCP/UDP port, target IP address, target =
TCP/UDP port )

=20

Routers use IP addresses to send and receive the information. NAPT can =
change the insider interface IP and port and TCP/UDP sessions are ok.

=20

Multicast routing is different because it uses also the multicast group =
address for routing.  I think at least this is a point to consider and =
to debate when talking about a multicast NAPT. This was the reason of my =
e-mail of yesterday with the example of the problem of two senders =
behind NAPT using the same multicast group and how outside routers =
consider this traffic as coming from only one source.

=20

Ideally a multicast NAPT should do the same function of a unicast NAPT: =
establish multicast sessions changing parameters that uniquely identify =
each multicast session.

=20

One problem is how the receiver will contact with the sender behind the =
NAPT. But there is still the same problem if the NAPT don't change the =
multicast address. I think this problem is out of the scope of the =
BEHAVE draft.

=20

I think it is difficult to know today how multicast sessions will be =
used in the future and which parameters will be used to establish =
sessions and how sessions will be started. One solution is to consider =
this out of the scope of the document but I think is a good idea to =
mention it.

=20

Regards


Alvaro


________________________________

De: Brian Haberman [mailto:brian@innovationslab.net]
Enviado el: jue 07/06/2007 15:15
Para: Dan Wing
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





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.

Regards,
Brian



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style=3D"mso-ansi-language: EN-GB"><o:p><FONT face=3DArial =
size=3D2>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT face=3DArial><FONT =
size=3D2>Multicast routing is different because it uses also the =
multicast group address for routing. <SPAN style=3D"mso-spacerun: =
yes">&nbsp;</SPAN>I think at least this is a point to consider and to =
debate when talking about a multicast NAPT. This was the reason of my =
e-mail of yesterday with the example of the problem of two senders =
behind NAPT using the same multicast group and how outside routers =
consider this traffic as coming from only one =
source.<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><o:p><FONT face=3DArial =
size=3D2>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT face=3DArial><FONT =
size=3D2>Ideally a multicast NAPT should do the same function of a =
unicast NAPT: establish multicast sessions changing parameters that =
uniquely identify each multicast =
session.<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><o:p><FONT face=3DArial =
size=3D2>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT face=3DArial><FONT size=3D2>One =
problem is how the receiver will contact with the sender behind the =
NAPT. But there is still the same problem if the NAPT don&#8217;t change =
the multicast address. I think this problem is out of the scope of the =
BEHAVE draft.<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><o:p><FONT face=3DArial =
size=3D2>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT face=3DArial><FONT size=3D2>I =
think it is difficult to know today how multicast sessions will be used =
in the future and which parameters will be used to establish sessions =
and how sessions will be started. </FONT></FONT></SPAN><SPAN =
lang=3DEN-GB style=3D"mso-ansi-language: EN-GB"><FONT face=3DArial><FONT =
size=3D2>One solution is to consider this out of the scope of the =
document but I think is a good idea to mention =
it.<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><o:p><FONT face=3DArial =
size=3D2>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><FONT face=3DArial><FONT =
size=3D2>Regards<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-GB =
style=3D"mso-ansi-language: EN-GB"><BR><FONT face=3DArial><FONT =
size=3D2>Alvaro<o:p></o:p></FONT></FONT></SPAN></P></FONT></DIV></DIV>=0A=
<DIV dir=3Dltr><BR>=0A=
<HR tabIndex=3D-1>=0A=
<FONT face=3DTahoma size=3D2><B>De:</B> Brian Haberman =
[mailto:brian@innovationslab.net]<BR><B>Enviado el:</B> jue 07/06/2007 =
15:15<BR><B>Para:</B> Dan Wing<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><BR><BR>=0A=
<P><FONT size=3D2>Dan Wing wrote:<BR>&gt;&gt; The NAPT knows what is =
SSM. Couldn't it also do a Multicast Group&nbsp;<BR>&gt;&gt; Translation =
for SSM groups ? Why<BR>&gt;&gt; is this more difficult than, say, port =
translation ?<BR>&gt;<BR>&gt; It may be fairly easy -- except for =
rewriting the advertisements in SAP,<BR>&gt; HTTP, HTTPS (a big problem =
there), email, SIP, RTSP, ....<BR>&gt;<BR>&gt; And NAPTs don't do that =
kind of rewriting today; draft-ietf-behave-multicast<BR>&gt; is only a =
BCP and trying to describe what a NAT should best be doing as of<BR>&gt; =
its date of publication.<BR>&gt;<BR>&gt;<BR>&gt; If we want NAPTs to do =
this, we would want a different document that was<BR>&gt; =
standards-track and that document would define how we wanted NAPTs =
to<BR>&gt; perform this function.&nbsp; However, I don't see a =
successful path if such an<BR>&gt; operation would require the NAPT to =
have an ALG for SAP, HTTP, SIP, and RTSP<BR>&gt; and requires the NAT =
also rewrite those packets.&nbsp; We'd need something else.<BR>&gt; ALGs =
don't work well for a variety of reasons.<BR><BR>Given that, I would =
suggest that the document state that administrators<BR>of the network =
behind the NAPT assign sub-ranges of the SSM space to<BR>their multicast =
sources.&nbsp; This could be done manually or *possibly* with<BR>the =
tools developed by the MALLOC WG.&nbsp; For the ma &amp; pa type =
residential<BR>networks, I doubt you will see multiple machines behind a =
NAPT sourcing<BR>SSM.&nbsp; For more corporate types, the network admin =
can control the<BR>allocation of the SSM =
range.<BR><BR>Regards,<BR>Brian<BR></FONT></P></DIV></BODY></HTML>
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From: "David Barrett" <dbarrett@quinthar.com>
To: <behave@ietf.org>
Subject: RE: [BEHAVE] BoF request: STUN Control Usage
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> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
> Subject: [BEHAVE] BoF request: STUN Control Usage
> 
>  1. ICE works.  This has been well demonstrated by its successful
>     deployment on the Internet by millions of endpoints.  It works
>     without explicit support by subscriber's NATs or ISP's NATs.

Just curious, what statistics are available on ICE's real-world usage and
NAT-penetration success rates?  Specifically:

- How did you come up with the "millions of endpoints" number?

- What does "works" mean?  (I'd suggest this be measured in "direct peer
connectivity ratio" and "connection setup time", but any measure will do.)

Granted, it might be a bit OT given that you're really trying to talk about
STUN, but I'm always interested in new real-world statistics.

-david



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From: "Dan Wing" <dwing@cisco.com>
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Subject: RE: [BEHAVE] BoF request: STUN Control Usage
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> > -----Original Message-----
> > From: Dan Wing [mailto:dwing@cisco.com]
> > Subject: [BEHAVE] BoF request: STUN Control Usage
> > 
> >  1. ICE works.  This has been well demonstrated by its successful
> >     deployment on the Internet by millions of endpoints.  It works
> >     without explicit support by subscriber's NATs or ISP's NATs.
> 
> Just curious, what statistics are available on ICE's 
> real-world usage and NAT-penetration success rates?  Specifically:
> 
> - How did you come up with the "millions of endpoints" number?

GoogleTalk's API, Google Talk itself, Microsoft's Windows Live
Messenger (formerly called MSN), and Yahoo Messenger all use 
ICE.

> - What does "works" mean?  (I'd suggest this be measured in 
> "direct peer connectivity ratio" and "connection setup time", 
> but any measure will do.)

I found this, "Improving ICE Service Selection in a P2P System 
using the Gradient Topology", 
http://www.jimdowling.info/pubs/dowling-j_icep2p_saso07.pdf
which includes the following quote:

  [...] Google's estimate that 8% of the GoogleTalk traffic 
  is routed using relay nodes.

which cites
  Google. Google talk, Accessed, Feb 22, 2007.
  http://code.google.com/apis/talk/libjingle

which leads one to:
  http://code.google.com/apis/talk/libjingle/important_concepts.html

> Granted, it might be a bit OT given that you're really trying 
> to talk about STUN, but I'm always interested in new real-world 
> statistics.

It's a worthwhile discussion, even if a bit off topic.


If STUN/ICE were more, or less, common on the Internet would you
find a BoF on optimizing STUN more, or less, important?

-d


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Subject: RE: [BEHAVE] BoF request: STUN Control Usage
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> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
> 
> > > -----Original Message-----
> > > From: Dan Wing [mailto:dwing@cisco.com]
> > > Subject: [BEHAVE] BoF request: STUN Control Usage
> > >
> > >  1. ICE works.  This has been well demonstrated by its successful
> > >     deployment on the Internet by millions of endpoints.  It works
> > >     without explicit support by subscriber's NATs or ISP's NATs.
> >
> > Just curious, what statistics are available on ICE's
> > real-world usage and NAT-penetration success rates?  Specifically:
> >
> > - How did you come up with the "millions of endpoints" number?
> 
> GoogleTalk's API, Google Talk itself, Microsoft's Windows Live
> Messenger (formerly called MSN), and Yahoo Messenger all use
> ICE.

Ok, that's true, ICE code exists on millions of computers.  I wonder what
fraction of those clients actually use the VoIP features, however.  It'd be
neat to know how (or estimate) many ICE sessions are established daily,
especially as a fraction of total P2P VoIP connections.

Incidentally, do you know if it's possible to establish VoIP communications
between these networks?  Has it been demonstrated that these different ICE
implementations are in fact compatible, or are they more "ICE-influenced"?


> > - What does "works" mean?  (I'd suggest this be measured in
> > "direct peer connectivity ratio" and "connection setup time",
> > but any measure will do.)
> 
> I found this, "Improving ICE Service Selection in a P2P System
> using the Gradient Topology",
> http://www.jimdowling.info/pubs/dowling-j_icep2p_saso07.pdf
> which includes the following quote:
> 
>   [...] Google's estimate that 8% of the GoogleTalk traffic
>   is routed using relay nodes.

Very cool, thanks!


> > Granted, it might be a bit OT given that you're really trying
> > to talk about STUN, but I'm always interested in new real-world
> > statistics.
> 
> It's a worthwhile discussion, even if a bit off topic.
> 
> If STUN/ICE were more, or less, common on the Internet would you
> find a BoF on optimizing STUN more, or less, important?

Well, I don't know the answers to the above questions I pose, but I'd wager
that the actual utility of ICE in the real world today is less dramatic than
"millions of endpoints" would suggest.  If that's true (and I have no data
to suggest it is or isn't), I'd suggest that perhaps we should work on
determining why ICE isn't more widely used and focus on that.  

For example, has it been proven that different ICE implementations
interoperate as smoothly in practice as they should in theory?  If not, what
real-world lessons can be drawn and applied to perhaps improve practical
interoperability going forward?  92% peer connectivity is pretty good --
perhaps there are bigger fish to fry than getting to 95% or 99%?

And while I realize this is all about ICE, I think this might be relevant to
STUN because there are a finite number of people in the IETF working on P2P,
and their time is a scarce commodity.  Given that ICE is STUN's primary
customer, STUN's fortunes sink or swim with ICE's (just as ICE's fortunes
sink or swim on SIP's).  

As such, until it's been demonstrated that ICE and SIP are already swimming
(which they might be), I'd be hesitant to spend much energy on expanding
STUN beyond the strict functionality required by ICE and would instead
refocus that effort on making ICE and SIP more relevant in the real world.

Said another way, until some measurable, significant fraction (25%?) of the
world's P2P VoIP calls use a SIP/ICE/STUN stack, I think the focus should be
on smoothing out the practical wrinkles that prevent greater adoption over
improving the theoretical capabilities of an underutilized protocol.

Reducing chattiness of STUN sounds great, but the amount of energy it'll
take to not only standardize the protocol but push standard implementation
into a significant portion of the world's NATs seems high.  If we could
measure all that energy into manhours, I wonder what better way those hours
might be spent?

-david



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From: "Dan Wing" <dwing@cisco.com>
To: "'David Barrett'" <dbarrett@quinthar.com>, <behave@ietf.org>
Subject: RE: [BEHAVE] BoF request: STUN Control Usage
Date: Thu, 7 Jun 2007 20:36:46 -0700
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> > > Just curious, what statistics are available on ICE's
> > > real-world usage and NAT-penetration success rates?  Specifically:
> > >
> > > - How did you come up with the "millions of endpoints" number?
> > 
> > GoogleTalk's API, Google Talk itself, Microsoft's Windows Live
> > Messenger (formerly called MSN), and Yahoo Messenger all use
> > ICE.
> 
> Ok, that's true, ICE code exists on millions of computers.  I 
> wonder what fraction of those clients actually use the VoIP 
> features, however. 

Dunno.

> It'd be neat to know how (or estimate) many ICE sessions are 
> established daily, especially as a fraction of total P2P 
> VoIP connections.
> 
> Incidentally, do you know if it's possible to establish VoIP 
> communications between these networks? 

I don't know.  I know Microsoft and Yahoo have collaborated
with their IM systems, but I don't know if that includes VoIP.

> Has it been demonstrated that these different ICE
> implementations are in fact compatible, or are they more 
> "ICE-influenced"?

ICE isn't finished, and early ICE versions are not compatible
with each other.  Thus, there isn't much hope those different
networks would be compatible until until ICE has gone to RFC
or if those networks had worked out a version of ICE they 
mutually agreed they would interoperate with (thus creating a
defacto standard).

> > > - What does "works" mean?  (I'd suggest this be measured in
> > > "direct peer connectivity ratio" and "connection setup time",
> > > but any measure will do.)
> > 
> > I found this, "Improving ICE Service Selection in a P2P System
> > using the Gradient Topology",
> > http://www.jimdowling.info/pubs/dowling-j_icep2p_saso07.pdf
> > which includes the following quote:
> > 
> >   [...] Google's estimate that 8% of the GoogleTalk traffic
> >   is routed using relay nodes.
> 
> Very cool, thanks!
> 
> 
> > > Granted, it might be a bit OT given that you're really trying
> > > to talk about STUN, but I'm always interested in new real-world
> > > statistics.
> > 
> > It's a worthwhile discussion, even if a bit off topic.
> > 
> > If STUN/ICE were more, or less, common on the Internet would you
> > find a BoF on optimizing STUN more, or less, important?
> 
> Well, I don't know the answers to the above questions I pose, 
> but I'd wager that the actual utility of ICE in the real world 
> today is less dramatic than "millions of endpoints" would suggest.  
> If that's true (and I have no data to suggest it is or isn't), 
> I'd suggest that perhaps we should work on determining why ICE 
> isn't more widely used and focus on that.  
> 
> For example, has it been proven that different ICE implementations
> interoperate as smoothly in practice as they should in 
> theory?  If not, what real-world lessons can be drawn and applied 
> to perhaps improve practical interoperability going forward?  92% 
> peer connectivity is pretty good -- perhaps there are bigger 
> fish to fry than getting to 95% or 99%?

Google's analysis is based on their own (GoogleTalk) network, I 
would expect -- not GoogleTalk-to-something-else.  I expect their
92% number is based on how often two users, each behind 
NATs with the 'Address and Port-Dependent Mapping' property
(RFC4787) attempt to talk with each other (this property was
formerly called 'symmetric NAT').  With ICE, since approximately
ice-10, that is the only case that causes a media relay to be
utilized.

> And while I realize this is all about ICE, I think this might 
> be relevant to STUN because there are a finite number of people 
> in the IETF working on P2P, and their time is a scarce 
> commodity.  Given that ICE is STUN's primary customer, STUN's 
> fortunes sink or swim with ICE's (just as ICE's fortunes
> sink or swim on SIP's).  
> 
> As such, until it's been demonstrated that ICE and SIP are 
> already swimming (which they might be), I'd be hesitant to 
> spend much energy on expanding STUN beyond the strict 
> functionality required by ICE and would instead refocus that 
> effort on making ICE and SIP more relevant in 
> the real world.
> 
> Said another way, until some measurable, significant fraction 
> (25%?) of the world's P2P VoIP calls use a SIP/ICE/STUN stack, 
> I think the focus should be on smoothing out the practical 
> wrinkles that prevent greater adoption over improving the 
> theoretical capabilities of an underutilized protocol.

Not sure I follow -- all of the major Internet-based IM clients
with voice and video capabilities use ICE (except Skype which
uses its own techniques), even though ICE isn't even an RFC.  The
value is so great these products have utilized a pre-standard
specification.  This seems quite compelling to me, especially
considering the difficulties usually incurred in implementing
pre-standard specifications in commercial products.

> Reducing chattiness of STUN sounds great, but the amount of 
> energy it'll take to not only standardize the protocol but 
> push standard implementation into a significant portion of 
> the world's NATs seems high.  If we could measure all that 
> energy into manhours, I wonder what better way those hours
> might be spent?

Yes, that's the question.

-d


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Subject: RE: [BEHAVE] BoF request: STUN Control Usage
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> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
> 
> Google's analysis is based on their own (GoogleTalk) network, I
> would expect -- not GoogleTalk-to-something-else.  I expect their
> 92% number is based on how often two users, each behind
> NATs with the 'Address and Port-Dependent Mapping' property
> (RFC4787) attempt to talk with each other (this property was
> formerly called 'symmetric NAT').  With ICE, since approximately
> ice-10, that is the only case that causes a media relay to be
> utilized.

Ah, ok, so it's a theoretical number.  Does anyone know the real number?
Until it's been measured empirically, at scale, in the real world, we simply
don't know.  In my experience, implementing the algorithms is easy, but
tweaking them to work in the real world is very, very hard.


> > Said another way, until some measurable, significant fraction
> > (25%?) of the world's P2P VoIP calls use a SIP/ICE/STUN stack,
> > I think the focus should be on smoothing out the practical
> > wrinkles that prevent greater adoption over improving the
> > theoretical capabilities of an underutilized protocol.
> 
> Not sure I follow -- all of the major Internet-based IM clients
> with voice and video capabilities use ICE (except Skype which
> uses its own techniques), even though ICE isn't even an RFC.  The
> value is so great these products have utilized a pre-standard
> specification.  This seems quite compelling to me, especially
> considering the difficulties usually incurred in implementing
> pre-standard specifications in commercial products.

It's no challenge to get an implementation of a protocol to interoperate
with itself (well, it *is* challenging, but that's not the point).  The real
challenge is having multiple implementations of the same protocol "just
work" without out-of-band coordination between the implementers.

Saying "these networks support ICE but not each other" is exactly equivalent
to saying "these networks have similar protocols -- all called ICE -- and
its unknown how much work it'll take to make them speak the same protocol".


> > Reducing chattiness of STUN sounds great, but the amount of
> > energy it'll take to not only standardize the protocol but
> > push standard implementation into a significant portion of
> > the world's NATs seems high.  If we could measure all that
> > energy into manhours, I wonder what better way those hours
> > might be spent?
> 
> Yes, that's the question.

I don't mean to bash on ICE, but merely to say that until we prove ICE
works, ICE hasn't been proven to work.  I'd suggest "prove" means
"demonstrate in the real world at significant scale" and "work" means:

a) Achieves some minimum level of peer connectivity (90%?)
b) Is sufficiently unambiguous to interoperate without OOB coordination

If it hasn't already been done (and hopefully it has been), these should be
the focus.  Until then, it's wishful thinking to assume all's in order and
move on to something else.

Said another way, I think we've certainly achieved "documenting algorithms
that can be used to construct a proprietary protocol that achieves an
unknown level of NAT penetration".  But "standardizing a protocol that, when
implemented, ensures a minimum level of effectiveness and a high degree of
interoperation on a global scale" remains to be seen.

-david



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From behave-bounces@ietf.org Fri Jun 08 02:44:24 2007
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Subject: RE: [BEHAVE] BoF request: STUN Control Usage
Date: Fri, 8 Jun 2007 09:44:12 +0300
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From: <Erkki.Koivusalo@nokia.com>
To: <dbarrett@quinthar.com>
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Hi,=20

>Given that ICE is STUN's primary customer, STUN's fortunes=20
>sink or swim with ICE's (just as ICE's fortunes
>sink or swim on SIP's). =20

I think this statement is a bit exaggerated. There
are already SIP VoIP UA implementations with RFC
3489 version of STUN but without ICE. They are
succesfully deployed with environments where the
NATs implement Endpoint-Independent Mapping behaviour.

Thus fates of STUN and ICE are not as tightly connected
to each other as you suggest. ICE provides added value
for complex environments, but there are significant
deployments which already work pretty well just with
SIP combined with STUN.

>As such, until it's been demonstrated that ICE and SIP are=20
>already swimming (which they might be), I'd be hesitant to=20
>spend much energy on expanding STUN beyond the strict=20
>functionality required by ICE and would instead refocus=20
>that effort on making ICE and SIP more relevant in the=20
>real world.

For the deployments I am referring to, it would be useful
to have STUN extended with a mechanism to control the
required keepalive interval for NAT binding:

- Extensions needed for the already deployed set of=20
  protocols would be minimal.

- There would be significant benefits both to the
  traffic in the network and the battery consumption
  of mobile VoIP devices.

So I think this effort is valuable regardless on what
happens to ICE.

Regards,

Erkki


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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
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Hi,

I think that SIP outbound is as "primary" customer of STUN as ICE is.

Regards,

Christer

> -----Original Message-----
> From: Erkki.Koivusalo@nokia.com [mailto:Erkki.Koivusalo@nokia.com]=20
> Sent: 8. kes=E4kuuta 2007 9:44
> To: dbarrett@quinthar.com
> Cc: behave@ietf.org
> Subject: RE: [BEHAVE] BoF request: STUN Control Usage
>=20
> Hi,=20
>=20
> >Given that ICE is STUN's primary customer, STUN's fortunes=20
> sink or swim=20
> >with ICE's (just as ICE's fortunes sink or swim on SIP's).
>=20
> I think this statement is a bit exaggerated. There are=20
> already SIP VoIP UA implementations with RFC
> 3489 version of STUN but without ICE. They are succesfully=20
> deployed with environments where the NATs implement=20
> Endpoint-Independent Mapping behaviour.
>=20
> Thus fates of STUN and ICE are not as tightly connected to=20
> each other as you suggest. ICE provides added value for=20
> complex environments, but there are significant deployments=20
> which already work pretty well just with SIP combined with STUN.
>=20
> >As such, until it's been demonstrated that ICE and SIP are already=20
> >swimming (which they might be), I'd be hesitant to spend=20
> much energy on=20
> >expanding STUN beyond the strict functionality required by ICE and=20
> >would instead refocus that effort on making ICE and SIP more=20
> relevant=20
> >in the real world.
>=20
> For the deployments I am referring to, it would be useful to=20
> have STUN extended with a mechanism to control the required=20
> keepalive interval for NAT binding:
>=20
> - Extensions needed for the already deployed set of
>   protocols would be minimal.
>=20
> - There would be significant benefits both to the
>   traffic in the network and the battery consumption
>   of mobile VoIP devices.
>=20
> So I think this effort is valuable regardless on what happens to ICE.
>=20
> Regards,
>=20
> Erkki
>=20
>=20
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
>=20


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



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From: "Pete Cordell" <pete.cordell@tech-know-ware.com>
To: "David Barrett" <dbarrett@quinthar.com>, <behave@ietf.org>
References: <E1HwUiQ-000746-W1@megatron.ietf.org>
Subject: Re: [BEHAVE] BoF request: STUN Control Usage
Date: Fri, 8 Jun 2007 09:38:45 +0100
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----- Original Message From: "David Barrett" <dbarrett@quinthar.com>

> Reducing chattiness of STUN sounds great, but the amount of energy it'll
> take to not only standardize the protocol but push standard implementation
> into a significant portion of the world's NATs seems high.  If we could
> measure all that energy into man-hours, I wonder what better way those 
> hours
> might be spent?

Alas, IMO, nobody has a better idea of where such man-hours could be spent.

I think STUN control is a small incremental piece of work and it makes sense 
to do it while the expertise is still present in the behave WG and all the 
various issues and interactions are current in peoples' minds.

Also, for better or worse, things like STUN, ICE, outbound are all 
inter-related and it seems sensible to deliver them as a package rather than 
trying to bake one and then move onto the next.  At least then, if, in the 
unlikely event that ICE requires a tweak to allow for STUN control, then it 
can be done.

Finally, it's seemed obvious to me from day 1 that 'the' NAT would be the 
ideal location for the STUN server.  So, IMO, the work should go ahead.

Regards,

Pete.
--
=============================================
Pete Cordell
Codalogic Ltd
for XML Schema to C++ data binding visit
 http://www.codalogic.com/lmx/
=============================================




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From: "Alvaro Fernandez" <Alvaro@soportemv.com>
To: "Leymann, Nicolai" <Nicolai.Leymann@t-systems.com>,
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Leyman:

=20

One possible solution ( out of the scope of the draft)

=20

1. The sender behind the  NAPT sends the channel (Si,Gj) using a random =
Gj inside the multicast SSS range

2. Logically the sender expects that some one will be interested in the =
data, so the sender sends a unicast message (for example http)  to a =
"multicast index portal" describing the content and the channel (Si,Gj)

3. The NAPT changes the internal Si into a IP public in both cases: the =
unicast data and the multicast data using the same IP public.

4. The multicast portal knows the IP public and tries to connect with =
the multicast channel ( IP public, Gj)

If the multicast portal connects with ( IP public, Gj) there is no Gj =
collision and the portal shows the channel (IP public, Gj) in its index. =
Everybody can connect.

5. If after a few seconds the channel don't appears in the index, the =
senders understands that there is a multicast Group collision and =
changes the Gj into another randomly selected Gk.

=20

Another business for the search engines.

=20

Regards


Alvaro


________________________________

De: Leymann, Nicolai [mailto:Nicolai.Leymann@t-systems.com]
Enviado el: vie 08/06/2007 11:44
Para: Alvaro Fernandez; albert.e.manfredi@boeing.com; behave@ietf.org; =
mboned@ietf.org; magma@ietf.org
Asunto: AW: [magma] RE: [MBONED] FW: WGLC: =
draft-ietf-behave-multicast-06.txt



Hi Everybody,

> Hi all.
> If you have two sources from two different inside interfaces
> sending multicast data to the outside interface and both
> sources use the same multicast address (G1) then outside
> routers, using for example PIM-SM, will consider that there
> is only one multicast channel and will send all the traffic.
> Also hosts receiving all the traffic need to separate it.
Right. This is from my point of view not the solution you want.
The reason for SSM is to have different senders (source) using
the same group and to be able to route based on (S,G) information.

> One solution is to reserve an IP address range inside SSM range
> ( for example 232.0.0.0 to 232.0.255.255) and let the NAPT change,
> not only the source address of the sender, but also the multicast
> address of the sender to be unique in this range. In this way
> outside routers ( and receivers ) will consider traffic as coming
> from different channels (IP public ,G2) and ( IP public, G3) with
> G2 and G3 inside the said range.
The main problem is how to signal either the (S,G) channel or the
multicast group if this information is changed by a NAPT device.

E.g. in a typical IPTV scenario the signalling of the group or
channel information is done out-bband using servers providing
information about the IPTV channels. Clients are receiving this
information from the server (http/xml) to join the appropriate
IPTV channels.

If a sender (S1) sends multicast traffic to a group (G1) the receiver
joins (S1,G1) due to the information it got from the server. If the
NAPT device changes this to (S2,G1) or (S1,G2) the receiver is not
going to receive any traffic.

I'm not sure if this problem can be solved without making to many
assumptions about the applications deploying IP-Multicast.

  Regards

     Nic

________________________________

        De: Manfredi, Albert E [mailto:albert.e.manfredi@boeing.com]
        Enviado el: mar 05/06/2007 16:02
        Para: behave@ietf.org; mboned@ietf.org; magma@ietf.org
        Asunto: [magma] RE: [MBONED] FW: WGLC:
draft-ietf-behave-multicast-06.txt
      =20
      =20

        > -----Original Message-----
        > From: Dan Wing [mailto:dwing@cisco.com]
        > Sent: Monday, June 04, 2007 7:38 PM
        > To: mboned@ietf.org; magma@ietf.org
        > Cc: 'Behave WG'
        > Subject: [MBONED] FW: WGLC: draft-ietf-behave-multicast-06.txt
        >
        > The BEHAVE working group concluded its WGLC of
        > draft-ietf-behave-multicast-06 with no comments received.  I
        > would like
        > MBONED and MAGMA to please take a look at this draft prior to
        > its submission
        > to IESG.
        >
        > Please send replies to behave@ietf.org.  Thanks!
      =20
      =20
http://www.ietf.org/internet-drafts/draft-ietf-behave-multicast-06.txt:
      =20
        What if the NAPT or NAT device just passed IGMP queries and
reports
        through, changing only the source address of the reports, and
let the
        multicast router outside do all the hard work? I'm wondering
whether the
        IGMP aggregation function is mandatory in NAT/NATP.
      =20
        Bert
      =20
        _______________________________________________
        magma mailing list
        magma@ietf.org
        https://www1.ietf.org/mailman/listinfo/magma
      =20




------_=_NextPart_001_01C7A9B7.179BD85E
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<HTML dir=3Dltr><HEAD><TITLE>AW: [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 6.00.2900.3086" name=3DGENERATOR></HEAD>=0A=
<BODY>=0A=
<DIV id=3DidOWAReplyText25175 dir=3Dltr>=0A=
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 size=3D2>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><FONT size=3D3><FONT face=3D"Times =
New Roman">Leyman:<?xml:namespace prefix =3D o ns =3D =
"urn:schemas-microsoft-com:office:office" =
/><o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><o:p><FONT face=3D"Times New Roman" =
size=3D3>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><FONT size=3D3><FONT face=3D"Times =
New Roman">One possible solution ( out of the scope of the =
draft)</FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><o:p><FONT face=3D"Times New Roman" =
size=3D3>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><FONT size=3D3><FONT face=3D"Times =
New Roman">1. The sender behind the<SPAN style=3D"mso-spacerun: =
yes">&nbsp; </SPAN>NAPT sends the channel (Si,Gj) using a random Gj =
inside the multicast SSS range<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><FONT size=3D3><FONT face=3D"Times =
New Roman">2. Logically the sender expects that some one will be =
interested in the data, so the sender sends a unicast message (for =
example http) <SPAN style=3D"mso-spacerun: yes">&nbsp;</SPAN>to a =
&#8220;multicast index portal&#8221; describing the content and the =
channel (Si,Gj)<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><FONT size=3D3><FONT face=3D"Times =
New Roman">3. The NAPT changes the internal Si into a IP public in both =
cases: the unicast data and the multicast data using the same IP =
public.<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><FONT size=3D3><FONT face=3D"Times =
New Roman">4. The multicast portal knows the IP public and tries to =
connect with the multicast channel ( IP public, =
Gj)<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><FONT size=3D3><FONT face=3D"Times =
New Roman">If the multicast portal connects with ( IP public, Gj) there =
is no Gj collision and the portal shows the channel (IP public, Gj) in =
its index. Everybody can connect.<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><FONT size=3D3><FONT face=3D"Times =
New Roman">5. If after a few seconds the channel don&#8217;t appears in =
the index, the senders understands that there is a multicast Group =
collision and changes the Gj into another randomly selected =
Gk.<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><o:p><FONT face=3D"Times New Roman" =
size=3D3>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><FONT size=3D3><FONT face=3D"Times =
New Roman">Another business for the search =
engines.<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><o:p><FONT face=3D"Times New Roman" =
size=3D3>&nbsp;</FONT></o:p></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><FONT size=3D3><FONT face=3D"Times =
New Roman">Regards<o:p></o:p></FONT></FONT></SPAN></P>=0A=
<P class=3DMsoNormal style=3D"MARGIN: 0cm 0cm 0pt"><SPAN lang=3DEN-US =
style=3D"mso-ansi-language: EN-US"><BR><FONT size=3D3><FONT =
face=3D"Times New =
Roman">Alvaro<o:p></o:p></FONT></FONT></SPAN></P></FONT></DIV></DIV>=0A=
<DIV dir=3Dltr><BR>=0A=
<HR tabIndex=3D-1>=0A=
<FONT face=3DTahoma size=3D2><B>De:</B> Leymann, Nicolai =
[mailto:Nicolai.Leymann@t-systems.com]<BR><B>Enviado el:</B> vie =
08/06/2007 11:44<BR><B>Para:</B> Alvaro Fernandez; =
albert.e.manfredi@boeing.com; behave@ietf.org; mboned@ietf.org; =
magma@ietf.org<BR><B>Asunto:</B> AW: [magma] RE: [MBONED] FW: WGLC: =
draft-ietf-behave-multicast-06.txt<BR></FONT><BR></DIV>=0A=
<DIV>=0A=
<P><FONT size=3D2>Hi Everybody,<BR><BR>&gt; Hi all.<BR>&gt; If you have =
two sources from two different inside interfaces<BR>&gt; sending =
multicast data to the outside interface and both<BR>&gt; sources use the =
same multicast address (G1) then outside<BR>&gt; routers, using for =
example PIM-SM, will consider that there<BR>&gt; is only one multicast =
channel and will send all the traffic.<BR>&gt; Also hosts receiving all =
the traffic need to separate it.<BR>Right. This is from my point of view =
not the solution you want.<BR>The reason for SSM is to have different =
senders (source) using<BR>the same group and to be able to route based =
on (S,G) information.<BR><BR>&gt; One solution is to reserve an IP =
address range inside SSM range<BR>&gt; ( for example 232.0.0.0 to =
232.0.255.255) and let the NAPT change,<BR>&gt; not only the source =
address of the sender, but also the multicast<BR>&gt; address of the =
sender to be unique in this range. In this way<BR>&gt; outside routers ( =
and receivers ) will consider traffic as coming<BR>&gt; from different =
channels (IP public ,G2) and ( IP public, G3) with<BR>&gt; G2 and G3 =
inside the said range.<BR>The main problem is how to signal either the =
(S,G) channel or the<BR>multicast group if this information is changed =
by a NAPT device.<BR><BR>E.g. in a typical IPTV scenario the signalling =
of the group or<BR>channel information is done out-bband using servers =
providing<BR>information about the IPTV channels. Clients are receiving =
this<BR>information from the server (http/xml) to join the =
appropriate<BR>IPTV channels.<BR><BR>If a sender (S1) sends multicast =
traffic to a group (G1) the receiver<BR>joins (S1,G1) due to the =
information it got from the server. If the<BR>NAPT device changes this =
to (S2,G1) or (S1,G2) the receiver is not<BR>going to receive any =
traffic.<BR><BR>I'm not sure if this problem can be solved without =
making to many<BR>assumptions about the applications deploying =
IP-Multicast.<BR><BR>&nbsp; Regards<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp; =
Nic<BR><BR>________________________________<BR><BR>&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; De: Manfredi, Albert E [<A =
href=3D"mailto:albert.e.manfredi@boeing.com">mailto:albert.e.manfredi@boe=
ing.com</A>]<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Enviado el: =
mar 05/06/2007 16:02<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Para: =
behave@ietf.org; mboned@ietf.org; =
magma@ietf.org<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Asunto: =
[magma] RE: [MBONED] FW: =
WGLC:<BR>draft-ietf-behave-multicast-06.txt<BR>&nbsp;&nbsp;&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR><BR>&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; -----Original =
Message-----<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; From: =
Dan Wing [<A =
href=3D"mailto:dwing@cisco.com">mailto:dwing@cisco.com</A>]<BR>&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Sent: Monday, June 04, 2007 7:38 =
PM<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; To: =
mboned@ietf.org; =
magma@ietf.org<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Cc: =
'Behave WG'<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Subject: =
[MBONED] FW: WGLC: =
draft-ietf-behave-multicast-06.txt<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;&nbsp; &gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; The =
BEHAVE working group concluded its WGLC =
of<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; =
draft-ietf-behave-multicast-06 with no comments received.&nbsp; =
I<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; would =
like<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; MBONED and MAGMA =
to please take a look at this draft prior =
to<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; its =
submission<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; to =
IESG.<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
&gt;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &gt; Please send =
replies to behave@ietf.org.&nbsp; =
Thanks!<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR>&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp;&nbsp;<BR>http://www.ietf.org/internet-drafts/draft-i=
etf-behave-multicast-06.txt:<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp=
;<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; What if the NAPT or NAT =
device just passed IGMP queries =
and<BR>reports<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; through, =
changing only the source address of the reports, and<BR>let =
the<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; multicast router =
outside do all the hard work? I'm wondering<BR>whether =
the<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; IGMP aggregation =
function is mandatory in =
NAT/NATP.<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR>&nbsp;&nbsp;&n=
bsp;&nbsp;&nbsp;&nbsp;&nbsp; =
Bert<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;<BR>&nbsp;&nbsp;&nbsp;&=
nbsp;&nbsp;&nbsp;&nbsp; =
_______________________________________________<BR>&nbsp;&nbsp;&nbsp;&nbs=
p;&nbsp;&nbsp;&nbsp; magma mailing =
list<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
magma@ietf.org<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <A =
href=3D"https://www1.ietf.org/mailman/listinfo/magma">https://www1.ietf.o=
rg/mailman/listinfo/magma</A><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbs=
p;<BR><BR></FONT></P></DIV></BODY></HTML>
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From: Jonathan Rosenberg <jdrosen@cisco.com>
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To: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
Subject: Re: [BEHAVE] BoF request: STUN Control Usage
References: <02c601c7a957$66172430$d2dc46ab@amer.cisco.com><E1HwUiQ-000746-W1@megatron.ietf.org>	<8B1D53AEF7B03449A6D3771B3B7F850F03CADB8F@esebe103.NOE.Nokia.com>
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To take this a step further, I think sip-outbound is the primary 
benefactor of stun-control, not ICE.

The big pain point today are these awful keepalives we need to 
constantly be sending for sip-outbound. WHen UDP is used, its every 20 
or 30 seconds. That represents a huge load on the servers even when its 
STUN (though much better than REGISTER like what SBCs use today). With 
stun control, we could control this so that SIP servers would see 
reduction in stun volume by one or even two orders of magnitude. That is 
a real big bang for the buck. Also benefits wireless applications.

-Jonathan R.

Christer Holmberg (JO/LMF) wrote:

> Hi,
> 
> I think that SIP outbound is as "primary" customer of STUN as ICE is.
> 
> Regards,
> 
> Christer
> 
> 
>>-----Original Message-----
>>From: Erkki.Koivusalo@nokia.com [mailto:Erkki.Koivusalo@nokia.com] 
>>Sent: 8. kesäkuuta 2007 9:44
>>To: dbarrett@quinthar.com
>>Cc: behave@ietf.org
>>Subject: RE: [BEHAVE] BoF request: STUN Control Usage
>>
>>Hi, 
>>
>>
>>>Given that ICE is STUN's primary customer, STUN's fortunes 
>>
>>sink or swim 
>>
>>>with ICE's (just as ICE's fortunes sink or swim on SIP's).
>>
>>I think this statement is a bit exaggerated. There are 
>>already SIP VoIP UA implementations with RFC
>>3489 version of STUN but without ICE. They are succesfully 
>>deployed with environments where the NATs implement 
>>Endpoint-Independent Mapping behaviour.
>>
>>Thus fates of STUN and ICE are not as tightly connected to 
>>each other as you suggest. ICE provides added value for 
>>complex environments, but there are significant deployments 
>>which already work pretty well just with SIP combined with STUN.
>>
>>
>>>As such, until it's been demonstrated that ICE and SIP are already 
>>>swimming (which they might be), I'd be hesitant to spend 
>>
>>much energy on 
>>
>>>expanding STUN beyond the strict functionality required by ICE and 
>>>would instead refocus that effort on making ICE and SIP more 
>>
>>relevant 
>>
>>>in the real world.
>>
>>For the deployments I am referring to, it would be useful to 
>>have STUN extended with a mechanism to control the required 
>>keepalive interval for NAT binding:
>>
>>- Extensions needed for the already deployed set of
>>  protocols would be minimal.
>>
>>- There would be significant benefits both to the
>>  traffic in the network and the battery consumption
>>  of mobile VoIP devices.
>>
>>So I think this effort is valuable regardless on what happens to ICE.
>>
>>Regards,
>>
>>Erkki
>>
>>
>>_______________________________________________
>>Behave mailing list
>>Behave@ietf.org
>>https://www1.ietf.org/mailman/listinfo/behave
>>
> 
> 
> 
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
> 

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


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From behave-bounces@ietf.org Fri Jun 08 10:30:15 2007
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Subject: [BEHAVE] Re: [AVT] Moving draft-marjou-behave-app-rtp-keepalive to
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On 1 Jun 2007, at 18:15, Colin Perkins wrote:
> There has been some private discussion on adopting the draft on  
> Application Mechanisms for maintaining alive NAT mappings  
> associated with RTP flows (draft-marjou-behave-app-rtp- 
> keepalive-01.txt) as an AVT working group draft. This has  
> previously been discussed in AVT, BEHAVE and MMUSIC. Opinions on  
> whether this draft should become a working group draft, and on the  
> appropriate home for it, are solicited by 8 June 2007.

There were several comments in favour of this proposal, and no  
objections. Accordingly, we will take this draft as an AVT work item.

-- 
Colin Perkins
http://csperkins.org/




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Hi Everybody,

> Hi all.
> If you have two sources from two different inside interfaces=20
> sending multicast data to the outside interface and both=20
> sources use the same multicast address (G1) then outside=20
> routers, using for example PIM-SM, will consider that there=20
> is only one multicast channel and will send all the traffic.=20
> Also hosts receiving all the traffic need to separate it.
Right. This is from my point of view not the solution you want.
The reason for SSM is to have different senders (source) using
the same group and to be able to route based on (S,G) information.

> One solution is to reserve an IP address range inside SSM range=20
> ( for example 232.0.0.0 to 232.0.255.255) and let the NAPT change,=20
> not only the source address of the sender, but also the multicast=20
> address of the sender to be unique in this range. In this way=20
> outside routers ( and receivers ) will consider traffic as coming=20
> from different channels (IP public ,G2) and ( IP public, G3) with=20
> G2 and G3 inside the said range.
The main problem is how to signal either the (S,G) channel or the=20
multicast group if this information is changed by a NAPT device.

E.g. in a typical IPTV scenario the signalling of the group or
channel information is done out-bband using servers providing
information about the IPTV channels. Clients are receiving this=20
information from the server (http/xml) to join the appropriate=20
IPTV channels.

If a sender (S1) sends multicast traffic to a group (G1) the receiver
joins (S1,G1) due to the information it got from the server. If the
NAPT device changes this to (S2,G1) or (S1,G2) the receiver is not
going to receive any traffic.

I'm not sure if this problem can be solved without making to many
assumptions about the applications deploying IP-Multicast.

  Regards

     Nic

________________________________

	De: Manfredi, Albert E [mailto:albert.e.manfredi@boeing.com]
	Enviado el: mar 05/06/2007 16:02
	Para: behave@ietf.org; mboned@ietf.org; magma@ietf.org
	Asunto: [magma] RE: [MBONED] FW: WGLC:
draft-ietf-behave-multicast-06.txt
=09
=09

	> -----Original Message-----
	> From: Dan Wing [mailto:dwing@cisco.com]
	> Sent: Monday, June 04, 2007 7:38 PM
	> To: mboned@ietf.org; magma@ietf.org
	> Cc: 'Behave WG'
	> Subject: [MBONED] FW: WGLC: draft-ietf-behave-multicast-06.txt
	>
	> The BEHAVE working group concluded its WGLC of
	> draft-ietf-behave-multicast-06 with no comments received.  I
	> would like
	> MBONED and MAGMA to please take a look at this draft prior to
	> its submission
	> to IESG.
	>
	> Please send replies to behave@ietf.org.  Thanks!
=09
=09
http://www.ietf.org/internet-drafts/draft-ietf-behave-multicast-06.txt:
=09
	What if the NAPT or NAT device just passed IGMP queries and
reports
	through, changing only the source address of the reports, and
let the
	multicast router outside do all the hard work? I'm wondering
whether the
	IGMP aggregation function is mandatory in NAT/NATP.
=09
	Bert
=09
	_______________________________________________
	magma mailing list
	magma@ietf.org
	https://www1.ietf.org/mailman/listinfo/magma
=09



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To: "'Jonathan Rosenberg'" <jdrosen@cisco.com>,
	"'Christer Holmberg \(JO/LMF\)'" <christer.holmberg@ericsson.com>
Subject: RE: [BEHAVE] BoF request: STUN Control Usage
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> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]
> Subject: Re: [BEHAVE] BoF request: STUN Control Usage
> 
> The big pain point today are these awful keepalives we need to
> constantly be sending for sip-outbound. WHen UDP is used, its every 20
> or 30 seconds. That represents a huge load on the servers even when its
> STUN (though much better than REGISTER like what SBCs use today).

Again, just to push back on this a bit, how "huge" is a huge load?  In my
experience the keepalive traffic is the best kind: negligible, totally
predictable, and scales linearly with userbase.  All things being equal I'd
prefer it weren't necessary, but it's certainly not on the radar as being a
problem worth worrying about.  (If traffic is really such a huge problem,
let's just reduce the packet size -- a single packet is all that's really
needed to get the job done.)

Rather, the "cost" of unknown timeouts (the real problem STUN Control is
meant to solve) is engineering to accommodate them.  But unfortunately,
until STUN Control has near universal deployment -- and especially given
that there's no way to detect if a non-STUN-Control-compliant gateway is in
place -- we'll all still need to code exactly as if the STUN-Control
protocol didn't exist.  (Indeed, it'll be *more* costly as now we need to
support another protocol.)

In effect, STUN Control has a value curve looking like a square wave of
near-zero (even sub-zero) reduction of P2P software complexity up to
something like 90% global deployment (at which point application developers
are ok with just accepting a smaller userbase as the price of convenience).

I don't mean to be negative, but it just seems like a huge burden to
accomplish this and -- if I as a P2P software developer am meant to be the
benefactor -- I'd suggest the gain isn't worth the cost.

Personally, I would *much* rather that we just spend that same energy
getting BEHAVE deployed faster, or getting SIP/ICE/STUN used more widely
(starting with just quantifying how widely in fact they are being used).
Otherwise I fear we'd just create a document describing a neat protocol that
in theory does great things but in practice has limited (or unknown)
utility.

-david




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> -----Original Message-----
> From: ietf-secretariat@ietf.org [mailto:ietf-secretariat@ietf.org] 
> Sent: Thursday, June 07, 2007 9:00 PM
> To: ietf-announce@ietf.org
> Subject: Internet-Drafts Submission Cutoff Dates for the 69th 
> IETF Meeting in Chicago, IL, USA 
> 
> 
> There are two (2) Internet-Draft cutoff dates for the 69th 
> IETF Meeting in Chicago, IL, USA:
> 
> July 2nd: Cutoff Date for Initial (i.e., version -00) 
> Internet-Draft Submissions 
> 
> All initial Internet-Drafts (version -00) must be submitted 
> by Monday, 
> July 2nd at 9:00 AM ET. As always, all initial submissions with a 
> filename beginning with "draft-ietf" must be approved by the 
> appropriate WG Chair before they can be processed or announced.  The 
> Secretariat would appreciate receiving WG Chair approval by Monday, 
> June 25th at 9:00 AM ET.
> 
> July 9th: Cutoff Date for Revised (i.e., version -01 and higher) 
> Internet-Draft Submissions 
> 
> All revised Internet-Drafts (version -01 and higher) must be 
> submitted 
> by Monday, July 9th at 9:00 AM ET.
> 
> Initial and revised Internet-Drafts received after their respective 
> cutoff dates will not be made available in the Internet-Drafts 
> directory or announced until on or after Monday, July 23rd at 9:00 
> AM ET, when Internet-Draft posting resumes.  Please do not wait until 
> the last minute to submit.
> 
> Thank you for your understanding and cooperation. If you have any 
> questions or concerns, then please send a message to 
> internet-drafts@ietf.org.
> 
> The IETF Secretariat
> 
> FYI: The Internet-Draft cutoff dates as well as other 
> significant dates
> for the 69th IETF Meeting can be found at 
> http://www.ietf.org/meetings/cutoff_dates_69.html.
> 
> _______________________________________________
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Date: Fri, 8 Jun 2007 15:57:02 -0400
From: "Adam Fisk" <adamfisk@gmail.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>
Subject: Re: [BEHAVE] BoF request: STUN Control Usage
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On 6/8/07, Jonathan Rosenberg <jdrosen@cisco.com> wrote:
>
> To take this a step further, I think sip-outbound is the primary
> benefactor of stun-control, not ICE.


Agreed.

The big pain point today are these awful keepalives we need to
> constantly be sending for sip-outbound. WHen UDP is used, its every 20
> or 30 seconds. That represents a huge load on the servers even when its
> STUN (though much better than REGISTER like what SBCs use today). With
> stun control, we could control this so that SIP servers would see
> reduction in stun volume by one or even two orders of magnitude. That is
> a real big bang for the buck. Also benefits wireless applications.


Any idea how many of the deployed SIP servers use UDP?  In my understanding,
more and more implementations were switching to TCP because SIP messages
were hitting up against the 1500 byte MTU limit for UDP.  Maybe the wireless
guys are using UDP?  I never really considered using the UDP implementation
myself just because I didn't see the point.  I guess the point would be
better performance, but the added complication doesn't seem worth it.

I've even heard talk of excluding UDP altogether if we were to write SIP
over again.  I bring this up because the double CRLF keep alive for TCP with
SIP outbound is a really trivial addition.  If almost everyone's using TCP,
then your reasoning above for STUN control breaks down.

-Adam

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<br><br><div><span class="gmail_quote">On 6/8/07, <b class="gmail_sendername">Jonathan Rosenberg</b> &lt;<a href="mailto:jdrosen@cisco.com">jdrosen@cisco.com</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
To take this a step further, I think sip-outbound is the primary<br>benefactor of stun-control, not ICE.</blockquote><div><br>Agreed. <br></div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
The big pain point today are these awful keepalives we need to<br>constantly be sending for sip-outbound. WHen UDP is used, its every 20<br>or 30 seconds. That represents a huge load on the servers even when its<br>STUN (though much better than REGISTER like what SBCs use today). With
<br>stun control, we could control this so that SIP servers would see<br>reduction in stun volume by one or even two orders of magnitude. That is<br>a real big bang for the buck. Also benefits wireless applications.</blockquote>
<div><br>Any idea how many of the deployed SIP servers use UDP?&nbsp; In my understanding, more and more implementations were switching to TCP because SIP messages were hitting up against the 1500 byte MTU limit for UDP.&nbsp; Maybe the wireless guys are using UDP?&nbsp; I never really considered using the UDP implementation myself just because I didn&#39;t see the point.&nbsp; I guess the point would be better performance, but the added complication doesn&#39;t seem worth it.&nbsp; 
<br><br>I&#39;ve even heard talk of excluding UDP altogether if we were to write SIP over again.&nbsp; I bring this up because the double CRLF keep alive for TCP with SIP outbound is a really trivial addition.&nbsp; If almost everyone&#39;s using TCP, then your reasoning above for STUN control breaks down.
<br><br>-Adam<br></div><br></div><br>

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I'm unclear why the TURN server is expected to attempt to open up a TCP
connection to the remote host when it receives a connect request.  If the
TURN server is able to successfully make the connection to the remote host,
why in the world would the client be using TURN in the first place?  Why
wouldn't the client just bypass the costly relay and make the direct
connection?  Is this for highly controlled networks where clients may only
have outgoing access to the TURN server and not the remote host?

I can understand the allocating of permissions in response to a connect
request.  As such, it would make more sense to me that the connect request
be a "permission request" where no outgoing connection attempt is made.  As
far as I can tell, that outgoing connection attempt is a waste of precious
server resources in the vast majority of cases because it will almost always
fail.

Am I missing something?

Thanks.

-Adam Fisk

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I&#39;m unclear why the TURN server is expected to attempt to open up a TCP connection to the remote host when it receives a connect request.&nbsp; If the TURN server is able to successfully make the connection to the remote host, why in the world would the client be using TURN in the first place?&nbsp; Why wouldn&#39;t the client just bypass the costly relay and make the direct connection?&nbsp; Is this for highly controlled networks where clients may only have outgoing access to the TURN server and not the remote host?
<br><br>I can understand the allocating of permissions in response to a connect request.&nbsp; As such, it would make more sense to me that the connect request be a &quot;permission request&quot; where no outgoing connection attempt is made.&nbsp; As far as I can tell, that outgoing connection attempt is a waste of precious server resources in the vast majority of cases because it will almost always fail.
<br><br>Am I missing something?<br><br>Thanks.<br><br>-Adam Fisk<br>

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From behave-bounces@ietf.org Fri Jun 08 16:48:05 2007
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From: "Dan Wing" <dwing@cisco.com>
To: "'Adam Fisk'" <adamfisk@gmail.com>,
	"'Jonathan Rosenberg'" <jdrosen@cisco.com>
Subject: RE: [BEHAVE] BoF request: STUN Control Usage
Date: Fri, 8 Jun 2007 13:47:58 -0700
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> Any idea how many of the deployed SIP servers use UDP? 

There is some information from SIPit 20,
https://www.sipit.net/SIPit20_Summary but of course that
lacks market share information.

> In my 
> understanding, more and more implementations were switching 
> to TCP because SIP messages were hitting up against the 1500 
> byte MTU limit for UDP. 

Based on that SIPit information, 71% claimed to work fine
with fragmented UDP packets and 10% didn't know.  I guess the
remainder (19%) know they break with fragmented UDP.  I dunno
what those implementations plan to do -- maybe choose a 
better IP stack?

Additionally, under high traffic rates from a single IP 
address (such as a bunch of SIP clients behind a single NAPT), 
draft-heffner-frag-harmful describes problems with reassembly
which I doubt are sufficiently appreciated by those running
SIP over UDP.

> Maybe the wireless guys are using UDP?  I never really considered
> using the UDP implementation myself just because I didn't see the
> point.  I guess the point would be better performance, but the added
> complication doesn't seem worth it.
> 
> I've even heard talk of excluding UDP altogether if we were 
> to write SIP over again.  I bring this up because the double 
> CRLF keep alive for TCP with SIP outbound is a really trivial 
> addition.  If almost everyone's using TCP, then your 
> reasoning above for STUN control breaks down. 

To summarize, I believe you're saying if UDP keepalives create 
too much load on your server or network, just use TCP.

-d


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On 6/8/07, Dan Wing <dwing@cisco.com> wrote:
>
> > Any idea how many of the deployed SIP servers use UDP?
>
> There is some information from SIPit 20,
> https://www.sipit.net/SIPit20_Summary but of course that
> lacks market share information.


Interesting.  So it looks like most implementations support both.  I wonder
how many of those actually use the UDP implementation over the TCP in terms
of what clients actually connect with.  I know I wouldn't, but I wonder.


> Additionally, under high traffic rates from a single IP
> address (such as a bunch of SIP clients behind a single NAPT),
> draft-heffner-frag-harmful describes problems with reassembly
> which I doubt are sufficiently appreciated by those running
> SIP over UDP.


I would think there would be many issues along these lines.  It has to be
one of the thorniest parts of any implementation in any case.


> To summarize, I believe you're saying if UDP keepalives create
> too much load on your server or network, just use TCP.


Basically.  Put a slightly different way, if no one's using UDP in practice
anyway, then the TCP keep alive mechanism doesn't seem too onerous.  There
unfortunately seems to be little way to get a solid answer to that question,
though, so I'm not sure where to go with it.

Thanks.

-Adam

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<br><br><div><span class="gmail_quote">On 6/8/07, <b class="gmail_sendername">Dan Wing</b> &lt;<a href="mailto:dwing@cisco.com">dwing@cisco.com</a>&gt; wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
&gt; Any idea how many of the deployed SIP servers use UDP?<br><br>There is some information from SIPit 20,<br><a href="https://www.sipit.net/SIPit20_Summary">https://www.sipit.net/SIPit20_Summary</a> but of course that<br>
lacks market share information.</blockquote><div><br>Interesting.&nbsp; So it looks like most implementations support both.&nbsp; I wonder how many of those actually use the UDP implementation over the TCP in terms of what clients actually connect with.&nbsp; I know I wouldn&#39;t, but I wonder.
<br><br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>Additionally, under high traffic rates from a single IP<br>address (such as a bunch of SIP clients behind a single NAPT),
<br>draft-heffner-frag-harmful describes problems with reassembly<br>which I doubt are sufficiently appreciated by those running<br>SIP over UDP.</blockquote><div><br>I would think there would be many issues along these lines.&nbsp; It has to be one of the thorniest parts of any implementation in any case.
<br></div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>To summarize, I believe you&#39;re saying if UDP keepalives create<br>too much load on your server or network, just use TCP.
</blockquote><div><br>Basically.&nbsp; Put a slightly different way, if no one&#39;s using UDP in practice anyway, then the TCP keep alive mechanism doesn&#39;t seem too onerous.&nbsp; There unfortunately seems to be little way to get a solid answer to that question, though, so I&#39;m not sure where to go with it.
<br><br>Thanks.<br><br>-Adam<br></div><br></div><br>

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From behave-bounces@ietf.org Fri Jun 08 18:18:58 2007
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Date: Fri, 08 Jun 2007 18:18:30 -0400
From: Jonathan Rosenberg <jdrosen@cisco.com>
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Subject: [BEHAVE] STUN feature for v4/v6 transition
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STUNbis has some mysterious text in it, dealing with v4/v6 and mapped 
addreses. It says this:

  It is possible for an IPv4 host to receive a MAPPED-ADDRESS
    containing an IPv6 address, or for an IPv6 host to receive a MAPPED-
    ADDRESS containing an IPv4 address.  Clients MUST be prepared for
    this case.


The spec is entirely unclear on what it means to be prepared for this. 
Indeed, when would this even happen? Here is the deployment model 
envisioned for this case:


    Please view in a fixed-width font such as
                    Courier.



        +-----------+
        |    v4     |
        |  STUN     |
        |   Server  |
        |           |
        +-----------+               +----------+
                                    |          |
                                    | v4-only  |
                                    |   host   |
               v4 Internet          |          |
                                    +----------+







        +-----------------+
        |   v4 to v6 NAT  |
        +-----------------+


            v6 local net


         +----------+
         |          |
         | v6-only  |
         |  host    |
         |          |
         +----------+



There is a v6-only host behind a NAT. This NAT is v6 on the private side 
and v4 on the public side. It does twice-NAT; for packets outbound from 
the private to public sides, it converts the v6 source IP to a v4 
IP/port from a pool on the v4 public Internet side. It also converts the 
destination addresses from v6 to v4; the v6 address would need to be a 
v4-encapsulated v6 address or something for which there is a static 
mapping.

So the v6 only host sends a STUN request to a v6 destination. This goes 
through the NAT, which allocates the host a v4 binding and translates 
the destination to the corresponding v4 destination - the STUN server. 
The STUN server returns the v4 mapped address in the MAPPED-ADDRESS.

Now, this v6 host is going to get a STUN response that has a 
MAPPED-ADDRESS that is v4. If we consider the usage of STUN binding 
indication without ICE as a simple example, the v6-only host would 
include the v4 address into the SDP of an INVITE, and send it out. If 
the target host is v4-only as in the picture, the media will work from 
the v4-only host to the v6 host.

The story is similar with ICE and it works quite well in fact.

None of this requires the v6-only host to have a v4 address, it just 
needs to 'be prepared' to use a v4 address in this way. Now, there are 
lots of other things you need to assume are working in this story for 
the whole thing to come together, but this is the envisioned use case.

So, are people interested in this use case? Is this a valid transition 
scenario? If not, we can remove this vague text from 3489bis. If so, we 
need to add some clarity about what it means.

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


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To: "'Leymann, Nicolai'" <Nicolai.Leymann@t-systems.com>,
	<Alvaro@soportemv.com>, <albert.e.manfredi@boeing.com>,
	<behave@ietf.org>, <mboned@ietf.org>, <magma@ietf.org>
Subject: RE: [BEHAVE] AW: [magma] RE: [MBONED] FW:
	WGLC:draft-ietf-behave-multicast-06.txt
Date: Fri, 8 Jun 2007 18:06:57 -0700
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> Hi Everybody,
> 
> > Hi all.
> > If you have two sources from two different inside interfaces 
> > sending multicast data to the outside interface and both 
> > sources use the same multicast address (G1) then outside 
> > routers, using for example PIM-SM, will consider that there 
> > is only one multicast channel and will send all the traffic. 
> > Also hosts receiving all the traffic need to separate it.
>
> Right. This is from my point of view not the solution you want.
> The reason for SSM is to have different senders (source) using
> the same group and to be able to route based on (S,G) information.
> 
> > One solution is to reserve an IP address range inside SSM range 
> > ( for example 232.0.0.0 to 232.0.255.255) and let the NAPT change, 
> > not only the source address of the sender, but also the multicast 
> > address of the sender to be unique in this range. In this way 
> > outside routers ( and receivers ) will consider traffic as coming 
> > from different channels (IP public ,G2) and ( IP public, G3) with 
> > G2 and G3 inside the said range.
>
> The main problem is how to signal either the (S,G) channel or the 
> multicast group if this information is changed by a NAPT device.
> 
> E.g. in a typical IPTV scenario the signalling of the group or
> channel information is done out-bband using servers providing
> information about the IPTV channels. Clients are receiving this 
> information from the server (http/xml) to join the appropriate 
> IPTV channels.
> 
> If a sender (S1) sends multicast traffic to a group (G1) the receiver
> joins (S1,G1) due to the information it got from the server. If the
> NAPT device changes this to (S2,G1) or (S1,G2) the receiver is not
> going to receive any traffic.
> 
> I'm not sure if this problem can be solved without making to many
> assumptions about the applications deploying IP-Multicast.

I agree, and I'll leave such a recommendation out of the -07
revision of this document.  I expect to have -07 ready next week.

-d


>   Regards
> 
>      Nic
> 
> ________________________________
> 
> 	De: Manfredi, Albert E [mailto:albert.e.manfredi@boeing.com]
> 	Enviado el: mar 05/06/2007 16:02
> 	Para: behave@ietf.org; mboned@ietf.org; magma@ietf.org
> 	Asunto: [magma] RE: [MBONED] FW: WGLC:
> draft-ietf-behave-multicast-06.txt
> 	
> 	
> 
> 	> -----Original Message-----
> 	> From: Dan Wing [mailto:dwing@cisco.com]
> 	> Sent: Monday, June 04, 2007 7:38 PM
> 	> To: mboned@ietf.org; magma@ietf.org
> 	> Cc: 'Behave WG'
> 	> Subject: [MBONED] FW: WGLC: draft-ietf-behave-multicast-06.txt
> 	>
> 	> The BEHAVE working group concluded its WGLC of
> 	> draft-ietf-behave-multicast-06 with no comments received.  I
> 	> would like
> 	> MBONED and MAGMA to please take a look at this draft prior to
> 	> its submission
> 	> to IESG.
> 	>
> 	> Please send replies to behave@ietf.org.  Thanks!
> 	
> 	
> http://www.ietf.org/internet-drafts/draft-ietf-behave-multicas
> t-06.txt:
> 	
> 	What if the NAPT or NAT device just passed IGMP queries and
> reports
> 	through, changing only the source address of the reports, and
> let the
> 	multicast router outside do all the hard work? I'm wondering
> whether the
> 	IGMP aggregation function is mandatory in NAT/NATP.
> 	
> 	Bert
> 	
> 	_______________________________________________
> 	magma mailing list
> 	magma@ietf.org
> 	https://www1.ietf.org/mailman/listinfo/magma
> 	
> 
> 
> 
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave


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From: "Dan Wing" <dwing@cisco.com>
To: "'Bharat Joshi'" <bharat_joshi@infosys.com>
Subject: RE: [BEHAVE] Re: [MBONED] FW: WGLC: draft-ietf-behave-multicast-06.txt
Date: Fri, 8 Jun 2007 18:09:44 -0700
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> > > * In section 4.2, why is it that a NAPT device 
> > > implementing this draft
> > > needs to aggregate IGMP message only for IGMPv3? I think if it is
> > > implementing 'RFC4605', it will anyway do the aggregation.
> > 
> > It's my understanding that things will work -- somewhat 
> > sub-optimally --
> > with IGMPv1 or IGMPv2 without the NAT doing aggregation of IGMPv1/v2
> > messages and blindly just forwarding them upstream.
> > 
> > If my understanding is incorrect, please let me know (it 
> > may well be; multicast is not my expertise).

(By the way, to address that weakness, I have recently added 
Toerless Eckert as a co-author.)

> Things should work even if we do not aggregate IGMPv3 message 
> also. Why we do aggregation is that we do not want to send multiple 
> messages upstream?  Because upstream would forward multicast data 
> traffic if he knows that there is atleast one host interested in 
> a group.

But if blind forwarding is done, with no aggregation, the upstream
router will see arrival and leave packets arrive in the wrong order.
We adjusted the text in -07 and I think it should be clearer now.
It now reads:

***
   4.3.  IGMPv3 NAPT

   When a IGMPv3 proxying device receives an IGMP membership on an
   inside interface, it creates its own IGMP proxying membership state
   and its own IGMP forwarding table.  It then creates an independent
   IGMP membership report on its outside interface reporting the
   multicast groups/channels -- but there is no direct relationship or
   "relaying" of IGMP membership reports or queries across the
   interfaces.  The NAPT device will subsequently receive a multicast
   data packet on the outside ('public') interface and forward the
   multicast packet to inside ('internal') interfaces based on its IGMP
   forwarding table.

   By performing NAPT on IGMPv3 membership reports, the membership
   reports appear to originate from a single IGMPv3 reporter instead of
   different reporters.  Because IGMPv3 has different types of
   membership reports differentiating between status (IS_INCLUDE,
   IS_EXCLUDE) and change indication (e.g., TO_INCLUDE, TO_EXCLUDE), if
   a NAPT were to interleave reports from two or more reporters (joining
   and leaving the same groups) the NAPT would create a sequence of
   packets that are incompliant with an IGMPv3 reporter [RFC3376].  For
   this reason,

   REQ-4:  If a NAPT supports IGMPv3, the NAPT MUST implement [RFC4605].
           Such compliance causes the NAPT to aggregate the IGMPv3
           membership reports and report only the aggregated information
           upstream.

   Failure to do this aggregation will cause undesired temporary
   blackholing of multicast traffic.  For example, consider two hosts
   behind the same NAPT.  If one host is joining a session at the same
   time another is leaving the session, and the NAPT were to merely
   relay the join and leave upstream, the session will be terminated,
   and the join and leave announcements would not comply with section 5
   of [RFC3376].
***

> Aggregating IGMPv3 is a little more complex than V1 and V2.

Yes, agreed.

> I just wanted to point out that if an NAPT is implementing 
> RFC 4605 than it should be aggregating all versions of IGMP. 
>
> So why do you want to
> specifically mention the case for IGMPv3?

But IGMPv1 and IGMPv2 predated RFC4605, and there have been
and still are IGMPv1/v2-aware NATs that should "work" (but
perhaps not optimally) without implementing RFC4605.  I have
removed the old REQ-1 in -07 for this reason (and based on 
some other feedback).

In -07, I adjusted the requirements so that RFC4605 is only a 
requirement for NATs that support IGMPv3, and not a requirement 
for NATs that only do IGMPv1 or only IGMPv2.  I expect to publish
-07 in the middle of next week.

> > > * In case of IGMPv3, why is it that aggregation is a MUST? 
> > > Also I think
> > > when one of the hosts behind the NAPT generates LEAVE and another
> > > generates JOIN simultaneously, the session will be UP till the
> > > group-timer timesout and as JOIN is also received, it 
> > > will be updated
> > > and so session will always be UP. Am I mixing the session 
> > > created in
> > > NAPT with the state in upstream router?
> > 
> > Slightly after that requirement in the document is this explanation:
> > 
> >     >
> >     >  Failure to do this aggregation will cause undesired
> >     >  temporary blackholing of multicast traffic.  For example,
> >     >  consider two hosts behind the same NAPT. If one host is
> >     >  joining a session at the same time another is leaving the
> >     >  session, and the NAPT merely relays the join and leave
> >     >  upstream, the session will be terminated and the join and
> >     >  leave announcements do not comply with section 5 of <xref
> >     >  target="RFC3376"></xref>.
> >     >
> 
> Thats exactly why I raised this question. How does this creates an
> issue? This can happen in a host as well. Lets say application just
> started and send a join and immediately after this because of some
> failure, it goes down and send a leave. I am not sure what the problem
> is.

If it's a single host, it brought this behavior on itself.  I believe
this is identical to the behavior without a NAPT.

> > > * In section 4.4, though complete Multicast Range is 
> > > 224/4 but out of
> > > this range 232/8 is used for Source Specific Multicast or 
> > > Single-Source
> > > Multicast. So is the following statement syntactically correct
> > > 
> > > "Any-sosurce multicast (ASM) uses the IP addresses in the 
> > > 224.0.0.0 -
> > > 239.255.255.255 range [IANA-ALLOC]."
> > > 
> > > What I meant is that it looks to be incorrect to call this 
> > > range as ASM range.
> > 
> > You're right.  I pulled this from the Terminology section 
> > of RFC3569 
> > ("An Overview of Source-Specific Multicast (SSM)").  I have adjusted
> > my document so that it describes the ASM space as 224/8 through
> > 231/8 and 233/8 through 239/8 (with a hole for 224/8).
> > 
> 
> I think this may not be completely correct. This is because 239/8 is
> available to administrator and they can use this for SSM 
> within a given domain.

Is there some other way a NAT could make this determination?  Perhaps,
(see below) the NAT could determine the host is doing ASM because the
host is sending to the same multicast group; that may well be far easier
anyway...   Although, with SSM, RTP fixed this problem (in 
draft-ietf-avt-rtcpssm, where it suggests identifying hosts using
information other than the source transport address), thus with 
SSM traffic the NAT doesn't need to keep the UDP binding open for
60 minutes.  It's ASM RTP applications which don't follow 
draft-ietf-avt-rtcpssm's suggestions for identifying hosts that 
are the reason that requirement exists in draft-ietf-behave-multicast.

> > > * In section 4.4,
> > > 
> > >      a.  This UDP mapping SHOULD be destroyed when the host leaves
> > >          that host group.
> > > 
> > > How do we find out that host has left the group?  Please correct
> > > me if I am wrong.  We are talking about a host behind the NAPT
> > > device generating Multicast data traffic [UDP traffic].  A
> > > multicast host does not need to join the multicast group [By
> > > sending an IGMP join message] to send multicast traffic to that
> > > group.  So how does NAPT finds out when it can destroy this
> > > mapping and so this mapping will have to timeout before it can
> > > be removed.
> > 
> > Ah, good point which I hadn't considered.  I have removed that 
> > requirement.

Sorry -- while making this edit, I remembered the genesis of the
problem for RTP.  The problem only occurs when a host is receiving 
the same multicast stream that it is also sending to.  Specifically, 
when the host is receiving an RTP stream it will occasionally 
send RTCP reports.  With ASM, these RTCP reports are sent to the 
multicast address (so that all participants can see all other 
participants RTCP reports and adjust some RTP behaviors according 
to the size of the group).

If a host is only sending packets into an ASM stream (such as 
sending RTP packets) and isn't also listening to the stream,
we don't have this problem with RTP.  So we don't need to require
a NAT to have this 60-minute timeout.

So, I'm going to leave this requirement in -07, and have tried
to adjust the discussion/justification to describe why this
requirement is only necessary when a host is both a member of
the multicast group _and_ sends traffic to that same multicast
group.



Related, I have previously received pushback for this 60-minute 
timer being a MUST, because the difficulties this causes RTP 
aren't sufficiently significant to flat-out break interoperability 
if a NAT didn't implement that MUST.  I tend to agree with this 
comment, but would sure like to hear from others before I adjust
it.

-d

> Ok. So now how do you remove that mapping? I guess only once 
> that times out. Right?



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Subject: [BEHAVE] Re: STUN feature for v4/v6 transition
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Hi,

I think we should remove the text from the STUNbis spec. We should talk 
about these issues in draft-ietf-behave-turn-ipv6-02.txt. We have had 
some good discussions on this draft on this list and I will be putting 
together a new revision reflecting our conclusions as soon as I can 
(right now, I am on the road).

Regarding the use case you provide in your email, note that the NAT-PT 
spec is being moved to Historic. Therefore, you are right to question if 
that is a valid transition scenario:

http://www.ietf.org/internet-drafts/draft-ietf-v6ops-natpt-to-historic-00.txt

In any case, I agree with you that the STUNbis spec is not the place to 
deal with those issues. If we wanted to cover the few legacy networks 
using NAT-PT for v4/v6 transition directly in the STUNbis spec, we could 
say that a client receiving an address whose address family it 
understands, it can use it. If the address family, and thus the address 
itself, is not understood, it is discarded.

Cheers,

Gonzalo



Jonathan Rosenberg wrote:
> STUNbis has some mysterious text in it, dealing with v4/v6 and mapped 
> addreses. It says this:
> 
>  It is possible for an IPv4 host to receive a MAPPED-ADDRESS
>    containing an IPv6 address, or for an IPv6 host to receive a MAPPED-
>    ADDRESS containing an IPv4 address.  Clients MUST be prepared for
>    this case.
> 
> 
> The spec is entirely unclear on what it means to be prepared for this. 
> Indeed, when would this even happen? Here is the deployment model 
> envisioned for this case:
> 
> 
>    Please view in a fixed-width font such as
>                    Courier.
> 
> 
> 
>        +-----------+
>        |    v4     |
>        |  STUN     |
>        |   Server  |
>        |           |
>        +-----------+               +----------+
>                                    |          |
>                                    | v4-only  |
>                                    |   host   |
>               v4 Internet          |          |
>                                    +----------+
> 
> 
> 
> 
> 
> 
> 
>        +-----------------+
>        |   v4 to v6 NAT  |
>        +-----------------+
> 
> 
>            v6 local net
> 
> 
>         +----------+
>         |          |
>         | v6-only  |
>         |  host    |
>         |          |
>         +----------+
> 
> 
> 
> There is a v6-only host behind a NAT. This NAT is v6 on the private side 
> and v4 on the public side. It does twice-NAT; for packets outbound from 
> the private to public sides, it converts the v6 source IP to a v4 
> IP/port from a pool on the v4 public Internet side. It also converts the 
> destination addresses from v6 to v4; the v6 address would need to be a 
> v4-encapsulated v6 address or something for which there is a static 
> mapping.
> 
> So the v6 only host sends a STUN request to a v6 destination. This goes 
> through the NAT, which allocates the host a v4 binding and translates 
> the destination to the corresponding v4 destination - the STUN server. 
> The STUN server returns the v4 mapped address in the MAPPED-ADDRESS.
> 
> Now, this v6 host is going to get a STUN response that has a 
> MAPPED-ADDRESS that is v4. If we consider the usage of STUN binding 
> indication without ICE as a simple example, the v6-only host would 
> include the v4 address into the SDP of an INVITE, and send it out. If 
> the target host is v4-only as in the picture, the media will work from 
> the v4-only host to the v6 host.
> 
> The story is similar with ICE and it works quite well in fact.
> 
> None of this requires the v6-only host to have a v4 address, it just 
> needs to 'be prepared' to use a v4 address in this way. Now, there are 
> lots of other things you need to assume are working in this story for 
> the whole thing to come together, but this is the envisioned use case.
> 
> So, are people interested in this use case? Is this a valid transition 
> scenario? If not, we can remove this vague text from 3489bis. If so, we 
> need to add some clarity about what it means.
> 
> Thanks,
> Jonathan R.



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Date: Sun, 10 Jun 2007 10:20:37 -0400
From: Jonathan Rosenberg <jdrosen@cisco.com>
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Subject: [BEHAVE] Re: STUN feature for v4/v6 transition
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OK, so suggest that perhaps the text say that handling of a 
mapped-address using a different address family is outside the scope of 
this specification, and clients that don't know what to do with it, 
should treat it as an error.

-Jonathan R.

Gonzalo Camarillo wrote:

> Hi,
> 
> I think we should remove the text from the STUNbis spec. We should talk 
> about these issues in draft-ietf-behave-turn-ipv6-02.txt. We have had 
> some good discussions on this draft on this list and I will be putting 
> together a new revision reflecting our conclusions as soon as I can 
> (right now, I am on the road).
> 
> Regarding the use case you provide in your email, note that the NAT-PT 
> spec is being moved to Historic. Therefore, you are right to question if 
> that is a valid transition scenario:
> 
> http://www.ietf.org/internet-drafts/draft-ietf-v6ops-natpt-to-historic-00.txt 
> 
> 
> In any case, I agree with you that the STUNbis spec is not the place to 
> deal with those issues. If we wanted to cover the few legacy networks 
> using NAT-PT for v4/v6 transition directly in the STUNbis spec, we could 
> say that a client receiving an address whose address family it 
> understands, it can use it. If the address family, and thus the address 
> itself, is not understood, it is discarded.
> 
> Cheers,
> 
> Gonzalo
> 
> 
> 
> Jonathan Rosenberg wrote:
> 
>> STUNbis has some mysterious text in it, dealing with v4/v6 and mapped 
>> addreses. It says this:
>>
>>  It is possible for an IPv4 host to receive a MAPPED-ADDRESS
>>    containing an IPv6 address, or for an IPv6 host to receive a MAPPED-
>>    ADDRESS containing an IPv4 address.  Clients MUST be prepared for
>>    this case.
>>
>>
>> The spec is entirely unclear on what it means to be prepared for this. 
>> Indeed, when would this even happen? Here is the deployment model 
>> envisioned for this case:
>>
>>
>>    Please view in a fixed-width font such as
>>                    Courier.
>>
>>
>>
>>        +-----------+
>>        |    v4     |
>>        |  STUN     |
>>        |   Server  |
>>        |           |
>>        +-----------+               +----------+
>>                                    |          |
>>                                    | v4-only  |
>>                                    |   host   |
>>               v4 Internet          |          |
>>                                    +----------+
>>
>>
>>
>>
>>
>>
>>
>>        +-----------------+
>>        |   v4 to v6 NAT  |
>>        +-----------------+
>>
>>
>>            v6 local net
>>
>>
>>         +----------+
>>         |          |
>>         | v6-only  |
>>         |  host    |
>>         |          |
>>         +----------+
>>
>>
>>
>> There is a v6-only host behind a NAT. This NAT is v6 on the private 
>> side and v4 on the public side. It does twice-NAT; for packets 
>> outbound from the private to public sides, it converts the v6 source 
>> IP to a v4 IP/port from a pool on the v4 public Internet side. It also 
>> converts the destination addresses from v6 to v4; the v6 address would 
>> need to be a v4-encapsulated v6 address or something for which there 
>> is a static mapping.
>>
>> So the v6 only host sends a STUN request to a v6 destination. This 
>> goes through the NAT, which allocates the host a v4 binding and 
>> translates the destination to the corresponding v4 destination - the 
>> STUN server. The STUN server returns the v4 mapped address in the 
>> MAPPED-ADDRESS.
>>
>> Now, this v6 host is going to get a STUN response that has a 
>> MAPPED-ADDRESS that is v4. If we consider the usage of STUN binding 
>> indication without ICE as a simple example, the v6-only host would 
>> include the v4 address into the SDP of an INVITE, and send it out. If 
>> the target host is v4-only as in the picture, the media will work from 
>> the v4-only host to the v6 host.
>>
>> The story is similar with ICE and it works quite well in fact.
>>
>> None of this requires the v6-only host to have a v4 address, it just 
>> needs to 'be prepared' to use a v4 address in this way. Now, there are 
>> lots of other things you need to assume are working in this story for 
>> the whole thing to come together, but this is the envisioned use case.
>>
>> So, are people interested in this use case? Is this a valid transition 
>> scenario? If not, we can remove this vague text from 3489bis. If so, 
>> we need to add some clarity about what it means.
>>
>> Thanks,
>> Jonathan R.
> 
> 

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


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On Friday 08 June 2007 21:14:34 ext David Barrett wrote:
> Again, just to push back on this a bit, how "huge" is a huge load?  In my
> experience the keepalive traffic is the best kind: negligible, totally
> predictable, and scales linearly with userbase.

Handling keep-alives may well be totally neglectible on a modern and=20
well-implemented SIP server. And it is of course of neglectible CPU cost to=
 a=20
client.

*But* it is a HUGE battery W-h sink for mobile devices that have to wake-up=
=20
their radio every 30 seconds to send something. VoIP is not going to fly if=
=20
battery life is 5 times less than circuit-switched...

(...)
> Personally, I would *much* rather that we just spend that same energy
> getting BEHAVE deployed faster, or getting SIP/ICE/STUN used more widely
> (starting with just quantifying how widely in fact they are being used).
> Otherwise I fear we'd just create a document describing a neat protocol
> that in theory does great things but in practice has limited (or unknown)
> utility.

STUN control puts no extra requirement to the server side, and is entirely=
=20
optional on the client side, so I would not see it as a problem if you thin=
k=20
you do not need it.

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


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Subject: Re: [BEHAVE] connect request in draft-ietf-behave-turn-03.txt
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On Friday 08 June 2007 23:19:55 ext Adam Fisk wrote:
> I'm unclear why the TURN server is expected to attempt to open up a TCP
> connection to the remote host when it receives a connect request.  If the
> TURN server is able to successfully make the connection to the remote hos=
t,
> why in the world would the client be using TURN in the first place?

The other side could be using TURN too. If TURN servers were not capable of=
=20
actively connecting, there'd be a dead-lock.

> Why wouldn't the client just bypass the costly relay and make the direct
> connection?

If the either side does not implement ICE-TCP, it is typically not possible=
 to=20
negociate this properly; TURN becomes the only catch-all connectivity=20
solution.

This is particularly true when you act as the offerer, know that active=20
connections to you might not work, and have no idea what the answerer=20
situation is.

Or then we need to specify that the answerer must not use TURN when the=20
offerer does (or more generally when the offerer can be connected to).

But then...

> Is this for highly controlled networks where clients may only
> have outgoing access to the TURN server and not the remote host?

=2E..that would still be a problem.

=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
Subject: Re: [BEHAVE] STUN feature for v4/v6 transition
Date: Mon, 11 Jun 2007 10:02:29 +0300
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On Saturday 09 June 2007 01:18:30 ext Jonathan Rosenberg wrote:
> STUNbis has some mysterious text in it, dealing with v4/v6 and mapped
> addreses. It says this:
>
>   It is possible for an IPv4 host to receive a MAPPED-ADDRESS
>     containing an IPv6 address, or for an IPv6 host to receive a MAPPED-
>     ADDRESS containing an IPv4 address.  Clients MUST be prepared for
>     this case.
>
>
> The spec is entirely unclear on what it means to be prepared for this.
> Indeed, when would this even happen?

My main concern with this text is that it breaks terribly whenever the serv=
er=20
sends a redirect to another server, as the client has no idea how to send=20
traffic to a server that has a different address family than the one it is=
=20
using.

That being said, while v6/v6 NATs are not supposed to be used (so that IPv6=
=20
binding discovery should not be ever needed, though keep-alives might well=
=20
be), there are v6-to-v4 NATs. I just do not know if we need to care about=20
them, since I assume they are pretty scarce. There might also be v4-to-v6=20
NATs later on to connect legacy devices, but I guess they won't handle IPv6=
=20
addresses from STUN properly anyway.


Therefore, I am overall fine with suppressing the sentence.

Regards,

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


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Subject: RE: [BEHAVE] BoF request: STUN Control Usage
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> -----Original Message-----
> From: R=E9mi Denis-Courmont [mailto:remi.denis-courmont@nokia.com]
> Subject: Re: [BEHAVE] BoF request: STUN Control Usage
>=20
> Handling keep-alives may well be totally neglectible on a modern and
> well-implemented SIP server. And it is of course of neglectible CPU =
cost
> to a client.

I'm glad you agree that for non-mobile applications, keepalives are
negligible for clients and servers alike.  Is there anyone else who =
thinks
keepalive traffic presents real-world CPU or bandwidth problems for
non-mobile applications (and can show some kind of reality-based math to
back up the assertion), or can we lay the matter to rest?


> STUN control puts no extra requirement to the server side, and is =
entirely
> optional on the client side, so I would not see it as a problem if you
> think you do not need it.

The problem is its existence takes energy away from other incomplete,
higher-priority projects.  Even worse, merely discussing another =
reinvention
of STUN adds further delays to BEHAVE and SIP/ICE/STUN adoption as =
everyone
who might have considered it thinks "Whelp, I'll just do my own thing =
and
wait another year or so until they figure out what I'm supposed to do."

To repeat, I'd love it to exist and be widely deployed.  But until this
solution is deployed in a standard form with >90% coverage of all the
world's NAT's, it offers near zero value because every P2P application =
would
still need to build the exact same technology as today to accommodate
unknown timeouts.

Furthermore, to obtain >90% penetration of NATs will take *a lot* of =
time
and hard work.  This is a project that won't see the light of day for a
decade, at best.  I'd prefer those years of diligent, hard work were =
instead
focused on just getting BEHAVE more widely deployed in its current form, =
or
even just focused on increasing SIP/ICE/STUN's overall adoption.

-david



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Hi,

I have decided to not approve this BOF for Chicago. The main reason is 
that we should focus on getting the all the proposals currently in the 
pipe out of it. BEHAVE for example has a number of things, including 
STUN-BIS and TURN that we should get out in the near future. Having this 
BOF will be a distraction for this. I also expect MIDCOM and NSIS to 
have made progress towards publishing their solutions as RFCs.

I would recommend the proponents, interested and people seeing issue to 
keep a dialog and continue to prepare for a future BOF. I think the next 
IETF meeting in Vancouver in the fall will be a more suitable time for 
discussing the general topic on NAT control and any further work in IETF 
on this topic.

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] NAT control and STUN - some thoughts
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Hi,

Even if the BoF for IETF 69 was not approved, I think it's useful to
continue the discussion. Here are some thoughts I have about NAT/FW
control and STUN usage for it.

- As methodologies such as ICE and real-world applications such as Skype
or GoogleTalk have shown, NAT/FW traversal *can* be done relatively
effectively in general. For fixed devices and networks even the constant
keepalives to maintain state in the middleboxes is not an issue.
However, for wireless devices and networks the situation is quite
different. Keepalives, especially for UDP, consume the battery
relatively fast, as the radio transmitter/receiver needs to wake up
periodically to send otherwise unneeded packets. In practice any
application that needs to maintain a UDP mapping/pinhole, could not be
sustained for more than a few hours (as opposite to few days) in any
wide area cellular network, such as WCDMA/HSDPA or WiMAX (802.16e). TCP
already works much better, so its use e.g. for SIP is highly
recommended, but it still has the same problem in general.

- Most difficult are applications, where always-on connectivity needs to
be maintained. Examples include VoIP/IM/Presence. SIP, XMPP, Skype all
share the same problem, although typically SIP would be worst as UDP is
often used as the transport. Also, any protocols relying on UDP
encapsulation and maintaining always-on/long sessions, have the same
issues. For instance IPSec VPNs, Mobile IP, Teredo, ...

- The solution space is very challenging. The impact of IPv6 is still
very minor, also it has the same basic problem with firewalls. Of
course, the NAT/FW devices in cellular access could be configured with
longer timeouts (another useful idea in addition to TCP usage), but
there is no trivial way for applications to detect this.

- Something like STUN NAT control would be tempting for wireless devices
to implement, *if* some of the NAT box vendors would support it too, so
there would be at least some deployment potential. The biggest use cases
for UDP control would be in my opinion:
  - 1. Controlling the mappings for protocols using UDP encapsulation
such as IPSec, Mobile IP and Teredo. If the "path to the public address
space" could be controlled, these kind of protocols could allow quite
powerful connectivity with just a single UDP mapping.
  - 2. ICE optimization. Gathering candidates before sending offer or
answer can cause some delay. With STUN control, at least one candidate
could be maintained quite cheaply just waiting for a call to be
made/received, so that the offer/answer could be generated immediately.
  - I'm not so sure on SIP outbound. TCP should be used for that. More
later on.

- Another useful thing would be something that would work for TCP, since
that would help apps like SIP, e-mail etc. I'm not sure how feasible
STUN control would be. I guess in principle the tagging method in -02
draft would work, but also the TCP peer (server) would need to run a
STUN server in the same port as the app protocol and mux the two.
Current deployment of STUN would not help in bootsraping this. (Actually
SIP outbound had the muxing stuff for TCP in an earlier version. I was
against if for different reasons. I don't want to go and open that
anymore, but certainly the debate can now be seen in a bit differet
light too.)

- The biggest problem I see (in addition to usual issues with
deployment, topologies etc.) is that of a non-STUN capable firewall on
the path. This could not be detected. So, it would be hard to know when
you really have the path under control and can reduce the keepalive
messages.

Personally I think we should continue the work in this area, I don't see
the problem going away for the next few years, actually it will probably
hit the headlines sooner or later as wireless broadband starts to become
available and people notice that they still can't really run their
Skype, Gizmo or Gtalk clients in the way they would want. Some say that
the ISPs should just go for v6, but I'm doubtful.

Regards,
	Markus =20


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Hi Magnus,

I can understand your reasoning. I just hope that we don't end up in a
situation where there is the formally agreed solution X that no-one
implements and whose only function would be to block the development of
solutions Y and Z, which perhaps could have more implementation
interest.=20

Instead of a BoF, I think it would still be useful to have slot in
BEHAVE to discuss the proposal. A bar BoF should definitely be held, at
least ;)

Markus
=20

>-----Original Message-----
>From: ext Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
>Sent: 11 June, 2007 17:17
>To: behave@ietf.org
>Subject: Re: [BEHAVE] BoF request: STUN Control Usage
>
>Hi,
>
>I have decided to not approve this BOF for Chicago. The main=20
>reason is that we should focus on getting the all the=20
>proposals currently in the pipe out of it. BEHAVE for example=20
>has a number of things, including STUN-BIS and TURN that we=20
>should get out in the near future. Having this BOF will be a=20
>distraction for this. I also expect MIDCOM and NSIS to have=20
>made progress towards publishing their solutions as RFCs.
>
>I would recommend the proponents, interested and people seeing=20
>issue to keep a dialog and continue to prepare for a future=20
>BOF. I think the next IETF meeting in Vancouver in the fall=20
>will be a more suitable time for discussing the general topic=20
>on NAT control and any further work in IETF on this topic.
>
>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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>


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Date: Mon, 11 Jun 2007 16:59:39 -0400
Subject: Re: [BEHAVE] NAT control and STUN - some thoughts
From: Melinda Shore <mshore@cisco.com>
To: <Markus.Isomaki@nokia.com>, <behave@ietf.org>, Dan Wing <dwing@cisco.com>
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On 6/11/07 4:07 PM, "Markus.Isomaki@nokia.com" <Markus.Isomaki@nokia.com>
wrote:
> - As methodologies such as ICE and real-world applications such as Skype
> or GoogleTalk have shown, NAT/FW traversal *can* be done relatively
> effectively in general.

One quick point - Skype in particular works by bypassing firewall
access control decisions, and that isn't a good thing.  This is
one area in which the distinction between firewall and NAT matters
a great deal.  Firewalls enforce access policies, occasionally
very complex access policies.  NATs, to the extent they deal with
policy at all, deal with address policy, and in the pure NAT case
you're not dealing with authorization questions as you are with
firewalls.

Anything that goes through a firewall without giving the firewall
adequate opportunity to say "yes" or "no" is probably not okay, and
consequently it's important to draw distinctions between firewalls
and NATs in this context.

Melinda
 


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Subject: Re: [BEHAVE] BoF request: STUN Control Usage
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On 8-jun-2007, at 22:47, Dan Wing wrote:

> Based on that SIPit information, 71% claimed to work fine
> with fragmented UDP packets and 10% didn't know.  I guess the
> remainder (19%) know they break with fragmented UDP.  I dunno
> what those implementations plan to do -- maybe choose a
> better IP stack?

<delurk>

A problem with fragmentation is that the port numbers are only in the  
first fragment so fragmented packets often have trouble getting by  
firewalls.

</delurk>


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Subject: RE: [BEHAVE] BoF request: STUN Control Usage
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> > Based on that SIPit information, 71% claimed to work fine
> > with fragmented UDP packets and 10% didn't know.  I guess the
> > remainder (19%) know they break with fragmented UDP.  I dunno
> > what those implementations plan to do -- maybe choose a
> > better IP stack?
> 
> <delurk>
> 
> A problem with fragmentation is that the port numbers are 
> only in the  
> first fragment so fragmented packets often have trouble getting by  
> firewalls.
> 
> </delurk>

Good point, thanks for delurking briefly.  NATs have a similar difficultly.
I don't know if the ~20% of stacks reported at SIPit that are known to break
with fragmented UDP were considering a firewall (or NAT) between their stack
and their next-hop SIP device, though.

-d


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Hi,

Gorry and me just submitted a new -01 version of the "UDP Usage  
Guidelines for Application Designers" draft that TSVWG is putting  
together as a BCP.

Compared to the -00 version, this draft adds a new Section 3.5 that  
gives middlebox traversal guidelines.

(I'm cross-posting to AVT, MMUSIC and BEHAVE, because this addition  
was triggered by the recent discussion around draft-marjou-behave-app- 
rtp-keepalive. Please discuss draft-ietf-tsvwg-udp-guidelines on the  
TSVWG list; I'm setting the reply-to accordingly.)

We'd be interested in comments, both from TSV folks but especially  
from RAI people, who might not have had this on their radar.

Lars

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From behave-bounces@ietf.org Tue Jun 12 04:33:05 2007
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Subject: RE: [BEHAVE] NAT control and STUN - some thoughts
Date: Tue, 12 Jun 2007 11:32:12 +0300
Message-ID: <C84E0A4ABA6DD74DA5221E0833A35DF309BF5018@esebe101.NOE.Nokia.com>
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From: <Markus.Isomaki@nokia.com>
To: <mshore@cisco.com>, <behave@ietf.org>, <dwing@cisco.com>
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Hi,

Yes, understood and agreed.

For a client vendor/implemeter it's an interesting dilemma whether to
support something like HTTP/TLS "tunneling". On one hand it would make
your client work in more environments (users would typically see this as
a good thing), while on the other hand you are giving tools to break
network policies (enterprise/network admins see this as a bad thing).

Markus
 =20

>-----Original Message-----
>From: ext Melinda Shore [mailto:mshore@cisco.com]=20
>Sent: 12 June, 2007 00:00
>To: Isomaki Markus (Nokia-SIR/Espoo); behave@ietf.org; Dan Wing
>Subject: Re: [BEHAVE] NAT control and STUN - some thoughts
>
>On 6/11/07 4:07 PM, "Markus.Isomaki@nokia.com"=20
><Markus.Isomaki@nokia.com>
>wrote:
>> - As methodologies such as ICE and real-world applications such as=20
>> Skype or GoogleTalk have shown, NAT/FW traversal *can* be done=20
>> relatively effectively in general.
>
>One quick point - Skype in particular works by bypassing=20
>firewall access control decisions, and that isn't a good=20
>thing.  This is one area in which the distinction between=20
>firewall and NAT matters a great deal.  Firewalls enforce=20
>access policies, occasionally very complex access policies. =20
>NATs, to the extent they deal with policy at all, deal with=20
>address policy, and in the pure NAT case you're not dealing=20
>with authorization questions as you are with firewalls.
>
>Anything that goes through a firewall without giving the=20
>firewall adequate opportunity to say "yes" or "no" is probably=20
>not okay, and consequently it's important to draw distinctions=20
>between firewalls and NATs in this context.
>
>Melinda
>=20
>


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From behave-bounces@ietf.org Thu Jun 14 05:51:52 2007
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Date: Thu, 14 Jun 2007 11:51:34 +0200
From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [BEHAVE] BoF request: STUN Control Usage
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Markus.Isomaki@nokia.com skrev:
> Hi Magnus,
> 
> I can understand your reasoning. I just hope that we don't end up in a
> situation where there is the formally agreed solution X that no-one
> implements and whose only function would be to block the development of
> solutions Y and Z, which perhaps could have more implementation
> interest. 

This is clearly one of the topics that a future BOF to some degree will 
need to deal with. There are already developed solutions, and the 
question that any proponent needs to answer is why they aren't working. 
I think STUN control have some appealing arguments, but I don't think 
this is the end of the discussion.

> 
> Instead of a BoF, I think it would still be useful to have slot in
> BEHAVE to discuss the proposal. A bar BoF should definitely be held, at
> least ;)
> 

No, not a slot in BEHAVE. As I said, lets focus BEHAVE on the WG's 
current issues. We also have requested a new mailing list to move the 
STUN control discussion too. It will be announced as soon it is working. 
A Bar BOF is fine and in fact encouraged.

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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Subject: announcing SAFE mailing list [was: RE: [BEHAVE] BoF request: STUN
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Magnus Westerlund wrote:
> Markus.Isomaki@nokia.com skrev:
...
> > Instead of a BoF, I think it would still be useful to have slot in
> > BEHAVE to discuss the proposal.  A bar BoF should definitely be
> > held, at least ;)
>
> No, not a slot in BEHAVE. As I said, lets focus BEHAVE on the WG's
> current issues.  We also have requested a new mailing list to move
> the STUN control discussion too.  It will be announced as soon it is
> working.

The new mailing list has been created, "Self-Address Fixing Evolution"
(SAFE):
 
     Evolution of the protocol and procedures of RFC3489 and 
     draft-ietf-behave-rfc3489bis, so that middleboxes can be 
     discovered, queried, and controlled.

Subscribe at:  https://www1.ietf.org/mailman/listinfo/safe

Please join the SAFE mailing list if you are interested in this topic.

> A Bar BOF is fine and in fact encouraged.

I'll schedule that in a few days on the SAFE mailing list.  I am
leaning towards Monday or Thursday for that, after dinner (Tuesday 
is the social and a SPIT bar BoF has been called for Wednesday).

-d


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From: "Dan Wing" <dwing@cisco.com>
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Date: Sat, 16 Jun 2007 10:28:33 -0700
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Subject: [BEHAVE] need more Nomcom 2007-8 volunteers
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If you're selected, it's typically only a few hours of effort.

-d

> -----Original Message-----
> From: Lakshminath Dondeti [mailto:ldondeti@qualcomm.com] 
> Sent: Saturday, June 16, 2007 9:55 AM
> To: wgchairs@ietf.org; irsg@isi.edu; bofchairs@ietf.org
> Subject: Request to forward the Nomcom 2007-8 Announcement to 
> lists youmanage
> 
> Folks,
> 
> We have around 50 volunteers so far for Nomcom 2007-8 voting member 
> selection.  We need a lot more!  Not everyone subscribes to the 
> IETF-Announce list.  Could you please forward the announcement 
> https://datatracker.ietf.org/public/show_nomcom_message.cgi?id
> =1234 to 
> WG, RG, BoF, Area lists that you manage?
> 
> Thanks in advance.
> 
> best regards,
> Lakshminath


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From behave-bounces@ietf.org Wed Jun 20 12:45:30 2007
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Subject: [BEHAVE] FW: Nomcom 2007-8: Notes on Voting Member Time Commitment
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> -----Original Message-----
> From: Lakshminath Dondeti [mailto:ldondeti@qualcomm.com] 
> Sent: Tuesday, June 19, 2007 11:35 PM
> To: ietf-announce@ietf.org
> Subject: Nomcom 2007-8: Notes on Voting Member Time Commitment
> 
> Folks
> 
> Some of you have asked for clarifications on the time commitments of 
> voting members.  BCP 10 has the details of the process and 
> the role of 
> the voting and non-voting members.  Here are some notes for 
> your quick 
> reference on how it might be in practice:
> 
> If all goes well with the selection process, the members begin their 
> work at the Chicago IETF meeting.  The plan is for the 
> members to attend 
> the Chicago and Vancouver IETF meetings to interview and/or gather 
> feedback and input from the community, including people in leadership 
> positions: WG and RG chairs, the IAOC, the IAB and the IESG.  We can 
> address exceptional circumstances on a case-by-case basis, 
> but that is 
> the general expectation.
> 
> The Nomcom will then spend about a month (4-5 weeks) self-organizing. 
> There will be 1-2 hrs per week of conference calls and BCP 10 as the 
> reading assignment.  As with anything IETF, there will be some email 
> traffic too.
> 
> Subsequent to that, the candidate selection phase begins which lasts 
> about 4 months.  During that phase there is a good amount of reading: 
> especially candidate statements and community feedback.  
> There will also 
> be discussions and deliberations via 2 hrs per week of 
> conference calls 
> and email.  That's under normal operating conditions.  If there are 
> interim openings, there will be slightly additional work.
> 
> We (the entire nomcom) can make efforts to distribute the work evenly 
> over that period of 5 months, so as to be as non-disruptive 
> to our day 
> jobs as possible.
> 
> If you have any specific questions or requests for clarifications, 
> please do not hesitate to contact me.
> 
> thanks,
> Lakshminath
> Nomcom 2007-8 Chair
> 
> _______________________________________________
> IETF-Announce mailing list
> IETF-Announce@ietf.org
> https://www1.ietf.org/mailman/listinfo/ietf-announce


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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		: Multicast Requirements for a Network Address (and port) Translator (NAT)
	Author(s)	: T. Eckert, D. Wing
	Filename	: draft-ietf-behave-multicast-07.txt
	Pages		: 15
	Date		: 2007-6-21
	
This document specifies requirements for a Network Address (and port)
   Translator (NAT) that supports any source multicast or source
   specific IP multicast.  A multicast-capable NAT device that adheres
   to the requirements of this document can optimize the operation of
   multicast applications that are generally unaware of multicast NAT
   devices.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-behave-multicast-07.txt

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From behave-bounces@ietf.org Thu Jun 21 20:18:24 2007
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Subject: [BEHAVE] multicast senders behind an IGMP Proxy
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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



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A day after submitting -07, I noticed there is some confusing 
text with IGMPv3 (REQ-7)'s ASM/SSM send/receive requirements
and the transport protocol requirements (REQ-9/REQ-10).

I will be clarifying this by moving some text around, so please
don't waste your time with -07.  -08 will be out shortly.

-d


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From behave-bounces@ietf.org Fri Jun 22 06:02:20 2007
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Date: Fri, 22 Jun 2007 15:31:16 +0530
From: Prashant Jhingran <prashantj@huawei.com>
In-reply-to: <1d0c01c7b462$d4c148a0$c2f0200a@amer.cisco.com>
To: 'Dan Wing' <dwing@cisco.com>, behave@ietf.org, magma@ietf.org
Message-id: <000601c7b4b4$46c224f0$5f17120a@china.huawei.com>
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Subject: [BEHAVE] RE: [magma] multicast senders behind an IGMP Proxy
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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.

- 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 Fri Jun 22 08:02:53 2007
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From: Brian Haberman <brian@innovationslab.net>
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Cc: magma@ietf.org, behave@ietf.org, 'Dan Wing' <dwing@cisco.com>
Subject: [BEHAVE] Re: [magma] multicast senders behind an IGMP Proxy
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In IPv6, the scoping can be accomplished simply with the existing scope
bits without having to rely on the Hop Count.

The home router simply acts as a scope border and does not forward
traffic to the Internet.

Regards,
Brian


Prashant Jhingran wrote:
> 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.
> 
> - 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
> 
> 
> 
> _______________________________________________
> magma mailing list
> magma@ietf.org
> https://www1.ietf.org/mailman/listinfo/magma


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From behave-bounces@ietf.org Fri Jun 22 08:15:37 2007
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Date: Fri, 22 Jun 2007 17:44:36 +0530
From: Prashant Jhingran <prashantj@huawei.com>
In-reply-to: <467BBA4D.2080705@innovationslab.net>
To: 'Brian Haberman' <brian@innovationslab.net>
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Cc: magma@ietf.org, behave@ietf.org, 'Dan Wing' <dwing@cisco.com>
Subject: [BEHAVE] RE: [magma] multicast senders behind an IGMP Proxy
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Oh yes, multicast link-local addressing scheme should be
good enough to tackle this issue for IPv6.

My suggestions were applicable for IPv4 alone.

Regards,
Prashant Jhingran

Huawei Technologies
+91-80-41117676 Ext -7175
+91-9448927814

"Sing, dance, serve, meditate and celebrate" 
 - www.artofliving.org  

-----Original Message-----
From: Brian Haberman [mailto:brian@innovationslab.net] 
Sent: Friday, June 22, 2007 5:32 PM
To: prashantj@huawei.com
Cc: 'Dan Wing'; behave@ietf.org; magma@ietf.org; 'Toerless
Eckert'
Subject: Re: [magma] multicast senders behind an IGMP Proxy

In IPv6, the scoping can be accomplished simply with the
existing scope
bits without having to rely on the Hop Count.

The home router simply acts as a scope border and does not
forward
traffic to the Internet.

Regards,
Brian


Prashant Jhingran wrote:
> 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.
> 
> - 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
> 
> 
> 
> _______________________________________________
> magma mailing list
> magma@ietf.org
> https://www1.ietf.org/mailman/listinfo/magma




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From: "Dan Wing" <dwing@cisco.com>
To: "'John Zwiebel'" <jzwiebel@cisco.com>
Date: Sun, 24 Jun 2007 21:48:08 -0700
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Cc: magma@ietf.org, behave@ietf.org
Subject: [BEHAVE] RE: [magma] multicast senders behind an IGMP Proxy
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> >  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).
> >
> 
> I have a difficult time imagining a home network that would have to
> be subnetted.
> 
> And although I can imagine 2 HDTVs watching the same channel 
> sometimes, I don't believe it will happen often enough that 
> there would be any BW saving inside one's home.

So you're saying nobody would use multicast within their own 
network?  Here is what we plan on adding to the next revision of
the NAT Multicast document:

<begin>
   As many NATs are located adjacent to bandwidth-constrained access
   links, it is important that multicast senders communicating with
   multicast receivers behind the NAT not have their flows consume
   bandwidth on the access link.  This is accomplished by applications
   using administratively scoped IP addresses.

   REQ-6:   A NAT MUST NOT forward administratively scoped multicast
            traffic (239/8) [RFC2365] from its 'inside' interface(s) to
            its 'outside' interface, unless the NAT has been configured
            to do so.
<end>

Please let me know if you object to that text.

-d


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Subject: [BEHAVE] Re: [magma] multicast senders behind an IGMP Proxy
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On Jun 21, 2007, at 5:18 PM, Dan Wing wrote:

>  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).
>

I have a difficult time imagining a home network that would have to
be subnetted.

And although I can imagine 2 HDTVs watching the same channel sometimes,
I don't believe it will happen often enough that there would be any
BW saving inside one's home.


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