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Date: Mon, 1 Jan 2007 18:18:19 +0100
From: Marc Petit-Huguenin <petithug@acm.org>
To: jdrosen@cisco.com
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Subject: [BEHAVE] Review of draft-ietf-behave-rfc3489bis-05 [update]
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I found an additional issue that I would like to add to my review of
draft-ietf-behave-rfc3489bis-05:

9. Section 8.3.2, paragraph 17:

"Any response response between 100 and 299 MUST result in the cessation
of request retransmissions, but otherwise is discarded."

The problem is that the "final" response that will be sent before the
7900ms timeout will not be retransmitted if it is lost.  The request
will eventually be retried on the next server, but it is still a problem
if the usage does not use DNS, or if there is only one SRV RR.

Here some suggestions to fix this:

o The simplest way is to remove this sentence, but I think that there is
a real need to have a way to stop the retransmissions until the response
can be calculated.  An example that comes to mind would be if the STUN
server needs to contact a authentication server (e.g. RFC 4590) that
will take more than 100ms to respond.

o Another way would be to keep the sentence and do what was done with
the 200/ACK for SIP INVITE.  For STUN this would mean to add an ACK
Indication, and retransmit the "final" response until an ACK Indication
with the same transaction id is received.

o Another way would be to remove the sentence, and add a new error
response code 448 Response Delayed that will signal that the response
will be sent later in a new Update request, but in the opposite
direction.  Here's an example for an Allocate request:

 UA                              TURN
  | Allocate request              |
1 |------------------------------>|
  |   Allocate Error Response 448 |
2 |<------------------------------|
  :                               :
  |                Update Request |
3 |<------------------------------|
  | Update Response               |
4 |------------------------------>|

1: An Allocate request is sent to the TURN server.
2: The TURN server needs more time to allocate the ports, and so
responds with a 448 response that stops the retransmissions.
3: After some time the TURN server manages to allocate the ports and so
sends an Update Request in the direction of the UA.  It copies in the
request all the attributes that it should have copied in the Allocate
Response if it was able to do it. The request also contains an
additional attribute containing the transaction id of the original
transaction.  It needs to send the request in the next 30 seconds after
receiving the Allocate request to be sure that the UDP binding is still
active in the NAT.
4: The UA responds with an Update Response to stop the retransmissions.

The advantage of this last method is that it does not change the STUN
transaction behavior.

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From behave-bounces@ietf.org Wed Jan 03 16:17:32 2007
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Thread-Topic: STUN FINGERPRINT
Thread-Index: AccvfH/BzK5HCLFBQOSM+DLZ0d22Zw==
From: "Kevin Johns" <K.Johns@CableLabs.com>
To: <behave@ietf.org>
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Subject: [BEHAVE] STUN FINGERPRINT
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All,
=20
Per the text of the current STUN draft (-05), it states that the value
is computed as the CRC-32 of the STUN message up to (but excluding) the
FINGERPRINT attribute itself.
=20
The question is on the length value used in the message length field of
the STUN header. Is the FINGERPRINT calculated using the message length
including the FINGERPRINT attribute or the message length minus the
FINGERPRINT attribute?
=20
I could not find this discussed anywhere in the draft and we need to
know which way to implement.
=20
Regards,
Kevin Johns

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.2963" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2>All,</FONT></SPAN></DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial size=3D2>Per =
the text of the=20
current STUN draft (-05), it states that the value is computed as the =
CRC-32 of=20
the STUN message up to (but excluding) the FINGERPRINT attribute=20
itself.</FONT></SPAN></DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial size=3D2>The =
question is on=20
the length value used in the message length field of the STUN header. Is =
the=20
FINGERPRINT calculated using the message length including the =
FINGERPRINT=20
attribute or the message length minus the FINGERPRINT=20
attribute?</FONT></SPAN></DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial size=3D2>I =
could not find=20
this discussed anywhere in the draft and we need to know which way to=20
implement.</FONT></SPAN></DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial size=3D2>Kevin=20
Johns</FONT></SPAN></DIV></BODY></HTML>

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From behave-bounces@ietf.org Fri Jan 05 06:18:36 2007
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Hi,

A bit late for the rfc3489bis-05 WGLC, but better late than never.
This is a great spec, but there's still some issues to discuss.
I tried not to repeat the already commented issues.

Here's the comments:

-6. (page 13): Requirement about selecting a new TID value is a bit
iffy.
 It requires that the transaction ID's should be uniformly and randomly
 selected on the 0 to 2**96 - 1 scale, but then again, a client should
not
 repeat previous TID's.
 These two requirements are conflicting with each other. A client really
 can not remember all of its used old TID's forever. Neither can it use
an
 incrementing counter as a part of the TID to achieve the non-repeating
 characteristic due to the random selection requirement. AFAIK for basic
 STUN the requirement of uniform & random value selection is enough. If
we
 really do need the "not repeating" part for something specific, it
should
 be clearly stated here.

-7.1 (page 14): The retry timeout value calculation with this new style
 might be a bit of an overkill. Is it really reasonable to specify such
 an adaptive algorithm just for retransmissions? Would't it just be
enough
 that the implementations would be allowed to configure and/or the retry
 starting time value, and then double it (and so on like in previous
 rfc3489(-bis) versions)?
 I do understand that this mechanism is just a SHOULD level requirement,
 but still, do we have a need to make it so "dynamic and wise"?

-8.1 (page 16): Service discovery (DNS SRV & port) specifications in
 rfc3489bis-05 are not backwards compatible with RFC3489 (as I stated
earlier).
 Rohan Mahy had a nice solution to this (I'll second this):
 "I think we want to switch to _tls-tcp going forward and allow
 _tls for backwards compatibility."

-8.1 (page 16): There is (still) no default port for the STUN over
 TLS (shared secret usage). I'd really like this specification to
include
 one. I havent heard of any (good) reasons why not to do so.
 Here's a few reasons why to specify this default port definition also
 for STUN over (TCP) TLS:
 a) This enables (minimal) implementations not to include the DNS SRV
query
 part in the implementation. Also, disabling the DNS SRV query part with
 settings (like by having just a direct STUN server address in settings)
 would be enabled.
 b) The TLS STUN services (if implemented) will be in some port anyway.
 Why not specify a default port so that the service providers could use
 a specific port by default, and not scatter the port selections to
 random ports?
 c) I'm not much of a security specialist, but would't this
specification ...
 "For usages that
  require TLS, such as the short term password usage, lack of SRV
  records is equivalent to a failure of the transaction, since the
  request or indication MUST NOT be sent unless SRV records provided a
  transport address specifically for TLS."
 ... enable a potential down-negotiation weakness if the attacker can
somehow
 block the DNS SRV queries the client makes? By this I mean of course
possible
 fall-back to not using short term passwords, and continuing by assuming
 that the server supports STUN binding query without any shared secrets.

-8.3.2 (page 19): Acceptance of response messages that unexpectedly
include a
 MESSAGE-INTEGRITY attribute. How can a server include a
MESSAGE-INTEGRITY
 attribute in a response if the client did not use one in the request?
 What's the point of accepting such response?

-11.2 & 11.3 (pages 30 & 31): The attributes USERNAME & PASSWORD are not
 backwards compatible with RFC3489 due to the changed length and padding
 requirements. The new way is IMHO better, but shouldn't this be
mentioned
 somewhere in the spec?

-11.6: The error code 300 and usage of an alternate server attribute
 ALTERNATE-SERVER. There are already ways to select alternate servers
 (e.g. trying the next DNS SRV result).
 As stated earlier, this also causes a security threat.
 Also, since this is not much usable for plain STUN use cases, I'd not
 include this at least in this STUN RFC.

-General mumble: TLS implementation: this short term secret usage
requiring
 TLS implementation has obviously been seen as a cumbersome feature, and
thus
 it is AFAIK not (at least yet) generally adopted in STUN
implementations.
 Should we still reconcider options which would be lighter to implement,
 and thus would be easier to get widely adopted?
 The long term credentials is one option, but it has it's own
weaknesses,
 like the two-part message sequence due to the challenges.
 I'm just concerned about the risk that this feature won't be
implemented
 later on either due to the too high benefit/cost ratio.
 I do however like the general idea of having a secure way to do STUN
 without much of settings in either the client nor the server.

--
Tomi Kohonen
Nokia Corporation


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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		: NAT Behavioral Requirements for TCP
	Author(s)	: S. Guha, et al.
	Filename	: draft-ietf-behave-tcp-03.txt
	Pages		: 20
	Date		: 2007-1-5
	
This document defines a set of requirements for NATs that handle TCP
   that would allow many applications, such as peer-to-peer applications
   and on-line games, to work consistently.  Developing NATs that meet
   this set of requirements will greatly increase the likelihood that
   these applications will function properly.

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From behave-bounces@ietf.org Fri Jan 05 18:42:26 2007
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Subject: Re: [BEHAVE] I-D ACTION:draft-ietf-behave-tcp-03.txt
From: Saikat Guha <saikat@cs.cornell.edu>
To: behave <behave@ietf.org>
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On Fri, 2007-01-05 at 15:50 -0500, Internet-Drafts@ietf.org wrote:
> 	Title		: NAT Behavioral Requirements for TCP
> 	Author(s)	: S. Guha, et al.
> 	Filename	: draft-ietf-behave-tcp-03.txt
> 	Pages		: 20
> 	Date		: 2007-1-5
> =09
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-behave-tcp-03.txt

A diff of the changes from -02 is at http://tinyurl.com/ylbcep

The intended status of this draft will be _BCP_; this is not reflected
in the version linked above.

The key change in -03 is a rewording of REQ-4 (6s delayed ICMP) to
preserve the original intent and meaning but to also allow NATs that
always silently drop the SYN due to security concerns (as allowed by the
SHOULD) to stay "fully compliant" with the spec, not just "compliant".
This was discussed at the last meeting.

Other changes include explaining why port-range preservation for TCP is
left unspecified and dropping the MUST in section 7.3 text, both per
WG consensus in Nov.

cheers and a happy new year to all,
--=20
Saikat

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From behave-bounces@ietf.org Wed Jan 10 19:04:05 2007
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Subject: FW: [BEHAVE] STUN FINGERPRINT
Date: Wed, 10 Jan 2007 17:03:33 -0700
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All,
=20
I sent this last week and did not see a response. Given the original
postings close proximity to the new year, I suspect several folks were
on vacation and thus missed it.
=20
Please respond if you have any thoughts on this.
=20
Regards,
Kevin Johns

  _____ =20

From: Kevin Johns=20
Sent: Wednesday, January 03, 2007 2:17 PM
To: behave@ietf.org
Subject: [BEHAVE] STUN FINGERPRINT


All,
=20
Per the text of the current STUN draft (-05), it states that the value
is computed as the CRC-32 of the STUN message up to (but excluding) the
FINGERPRINT attribute itself.
=20
The question is on the length value used in the message length field of
the STUN header. Is the FINGERPRINT calculated using the message length
including the FINGERPRINT attribute or the message length minus the
FINGERPRINT attribute?
=20
I could not find this discussed anywhere in the draft and we need to
know which way to implement.
=20
Regards,
Kevin Johns

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.2963" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D749200100-11012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>All,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D749200100-11012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D749200100-11012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>I sent this last week and did not see a =
response.=20
Given&nbsp;the original postings&nbsp;close proximity to the new year, I =
suspect=20
several folks were on vacation and thus missed it.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D749200100-11012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D749200100-11012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Please respond if you have any thoughts on=20
this.</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D749200100-11012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D749200100-11012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D749200100-11012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Kevin Johns</FONT></SPAN></DIV><BR>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Kevin Johns <BR><B>Sent:</B> =
Wednesday,=20
January 03, 2007 2:17 PM<BR><B>To:</B> =
behave@ietf.org<BR><B>Subject:</B>=20
[BEHAVE] STUN FINGERPRINT<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2>All,</FONT></SPAN></DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial size=3D2>Per =
the text of the=20
current STUN draft (-05), it states that the value is computed as the =
CRC-32 of=20
the STUN message up to (but excluding) the FINGERPRINT attribute=20
itself.</FONT></SPAN></DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial size=3D2>The =
question is on=20
the length value used in the message length field of the STUN header. Is =
the=20
FINGERPRINT calculated using the message length including the =
FINGERPRINT=20
attribute or the message length minus the FINGERPRINT=20
attribute?</FONT></SPAN></DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial size=3D2>I =
could not find=20
this discussed anywhere in the draft and we need to know which way to=20
implement.</FONT></SPAN></DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial=20
size=3D2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D916320821-03012007><FONT face=3DArial size=3D2>Kevin=20
Johns</FONT></SPAN></DIV></BODY></HTML>

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Sorry for not replying Kevin. This is a good question. I think it should 
be including the FINGERPRINT. THis way, when the fingerprint is 
verified, the recipient does not need to modify the message prior to 
verification.

I'll clarify along the with the other gazillion fixes required from 
comments :(

-Jonathan R.

Kevin Johns wrote:

> All,
>  
> I sent this last week and did not see a response. Given the original 
> postings close proximity to the new year, I suspect several folks were 
> on vacation and thus missed it.
>  
> Please respond if you have any thoughts on this.
>  
> Regards,
> Kevin Johns
> 
> ------------------------------------------------------------------------
> *From:* Kevin Johns
> *Sent:* Wednesday, January 03, 2007 2:17 PM
> *To:* behave@ietf.org
> *Subject:* [BEHAVE] STUN FINGERPRINT
> 
> All,
>  
> Per the text of the current STUN draft (-05), it states that the value 
> is computed as the CRC-32 of the STUN message up to (but excluding) the 
> FINGERPRINT attribute itself.
>  
> The question is on the length value used in the message length field of 
> the STUN header. Is the FINGERPRINT calculated using the message length 
> including the FINGERPRINT attribute or the message length minus the 
> FINGERPRINT attribute?
>  
> I could not find this discussed anywhere in the draft and we need to 
> know which way to implement.
>  
> Regards,
> Kevin Johns
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> 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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Subject: RE: FW: [BEHAVE] STUN FINGERPRINT
Date: Thu, 11 Jan 2007 11:15:47 -0700
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From: "Kevin Johns" <K.Johns@CableLabs.com>
To: "Jonathan Rosenberg" <jdrosen@cisco.com>
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Jonathan,

Thanks for the response. I assume the same would go for the Message
Integrity as well?

Kevin=20

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
Sent: Thursday, January 11, 2007 10:57 AM
To: Kevin Johns
Cc: behave@ietf.org
Subject: Re: FW: [BEHAVE] STUN FINGERPRINT

Sorry for not replying Kevin. This is a good question. I think it should
be including the FINGERPRINT. THis way, when the fingerprint is
verified, the recipient does not need to modify the message prior to
verification.

I'll clarify along the with the other gazillion fixes required from
comments :(

-Jonathan R.

Kevin Johns wrote:

> All,
> =20
> I sent this last week and did not see a response. Given the original=20
> postings close proximity to the new year, I suspect several folks were

> on vacation and thus missed it.
> =20
> Please respond if you have any thoughts on this.
> =20
> Regards,
> Kevin Johns
>=20
> ----------------------------------------------------------------------
> --
> *From:* Kevin Johns
> *Sent:* Wednesday, January 03, 2007 2:17 PM
> *To:* behave@ietf.org
> *Subject:* [BEHAVE] STUN FINGERPRINT
>=20
> All,
> =20
> Per the text of the current STUN draft (-05), it states that the value

> is computed as the CRC-32 of the STUN message up to (but excluding)=20
> the FINGERPRINT attribute itself.
> =20
> The question is on the length value used in the message length field=20
> of the STUN header. Is the FINGERPRINT calculated using the message=20
> length including the FINGERPRINT attribute or the message length minus

> the FINGERPRINT attribute?
> =20
> I could not find this discussed anywhere in the draft and we need to=20
> know which way to implement.
> =20
> Regards,
> Kevin Johns
>=20
>=20
> ----------------------------------------------------------------------
> --
>=20
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
>=20
>=20
> ----------------------------------------------------------------------
> --
>=20
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave

--=20
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 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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Yes.

-Jonathan R.

Kevin Johns wrote:

> Jonathan,
> 
> Thanks for the response. I assume the same would go for the Message
> Integrity as well?
> 
> Kevin 
> 
> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
> Sent: Thursday, January 11, 2007 10:57 AM
> To: Kevin Johns
> Cc: behave@ietf.org
> Subject: Re: FW: [BEHAVE] STUN FINGERPRINT
> 
> Sorry for not replying Kevin. This is a good question. I think it should
> be including the FINGERPRINT. THis way, when the fingerprint is
> verified, the recipient does not need to modify the message prior to
> verification.
> 
> I'll clarify along the with the other gazillion fixes required from
> comments :(
> 
> -Jonathan R.
> 
> Kevin Johns wrote:
> 
> 
>>All,
>> 
>>I sent this last week and did not see a response. Given the original 
>>postings close proximity to the new year, I suspect several folks were
> 
> 
>>on vacation and thus missed it.
>> 
>>Please respond if you have any thoughts on this.
>> 
>>Regards,
>>Kevin Johns
>>
>>----------------------------------------------------------------------
>>--
>>*From:* Kevin Johns
>>*Sent:* Wednesday, January 03, 2007 2:17 PM
>>*To:* behave@ietf.org
>>*Subject:* [BEHAVE] STUN FINGERPRINT
>>
>>All,
>> 
>>Per the text of the current STUN draft (-05), it states that the value
> 
> 
>>is computed as the CRC-32 of the STUN message up to (but excluding) 
>>the FINGERPRINT attribute itself.
>> 
>>The question is on the length value used in the message length field 
>>of the STUN header. Is the FINGERPRINT calculated using the message 
>>length including the FINGERPRINT attribute or the message length minus
> 
> 
>>the FINGERPRINT attribute?
>> 
>>I could not find this discussed anywhere in the draft and we need to 
>>know which way to implement.
>> 
>>Regards,
>>Kevin Johns
>>
>>
>>----------------------------------------------------------------------
>>--
>>
>>_______________________________________________
>>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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I am starting to build the agenda for BEHAVE.  If you would like a slot on
the BEHAVE agenda for IETF 68 in Prague, please email me the I-D and
duration you'd like.  We have a two-hour slot.

Thanks,
-d

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We should add the two attributes defined in ICE:

    0x0024 PRIORITY
    0x0025 USE-CANDIDATE

-Jonathan R.

Dan Wing wrote:

> In case this helps anyone else, I built a table showing all of the STUN
> attributes and where they're defined,
> http://www.employees.org/behave/stun-attributes.html.  As IANA will only
> maintain a registry after an RFC is published, we will probably maintain
> this page to coordinate STUN attribute values amongst Internet Drafts.
> 
> -d
> 
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
> 

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

In San Diego, Jonathan presented our
draft-wing-behave-nat-control-stun-usage document,
<http://www3.ietf.org/proceedings/06nov/slides/behave-5/sld1.htm>, 
<http://tools.ietf.org/html/draft-wing-behave-nat-control-stun-usage-00>:
   Simple Traversal Underneath NAT (STUN) is a mechanism for traversing
   NATs.  STUN requests are transmitted through a NAT to external STUN
   servers.  While this works very well, its two primary drawbacks are
   the inability to modify the properties of a NAT binding and the need
   to query a public STUN server for every NAT binding.  These drawbacks
   require frequent messages which present a load on servers (like SIP
   servers and STUN servers) and are bad for low speed access networks,
   such as cellular.  This document proposes that the STUN server be
   embedded in the NAT itself, and describes how these STUN servers can
   be readily discovered and utilized to reduce queries to public STUN
   servers and to reduce NAT keepalive traffic.

As Jonathan indicated at the meeting, Cisco claims IPR on this technique.
Terms are here:
http://www.ietf.org/ietf/IPR/cisco-ipr-draft-wing-behave-nat-control-stun-us
age-00.txt


At the San Diego meeting there were positive comments around this approach.

Is there interest in moving this work forward?

-d

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To: "'Jonathan Rosenberg'" <jdrosen@cisco.com>
Subject: RE: [BEHAVE] STUN attributes
Date: Fri, 12 Jan 2007 11:31:07 -0800
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Added.  Thanks.
-d
 

> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
> Sent: Friday, January 12, 2007 9:37 AM
> To: Dan Wing
> Cc: behave@ietf.org
> Subject: Re: [BEHAVE] STUN attributes
> 
> We should add the two attributes defined in ICE:
> 
>     0x0024 PRIORITY
>     0x0025 USE-CANDIDATE
> 
> -Jonathan R.
> 
> Dan Wing wrote:
> 
> > In case this helps anyone else, I built a table showing all 
> of the STUN
> > attributes and where they're defined,
> > http://www.employees.org/behave/stun-attributes.html.  As 
> IANA will only
> > maintain a registry after an RFC is published, we will 
> probably maintain
> > this page to coordinate STUN attribute values amongst 
> Internet Drafts.
> > 
> > -d
> > 
> > _______________________________________________
> > 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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> In San Diego, Jonathan presented our
> draft-wing-behave-nat-control-stun-usage document,
> <http://www3.ietf.org/proceedings/06nov/slides/behave-5/sld1.htm>,
> <http://tools.ietf.org/html/draft-wing-behave-nat-control-stun-usage-
> 00>:
>    Simple Traversal Underneath NAT (STUN) is a mechanism for
traversing
>    NATs.  STUN requests are transmitted through a NAT to external STUN
>    servers.  While this works very well, its two primary drawbacks are
>    the inability to modify the properties of a NAT binding and the
need
>    to query a public STUN server for every NAT binding.  These
> drawbacks
>    require frequent messages which present a load on servers (like SIP
>    servers and STUN servers) and are bad for low speed access
networks,
>    such as cellular.  This document proposes that the STUN server be
>    embedded in the NAT itself, and describes how these STUN servers
can
>    be readily discovered and utilized to reduce queries to public STUN
>    servers and to reduce NAT keepalive traffic.
>=20
> As Jonathan indicated at the meeting, Cisco claims IPR on this
> technique.
> Terms are here:
> http://www.ietf.org/ietf/IPR/cisco-ipr-draft-wing-behave-nat-control-
> stun-us
> age-00.txt
>=20
>=20
> At the San Diego meeting there were positive comments around this
> approach.
>=20
> Is there interest in moving this work forward?

If you want to get mapping control between the client and the NAT, why
not just use UPNP? The protocol has been used since 2001, there are many
implementations...

-- Christian Huitema

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I noticed in Section 15.2:

    0x8023: FINGERPRINT
    0x8023: ALTERNATE-SERVER

Are these two attributes supposed to have the same value?  My apologies if I overlooked something...

Thanks,
FM



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From: "Dan Wing" <dwing@cisco.com>
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Subject: RE: [BEHAVE] NAT Control STUN Usage
Date: Sun, 14 Jan 2007 18:29:07 -0800
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> If you want to get mapping control between the client and the NAT, 
> why not just use UPNP? The protocol has been used since 2001, 
> there are many implementations...

* UPnP Doesn't work with nested NATs (a NAT in my apartment (or dorm room)
and another in the basement of my apartment/dorm building; or two NATs at
home),

* UPnP is said to be slow
<http://www.voipsa.org/pipermail/voipsec_voipsa.org/2006-June/001832.html>,

* some NAT vendors don't implement UPnP,

* UPnP doesn't work in a routed network.  Such as an enterprise network.

* To work in the above situations, endpoints need to implement STUN anyway.
Designing an endpoint that only works in UPnP environments restricts the
endpoint to only ever working in the presence of UPnP.

-d

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> I noticed in Section 15.2:
> 
>     0x8023: FINGERPRINT
>     0x8023: ALTERNATE-SERVER
> 
> Are these two attributes supposed to have the same value?  My 
> apologies if I overlooked something...

No, they're not supposed to have the same value.  The double
assignment problem was mentioned at
http://www.employees.org/behave/stun-attributes.html but
it's also interesting how ALTERNATE-SERVERs values have evolved:

  grep -i "0x.*alternate-server" draft*turn*.txt
  draft-rosenberg-midcom-turn-00.txt:   0x0007: ALTERNATE-SERVER
  draft-rosenberg-midcom-turn-01.txt:   0x000e: ALTERNATE-SERVER
  draft-rosenberg-midcom-turn-02.txt:   0x000e: ALTERNATE-SERVER
  draft-rosenberg-midcom-turn-03.txt:   0x000e: ALTERNATE-SERVER
  draft-rosenberg-midcom-turn-04.txt:   0x000e: ALTERNATE-SERVER
  draft-rosenberg-midcom-turn-05.txt:   0x000e: ALTERNATE-SERVER
  draft-rosenberg-midcom-turn-06.txt:   0x000e: ALTERNATE-SERVER
  draft-rosenberg-midcom-turn-07.txt:   0x000e: ALTERNATE-SERVER
  draft-rosenberg-midcom-turn-08.txt:   0x000e: ALTERNATE-SERVER

  grep -i "0x.*alternate-server" draft-ietf-behave-rfc3489bis*.txt
  draft-ietf-behave-rfc3489bis-03.txt:        0x000E: ALTERNATE-SERVER
  draft-ietf-behave-rfc3489bis-03.txt:        0x8023: ALTERNATE-SERVER
  draft-ietf-behave-rfc3489bis-04.txt:    0x8023: ALTERNATE-SERVER
  draft-ietf-behave-rfc3489bis-05.txt:    0x8023: ALTERNATE-SERVER

I'm not sure if there are implementations that rely on those values.

-d

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To keep this topic alive, I plan to update the following individual draft:
http://tools.ietf.org/wg/behave/draft-marjou-behave-app-rtp-keepalive-00.tx=
t

I am therefore interested of any feedback, especially on which
mechanism(s) is best.

Xavier

> On 12/3/06, Dan Wing <dwing@cisco.com> wrote:
> Minutes of the last BEHAVE meeting have been posted to:
> http://www3.ietf.org/proceedings/06nov/minutes/behave.html
>
> Application Keepalive for NATs, Xavier Marjou -
> draft-marjou-behave-app-rtp-keepalive-00
> status:  separate from ICE, or integrate into Application milestone docum=
ent?
>
> JR =96 this is app design guidelines. How about if we charter to create
> that document and put this text, along with other text around it in
> STUN all in one document?
> Xavier Marjou (speaker, XM) =96 take to list.
> Chair =96 question of which mechanisms might be best asked in AVT FA =96
> good to ask AVT for sure. But we should discuss here Magnus =96 both
> groups working on it Chair =96 consensus to fold this into application
> document, per JR's proposal.
> Will figure out what to do on list.
> FA =96 support. Think it's a good idea. But do we have an editor for
> that, will it get done? I don't want to send this to dev/null.
> PM =96 in our BEHAVE app document are we the best place to have that
> discussion?
> chair - don't have a clear decision on what to do, will work to figure
> out where this goes.

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=20
One of the properties of STUN relay is that the STUN relay server
handles both STUN messages and media on the same IP and Port. This makes
policy control a bit difficult as the policy enforcement point has to be
able to identify STUN messages over Media in order to enforce policy.
Has there been any discussion around having a management only IP/Port
and as part of the allocation process allocate a local transport address
(the address and port the UA requesting the allocation sends and
receives from) in addition to the currently allocated relayed transport
address?
=20
Such and approach would facilitate policy management as STUN Relay (and
STUN for that matter) can be classified separately from the media
traffic allowing for separate policy enforcement.
=20
Regards,
Kevin

------_=_NextPart_001_01C738C4.F8DC9FD1
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
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charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.2963" name=3DGENERATOR></HEAD>
<BODY>
<DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D736143616-15012007><FONT face=3DArial size=3D2>One of =
the=20
properties of STUN relay is that the STUN relay server handles both STUN =

messages and media on the same IP and Port. This makes policy control a =
bit=20
difficult as the policy enforcement point has to be able to identify =
STUN=20
messages over Media in order to enforce policy. Has there been any =
discussion=20
around having a management only IP/Port and as part of the allocation =
process=20
allocate a local transport address (the address and port the UA =
requesting the=20
allocation sends and receives from) in addition to the currently =
allocated=20
relayed transport address?</FONT></SPAN></DIV>
<DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D736143616-15012007><FONT face=3DArial size=3D2>Such =
and approach=20
would facilitate policy management as STUN Relay (and STUN for that =
matter) can=20
be classified separately from the media traffic allowing for separate =
policy=20
enforcement.</FONT></SPAN></DIV>
<DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
size=3D2>Regards,</FONT></SPAN></DIV>
<DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
size=3D2>Kevin</FONT></SPAN></DIV></BODY></HTML>

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Subject: RE: [BEHAVE] STUN Relay
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What sort of policy management separation are you looking for? I am
assuming that you are talking about gating behavior at the relay?
=20
Regards,
Brian


________________________________

	From: Kevin Johns [mailto:K.Johns@CableLabs.com]=20
	Sent: Monday, January 15, 2007 10:48 AM
	To: behave@ietf.org
	Subject: [BEHAVE] STUN Relay
=09
=09
	=20
	One of the properties of STUN relay is that the STUN relay
server handles both STUN messages and media on the same IP and Port.
This makes policy control a bit difficult as the policy enforcement
point has to be able to identify STUN messages over Media in order to
enforce policy. Has there been any discussion around having a management
only IP/Port and as part of the allocation process allocate a local
transport address (the address and port the UA requesting the allocation
sends and receives from) in addition to the currently allocated relayed
transport address?
	=20
	Such and approach would facilitate policy management as STUN
Relay (and STUN for that matter) can be classified separately from the
media traffic allowing for separate policy enforcement.
	=20
	Regards,
	Kevin


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charset=3Dus-ascii">
<META content=3D"MSHTML 6.00.2900.3020" name=3DGENERATOR></HEAD>
<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D287225316-15012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>What sort of policy management separation are =
you looking=20
for? I am assuming that you are talking about gating behavior at the=20
relay?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D287225316-15012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D287225316-15012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D287225316-15012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Brian</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Kevin Johns=20
  [mailto:K.Johns@CableLabs.com] <BR><B>Sent:</B> Monday, January 15, =
2007 10:48=20
  AM<BR><B>To:</B> behave@ietf.org<BR><B>Subject:</B> [BEHAVE] STUN=20
  Relay<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial size=3D2>One =
of the=20
  properties of STUN relay is that the STUN relay server handles both =
STUN=20
  messages and media on the same IP and Port. This makes policy control =
a bit=20
  difficult as the policy enforcement point has to be able to identify =
STUN=20
  messages over Media in order to enforce policy. Has there been any =
discussion=20
  around having a management only IP/Port and as part of the allocation =
process=20
  allocate a local transport address (the address and port the UA =
requesting the=20
  allocation sends and receives from) in addition to the currently =
allocated=20
  relayed transport address?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial size=3D2>Such =
and approach=20
  would facilitate policy management as STUN Relay (and STUN for that =
matter)=20
  can be classified separately from the media traffic allowing for =
separate=20
  policy enforcement.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
  size=3D2>Regards,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
  size=3D2>Kevin</FONT></SPAN></DIV></BLOCKQUOTE></BODY></HTML>

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From behave-bounces@ietf.org Mon Jan 15 12:26:37 2007
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Brian,
=20
The concern is this, a UA needs to send a STUN message to a STUN Relay
server to get an allocated address to put in its SDP. To allow this, the
policy enforcement point (CMTS, GGSN, AP, etc.) needs to have a policy
which allows the UA to send and receive messages from the STUN Relay.
Given the UA needs to communicate with the STUN Relay prior to any
session establishment, the policy needs to be static and apply to all
UAs. Now you have effectively installed a policy which allows UAs to
communicate freely via this static default policy. Now, this may not be
a big issue if the network installs a dynamic policy once the session is
established and the dynamic policy takes precedence over the static.
However, if the network does not install a dynamic policy, the UA still
has a free path to communicate with another UA.
=20
So it would be nice to be able to install a default policy which only
allows for STUN management traffic. Thus, if a UA requests a relay
address allocation the relay would happen on a different address/port
and thus block the UAs ability to relay media without a policy.
=20
Kevin

  _____ =20

From: Brian Stucker [mailto:bstucker@nortel.com]=20
Sent: Monday, January 15, 2007 9:54 AM
To: Kevin Johns; behave@ietf.org
Subject: RE: [BEHAVE] STUN Relay


What sort of policy management separation are you looking for? I am
assuming that you are talking about gating behavior at the relay?
=20
Regards,
Brian


  _____ =20

	From: Kevin Johns [mailto:K.Johns@CableLabs.com]=20
	Sent: Monday, January 15, 2007 10:48 AM
	To: behave@ietf.org
	Subject: [BEHAVE] STUN Relay
=09
=09
	=20
	One of the properties of STUN relay is that the STUN relay
server handles both STUN messages and media on the same IP and Port.
This makes policy control a bit difficult as the policy enforcement
point has to be able to identify STUN messages over Media in order to
enforce policy. Has there been any discussion around having a management
only IP/Port and as part of the allocation process allocate a local
transport address (the address and port the UA requesting the allocation
sends and receives from) in addition to the currently allocated relayed
transport address?
	=20
	Such and approach would facilitate policy management as STUN
Relay (and STUN for that matter) can be classified separately from the
media traffic allowing for separate policy enforcement.
	=20
	Regards,
	Kevin


------_=_NextPart_001_01C738CA.46A59D51
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<BODY>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D634371517-15012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Brian,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D634371517-15012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D634371517-15012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>The concern is this, a UA needs to send a STUN =
message to a=20
STUN Relay server to get an allocated address to put in its SDP. To =
allow this,=20
the policy enforcement point (CMTS, GGSN, AP, etc.) needs to have a =
policy which=20
allows the UA to send and receive messages from the STUN Relay. Given =
the UA=20
needs to communicate with the STUN Relay prior to any session =
establishment, the=20
policy needs to be static and apply to all UAs. Now you have effectively =

installed a policy which allows UAs to communicate freely via this =
static=20
default policy. Now, this may not be a big issue if the network installs =
a=20
dynamic policy once the session is established and the dynamic policy =
takes=20
precedence over the static. However, if the network does not install a =
dynamic=20
policy, the UA still has a free path to communicate with another=20
UA.</FONT></SPAN></DIV>
<DIV><SPAN class=3D634371517-15012007><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D634371517-15012007><FONT face=3DArial color=3D#0000ff =
size=3D2>So it=20
would be nice to be able to install a default policy which only allows =
for STUN=20
management traffic. Thus, if a UA requests a relay address allocation =
the relay=20
would happen on a different address/port and thus block the UAs ability =
to relay=20
media without a policy.</FONT></SPAN></DIV>
<DIV><SPAN class=3D634371517-15012007><FONT face=3DArial color=3D#0000ff =

size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV><SPAN class=3D634371517-15012007><FONT face=3DArial color=3D#0000ff =

size=3D2>Kevin</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><BR></DIV>
<DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
<HR tabIndex=3D-1>
<FONT face=3DTahoma size=3D2><B>From:</B> Brian Stucker =
[mailto:bstucker@nortel.com]=20
<BR><B>Sent:</B> Monday, January 15, 2007 9:54 AM<BR><B>To:</B> Kevin =
Johns;=20
behave@ietf.org<BR><B>Subject:</B> RE: [BEHAVE] STUN =
Relay<BR></FONT><BR></DIV>
<DIV></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D287225316-15012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>What sort of policy management separation are =
you looking=20
for? I am assuming that you are talking about gating behavior at the=20
relay?</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D287225316-15012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2></FONT></SPAN>&nbsp;</DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D287225316-15012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Regards,</FONT></SPAN></DIV>
<DIV dir=3Dltr align=3Dleft><SPAN class=3D287225316-15012007><FONT =
face=3DArial=20
color=3D#0000ff size=3D2>Brian</FONT></SPAN></DIV><BR>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px =
solid; MARGIN-RIGHT: 0px">
  <DIV class=3DOutlookMessageHeader lang=3Den-us dir=3Dltr align=3Dleft>
  <HR tabIndex=3D-1>
  <FONT face=3DTahoma size=3D2><B>From:</B> Kevin Johns=20
  [mailto:K.Johns@CableLabs.com] <BR><B>Sent:</B> Monday, January 15, =
2007 10:48=20
  AM<BR><B>To:</B> behave@ietf.org<BR><B>Subject:</B> [BEHAVE] STUN=20
  Relay<BR></FONT><BR></DIV>
  <DIV></DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial size=3D2>One =
of the=20
  properties of STUN relay is that the STUN relay server handles both =
STUN=20
  messages and media on the same IP and Port. This makes policy control =
a bit=20
  difficult as the policy enforcement point has to be able to identify =
STUN=20
  messages over Media in order to enforce policy. Has there been any =
discussion=20
  around having a management only IP/Port and as part of the allocation =
process=20
  allocate a local transport address (the address and port the UA =
requesting the=20
  allocation sends and receives from) in addition to the currently =
allocated=20
  relayed transport address?</FONT></SPAN></DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial size=3D2>Such =
and approach=20
  would facilitate policy management as STUN Relay (and STUN for that =
matter)=20
  can be classified separately from the media traffic allowing for =
separate=20
  policy enforcement.</FONT></SPAN></DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
  size=3D2></FONT></SPAN>&nbsp;</DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
  size=3D2>Regards,</FONT></SPAN></DIV>
  <DIV><SPAN class=3D736143616-15012007><FONT face=3DArial=20
  size=3D2>Kevin</FONT></SPAN></DIV></BLOCKQUOTE></BODY></HTML>

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From behave-bounces@ietf.org Mon Jan 15 14:33:21 2007
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From: Philip Matthews <philip_matthews@magma.ca>
Subject: Re: [BEHAVE] NAT Control STUN Usage
Date: Mon, 15 Jan 2007 14:33:17 -0500
To: Dan Wing <dwing@cisco.com>
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I would like to see BEHAVE take on work in this area.
I think that a solution to this problem is of interest.

May I suggest that the work begin by writing a problem statement,
so that we can agree on exactly what we are trying to achieve?
This could be just a section or two at the beginning of the existing
document (i.e., I am NOT suggesting a separate requirements
document).

- Philip

On 12-Jan-07, at 13:09 , Dan Wing wrote:

> (as individual contributor:)
>
> In San Diego, Jonathan presented our
> draft-wing-behave-nat-control-stun-usage document,
> <http://www3.ietf.org/proceedings/06nov/slides/behave-5/sld1.htm>,
> <http://tools.ietf.org/html/draft-wing-behave-nat-control-stun- 
> usage-00>:
>    Simple Traversal Underneath NAT (STUN) is a mechanism for  
> traversing
>    NATs.  STUN requests are transmitted through a NAT to external STUN
>    servers.  While this works very well, its two primary drawbacks are
>    the inability to modify the properties of a NAT binding and the  
> need
>    to query a public STUN server for every NAT binding.  These  
> drawbacks
>    require frequent messages which present a load on servers (like SIP
>    servers and STUN servers) and are bad for low speed access  
> networks,
>    such as cellular.  This document proposes that the STUN server be
>    embedded in the NAT itself, and describes how these STUN servers  
> can
>    be readily discovered and utilized to reduce queries to public STUN
>    servers and to reduce NAT keepalive traffic.
>
> As Jonathan indicated at the meeting, Cisco claims IPR on this  
> technique.
> Terms are here:
> http://www.ietf.org/ietf/IPR/cisco-ipr-draft-wing-behave-nat- 
> control-stun-us
> age-00.txt
>
>
> At the San Diego meeting there were positive comments around this  
> approach.
>
> Is there interest in moving this work forward?
>
> -d
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave


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From: "Neil Deason" <neildeason@gmail.com>
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Hello Folks.

The use of long term credentials to authenticate STUN requests within relay
sessions opens a number of replay type attacks. For example:

- DoS on server by using up ports through replaying Allocate requests
- DoS on clients by replaying Send requests
- Session hijacking by replaying SetActiveDestination requests

Possible options to address this:

a) Add a sequence number in STUN messages

This could be a new extension attribute or possibly the transaction ID
could be used. The transaction ID could be used by reserving the last 4
bytes to be sequence number or alternatively within a relay session only the
first Allocate request uses a randomly chosen transaction ID and then
subsequent requests in the session increment it.

b) Add a nonce count for STUN messages authenticated with long term creds

Add an nonce-count attribute to record the number of times a nonce has been
used within requests. Client will need to include this count in every
request that carries the nonce, i.e. the first request sent carries a
nonce-count of 1, the next 2 etc. Servers then need to maintain their own
copy of this count and if the same value is seen twice it is a potential
replay.

------=_Part_93323_18975244.1168900084883
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<div>Hello Folks.</div>
<div>&nbsp;</div>
<div>The use of long term credentials to authenticate STUN requests within relay sessions opens a number of replay type attacks. For example:</div>
<div>&nbsp;</div>
<div>- DoS on server by using up ports through replaying Allocate requests </div>
<div>- DoS on clients by replaying Send requests </div>
<div>- Session hijacking by replaying SetActiveDestination requests</div>
<div>&nbsp;</div>
<div>Possible options to address this:</div>
<div>&nbsp;</div>
<div>a)&nbsp;Add a&nbsp;sequence number&nbsp;in STUN messages</div>
<div>&nbsp;</div>
<div>This could be a new extension attribute or possibly the transaction ID could&nbsp;be used. The transaction ID&nbsp;could be used by reserving the last 4 bytes to be sequence number or alternatively within a relay session only the first Allocate request uses a randomly chosen transaction ID and then subsequent requests in the session increment it.
</div>
<div>&nbsp;</div>
<div>b)&nbsp;Add a nonce count for STUN messages authenticated with long term creds</div>
<div>&nbsp;</div>
<div>Add an nonce-count attribute to record the number of times a nonce has been used within requests. Client will need to include this count in every request that carries the nonce, i.e. the first request sent carries a nonce-count of 1, the next 2 etc. Servers then need to maintain their own copy of this count and if the same value is seen twice it is a potential replay.
</div>

------=_Part_93323_18975244.1168900084883--


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From behave-bounces@ietf.org Tue Jan 16 00:02:37 2007
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Is the magic cookie field treated as a 32-bit value, i.e. do you apply
htonl() on it before sending or is it a byte array like the transaction
ID?

Thanks,
FM



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From behave-bounces@ietf.org Tue Jan 16 01:27:51 2007
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From: "Dan Wing" <dwing@cisco.com>
To: "'Kevin Johns'" <K.Johns@CableLabs.com>,
	"'Brian Stucker'" <bstucker@nortel.com>, <behave@ietf.org>
Subject: RE: [BEHAVE] STUN Relay
Date: Mon, 15 Jan 2007 22:27:34 -0800
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You need a bit more than STUN management traffic -- the endpoint needs to
also be able to send ICE connectivity checks through the TURN server.
Otherwise, the endpoint will never know that the path through the TURN
server is a viable path.

Implementing additional functionality in the TURN server, akin to what I
described in draft-wing-session-auth-00, should provide the hooks necessary
to block or detect attempts to use a TURN server without first signaling via
SIP.

-d


> -----Original Message-----
> From: Kevin Johns [mailto:K.Johns@CableLabs.com] 
> Sent: Monday, January 15, 2007 9:26 AM
> To: Brian Stucker; behave@ietf.org
> Subject: RE: [BEHAVE] STUN Relay
> 
> Brian,
>  
> The concern is this, a UA needs to send a STUN message to a 
> STUN Relay server to get an allocated address to put in its 
> SDP. To allow this, the policy enforcement point (CMTS, GGSN, 
> AP, etc.) needs to have a policy which allows the UA to send 
> and receive messages from the STUN Relay. Given the UA needs 
> to communicate with the STUN Relay prior to any session 
> establishment, the policy needs to be static and apply to all 
> UAs. Now you have effectively installed a policy which allows 
> UAs to communicate freely via this static default policy. 
> Now, this may not be a big issue if the network installs a 
> dynamic policy once the session is established and the 
> dynamic policy takes precedence over the static. However, if 
> the network does not install a dynamic policy, the UA still 
> has a free path to communicate with another UA.
>  
> So it would be nice to be able to install a default policy 
> which only allows for STUN management traffic. Thus, if a UA 
> requests a relay address allocation the relay would happen on 
> a different address/port and thus block the UAs ability to 
> relay media without a policy.
>  
> Kevin
> 
> ________________________________
> 
> From: Brian Stucker [mailto:bstucker@nortel.com] 
> Sent: Monday, January 15, 2007 9:54 AM
> To: Kevin Johns; behave@ietf.org
> Subject: RE: [BEHAVE] STUN Relay
> 
> 
> What sort of policy management separation are you looking 
> for? I am assuming that you are talking about gating behavior 
> at the relay?
>  
> Regards,
> Brian
> 
> 
> ________________________________
> 
> 	From: Kevin Johns [mailto:K.Johns@CableLabs.com] 
> 	Sent: Monday, January 15, 2007 10:48 AM
> 	To: behave@ietf.org
> 	Subject: [BEHAVE] STUN Relay
> 	
> 	
> 	 
> 	One of the properties of STUN relay is that the STUN 
> relay server handles both STUN messages and media on the same 
> IP and Port. This makes policy control a bit difficult as the 
> policy enforcement point has to be able to identify STUN 
> messages over Media in order to enforce policy. Has there 
> been any discussion around having a management only IP/Port 
> and as part of the allocation process allocate a local 
> transport address (the address and port the UA requesting the 
> allocation sends and receives from) in addition to the 
> currently allocated relayed transport address?
> 	 
> 	Such and approach would facilitate policy management as 
> STUN Relay (and STUN for that matter) can be classified 
> separately from the media traffic allowing for separate 
> policy enforcement.
> 	 
> 	Regards,
> 	Kevin
> 
> 

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Le Tuesday 16 January 2007 07:01, ext Frank W. Miller a =C3=A9crit=C2=A0:
> Is the magic cookie field treated as a 32-bit value, i.e. do you apply
> htonl() on it before sending or is it a byte array like the transaction
> ID?

How you read/write the cookie is an implementation detail. Of course, the=20
cookie value in the specification is in network byte order.
=2D-=20
R=C3=A9mi Denis-Courmont <Remi.Denis-Courmont@nokia.com>
Research Engineer

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Subject: [BEHAVE] Re: [AVT] Update of rtp-keepalive draft
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I'd vote for 4.1 first, followed by 4.5 (which sounds a lot like 4.7 to
me only it's defined).

reasons:

sending a 0-byte udp packet is solving a network issue, not an RTP issue
and shouldn't pretend to be RTP.

the cons specified:
 - udp specific, well yes but this isn't an issue with tcp anyway.
 - rx-ing party might not handle this. Perhaps, but since these are
intended to be used for NAT traversal I would assume the rx-ing party
would be most probably aware since the address:port of the received udp
needs to be communicated to the tx-ing party anyway, and they generally
need to share the same socket(or at least appear to).

Otherwise I'd pick an unused RTP payload type value and define it as
reserved for NAT keepalive usage. I guess this most closely maps to 4.5.

Regards,

Lee Dilkie

(we implemented the 0-byte udp in our solution, FWIW, although we choose
30 second intervals).

Xavier Marjou wrote:
> To keep this topic alive, I plan to update the following individual
> draft:
> http://tools.ietf.org/wg/behave/draft-marjou-behave-app-rtp-keepalive-0=
0.txt
>
>
> I am therefore interested of any feedback, especially on which
> mechanism(s) is best.
>
> Xavier
>
>> On 12/3/06, Dan Wing <dwing@cisco.com> wrote:
>> Minutes of the last BEHAVE meeting have been posted to:
>> http://www3.ietf.org/proceedings/06nov/minutes/behave.html
>>
>> Application Keepalive for NATs, Xavier Marjou -
>> draft-marjou-behave-app-rtp-keepalive-00
>> status:  separate from ICE, or integrate into Application milestone
>> document?
>>
>> JR =96 this is app design guidelines. How about if we charter to creat=
e
>> that document and put this text, along with other text around it in
>> STUN all in one document?
>> Xavier Marjou (speaker, XM) =96 take to list.
>> Chair =96 question of which mechanisms might be best asked in AVT FA =96
>> good to ask AVT for sure. But we should discuss here Magnus =96 both
>> groups working on it Chair =96 consensus to fold this into application
>> document, per JR's proposal.
>> Will figure out what to do on list.
>> FA =96 support. Think it's a good idea. But do we have an editor for
>> that, will it get done? I don't want to send this to dev/null.
>> PM =96 in our BEHAVE app document are we the best place to have that
>> discussion?
>> chair - don't have a clear decision on what to do, will work to figure
>> out where this goes.
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt

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From behave-bounces@ietf.org Tue Jan 16 11:59:12 2007
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From: Colin Perkins <csp@csperkins.org>
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Subject: [BEHAVE] Re: [AVT] Update of rtp-keepalive draft
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Do we need to mandate an alternative? All of the alternatives =20
specified will work, and RTP implementations have to be robust to =20
receiving all seven types of packet anyway (otherwise, there is a =20
trivial denial-of-service attack on the application). As long as =20
something is sent at an appropriate interval, the NAT will be happy, =20
and the application can use whatever mechanism is appropriate for it.

Colin





On 15 Jan 2007, at 19:04, Lee Dilkie wrote:
> I'd vote for 4.1 first, followed by 4.5 (which sounds a lot like =20
> 4.7 to
> me only it's defined).
>
> reasons:
>
> sending a 0-byte udp packet is solving a network issue, not an RTP =20
> issue
> and shouldn't pretend to be RTP.
>
> the cons specified:
>  - udp specific, well yes but this isn't an issue with tcp anyway.
>  - rx-ing party might not handle this. Perhaps, but since these are
> intended to be used for NAT traversal I would assume the rx-ing party
> would be most probably aware since the address:port of the received =20=

> udp
> needs to be communicated to the tx-ing party anyway, and they =20
> generally
> need to share the same socket(or at least appear to).
>
> Otherwise I'd pick an unused RTP payload type value and define it as
> reserved for NAT keepalive usage. I guess this most closely maps to =20=

> 4.5.
>
> Regards,
>
> Lee Dilkie
>
> (we implemented the 0-byte udp in our solution, FWIW, although we =20
> choose
> 30 second intervals).
>
> Xavier Marjou wrote:
>> To keep this topic alive, I plan to update the following individual
>> draft:
>> http://tools.ietf.org/wg/behave/draft-marjou-behave-app-rtp-=20
>> keepalive-00.txt
>>
>>
>> I am therefore interested of any feedback, especially on which
>> mechanism(s) is best.
>>
>> Xavier
>>
>>> On 12/3/06, Dan Wing <dwing@cisco.com> wrote:
>>> Minutes of the last BEHAVE meeting have been posted to:
>>> http://www3.ietf.org/proceedings/06nov/minutes/behave.html
>>>
>>> Application Keepalive for NATs, Xavier Marjou -
>>> draft-marjou-behave-app-rtp-keepalive-00
>>> status:  separate from ICE, or integrate into Application milestone
>>> document?
>>>
>>> JR =96 this is app design guidelines. How about if we charter to =20
>>> create
>>> that document and put this text, along with other text around it in
>>> STUN all in one document?
>>> Xavier Marjou (speaker, XM) =96 take to list.
>>> Chair =96 question of which mechanisms might be best asked in AVT FA =
=96
>>> good to ask AVT for sure. But we should discuss here Magnus =96 both
>>> groups working on it Chair =96 consensus to fold this into =
application
>>> document, per JR's proposal.
>>> Will figure out what to do on list.
>>> FA =96 support. Think it's a good idea. But do we have an editor for
>>> that, will it get done? I don't want to send this to dev/null.
>>> PM =96 in our BEHAVE app document are we the best place to have that
>>> discussion?
>>> chair - don't have a clear decision on what to do, will work to =20
>>> figure
>>> out where this goes.
>>
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www1.ietf.org/mailman/listinfo/avt
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt

--=20
Colin Perkins
http://csperkins.org/




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For STUN client to determine the NAT type we may have to run it on the port
over which the needy application is already running.
  How can we achieve this functionality as now we have to run stun client
using the same port  and same protocol as our application is using?

Regards,
Vikas

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<br>
For STUN client to determine the NAT type we may have to run it on the port over which the needy application is already running.<br>
&nbsp; How can we achieve this functionality as now we have to run stun client using the same port&nbsp; and same protocol as our application is using?<br>
<br>
Regards,<br>
Vikas 

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From behave-bounces@ietf.org Tue Jan 16 12:35:05 2007
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Dan,

I took a quick look at your draft. The main drawback I see is that it
requires the firewall to implement it. Given the vast number of consumer
NAT/Firewall devices this seems to have limited applicability.

Having not thought through all the details, I was expecting that the UA
could send a STUN allocate request and upon getting a response, send a
binding request to the IP/Port assigned to the UA to associate the STUN
Relay with the UAs address and to open the NAT/Firewall binding. This
would then allow the ICE connectivity checks.

Alternatively, the UA could wait till it gets the SDP answer with the
remote UAs address info then send the connectivity checks which would
open the NAT bindings as well. The drawback to this is that the remote
UA might start connectivity checks as soon as they send the answer.
These would fail (no response) as the local UAs NAT will not have any
bindings to allow the connectivity checks through (although this is
currently the case and the receipt of the connectivity check cases a
trigger check to cut down on the delay of waiting for a timer to fire).

Kevin

-----Original Message-----
From: Dan Wing [mailto:dwing@cisco.com]=20
Sent: Monday, January 15, 2007 11:28 PM
To: Kevin Johns; 'Brian Stucker'; behave@ietf.org
Subject: RE: [BEHAVE] STUN Relay

You need a bit more than STUN management traffic -- the endpoint needs
to also be able to send ICE connectivity checks through the TURN server.
Otherwise, the endpoint will never know that the path through the TURN
server is a viable path.

Implementing additional functionality in the TURN server, akin to what I
described in draft-wing-session-auth-00, should provide the hooks
necessary to block or detect attempts to use a TURN server without first
signaling via SIP.

-d


> -----Original Message-----
> From: Kevin Johns [mailto:K.Johns@CableLabs.com]
> Sent: Monday, January 15, 2007 9:26 AM
> To: Brian Stucker; behave@ietf.org
> Subject: RE: [BEHAVE] STUN Relay
>=20
> Brian,
> =20
> The concern is this, a UA needs to send a STUN message to a STUN Relay

> server to get an allocated address to put in its SDP. To allow this,=20
> the policy enforcement point (CMTS, GGSN, AP, etc.) needs to have a=20
> policy which allows the UA to send and receive messages from the STUN=20
> Relay. Given the UA needs to communicate with the STUN Relay prior to=20
> any session establishment, the policy needs to be static and apply to=20
> all UAs. Now you have effectively installed a policy which allows UAs=20
> to communicate freely via this static default policy.
> Now, this may not be a big issue if the network installs a dynamic=20
> policy once the session is established and the dynamic policy takes=20
> precedence over the static. However, if the network does not install a

> dynamic policy, the UA still has a free path to communicate with=20
> another UA.
> =20
> So it would be nice to be able to install a default policy which only=20
> allows for STUN management traffic. Thus, if a UA requests a relay=20
> address allocation the relay would happen on a different address/port=20
> and thus block the UAs ability to relay media without a policy.
> =20
> Kevin
>=20
> ________________________________
>=20
> From: Brian Stucker [mailto:bstucker@nortel.com]
> Sent: Monday, January 15, 2007 9:54 AM
> To: Kevin Johns; behave@ietf.org
> Subject: RE: [BEHAVE] STUN Relay
>=20
>=20
> What sort of policy management separation are you looking for? I am=20
> assuming that you are talking about gating behavior at the relay?
> =20
> Regards,
> Brian
>=20
>=20
> ________________________________
>=20
> 	From: Kevin Johns [mailto:K.Johns@CableLabs.com]=20
> 	Sent: Monday, January 15, 2007 10:48 AM
> 	To: behave@ietf.org
> 	Subject: [BEHAVE] STUN Relay
> =09
> =09
> 	=20
> 	One of the properties of STUN relay is that the STUN relay
server=20
> handles both STUN messages and media on the same IP and Port. This=20
> makes policy control a bit difficult as the policy enforcement point=20
> has to be able to identify STUN messages over Media in order to=20
> enforce policy. Has there been any discussion around having a=20
> management only IP/Port and as part of the allocation process allocate

> a local transport address (the address and port the UA requesting the=20
> allocation sends and receives from) in addition to the currently=20
> allocated relayed transport address?
> 	=20
> 	Such and approach would facilitate policy management as STUN
Relay=20
> (and STUN for that matter) can be classified separately from the media

> traffic allowing for separate policy enforcement.
> 	=20
> 	Regards,
> 	Kevin
>=20
>=20


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> I took a quick look at your draft. The main drawback I see is that it
> requires the firewall to implement it.

I realize my draft talks about firewalls.  However, that same function
(of deciding when to open/close a pinhole) can be implemented by a TURN
server. 

> Given the vast number of consumer
> NAT/Firewall devices this seems to have limited applicability.

That isn't what I suggested.

> Having not thought through all the details, I was expecting 
> that the UA
> could send a STUN allocate request and upon getting a response, send a
> binding request to the IP/Port assigned to the UA to 
> associate the STUN
> Relay with the UAs address and to open the NAT/Firewall binding. This
> would then allow the ICE connectivity checks.
> 
> Alternatively, the UA could wait till it gets the SDP answer with the
> remote UAs address info

For TURN's connectivity checks to work, the offering UA needs to wait 
for the answer anyway.  This is because TURN purposefully does not allow
receiving packets from any IP address:  the transport address assigned
by the TURN server will only allow packets from IP addresses specified
by the STUN client in a Set Active Destination request.  See the 
top of http://tools.ietf.org/html/draft-ietf-behave-turn-02#section-9.2

> then send the connectivity checks which would
> open the NAT bindings as well. The drawback to this is that the remote
> UA might start connectivity checks as soon as they send the answer.

That's fine -- they are retransmitted by that remote UA until they're
successful.

-d

> These would fail (no response) as the local UAs NAT will not have any
> bindings to allow the connectivity checks through (although this is
> currently the case and the receipt of the connectivity check cases a
> trigger check to cut down on the delay of waiting for a timer 
> to fire).
> 
> Kevin
> 
> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com] 
> Sent: Monday, January 15, 2007 11:28 PM
> To: Kevin Johns; 'Brian Stucker'; behave@ietf.org
> Subject: RE: [BEHAVE] STUN Relay
> 
> You need a bit more than STUN management traffic -- the endpoint needs
> to also be able to send ICE connectivity checks through the 
> TURN server.
> Otherwise, the endpoint will never know that the path through the TURN
> server is a viable path.
> 
> Implementing additional functionality in the TURN server, 
> akin to what I
> described in draft-wing-session-auth-00, should provide the hooks
> necessary to block or detect attempts to use a TURN server 
> without first
> signaling via SIP.
> 
> -d
> 
> 
> > -----Original Message-----
> > From: Kevin Johns [mailto:K.Johns@CableLabs.com]
> > Sent: Monday, January 15, 2007 9:26 AM
> > To: Brian Stucker; behave@ietf.org
> > Subject: RE: [BEHAVE] STUN Relay
> > 
> > Brian,
> >  
> > The concern is this, a UA needs to send a STUN message to a 
> STUN Relay
> 
> > server to get an allocated address to put in its SDP. To 
> allow this, 
> > the policy enforcement point (CMTS, GGSN, AP, etc.) needs to have a 
> > policy which allows the UA to send and receive messages 
> from the STUN 
> > Relay. Given the UA needs to communicate with the STUN 
> Relay prior to 
> > any session establishment, the policy needs to be static 
> and apply to 
> > all UAs. Now you have effectively installed a policy which 
> allows UAs 
> > to communicate freely via this static default policy.
> > Now, this may not be a big issue if the network installs a dynamic 
> > policy once the session is established and the dynamic policy takes 
> > precedence over the static. However, if the network does 
> not install a
> 
> > dynamic policy, the UA still has a free path to communicate with 
> > another UA.
> >  
> > So it would be nice to be able to install a default policy 
> which only 
> > allows for STUN management traffic. Thus, if a UA requests a relay 
> > address allocation the relay would happen on a different 
> address/port 
> > and thus block the UAs ability to relay media without a policy.
> >  
> > Kevin
> > 
> > ________________________________
> > 
> > From: Brian Stucker [mailto:bstucker@nortel.com]
> > Sent: Monday, January 15, 2007 9:54 AM
> > To: Kevin Johns; behave@ietf.org
> > Subject: RE: [BEHAVE] STUN Relay
> > 
> > 
> > What sort of policy management separation are you looking for? I am 
> > assuming that you are talking about gating behavior at the relay?
> >  
> > Regards,
> > Brian
> > 
> > 
> > ________________________________
> > 
> > 	From: Kevin Johns [mailto:K.Johns@CableLabs.com] 
> > 	Sent: Monday, January 15, 2007 10:48 AM
> > 	To: behave@ietf.org
> > 	Subject: [BEHAVE] STUN Relay
> > 	
> > 	
> > 	 
> > 	One of the properties of STUN relay is that the STUN relay
> server 
> > handles both STUN messages and media on the same IP and Port. This 
> > makes policy control a bit difficult as the policy 
> enforcement point 
> > has to be able to identify STUN messages over Media in order to 
> > enforce policy. Has there been any discussion around having a 
> > management only IP/Port and as part of the allocation 
> process allocate
> 
> > a local transport address (the address and port the UA 
> requesting the 
> > allocation sends and receives from) in addition to the currently 
> > allocated relayed transport address?
> > 	 
> > 	Such and approach would facilitate policy management as STUN
> Relay 
> > (and STUN for that matter) can be classified separately 
> from the media
> 
> > traffic allowing for separate policy enforcement.
> > 	 
> > 	Regards,
> > 	Kevin
> > 
> > 

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From: "Dan Wing" <dwing@cisco.com>
To: "'Vikas'" <pjain01@gmail.com>, <behave@ietf.org>
Subject: RE: [BEHAVE] running two application on same port
Date: Tue, 16 Jan 2007 10:28:04 -0800
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> For STUN client to determine the NAT type we may have to run 
> it on the port over which the needy application is already running.

RFC3489 recommended determining the NAT type.  However, in practice,
this has been found to not work well because some NATs change their
behavior after a several round-trips of UDP messages (becoming full 
cone, for example), and others change their behavior to "symmetric"
if multiple hosts behind the NAT use the same source UDP port.  

Because of this, it is now considered 'state of the art' to _not_
probe to determine the kind of NAT you're behind.

>   How can we achieve this functionality as now we have to run 
> stun client using the same port  and same protocol as our 
> application is using?

You have to multiplex STUN with your application protocol.  Some
application protocols, such as RTP (RFC3550) are easily multiplexed
with STUN because the first few bits of a STUN packet and of an
RTP packet are different.  DTLS happens to demultiplex well, too
(see table in 
http://tools.ietf.org/html/draft-mcgrew-tls-srtp-00#section-3.6.2).

If your application protocol doesn't demultiplex as easily, you
may find it useful to rely on STUN's MAGIC-COOKIE attribute and
FINGERPRINT attributes to demultiplex STUN and your application 
protocol.  You could, of course, prefix all of your 
application protocol messages with an 8-bit value that 
doesn't overlap with STUN's first 8 bits, but that shouldn't be
necessary considering STUN's MAGIC-COOKIE and FINGERPRINT 
attributes, which were designed to make it possible to 
demultiplex STUN with arbitrary application protocols.

-d

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From: "Kevin Johns" <K.Johns@CableLabs.com>
To: "Dan Wing" <dwing@cisco.com>, "Brian Stucker" <bstucker@nortel.com>,
	<behave@ietf.org>
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Dan,

Thanks for the clarification on your draft. I will re-read with that in
mind.

Kevin=20

-----Original Message-----
From: Dan Wing [mailto:dwing@cisco.com]=20
Sent: Tuesday, January 16, 2007 11:19 AM
To: Kevin Johns; 'Brian Stucker'; behave@ietf.org
Subject: RE: [BEHAVE] STUN Relay

> I took a quick look at your draft. The main drawback I see is that it=20
> requires the firewall to implement it.

I realize my draft talks about firewalls.  However, that same function
(of deciding when to open/close a pinhole) can be implemented by a TURN
server.=20

> Given the vast number of consumer
> NAT/Firewall devices this seems to have limited applicability.

That isn't what I suggested.

> Having not thought through all the details, I was expecting that the=20
> UA could send a STUN allocate request and upon getting a response,=20
> send a binding request to the IP/Port assigned to the UA to associate=20
> the STUN Relay with the UAs address and to open the NAT/Firewall=20
> binding. This would then allow the ICE connectivity checks.
>=20
> Alternatively, the UA could wait till it gets the SDP answer with the=20
> remote UAs address info

For TURN's connectivity checks to work, the offering UA needs to wait
for the answer anyway.  This is because TURN purposefully does not allow
receiving packets from any IP address:  the transport address assigned
by the TURN server will only allow packets from IP addresses specified
by the STUN client in a Set Active Destination request.  See the top of
http://tools.ietf.org/html/draft-ietf-behave-turn-02#section-9.2

> then send the connectivity checks which would open the NAT bindings as

> well. The drawback to this is that the remote UA might start=20
> connectivity checks as soon as they send the answer.

That's fine -- they are retransmitted by that remote UA until they're
successful.

-d

> These would fail (no response) as the local UAs NAT will not have any=20
> bindings to allow the connectivity checks through (although this is=20
> currently the case and the receipt of the connectivity check cases a=20
> trigger check to cut down on the delay of waiting for a timer to=20
> fire).
>=20
> Kevin
>=20
> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
> Sent: Monday, January 15, 2007 11:28 PM
> To: Kevin Johns; 'Brian Stucker'; behave@ietf.org
> Subject: RE: [BEHAVE] STUN Relay
>=20
> You need a bit more than STUN management traffic -- the endpoint needs

> to also be able to send ICE connectivity checks through the TURN=20
> server.
> Otherwise, the endpoint will never know that the path through the TURN

> server is a viable path.
>=20
> Implementing additional functionality in the TURN server, akin to what

> I described in draft-wing-session-auth-00, should provide the hooks=20
> necessary to block or detect attempts to use a TURN server without=20
> first signaling via SIP.
>=20
> -d
>=20
>=20
> > -----Original Message-----
> > From: Kevin Johns [mailto:K.Johns@CableLabs.com]
> > Sent: Monday, January 15, 2007 9:26 AM
> > To: Brian Stucker; behave@ietf.org
> > Subject: RE: [BEHAVE] STUN Relay
> >=20
> > Brian,
> > =20
> > The concern is this, a UA needs to send a STUN message to a
> STUN Relay
>=20
> > server to get an allocated address to put in its SDP. To
> allow this,
> > the policy enforcement point (CMTS, GGSN, AP, etc.) needs to have a=20
> > policy which allows the UA to send and receive messages
> from the STUN
> > Relay. Given the UA needs to communicate with the STUN
> Relay prior to
> > any session establishment, the policy needs to be static
> and apply to
> > all UAs. Now you have effectively installed a policy which
> allows UAs
> > to communicate freely via this static default policy.
> > Now, this may not be a big issue if the network installs a dynamic=20
> > policy once the session is established and the dynamic policy takes=20
> > precedence over the static. However, if the network does
> not install a
>=20
> > dynamic policy, the UA still has a free path to communicate with=20
> > another UA.
> > =20
> > So it would be nice to be able to install a default policy
> which only
> > allows for STUN management traffic. Thus, if a UA requests a relay=20
> > address allocation the relay would happen on a different
> address/port
> > and thus block the UAs ability to relay media without a policy.
> > =20
> > Kevin
> >=20
> > ________________________________
> >=20
> > From: Brian Stucker [mailto:bstucker@nortel.com]
> > Sent: Monday, January 15, 2007 9:54 AM
> > To: Kevin Johns; behave@ietf.org
> > Subject: RE: [BEHAVE] STUN Relay
> >=20
> >=20
> > What sort of policy management separation are you looking for? I am=20
> > assuming that you are talking about gating behavior at the relay?
> > =20
> > Regards,
> > Brian
> >=20
> >=20
> > ________________________________
> >=20
> > 	From: Kevin Johns [mailto:K.Johns@CableLabs.com]=20
> > 	Sent: Monday, January 15, 2007 10:48 AM
> > 	To: behave@ietf.org
> > 	Subject: [BEHAVE] STUN Relay
> > =09
> > =09
> > 	=20
> > 	One of the properties of STUN relay is that the STUN relay
> server
> > handles both STUN messages and media on the same IP and Port. This=20
> > makes policy control a bit difficult as the policy
> enforcement point
> > has to be able to identify STUN messages over Media in order to=20
> > enforce policy. Has there been any discussion around having a=20
> > management only IP/Port and as part of the allocation
> process allocate
>=20
> > a local transport address (the address and port the UA
> requesting the
> > allocation sends and receives from) in addition to the currently=20
> > allocated relayed transport address?
> > 	=20
> > 	Such and approach would facilitate policy management as STUN
> Relay
> > (and STUN for that matter) can be classified separately
> from the media
>=20
> > traffic allowing for separate policy enforcement.
> > 	=20
> > 	Regards,
> > 	Kevin
> >=20
> >=20


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I understand why you are asking for this capability, but it is a 
tremendous change to the spec at this late hour, and I think there are 
other ways to achieve your goal.

I'll also point out that the approach you describe is hard to make work 
securely. It introduces an attack whereby an observer can watch the 
allocate message, learn the allocated port and associated cookie/token 
used to correlate the requests, and inject a faked request from their 
own port. This attack would require something akin to a shared secret 
request over TLS for each call. It also adds a lot of machinery to the 
spec. We considered an approach like this waaaayyyy back in the early 
days of TURN.

The easiest thing, IMHO, is for the firewall to just discard anything 
which is not a STUN message until authorization, and require the UA to 
set an active destination within a few seconds of the allocate request 
(the latter would be enforced on the stun relay). The authorization that 
gets installed then would be to allow non-stun messages. This requires 
the firewall function to peer a bit deeper than ports, but stun was 
designed to be easily recognizable by intermediate boxes.

Thanks,
Jonathan R.


Kevin Johns wrote:

> Dan,
> 
> I took a quick look at your draft. The main drawback I see is that it
> requires the firewall to implement it. Given the vast number of consumer
> NAT/Firewall devices this seems to have limited applicability.
> 
> Having not thought through all the details, I was expecting that the UA
> could send a STUN allocate request and upon getting a response, send a
> binding request to the IP/Port assigned to the UA to associate the STUN
> Relay with the UAs address and to open the NAT/Firewall binding. This
> would then allow the ICE connectivity checks.
> 
> Alternatively, the UA could wait till it gets the SDP answer with the
> remote UAs address info then send the connectivity checks which would
> open the NAT bindings as well. The drawback to this is that the remote
> UA might start connectivity checks as soon as they send the answer.
> These would fail (no response) as the local UAs NAT will not have any
> bindings to allow the connectivity checks through (although this is
> currently the case and the receipt of the connectivity check cases a
> trigger check to cut down on the delay of waiting for a timer to fire).
> 
> Kevin
> 
> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com] 
> Sent: Monday, January 15, 2007 11:28 PM
> To: Kevin Johns; 'Brian Stucker'; behave@ietf.org
> Subject: RE: [BEHAVE] STUN Relay
> 
> You need a bit more than STUN management traffic -- the endpoint needs
> to also be able to send ICE connectivity checks through the TURN server.
> Otherwise, the endpoint will never know that the path through the TURN
> server is a viable path.
> 
> Implementing additional functionality in the TURN server, akin to what I
> described in draft-wing-session-auth-00, should provide the hooks
> necessary to block or detect attempts to use a TURN server without first
> signaling via SIP.
> 
> -d
> 
> 
> 
>>-----Original Message-----
>>From: Kevin Johns [mailto:K.Johns@CableLabs.com]
>>Sent: Monday, January 15, 2007 9:26 AM
>>To: Brian Stucker; behave@ietf.org
>>Subject: RE: [BEHAVE] STUN Relay
>>
>>Brian,
>> 
>>The concern is this, a UA needs to send a STUN message to a STUN Relay
> 
> 
>>server to get an allocated address to put in its SDP. To allow this, 
>>the policy enforcement point (CMTS, GGSN, AP, etc.) needs to have a 
>>policy which allows the UA to send and receive messages from the STUN 
>>Relay. Given the UA needs to communicate with the STUN Relay prior to 
>>any session establishment, the policy needs to be static and apply to 
>>all UAs. Now you have effectively installed a policy which allows UAs 
>>to communicate freely via this static default policy.
>>Now, this may not be a big issue if the network installs a dynamic 
>>policy once the session is established and the dynamic policy takes 
>>precedence over the static. However, if the network does not install a
> 
> 
>>dynamic policy, the UA still has a free path to communicate with 
>>another UA.
>> 
>>So it would be nice to be able to install a default policy which only 
>>allows for STUN management traffic. Thus, if a UA requests a relay 
>>address allocation the relay would happen on a different address/port 
>>and thus block the UAs ability to relay media without a policy.
>> 
>>Kevin
>>
>>________________________________
>>
>>From: Brian Stucker [mailto:bstucker@nortel.com]
>>Sent: Monday, January 15, 2007 9:54 AM
>>To: Kevin Johns; behave@ietf.org
>>Subject: RE: [BEHAVE] STUN Relay
>>
>>
>>What sort of policy management separation are you looking for? I am 
>>assuming that you are talking about gating behavior at the relay?
>> 
>>Regards,
>>Brian
>>
>>
>>________________________________
>>
>>	From: Kevin Johns [mailto:K.Johns@CableLabs.com] 
>>	Sent: Monday, January 15, 2007 10:48 AM
>>	To: behave@ietf.org
>>	Subject: [BEHAVE] STUN Relay
>>	
>>	
>>	 
>>	One of the properties of STUN relay is that the STUN relay
> 
> server 
> 
>>handles both STUN messages and media on the same IP and Port. This 
>>makes policy control a bit difficult as the policy enforcement point 
>>has to be able to identify STUN messages over Media in order to 
>>enforce policy. Has there been any discussion around having a 
>>management only IP/Port and as part of the allocation process allocate
> 
> 
>>a local transport address (the address and port the UA requesting the 
>>allocation sends and receives from) in addition to the currently 
>>allocated relayed transport address?
>>	 
>>	Such and approach would facilitate policy management as STUN
> 
> Relay 
> 
>>(and STUN for that matter) can be classified separately from the media
> 
> 
>>traffic allowing for separate policy enforcement.
>>	 
>>	Regards,
>>	Kevin
>>
>>
> 
> 
> 
> _______________________________________________
> 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 Tue Jan 16 17:43:21 2007
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Subject: RE: [BEHAVE] STUN Relay
Date: Tue, 16 Jan 2007 16:42:54 -0600
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Thread-Topic: [BEHAVE] STUN Relay
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This approach has a side effect...

If you rely on the client sending a SAD request to the TURN server
'within a few seconds of the allocate request' then you're very likely
going to clip the first few seconds of any early media going back to the
client using the existing mechanism.

If a gate is set by a component in the SIP signaling path, the clipping
can still occur, but should be the minimum interval possible given that
a gate exists in the network at all. I'm not suggesting changes to TURN
in regards to handling clipping other than to point out that we're
essentially requiring the originator to send packets (STUN or otherwise)
on the media path in order to get any media back from far ends, even if
that far end is a trusted network entity.

Should we perhaps state that the relay client should be prepared to
receive media from the relay server in the absence of any permissions
that the client may have created on the allocation via SAD/SI messages?
That would allow local network policy to stream early media to the relay
client without waiting for the client to interact with the relay server
outside of requesting the allocation (presumably as part of a candidate
address gathering exercise).

Regards,
Brian


> -----Original Message-----
> From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
> Sent: Tuesday, January 16, 2007 3:09 PM
> To: Kevin Johns
> Cc: Dan Wing; Stucker, Brian (RICH1:AR00); behave@ietf.org
> Subject: Re: [BEHAVE] STUN Relay
>=20
> I understand why you are asking for this capability, but it=20
> is a tremendous change to the spec at this late hour, and I=20
> think there are other ways to achieve your goal.
>=20
> I'll also point out that the approach you describe is hard to=20
> make work securely. It introduces an attack whereby an=20
> observer can watch the allocate message, learn the allocated=20
> port and associated cookie/token used to correlate the=20
> requests, and inject a faked request from their own port.=20
> This attack would require something akin to a shared secret=20
> request over TLS for each call. It also adds a lot of=20
> machinery to the spec. We considered an approach like this=20
> waaaayyyy back in the early days of TURN.
>=20
> The easiest thing, IMHO, is for the firewall to just discard=20
> anything which is not a STUN message until authorization, and=20
> require the UA to set an active destination within a few=20
> seconds of the allocate request (the latter would be enforced=20
> on the stun relay). The authorization that gets installed=20
> then would be to allow non-stun messages. This requires the=20
> firewall function to peer a bit deeper than ports, but stun=20
> was designed to be easily recognizable by intermediate boxes.
>=20
> Thanks,
> Jonathan R.
>=20
>=20
> Kevin Johns wrote:
>=20
> > Dan,
> >=20
> > I took a quick look at your draft. The main drawback I see=20
> is that it=20
> > requires the firewall to implement it. Given the vast number of=20
> > consumer NAT/Firewall devices this seems to have limited=20
> applicability.
> >=20
> > Having not thought through all the details, I was expecting=20
> that the=20
> > UA could send a STUN allocate request and upon getting a response,=20
> > send a binding request to the IP/Port assigned to the UA to=20
> associate=20
> > the STUN Relay with the UAs address and to open the NAT/Firewall=20
> > binding. This would then allow the ICE connectivity checks.
> >=20
> > Alternatively, the UA could wait till it gets the SDP=20
> answer with the=20
> > remote UAs address info then send the connectivity checks=20
> which would=20
> > open the NAT bindings as well. The drawback to this is that=20
> the remote=20
> > UA might start connectivity checks as soon as they send the answer.
> > These would fail (no response) as the local UAs NAT will=20
> not have any=20
> > bindings to allow the connectivity checks through (although this is=20
> > currently the case and the receipt of the connectivity=20
> check cases a=20
> > trigger check to cut down on the delay of waiting for a=20
> timer to fire).
> >=20
> > Kevin
> >=20
> > -----Original Message-----
> > From: Dan Wing [mailto:dwing@cisco.com]
> > Sent: Monday, January 15, 2007 11:28 PM
> > To: Kevin Johns; 'Brian Stucker'; behave@ietf.org
> > Subject: RE: [BEHAVE] STUN Relay
> >=20
> > You need a bit more than STUN management traffic -- the=20
> endpoint needs=20
> > to also be able to send ICE connectivity checks through the=20
> TURN server.
> > Otherwise, the endpoint will never know that the path=20
> through the TURN=20
> > server is a viable path.
> >=20
> > Implementing additional functionality in the TURN server,=20
> akin to what=20
> > I described in draft-wing-session-auth-00, should provide the hooks=20
> > necessary to block or detect attempts to use a TURN server without=20
> > first signaling via SIP.
> >=20
> > -d
> >=20
> >=20
> >=20
> >>-----Original Message-----
> >>From: Kevin Johns [mailto:K.Johns@CableLabs.com]
> >>Sent: Monday, January 15, 2007 9:26 AM
> >>To: Brian Stucker; behave@ietf.org
> >>Subject: RE: [BEHAVE] STUN Relay
> >>
> >>Brian,
> >>=20
> >>The concern is this, a UA needs to send a STUN message to a=20
> STUN Relay
> >=20
> >=20
> >>server to get an allocated address to put in its SDP. To=20
> allow this,=20
> >>the policy enforcement point (CMTS, GGSN, AP, etc.) needs to have a=20
> >>policy which allows the UA to send and receive messages=20
> from the STUN=20
> >>Relay. Given the UA needs to communicate with the STUN=20
> Relay prior to=20
> >>any session establishment, the policy needs to be static=20
> and apply to=20
> >>all UAs. Now you have effectively installed a policy which=20
> allows UAs=20
> >>to communicate freely via this static default policy.
> >>Now, this may not be a big issue if the network installs a dynamic=20
> >>policy once the session is established and the dynamic policy takes=20
> >>precedence over the static. However, if the network does=20
> not install a
> >=20
> >=20
> >>dynamic policy, the UA still has a free path to communicate with=20
> >>another UA.
> >>=20
> >>So it would be nice to be able to install a default policy=20
> which only=20
> >>allows for STUN management traffic. Thus, if a UA requests a relay=20
> >>address allocation the relay would happen on a different=20
> address/port=20
> >>and thus block the UAs ability to relay media without a policy.
> >>=20
> >>Kevin
> >>
> >>________________________________
> >>
> >>From: Brian Stucker [mailto:bstucker@nortel.com]
> >>Sent: Monday, January 15, 2007 9:54 AM
> >>To: Kevin Johns; behave@ietf.org
> >>Subject: RE: [BEHAVE] STUN Relay
> >>
> >>
> >>What sort of policy management separation are you looking for? I am=20
> >>assuming that you are talking about gating behavior at the relay?
> >>=20
> >>Regards,
> >>Brian
> >>
> >>
> >>________________________________
> >>
> >>	From: Kevin Johns [mailto:K.Johns@CableLabs.com]=20
> >>	Sent: Monday, January 15, 2007 10:48 AM
> >>	To: behave@ietf.org
> >>	Subject: [BEHAVE] STUN Relay
> >>=09
> >>=09
> >>	=20
> >>	One of the properties of STUN relay is that the STUN relay
> >=20
> > server
> >=20
> >>handles both STUN messages and media on the same IP and Port. This=20
> >>makes policy control a bit difficult as the policy=20
> enforcement point=20
> >>has to be able to identify STUN messages over Media in order to=20
> >>enforce policy. Has there been any discussion around having a=20
> >>management only IP/Port and as part of the allocation=20
> process allocate
> >=20
> >=20
> >>a local transport address (the address and port the UA=20
> requesting the=20
> >>allocation sends and receives from) in addition to the currently=20
> >>allocated relayed transport address?
> >>	=20
> >>	Such and approach would facilitate policy management as STUN
> >=20
> > Relay
> >=20
> >>(and STUN for that matter) can be classified separately=20
> from the media
> >=20
> >=20
> >>traffic allowing for separate policy enforcement.
> >>	=20
> >>	Regards,
> >>	Kevin
> >>
> >>
> >=20
> >=20
> >=20
> > _______________________________________________
> > Behave mailing list
> > Behave@ietf.org
> > https://www1.ietf.org/mailman/listinfo/behave
> >=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Cisco Fellow                                   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@cisco.com                              FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20

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Thanks for the history. If this was considered in the past and rejected
I am ok with that, was just wondering if it was considered and if so why
it was not done.

Your suggested approach seems to be along the same lines as Dans. I'll
plan on taking a more detailed review of his draft to better understand
the architecture and potential implications of such an approach.

Kevin

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20
Sent: Tuesday, January 16, 2007 2:09 PM
To: Kevin Johns
Cc: Dan Wing; Brian Stucker; behave@ietf.org
Subject: Re: [BEHAVE] STUN Relay

I understand why you are asking for this capability, but it is a
tremendous change to the spec at this late hour, and I think there are
other ways to achieve your goal.

I'll also point out that the approach you describe is hard to make work
securely. It introduces an attack whereby an observer can watch the
allocate message, learn the allocated port and associated cookie/token
used to correlate the requests, and inject a faked request from their
own port. This attack would require something akin to a shared secret
request over TLS for each call. It also adds a lot of machinery to the
spec. We considered an approach like this waaaayyyy back in the early
days of TURN.

The easiest thing, IMHO, is for the firewall to just discard anything
which is not a STUN message until authorization, and require the UA to
set an active destination within a few seconds of the allocate request
(the latter would be enforced on the stun relay). The authorization that
gets installed then would be to allow non-stun messages. This requires
the firewall function to peer a bit deeper than ports, but stun was
designed to be easily recognizable by intermediate boxes.

Thanks,
Jonathan R.


Kevin Johns wrote:

> Dan,
>=20
> I took a quick look at your draft. The main drawback I see is that it=20
> requires the firewall to implement it. Given the vast number of=20
> consumer NAT/Firewall devices this seems to have limited
applicability.
>=20
> Having not thought through all the details, I was expecting that the=20
> UA could send a STUN allocate request and upon getting a response,=20
> send a binding request to the IP/Port assigned to the UA to associate=20
> the STUN Relay with the UAs address and to open the NAT/Firewall=20
> binding. This would then allow the ICE connectivity checks.
>=20
> Alternatively, the UA could wait till it gets the SDP answer with the=20
> remote UAs address info then send the connectivity checks which would=20
> open the NAT bindings as well. The drawback to this is that the remote

> UA might start connectivity checks as soon as they send the answer.
> These would fail (no response) as the local UAs NAT will not have any=20
> bindings to allow the connectivity checks through (although this is=20
> currently the case and the receipt of the connectivity check cases a=20
> trigger check to cut down on the delay of waiting for a timer to
fire).
>=20
> Kevin
>=20
> -----Original Message-----
> From: Dan Wing [mailto:dwing@cisco.com]
> Sent: Monday, January 15, 2007 11:28 PM
> To: Kevin Johns; 'Brian Stucker'; behave@ietf.org
> Subject: RE: [BEHAVE] STUN Relay
>=20
> You need a bit more than STUN management traffic -- the endpoint needs

> to also be able to send ICE connectivity checks through the TURN
server.
> Otherwise, the endpoint will never know that the path through the TURN

> server is a viable path.
>=20
> Implementing additional functionality in the TURN server, akin to what

> I described in draft-wing-session-auth-00, should provide the hooks=20
> necessary to block or detect attempts to use a TURN server without=20
> first signaling via SIP.
>=20
> -d
>=20
>=20
>=20
>>-----Original Message-----
>>From: Kevin Johns [mailto:K.Johns@CableLabs.com]
>>Sent: Monday, January 15, 2007 9:26 AM
>>To: Brian Stucker; behave@ietf.org
>>Subject: RE: [BEHAVE] STUN Relay
>>
>>Brian,
>>=20
>>The concern is this, a UA needs to send a STUN message to a STUN Relay
>=20
>=20
>>server to get an allocated address to put in its SDP. To allow this,=20
>>the policy enforcement point (CMTS, GGSN, AP, etc.) needs to have a=20
>>policy which allows the UA to send and receive messages from the STUN=20
>>Relay. Given the UA needs to communicate with the STUN Relay prior to=20
>>any session establishment, the policy needs to be static and apply to=20
>>all UAs. Now you have effectively installed a policy which allows UAs=20
>>to communicate freely via this static default policy.
>>Now, this may not be a big issue if the network installs a dynamic=20
>>policy once the session is established and the dynamic policy takes=20
>>precedence over the static. However, if the network does not install a
>=20
>=20
>>dynamic policy, the UA still has a free path to communicate with=20
>>another UA.
>>=20
>>So it would be nice to be able to install a default policy which only=20
>>allows for STUN management traffic. Thus, if a UA requests a relay=20
>>address allocation the relay would happen on a different address/port=20
>>and thus block the UAs ability to relay media without a policy.
>>=20
>>Kevin
>>
>>________________________________
>>
>>From: Brian Stucker [mailto:bstucker@nortel.com]
>>Sent: Monday, January 15, 2007 9:54 AM
>>To: Kevin Johns; behave@ietf.org
>>Subject: RE: [BEHAVE] STUN Relay
>>
>>
>>What sort of policy management separation are you looking for? I am=20
>>assuming that you are talking about gating behavior at the relay?
>>=20
>>Regards,
>>Brian
>>
>>
>>________________________________
>>
>>	From: Kevin Johns [mailto:K.Johns@CableLabs.com]=20
>>	Sent: Monday, January 15, 2007 10:48 AM
>>	To: behave@ietf.org
>>	Subject: [BEHAVE] STUN Relay
>>=09
>>=09
>>	=20
>>	One of the properties of STUN relay is that the STUN relay
>=20
> server
>=20
>>handles both STUN messages and media on the same IP and Port. This=20
>>makes policy control a bit difficult as the policy enforcement point=20
>>has to be able to identify STUN messages over Media in order to=20
>>enforce policy. Has there been any discussion around having a=20
>>management only IP/Port and as part of the allocation process allocate
>=20
>=20
>>a local transport address (the address and port the UA requesting the=20
>>allocation sends and receives from) in addition to the currently=20
>>allocated relayed transport address?
>>	=20
>>	Such and approach would facilitate policy management as STUN
>=20
> Relay
>=20
>>(and STUN for that matter) can be classified separately from the media
>=20
>=20
>>traffic allowing for separate policy enforcement.
>>	=20
>>	Regards,
>>	Kevin
>>
>>
>=20
>=20
>=20
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
>=20

--=20
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Cisco Fellow                                   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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



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From: "Dan Wing" <dwing@cisco.com>
To: "'Brian Stucker'" <bstucker@nortel.com>,
	"'Jonathan Rosenberg'" <jdrosen@cisco.com>,
	"'Kevin Johns'" <K.Johns@CableLabs.com>
Subject: RE: [BEHAVE] STUN Relay
Date: Tue, 16 Jan 2007 15:10:17 -0800
Keywords: direct-to-dwing
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(Assuming I understood your proposal correctly:)

One of TURN's design goals was to _not_ allow TURN to be used
to create a server on the 'far' side of a firewall.  If we
allow a TURN server to relay IP packets from arbitrary hosts
on the Internet to the STUN client, it seems to break that
design goal.

-d


> -----Original Message-----
> From: Brian Stucker [mailto:bstucker@nortel.com] 
> Sent: Tuesday, January 16, 2007 2:43 PM
> To: Jonathan Rosenberg; Kevin Johns
> Cc: Dan Wing; behave@ietf.org
> Subject: RE: [BEHAVE] STUN Relay
> 
> This approach has a side effect...
> 
> If you rely on the client sending a SAD request to the TURN server
> 'within a few seconds of the allocate request' then you're very likely
> going to clip the first few seconds of any early media going 
> back to the
> client using the existing mechanism.
> 
> If a gate is set by a component in the SIP signaling path, 
> the clipping
> can still occur, but should be the minimum interval possible 
> given that
> a gate exists in the network at all. I'm not suggesting 
> changes to TURN
> in regards to handling clipping other than to point out that we're
> essentially requiring the originator to send packets (STUN or 
> otherwise)
> on the media path in order to get any media back from far 
> ends, even if
> that far end is a trusted network entity.
> 
> Should we perhaps state that the relay client should be prepared to
> receive media from the relay server in the absence of any permissions
> that the client may have created on the allocation via SAD/SI 
> messages?
> That would allow local network policy to stream early media 
> to the relay
> client without waiting for the client to interact with the 
> relay server
> outside of requesting the allocation (presumably as part of a 
> candidate
> address gathering exercise).
> 
> Regards,
> Brian
> 
> 
> > -----Original Message-----
> > From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] 
> > Sent: Tuesday, January 16, 2007 3:09 PM
> > To: Kevin Johns
> > Cc: Dan Wing; Stucker, Brian (RICH1:AR00); behave@ietf.org
> > Subject: Re: [BEHAVE] STUN Relay
> > 
> > I understand why you are asking for this capability, but it 
> > is a tremendous change to the spec at this late hour, and I 
> > think there are other ways to achieve your goal.
> > 
> > I'll also point out that the approach you describe is hard to 
> > make work securely. It introduces an attack whereby an 
> > observer can watch the allocate message, learn the allocated 
> > port and associated cookie/token used to correlate the 
> > requests, and inject a faked request from their own port. 
> > This attack would require something akin to a shared secret 
> > request over TLS for each call. It also adds a lot of 
> > machinery to the spec. We considered an approach like this 
> > waaaayyyy back in the early days of TURN.
> > 
> > The easiest thing, IMHO, is for the firewall to just discard 
> > anything which is not a STUN message until authorization, and 
> > require the UA to set an active destination within a few 
> > seconds of the allocate request (the latter would be enforced 
> > on the stun relay). The authorization that gets installed 
> > then would be to allow non-stun messages. This requires the 
> > firewall function to peer a bit deeper than ports, but stun 
> > was designed to be easily recognizable by intermediate boxes.
> > 
> > Thanks,
> > Jonathan R.
> > 
> > 
> > Kevin Johns wrote:
> > 
> > > Dan,
> > > 
> > > I took a quick look at your draft. The main drawback I see 
> > is that it 
> > > requires the firewall to implement it. Given the vast number of 
> > > consumer NAT/Firewall devices this seems to have limited 
> > applicability.
> > > 
> > > Having not thought through all the details, I was expecting 
> > that the 
> > > UA could send a STUN allocate request and upon getting a 
> response, 
> > > send a binding request to the IP/Port assigned to the UA to 
> > associate 
> > > the STUN Relay with the UAs address and to open the NAT/Firewall 
> > > binding. This would then allow the ICE connectivity checks.
> > > 
> > > Alternatively, the UA could wait till it gets the SDP 
> > answer with the 
> > > remote UAs address info then send the connectivity checks 
> > which would 
> > > open the NAT bindings as well. The drawback to this is that 
> > the remote 
> > > UA might start connectivity checks as soon as they send 
> the answer.
> > > These would fail (no response) as the local UAs NAT will 
> > not have any 
> > > bindings to allow the connectivity checks through 
> (although this is 
> > > currently the case and the receipt of the connectivity 
> > check cases a 
> > > trigger check to cut down on the delay of waiting for a 
> > timer to fire).
> > > 
> > > Kevin
> > > 
> > > -----Original Message-----
> > > From: Dan Wing [mailto:dwing@cisco.com]
> > > Sent: Monday, January 15, 2007 11:28 PM
> > > To: Kevin Johns; 'Brian Stucker'; behave@ietf.org
> > > Subject: RE: [BEHAVE] STUN Relay
> > > 
> > > You need a bit more than STUN management traffic -- the 
> > endpoint needs 
> > > to also be able to send ICE connectivity checks through the 
> > TURN server.
> > > Otherwise, the endpoint will never know that the path 
> > through the TURN 
> > > server is a viable path.
> > > 
> > > Implementing additional functionality in the TURN server, 
> > akin to what 
> > > I described in draft-wing-session-auth-00, should provide 
> the hooks 
> > > necessary to block or detect attempts to use a TURN 
> server without 
> > > first signaling via SIP.
> > > 
> > > -d
> > > 
> > > 
> > > 
> > >>-----Original Message-----
> > >>From: Kevin Johns [mailto:K.Johns@CableLabs.com]
> > >>Sent: Monday, January 15, 2007 9:26 AM
> > >>To: Brian Stucker; behave@ietf.org
> > >>Subject: RE: [BEHAVE] STUN Relay
> > >>
> > >>Brian,
> > >> 
> > >>The concern is this, a UA needs to send a STUN message to a 
> > STUN Relay
> > > 
> > > 
> > >>server to get an allocated address to put in its SDP. To 
> > allow this, 
> > >>the policy enforcement point (CMTS, GGSN, AP, etc.) needs 
> to have a 
> > >>policy which allows the UA to send and receive messages 
> > from the STUN 
> > >>Relay. Given the UA needs to communicate with the STUN 
> > Relay prior to 
> > >>any session establishment, the policy needs to be static 
> > and apply to 
> > >>all UAs. Now you have effectively installed a policy which 
> > allows UAs 
> > >>to communicate freely via this static default policy.
> > >>Now, this may not be a big issue if the network installs 
> a dynamic 
> > >>policy once the session is established and the dynamic 
> policy takes 
> > >>precedence over the static. However, if the network does 
> > not install a
> > > 
> > > 
> > >>dynamic policy, the UA still has a free path to communicate with 
> > >>another UA.
> > >> 
> > >>So it would be nice to be able to install a default policy 
> > which only 
> > >>allows for STUN management traffic. Thus, if a UA 
> requests a relay 
> > >>address allocation the relay would happen on a different 
> > address/port 
> > >>and thus block the UAs ability to relay media without a policy.
> > >> 
> > >>Kevin
> > >>
> > >>________________________________
> > >>
> > >>From: Brian Stucker [mailto:bstucker@nortel.com]
> > >>Sent: Monday, January 15, 2007 9:54 AM
> > >>To: Kevin Johns; behave@ietf.org
> > >>Subject: RE: [BEHAVE] STUN Relay
> > >>
> > >>
> > >>What sort of policy management separation are you looking 
> for? I am 
> > >>assuming that you are talking about gating behavior at the relay?
> > >> 
> > >>Regards,
> > >>Brian
> > >>
> > >>
> > >>________________________________
> > >>
> > >>	From: Kevin Johns [mailto:K.Johns@CableLabs.com] 
> > >>	Sent: Monday, January 15, 2007 10:48 AM
> > >>	To: behave@ietf.org
> > >>	Subject: [BEHAVE] STUN Relay
> > >>	
> > >>	
> > >>	 
> > >>	One of the properties of STUN relay is that the STUN relay
> > > 
> > > server
> > > 
> > >>handles both STUN messages and media on the same IP and 
> Port. This 
> > >>makes policy control a bit difficult as the policy 
> > enforcement point 
> > >>has to be able to identify STUN messages over Media in order to 
> > >>enforce policy. Has there been any discussion around having a 
> > >>management only IP/Port and as part of the allocation 
> > process allocate
> > > 
> > > 
> > >>a local transport address (the address and port the UA 
> > requesting the 
> > >>allocation sends and receives from) in addition to the currently 
> > >>allocated relayed transport address?
> > >>	 
> > >>	Such and approach would facilitate policy management as STUN
> > > 
> > > Relay
> > > 
> > >>(and STUN for that matter) can be classified separately 
> > from the media
> > > 
> > > 
> > >>traffic allowing for separate policy enforcement.
> > >>	 
> > >>	Regards,
> > >>	Kevin
> > >>
> > >>
> > > 
> > > 
> > > 
> > > _______________________________________________
> > > Behave mailing list
> > > Behave@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/behave
> > > 
> > 
> > -- 
> > Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> > Cisco Fellow                                   Parsippany, NJ 
> > 07054-2711
> > Cisco Systems
> > jdrosen@cisco.com                              FAX:   (973) 952-5050
> > http://www.jdrosen.net                         PHONE: (973) 952-5000
> > http://www.cisco.com
> > 

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



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Subject: [BEHAVE] RE: [AVT] Update of rtp-keepalive draft
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The issue of keepalive is also related to the RTP endpoint - If the =
remote side stops the voice connection because of some problem during =
silence, there is no certain way to discover it.
=20
So I think the recommended alternative should solve both issues (4.5 =
does the job)
=20
Regards,
Yuval

________________________________

=EE=E0=FA: Colin Perkins [mailto:csp@csperkins.org]
=F0=F9=EC=E7: =E2 16/01/2007 18:58
=E0=EC: Lee Dilkie
=F2=E5=FA=F7 =EC=E9=E3=E9=F2=E4: behave@ietf.org; Xavier Marjou; =
avt@ietf.org
=F0=E5=F9=E0: Re: [AVT] Update of rtp-keepalive draft



Do we need to mandate an alternative? All of the alternatives=20
specified will work, and RTP implementations have to be robust to=20
receiving all seven types of packet anyway (otherwise, there is a=20
trivial denial-of-service attack on the application). As long as=20
something is sent at an appropriate interval, the NAT will be happy,=20
and the application can use whatever mechanism is appropriate for it.

Colin





On 15 Jan 2007, at 19:04, Lee Dilkie wrote:
> I'd vote for 4.1 first, followed by 4.5 (which sounds a lot like=20
> 4.7 to
> me only it's defined).
>
> reasons:
>
> sending a 0-byte udp packet is solving a network issue, not an RTP=20
> issue
> and shouldn't pretend to be RTP.
>
> the cons specified:
>  - udp specific, well yes but this isn't an issue with tcp anyway.
>  - rx-ing party might not handle this. Perhaps, but since these are
> intended to be used for NAT traversal I would assume the rx-ing party
> would be most probably aware since the address:port of the received=20
> udp
> needs to be communicated to the tx-ing party anyway, and they=20
> generally
> need to share the same socket(or at least appear to).
>
> Otherwise I'd pick an unused RTP payload type value and define it as
> reserved for NAT keepalive usage. I guess this most closely maps to=20
> 4.5.
>
> Regards,
>
> Lee Dilkie
>
> (we implemented the 0-byte udp in our solution, FWIW, although we=20
> choose
> 30 second intervals).
>
> Xavier Marjou wrote:
>> To keep this topic alive, I plan to update the following individual
>> draft:
>> http://tools.ietf.org/wg/behave/draft-marjou-behave-app-rtp-
>> keepalive-00.txt
>>
>>
>> I am therefore interested of any feedback, especially on which
>> mechanism(s) is best.
>>
>> Xavier
>>
>>> On 12/3/06, Dan Wing <dwing@cisco.com> wrote:
>>> Minutes of the last BEHAVE meeting have been posted to:
>>> http://www3.ietf.org/proceedings/06nov/minutes/behave.html
>>>
>>> Application Keepalive for NATs, Xavier Marjou -
>>> draft-marjou-behave-app-rtp-keepalive-00
>>> status:  separate from ICE, or integrate into Application milestone
>>> document?
>>>
>>> JR =96 this is app design guidelines. How about if we charter to=20
>>> create
>>> that document and put this text, along with other text around it in
>>> STUN all in one document?
>>> Xavier Marjou (speaker, XM) =96 take to list.
>>> Chair =96 question of which mechanisms might be best asked in AVT FA =
=96
>>> good to ask AVT for sure. But we should discuss here Magnus =96 both
>>> groups working on it Chair =96 consensus to fold this into =
application
>>> document, per JR's proposal.
>>> Will figure out what to do on list.
>>> FA =96 support. Think it's a good idea. But do we have an editor for
>>> that, will it get done? I don't want to send this to dev/null.
>>> PM =96 in our BEHAVE app document are we the best place to have that
>>> discussion?
>>> chair - don't have a clear decision on what to do, will work to=20
>>> figure
>>> out where this goes.
>>
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www1.ietf.org/mailman/listinfo/avt
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt

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




_______________________________________________
Audio/Video Transport Working Group
avt@ietf.org
https://www1.ietf.org/mailman/listinfo/avt



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<HTML>=0A=
<HEAD>=0A=
=0A=
<META NAME=3D"Generator" CONTENT=3D"MS Exchange Server version =
6.5.7638.1">=0A=
<TITLE>Re: [AVT] Update of rtp-keepalive draft</TITLE>=0A=
</HEAD>=0A=
<BODY>=0A=
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 size=3D2>The issue of =
keepalive is =0A=
also related to the RTP endpoint -&nbsp;If the remote side stops the =
voice =0A=
connection because of some problem during silence, there is no certain =
way to =0A=
discover it.</FONT></DIV>=0A=
<DIV dir=3Dltr>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial color=3D#000000 size=3D2>So I think =
the recommended =0A=
alternative should solve both issues (4.5 does the job)</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2></FONT>&nbsp;</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Regards,</FONT></DIV>=0A=
<DIV dir=3Dltr><FONT face=3DArial size=3D2>Yuval</FONT></DIV>=0A=
<DIV dir=3Dltr><BR></DIV>=0A=
<DIV dir=3Dltr>=0A=
<HR tabIndex=3D-1>=0A=
</DIV>=0A=
<DIV dir=3Dltr><FONT face=3DTahoma size=3D2><B>=EE=E0=FA:</B> Colin =
Perkins =0A=
[mailto:csp@csperkins.org]<BR><B>=F0=F9=EC=E7:</B> =E2 16/01/2007 =
18:58<BR><B>=E0=EC:</B> Lee =0A=
Dilkie<BR><B>=F2=E5=FA=F7 =EC=E9=E3=E9=F2=E4:</B> behave@ietf.org; =
Xavier Marjou; =0A=
avt@ietf.org<BR><B>=F0=E5=F9=E0:</B> Re: [AVT] Update of rtp-keepalive =0A=
draft<BR></FONT><BR></DIV>=0A=
<P dir=3Dltr><FONT size=3D2>Do we need to mandate an alternative? All of =
the =0A=
alternatives&nbsp;<BR>specified will work, and RTP implementations have =
to be =0A=
robust to&nbsp;<BR>receiving all seven types of packet anyway =
(otherwise, there =0A=
is a&nbsp;<BR>trivial denial-of-service attack on the application). As =
long =0A=
as&nbsp;<BR>something is sent at an appropriate interval, the NAT will =
be =0A=
happy,&nbsp;<BR>and the application can use whatever mechanism is =
appropriate =0A=
for it.<BR><BR>Colin<BR><BR><BR><BR><BR><BR>On 15 Jan 2007, at 19:04, =
Lee Dilkie =0A=
wrote:<BR>&gt; I'd vote for 4.1 first, followed by 4.5 (which sounds a =
lot =0A=
like&nbsp;<BR>&gt; 4.7 to<BR>&gt; me only it's defined).<BR>&gt;<BR>&gt; =0A=
reasons:<BR>&gt;<BR>&gt; sending a 0-byte udp packet is solving a =
network issue, =0A=
not an RTP&nbsp;<BR>&gt; issue<BR>&gt; and shouldn't pretend to be =0A=
RTP.<BR>&gt;<BR>&gt; the cons specified:<BR>&gt;&nbsp; - udp specific, =
well yes =0A=
but this isn't an issue with tcp anyway.<BR>&gt;&nbsp; - rx-ing party =
might not =0A=
handle this. Perhaps, but since these are<BR>&gt; intended to be used =
for NAT =0A=
traversal I would assume the rx-ing party<BR>&gt; would be most probably =
aware =0A=
since the address:port of the received&nbsp;<BR>&gt; udp<BR>&gt; needs =
to be =0A=
communicated to the tx-ing party anyway, and they&nbsp;<BR>&gt; =0A=
generally<BR>&gt; need to share the same socket(or at least appear =0A=
to).<BR>&gt;<BR>&gt; Otherwise I'd pick an unused RTP payload type value =
and =0A=
define it as<BR>&gt; reserved for NAT keepalive usage. I guess this most =
closely =0A=
maps to&nbsp;<BR>&gt; 4.5.<BR>&gt;<BR>&gt; Regards,<BR>&gt;<BR>&gt; Lee =0A=
Dilkie<BR>&gt;<BR>&gt; (we implemented the 0-byte udp in our solution, =
FWIW, =0A=
although we&nbsp;<BR>&gt; choose<BR>&gt; 30 second =
intervals).<BR>&gt;<BR>&gt; =0A=
Xavier Marjou wrote:<BR>&gt;&gt; To keep this topic alive, I plan to =
update the =0A=
following individual<BR>&gt;&gt; draft:<BR>&gt;&gt; <A =0A=
href=3D"http://tools.ietf.org/wg/behave/draft-marjou-behave-app-rtp-">htt=
p://tools.ietf.org/wg/behave/draft-marjou-behave-app-rtp-</A><BR>&gt;&gt;=
 =0A=
keepalive-00.txt<BR>&gt;&gt;<BR>&gt;&gt;<BR>&gt;&gt; I am therefore =
interested =0A=
of any feedback, especially on which<BR>&gt;&gt; mechanism(s) is =0A=
best.<BR>&gt;&gt;<BR>&gt;&gt; Xavier<BR>&gt;&gt;<BR>&gt;&gt;&gt; On =
12/3/06, Dan =0A=
Wing &lt;dwing@cisco.com&gt; wrote:<BR>&gt;&gt;&gt; Minutes of the last =
BEHAVE =0A=
meeting have been posted to:<BR>&gt;&gt;&gt; <A =0A=
href=3D"http://www3.ietf.org/proceedings/06nov/minutes/behave.html">http:=
//www3.ietf.org/proceedings/06nov/minutes/behave.html</A><BR>&gt;&gt;&gt;=
<BR>&gt;&gt;&gt; =0A=
Application Keepalive for NATs, Xavier Marjou -<BR>&gt;&gt;&gt; =0A=
draft-marjou-behave-app-rtp-keepalive-00<BR>&gt;&gt;&gt; status:&nbsp; =
separate =0A=
from ICE, or integrate into Application milestone<BR>&gt;&gt;&gt; =0A=
document?<BR>&gt;&gt;&gt;<BR>&gt;&gt;&gt; JR =96 this is app design =
guidelines. =0A=
How about if we charter to&nbsp;<BR>&gt;&gt;&gt; create<BR>&gt;&gt;&gt; =
that =0A=
document and put this text, along with other text around it =
in<BR>&gt;&gt;&gt; =0A=
STUN all in one document?<BR>&gt;&gt;&gt; Xavier Marjou (speaker, XM) =
=96 take to =0A=
list.<BR>&gt;&gt;&gt; Chair =96 question of which mechanisms might be =
best asked =0A=
in AVT FA =96<BR>&gt;&gt;&gt; good to ask AVT for sure. But we should =
discuss here =0A=
Magnus =96 both<BR>&gt;&gt;&gt; groups working on it Chair =96 consensus =
to fold =0A=
this into application<BR>&gt;&gt;&gt; document, per JR's =0A=
proposal.<BR>&gt;&gt;&gt; Will figure out what to do on =
list.<BR>&gt;&gt;&gt; FA =0A=
=96 support. Think it's a good idea. But do we have an editor =
for<BR>&gt;&gt;&gt; =0A=
that, will it get done? I don't want to send this to =
dev/null.<BR>&gt;&gt;&gt; =0A=
PM =96 in our BEHAVE app document are we the best place to have =0A=
that<BR>&gt;&gt;&gt; discussion?<BR>&gt;&gt;&gt; chair - don't have a =
clear =0A=
decision on what to do, will work to&nbsp;<BR>&gt;&gt;&gt; =0A=
figure<BR>&gt;&gt;&gt; out where this goes.<BR>&gt;&gt;<BR>&gt;&gt; =0A=
_______________________________________________<BR>&gt;&gt; Audio/Video =0A=
Transport Working Group<BR>&gt;&gt; avt@ietf.org<BR>&gt;&gt; <A =0A=
href=3D"https://www1.ietf.org/mailman/listinfo/avt">https://www1.ietf.org=
/mailman/listinfo/avt</A><BR>&gt;<BR>&gt; =0A=
_______________________________________________<BR>&gt; Audio/Video =
Transport =0A=
Working Group<BR>&gt; avt@ietf.org<BR>&gt; <A =0A=
href=3D"https://www1.ietf.org/mailman/listinfo/avt">https://www1.ietf.org=
/mailman/listinfo/avt</A><BR><BR>--<BR>Colin =0A=
Perkins<BR><A =0A=
href=3D"http://csperkins.org/">http://csperkins.org/</A><BR><BR><BR><BR><=
BR>_______________________________________________<BR>Audio/Video =0A=
Transport Working Group<BR>avt@ietf.org<BR><A =0A=
href=3D"https://www1.ietf.org/mailman/listinfo/avt">https://www1.ietf.org=
/mailman/listinfo/avt</A><BR></FONT></P>=0A=
=0A=
</BODY>=0A=
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Subject: Re: [BEHAVE] RE: [AVT] Update of rtp-keepalive draft
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Yuval Nissan skrev:
> The issue of keepalive is also related to the RTP endpoint - If the remote side stops the voice connection because of some problem during silence, there is no certain way to discover it.

The solution to that is called RTCP. It is there, it also provides other 
features. Use it instead of inventing new mechanisms.

Cheers

Magnus Westerlund

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

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From behave-bounces@ietf.org Wed Jan 17 04:12:44 2007
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Subject: RE: [BEHAVE] RE: [AVT] Update of rtp-keepalive draft
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RTCP is the right solution. However in practice not all devices support
RTCP, and using mechanism like the No-Op (that puts limit on the maximum
time of no data in the connection) can help in these cases.

Regards,
Yuval=20

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]=20
Sent: Wednesday, January 17, 2007 10:10 AM
To: Yuval Nissan
Cc: Colin Perkins; Lee Dilkie; Xavier Marjou; behave@ietf.org;
avt@ietf.org
Subject: Re: [BEHAVE] RE: [AVT] Update of rtp-keepalive draft

Yuval Nissan skrev:
> The issue of keepalive is also related to the RTP endpoint - If the
remote side stops the voice connection because of some problem during
silence, there is no certain way to discover it.

The solution to that is called RTCP. It is there, it also provides other

features. Use it instead of inventing new mechanisms.

Cheers

Magnus Westerlund

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

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From behave-bounces@ietf.org Wed Jan 17 04:19:59 2007
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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [BEHAVE] RE: [AVT] Update of rtp-keepalive draft
Date: Wed, 17 Jan 2007 09:19:47 +0000
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Or you could just implement RTCP. It's not like it's difficult to do  
a minimal implementation...

Colin


On 17 Jan 2007, at 09:12, Yuval Nissan wrote:
> RTCP is the right solution. However in practice not all devices  
> support
> RTCP, and using mechanism like the No-Op (that puts limit on the  
> maximum
> time of no data in the connection) can help in these cases.
>
> Regards,
> Yuval
>
> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Wednesday, January 17, 2007 10:10 AM
> To: Yuval Nissan
> Cc: Colin Perkins; Lee Dilkie; Xavier Marjou; behave@ietf.org;
> avt@ietf.org
> Subject: Re: [BEHAVE] RE: [AVT] Update of rtp-keepalive draft
>
> Yuval Nissan skrev:
>> The issue of keepalive is also related to the RTP endpoint - If the
> remote side stops the voice connection because of some problem during
> silence, there is no certain way to discover it.
>
> The solution to that is called RTCP. It is there, it also provides  
> other
>
> features. Use it instead of inventing new mechanisms.
>
> Cheers
>
> Magnus Westerlund
>
> IETF Transport Area Director & TSVWG Chair
> ----------------------------------------------------------------------
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
> ----------------------------------------------------------------------

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




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From: Lee Dilkie <lee_dilkie@mitel.com>
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	<2EB1317C-B746-460A-BAE4-B107429E807A@csperkins.org>
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It's been a while since I read the specs but doesn't RTCP use different
(+1) ports than RTP? If so then it isn't a solution.

-lee


Colin Perkins wrote:
> Or you could just implement RTCP. It's not like it's difficult to do a
> minimal implementation...
>
> Colin
>
>
> On 17 Jan 2007, at 09:12, Yuval Nissan wrote:
>> RTCP is the right solution. However in practice not all devices support
>> RTCP, and using mechanism like the No-Op (that puts limit on the maximum
>> time of no data in the connection) can help in these cases.
>>
>> Regards,
>> Yuval
>>
>> -----Original Message-----
>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>> Sent: Wednesday, January 17, 2007 10:10 AM
>> To: Yuval Nissan
>> Cc: Colin Perkins; Lee Dilkie; Xavier Marjou; behave@ietf.org;
>> avt@ietf.org
>> Subject: Re: [BEHAVE] RE: [AVT] Update of rtp-keepalive draft
>>
>> Yuval Nissan skrev:
>>> The issue of keepalive is also related to the RTP endpoint - If the
>> remote side stops the voice connection because of some problem during
>> silence, there is no certain way to discover it.
>>
>> The solution to that is called RTCP. It is there, it also provides other
>>
>> features. Use it instead of inventing new mechanisms.
>>
>> Cheers
>>
>> Magnus Westerlund
>>
>> IETF Transport Area Director & TSVWG Chair
>> ----------------------------------------------------------------------
>> Multimedia Technologies, Ericsson Research EAB/TVA/A
>> ----------------------------------------------------------------------
>> Ericsson AB                | Phone +46 8 4048287
>> Torshamsgatan 23           | Fax   +46 8 7575550
>> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>> ----------------------------------------------------------------------
>
> --Colin Perkins
> http://csperkins.org/
>
>

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From behave-bounces@ietf.org Wed Jan 17 10:07:31 2007
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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [BEHAVE] RE: [AVT] Update of rtp-keepalive draft
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Yes, but there is a proposal to allow RTCP on the RTP port for this  
reason. See draft-ietf-avt-rtp-and-rtcp-mux-03.txt.

Colin



On 17 Jan 2007, at 14:58, Lee Dilkie wrote:
> It's been a while since I read the specs but doesn't RTCP use  
> different
> (+1) ports than RTP? If so then it isn't a solution.
>
> -lee
>
>
> Colin Perkins wrote:
>> Or you could just implement RTCP. It's not like it's difficult to  
>> do a
>> minimal implementation...
>>
>> Colin
>>
>>
>> On 17 Jan 2007, at 09:12, Yuval Nissan wrote:
>>> RTCP is the right solution. However in practice not all devices  
>>> support
>>> RTCP, and using mechanism like the No-Op (that puts limit on the  
>>> maximum
>>> time of no data in the connection) can help in these cases.
>>>
>>> Regards,
>>> Yuval
>>>
>>> -----Original Message-----
>>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>>> Sent: Wednesday, January 17, 2007 10:10 AM
>>> To: Yuval Nissan
>>> Cc: Colin Perkins; Lee Dilkie; Xavier Marjou; behave@ietf.org;
>>> avt@ietf.org
>>> Subject: Re: [BEHAVE] RE: [AVT] Update of rtp-keepalive draft
>>>
>>> Yuval Nissan skrev:
>>>> The issue of keepalive is also related to the RTP endpoint - If the
>>> remote side stops the voice connection because of some problem  
>>> during
>>> silence, there is no certain way to discover it.
>>>
>>> The solution to that is called RTCP. It is there, it also  
>>> provides other
>>>
>>> features. Use it instead of inventing new mechanisms.
>>>
>>> Cheers
>>>
>>> Magnus Westerlund
>>>
>>> IETF Transport Area Director & TSVWG Chair
>>> -------------------------------------------------------------------- 
>>> --
>>> Multimedia Technologies, Ericsson Research EAB/TVA/A
>>> -------------------------------------------------------------------- 
>>> --
>>> Ericsson AB                | Phone +46 8 4048287
>>> Torshamsgatan 23           | Fax   +46 8 7575550
>>> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>> -------------------------------------------------------------------- 
>>> --
>>
>> --Colin Perkins
>> http://csperkins.org/
>>
>>

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




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Seems a little heavyweight to solve what is essentially a network layer
problem. I'm not a fan of trying to guess at protocols in realtime based
on what's put in front of me, I worry about the security implications.
For that reason I like the 0-byte udp approach, it cannot be confused
with anything else (RTP or RTCP). As well, the 0-byte technique can be
used for other protocols, like RTCP, to aid them in traversing NAT.

It would be nice if we could use a time machine to go baaaack in order
to fix the future (wasn't that a movie?) but we have what we have I guess.

-lee

Colin Perkins wrote:
> Yes, but there is a proposal to allow RTCP on the RTP port for this
> reason. See draft-ietf-avt-rtp-and-rtcp-mux-03.txt.
>
> Colin
>
>
>
> On 17 Jan 2007, at 14:58, Lee Dilkie wrote:
>> It's been a while since I read the specs but doesn't RTCP use different
>> (+1) ports than RTP? If so then it isn't a solution.
>>
>> -lee
>>
>>
>> Colin Perkins wrote:
>>> Or you could just implement RTCP. It's not like it's difficult to do a
>>> minimal implementation...
>>>
>>> Colin
>>>
>>>
>>> On 17 Jan 2007, at 09:12, Yuval Nissan wrote:
>>>> RTCP is the right solution. However in practice not all devices
>>>> support
>>>> RTCP, and using mechanism like the No-Op (that puts limit on the
>>>> maximum
>>>> time of no data in the connection) can help in these cases.
>>>>
>>>> Regards,
>>>> Yuval
>>>>
>>>> -----Original Message-----
>>>> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
>>>> Sent: Wednesday, January 17, 2007 10:10 AM
>>>> To: Yuval Nissan
>>>> Cc: Colin Perkins; Lee Dilkie; Xavier Marjou; behave@ietf.org;
>>>> avt@ietf.org
>>>> Subject: Re: [BEHAVE] RE: [AVT] Update of rtp-keepalive draft
>>>>
>>>> Yuval Nissan skrev:
>>>>> The issue of keepalive is also related to the RTP endpoint - If the
>>>> remote side stops the voice connection because of some problem during
>>>> silence, there is no certain way to discover it.
>>>>
>>>> The solution to that is called RTCP. It is there, it also provides
>>>> other
>>>>
>>>> features. Use it instead of inventing new mechanisms.
>>>>
>>>> Cheers
>>>>
>>>> Magnus Westerlund
>>>>
>>>> IETF Transport Area Director & TSVWG Chair
>>>> ----------------------------------------------------------------------
>>>> Multimedia Technologies, Ericsson Research EAB/TVA/A
>>>> ----------------------------------------------------------------------
>>>> Ericsson AB                | Phone +46 8 4048287
>>>> Torshamsgatan 23           | Fax   +46 8 7575550
>>>> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>>>> ----------------------------------------------------------------------
>>>
>>> --Colin Perkins
>>> http://csperkins.org/
>>>
>>>
>
> --Colin Perkins
> http://csperkins.org/
>
>
>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt

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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		: NAT Behavioral Requirements for TCP
	Author(s)	: S. Guha, et al.
	Filename	: draft-ietf-behave-tcp-04.txt
	Pages		: 20
	Date		: 2007-1-17
	
This document defines a set of requirements for NATs that handle TCP
   that would allow many applications, such as peer-to-peer applications
   and on-line games, to work consistently.  Developing NATs that meet
   this set of requirements will greatly increase the likelihood that
   these applications will function properly.

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> Do we need to mandate an alternative?
Personally, I would really prefer to only have one alternative
recommended when STUN is not yet available.

When too many alternatives exist for one same thing, this results in
additional complexity in interoperability and also in future
specifications that would rely on that thing.

Xavier

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Subject: [BEHAVE] TURN major issues and proposed fixes
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Hi Folks,

The following issues have all come up from implementors of TURN and
are big enough issues that I wanted to get some list feedback before
incorporating these in a new rev.

1. Allocations, Bindings, Permissions, Timeouts, and the Active Destination.

The current text is still a bit fuzzy and inconsistent about what is a
difference between permission and a binding and what keeps bindings
open. Also, many implementors have expressed confusion between the
behavior of an ordinary NAT binding and a "STUN Relay binding", so I
am proposing we skip the binding terminology in favor of "permission".
Much of the following came from consensus at one of the meetings. I've
come up with the following proposal which fills out a consistent set
of rules:
- Remove the terminology "binding" altogether inside the TURN server.
Only use "binding" when talking about a "real" NAT (ex: between the
TURN client and the TURN server)
- All requests and indications except for initial Allocation requests
are sent in the context of a matching valid allocation, otherwise they
are rejected (requests) or discarded silently (Indications).
- Allocations are explicitly refreshed via subsequent allocation
requests.  Allocation timeout values are on the order of minutes. Send
and Data indications do not refresh Allocations. Allocation timers are
based on the LIFETIME attribute.
- Each allocation can have zero or more permissions
- When allocations are destroyed all related permissions are deleted.
- Permissions are only created by Send Indications over UDP and
Connect Requests over TCP.
- Data sent to a peer (via send Indications and unencapsulated data
sent to the active destination) by the TURN client refresh the
relevant permission. Data to one permission does not refresh other
permissions associated with the same binding.
- The expected timeout interval for all permissions for an allocation
is conveyed from the server to the client using the REFRESH-INTERVAL
attribute in an initial successful Allocate Response.
- The active destination is changed only with the Set Active
Destination Request for an allocation. The active destination is
completely independent of any permissions.

2. SetActiveDestination should not be limited to existing permissions
Current text in some places says that the active destination needs to
be an existing permission. This is actually unnecessary however, and
loosening this restriction is useful.
Take the case where the TURN client receives a SIP INVITE with an
offer, and the TURN client wishes to accept the offer. The TURN client
should be able to set the active destination immediately before
issuing its first Send request (to this destination). Using this order
of operations insures that the TURN client receives all data traffic
unencapsulated. This provides for more consistent bandwidth usage in
this case.

The proposal is to change the current text to completely decouple the
active destination from any permissions.  A permission controls
whether traffic flows.  The active destination controls whether
traffic flowing over a binding is encapsulated of not.

An alternative to this proposal is to create a new permission if no
permission for the active destination exists.  The problem is that
this does not result in sending any data to the peer (as described in
the applicability statement and UNSAF considerations), it is
completely implicit behavior, and the opposite operation (clearing the
active destination) does not delete permissions.

3. SetActiveDestination changing destinations

Currently there are several pages of text devoted to a timer based
state machine that needs to be implemented on both the server and the
client to handle the case of switching the active destination from one
address to another. The problem is that  over UDP, the client cannot
tell which unencapsulated data is from the old active destination vs.
the new active destination.

However, requiring timers and a state machine on the server is a
significant implementation burden which provides no benefit for the
TURN server or its administrative domain.

The proposal is to put this problem completely into the hands of the
client.  Some clients may never wish to change the active destination
for example and will have no need to implement anything special.
Those clients that want to change the active destination over UDP are
required to clear the active destination, wait for an appropriate
interval (5 seconds is suggested), then set the active destination to
the new address.

Specific proposed text for items 2 and 3 and also delete the TIMER-VAL
attribute:

<section title="Set Active Destination Request">
<section title="Client Behavior">
<t>
The Set Active Destination address allows the client to create an
optimized relay function between it and the server. When the server
receives packets from a particular preferred external client, the
server will forward those packets towards the client without
encapsulating them in a Data Indication. Similarly, the client can
send non-STUN packets to the server without encapsulation, and these
are forwarded to the external client. Sending and receiving data in
unencapsulated form is critical for efficiency purposes. One of the
primary use cases for the STUN relay usage is in support of Voice over
IP (VoIP), which uses very small UDP packets to begin with. The extra
overhead of an additional layer of encapsulation is considered
unacceptable.
</t><t>
The Set Active Destination request is used by the client to provide
the identity of this preferred external client. The Set Active
Destination address MAY contain a REMOTE-ADDRESS attribute. This
attribute, when present, provides the address of the preferred
external client to the server. When absent, it clears the value of the
preferred external client. This address has no effect on the existence
of any permissions.
</t><t>
The client MUST NOT send a Set Active Destination request with a
REMOTE-ADDRESS attribute over an unreliable link (ex: UDP) if an
active destination is already set for that allocation. If the client
wishes to set a new active destination, it MUST wait until 5 seconds
after a successful response is received to a Set Destination Request
removing the active destination. Failure to wait could cause the
client to receive and attribute late data forwarded by the STUN relay
server to the wrong peer.
</t><t>
<list><t>Consider the case where the active destination is set, and
the server is relaying packets towards the client. The client knows
the IP address and port where the packets came from - the current
value of the active destination. The client issues a Set Active
Destination Request to change the active destination, and receives a
response. A moment later, a data packet is received, not encapsulated
in a STUN Data Indication. What is the source if this packet? Is it
the active destination that existed prior to the Set Active
Destination request, or the one after? If the transport between the
client and the STUN server is not reliable, there is no way to
know.</t></list>
</t>
</section>

<section title="Server Behavior">
<t>
The Set Active Destination Request is used by a client to set the
forwarding destination of all data that is not encapsulated in STUN
Send Indications. In addition, when a matching permission is present,
all data received from that external client will be forwarded to the
STUN client without being encapsulated in a Data Indication. This
request does not modify any permissions.
</t><t>
The request MUST be authenticated using the same shared secret as the
one associated with the allocation, or be authenticated using a short
term password derived from that shared secret. If the request was
authenticated but not with such a matching credential, the server MUST
generate an error response with a 441 response code.
</t><t>
If the Set Active Destination request does not contain a
REMOTE-ADDRESS attribute, the value of the active destination is
cleared. If the Set Active Destination request contains a
REMOTE-ADDRESS attribute, and the active destination is not set, the
active destination is set to that IP address and port. If an active
destination is already set, and the request was received over a
reliable transport, the active destination is changed to the new
value.  If the active destination is already set and the request was
received over UDP, the Set Active Destination request is rejected with
a 439 Active Destination Already Set error response.  This prevents
the race condition described in the previous section.
</t>
</section>
</section>

4. Opening TCP permissions

Currently TCP to TCP traffic between TURN servers requires the TURN
servers to implement simultaneous open.

SYN ->
<- SYN
SYN-ACK ->
<- ACK

Option A is to leave this as is and to clarify this non-obvious
implication to implementors (you cannot receive an incoming connection
unless you first try to open one with the Connect request.  Option B
is to include an explicit Open Permission Request to create a specific
permission for the TURN server to receive a TCP SYN from one specific
IP address and port number.

I will implement Option A unless there is strong desire on the list to
do something else.

thanks,
-rohan

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Subject: RE: [BEHAVE] TURN major issues and proposed fixes
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=20

> -----Original Message-----
> From: Rohan Mahy [mailto:rohan.mahy@gmail.com]=20
> Sent: Friday, January 19, 2007 10:03 PM
> To: behave@ietf.org
> Cc: Rohan Mahy
> Subject: [BEHAVE] TURN major issues and proposed fixes
>=20
> Hi Folks,
>=20
> The following issues have all come up from implementors of=20
> TURN and are big enough issues that I wanted to get some list=20
> feedback before incorporating these in a new rev.
>=20
> 1. Allocations, Bindings, Permissions, Timeouts, and the=20
> Active Destination.
>=20
> The current text is still a bit fuzzy and inconsistent about=20
> what is a difference between permission and a binding and=20
> what keeps bindings open. Also, many implementors have=20
> expressed confusion between the behavior of an ordinary NAT=20
> binding and a "STUN Relay binding", so I am proposing we skip=20
> the binding terminology in favor of "permission".
> Much of the following came from consensus at one of the=20
> meetings. I've come up with the following proposal which=20
> fills out a consistent set of rules:
> - Remove the terminology "binding" altogether inside the TURN server.
> Only use "binding" when talking about a "real" NAT (ex:=20
> between the TURN client and the TURN server)

Sounds good.

> - All requests and indications except for initial Allocation=20
> requests are sent in the context of a matching valid=20
> allocation, otherwise they are rejected (requests) or=20
> discarded silently (Indications).

Sounds good.

> - Allocations are explicitly refreshed via subsequent=20
> allocation requests.  Allocation timeout values are on the=20
> order of minutes. Send and Data indications do not refresh=20
> Allocations. Allocation timers are based on the LIFETIME attribute.

I'm confused. What's the relation between LIFETIME and REFRESH-INTERVAL
below? If the allocation expires, then the permissions go with it
because the permissions were created in the context of an allocation.

> - Each allocation can have zero or more permissions
> - When allocations are destroyed all related permissions are deleted.
> - Permissions are only created by Send Indications over UDP=20
> and Connect Requests over TCP.

This means that an offerer would have to generate RTP traffic to an
answerer to get RTP traffic sent back to it. What if I want to setup a
one-way stream to me in which it's inappropriate to generate traffic in
the other direction, only to make the TURN server happy? Why not let Set
Active Destination also create a permission?

> - Data sent to a peer (via send Indications and=20
> unencapsulated data sent to the active destination) by the=20
> TURN client refresh the relevant permission. Data to one=20
> permission does not refresh other permissions associated with=20
> the same binding.

Sounds good.

> - The expected timeout interval for all permissions for an=20
> allocation is conveyed from the server to the client using=20
> the REFRESH-INTERVAL attribute in an initial successful=20
> Allocate Response.

How does this relate to LIFETIME (see above)?

> - The active destination is changed only with the Set Active=20
> Destination Request for an allocation. The active destination=20
> is completely independent of any permissions.

There's some text about the Set Active Destination Request creating a
permission in section 9.2 that's been floating about on the list as
being used to allow an originator to tell the TURN server to allow
traffic from that destination. This would make it possible for a
permission to be created without requiring the originator to generate
traffic to the actual intended destination. I think this should be
allowed. Are you proposing changing this?

>=20
> 2. SetActiveDestination should not be limited to existing=20
> permissions Current text in some places says that the active=20
> destination needs to be an existing permission. This is=20
> actually unnecessary however, and loosening this restriction=20
> is useful.
> Take the case where the TURN client receives a SIP INVITE=20
> with an offer, and the TURN client wishes to accept the=20
> offer. The TURN client should be able to set the active=20
> destination immediately before issuing its first Send request=20
> (to this destination). Using this order of operations insures=20
> that the TURN client receives all data traffic=20
> unencapsulated. This provides for more consistent bandwidth=20
> usage in this case.

This does not (to me) appear to mesh with some of the points you've made
above. I thought paragraph 5 on page 19 of the draft made the
expectation pretty clear:

   If the Set Active Destination request contains a REMOTE-ADDRESS
   attribute, the IP address contained within it is added to the
   permissions for this allocation, if it was not already present.

>=20
> The proposal is to change the current text to completely=20
> decouple the active destination from any permissions.  A=20
> permission controls whether traffic flows.  The active=20
> destination controls whether traffic flowing over a binding=20
> is encapsulated of not.
>=20
> An alternative to this proposal is to create a new permission=20
> if no permission for the active destination exists.  The=20
> problem is that this does not result in sending any data to=20
> the peer (as described in the applicability statement and=20
> UNSAF considerations), it is completely implicit behavior,=20
> and the opposite operation (clearing the active destination)=20
> does not delete permissions.

Why not add an operation to explicitly clear the active destination
then?

>=20
> 3. SetActiveDestination changing destinations
>=20
> Currently there are several pages of text devoted to a timer=20
> based state machine that needs to be implemented on both the=20
> server and the client to handle the case of switching the=20
> active destination from one address to another. The problem=20
> is that  over UDP, the client cannot tell which=20
> unencapsulated data is from the old active destination vs.
> the new active destination.
>=20
> However, requiring timers and a state machine on the server=20
> is a significant implementation burden which provides no=20
> benefit for the TURN server or its administrative domain.
>=20
> The proposal is to put this problem completely into the hands=20
> of the client.  Some clients may never wish to change the=20
> active destination for example and will have no need to=20
> implement anything special.
> Those clients that want to change the active destination over=20
> UDP are required to clear the active destination, wait for an=20
> appropriate interval (5 seconds is suggested), then set the=20
> active destination to the new address.
>=20
> Specific proposed text for items 2 and 3 and also delete the TIMER-VAL
> attribute:
>=20
> <section title=3D"Set Active Destination Request"> <section=20
> title=3D"Client Behavior"> <t> The Set Active Destination=20
> address allows the client to create an optimized relay=20
> function between it and the server. When the server receives=20
> packets from a particular preferred external client, the=20
> server will forward those packets towards the client without=20
> encapsulating them in a Data Indication. Similarly, the=20
> client can send non-STUN packets to the server without=20
> encapsulation, and these are forwarded to the external=20
> client. Sending and receiving data in unencapsulated form is=20
> critical for efficiency purposes. One of the primary use=20
> cases for the STUN relay usage is in support of Voice over IP=20
> (VoIP), which uses very small UDP packets to begin with. The=20
> extra overhead of an additional layer of encapsulation is=20
> considered unacceptable.
> </t><t>
> The Set Active Destination request is used by the client to=20
> provide the identity of this preferred external client. The=20
> Set Active Destination address MAY contain a REMOTE-ADDRESS=20
> attribute. This attribute, when present, provides the address=20
> of the preferred external client to the server. When absent,=20
> it clears the value of the preferred external client. This=20
> address has no effect on the existence of any permissions.
> </t><t>
> The client MUST NOT send a Set Active Destination request=20
> with a REMOTE-ADDRESS attribute over an unreliable link (ex:=20
> UDP) if an active destination is already set for that=20
> allocation. If the client wishes to set a new active=20
> destination, it MUST wait until 5 seconds after a successful=20
> response is received to a Set Destination Request removing=20
> the active destination. Failure to wait could cause the=20
> client to receive and attribute late data forwarded by the=20
> STUN relay server to the wrong peer.
> </t><t>
> <list><t>Consider the case where the active destination is=20
> set, and the server is relaying packets towards the client.=20
> The client knows the IP address and port where the packets=20
> came from - the current value of the active destination. The=20
> client issues a Set Active Destination Request to change the=20
> active destination, and receives a response. A moment later,=20
> a data packet is received, not encapsulated in a STUN Data=20
> Indication. What is the source if this packet? Is it the=20
> active destination that existed prior to the Set Active=20
> Destination request, or the one after? If the transport=20
> between the client and the STUN server is not reliable, there=20
> is no way to know.</t></list> </t> </section>
>=20
> <section title=3D"Server Behavior">
> <t>
> The Set Active Destination Request is used by a client to set=20
> the forwarding destination of all data that is not=20
> encapsulated in STUN Send Indications. In addition, when a=20
> matching permission is present, all data received from that=20
> external client will be forwarded to the STUN client without=20
> being encapsulated in a Data Indication. This request does=20
> not modify any permissions.
> </t><t>
> The request MUST be authenticated using the same shared=20
> secret as the one associated with the allocation, or be=20
> authenticated using a short term password derived from that=20
> shared secret. If the request was authenticated but not with=20
> such a matching credential, the server MUST generate an error=20
> response with a 441 response code.
> </t><t>
> If the Set Active Destination request does not contain a=20
> REMOTE-ADDRESS attribute, the value of the active destination=20
> is cleared. If the Set Active Destination request contains a=20
> REMOTE-ADDRESS attribute, and the active destination is not=20
> set, the active destination is set to that IP address and=20
> port. If an active destination is already set, and the=20
> request was received over a reliable transport, the active=20
> destination is changed to the new value.  If the active=20
> destination is already set and the request was received over=20
> UDP, the Set Active Destination request is rejected with a=20
> 439 Active Destination Already Set error response.  This=20
> prevents the race condition described in the previous section.
> </t>
> </section>
> </section>
>=20
> 4. Opening TCP permissions
>=20
> Currently TCP to TCP traffic between TURN servers requires=20
> the TURN servers to implement simultaneous open.
>=20
> SYN ->
> <- SYN
> SYN-ACK ->
> <- ACK
>=20
> Option A is to leave this as is and to clarify this=20
> non-obvious implication to implementors (you cannot receive=20
> an incoming connection unless you first try to open one with=20
> the Connect request.  Option B is to include an explicit Open=20
> Permission Request to create a specific permission for the=20
> TURN server to receive a TCP SYN from one specific IP address=20
> and port number.
>=20
> I will implement Option A unless there is strong desire on=20
> the list to do something else.
>=20
> thanks,
> -rohan
>=20
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
>=20

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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [BEHAVE] TURN major issues and proposed fixes
Date: Tue, 23 Jan 2007 08:24:40 -0800
To: "Brian Stucker" <bstucker@nortel.com>
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Hi Brian,

Thanks for your comments.  My responses are inline.

On Jan 22, 2007, at 16:21, Brian Stucker wrote:
>> -----Original Message-----
>> From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
>> Sent: Friday, January 19, 2007 10:03 PM
>> To: behave@ietf.org
>> Cc: Rohan Mahy
>> Subject: [BEHAVE] TURN major issues and proposed fixes
>>
>> Hi Folks,
>>
>> The following issues have all come up from implementors of
>> TURN and are big enough issues that I wanted to get some list
>> feedback before incorporating these in a new rev.
>>
>> 1. Allocations, Bindings, Permissions, Timeouts, and the
>> Active Destination.
>>
>> The current text is still a bit fuzzy and inconsistent about
>> what is a difference between permission and a binding and
>> what keeps bindings open. Also, many implementors have
>> expressed confusion between the behavior of an ordinary NAT
>> binding and a "STUN Relay binding", so I am proposing we skip
>> the binding terminology in favor of "permission".
>> Much of the following came from consensus at one of the
>> meetings. I've come up with the following proposal which
>> fills out a consistent set of rules:
>> - Remove the terminology "binding" altogether inside the TURN server.
>> Only use "binding" when talking about a "real" NAT (ex:
>> between the TURN client and the TURN server)
>
> Sounds good.
>
>> - All requests and indications except for initial Allocation
>> requests are sent in the context of a matching valid
>> allocation, otherwise they are rejected (requests) or
>> discarded silently (Indications).
>
> Sounds good.
>
>> - Allocations are explicitly refreshed via subsequent
>> allocation requests.  Allocation timeout values are on the
>> order of minutes. Send and Data indications do not refresh
>> Allocations. Allocation timers are based on the LIFETIME attribute.
>
> I'm confused. What's the relation between LIFETIME and REFRESH-INTERVAL
> below? If the allocation expires, then the permissions go with it
> because the permissions were created in the context of an allocation.

I expect that the LIFETIME might be on the order of 15 minutes, and the 
REFRESH-INTERVAL for a UDP permission might be on the order of 30 
seconds.

Under the proposal here, you need to send data to the target of each 
permission at least every 30 seconds to keep the permission alive.  I 
assume this also is likely to have the side effect of keeping NAT 
bindings alive as well.  You separately need to send a (re)Allocate 
Request to the TURN server at least every 15 minutes to refresh the 
allocation.  As you pointed out, if you don't do this, the whole 
allocation goes away and all the related permissions with it.

>> - Each allocation can have zero or more permissions
>> - When allocations are destroyed all related permissions are deleted.
>> - Permissions are only created by Send Indications over UDP
>> and Connect Requests over TCP.
>
> This means that an offerer would have to generate RTP traffic to an
> answerer to get RTP traffic sent back to it. What if I want to setup a
> one-way stream to me in which it's inappropriate to generate traffic in
> the other direction, only to make the TURN server happy? Why not let 
> Set
> Active Destination also create a permission?

This would be a change from the Applicability, Introduction, and UNSAF 
considerations as I read them.  There is a lot of text in those 
sections about how we only create a permission when we have sent data 
to that target.

>> - Data sent to a peer (via send Indications and
>> unencapsulated data sent to the active destination) by the
>> TURN client refresh the relevant permission. Data to one
>> permission does not refresh other permissions associated with
>> the same binding.
>
> Sounds good.
>
>> - The expected timeout interval for all permissions for an
>> allocation is conveyed from the server to the client using
>> the REFRESH-INTERVAL attribute in an initial successful
>> Allocate Response.
>
> How does this relate to LIFETIME (see above)?
>
>> - The active destination is changed only with the Set Active
>> Destination Request for an allocation. The active destination
>> is completely independent of any permissions.
>
> There's some text about the Set Active Destination Request creating a
> permission in section 9.2 that's been floating about on the list as
> being used to allow an originator to tell the TURN server to allow
> traffic from that destination. This would make it possible for a
> permission to be created without requiring the originator to generate
> traffic to the actual intended destination. I think this should be
> allowed. Are you proposing changing this?

can you paste that into this thread please?

>> 2. SetActiveDestination should not be limited to existing
>> permissions Current text in some places says that the active
>> destination needs to be an existing permission. This is
>> actually unnecessary however, and loosening this restriction
>> is useful.
>> Take the case where the TURN client receives a SIP INVITE
>> with an offer, and the TURN client wishes to accept the
>> offer. The TURN client should be able to set the active
>> destination immediately before issuing its first Send request
>> (to this destination). Using this order of operations insures
>> that the TURN client receives all data traffic
>> unencapsulated. This provides for more consistent bandwidth
>> usage in this case.
>
> This does not (to me) appear to mesh with some of the points you've 
> made
> above.

Can you be more specific about that?
This proposed change makes the active destination used by the TURN 
server only for deciding when to send unwrapped data and to decide how 
to process unwrapped data it receives.
You can set the active destination to anything you'd like.  IIF traffic 
is received and there is a permission (because you sent traffic there) 
it gets wrapped in a Data indication unless it was received from the 
active destination.

Also, if you set the active destination, there may be no need to wrap 
your first bit of outgoing data in a Send request.


> I thought paragraph 5 on page 19 of the draft made the
> expectation pretty clear:
>
>    If the Set Active Destination request contains a REMOTE-ADDRESS
>    attribute, the IP address contained within it is added to the
>    permissions for this allocation, if it was not already present.

This paragraph is later contradicted a few paragraphs later.  We need 
to change one of the paragraphs.

>> The proposal is to change the current text to completely
>> decouple the active destination from any permissions.  A
>> permission controls whether traffic flows.  The active
>> destination controls whether traffic flowing over a binding
>> is encapsulated of not.
>>
>> An alternative to this proposal is to create a new permission
>> if no permission for the active destination exists.  The
>> problem is that this does not result in sending any data to
>> the peer (as described in the applicability statement and
>> UNSAF considerations), it is completely implicit behavior,
>> and the opposite operation (clearing the active destination)
>> does not delete permissions.
>
> Why not add an operation to explicitly clear the active destination
> then?

We can clear the active destination just fine now, its just that that 
has no effect on permissions.

What we don't have is a way to explicitly delete a permission.  I 
proposed a Close Binding Request verb back in Montreal and there was 
pretty overwhelming reaction to get rid of it.

>> 3. SetActiveDestination changing destinations

Do you have any comments on proposals 3 or 4?

thanks,
-rohan

>> Currently there are several pages of text devoted to a timer
>> based state machine that needs to be implemented on both the
>> server and the client to handle the case of switching the
>> active destination from one address to another. The problem
>> is that  over UDP, the client cannot tell which
>> unencapsulated data is from the old active destination vs.
>> the new active destination.
>>
>> However, requiring timers and a state machine on the server
>> is a significant implementation burden which provides no
>> benefit for the TURN server or its administrative domain.
>>
>> The proposal is to put this problem completely into the hands
>> of the client.  Some clients may never wish to change the
>> active destination for example and will have no need to
>> implement anything special.
>> Those clients that want to change the active destination over
>> UDP are required to clear the active destination, wait for an
>> appropriate interval (5 seconds is suggested), then set the
>> active destination to the new address.
>>
>> Specific proposed text for items 2 and 3 and also delete the TIMER-VAL
>> attribute:
>>
>> <section title="Set Active Destination Request"> <section
>> title="Client Behavior"> <t> The Set Active Destination
>> address allows the client to create an optimized relay
>> function between it and the server. When the server receives
>> packets from a particular preferred external client, the
>> server will forward those packets towards the client without
>> encapsulating them in a Data Indication. Similarly, the
>> client can send non-STUN packets to the server without
>> encapsulation, and these are forwarded to the external
>> client. Sending and receiving data in unencapsulated form is
>> critical for efficiency purposes. One of the primary use
>> cases for the STUN relay usage is in support of Voice over IP
>> (VoIP), which uses very small UDP packets to begin with. The
>> extra overhead of an additional layer of encapsulation is
>> considered unacceptable.
>> </t><t>
>> The Set Active Destination request is used by the client to
>> provide the identity of this preferred external client. The
>> Set Active Destination address MAY contain a REMOTE-ADDRESS
>> attribute. This attribute, when present, provides the address
>> of the preferred external client to the server. When absent,
>> it clears the value of the preferred external client. This
>> address has no effect on the existence of any permissions.
>> </t><t>
>> The client MUST NOT send a Set Active Destination request
>> with a REMOTE-ADDRESS attribute over an unreliable link (ex:
>> UDP) if an active destination is already set for that
>> allocation. If the client wishes to set a new active
>> destination, it MUST wait until 5 seconds after a successful
>> response is received to a Set Destination Request removing
>> the active destination. Failure to wait could cause the
>> client to receive and attribute late data forwarded by the
>> STUN relay server to the wrong peer.
>> </t><t>
>> <list><t>Consider the case where the active destination is
>> set, and the server is relaying packets towards the client.
>> The client knows the IP address and port where the packets
>> came from - the current value of the active destination. The
>> client issues a Set Active Destination Request to change the
>> active destination, and receives a response. A moment later,
>> a data packet is received, not encapsulated in a STUN Data
>> Indication. What is the source if this packet? Is it the
>> active destination that existed prior to the Set Active
>> Destination request, or the one after? If the transport
>> between the client and the STUN server is not reliable, there
>> is no way to know.</t></list> </t> </section>
>>
>> <section title="Server Behavior">
>> <t>
>> The Set Active Destination Request is used by a client to set
>> the forwarding destination of all data that is not
>> encapsulated in STUN Send Indications. In addition, when a
>> matching permission is present, all data received from that
>> external client will be forwarded to the STUN client without
>> being encapsulated in a Data Indication. This request does
>> not modify any permissions.
>> </t><t>
>> The request MUST be authenticated using the same shared
>> secret as the one associated with the allocation, or be
>> authenticated using a short term password derived from that
>> shared secret. If the request was authenticated but not with
>> such a matching credential, the server MUST generate an error
>> response with a 441 response code.
>> </t><t>
>> If the Set Active Destination request does not contain a
>> REMOTE-ADDRESS attribute, the value of the active destination
>> is cleared. If the Set Active Destination request contains a
>> REMOTE-ADDRESS attribute, and the active destination is not
>> set, the active destination is set to that IP address and
>> port. If an active destination is already set, and the
>> request was received over a reliable transport, the active
>> destination is changed to the new value.  If the active
>> destination is already set and the request was received over
>> UDP, the Set Active Destination request is rejected with a
>> 439 Active Destination Already Set error response.  This
>> prevents the race condition described in the previous section.
>> </t>
>> </section>
>> </section>
>>
>> 4. Opening TCP permissions
>>
>> Currently TCP to TCP traffic between TURN servers requires
>> the TURN servers to implement simultaneous open.
>>
>> SYN ->
>> <- SYN
>> SYN-ACK ->
>> <- ACK
>>
>> Option A is to leave this as is and to clarify this
>> non-obvious implication to implementors (you cannot receive
>> an incoming connection unless you first try to open one with
>> the Connect request.  Option B is to include an explicit Open
>> Permission Request to create a specific permission for the
>> TURN server to receive a TCP SYN from one specific IP address
>> and port number.
>>
>> I will implement Option A unless there is strong desire on
>> the list to do something else.
>>
>> thanks,
>> -rohan
>>
>> _______________________________________________
>> Behave mailing list
>> Behave@ietf.org
>> https://www1.ietf.org/mailman/listinfo/behave
>>


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From behave-bounces@ietf.org Tue Jan 23 12:41:35 2007
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Subject: RE: [BEHAVE] TURN major issues and proposed fixes
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Rohan,

Two questions on your proposed changes:

I do have some concern on requiring the client to refresh the
allocation. What is the driver for having the client refresh the
allocation even in the presence of relayed data?

The new text implies that only a single active destination is allowed.
Is my read correct or can the client set multiple active destinations?

Kevin

-----Original Message-----
From: Rohan Mahy [mailto:rohan.mahy@gmail.com]=20
Sent: Friday, January 19, 2007 9:03 PM
To: behave@ietf.org
Cc: Rohan Mahy
Subject: [BEHAVE] TURN major issues and proposed fixes

Hi Folks,

The following issues have all come up from implementors of TURN and
are big enough issues that I wanted to get some list feedback before
incorporating these in a new rev.

1. Allocations, Bindings, Permissions, Timeouts, and the Active
Destination.

The current text is still a bit fuzzy and inconsistent about what is a
difference between permission and a binding and what keeps bindings
open. Also, many implementors have expressed confusion between the
behavior of an ordinary NAT binding and a "STUN Relay binding", so I
am proposing we skip the binding terminology in favor of "permission".
Much of the following came from consensus at one of the meetings. I've
come up with the following proposal which fills out a consistent set
of rules:
- Remove the terminology "binding" altogether inside the TURN server.
Only use "binding" when talking about a "real" NAT (ex: between the
TURN client and the TURN server)
- All requests and indications except for initial Allocation requests
are sent in the context of a matching valid allocation, otherwise they
are rejected (requests) or discarded silently (Indications).
- Allocations are explicitly refreshed via subsequent allocation
requests.  Allocation timeout values are on the order of minutes. Send
and Data indications do not refresh Allocations. Allocation timers are
based on the LIFETIME attribute.
- Each allocation can have zero or more permissions
- When allocations are destroyed all related permissions are deleted.
- Permissions are only created by Send Indications over UDP and
Connect Requests over TCP.
- Data sent to a peer (via send Indications and unencapsulated data
sent to the active destination) by the TURN client refresh the
relevant permission. Data to one permission does not refresh other
permissions associated with the same binding.
- The expected timeout interval for all permissions for an allocation
is conveyed from the server to the client using the REFRESH-INTERVAL
attribute in an initial successful Allocate Response.
- The active destination is changed only with the Set Active
Destination Request for an allocation. The active destination is
completely independent of any permissions.

2. SetActiveDestination should not be limited to existing permissions
Current text in some places says that the active destination needs to
be an existing permission. This is actually unnecessary however, and
loosening this restriction is useful.
Take the case where the TURN client receives a SIP INVITE with an
offer, and the TURN client wishes to accept the offer. The TURN client
should be able to set the active destination immediately before
issuing its first Send request (to this destination). Using this order
of operations insures that the TURN client receives all data traffic
unencapsulated. This provides for more consistent bandwidth usage in
this case.

The proposal is to change the current text to completely decouple the
active destination from any permissions.  A permission controls
whether traffic flows.  The active destination controls whether
traffic flowing over a binding is encapsulated of not.

An alternative to this proposal is to create a new permission if no
permission for the active destination exists.  The problem is that
this does not result in sending any data to the peer (as described in
the applicability statement and UNSAF considerations), it is
completely implicit behavior, and the opposite operation (clearing the
active destination) does not delete permissions.

3. SetActiveDestination changing destinations

Currently there are several pages of text devoted to a timer based
state machine that needs to be implemented on both the server and the
client to handle the case of switching the active destination from one
address to another. The problem is that  over UDP, the client cannot
tell which unencapsulated data is from the old active destination vs.
the new active destination.

However, requiring timers and a state machine on the server is a
significant implementation burden which provides no benefit for the
TURN server or its administrative domain.

The proposal is to put this problem completely into the hands of the
client.  Some clients may never wish to change the active destination
for example and will have no need to implement anything special.
Those clients that want to change the active destination over UDP are
required to clear the active destination, wait for an appropriate
interval (5 seconds is suggested), then set the active destination to
the new address.

Specific proposed text for items 2 and 3 and also delete the TIMER-VAL
attribute:

<section title=3D"Set Active Destination Request">
<section title=3D"Client Behavior">
<t>
The Set Active Destination address allows the client to create an
optimized relay function between it and the server. When the server
receives packets from a particular preferred external client, the
server will forward those packets towards the client without
encapsulating them in a Data Indication. Similarly, the client can
send non-STUN packets to the server without encapsulation, and these
are forwarded to the external client. Sending and receiving data in
unencapsulated form is critical for efficiency purposes. One of the
primary use cases for the STUN relay usage is in support of Voice over
IP (VoIP), which uses very small UDP packets to begin with. The extra
overhead of an additional layer of encapsulation is considered
unacceptable.
</t><t>
The Set Active Destination request is used by the client to provide
the identity of this preferred external client. The Set Active
Destination address MAY contain a REMOTE-ADDRESS attribute. This
attribute, when present, provides the address of the preferred
external client to the server. When absent, it clears the value of the
preferred external client. This address has no effect on the existence
of any permissions.
</t><t>
The client MUST NOT send a Set Active Destination request with a
REMOTE-ADDRESS attribute over an unreliable link (ex: UDP) if an
active destination is already set for that allocation. If the client
wishes to set a new active destination, it MUST wait until 5 seconds
after a successful response is received to a Set Destination Request
removing the active destination. Failure to wait could cause the
client to receive and attribute late data forwarded by the STUN relay
server to the wrong peer.
</t><t>
<list><t>Consider the case where the active destination is set, and
the server is relaying packets towards the client. The client knows
the IP address and port where the packets came from - the current
value of the active destination. The client issues a Set Active
Destination Request to change the active destination, and receives a
response. A moment later, a data packet is received, not encapsulated
in a STUN Data Indication. What is the source if this packet? Is it
the active destination that existed prior to the Set Active
Destination request, or the one after? If the transport between the
client and the STUN server is not reliable, there is no way to
know.</t></list>
</t>
</section>

<section title=3D"Server Behavior">
<t>
The Set Active Destination Request is used by a client to set the
forwarding destination of all data that is not encapsulated in STUN
Send Indications. In addition, when a matching permission is present,
all data received from that external client will be forwarded to the
STUN client without being encapsulated in a Data Indication. This
request does not modify any permissions.
</t><t>
The request MUST be authenticated using the same shared secret as the
one associated with the allocation, or be authenticated using a short
term password derived from that shared secret. If the request was
authenticated but not with such a matching credential, the server MUST
generate an error response with a 441 response code.
</t><t>
If the Set Active Destination request does not contain a
REMOTE-ADDRESS attribute, the value of the active destination is
cleared. If the Set Active Destination request contains a
REMOTE-ADDRESS attribute, and the active destination is not set, the
active destination is set to that IP address and port. If an active
destination is already set, and the request was received over a
reliable transport, the active destination is changed to the new
value.  If the active destination is already set and the request was
received over UDP, the Set Active Destination request is rejected with
a 439 Active Destination Already Set error response.  This prevents
the race condition described in the previous section.
</t>
</section>
</section>

4. Opening TCP permissions

Currently TCP to TCP traffic between TURN servers requires the TURN
servers to implement simultaneous open.

SYN ->
<- SYN
SYN-ACK ->
<- ACK

Option A is to leave this as is and to clarify this non-obvious
implication to implementors (you cannot receive an incoming connection
unless you first try to open one with the Connect request.  Option B
is to include an explicit Open Permission Request to create a specific
permission for the TURN server to receive a TCP SYN from one specific
IP address and port number.

I will implement Option A unless there is strong desire on the list to
do something else.

thanks,
-rohan

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From: Rohan Mahy <rohan@ekabal.com>
Subject: Re: [BEHAVE] TURN major issues and proposed fixes
Date: Tue, 23 Jan 2007 21:12:00 -0800
To: "Kevin Johns" <K.Johns@CableLabs.com>
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Hi Kevin,

On Jan 23, 2007, at 9:34, Kevin Johns wrote:
> Rohan,
>
> Two questions on your proposed changes:
>
> I do have some concern on requiring the client to refresh the
> allocation. What is the driver for having the client refresh the
> allocation even in the presence of relayed data?

A few come to mind:
1. The WG came to consensus on general principal that an explicit 
reallocation mechanism was preferable to an implicit one.
2. Keeping timers for permissions and allocations conjoined is messy.  
What happens after all the permissions time out?  That MUST NOT cause 
the allocation to suddenly expire.  To get sane behavior, the TURN 
server would need to start the allocation timer after all the 
permissions went away.  This seems overly complicated on the server 
side.
3. If a TURN client has an open allocation, and there is no explicit 
reallocation required, another host that spoofs the client's source IP 
address can send traffic through the TURN server toward an established 
peer indefinitely.  If reallocation is required, the attacker would be 
limited to a fixed period of time unless the attacker had the client's 
TURN credentials.

> The new text implies that only a single active destination is allowed.
> Is my read correct or can the client set multiple active destinations?

By definition, there can only be one active destination.  Note that the 
active destination does not mean it is the only destination for which 
traffic can be flowing. The active destination means the only host for 
which end-to-end data is not wrapped in Send and Data indications.

thanks,
-rohan

> Kevin
>
> -----Original Message-----
> From: Rohan Mahy [mailto:rohan.mahy@gmail.com]
> Sent: Friday, January 19, 2007 9:03 PM
> To: behave@ietf.org
> Cc: Rohan Mahy
> Subject: [BEHAVE] TURN major issues and proposed fixes
>
> Hi Folks,
>
> The following issues have all come up from implementors of TURN and
> are big enough issues that I wanted to get some list feedback before
> incorporating these in a new rev.
>
> 1. Allocations, Bindings, Permissions, Timeouts, and the Active
> Destination.
>
> The current text is still a bit fuzzy and inconsistent about what is a
> difference between permission and a binding and what keeps bindings
> open. Also, many implementors have expressed confusion between the
> behavior of an ordinary NAT binding and a "STUN Relay binding", so I
> am proposing we skip the binding terminology in favor of "permission".
> Much of the following came from consensus at one of the meetings. I've
> come up with the following proposal which fills out a consistent set
> of rules:
> - Remove the terminology "binding" altogether inside the TURN server.
> Only use "binding" when talking about a "real" NAT (ex: between the
> TURN client and the TURN server)
> - All requests and indications except for initial Allocation requests
> are sent in the context of a matching valid allocation, otherwise they
> are rejected (requests) or discarded silently (Indications).
> - Allocations are explicitly refreshed via subsequent allocation
> requests.  Allocation timeout values are on the order of minutes. Send
> and Data indications do not refresh Allocations. Allocation timers are
> based on the LIFETIME attribute.
> - Each allocation can have zero or more permissions
> - When allocations are destroyed all related permissions are deleted.
> - Permissions are only created by Send Indications over UDP and
> Connect Requests over TCP.
> - Data sent to a peer (via send Indications and unencapsulated data
> sent to the active destination) by the TURN client refresh the
> relevant permission. Data to one permission does not refresh other
> permissions associated with the same binding.
> - The expected timeout interval for all permissions for an allocation
> is conveyed from the server to the client using the REFRESH-INTERVAL
> attribute in an initial successful Allocate Response.
> - The active destination is changed only with the Set Active
> Destination Request for an allocation. The active destination is
> completely independent of any permissions.
>
> 2. SetActiveDestination should not be limited to existing permissions
> Current text in some places says that the active destination needs to
> be an existing permission. This is actually unnecessary however, and
> loosening this restriction is useful.
> Take the case where the TURN client receives a SIP INVITE with an
> offer, and the TURN client wishes to accept the offer. The TURN client
> should be able to set the active destination immediately before
> issuing its first Send request (to this destination). Using this order
> of operations insures that the TURN client receives all data traffic
> unencapsulated. This provides for more consistent bandwidth usage in
> this case.
>
> The proposal is to change the current text to completely decouple the
> active destination from any permissions.  A permission controls
> whether traffic flows.  The active destination controls whether
> traffic flowing over a binding is encapsulated of not.
>
> An alternative to this proposal is to create a new permission if no
> permission for the active destination exists.  The problem is that
> this does not result in sending any data to the peer (as described in
> the applicability statement and UNSAF considerations), it is
> completely implicit behavior, and the opposite operation (clearing the
> active destination) does not delete permissions.
>
> 3. SetActiveDestination changing destinations
>
> Currently there are several pages of text devoted to a timer based
> state machine that needs to be implemented on both the server and the
> client to handle the case of switching the active destination from one
> address to another. The problem is that  over UDP, the client cannot
> tell which unencapsulated data is from the old active destination vs.
> the new active destination.
>
> However, requiring timers and a state machine on the server is a
> significant implementation burden which provides no benefit for the
> TURN server or its administrative domain.
>
> The proposal is to put this problem completely into the hands of the
> client.  Some clients may never wish to change the active destination
> for example and will have no need to implement anything special.
> Those clients that want to change the active destination over UDP are
> required to clear the active destination, wait for an appropriate
> interval (5 seconds is suggested), then set the active destination to
> the new address.
>
> Specific proposed text for items 2 and 3 and also delete the TIMER-VAL
> attribute:
>
> <section title="Set Active Destination Request">
> <section title="Client Behavior">
> <t>
> The Set Active Destination address allows the client to create an
> optimized relay function between it and the server. When the server
> receives packets from a particular preferred external client, the
> server will forward those packets towards the client without
> encapsulating them in a Data Indication. Similarly, the client can
> send non-STUN packets to the server without encapsulation, and these
> are forwarded to the external client. Sending and receiving data in
> unencapsulated form is critical for efficiency purposes. One of the
> primary use cases for the STUN relay usage is in support of Voice over
> IP (VoIP), which uses very small UDP packets to begin with. The extra
> overhead of an additional layer of encapsulation is considered
> unacceptable.
> </t><t>
> The Set Active Destination request is used by the client to provide
> the identity of this preferred external client. The Set Active
> Destination address MAY contain a REMOTE-ADDRESS attribute. This
> attribute, when present, provides the address of the preferred
> external client to the server. When absent, it clears the value of the
> preferred external client. This address has no effect on the existence
> of any permissions.
> </t><t>
> The client MUST NOT send a Set Active Destination request with a
> REMOTE-ADDRESS attribute over an unreliable link (ex: UDP) if an
> active destination is already set for that allocation. If the client
> wishes to set a new active destination, it MUST wait until 5 seconds
> after a successful response is received to a Set Destination Request
> removing the active destination. Failure to wait could cause the
> client to receive and attribute late data forwarded by the STUN relay
> server to the wrong peer.
> </t><t>
> <list><t>Consider the case where the active destination is set, and
> the server is relaying packets towards the client. The client knows
> the IP address and port where the packets came from - the current
> value of the active destination. The client issues a Set Active
> Destination Request to change the active destination, and receives a
> response. A moment later, a data packet is received, not encapsulated
> in a STUN Data Indication. What is the source if this packet? Is it
> the active destination that existed prior to the Set Active
> Destination request, or the one after? If the transport between the
> client and the STUN server is not reliable, there is no way to
> know.</t></list>
> </t>
> </section>
>
> <section title="Server Behavior">
> <t>
> The Set Active Destination Request is used by a client to set the
> forwarding destination of all data that is not encapsulated in STUN
> Send Indications. In addition, when a matching permission is present,
> all data received from that external client will be forwarded to the
> STUN client without being encapsulated in a Data Indication. This
> request does not modify any permissions.
> </t><t>
> The request MUST be authenticated using the same shared secret as the
> one associated with the allocation, or be authenticated using a short
> term password derived from that shared secret. If the request was
> authenticated but not with such a matching credential, the server MUST
> generate an error response with a 441 response code.
> </t><t>
> If the Set Active Destination request does not contain a
> REMOTE-ADDRESS attribute, the value of the active destination is
> cleared. If the Set Active Destination request contains a
> REMOTE-ADDRESS attribute, and the active destination is not set, the
> active destination is set to that IP address and port. If an active
> destination is already set, and the request was received over a
> reliable transport, the active destination is changed to the new
> value.  If the active destination is already set and the request was
> received over UDP, the Set Active Destination request is rejected with
> a 439 Active Destination Already Set error response.  This prevents
> the race condition described in the previous section.
> </t>
> </section>
> </section>
>
> 4. Opening TCP permissions
>
> Currently TCP to TCP traffic between TURN servers requires the TURN
> servers to implement simultaneous open.
>
> SYN ->
> <- SYN
> SYN-ACK ->
> <- ACK
>
> Option A is to leave this as is and to clarify this non-obvious
> implication to implementors (you cannot receive an incoming connection
> unless you first try to open one with the Connect request.  Option B
> is to include an explicit Open Permission Request to create a specific
> permission for the TURN server to receive a TCP SYN from one specific
> IP address and port number.
>
> I will implement Option A unless there is strong desire on the list to
> do something else.
>
> thanks,
> -rohan
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave


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BEHAVE, TCPM:

We are starting a two-week working group last call in the BEHAVE working
group for a TCP-related document, draft-ietf-behave-tcp-04, which describes
how to NAT TCP traffic.  This document is expected to become a BCP.

http://www.ietf.org/internet-drafts/draft-ietf-behave-tcp-04.txt

     Abstract:
     This document defines a set of requirements for NATs that
     handle TCP that would allow many applications, such as
     peer-to-peer applications and on-line games, to work
     consistently.  Developing NATs that meet this set of
     requirements will greatly increase the likelihood that
     these applications will function properly.
    
Please send comments to the authors (CC'd) or behave@ietf.org.

WGLC on this document ends on Wednesday, 7-Feb-2007.

-Dan Wing
 chair, BEHAVE working group

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Subject: [BEHAVE] Difference between STUN Binding Discovery and NAT Keepalive
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Hi


I am trying to understand the difference between the two STUN usages
defined in rfc3489-bis05, in sections 12.1 and 12.2, Binding Discovery
and NAT Keepalive. My current understanding of the two is as follows:


* Binding Discovery:
   - Used to get publicly mapped address:port
   - Used for both SIP and RTP/RTCP
   - Backwards compatible with RFC3489
   - authentication and message integrity is RECOMMENDED
   - requests SHOULD be challenged

* NAT Keepalive
   - periodic keepalive on same flow
   - detection of flow failure
   - multiplexed sockets with application
   - client SHOULD NOT use message integrity
   - server SHOULD not authenticate
   - used by outbound draft for flow keepalive


this leads to the following question:

   How can a STUN server providing both usages differentiate between the
   two usages, when both send a STUN Binding Request?


If we assume that for a particular deployed domain, there are two STUN
servers, one multiplexed with SIP on port 5060 and one standalone STUN
server on port 3478. In this case the two servers know which usage they
implement and can respond accordingly. However, the information from the
Binding Discovery STUN server on port 3478 is totally useless if the UA
is behind symmetrical NAT.. For a UA behind NAT to retrieve its publicly
mapped address, with respect to the SIP server, the SIP and STUN traffic
must follow exactly the same flow from UA via NAT to SIP/STUN server.


 From rfc3489-bis05:


12.2.7.  Client Procedures

    If the STUN Response indicates the client's mapped address has
    changed from the client's expected mapped address, the client SHOULD
    inform other applications of its new mapped address.  For example, a
    SIP client could use the binding discovery usage to obtain a new
    mapped address, and then register it using SIP registration
    procedures.


This scenario will not work for symmetrical NATs.


Typically in my UA one usage will be enough, periodically sending STUN
Binding Request messages to the SIP server, to get the publicly mapped
address:port AND keep the flow alive. If the SIP server I am currently
talking to does not have the "sip-stun" option tag, I could always
locate the STUN server for that domain using DNS SRV, but that will not
help if my UA is behind symmetrical NAT..


hopefully all my questions made sense..



/alfred

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From: Vikas <pjain01@gmail.com>
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Subject: Re: [BEHAVE] Difference between STUN Binding Discovery and NAT
	Keepalive
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On 28/01/07, Alfred E. Heggestad <aeh@db.org> wrote:
>
> Hi
>
>
> I am trying to understand the difference between the two STUN usages
> defined in rfc3489-bis05, in sections 12.1 and 12.2, Binding Discovery
> and NAT Keepalive. My current understanding of the two is as follows:
>
>
> * Binding Discovery:
>    - Used to get publicly mapped address:port
>    - Used for both SIP and RTP/RTCP
>    - Backwards compatible with RFC3489
>    - authentication and message integrity is RECOMMENDED
>    - requests SHOULD be challenged
>
> * NAT Keepalive
>    - periodic keepalive on same flow
>    - detection of flow failure
>    - multiplexed sockets with application
>    - client SHOULD NOT use message integrity
>    - server SHOULD not authenticate
>    - used by outbound draft for flow keepalive
>
>
> this leads to the following question:
>
>    How can a STUN server providing both usages differentiate between the
>    two usages, when both send a STUN Binding Request?


>>>>Before authentication it can see the request type and can
challenge if it is a binding request(see section 6 - A class of 0 is a
   Request, a class of 1 is an indication, a class of 2 is a success
response, and a class of 3 is an error response).



If we assume that for a particular deployed domain, there are two STUN
> servers, one multiplexed with SIP on port 5060 and one standalone STUN
> server on port 3478. In this case the two servers know which usage they
> implement and can respond accordingly. However, the information from the
> Binding Discovery STUN server on port 3478 is totally useless if the UA
> is behind symmetrical NAT.. For a UA behind NAT to retrieve its publicly
> mapped address, with respect to the SIP server, the SIP and STUN traffic
> must follow exactly the same flow from UA via NAT to SIP/STUN server.
>
>
> From rfc3489-bis05:
>
>
> 12.2.7.  Client Procedures
>
>     If the STUN Response indicates the client's mapped address has
>     changed from the client's expected mapped address, the client SHOULD
>     inform other applications of its new mapped address.  For example, a
>     SIP client could use the binding discovery usage to obtain a new
>     mapped address, and then register it using SIP registration
>     procedures.
>
>
> This scenario will not work for symmetrical NATs.
>
> >>>>>>This functionality can't be achieved only by these two STUN usage.You may have to implement ICE for that.(see section 1- It will allow incoming UDP
>    packets through NAT, but only through a subset of existing NAT types.
>    In particular, the STUN binding usage by itself does not enable
>    incoming UDP packets through NATs whose mapping property is address
>    dependent or address and port dependent [14])
>
>
>
> Typically in my UA one usage will be enough, periodically sending STUN
> Binding Request messages to the SIP server, to get the publicly mapped
> address:port AND keep the flow alive. If the SIP server I am currently
> talking to does not have the "sip-stun" option tag, I could always
> locate the STUN server for that domain using DNS SRV, but that will not
> help if my UA is behind symmetrical NAT..
>
>
> hopefully all my questions made sense..
>
>
>
> /alfred
>
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave
>

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<br><br><div><span class="gmail_quote">On 28/01/07, <b class="gmail_sendername">Alfred E. Heggestad</b> &lt;<a href="mailto:aeh@db.org">aeh@db.org</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;">
Hi<br><br><br>I am trying to understand the difference between the two STUN usages<br>defined in rfc3489-bis05, in sections 12.1 and 12.2, Binding Discovery<br>and NAT Keepalive. My current understanding of the two is as follows:
<br><br><br>* Binding Discovery:<br>&nbsp;&nbsp; - Used to get publicly mapped address:port<br>&nbsp;&nbsp; - Used for both SIP and RTP/RTCP<br>&nbsp;&nbsp; - Backwards compatible with RFC3489<br>&nbsp;&nbsp; - authentication and message integrity is RECOMMENDED
<br>&nbsp;&nbsp; - requests SHOULD be challenged<br><br>* NAT Keepalive<br>&nbsp;&nbsp; - periodic keepalive on same flow<br>&nbsp;&nbsp; - detection of flow failure<br>&nbsp;&nbsp; - multiplexed sockets with application<br>&nbsp;&nbsp; - client SHOULD NOT use message integrity
<br>&nbsp;&nbsp; - server SHOULD not authenticate<br>&nbsp;&nbsp; - used by outbound draft for flow keepalive<br><br><br>this leads to the following question:<br><br>&nbsp;&nbsp; How can a STUN server providing both usages differentiate between the<br>
&nbsp;&nbsp; two usages, when both send a STUN Binding Request?</blockquote><div><br><pre>&gt;&gt;&gt;&gt;Before authentication it can see the request type and can challenge if it is a binding request(see section 6 - A class of 0 is a
<br>   Request, a class of 1 is an indication, a class of 2 is a success response, and a class of 3 is an error response).<br></pre>
&nbsp;</div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">If we assume that for a particular deployed domain, there are two STUN<br>servers, one multiplexed with SIP on port 5060 and one standalone STUN
<br>server on port 3478. In this case the two servers know which usage they<br>implement and can respond accordingly. However, the information from the<br>Binding Discovery STUN server on port 3478 is totally useless if the UA
<br>is behind symmetrical NAT.. For a UA behind NAT to retrieve its publicly<br>mapped address, with respect to the SIP server, the SIP and STUN traffic<br>must follow exactly the same flow from UA via NAT to SIP/STUN server.
<br><br><br> From rfc3489-bis05:<br><br><br>12.2.7.&nbsp;&nbsp;Client Procedures<br><br>&nbsp;&nbsp;&nbsp;&nbsp;If the STUN Response indicates the client&#39;s mapped address has<br>&nbsp;&nbsp;&nbsp;&nbsp;changed from the client&#39;s expected mapped address, the client SHOULD
<br>&nbsp;&nbsp;&nbsp;&nbsp;inform other applications of its new mapped address.&nbsp;&nbsp;For example, a<br>&nbsp;&nbsp;&nbsp;&nbsp;SIP client could use the binding discovery usage to obtain a new<br>&nbsp;&nbsp;&nbsp;&nbsp;mapped address, and then register it using SIP registration<br>&nbsp;&nbsp;&nbsp;&nbsp;procedures.
<br><br><br>This scenario will not work for symmetrical NATs.<br><pre>&gt;&gt;&gt;&gt;&gt;&gt;This functionality can&#39;t be achieved only by these two STUN usage.You may have to implement ICE for that.(see section 1- It will allow incoming UDP
<br>   packets through NAT, but only through a subset of existing NAT types.<br>   In particular, the STUN binding usage by itself does not enable<br>   incoming UDP packets through NATs whose mapping property is address<br>
   dependent or address and port dependent [14])<br></pre>
<br><br>Typically in my UA one usage will be enough, periodically sending STUN<br>Binding Request messages to the SIP server, to get the publicly mapped<br>address:port AND keep the flow alive. If the SIP server I am currently
<br>talking to does not have the &quot;sip-stun&quot; option tag, I could always<br>locate the STUN server for that domain using DNS SRV, but that will not<br>help if my UA is behind symmetrical NAT..<br><br><br>hopefully all my questions made sense..
<br><br><br><br>/alfred<br><br>_______________________________________________<br>Behave mailing list<br><a href="mailto:Behave@ietf.org">Behave@ietf.org</a><br><a href="https://www1.ietf.org/mailman/listinfo/behave">https://www1.ietf.org/mailman/listinfo/behave
</a><br></blockquote></div><br>

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From: "Dan Wing" <dwing@cisco.com>
To: "'Alfred E. Heggestad'" <aeh@db.org>, <behave@ietf.org>
Subject: RE: [BEHAVE] Difference between STUN Binding Discovery and NAT
	Keepalive
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> I am trying to understand the difference between the two STUN usages
> defined in rfc3489-bis05, in sections 12.1 and 12.2, Binding Discovery
> and NAT Keepalive. My current understanding of the two is as follows:
> 
> 
> * Binding Discovery:
>    - Used to get publicly mapped address:port
>    - Used for both SIP and RTP/RTCP
>    - Backwards compatible with RFC3489
>    - authentication and message integrity is RECOMMENDED
>    - requests SHOULD be challenged

The last two bullets provide protection against an attacker that 
is on the path between your STUN client and your STUN server.

> * NAT Keepalive
>    - periodic keepalive on same flow
>    - detection of flow failure
>    - multiplexed sockets with application
>    - client SHOULD NOT use message integrity
>    - server SHOULD not authenticate
>    - used by outbound draft for flow keepalive

One nuance with NAT Keepalive is that in the most recent version
of ICE (draft-ietf-mmusic-ice-13.txt), the NAT keepalive used
in ICE was changed from a Binding Discovery / Binding Response
exchange to an Indication.  The Indication is a unidirectional
STUN message, which has no response.

draft-ietf-behave-rfc3489bis has not yet been updated to align
with this change to draft-ietf-mmusic-ice-13.

> this leads to the following question:
> 
>    How can a STUN server providing both usages differentiate 
> between the two usages, when both send a STUN Binding Request?

The STUN server listening on UDP/5060 will only receive STUN
messages for the Keepalive Usage.

> If we assume that for a particular deployed domain, there are two
> STUN servers, one multiplexed with SIP on port 5060 and one
> standalone STUN server on port 3478. In this case the two servers
> know which usage they implement and can respond accordingly.
> However, the information from the Binding Discovery STUN server on
> port 3478 is totally useless if the UA is behind symmetrical NAT..

It's not totally useless -- even if you're behind a symmetric NAT
you can still establish bi-directional UDP communication with 
another endpoint if that other endpoint knows its transport 
address.  

However, if that other endpoint is _also_ behind a symmetric NAT, 
you a media relay (TURN server or SBC) is necessary.

ICE describes the procedures to do this.

> For a UA behind NAT to retrieve its publicly mapped address, with
> respect to the SIP server, the SIP and STUN traffic must follow
> exactly the same flow from UA via NAT to SIP/STUN server.

A SIP client, which wants to learn its public transport address
for SIP traffic, doesn't need to talk to the STUN server 
listening on UDP/3478 -- it only needs to talk to the STUN
server embedded with its SIP proxy on UDP/5060.

Are you concerned about the different SHOULDs around 
message integrity for the Binding Discovery usage?

>  From rfc3489-bis05:
> 
> 
> 12.2.7.  Client Procedures
> 
>     If the STUN Response indicates the client's mapped address has
>     changed from the client's expected mapped address, the 
> client SHOULD
>     inform other applications of its new mapped address.  For 
> example, a
>     SIP client could use the binding discovery usage to obtain a new
>     mapped address, and then register it using SIP registration
>     procedures.
> 
> 
> This scenario will not work for symmetrical NATs.

It will work if the STUN Binding Request is sent to the same
port as all SIP traffic (UDP/5060), and demultiplexed by the
SIP proxy.

> Typically in my UA one usage will be enough, periodically sending STUN
> Binding Request messages to the SIP server, to get the publicly mapped
> address:port AND keep the flow alive.

I just searched draft-ietf-sip-outbound-07, and can't find where
it is necessary for the SIP UA to learn its public IP address and
UDP port.  That is, other than learning that it *changed* (which
it can learn via the periodic keepalives), it seems that 
draft-ietf-sip-outbound-07 doesn't say the SIP UA needs to
learn its public transport address using STUN.   It's not
necessary, anyway -- the SIP proxy knows the transport address
of an incoming UDP packet when the SIP proxy receives the packet.

> If the SIP server I am currently
> talking to does not have the "sip-stun" option tag, I could always
> locate the STUN server for that domain using DNS SRV, but 
> that will not help if my UA is behind symmetrical NAT..

-d

> hopefully all my questions made sense..
> 
> 
> 
> /alfred
> 
> _______________________________________________
> Behave mailing list
> Behave@ietf.org
> https://www1.ietf.org/mailman/listinfo/behave

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From behave-bounces@ietf.org Wed Jan 31 17:40:26 2007
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Subject: Re: [BEHAVE] Difference between STUN Binding Discovery and NAT
	Keepalive
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Hi Dan,

thanks for your reply, see inline.

Dan Wing wrote:
>> I am trying to understand the difference between the two STUN usages
>> defined in rfc3489-bis05, in sections 12.1 and 12.2, Binding Discovery
>> and NAT Keepalive. My current understanding of the two is as follows:
>>
>>
>> * Binding Discovery:
>>    - Used to get publicly mapped address:port
>>    - Used for both SIP and RTP/RTCP
>>    - Backwards compatible with RFC3489
>>    - authentication and message integrity is RECOMMENDED
>>    - requests SHOULD be challenged
> 
> The last two bullets provide protection against an attacker that 
> is on the path between your STUN client and your STUN server.
> 
>> * NAT Keepalive
>>    - periodic keepalive on same flow
>>    - detection of flow failure
>>    - multiplexed sockets with application
>>    - client SHOULD NOT use message integrity
>>    - server SHOULD not authenticate
>>    - used by outbound draft for flow keepalive
> 
> One nuance with NAT Keepalive is that in the most recent version
> of ICE (draft-ietf-mmusic-ice-13.txt), the NAT keepalive used
> in ICE was changed from a Binding Discovery / Binding Response
> exchange to an Indication.  The Indication is a unidirectional
> STUN message, which has no response.
> 
> draft-ietf-behave-rfc3489bis has not yet been updated to align
> with this change to draft-ietf-mmusic-ice-13.
> 

ah, ok. is unidirectional STUN indication (sent from UA) guaranteed
to work with all types of NAT boxes ? is it not so that certain
NATs assume bidirectional traffic to keep the flow open?

will STUN Binding Indication be a new STUN message in rfc3489bis-06 ?


>> this leads to the following question:
>>
>>    How can a STUN server providing both usages differentiate 
>> between the two usages, when both send a STUN Binding Request?
> 
> The STUN server listening on UDP/5060 will only receive STUN
> messages for the Keepalive Usage.
> 

ok fair enough.


>> If we assume that for a particular deployed domain, there are two
>> STUN servers, one multiplexed with SIP on port 5060 and one
>> standalone STUN server on port 3478. In this case the two servers
>> know which usage they implement and can respond accordingly.
>> However, the information from the Binding Discovery STUN server on
>> port 3478 is totally useless if the UA is behind symmetrical NAT..
> 
> It's not totally useless -- even if you're behind a symmetric NAT
> you can still establish bi-directional UDP communication with 
> another endpoint if that other endpoint knows its transport 
> address.  
> 
> However, if that other endpoint is _also_ behind a symmetric NAT, 
> you a media relay (TURN server or SBC) is necessary.
> 
> ICE describes the procedures to do this.
> 

OK maybe my question was a bit vague. I am aware of ICE/TURN and
how the media/NAT can be solved by that.

I was mainly asking STUN-related questions, with a view from the
outbound draft ;)


>> For a UA behind NAT to retrieve its publicly mapped address, with
>> respect to the SIP server, the SIP and STUN traffic must follow
>> exactly the same flow from UA via NAT to SIP/STUN server.
> 
> A SIP client, which wants to learn its public transport address
> for SIP traffic, doesn't need to talk to the STUN server 
> listening on UDP/3478 -- it only needs to talk to the STUN
> server embedded with its SIP proxy on UDP/5060.
> 
> Are you concerned about the different SHOULDs around 
> message integrity for the Binding Discovery usage?
> 

Yes, agreed. To get my public SIP address I only need to send
a STUN Binding Request to my SIP Proxy at UDP/5060.

I am not concerned about the message integrity, I am mainly
trying to understand why Binding Discover and NAT Keepalive
usage cant be just one usage (i.e. the same) instead of two
usages that are protocol-wise very similar ..


>>  From rfc3489-bis05:
>>
>>
>> 12.2.7.  Client Procedures
>>
>>     If the STUN Response indicates the client's mapped address has
>>     changed from the client's expected mapped address, the 
>> client SHOULD
>>     inform other applications of its new mapped address.  For 
>> example, a
>>     SIP client could use the binding discovery usage to obtain a new
>>     mapped address, and then register it using SIP registration
>>     procedures.
>>
>>
>> This scenario will not work for symmetrical NATs.
> 
> It will work if the STUN Binding Request is sent to the same
> port as all SIP traffic (UDP/5060), and demultiplexed by the
> SIP proxy.
> 

yes. so the new section in rfc3489-bis05 12.2 NAT Keepalives is
mainly a tool for outbound, and should go to UDP/5060 ?

but section 12.2.7 says "...SIP client could use the *binding discovery* usage
to obtain..." - why recommend that the client has to send a new STUN Binding
Request, when the client at the same to has just got the new publicly mapped
address:port from the current NAT Keepalive Binding Response ?


>> Typically in my UA one usage will be enough, periodically sending STUN
>> Binding Request messages to the SIP server, to get the publicly mapped
>> address:port AND keep the flow alive.
> 
> I just searched draft-ietf-sip-outbound-07, and can't find where
> it is necessary for the SIP UA to learn its public IP address and
> UDP port.  That is, other than learning that it *changed* (which
> it can learn via the periodic keepalives), it seems that 
> draft-ietf-sip-outbound-07 doesn't say the SIP UA needs to
> learn its public transport address using STUN.   It's not
> necessary, anyway -- the SIP proxy knows the transport address
> of an incoming UDP packet when the SIP proxy receives the packet.
> 

ok, it was my assumption that the publicly mapped address from
the STUN Binding Response was used to updated the Via: and Contact:
headers of the SIP UA. But you are right; the SIP proxy knows
where the SIP message came from.

should we ask the outbound people if this could be clarified?


so can we conclude with the following statement (simplified):

   port 3478: STUN Binding Discovery
   port 5060: SIP and STUN Keepalive


many thanks for your clarifications, I think things are bit
clearer now ..


/alfred


<snip>


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Subject: [BEHAVE] BCP 127 RFC 4787 on Network Address Translation (NAT)
	Behavioral Requirements for Unicast UDP
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A new Request for Comments is now available in online RFC libraries.

        BCP 127        
        RFC 4787

        Title:      Network Address Translation (NAT) Behavioral 
                    Requirements for Unicast UDP 
        Author:     F. Audet, Ed.,
                    C. Jennings
        Status:     Best Current Practice
        Date:       January 2007
        Mailbox:    audet@nortel.com, 
                    fluffy@cisco.com
        Pages:      29
        Characters: 68693
        Updates:    
        See-Also:   BCP0127

        I-D Tag:    draft-ietf-behave-nat-udp-08.txt

        URL:        http://www.rfc-editor.org/rfc/rfc4787.txt

This document defines basic terminology for describing different
types of Network Address Translation (NAT) behavior when handling
Unicast UDP and also defines a set of requirements that would allow
many applications, such as multimedia communications or online
gaming, to work consistently.  Developing NATs that meet this set of
requirements will greatly increase the likelihood that these
applications will function properly.  This document specifies an Internet 
Best Current Practices for the Internet Community, and requests 
discussion and suggestions for improvements.

This document is a product of the Behavior Engineering for Hindrance 
Avoidance Working Group of the IETF.

BEST CURRENT PRACTICE:
This document specifies an Internet Best Current Practices for the
Internet Community, and requests discussion and suggestions for
improvements.  Distribution of this memo is unlimited.


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