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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 Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: Connection-Establishment Preconditions in the 
			  Session Initiation Protocol (SIP)
	Author(s)	: G. Camarillo
	Filename	: draft-ietf-mmusic-connection-precon-00.txt
	Pages		: 8
	Date		: 2004-10-1
	
This document defines the connection-establishment precondition type
   for the SIP preconditions framework. Connection-establishment
   preconditions are met when a transport connection (e.g., a TCP
   connection) is successfully established between two endpoints.

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From mmusic-bounces@ietf.org  Mon Oct  4 00:14:53 2004
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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] Re: draft-ietf-mmusic-sdp-new-20
Date: Sun, 3 Oct 2004 17:11:04 +0100
To: sunil bhargo <marx_bhargav@yahoo.com>
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On 27 Sep 2004, at 11:59, sunil bhargo wrote:
>     in the following paragraph for the unicast IP
> sessions :-
> 1. In section 4.1 it is mentioned for unicast sessions
>
> "The semantics of this address and port depend on the
> media and transport protocol defined.  By default,
> this SHOULD be the remote address and remote port to
> which data is sent."
>
>     but how can the sender identify which port the
> receiver is ready to ready to receive the data on.

An offer/answer exchange, or some similar means, outside the scope of 
SDP.

> 2. also in section 5 it is mentioned that
>
>   " An SDP session description consists of a
> session-level section followed by zero or more
> media-level sections. "
>        this implies that the m= line is OPTIONAL.
>       but then in the same section where the
> differentiation of REQUIRED and OPTIONAL fields is
> given
> Media description
>          m=  (media name and transport address)
>   m= does not contain the '*" field  which means it is
> essential but this contradicts the earlier statement.

The "m=" line is mandatory if a media description is present, although 
the entire media description is optional. Would it clarify if the 
description before the list was changed to "OPTIONAL items in each 
section are marked with a *"?

Colin


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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-20.txt
Date: Sun, 3 Oct 2004 17:16:02 +0100
To: sunil bhargo <marx_bhargav@yahoo.com>
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Hi,

On 27 Sep 2004, at 13:18, sunil bhargo wrote:
>     Doubts regarding the following sections :-
>
> 1.
> "an SDP parser MUST completely ignore any session
> description that contains a type letter that it does
> not understand"
>
>        Isn't the word MUST having very strong
> implications which might not be necessary. For example
> intermediate SDP parsers between the client and server
> if everything that is not understood is rejected then
> it might be problem. First thing that comes to mind
> are the proxies(caching or dumb) they would have to
> deny the service instead of proxying the presentation.

True.

>      But somewhere else we mention as well
> "An SDP parser MUST ignore any attribute it doesn't
> understand."
>      So do we mean to say that any type other than
> attribute if not understood then deny service or do we
> mean something else.

Correct. SDP is extended via attributes and it's likely that an 
attribute is not understood (and therefore ignored). New type letters 
are rarely if ever defined (none have been defined to date), so the 
correct response to them is to discard the session description.

> 2. bwtype has missing RR and RS in section 5.8 or are
> these intentionally missing.

This is intentional, since they're defined elsewhere (in the referenced 
drafts).

Colin


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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-20.txt
Date: Sun, 3 Oct 2004 17:20:03 +0100
To: sunil bhargo <marx_bhargav@yahoo.com>
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I'm not sure I understand your question. For a layered encoding, "b=AS" 
would specify the bandwidth of the entire layered stream.

Colin


On 29 Sep 2004, at 11:25, sunil bhargo wrote:

> Hi,
>          If the presentation is encoded into number of
> streams(MBR) one expects to see equivalent number of
> m= sections if that is so then how can multiple
> streams refer to the same m= section as implied below
> :-
>
>       "For applications where hierarchically encoded
> streams are being sent to a unicast address, it may be
> necessary to specify multiple transport ports.  This
> is done using a similar notation to that used for IP
> multicast addresses in the "c=" field:
> m=<media> <port>/<number of ports> <transport> <fmt
> list>
>       In such a case, the ports used depend on the
> transport protocol.For RTP, the default is that only
> the even numbered ports are used for data with the
> corresponding one-higher odd ports used for the RTCP
> belonging to the RTP session, and the <number of
> ports>
> denoting the number of RTP sessions.  For example:
>           m=video 49170/2 RTP/AVP 31 "
>
>           If multiple streams don't refer to the same
> section then <port>/<number of ports>  might not be
> valid as each media section will give its own ports.
> If multiple streams are mapped to one m= section then
> how will the b=AS parameter be specified on per media
> basis. But I might be missing something.
>
> sunil bhargo
>
> --- Colin Perkins <csp@csperkins.org> wrote:
>
>> On 20 Sep 2004, at 20:42, Internet-Drafts@ietf.org
>> wrote:
>>> A New Internet-Draft is available from the on-line
>> Internet-Drafts
>>> directories.
>>> This draft is a work item of the Multiparty
>> Multimedia Session Control
>>> Working Group of the IETF.
>>>
>>> 	Title		: SDP: Session Description Protocol
>>> 	Author(s)	: M. Handley, et al.
>>> 	Filename	: draft-ietf-mmusic-sdp-new-20.txt
>>> 	Pages		: 45
>>> 	Date		: 2004-9-20
>>> 	
>>> This memo defines the Session Description Protocol
>> (SDP). SDP is
>>> intended for describing multimedia sessions for
>> the purposes of
>>> session announcement, session invitation, and
>> other forms of
>>> multimedia session initiation.
>>>
>>> A URL for this Internet-Draft is:
>>>
>>
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-new-20.txt
>>
>> There's a single change in this version: to remove
>> the underspecified
>> "control" and "data" media types, aligning the SDP
>> media types with
>> MIME media types. If there's a use for these media
>> types they can be
>> registered separately. They've been removed here in
>> the interests of
>> getting the base spec out.
>>
>> Colin
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>
>
> __________________________________________________
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>
>
-- 
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Subject: [MMUSIC] Internet Draft deadlines for IETF 61
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The deadlines for internet-draft submissions for the 61st IETF meeting 
in Washington DC are as follows:

  11 October	Pre-approval deadline for new working group drafts
  18 October	Submission deadline for new working group drafts
  25 October	Submission deadline for other drafts

The deadline is at 9:00am US Eastern time on the date specified.

Note the initial pre-approval deadline: if you are intending to submit 
a new working group draft (i.e. filename draft-ietf-mmusic-...-00.txt) 
you MUST contact the chairs to seek approval, in sufficient time that 
we can read the draft and send approval before 11th October.

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


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From mmusic-bounces@ietf.org  Tue Oct  5 14:45:02 2004
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From: "Hedayat, Kaynam" <khedayat@brixnet.com>
To: "'mmusic@ietf.org'" <mmusic@ietf.org>
Date: Tue, 5 Oct 2004 14:26:22 -0400 
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Subject: [MMUSIC] New internet draft on media loopback
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I would like to bring the attention of the group to a new internet draft on
media loopback capability in SIP sessions. The draft can be found at
http://www.ietf.org/internet-drafts/draft-hedayat-media-loopback-00.txt

Regards,
Kaynam


-----------------------------------------
Kaynam Hedayat
Chief Technology Officer
Brix Networks
285 Mill Road
Chelmsford, MA 01824

tel.   978.367.5611
fax.   978.367.5700
khedayat@brixnet.com
www.brixnet.com

See it before they hear it: Test the performance and quality of your VoIP at
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Subject: [MMUSIC] Call for Papers: WiOpt '05
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Dear Colleagues,

Please note that the deadline for WiOpt'05 has been

              EXTENDED UNTIL October 10!

The web site http://www.wiopt.org has full details.

Our apologies for duplicate copies of this call.

Best regards

Holger Karl
(on behalf of the Wiopt'05 organizers)

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



   The 3rd International Symposium on Modeling and Optimization in
         Mobile, Ad Hoc, and Wireless Networks (WiOpt'05)

         Riva del Garda, Trentino, Italy, April 3-7, 2005

                   http://www.wiopt.org


SCOPE OF THE SYMPOSIUM
----------------------

The complexity of mobile, ad hoc, and wireless networks requires thoroughly
founded models capturing their essential characteristics and new
optimization techniques building upon such models. Insights into both the
theoretical aspects of these problems as well as practical methods and tools
are necessary. This symposium, building upon the two successful WiOpt
workshops, intends to bring together researchers and practitioners working
on optimization issues from different perspectives, including performance
analysis, mobile networking, wireless communication, and optimization
theory.

Contributions to this conference should show new modeling and optimization
techniques suitable for mobile/wireless networks or innovative applications
of these techniques to improve network operation. Topics of interest
include, but are not limited to:

* Modeling techniques and tools for mobile/wireless networks

* Performance characterisation & optimisation of mobile/wireless networks
and systems

* Mobile network optimisation: Applications, algorithms, and tools

* Protocol design for optimal mobile network utilisation

* Models and optimization approaches for energy efficiency, QoS,  bandwidth
economy and other aspects of mobile/wireless networks

* Models and optimization techniques for non-standard network paradigms like
sensor, vehicular, body area networks 

* Security and co-operation models for mobile networks

* Pricing and incentives in mobile and ad-hoc networks

Papers should be submitted as extended abstracts of up to 8 pages (see
webpage for details). Selected papers from WiOpt 2005 will appear in a
special issue of ACM/Kluwer Monet.


ADJUNCT WORKSHOPS
-----------------

Several one-day workshops will accompany the main WiOpt Symposium on April
3rd and 7th (see webpage for details):


* WINMEE 2005: Measurement in Wireless Networks
    Chairs: L. Krishnamurthy (Intel), 
            D. Papagiannaki (Intel Research Cambridge)

* RAWNET 2005: Resource Allocation in Wireless Networks, 
    Chairs: A. Proutiere (France Telecom R&D), S. Borst (Bell Labs.-CWI)

* SPASWIN 2005: Stochastic Geometry for Wireless Networks, 
    Chairs: F. Baccelli (INRIA-ENS), B. Blaszczyszyn (INRIA-ENS)

* NETCOD 2005: Network Coding and Information Theory, 
    Chairs: R. W. Yeung (The Chinese U. of Hong Kong), 
            P. Chou (Microsoft Research)


SPONSORHSIP
-----------

WiOpt05 is sponsored by INRIA, Create-Net, ICST, eNext NoE, Intel, and
Siemens mobile; it is technically co-sponsored by the IEEE Information
Theory Society; it is organized in cooperation with ACM and IFIP WG 6.3.


IMPORTANT DATES
---------------

Submission deadline:		Oct. 10, 2004 (midnight EST) 
Notification of acceptance: 	Dec. 5, 2004 
Camera-ready copy: 		Jan. 5, 2005

Papers should be submitted in pdf or ps format through the EDAS system
(http://edas.info). For more details, please visit http://www.wiopt.org



General Chair
-------------
E. Altman, INRIA

Vice General Chair
------------------
H. Karl, U. of Paderbon

Program Co-Chairs
-----------------
R. R. Mazumdar, Purdue U.
I. Stavrakakis, U. of Athens

Steering Committee Chair
------------------------
I. Chlamtac, Create-Net

Program Committee
-----------------
E. Altman, INRIA
V. Anantharam, UC Berkeley
K. Avrachenkov, INRIA
T. Basar, U. of Illinois
A. Campbell, Columbia U. 
X. Cao, Hong Kong U. of Science and Technology 
A. Chockalingam, Indian Institute of Science 
M. Conti, CNR-ITT 
J. Crowcroft, U. of Cambridge 
P. Dube, IBM Research 
A. Ephremides, U. of Maryland 
S. Giordano, SUSPI 
A. Jukan, NSF CISE Networking Research 
H. Karl, U. of Paderborn
P. Key, Microsoft Research 
J-W. Lee, Princeton U. 
R. Malhame, Ecole Poly-technique de Montreal 
P. Marbach, U. of Toronto 
D. Mcauley, Intel Research Cambridge
A. Misra, IBM Research
Q. Ni, National U. of Ireland 
Y. Onozato, Guma University 
A. Orda, Technion 
A. Panagakis, U. of Athens
R. Rao, UC San Diego 
A. Richa, Arizona State U.
P. Rodriguez, Microsoft Research
C. Rosenberg, Purdue U.
V. Syrotiuk, Aritona State U.
T. Turletti, INRIA
J. Wieselthier, Naval Research Laboratory 
M. Zukerman, U. of Melbourne

Workshops Chair
---------------
C. Barakat, INRIA

Local Arrangements Chair
------------------------
G. Boata, U. of Trento

Publicity Chair
---------------
T. Turletti, INRIA

Publications Chair
------------------
E. Deriche, INRIA

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. What is the business model for WiMAX? 
. What do we learn from earlier deployments? 
. What about the future extensions of the standard? 
. How is addressed the interoperability challenge? 

These questions, among others, will be addressed during the WiMAX Summit to
be held in Paris next 5-8 April, 2005.

A call for papers has been launched online, effective until October 29.

 

Get all details at:

 <http://www.upperside.fr/wimax05/wimax2005intro.htm>
http://www.upperside.fr/wimax05/wimax2005intro.htm

 

 


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<body lang=3DFR link=3Dblue vlink=3Dpurple>

<div class=3DSection1>

<p class=3DMsoNormal><strong><b><font size=3D2 face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:10.0pt;font-family:Arial'>&#8226; =
</span></font></b></strong><font
size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:10.0pt;font-family:Arial'>What
is the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>business
model</span></font></b></strong> for WiMAX? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>What
do we learn from <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>earlier
deployments</span></font></b></strong>? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>What
about the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>future
extensions</span></font></b></strong> of the standard? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>How
is addressed the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>interoperability</span></font></b></strong>
challenge? <br>
<br>
These questions, among others, will be addressed during the WiMAX Summit =
to be
held in <st1:place w:st=3D"on"><st1:City =
w:st=3D"on">Paris</st1:City></st1:place>
next 5-8 April, 2005.<br>
<br>
<span class=3Dtextebold>A call for papers </span><span class=3Dtexte>has =
been
launched online, effective until </span><span class=3Dstyle15>October =
29</span>.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>&nbsp;<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'>Get all details =
at:<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DArial><span =
style=3D'font-size:
10.0pt;font-family:Arial;color:black'><a
href=3D"http://www.upperside.fr/wimax05/wimax2005intro.htm"
title=3D"http://www.upperside.fr/wimax05/wimax2005intro.htm"><span =
lang=3DEN-GB><span
title=3D"http://www.upperside.fr/wimax05/wimax2005intro.htm">http://www.u=
pperside.fr/wimax05/wimax2005intro.htm</span></span></a></span></font><fo=
nt
size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:10.0pt;font-family:Arial'><o:p></o:p></span></font></p=
>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
10.0pt;font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

</div>

</body>

</html>

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From mmusic-bounces@ietf.org  Sat Oct  9 17:03:57 2004
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From: Yali Wang <ylwang@soe.ucsc.edu>
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To Whom It May Concern:


COuld you please help me?

In my own screen, the text file of internet draft shows very
neatly, at the top or the end of each page, I put the page remarking
lines.

But once after attached text file by email, then re-download and
re-open it, it changes to messed-up lines, and looks very ugly.
No matter how I modify the original,which in fact looks very good,
as long as attached by email, then the the lines text file look ugly.

Furthermore, I have the figure in the file. After attached, the figure
always disppear.

How should I solve this problem ? I used the Microsoft word 2000 for editing.

Thanks.

yali

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From mmusic-bounces@ietf.org  Mon Oct 11 06:16:16 2004
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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] Internet Draft deadlines for IETF 61
Date: Mon, 11 Oct 2004 11:10:14 +0100
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Reminder: the internet-draft submission deadline for -00 drafts is one 
week from today.

Colin



On 3 Oct 2004, at 09:47, Colin Perkins wrote:
> The deadlines for internet-draft submissions for the 61st IETF meeting 
> in Washington DC are as follows:
>
>  11 October	Pre-approval deadline for new working group drafts
>  18 October	Submission deadline for new working group drafts
>  25 October	Submission deadline for other drafts
>
> The deadline is at 9:00am US Eastern time on the date specified.
>
> Note the initial pre-approval deadline: if you are intending to submit 
> a new working group draft (i.e. filename draft-ietf-mmusic-...-00.txt) 
> you MUST contact the chairs to seek approval, in sufficient time that 
> we can read the draft and send approval before 11th October.
>
> -- 
> Colin Perkins
> http://csperkins.org/


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From mmusic-bounces@ietf.org  Tue Oct 12 03:22:57 2004
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From: Tony Yat-Tung Cheung <tony.cheung@asiayeah.com>
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Subject: [MMUSIC] Some possible typos on draft-rosenberg-midcom-turn-05
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Hi,

We are looking at the TURN draft for implementation.

We have found a few possible typos so far,

1) Page 6.
"The TURN protocol will provide the STUN client with IP addresses that 
route to it from the public internet."

We may replace "STUN client" with "TURN client".

2) Page 7
"Once a host sends packets to that transport address, it is ``locked 
down'', meaning that the client cannot cause packets to be sent to that 
host through the relay."

This is rather hard to understand. Does that mean the following,

"Once a peer sends packets to that transport address, it is "locked 
down", meaning that other peers cannot cause packets to be sent to that 
host through the relay."

3) Page 30: "TRANSPORT-PREFERENCES"

"It is either 32 or 96 bits long". This is not true, because it contains 
a variable-size MAPPE-ADDRESS element. I think the size should be 32 
bits or 32 bits + sizeof(MAPPED-ADDRESS).

Thanks.

Best Regards,
Tony Cheung

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Date: Tue, 12 Oct 2004 11:29:19 -0700
To: mmusic@ietf.org
From: Rishi Mathew <rmathew@real.com>
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Subject: [MMUSIC] Transport Header Stipulation
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Magnus/All,

This is regarding the Transport header defined in Chapter 14.43 of the RTSP 
spec. Right now, it says:

    The usage of any parameter that was not defined in RFC 2326 or in an
    extended way requires that request or response contains a Require
    header with the "play.basic" feature tag.

It is claimed that this was done to prevent backwards compatibility 
problems. However, in our field implementation, this parameter has always 
had a different value than what was defined in the original RTSP spec. It 
looks like this:

         Transport: x-real-rdt/udp;client_port=6970;server_port=17738\r\n

This was legal in the original spec. Obviously, any such stipulation is 
going to create more problems than it solves on the field. Our suggestion 
is that you remove this stipulation.

Thanks,
Rishi.



Rishi Mathew
Software Development Engineer
Helix Development Support Group
RealNetworks, Inc.
rmathew@real.com
206.892.6126
http://www.helixcommunity.org
http://www.realnetworks.com/products/support/devsupport.html


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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: [MMUSIC] New draft on requirements for "Sharing and Remote Access
	to Applications"
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Jonathan Lennox, Jason Nieh, Ricardo Baratto and I have put together an 
initial rough draft collecting requirements on sharing and remote access 
to applications. We believe that application sharing, as in some of the 
T.120 documents, and remote access to applications, exemplified in tools 
such as vnc, exhibit significant commonality.

Until it appears in the archives, the draft can be found at

http://www.cs.columbia.edu/sip/draft/sharing/draft-schulzrinne-mmusic-sharing-00.txt
http://www.cs.columbia.edu/sip/draft/sharing/draft-schulzrinne-mmusic-sharing-00.html

We would value comments and would like to request a slot at the MMUSIC 
meeting if there is group interest.

Henning

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From: Euchner Martin <martin.euchner@siemens.com>
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Subject: [MMUSIC] re: draft-ietf-mmusic-kmgmt-ext-11.txt: KMID? IANA
	considerations section
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IANA Consideration section 8 of the draft allocates key management IDs (KMID) for the key management protocols. One KMID (not yet allocated by IANA) will be for MIKEY.

Now here is my question:

Assuming an example where the offerer advertises just 3 MIKEY protocols; e.g. MIKEY-preshared, MIKEY-DHsign and MIKEY-PK,
Then the SDP would hold something like
a=key-mgmt:mikey   a1234a54...
a=key-mgmt:mikey   b454646ca54...
a=key-mgmt:mikey   c95785656ef3....

and the generated protocol list would be "mikey;mikey;mikey", where 'mikey' represents the allocated KMID value for the MIKEY protocol family.

Q1) In the context of bidding-down protection, what kind of security does this list actually provide?

I believe that an attacker could easily substitute another, weak 4th mikey key management protocol ("mikey-password") against one of the other three offered ones; yet the (MIKEY) protected protocol list would remain the same and the answer would not be able to detect the attack.

My conclusion (needs confirmation) is that the design flaw is that the KMID is on a key management protocol family but it should rather be on the key management protocol option/varient.

For MIKEY, the fix should be straightforward: Allocate a KMID for each of the MIKEY protocol variants; i.e.
One for MIKEY-preshared, one for MIKEY-DHsign and one for MIKEY-PK.

Cheers 

Martin.



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Hi All,

An update to Internet-Draft "The IMG Envelope" is now available.

http://www.ietf.org/internet-drafts/draft-walsh-mmusic-img-envelope-00.txt

All comments are very welcome.

Regards,
Sami Peltotalo.

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Folks,

MMUSIC is scheduled to meet once in Washington DC, currently

     on Tuesday, 9 Nov 2004, 9:00-11:30

This mail is to solicit input on the agenda.  We have quite a
few work items to finish up, so clear preference will be given
to discuss and resolve issues with our deliverables.

Please send requests for agenda time to the chairs (if you have not
already done so) and please refer to the Internet Draft you will
be discussing (and to issues to be discussed).

Thanks,
Joerg






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Subject: [MMUSIC] Re: draft-ietf-mmusic-kmgmt-ext-11.txt: KMID? IANA
 considerations section
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Hi Martin,

Thanks for raising this issue. It is known to us and was discussed 
internally when deciding for the current protection mechanism. Perhaps
we should have motivated why we did not consider this a threat to
be taken care of, but now that you raise the issue, let's see if we
can settle it.

Euchner Martin wrote:

 > IANA Consideration section 8 of the draft allocates key management 
IDs (KMID) for the key management protocols. One KMID (not yet allocated 
by IANA) will be for MIKEY.
 >
 > Now here is my question:
 >
 > Assuming an example where the offerer advertises just 3 MIKEY 
protocols; e.g. MIKEY-preshared, MIKEY-DHsign and MIKEY-PK,
 >
 > Then the SDP would hold something like
 > a=key-mgmt:mikey   a1234a54...
 > a=key-mgmt:mikey   b454646ca54...
 > a=key-mgmt:mikey   c95785656ef3....


First of all, why would this be something that ever occurs?
The reason we allow several options is to let the parties
agree on a particular prefered *protocol*, but choosing options
within one and the same protocol is not intended to be done
at this level.

(Anyway, let's assume it can happen...)

 >
 > and the generated protocol list would be "mikey;mikey;mikey", where 
'mikey' represents the allocated KMID value for the MIKEY protocol family.


Right.

 >
 > Q1) In the context of bidding-down protection, what kind of security 
does this list actually provide?
 >
 > I believe that an attacker could easily substitute another, weak 4th 
mikey key management protocol ("mikey-password") against one of the 
other three offered ones; yet the (MIKEY) protected protocol list would 
remain the same and the answer would not be able to detect the attack.


Okay, let's assume he can. To accomplish this, the "mikey-password"
variant needs to be so weak that he can fake the authentiction data of
the a-line where this is inserted. That is, the authentication mechanism
of the password variant is completely insecure. But if this is so, the
attacker does not even need the "real" Initiator to be on-line: he can
at any  point in time impersonate the real Initiator, faking MIKEY
messages using the password variant. Therefore...

 >
 > My conclusion (needs confirmation) is that the design flaw is that 
the KMID is on a key management protocol family but it should rather be 
on the key management protocol option/varient.
 >
 > For MIKEY, the fix should be straightforward: Allocate a KMID for 
each of the MIKEY protocol variants; i.e.
 > One for MIKEY-preshared, one for MIKEY-DHsign and one for MIKEY-PK.


...this protection mechanism is totally inadequate for the case
you describe, the "real" security hole remains open even if
we were to add this feature.

Another scenario in which the suggested protection is not enough
is as follows: suppose preshared-MIKEY is extended by a cipher/MAC
combination which has poor security. It may then not be enough to
authenticate the fact MIKEY-preshared was offered, but rather the
exact options at the transform level (which *is* done in each a-line
individually, but not across a-lines). However, we would exclude this
as a threat for the same reason as above: if faked MIKEY lines can
be inserted by an attacker, the problem is not of bidding-down nature,
it is really a fundamental security flaw of the protocol as such.

Finally, if one is still concerned by this, we would suggest that
the Initiator uses MIKEY's option of requesting a verification message
back from the Responder. If an attacker modifies the offer to contain
something the Initiator actually didn't offer, the response back will
be generated by an incorrect method/key, and the Initiator will
detect this when checking the response.

Best,

/Mats and MIKEY gang




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Hi Mathew

See below

Rishi Mathew wrote:

> Magnus/All,
> 
> This is regarding the Transport header defined in Chapter 14.43 of the 
> RTSP spec. Right now, it says:
> 
>    The usage of any parameter that was not defined in RFC 2326 or in an
>    extended way requires that request or response contains a Require
>    header with the "play.basic" feature tag.
> 

What I refer to as parameter is the addition of what in the ABNF is 
called parameter, i.e. what comes after the protocol specifier. For 
example a new parameter would be the "new_parm" below:

Transport: x-real-rdt/udp;client_port=6970;server_port=17738; new_parm=XYZ

> It is claimed that this was done to prevent backwards compatibility 
> problems. However, in our field implementation, this parameter has 
> always had a different value than what was defined in the original RTSP 
> spec. It looks like this:
> 
>         Transport: x-real-rdt/udp;client_port=6970;server_port=17738\r\n
> 
> This was legal in the original spec. Obviously, any such stipulation is 
> going to create more problems than it solves on the field. Our 
> suggestion is that you remove this stipulation.
> 

"x-real-rdt/udp" is protocol specifier. However that you claim that it 
is legal in RFC 2326 is not correct. The ABNF for the transport header 
looks like this:

    Transport           =    "Transport" ":"
                             1\#transport-spec
    transport-spec      =    transport-protocol/profile[/lower-transport]
                             *parameter
    transport-protocol  =    "RTP"
    profile             =    "AVP"
    lower-transport     =    "TCP" | "UDP"

Which means that no extension allowance was made in the ABNF. Thus it is 
unclear that a parser will correctly handle any extension. I do agree 
that the intention was to allow for this. However due to the shortcoming 
of the RFC 2326 it is a dangerous proposition to use extension protocol 
specifiers or parameters. That is the reason the text is present in the 
draft.

If you compare the above ABNF with the new one present in the draft:

Transport       =  "Transport" ":" transport-spec
                       *("," LWS transport-spec)
    transport-spec  =  transport-id *parameter
    transport-id    =  transport-prot "/" profile ["/" lower-transport]
                       ; no LWS is allowed inside transport-id
    transport-prot    =  "RTP" / token
    profile           =  "AVP" / token
    lower-transport   =  "TCP" / "UDP" / token

You see that there now is specified that all these types can be extended.

The draft do also define IANA registries for these points of extension, 
see 20.5.

So I don't think it is appropriate to remove this, as it can't be 
guaranteed that your proprietary extensions would correctly work with 
another implementation following RFC 2326. When clients support the 
updated specification and indicates this correctly, then you can be 
certain that any correct implementation can safely ignore your 
proprietary transport protocol.

So I think I will need to clarify the note to apply to any extensions. I 
also should change the wording to not use the word "require". My 
proposed change is:

"The usage of any parameter or protocol specifier not defined in RFC 
2326 or which is extended, SHALL be indicated by the use of the Require 
header and a feature tag(s) to ensure support by the receiver. This
is to protect against failures with uncertain consequences, as the a 
receiver of a message may do anything between ignoring a parameter, to 
complete refuse the message."

Any Comments?


Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
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Subject: [MMUSIC] Change proposal for draft-ietf-mmusic-sdp-new-20.txt
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Hi Colin,

In the work on updating RFC 3267 (AMR and AMR-WB RTP payload format) 
which originally defines "a=maxptime" I have recieved a request for 
clarification that I think make sense and which do not create any 
backwards compatibility issues:

In Section 6: under attribue "a=maxptime:<maximum packet time>"

I would like to rephrase the sentence:
"The time SHOULD be a multiple of the frame size."

To:
"The time SHOULD be an integer multiple of the frame size.
                      ^^^^^^^^^

Comments?

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
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Hi,

Lets revive this issue so that I can edit in the resolution if possible.

I had proposed that we consider either of the solutions:

> 1. Define a separate proxy-supported header, similar to the 
> proxy-require header. This header would then be populated by the 
> requester with all proxy-related feature tags that it supports. Then the
> proxy would simply forward the header with the common set of tags it 
> support. The server simply echoes the header in the response. The 
> proxies should ignore the header in the response messages.
> 
> 2. Allow the server to modify the supported headers to only contain 
> functionality it also supports.

Sunil Bhargo:
>      Therefore i am ok with the suggestion number 2
> that you have mentioned. Please let me know your
> suggestions.

When I looked at this now, I think we have a serious problem with 
proposal number 2. The problem is that the proxy chain can remove 
feature tags that are applicable only to servers. This is not good as 
the proxies could prevent the client or server from learning about a 
feature that the proxy will simply do pass-through on.

The only solution that I currently can think of for this problem would 
be to prefix the feature tags with its type; proxy, server or both. 
However that may be problematic to deploy.

I would also like to respond to some of Sunil's comments:
>      We also need to take into consideration that the
> client sometime might not be aware of the existence of
> the proxy. This can be in the cases of transparency or
> accelerated proxy deployment. In such cases even if we
> add the proxy-Supported header the client is not going
> to send it but will send the Supported header. 

I don't think this is a real issue, if a proxy sees a Supported header 
but no Proxy-Supported, the first proxy in the chain simply adds it. It 
is after all also the proxy that needs to learn if it can support any 
functionality that the session may consider. Also I think that we in 
this specification can simply request that any updates client simply 
always add the proxy-supported header, if it intends to use supported.


>    Also one problem in having a separate header
> proxy-supported and Supported is that even if the
> client receives both the proxy-supported and supported
> header in the response it should use the common set of
> features in both the proxy-supported and nsupported
> headers and this can be acheived by only having the
> supported header.

Yes, I agree that one needs to consider what the common set is. However 
that is the issue for the requester, that desires to use a feature. If 
that feature is supported by the requester it needs to understand it, 
and will also be able to determine which server and/or proxy features 
that needs to be supported to make the desired feature work. Thus I 
think it much easier for the requesting peer to make that combination 
than for an entity in the middle to do it.

Having two headers also allows for a nice separation of any feature tag 
depending on its type. Server only features goes in supported, proxy 
only goes in proxy-supported and feature for both domains goes in both.

This also gives a client that are capable of bypassing proxies to do it 
if required to get a certain feature that are not supported by the proxy 
chain.

Please comment!


Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
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From: sunil bhargo <marx_bhargav@yahoo.com>
Subject: Re: [MMUSIC] Is Supported Header in RTSP changeable by a proxy ?
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Hi Magnus,
          Thanks for the reply. I agree to your
concerns regarding the option 2 of not modifying the
Supported header by the intervening proxy or server. 
I am ok with option 1. In case of proxy chain then the
client will receive the common set of features in the
proxy-supported header. But what is the solution about
clients supporting the previous versions ,that is only
the supported header. In that case i think the proxy
should be allowed to write the supported headers. 
          Just wanted add one thing that the
user/player should never make the bypassing decision
based on the Supported headers feature-tags as
inclusion of any feature-tag in Supported header
should be optional session feature and the
player/client should be able to live with/without this
feature tag. If something is mandatory then it should
be included  into require and proxy-require headers.
Can we add the  text that the client is advised to use
only the common set of features in proxy-supported and
supported headers ?

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

> Hi,
> 
> Lets revive this issue so that I can edit in the
> resolution if possible.
> 
> I had proposed that we consider either of the
> solutions:
> 
> > 1. Define a separate proxy-supported header,
> similar to the 
> > proxy-require header. This header would then be
> populated by the 
> > requester with all proxy-related feature tags that
> it supports. Then the
> > proxy would simply forward the header with the
> common set of tags it 
> > support. The server simply echoes the header in
> the response. The 
> > proxies should ignore the header in the response
> messages.
> > 
> > 2. Allow the server to modify the supported
> headers to only contain 
> > functionality it also supports.
> 
> Sunil Bhargo:
> >      Therefore i am ok with the suggestion number
> 2
> > that you have mentioned. Please let me know your
> > suggestions.
> 
> When I looked at this now, I think we have a serious
> problem with 
> proposal number 2. The problem is that the proxy
> chain can remove 
> feature tags that are applicable only to servers.
> This is not good as 
> the proxies could prevent the client or server from
> learning about a 
> feature that the proxy will simply do pass-through
> on.
> 
> The only solution that I currently can think of for
> this problem would 
> be to prefix the feature tags with its type; proxy,
> server or both. 
> However that may be problematic to deploy.
> 
> I would also like to respond to some of Sunil's
> comments:
> >      We also need to take into consideration that
> the
> > client sometime might not be aware of the
> existence of
> > the proxy. This can be in the cases of
> transparency or
> > accelerated proxy deployment. In such cases even
> if we
> > add the proxy-Supported header the client is not
> going
> > to send it but will send the Supported header. 
> 
> I don't think this is a real issue, if a proxy sees
> a Supported header 
> but no Proxy-Supported, the first proxy in the chain
> simply adds it. It 
> is after all also the proxy that needs to learn if
> it can support any 
> functionality that the session may consider. Also I
> think that we in 
> this specification can simply request that any
> updates client simply 
> always add the proxy-supported header, if it intends
> to use supported.
> 
> 
> >    Also one problem in having a separate header
> > proxy-supported and Supported is that even if the
> > client receives both the proxy-supported and
> supported
> > header in the response it should use the common
> set of
> > features in both the proxy-supported and
> nsupported
> > headers and this can be acheived by only having
> the
> > supported header.
> 
> Yes, I agree that one needs to consider what the
> common set is. However 
> that is the issue for the requester, that desires to
> use a feature. If 
> that feature is supported by the requester it needs
> to understand it, 
> and will also be able to determine which server
> and/or proxy features 
> that needs to be supported to make the desired
> feature work. Thus I 
> think it much easier for the requesting peer to make
> that combination 
> than for an entity in the middle to do it.
> 
> Having two headers also allows for a nice separation
> of any feature tag 
> depending on its type. Server only features goes in
> supported, proxy 
> only goes in proxy-supported and feature for both
> domains goes in both.
> 
> This also gives a client that are capable of
> bypassing proxies to do it 
> if required to get a certain feature that are not
> supported by the proxy 
> chain.
> 
> Please comment!
> 
> 
> Cheers
> 
> Magnus Westerlund
> 
> 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: Euchner Martin <martin.euchner@siemens.com>
To: =?iso-8859-1?Q?=27Mats_N=E4slund=27?= <mats.naslund@ericsson.com>,
        Euchner Martin <martin.euchner@siemens.com>
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        "Elisabetta Carrara \(KI/EAB\)" <elisabetta.carrara@ericsson.com>,
        "Fredrik Lindholm \(KI/EAB\)" <fredrik.lindholm@ericsson.com>,
        mmusic@ietf.org, "Jari Arkko \(JO/LMF\)" <jari.arkko@ericsson.com>
Subject: [MMUSIC] RE: draft-ietf-mmusic-kmgmt-ext-11.txt: KMID? IANA
	considerations section
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Mats,

Many thanks to you and the entire MIKEY crew for answering my question.

On one hand, I understand your response (see my acknowledgement below), =
one the other hand put some further comments for discussion and yet in =
some other areas I believe that there may be just a resolvable =
misunderstanding.

I'm addressing the issues inline below.

Cheers

Martin

 -----Original Message-----
From: 	Mats N=E4slund [mailto:mats.naslund@ericsson.com]=20
Sent:	Montag, 18. Oktober 2004 11:39
To:	Euchner Martin
Cc:	mmusic@ietf.org; Jari Arkko (JO/LMF); Elisabetta Carrara (KI/EAB); =
Fredrik Lindholm (KI/EAB); Karl Norrman (KI/EAB)
Subject:	Re: draft-ietf-mmusic-kmgmt-ext-11.txt: KMID? IANA =
considerations section

>
 > Assuming an example where the offerer advertises just 3 MIKEY=20
protocols; e.g. MIKEY-preshared, MIKEY-DHsign and MIKEY-PK,
 >
 > Then the SDP would hold something like
 > a=3Dkey-mgmt:mikey   a1234a54...
 > a=3Dkey-mgmt:mikey   b454646ca54...
 > a=3Dkey-mgmt:mikey   c95785656ef3....


First of all, why would this be something that ever occurs?
The reason we allow several options is to let the parties
agree on a particular prefered *protocol*, but choosing options
within one and the same protocol is not intended to be done
at this level.

MEU:> Ok this confirms that kmtext "negotiates" a key management =
protocol family, but does not actually negotiate any particular key =
management capability. I intentionally put negotiate in quotes, since =
this SDP-based negotiation is operating just on the family - level of =
the key management protocol but certainly not on protocol capabilities =
(such as a particular choice like MIKEY-PK, or even more fine grained =
on a crypto algorithm level). So in order to "negotiate" any deeper =
than the family-level, kmgtext offers the desired MIKEY protocol =
variants in parallel. Unfortunately, this increases the processing time =
of the offerer considerably; namely 3 times to instantiate and to =
compile 3 different MIKEY key management protocol messages. For the =
answerer, the work load is not as high as on the offerer, but still the =
base64 decoding requires some more effort than just a quick inspection =
of a specific KMID would cost; so this latter point is really a minor =
issue.

I should clearly say, that I don't see any better alternative for a =
full 2-way handshake negotiation process yet a 4-way handshake would =
not face the parallel explosion property.


(Anyway, let's assume it can happen...)

 >
 > and the generated protocol list would be "mikey;mikey;mikey", where=20
'mikey' represents the allocated KMID value for the MIKEY protocol =
family.


Right.

 >
 > Q1) In the context of bidding-down protection, what kind of security =

does this list actually provide?
 >
 > I believe that an attacker could easily substitute another, weak 4th =

mikey key management protocol ("mikey-password") against one of the=20
other three offered ones; yet the (MIKEY) protected protocol list would =

remain the same and the answer would not be able to detect the attack.


Okay, let's assume he can. To accomplish this, the "mikey-password"
variant needs to be so weak that he can fake the authentiction data of
the a-line where this is inserted. That is, the authentication =
mechanism
of the password variant is completely insecure. But if this is so, the
attacker does not even need the "real" Initiator to be on-line: he can
at any  point in time impersonate the real Initiator, faking MIKEY
messages using the password variant. Therefore...

MEU:> Here I sense a minor misunderstanding. I'm clearly not =
questioning the security of any MIKEY protocol nor am I saying that any =
MIKEY protocol is any close to being weak that it could be broken.

My only concern in the context of a bidding-down attack is, that the =
bidding-down attack could be accomplished within a spurious hop =
somewhere between the originating and terminating end systems, such =
that the attacker replaces a stronger offer by a weaker one of the same =
protocol family. For the bidding-down attack success, the strength of =
the substituted MIKEY protocol matters only as much as its security is =
(somewhat) less than the original protocols. If the reduced security =
level is sufficient to break MIKEY in any reasonable time, is a =
completely different issue.
The protected protocol list would not indicate the particular =
substitution attack.

However, I admit that there are two mechanisms in place that allow the =
offerer to detect such an attack:
a) The offerer would receive a SDP response from the answerer that =
holds a "foreign", non-offered kmgt option.
b) MIKEY luckily provides the verification message, unless the MIKEY =
key management protocol isn't already two-way.

Mechanism a) or b) are the only way for the offerer to detect any =
occurred bidding-down attack; the responder has no chance to detect =
such an attack just by inspecting the incoming offer.

For myself this fact is now clear. So my conclusion is that in fact the =
MIKEY verification message has a very strong value in protecting the =
offerer against bidding-down attacks. I simply cannot remember right =
now, if this has been adequately explained in MIKEY RFC or in the kmext =
draft and if that is apparent to the reader.


 >
 > My conclusion (needs confirmation) is that the design flaw is that=20
the KMID is on a key management protocol family but it should rather be =

on the key management protocol option/varient.
 >
 > For MIKEY, the fix should be straightforward: Allocate a KMID for=20
each of the MIKEY protocol variants; i.e.
 > One for MIKEY-preshared, one for MIKEY-DHsign and one for MIKEY-PK.


...this protection mechanism is totally inadequate for the case
you describe, the "real" security hole remains open even if
we were to add this feature.

MEU:> hmm, I'm not certain if I understand this fully.
Except for the problem that you correctly describe in the following =
paragraph, I believe that my proposed solution could at least provide =
some value for the answerer to get protection against bidding down =
attacks on the MIKEY protocol level, yet clearly not on any =
transform/alg level below.
This would challenge us to answer the question: Is the proposed yet =
non-perfect approach (i.e. allocating several KMIDs) worth the effort?

Maybe...when we don't want the entire security protection to depend =
just on the offerer, but give the answerer a certain chance at least to =
recover on its own.
I put this aspect for further discussion among the security experts.


Another scenario in which the suggested protection is not enough
is as follows: suppose preshared-MIKEY is extended by a cipher/MAC
combination which has poor security. It may then not be enough to
authenticate the fact MIKEY-preshared was offered, but rather the
exact options at the transform level (which *is* done in each a-line
individually, but not across a-lines).=20
MEU:> Yes I completely agree to this point. Taken to the most extreme =
case this would mean that the entire offering of security parameters =
would need to be secured. I clearly think that this is beyond what we =
want to achieve.

However, we would exclude this
as a threat for the same reason as above: if faked MIKEY lines can
be inserted by an attacker, the problem is not of bidding-down nature,
it is really a fundamental security flaw of the protocol as such.

MEU:> not quite. Please see my explanations on the spurious hop, above.

Finally, if one is still concerned by this, we would suggest that
the Initiator uses MIKEY's option of requesting a verification message
back from the Responder. If an attacker modifies the offer to contain
something the Initiator actually didn't offer, the response back will
be generated by an incorrect method/key, and the Initiator will
detect this when checking the response.

MEU:> Yes full agreement. Clearly the verification mechanism is a very =
powerful tool to protect the offerer from a bidding down attack. =
However, the answerer fully depends on the correct reaction and =
implementation of the offerer. The answerer has no means to detect an =
occurred bidding-down attack on its own.

Best,

/Mats and MIKEY gang



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From mmusic-bounces@ietf.org  Tue Oct 19 23:14:08 2004
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Date: Tue, 19 Oct 2004 11:27:08 -0700
From: Awais Ali <aali@i2telecom.com>
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	<4174F77E.70807@ericsson.com>
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Subject: [MMUSIC] purpose of derived-from in ICE SDP attribute
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Hi all,

I can't seem to figure out the purpose of  derived-from attribute in 
section 9 of draft-ietf-mmusic-ice-02. It is mentioned anywhere in the 
draft. Can anybody explain that to me?

Regards
Ali


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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] Change proposal for draft-ietf-mmusic-sdp-new-20.txt
Date: Tue, 19 Oct 2004 20:13:06 +0100
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On 19 Oct 2004, at 10:14, Magnus Westerlund wrote:
> In the work on updating RFC 3267 (AMR and AMR-WB RTP payload format) 
> which originally defines "a=maxptime" I have recieved a request for 
> clarification that I think make sense and which do not create any 
> backwards compatibility issues:
>
> In Section 6: under attribue "a=maxptime:<maximum packet time>"
>
> I would like to rephrase the sentence:
> "The time SHOULD be a multiple of the frame size."
>
> To:
> "The time SHOULD be an integer multiple of the frame size.
>                      ^^^^^^^^^
> Comments?

Looks reasonable to me. Unless there are objections in the next couple 
of days, I'll make the change.

Colin


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From: Jon Ringle <ml-mmusic@ringle.org>
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Date: Tue, 19 Oct 2004 16:46:14 -0400
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Subject: [MMUSIC] Confusion regarding ICE username fragment
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I'm a little confused regarding the use of username fragments in ICE. The 
following statement is in draft-ietf-mmusic-ice-02, Section 5.3.2:

   For each local transport address where a STUN server is running, the
   client MUST choose a username fragment and a password.  The username
   fragment created by the client will be concatenated with the fragment
   created by its peer.  The result will serve as the username provided
   by its peer in STUN requests.  By creating the username as a
   combination of information from each side of a call, it allows a
   client to correlate the source of the request with a candidate
   transport address.  This is discussed further below.

   The username fragment MUST be globally unique, so that no other host
   will select a username with the same value.  This username fragment
   and password will be passed to its peer in an initiate or accept
   message.  As such, the process described in this section will
   associate, with each local transport address, a username fragment and
   password.  The client also associates this same username fragment and
   password with any transport addresses derived from the local
   transport address.

Suppose phone-A has only one local transport address on eth0 and derives a 
STUN and TURN address from the eth0 local transport address. From what I read 
above, it seems like only a username fragment and password is chosen only for 
the eth0 address, and that the derived STUN/TURN addresses use this same 
username/password. So, this would result in alt's in SDP in the initiate 
message from phone-A to phone-B like this:

a=alt:1 1.0 : PhoneAeth0user password 172.16.1.102 1234
a=alt:2 0.8 : PhoneAeth0user password 1.1.1.4 8801
a=alt:3 0.4 : PhoneAeth0user password 1.1.1.5 3002

This seems to go against the examples that are shown in 
draft-rosenberg-sipping-nat-scenarios-03, where a separate username fragment 
is shown for each local and derived addresses.

It seems to me that separate username fragments would be needed for each 
gathered address for correlation to the candidate table.

I feel like I am missing a key concept here. Could someone please explain.

Thanks,

Jon



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From: lazzaro <lazzaro@eecs.berkeley.edu>
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Subject: [MMUSIC] maxptime
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Magnus writes:
> In the work on updating RFC 3267 (AMR and AMR-WB RTP payload format) 
> which originally defines "a=maxptime" I have recieved a request for 
> clarification that I think make sense and which do not create any 
> backwards compatibility issues:
>
> In Section 6: under attribue "a=maxptime:<maximum packet time>"
>
> I would like to rephrase the sentence:
> "The time SHOULD be a multiple of the frame size."
>
> To:
> "The time SHOULD be an integer multiple of the frame size.


If maxptime is now considered a parameter for use in
all codecs (like the "ptime" it was derived from), shouldn't
its definition admit that not all codecs are frame based?

The RTP MIDI I-Ds use ptime and maxptime, and RTP MIDI
is not frame based ... indeed, a ptime and maxptime of 0 are
perfectly valid and useful for RTP MIDI.  Zero signals a
one-command-per-packet sending style. A MIDI command
has no temporal extent -- a packet needs two commands
to have non-zero time, as a rule (that has exceptions ...).

---
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
lazzaro [at] cs [dot] berkeley [dot] edu
---


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Hi John,

The attribute is not intended to exclude non-frame based codecs. However 
it does in some degree assume that audio content has some type of duration:

       a=maxptime:<maximum packet time>

          The maximum amount of media which can be encapsulated in each
          packet, expressed as time in milliseconds.  The time SHALL be
          calculated as the sum of the time the media present in the
          packet represents.  The time SHOULD be a multiple of the frame
          size.  This attribute is probably only meaningful for audio
          data, but may be used with other media types if it makes
          sense.  It is a media attribute, and is not dependent on
          charset.  Note that this attribute was introduced after RFC
          2327, and non updated implementations will ignore this
          attribute.

We could clarify this so it want be misunderstood. I also received a 
comment from Dave Singer that "frame size" is not really correct. Thus I 
think one can modify the sentence to be more clear without affecting the 
intended semantics:

For any frame-based media, the maximum packet time SHOULD be an integer 
multiple of the frame duration.

Is that better?

Cheers

Magnus


lazzaro wrote:

> Magnus writes:
>  > In the work on updating RFC 3267 (AMR and AMR-WB RTP payload format)
>  > which originally defines "a=maxptime" I have recieved a request for
>  > clarification that I think make sense and which do not create any
>  > backwards compatibility issues:
>  >
>  > In Section 6: under attribue "a=maxptime:<maximum packet time>"
>  >
>  > I would like to rephrase the sentence:
>  > "The time SHOULD be a multiple of the frame size."
>  >
>  > To:
>  > "The time SHOULD be an integer multiple of the frame size.
> 
> 
> If maxptime is now considered a parameter for use in
> all codecs (like the "ptime" it was derived from), shouldn't
> its definition admit that not all codecs are frame based?
> 
> The RTP MIDI I-Ds use ptime and maxptime, and RTP MIDI
> is not frame based ... indeed, a ptime and maxptime of 0 are
> perfectly valid and useful for RTP MIDI.  Zero signals a
> one-command-per-packet sending style. A MIDI command
> has no temporal extent -- a packet needs two commands
> to have non-zero time, as a rule (that has exceptions ...).
> 
> ---
> John Lazzaro
> http://www.cs.berkeley.edu/~lazzaro
> lazzaro [at] cs [dot] berkeley [dot] edu
> ---

Magnus Westerlund

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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sunil bhargo wrote:

> Hi Magnus,
>           Thanks for the reply. I agree to your
> concerns regarding the option 2 of not modifying the
> Supported header by the intervening proxy or server.
> I am ok with option 1. In case of proxy chain then the
> client will receive the common set of features in the
> proxy-supported header. But what is the solution about
> clients supporting the previous versions ,that is only
> the supported header. In that case i think the proxy
> should be allowed to write the supported headers.

First if the client do not indicate support for the new specification. 
Then it will ignore the Supported header, because that is also added by 
this specification. RFC 2326 only has Require and Unsupported to make 
this negotiation. Thus I don't see a problem with supported. A RFC 2326 
client will never add it, and thus will never need to interpret it. If 
proxy adds the proxy-supported header to a request it should be ignored 
by the receiving client.

It might be an issue if there is a proxy in the chain which do not 
support the new specification. Thus one should consider using 
proxy-require: play.basic to ensure that all proxies do correctly handle 
the supported header.

Please note that it will be mandatory to handle the supported and 
proxy-supported headers.


>           Just wanted add one thing that the
> user/player should never make the bypassing decision
> based on the Supported headers feature-tags as
> inclusion of any feature-tag in Supported header
> should be optional session feature and the
> player/client should be able to live with/without this
> feature tag. If something is mandatory then it should
> be included  into require and proxy-require headers.


First, I like to clarify that if one uses Supported and proxy-supported 
it is to find out if the proxies and server do understand a feature that 
one likes to use. It might be that one requires the feature to work. 
Therefore I don't think that it is reasonable after a exchange with 
Supported and Proxy-Supported that results in that a feature is not 
supported should still do Require and Proxy-Require. Supported is 
intended to find out things before one actually tries them. Require is 
to ensure that this functionality is in place as the request actually 
requires them.

Thus as I see it a client may very well decided to bypass a proxy-chain 
based on the Proxy-Supported header.

 > Can we add the  text that the client is advised to use
 > only the common set of features in proxy-supported and
 > supported headers ?
 >

I do agree for a need to add text explaining that feature tags that are 
classified as needing support by both servers and proxies, needs to 
considered using the common set from both headers.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
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From: sunil bhargo <marx_bhargav@yahoo.com>
Subject: Re: [MMUSIC] Is Supported Header in RTSP changeable by a proxy ?
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Hi Magnus,
        thanks for the reply. I agree now on the
bypassing decision. but just one clarification below
:-

> First if the client do not indicate support for the
> new specification. 
> Then it will ignore the Supported header, because
> that is also added by
> this specification. RFC 2326 only has Require and
> Unsupported to make
> this negotiation. Thus I don't see a problem with
> supported. A RFC 2326
> client will never add it, and thus will never need
to
> interpret it. If 
> proxy adds the proxy-supported header to a request
it > should be ignored by the receiving client.

> It might be an issue if there is a proxy in the
chain > which do not 
> support the new specification. Thus one should
> consider using
> proxy-require: play.basic to ensure that all proxies
> do correctly handle
> the supported header.

> Please note that it will be mandatory to handle the
> supported and
> proxy-supported headers.
         What i meant by backward compatibility is not
with RFC2326 but with client complying to
rfc2326bis02-07. I am not sure whether a client can
say that it complies to a draft. If it can then the
client will only send the supported header and no
proxy-supported header. So should the new 
specification proxies fall to not modifying the
supported header or should it be modified.


thanks,
sunil bhargo


		
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Hi,

sunil bhargo wrote:


>          What i meant by backward compatibility is not
> with RFC2326 but with client complying to
> rfc2326bis02-07. I am not sure whether a client can
> say that it complies to a draft. If it can then the
> client will only send the supported header and no
> proxy-supported header. So should the new
> specification proxies fall to not modifying the
> supported header or should it be modified.
> 

A draft is a draft and may change at any point in time. If one starts 
implementing draft documents, one must be ready to update that 
implementation at any time when updates becomes available. Any draft 
implementation should not be commerially or large scale deployed at all. 
To be able to do good technical work we need to possibility to change 
things, thus we can't have such arguments that we need to preserver 
compatibility with a previous draft version. Especially not when draft 
still has some way to go before being ready for WG last call.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
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From mmusic-bounces@ietf.org  Wed Oct 20 10:21:54 2004
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Subject: [MMUSIC] ICE Issue: mid-session connectivity checks
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Folks,

One of the big changes in the -02 ICE spec was the addition of 
mid-session connectivity checks. These checks served several purposes:

1. detection of a network failure that causes an interface to become 
unreachable

2. in the case of a failure of a nat in the middle of the network, if 
the nat recovers, or a backup takes its place, a new binding will get 
assigned, and it can be discovered


The particular motivating use case for 2 is an enterprise that is 
multihomed. It has a NAT on each of its two interfaces to its providers. 
The enterprise uses the address prefix provided from provider A, and 
advertises reachability to this prefix into provider B. A client within 
the enterprise makes a call. The NAT and/or connection to provider A 
fails. Routing now causes outbound UDP packets to go over the interface 
to provider B. So, outbound UDP packets from the client now go through 
this other NAT. It assigns a new binding. With mid-session checks, the 
new binding can be learned by the peer, and communications established 
using it.

Now, the trick is that, if we are interested in VoIP as a driving use 
case for this, the amount of time to discover this has to be relatively 
short. If it takes 30 seconds, the users in the call would have already 
hung up. That requires the connectivity checks to be sent at a rapid 
clip. Furthermore, in the case of such a failure, signaling would fail 
too in all likelihood (depending where the servers are), which would 
require some SIP operations.

So, the issue is, is it worth it? I have gotten a few complaints about 
the complexity introduced by the mid-session checks. Thus, we have three 
choices as I see it:

1. remove the checks completely

ICE runs only on setup, to do an initial check. RTCP would, for example, 
allow you to detect some of these failure situations mid-call. Recovery 
would require re-establishing the call.

2. keep the checks

THe draft stays as is

3. Allow for the usage of the checks to be negotiated

In case a client doesn't want to implement support for the mid-session 
checks, we make them optional and allow it to be signaled.


I'd prefer NOT to do 3, in the interest of avoiding protocol options. 
The argument in favor of 1 is primarily one of simplification, and that 
you will have to make these connectivity checks really fast in order to 
be responsive enough to really recover from a failure, which could be a 
problem.

As I constantly hear these days that "ICE is too complicated", in the 
interests of simplification, I'd be game to remove the feature. It could 
be added back in as an extension later if needed.

Comments?

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


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Subject: [MMUSIC] ICE issue: STUN vs. STUN over RTP
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Since the last IETF, several folks have independently suggested to me 
that we tunnel STUN over RTP, and use that for the connectivity checks. 
The reasoning is that it avoids the ugliness of multiplexing STUN and 
RTP on the same port. There are pros and cons to doing this:

Cons:

- STUN responses would end up being in RTP packets sent in the reverse 
direction, even though RTP is not really a request/response protocol, so 
its a mismatch there

- You'd need to also tunnel STUN over RTCP somehow (app packets?), which 
is getting really ugly

- STUN packets aren't RTP, and things like statistics would get somewhat 
distorted (less severe if you don't do mid-session checks per my other 
note). There would be jumps in timestamps that don't really make sense 
since STUN really isn't real time media

- If SRTP is in use, you couldn't process the STUN packets until the 
SRTP keying had been done. Thats not a big deal, in that ICE requires 
the exchange of username portions to complete within SIP before the STUN 
packets can be processed

- makes the solution RTP specific

Pros:

+ avoids demuxing STUN and RTP on the same port




I think the cons really outweigh the pros, especially if we remove the 
mid-session checks. As such, I'd like to keep the current STUN approach.

Comments?
-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@dynamicsoft.com                        FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
http://www.cisco.com


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From: sunil bhargo <marx_bhargav@yahoo.com>
Subject: Re: [MMUSIC] Is Supported Header in RTSP changeable by a proxy ?
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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We have resoloution then if nobody has any additions.

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

> Hi,
> 
> sunil bhargo wrote:
> 
> 
> >          What i meant by backward compatibility is
> not
> > with RFC2326 but with client complying to
> > rfc2326bis02-07. I am not sure whether a client
> can
> > say that it complies to a draft. If it can then
> the
> > client will only send the supported header and no
> > proxy-supported header. So should the new
> > specification proxies fall to not modifying the
> > supported header or should it be modified.
> > 
> 
> A draft is a draft and may change at any point in
> time. If one starts 
> implementing draft documents, one must be ready to
> update that 
> implementation at any time when updates becomes
> available. Any draft 
> implementation should not be commerially or large
> scale deployed at all. 
> To be able to do good technical work we need to
> possibility to change 
> things, thus we can't have such arguments that we
> need to preserver 
> compatibility with a previous draft version.
> Especially not when draft 
> still has some way to go before being ready for WG
> last call.
> 
> Cheers
> 
> Magnus Westerlund
> 
> 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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Subject: [MMUSIC] Re: draft-ietf-mmusic-kmgmt-ext-11.txt: KMID? IANA
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Hi again Martin,

Thanks for your additional thoughts. Indeed, I think we
are getting closer, but as you write, there may still be some
mutual misunderstandings before we "meet". Hopefully, some more
elaboration below brings us even closer...


Euchner Martin wrote:


 > Many thanks to you and the entire MIKEY crew for answering my question.
 >
 > On one hand, I understand your response (see my acknowledgement
 >below), one the other hand put some further comments for discussion and
 >yet in some other areas I believe that there may be just a resolvable=20
 >misunderstanding.
 >
 > I'm addressing the issues inline below.
 >
 > Cheers
 >
 > Martin
 >
 >  -----Original Message-----
 > From:   Mats N=E4slund [mailto:mats.naslund@ericsson.com]
 > Sent:   Montag, 18. Oktober 2004 11:39
 > To:     Euchner Martin
 > Cc:     mmusic@ietf.org; Jari Arkko (JO/LMF); Elisabetta Carrara=20
(KI/EAB); Fredrik Lindholm (KI/EAB); Karl Norrman (KI/EAB)
 >
 > Subject:        Re: draft-ietf-mmusic-kmgmt-ext-11.txt: KMID? IANA=20
considerations section
 >
 >  >
 >  > Assuming an example where the offerer advertises just 3 MIKEY
 > protocols; e.g. MIKEY-preshared, MIKEY-DHsign and MIKEY-PK,
 >  >
 >  > Then the SDP would hold something like
 >  > a=3Dkey-mgmt:mikey   a1234a54...
 >  > a=3Dkey-mgmt:mikey   b454646ca54...
 >  > a=3Dkey-mgmt:mikey   c95785656ef3....
 >
 >
 > First of all, why would this be something that ever occurs?
 > The reason we allow several options is to let the parties
 > agree on a particular prefered *protocol*, but choosing options
 > within one and the same protocol is not intended to be done
 > at this level.
 >
 > MEU:> Ok this confirms that kmtext "negotiates" a key management
 >protocol family, but does not actually negotiate any particular key
 >management capability. I intentionally put negotiate in quotes, since


Well, we would e.g. assume that if a shared key is known, that method
would in general be preferred (even if you by chance also know a public
key), and the need to negotiate in that "dimension" would be small.

Still, we think also that sort of negotiation *could* be done
with a security that is not lower than if just single offers were
allowed, see below.


 > this SDP-based negotiation is operating just on the family - level of=20
 >the key management protocol but certainly not on protocol capabilities
 >(such as a particular choice like MIKEY-PK, or even more fine grained
 >on a crypto algorithm level). So in order to "negotiate" any deeper
 >than the family-level, kmgtext offers the desired MIKEY protocol
 >variants in parallel. Unfortunately, this increases the processing time
 >of the offerer considerably; namely 3 times to instantiate and to
 >compile 3 different MIKEY key management protocol messages. For the=20
 >answerer, the work load is not as high as on the offerer, but still the
 >base64 decoding requires some more effort than just a quick inspection
 >of a specific KMID would cost; so this latter point is really a minor=20
 >issue.


Yes, if you really don't have any clue
what the responder supports/prefers, you *could* trade the number of
single messages for the amount of processing involved in a producing a
single, but larger multi-offer.


 >
 > I should clearly say, that I don't see any better alternative for a
 >full 2-way handshake negotiation process yet a 4-way handshake would
 >not face the parallel explosion property.
 >
 >
 > (Anyway, let's assume it can happen...)
 >
 >  >
 >  > and the generated protocol list would be "mikey;mikey;mikey", where
 > 'mikey' represents the allocated KMID value for the MIKEY protocol=20
family.
 >
 >
 > Right.
 >
 >  >
 >  > Q1) In the context of bidding-down protection, what kind of securit=
y
 > does this list actually provide?
 >  >
 >  > I believe that an attacker could easily substitute another, weak 4t=
h
 > mikey key management protocol ("mikey-password") against one of the
 > other three offered ones; yet the (MIKEY) protected protocol list woul=
d
 > remain the same and the answer would not be able to detect the attack.
 >
 >
 > Okay, let's assume he can. To accomplish this, the "mikey-password"
 > variant needs to be so weak that he can fake the authentiction data of
 > the a-line where this is inserted. That is, the authentication mechani=
sm
 > of the password variant is completely insecure. But if this is so, the
 > attacker does not even need the "real" Initiator to be on-line: he can
 > at any  point in time impersonate the real Initiator, faking MIKEY
 > messages using the password variant. Therefore...
 >
 > MEU:> Here I sense a minor misunderstanding. I'm clearly not
 >questioning the security of any MIKEY protocol nor am I saying that any
 >MIKEY protocol is any close to being weak that it could be broken.
 >
 > My only concern in the context of a bidding-down attack is, that the=20
 >bidding-down attack could be accomplished within a spurious hop

What do you mean by "spurious hop" ?

 >somewhere between the originating and terminating end systems, such
 >that the attacker replaces a stronger offer by a weaker one of the same
 >protocol family. For the bidding-down attack success, the strength of

Neglecting that we don't understand "spurious hop", we still think
that the attack you describe is of the following form:

1. The Initiator makes an offer containing m > 1 MIKEY
    alternatives/lines.

2. Somehow, along the path, at least one of the m MIKEY offers
    get replaced by another (weaker) MIKEY offer that was not in
    the original offer.

Correct?

If so, let's call the weak MIKEY variant "MIKEY-X", be it password
based or otherwise.

Now we need to ask, how could it be that the inserted MIKEY-X offer gets
accepted as authentic by the Responder?

As we see it, that would only be possible if either:

A) the MIKEY-X offer is a replay of an old offer (which would normally
   be taken care of by MIKEY-X replay protection), or,

B) The MIKEY-X option is so weak in itself that it is possible
   to fake MIKEY-X messages without even resorting on replay.

But if either of these would be possible, there would be
a flaw in that specific MIKEY-X variant, and it really doesn't
matter that there were more than one offer in the original
message: the attacker could fake MIKEY-X offers according to that
(single) MIKEY-X version at any point in time, right? The real
Initiator doesn't even need to be on-line. Thus, you *could* say that
you have a bidding down attack, but that is not *the* real problem.
The problem is more fundamental than that.

Also, in addition (as we seem to agree), the bidding-down variant of the=20
attack can be discovered using the verification option. (However, the
more basic "faked-offer" is not mitigated by the V-message.)

 > the substituted MIKEY protocol matters only as much as its security is
 >(somewhat) less than the original protocols. If the reduced security >
 >level is sufficient to break MIKEY in any reasonable time, is a
 >completely different issue.


We think (see above) that it isn't enough if the substituted offers
are only "somewhat" less secure, they would need to be quite badly
insecure, otherwise the substitution as such would be impossible.

 >
 > The protected protocol list would not indicate the particular=20
 >substitution attack.

Correct. But even single offers containing only one MIKEY line
could be faked.

 >
 > However, I admit that there are two mechanisms in place that allow the
 >offerer to detect such an attack:
 > a) The offerer would receive a SDP response from the answerer that
 >holds a "foreign", non-offered kmgt option.
 > b) MIKEY luckily provides the verification message, unless the MIKEY
 >key management protocol isn't already two-way.
 >
 > Mechanism a) or b) are the only way for the offerer to detect any
 >occurred bidding-down attack; the responder has no chance to detect
 >such an attack just by inspecting the incoming offer.

Yes. But even so, attacks remain possible by just faking *single* offers
based on the weak variant.

 >
 > For myself this fact is now clear. So my conclusion is that in fact
 >the MIKEY verification message has a very strong value in protecting
 >the offerer against bidding-down attacks. I simply cannot remember
 >right now, if this has been adequately explained in MIKEY RFC or in the
 >kmext draft and if that is apparent to the reader.


Maybe not...but this was clearly one reason for adding it as an option.

 >  >
 >  > My conclusion (needs confirmation) is that the design flaw is that
 > the KMID is on a key management protocol family but it should rather
 >be
 > on the key management protocol option/varient.
 >  >
 >  > For MIKEY, the fix should be straightforward: Allocate a KMID for
 > each of the MIKEY protocol variants; i.e.
 >  > One for MIKEY-preshared, one for MIKEY-DHsign and one for MIKEY-PK.
 >
 >
 > ...this protection mechanism is totally inadequate for the case
 > you describe, the "real" security hole remains open even if
 > we were to add this feature.
 >
 > MEU:> hmm, I'm not certain if I understand this fully.
 > Except for the problem that you correctly describe in the following
 >paragraph, I believe that my proposed solution could at least provide
 >some value for the answerer to get protection against bidding down
 >attacks on the MIKEY protocol level, yet clearly not on any
 >transform/alg level below.

It is not clear it helps, if there is even a single very insecure
variant. Also, if a single, otherwise secure, variant contains
insecure "sub-options" you have an issue. We are therefore sceptic
to what it adds.

 >
 > This would challenge us to answer the question: Is the proposed yet
 >non-perfect approach (i.e. allocating several KMIDs) worth the effort?
 >
 > Maybe...when we don't want the entire security protection to depend
 >just on the offerer, but give the answerer a certain chance at least to
 >recover on its own.
 >
 > I put this aspect for further discussion among the security experts.
 >

We are listening to other possible feedback. We hope we have
clarified why *we* think the suggested approach protects against
"relevant" attacks.

 >
 > Another scenario in which the suggested protection is not enough
 > is as follows: suppose preshared-MIKEY is extended by a cipher/MAC
 > combination which has poor security. It may then not be enough to
 > authenticate the fact MIKEY-preshared was offered, but rather the
 > exact options at the transform level (which *is* done in each a-line
 > individually, but not across a-lines).
 > MEU:> Yes I completely agree to this point. Taken to the most extreme
 >case this would mean that the entire offering of security parameters
 >would need to be secured. I clearly think that this is beyond what we=20
 >want to achieve.

Ok.


 >
 > However, we would exclude this
 > as a threat for the same reason as above: if faked MIKEY lines can
 > be inserted by an attacker, the problem is not of bidding-down nature,
 > it is really a fundamental security flaw of the protocol as such.
 >
 > MEU:> not quite. Please see my explanations on the spurious hop, >abov=
e.

We don't see how that could happen, or, to be precise, if it
*could* happen, even simpler, non-bidding-down attacks would
occur.

 >
 > Finally, if one is still concerned by this, we would suggest that
 > the Initiator uses MIKEY's option of requesting a verification message
 > back from the Responder. If an attacker modifies the offer to contain
 > something the Initiator actually didn't offer, the response back will
 > be generated by an incorrect method/key, and the Initiator will
 > detect this when checking the response.
 >
 > MEU:> Yes full agreement. Clearly the verification mechanism is a very
 >powerful tool to protect the offerer from a bidding down attack.
 >However, the answerer fully depends on the correct reaction and
 >implementation of the offerer. The answerer has no means to detect an
 >occurred bidding-down attack on its own.
 >

Fully agree.

Cheers

/Mats and MIKEY gang





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Date: Wed, 20 Oct 2004 10:42:49 -0700
From: lazzaro <lazzaro@eecs.berkeley.edu>
Subject: Re: [MMUSIC] maxptime
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On Oct 19, 2004, at 11:51 PM, Magnus Westerlund wrote:
> The attribute is not intended to exclude non-frame based codecs.
> However it does in some degree assume that audio content has some type 
> of duration:

The RTP MIDI payload's duration is always has a defined value.
It just so happens that 0 ms is an acceptable value.

So, as long as 0 ms is an acceptable maxptime duration its OK
by me. 0 ms needn't be explicitly permitted in the text, but it should 
not
be explicitly forbidden.

> For any frame-based media, the maximum packet time SHOULD be an 
> integer multiple of the frame duration.
>
> Is that better?

Yes, sounds good to me.

---
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
lazzaro [at] cs [dot] berkeley [dot] edu
---


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From: Euchner Martin <martin.euchner@siemens.com>
To: =?ISO-8859-1?Q?=27Mats_N=E4slund=27?= <mats.naslund@ericsson.com>,
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Mats,

Once again many thanks for your good answers that indeed clarify a lot of the misunderstandings.

To quickly summarize: I believe that we are already very close in our views; I don't see any real disagreement.

However, I see some further related issues on how the answerer has implemented its system. Originally, I had the "insecure implementation" in mind, yet it was not immediately apparent to me that this is the main cause of the problem. I would at least ask for your confirmation, if that is correct; see the longer added statements with the counter example in the middle of the email.

Cherio

Martin.

> MEU:> Ok this confirms that kmtext "negotiates" a key management
 >protocol family, but does not actually negotiate any particular key
 >management capability. I intentionally put negotiate in quotes, since


Well, we would e.g. assume that if a shared key is known, that method
would in general be preferred (even if you by chance also know a public
key), and the need to negotiate in that "dimension" would be small.

Still, we think also that sort of negotiation *could* be done
with a security that is not lower than if just single offers were
allowed, see below.


MEU: ok.

> My only concern in the context of a bidding-down attack is, that the 
 >bidding-down attack could be accomplished within a spurious hop

What do you mean by "spurious hop" ?

MEU:> sorry for not having explained the attack scenario. It is exactly the case that you showed in 1. and in 2, where an evil Eve who is not trusted by the offerer nor the answerer runs a modification/substitution attack on at least one SDP a-line.

 >somewhere between the originating and terminating end systems, such
 >that the attacker replaces a stronger offer by a weaker one of the same
 >protocol family. For the bidding-down attack success, the strength of

Neglecting that we don't understand "spurious hop", we still think
that the attack you describe is of the following form:

1. The Initiator makes an offer containing m > 1 MIKEY
    alternatives/lines.

2. Somehow, along the path, at least one of the m MIKEY offers
    get replaced by another (weaker) MIKEY offer that was not in
    the original offer.

Correct?

MEU:> yes exactly.

If so, let's call the weak MIKEY variant "MIKEY-X", be it password
based or otherwise.

Now we need to ask, how could it be that the inserted MIKEY-X offer gets
accepted as authentic by the Responder?

As we see it, that would only be possible if either:

A) the MIKEY-X offer is a replay of an old offer (which would normally
   be taken care of by MIKEY-X replay protection), or,

MEU: actually I had in mind just this case as I did not assume case B to happen (so my assumption is also along the lines that MIKEY-X still provides some meaningful security level, yet it is possibly lower in security level than any of the current MIKEY protocol variants.)

B) The MIKEY-X option is so weak in itself that it is possible
   to fake MIKEY-X messages without even resorting on replay.

MEU:> You are right that this case would also exist.

But if either of these would be possible, there would be
a flaw in that specific MIKEY-X variant, and it really doesn't
matter that there were more than one offer in the original
message: the attacker could fake MIKEY-X offers according to that
(single) MIKEY-X version at any point in time, right? The real
Initiator doesn't even need to be on-line. Thus, you *could* say that
you have a bidding down attack, but that is not *the* real problem.
The problem is more fundamental than that.

MEU: I tend to agree with that given that Eve somehow must have substituted actually a replayed MIKEY protocol run. I really cannot think of any much different attack right now.
Now if Eve had done so, then the answerer could detect this attack, by verification that the integrity of >each< offered MIKEY protocol holds valid. For the replayed option, the verification would fail (e.g. re-used nonce or expired timestamp etc).
So I agree with you that the answerer has a real chance to detect a bidding-down attack. This is good and removes my original concern.

New concern: Unfortunately, this comes at a high price; namely the answerer MUST implement all MIKEY protocol variants even those that MIKEY calls just optional to implement such as MIKEY-DHSIGN,  in order to be able to protect against bidding-down attacks. Right?

I cannot fully believe this to be true right now, as this questions the choice of making certain MIKEY protocols optional altogether.

Why is it so?

So the answerer MUST first verify all received MIKEY protocol offers, and just after that actually would decide which option to choose (authenticate first then policy decide).
In fact, doing it the other way round (i.e. 1st make protocol choice then verify that the choice is authentic) would be insecure, even if the choice is not the replayed protocol offer! See counter example:

Assume that the offerer provides 3 alternatives: A1, A2, A3 where A1 < A2 < A3 and "<" means security level is lower than. Assume further that Eve substitutes B1 against A1 somewhere between the offerer and the answerer and that B1 < A1. 
Now assume further the security policy of the answerer is such that the answerer picks the medium offered option; i.e. discarding the highest and the lowest offered option; so in the case without attack, the decision taken is A2. (Please don't ask me if such a security policy is any realistic or why such a policy would make sense).

In the case where Eve attacks with a substitute B1 for A3, and the answerer does not verify the security validity first of A1, B1, and A2, the policy decision would pick A1 now. The answerer would not detect the successful bidding-down attack just by verifying A1 which is of course correct. The offerer is also not able to detect the occurred bidding-down attack even with a verification message response.

Unfortunately, the insecure variant is so much more compelling from the performance and implementation point of view that incorrect implementations could be tricked at least in certain cases depending on the security policy.

Unless I'm badly missing the point here (perhaps it's too late already?), I believe the correct behaviour of the answerer is not immediately obvious to everyone; and thus, it may be more than worth to document the correct behaviour in the security consideration section.

BTW: I also forgot to clarify in my previous email, what I understand as a successful bidding-down attack: This is an outcome of a negotiation procedure where the offerer and answerer (with the "help" of some attacker) agree on a security option that yields a less security level than without the attacker. The successful bidding-down attack does not imply that any security protocol has been broken nor that the reduced security option would actually be subject to a (feasible) attack.

Also, in addition (as we seem to agree), the bidding-down variant of the 
attack can be discovered using the verification option. (However, the
more basic "faked-offer" is not mitigated by the V-message.)

 > the substituted MIKEY protocol matters only as much as its security is
 >(somewhat) less than the original protocols. If the reduced security >
 >level is sufficient to break MIKEY in any reasonable time, is a
 >completely different issue.


We think (see above) that it isn't enough if the substituted offers
are only "somewhat" less secure, they would need to be quite badly
insecure, otherwise the substitution as such would be impossible.

 >
MEU: I agree with you, except the issue explained above.

 > The protected protocol list would not indicate the particular 
 >substitution attack.

Correct. But even single offers containing only one MIKEY line
could be faked.

MEU:> Yes ok, yet such cases could be detected with the V-message.

 >
 > However, I admit that there are two mechanisms in place that allow the
 >offerer to detect such an attack:
 > a) The offerer would receive a SDP response from the answerer that
 >holds a "foreign", non-offered kmgt option.
 > b) MIKEY luckily provides the verification message, unless the MIKEY
 >key management protocol isn't already two-way.
 >
 > Mechanism a) or b) are the only way for the offerer to detect any
 >occurred bidding-down attack; the responder has no chance to detect
 >such an attack just by inspecting the incoming offer.

Yes. But even so, attacks remain possible by just faking *single* offers
based on the weak variant.

MEU:> ok.

 >
 > For myself this fact is now clear. So my conclusion is that in fact
 >the MIKEY verification message has a very strong value in protecting
 >the offerer against bidding-down attacks. I simply cannot remember
 >right now, if this has been adequately explained in MIKEY RFC or in the
 >kmext draft and if that is apparent to the reader.


Maybe not...but this was clearly one reason for adding it as an option.

MEU: ok. 

 >  >
 >  > My conclusion (needs confirmation) is that the design flaw is that
 > the KMID is on a key management protocol family but it should rather
 >be
 > on the key management protocol option/varient.
 >  >
 >  > For MIKEY, the fix should be straightforward: Allocate a KMID for
 > each of the MIKEY protocol variants; i.e.
 >  > One for MIKEY-preshared, one for MIKEY-DHsign and one for MIKEY-PK.
 >
 >
 > ...this protection mechanism is totally inadequate for the case
 > you describe, the "real" security hole remains open even if
 > we were to add this feature.
 >
 > MEU:> hmm, I'm not certain if I understand this fully.
 > Except for the problem that you correctly describe in the following
 >paragraph, I believe that my proposed solution could at least provide
 >some value for the answerer to get protection against bidding down
 >attacks on the MIKEY protocol level, yet clearly not on any
 >transform/alg level below.

It is not clear it helps, if there is even a single very insecure
variant. Also, if a single, otherwise secure, variant contains
insecure "sub-options" you have an issue. We are therefore sceptic
to what it adds.

MEU:> Yes I understand your hesitations now.

 >
 > This would challenge us to answer the question: Is the proposed yet
 >non-perfect approach (i.e. allocating several KMIDs) worth the effort?
 >
 > Maybe...when we don't want the entire security protection to depend
 >just on the offerer, but give the answerer a certain chance at least to
 >recover on its own.
 >
 > I put this aspect for further discussion among the security experts.
 >

We are listening to other possible feedback. We hope we have
clarified why *we* think the suggested approach protects against
"relevant" attacks.

MEU:> I appreciate any further views.
 >
 > Another scenario in which the suggested protection is not enough
 > is as follows: suppose preshared-MIKEY is extended by a cipher/MAC
 > combination which has poor security. It may then not be enough to
 > authenticate the fact MIKEY-preshared was offered, but rather the
 > exact options at the transform level (which *is* done in each a-line
 > individually, but not across a-lines).
 > MEU:> Yes I completely agree to this point. Taken to the most extreme
 >case this would mean that the entire offering of security parameters
 >would need to be secured. I clearly think that this is beyond what we 
 >want to achieve.

Ok.


 >
 > However, we would exclude this
 > as a threat for the same reason as above: if faked MIKEY lines can
 > be inserted by an attacker, the problem is not of bidding-down nature,
 > it is really a fundamental security flaw of the protocol as such.
 >
 > MEU:> not quite. Please see my explanations on the spurious hop, >above.

We don't see how that could happen, or, to be precise, if it
*could* happen, even simpler, non-bidding-down attacks would
occur.

MEU:> yes, possibly. I've not analyzed the situation what all could go wrong, if Eve actually attacks the SDP but I believe it could get pretty nasty from the security point of view.

 >
 > Finally, if one is still concerned by this, we would suggest that
 > the Initiator uses MIKEY's option of requesting a verification message
 > back from the Responder. If an attacker modifies the offer to contain
 > something the Initiator actually didn't offer, the response back will
 > be generated by an incorrect method/key, and the Initiator will
 > detect this when checking the response.
 >
 > MEU:> Yes full agreement. Clearly the verification mechanism is a very
 >powerful tool to protect the offerer from a bidding down attack.
 >However, the answerer fully depends on the correct reaction and
 >implementation of the offerer. The answerer has no means to detect an
 >occurred bidding-down attack on its own.
 >

Fully agree.

MEU> Excellent!

Cheers

/Mats and MIKEY gang




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To: "'Jonathan Rosenberg'" <jdrosen@dynamicsoft.com>,
        "'IETF MMUSIC WG'"
	<mmusic@ietf.org>
Subject: RE: [MMUSIC] ICE Issue: mid-session connectivity checks
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I vote for 1.

Note that 1 could be dealt with at the application layer. So while it does
indeed
mean that the session would be re-established, it doen't NECESSARILY mean
that 
the use would hang-up the call. You could in theory re-establish it without
hanging-up.

So instead of saying "Recovery would require re-establishing the call" I'd
say
"Recovery would require re-establishing the session".

In fact, couldn't we just do a new Offer/Answer?




> -----Original Message-----
> From: mmusic-bounces@ietf.org 
> [mailto:mmusic-bounces@ietf.org] On Behalf Of Jonathan Rosenberg
> Sent: Wednesday, October 20, 2004 05:46
> To: IETF MMUSIC WG
> Subject: [MMUSIC] ICE Issue: mid-session connectivity checks
> 
> 
> Folks,
> 
> One of the big changes in the -02 ICE spec was the addition of 
> mid-session connectivity checks. These checks served several purposes:
> 
> 1. detection of a network failure that causes an interface to become 
> unreachable
> 
> 2. in the case of a failure of a nat in the middle of the network, if 
> the nat recovers, or a backup takes its place, a new binding will get 
> assigned, and it can be discovered
> 
> 
> The particular motivating use case for 2 is an enterprise that is 
> multihomed. It has a NAT on each of its two interfaces to its 
> providers. 
> The enterprise uses the address prefix provided from provider A, and 
> advertises reachability to this prefix into provider B. A 
> client within 
> the enterprise makes a call. The NAT and/or connection to provider A 
> fails. Routing now causes outbound UDP packets to go over the 
> interface 
> to provider B. So, outbound UDP packets from the client now 
> go through 
> this other NAT. It assigns a new binding. With mid-session 
> checks, the 
> new binding can be learned by the peer, and communications 
> established 
> using it.
> 
> Now, the trick is that, if we are interested in VoIP as a driving use 
> case for this, the amount of time to discover this has to be 
> relatively 
> short. If it takes 30 seconds, the users in the call would 
> have already 
> hung up. That requires the connectivity checks to be sent at a rapid 
> clip. Furthermore, in the case of such a failure, signaling 
> would fail 
> too in all likelihood (depending where the servers are), which would 
> require some SIP operations.
> 
> So, the issue is, is it worth it? I have gotten a few 
> complaints about 
> the complexity introduced by the mid-session checks. Thus, we 
> have three 
> choices as I see it:
> 
> 1. remove the checks completely
> 
> ICE runs only on setup, to do an initial check. RTCP would, 
> for example, 
> allow you to detect some of these failure situations 
> mid-call. Recovery 
> would require re-establishing the call.
> 
> 2. keep the checks
> 
> THe draft stays as is
> 
> 3. Allow for the usage of the checks to be negotiated
> 
> In case a client doesn't want to implement support for the 
> mid-session 
> checks, we make them optional and allow it to be signaled.
> 
> 
> I'd prefer NOT to do 3, in the interest of avoiding protocol options. 
> The argument in favor of 1 is primarily one of 
> simplification, and that 
> you will have to make these connectivity checks really fast 
> in order to 
> be responsive enough to really recover from a failure, which 
> could be a 
> problem.
> 
> As I constantly hear these days that "ICE is too complicated", in the 
> interests of simplification, I'd be game to remove the 
> feature. It could 
> be added back in as an extension later if needed.
> 
> Comments?
> 
> -Jonathan R.
> -- 
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ 
> 07054-2711
> Cisco Systems
> jdrosen@dynamicsoft.com                        FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
> 
> 
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<TITLE>RE: [MMUSIC] ICE Issue: mid-session connectivity checks</TITLE>
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<P><FONT SIZE=2>I vote for 1.</FONT>
</P>

<P><FONT SIZE=2>Note that 1 could be dealt with at the application layer. So while it does indeed</FONT>
<BR><FONT SIZE=2>mean that the session would be re-established, it doen't NECESSARILY mean that </FONT>
<BR><FONT SIZE=2>the use would hang-up the call. You could in theory re-establish it without</FONT>
<BR><FONT SIZE=2>hanging-up.</FONT>
</P>

<P><FONT SIZE=2>So instead of saying &quot;Recovery would require re-establishing the call&quot; I'd say</FONT>
<BR><FONT SIZE=2>&quot;Recovery would require re-establishing the session&quot;.</FONT>
</P>

<P><FONT SIZE=2>In fact, couldn't we just do a new Offer/Answer?</FONT>
</P>
<BR>
<BR>
<BR>

<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: mmusic-bounces@ietf.org </FONT>
<BR><FONT SIZE=2>&gt; [<A HREF="mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</A>] On Behalf Of Jonathan Rosenberg</FONT>
<BR><FONT SIZE=2>&gt; Sent: Wednesday, October 20, 2004 05:46</FONT>
<BR><FONT SIZE=2>&gt; To: IETF MMUSIC WG</FONT>
<BR><FONT SIZE=2>&gt; Subject: [MMUSIC] ICE Issue: mid-session connectivity checks</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Folks,</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; One of the big changes in the -02 ICE spec was the addition of </FONT>
<BR><FONT SIZE=2>&gt; mid-session connectivity checks. These checks served several purposes:</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 1. detection of a network failure that causes an interface to become </FONT>
<BR><FONT SIZE=2>&gt; unreachable</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 2. in the case of a failure of a nat in the middle of the network, if </FONT>
<BR><FONT SIZE=2>&gt; the nat recovers, or a backup takes its place, a new binding will get </FONT>
<BR><FONT SIZE=2>&gt; assigned, and it can be discovered</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; The particular motivating use case for 2 is an enterprise that is </FONT>
<BR><FONT SIZE=2>&gt; multihomed. It has a NAT on each of its two interfaces to its </FONT>
<BR><FONT SIZE=2>&gt; providers. </FONT>
<BR><FONT SIZE=2>&gt; The enterprise uses the address prefix provided from provider A, and </FONT>
<BR><FONT SIZE=2>&gt; advertises reachability to this prefix into provider B. A </FONT>
<BR><FONT SIZE=2>&gt; client within </FONT>
<BR><FONT SIZE=2>&gt; the enterprise makes a call. The NAT and/or connection to provider A </FONT>
<BR><FONT SIZE=2>&gt; fails. Routing now causes outbound UDP packets to go over the </FONT>
<BR><FONT SIZE=2>&gt; interface </FONT>
<BR><FONT SIZE=2>&gt; to provider B. So, outbound UDP packets from the client now </FONT>
<BR><FONT SIZE=2>&gt; go through </FONT>
<BR><FONT SIZE=2>&gt; this other NAT. It assigns a new binding. With mid-session </FONT>
<BR><FONT SIZE=2>&gt; checks, the </FONT>
<BR><FONT SIZE=2>&gt; new binding can be learned by the peer, and communications </FONT>
<BR><FONT SIZE=2>&gt; established </FONT>
<BR><FONT SIZE=2>&gt; using it.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Now, the trick is that, if we are interested in VoIP as a driving use </FONT>
<BR><FONT SIZE=2>&gt; case for this, the amount of time to discover this has to be </FONT>
<BR><FONT SIZE=2>&gt; relatively </FONT>
<BR><FONT SIZE=2>&gt; short. If it takes 30 seconds, the users in the call would </FONT>
<BR><FONT SIZE=2>&gt; have already </FONT>
<BR><FONT SIZE=2>&gt; hung up. That requires the connectivity checks to be sent at a rapid </FONT>
<BR><FONT SIZE=2>&gt; clip. Furthermore, in the case of such a failure, signaling </FONT>
<BR><FONT SIZE=2>&gt; would fail </FONT>
<BR><FONT SIZE=2>&gt; too in all likelihood (depending where the servers are), which would </FONT>
<BR><FONT SIZE=2>&gt; require some SIP operations.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; So, the issue is, is it worth it? I have gotten a few </FONT>
<BR><FONT SIZE=2>&gt; complaints about </FONT>
<BR><FONT SIZE=2>&gt; the complexity introduced by the mid-session checks. Thus, we </FONT>
<BR><FONT SIZE=2>&gt; have three </FONT>
<BR><FONT SIZE=2>&gt; choices as I see it:</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 1. remove the checks completely</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; ICE runs only on setup, to do an initial check. RTCP would, </FONT>
<BR><FONT SIZE=2>&gt; for example, </FONT>
<BR><FONT SIZE=2>&gt; allow you to detect some of these failure situations </FONT>
<BR><FONT SIZE=2>&gt; mid-call. Recovery </FONT>
<BR><FONT SIZE=2>&gt; would require re-establishing the call.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 2. keep the checks</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; THe draft stays as is</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; 3. Allow for the usage of the checks to be negotiated</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; In case a client doesn't want to implement support for the </FONT>
<BR><FONT SIZE=2>&gt; mid-session </FONT>
<BR><FONT SIZE=2>&gt; checks, we make them optional and allow it to be signaled.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I'd prefer NOT to do 3, in the interest of avoiding protocol options. </FONT>
<BR><FONT SIZE=2>&gt; The argument in favor of 1 is primarily one of </FONT>
<BR><FONT SIZE=2>&gt; simplification, and that </FONT>
<BR><FONT SIZE=2>&gt; you will have to make these connectivity checks really fast </FONT>
<BR><FONT SIZE=2>&gt; in order to </FONT>
<BR><FONT SIZE=2>&gt; be responsive enough to really recover from a failure, which </FONT>
<BR><FONT SIZE=2>&gt; could be a </FONT>
<BR><FONT SIZE=2>&gt; problem.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; As I constantly hear these days that &quot;ICE is too complicated&quot;, in the </FONT>
<BR><FONT SIZE=2>&gt; interests of simplification, I'd be game to remove the </FONT>
<BR><FONT SIZE=2>&gt; feature. It could </FONT>
<BR><FONT SIZE=2>&gt; be added back in as an extension later if needed.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Comments?</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; -Jonathan R.</FONT>
<BR><FONT SIZE=2>&gt; -- </FONT>
<BR><FONT SIZE=2>&gt; Jonathan D. Rosenberg, Ph.D.&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 600 Lanidex Plaza</FONT>
<BR><FONT SIZE=2>&gt; Director, Service Provider VoIP Architecture&nbsp;&nbsp; Parsippany, NJ </FONT>
<BR><FONT SIZE=2>&gt; 07054-2711</FONT>
<BR><FONT SIZE=2>&gt; Cisco Systems</FONT>
<BR><FONT SIZE=2>&gt; jdrosen@dynamicsoft.com&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; FAX:&nbsp;&nbsp; (973) 952-5050</FONT>
<BR><FONT SIZE=2>&gt; <A HREF="http://www.jdrosen.net" TARGET="_blank">http://www.jdrosen.net</A>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; PHONE: (973) 952-5000</FONT>
<BR><FONT SIZE=2>&gt; <A HREF="http://www.cisco.com" TARGET="_blank">http://www.cisco.com</A></FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; _______________________________________________</FONT>
<BR><FONT SIZE=2>&gt; mmusic mailing list</FONT>
<BR><FONT SIZE=2>&gt; mmusic@ietf.org</FONT>
<BR><FONT SIZE=2>&gt; <A HREF="https://www1.ietf.org/mailman/listinfo/mmusic" TARGET="_blank">https://www1.ietf.org/mailman/listinfo/mmusic</A></FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
</P>

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To: Awais Ali <aali@i2telecom.com>
Subject: Re: [MMUSIC] purpose of derived-from in ICE SDP attribute
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This attribute is a leftover from previous versions and is not needed. 
It'll be removed in the next rev of the draft.

Thanks,
Jonathan R.

Awais Ali wrote:

> Hi all,
> 
> I can't seem to figure out the purpose of  derived-from attribute in 
> section 9 of draft-ietf-mmusic-ice-02. It is mentioned anywhere in the 
> draft. Can anybody explain that to me?
> 
> Regards
> Ali
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

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

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Subject: [MMUSIC] Re: ICE Issue: mid-session connectivity checks
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Jonathan,

    I agree with you that we should remove the mid-call checks. In 
addition to points that you raised, they also make running ICE with non 
RTP media protocols much cleaner.

Aron Rosenberg
Founder and CTO
SightSpeed Inc.

>In case a client doesn't want to implement support for the mid-session 
>checks, we make them optional and allow it to be signaled.
>
>
>I'd prefer NOT to do 3, in the interest of avoiding protocol options. 
>The argument in favor of 1 is primarily one of simplification, and that 
>you will have to make these connectivity checks really fast in order to 
>be responsive enough to really recover from a failure, which could be a 
>problem.
>
>As I constantly hear these days that "ICE is too complicated", in the 
>interests of simplification, I'd be game to remove the feature. It could 
>be added back in as an extension later if needed.
>
>Comments?
>
>-Jonathan R.
>  
>

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Hi John,

lazzaro wrote:

> 
> The RTP MIDI payload's duration is always has a defined value.
> It just so happens that 0 ms is an acceptable value.
> 
> So, as long as 0 ms is an acceptable maxptime duration its OK
> by me. 0 ms needn't be explicitly permitted in the text, but it should
> not
> be explicitly forbidden.
> 

I think the usage of maxptime=0 may be a bit problematic. A 
interpretation is that no audio can be included in the packet, as no 
audio duration is allowed.

I see that you have made another interpretation in the MIDI draft 
(draft-ietf-avt-rtp-midi-format-06):

"0 ms is a reasonable media time value for MIDI packets.  In a packet
with a 0 ms media time, all commands execute at the instant coded by the
packet timestamp.  Prohibitions in [13] against 0 ms ptime values are
not relevant for MIDI streams, and may be ignored."

An RTP timestamp tick is strictly not an instant, it has a duration 
equal to 1/(RTP timestamp rate). However these is my interpretation. 
maxptime was never intended for this usage, where you would have 
interest to limit the duration to smaller intervals than a 1 ms. We 
where mostly thinking of the smallest units as something like 5 ms or 
so. Is really your intention of using maxptime=0 that for each timestamp 
tick there is a command(s), a new packet should be sent? I can 
understand using ptime=0 to recommend that you do this behaviour but 
using maxptime to enforce the behaviour?

Another question is how well parsers out there will handle 0. Hopefully 
well, but I am not certain. Especially any that is primarily to handle 
Offer/Answer, where a normative MUST be larger than 0 is present. I 
think you should consider the impact on interoperability.

So lets try to sort this out by answering some questions:

- How big is the need to be able to specify ptime=0 and maxptime=0, and 
what is the effect is the smallest one can use is ptime=1?

- Do any implementers of generic SDP parsers have any problems with 
ptime or maxptime equals 0?

- Why does RFC 3264 have a normative MUST against using ptime=0?

- Any further considerations?

If the answer to the above questions and any other input indicates that 
this is reasonable to do this, then I think a clarification is in order 
to the parameters about the interpretation of 0.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
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Torshamsgatan 23           | Fax   +46 8 7575550
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From mmusic-bounces@ietf.org  Thu Oct 21 09:21:58 2004
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Subject: Re: [MMUSIC] ICE Issue: mid-session connectivity checks
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I would prefer to keep the checks - choice 2.

Jonathan Rosenberg wrote:
> Folks,
> 
> One of the big changes in the -02 ICE spec was the addition of 
> mid-session connectivity checks. These checks served several purposes:
> 
> 1. detection of a network failure that causes an interface to become 
> unreachable
> 
> 2. in the case of a failure of a nat in the middle of the network, if 
> the nat recovers, or a backup takes its place, a new binding will get 
> assigned, and it can be discovered
> 
> 
> The particular motivating use case for 2 is an enterprise that is 
> multihomed. It has a NAT on each of its two interfaces to its providers. 
> The enterprise uses the address prefix provided from provider A, and 
> advertises reachability to this prefix into provider B. A client within 
> the enterprise makes a call. The NAT and/or connection to provider A 
> fails. Routing now causes outbound UDP packets to go over the interface 
> to provider B. So, outbound UDP packets from the client now go through 
> this other NAT. It assigns a new binding. With mid-session checks, the 
> new binding can be learned by the peer, and communications established 
> using it.
> 
> Now, the trick is that, if we are interested in VoIP as a driving use 
> case for this, the amount of time to discover this has to be relatively 
> short. If it takes 30 seconds, the users in the call would have already 
> hung up. That requires the connectivity checks to be sent at a rapid 
> clip. Furthermore, in the case of such a failure, signaling would fail 
> too in all likelihood (depending where the servers are), which would 
> require some SIP operations.
> 
> So, the issue is, is it worth it? I have gotten a few complaints about 
> the complexity introduced by the mid-session checks. Thus, we have three 
> choices as I see it:
> 
> 1. remove the checks completely
> 
> ICE runs only on setup, to do an initial check. RTCP would, for example, 
> allow you to detect some of these failure situations mid-call. Recovery 
> would require re-establishing the call.
> 
> 2. keep the checks
> 
> THe draft stays as is
> 
> 3. Allow for the usage of the checks to be negotiated
> 
> In case a client doesn't want to implement support for the mid-session 
> checks, we make them optional and allow it to be signaled.
> 
> 
> I'd prefer NOT to do 3, in the interest of avoiding protocol options. 
> The argument in favor of 1 is primarily one of simplification, and that 
> you will have to make these connectivity checks really fast in order to 
> be responsive enough to really recover from a failure, which could be a 
> problem.
> 
> As I constantly hear these days that "ICE is too complicated", in the 
> interests of simplification, I'd be game to remove the feature. It could 
> be added back in as an extension later if needed.
> 
> Comments?
> 
> -Jonathan R.


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Subject: [MMUSIC] RTSP: The normative level for implementation of OPTIONS
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In the latest RTSP core draft their is a MUST implement OPTIONS method 
for an client. I propose that we lower the bar to SHOULD implement 
OPTIONS and give a motivation as being the main capability negotiation 
mechanism.

Any objections to this proposal?

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
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From mmusic-bounces@ietf.org  Thu Oct 21 09:47:03 2004
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From: Jon Ringle <ml-mmusic@ringle.org>
To: mmusic@ietf.org
Subject: Re: [MMUSIC] ICE Issue: mid-session connectivity checks
Date: Thu, 21 Oct 2004 09:15:17 -0400
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Alternate suggestion inline.

On Wednesday 20 October 2004 08:46 am, Jonathan Rosenberg wrote:
> Folks,
>
> One of the big changes in the -02 ICE spec was the addition of
> mid-session connectivity checks. These checks served several purposes:
>
> 1. detection of a network failure that causes an interface to become
> unreachable
>
> 2. in the case of a failure of a nat in the middle of the network, if
> the nat recovers, or a backup takes its place, a new binding will get
> assigned, and it can be discovered
>
>
> The particular motivating use case for 2 is an enterprise that is
> multihomed. It has a NAT on each of its two interfaces to its providers.
> The enterprise uses the address prefix provided from provider A, and
> advertises reachability to this prefix into provider B. A client within
> the enterprise makes a call. The NAT and/or connection to provider A
> fails. Routing now causes outbound UDP packets to go over the interface
> to provider B. So, outbound UDP packets from the client now go through
> this other NAT. It assigns a new binding. With mid-session checks, the
> new binding can be learned by the peer, and communications established
> using it.
>
> Now, the trick is that, if we are interested in VoIP as a driving use
> case for this, the amount of time to discover this has to be relatively
> short. If it takes 30 seconds, the users in the call would have already
> hung up. That requires the connectivity checks to be sent at a rapid
> clip. Furthermore, in the case of such a failure, signaling would fail
> too in all likelihood (depending where the servers are), which would
> require some SIP operations.
>
> So, the issue is, is it worth it? I have gotten a few complaints about
> the complexity introduced by the mid-session checks. Thus, we have three
> choices as I see it:
>
> 1. remove the checks completely
>
> ICE runs only on setup, to do an initial check. RTCP would, for example,
> allow you to detect some of these failure situations mid-call. Recovery
> would require re-establishing the call.
>
> 2. keep the checks
>
> THe draft stays as is
>
> 3. Allow for the usage of the checks to be negotiated
>
> In case a client doesn't want to implement support for the mid-session
> checks, we make them optional and allow it to be signaled.

I offer another suggestion:

Remove the STUN mid-call connectivity checks (or keep them to a minimum only 
to refresh bindings). A timer could be placed based on the reception of RTP 
packets (since ICE is demuxing STUN/RTP it is in a position to do this). If 
the timer fires due to no RTP packets received within a specified time (codec 
dependant), and the session has not been terminated yet, then this endpoint 
knows that there is a possible problem with the peer endpoint being able to 
reach this endpoint. This endpoint could then perform STUN connectivity 
checks. When the peer receives a STUN bind request mid-call, it can then 
assume that the reason it is receiving a STUN bind request is because it's 
RTP media is not reaching it's peer. The peer endpoint can then perform it's 
own connectivity checks again.

Jon

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        mmusic@ietf.org, "Jari Arkko \(JO/LMF\)" <jari.arkko@ericsson.com>
Subject: [MMUSIC] Re: draft-ietf-mmusic-kmgmt-ext-11.txt: KMID? IANA
 considerations section
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Hi again Martin,

Yes, we think we are almost in complete agreement
now. Final(?) thoughts below:


Euchner Martin wrote:

 > Mats,
 >
 > Once again many thanks for your good answers that indeed clarify a lot
 > of the misunderstandings.
 >
 > To quickly summarize: I believe that we are already very close in our
 > views; I don't see any real disagreement.
 >
 > However, I see some further related issues on how the answerer has
 > implemented its system. Originally, I had the "insecure implementation"
 > in mind, yet it was not immediately apparent to me that this is the main
 > cause of the problem. I would at least ask for your confirmation, if
 > that is correct; see the longer added statements with the counter
 > example in the middle of the email.
 >
 > Cherio
 >
 > Martin.
 >
 >  > MEU:> Ok this confirms that kmtext "negotiates" a key management
 >  >protocol family, but does not actually negotiate any particular key
 >  >management capability. I intentionally put negotiate in quotes, since
 >
 >
 > Well, we would e.g. assume that if a shared key is known, that method
 > would in general be preferred (even if you by chance also know a public
 > key), and the need to negotiate in that "dimension" would be small.
 >
 > Still, we think also that sort of negotiation *could* be done
 > with a security that is not lower than if just single offers were
 > allowed, see below.
 >
 >
 > MEU: ok.
 >
 >  > My only concern in the context of a bidding-down attack is, that the
 >  >bidding-down attack could be accomplished within a spurious hop
 >
 > What do you mean by "spurious hop" ?
 >
 > MEU:> sorry for not having explained the attack scenario. It is exactly
 > the case that you showed in 1. and in 2, where an evil Eve who is not
 > trusted by the offerer nor the answerer runs a modification/substitution
 > attack on at least one SDP a-line.
 >
 >  >somewhere between the originating and terminating end systems, such
 >  >that the attacker replaces a stronger offer by a weaker one of the same
 >  >protocol family. For the bidding-down attack success, the strength of
 >
 > Neglecting that we don't understand "spurious hop", we still think
 > that the attack you describe is of the following form:
 >
 > 1. The Initiator makes an offer containing m > 1 MIKEY
 >     alternatives/lines.
 >
 > 2. Somehow, along the path, at least one of the m MIKEY offers
 >     get replaced by another (weaker) MIKEY offer that was not in
 >     the original offer.
 >
 > Correct?
 >
 > MEU:> yes exactly.
 >
 > If so, let's call the weak MIKEY variant "MIKEY-X", be it password
 > based or otherwise.
 >
 > Now we need to ask, how could it be that the inserted MIKEY-X offer
 > gets accepted as authentic by the Responder?
 >
 >
 > As we see it, that would only be possible if either:
 >
 > A) the MIKEY-X offer is a replay of an old offer (which would normally
 >    be taken care of by MIKEY-X replay protection), or,
 >
 >MEU: actually I had in mind just this case as I did not assume case B
 >to happen (so my assumption is also along the lines that MIKEY-X still
 >provides some meaningful security level, yet it is possibly lower in
 >security level than any of the current MIKEY protocol variants.)
 >
 > B) The MIKEY-X option is so weak in itself that it is possible
 >    to fake MIKEY-X messages without even resorting on replay.
 >
 > MEU:> You are right that this case would also exist.
 >
 > But if either of these would be possible, there would be
 > a flaw in that specific MIKEY-X variant, and it really doesn't
 > matter that there were more than one offer in the original
 > message: the attacker could fake MIKEY-X offers according to that
 > (single) MIKEY-X version at any point in time, right? The real
 > Initiator doesn't even need to be on-line. Thus, you *could* say that
 > you have a bidding down attack, but that is not *the* real problem.
 > The problem is more fundamental than that.
 >
 > MEU: I tend to agree with that given that Eve somehow must have
 >substituted actually a replayed MIKEY protocol run. I really cannot
 >think of any much different attack right now.

OK.

 >
 > Now if Eve had done so, then the answerer could detect this attack, by
 >verification that the integrity of >each< offered MIKEY protocol holds
 >valid. For the replayed option, the verification would fail (e.g.
 >re-used nonce or expired timestamp etc).
 >
 > So I agree with you that the answerer has a real chance to detect a
 >bidding-down attack. This is good and removes my original concern.

OK.

 >
 > New concern: Unfortunately, this comes at a high price; namely the
 >answerer MUST implement all MIKEY protocol variants even those that
 >MIKEY calls just optional to implement such as MIKEY-DHSIGN,  in order
 >to be able to protect against bidding-down attacks. Right?


No. The bidding down attack will only be *detected* if the Responder
supports the variant that was inserted (MIKEY-X in the above case).
If the Responder does not even support MIKEY-X, the *substitution* will
not be detected, but OTHO you cannot then even talk of "bidding down" in
this case, since it's simply not possible to fool the Responder into
using something he doesn't even support.


 >
 > I cannot fully believe this to be true right now, as this questions 
the choice of making certain MIKEY protocols optional altogether.
 >
 > Why is it so?


In summary, you are right in the sense that the modification cannot be
*detected* unless all variants are supported, but OTOH, you cannot
"bid down" to something that is not implemented. For instance, if you
don't support MIKEY-DH, how can I fool you into using it?

 >
 > So the answerer MUST first verify all received MIKEY protocol offers,
 >and just after that actually would decide which option to choose
 >(authenticate first then policy decide).
 >
 > In fact, doing it the other way round (i.e. 1st make protocol choice 
then verify that the choice is authentic) would be insecure, even if the 
choice is not the replayed protocol offer! See counter example:


This is in very true: you can in general not make secure decisions based 
on (possibly) faked information, agreed.

[snip]

We took the liberty of deleting your example of the above, we generally
agree with you.

Best,

/Mats and MIKEY team




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From: sunil bhargo <marx_bhargav@yahoo.com>
Subject: Re: [MMUSIC] RTSP: The normative level for implementation of OPTIONS
	request
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Magnus,
       Actually there is some contradiction in section
11.1 and D.1 (minimal client implementation). 11.1
says
that OPTIONS is must for client but according to D.1
this is not MUST probably this is what you are
pointing 
at.
       If so then shouldn't the server support for    
   OPTIONS be also changed from MUST to should in D.2
?
        
sunil bhargo

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

> In the latest RTSP core draft their is a MUST
> implement OPTIONS method 
> for an client. I propose that we lower the bar to
> SHOULD implement 
> OPTIONS and give a motivation as being the main
> capability negotiation 
> mechanism.
> 
> Any objections to this proposal?
> 
> Cheers
> 
> Magnus Westerlund
> 
> 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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> https://www1.ietf.org/mailman/listinfo/mmusic
> 


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Hello Mats and MIKEY gang,

Thanks once more for the helpful responses. I've added inline below my comments to the exchange before. I believe we are getting very close now.

So to summarize the discussion and where we stand:
1) There is nothing wrong with MIKEY; all known MIKEY protocol variants provide sufficient and good security to protect against manipulation attacks on the SDP key management phase.
2) I don't see kmgtext-11 protocol being flawed either.
3) I'm a bit concerned that the overall security very much depends on a correct implementation of the responder, in particular the right processing order of the steps. My concern is fostered by the feeling that while for security experts, the correct and secure processing order is obvious, I'm not so certain, if it is also that obvious for non-security experts; especially, since I could not find any advice/or caution anywhere in the kmgtext document.

So I would like to ask the WG:
Do you see a need to add some advice or caution (possibly in the security consideration section) that explains what to do on the responder side?

I would highly appreciate to see such a text as some generally useful non-obvious but important detail.

4) We clarified that the KMID is on the key management protocol level, the current KMID scheme does not reflect the MIKEY protocol variant level.
5) We are in agreement that the bidding-down attack can be mitigated at the responder as long as the responder implements all possible MIKEY protocol options, even those that are not of interest to the responder, and of course that the responder verifies each received option to detect if it is an injection/replacement/replay etc.
6) We are also in agreement that we are concerned about bidding-down attacks on the protocol level, but not on the algorithm/parameter level; the latter would assume a very insecure MIKEY protocol anyhow; we can de-scope this easily from our discussion.
7) I believe that we are further in agreement that the current mechanism operating on a key management protocol level and the procedures are secure but come at a high price (number of implemented options, performance).

I am clearly concerned about the consequences, namely that the existing mechanism is very costly. Do we really want this?? Or can we try to improve?

8) Where could we improve it? Which bidding-down attacks are worth it? This is a requirements question:

I would like to see
a) better (i.e. cheaper) yet equally secure protection mechanism in place that quickly detects a maliciously injected/substituted MIKEY protocol option that the responder has not implemented and/or doesn't want to support.
b) Possibility for the responder to quickly decide without full verification of all offers not to consider further a particular offer.

c) The responder being able to quickly sort out attacks from a bidding-down attempt.

So in essence I believe I'm requiring that the MIKEY protocol identifier becomes an authenticated one.

Imagine an example with a PKI enabled environment where MIKEY-DHSIGN or MIKEY-PK are possible but MIKEY-PS is also available and implemented and where the security policy of the responder would always prefer a PKI option unless such one is not offered. 

In this example, the responder upon reception of (DHSIGN, PK, PS) offer would just decide which of both DHSIGN or PK to take, and then verify the offered and chosen MIKEY protocol string. The responder does not need to verify the other offered MIKEY strings. So there is only one MIKEY instantiation instead of three. Wouldn't that be worth it?
In the example above where the responder receives an attacked offer of (DHSIGN, PK; PW) and PW is not supported or implemented, the PW option can instantly be sorted out and would not be processed any further. This would even improve resistance against certain DoS attacks, but that's a different issue not of my concern here.

9) I wouldn't have argued so long, without offering a solution that (hopefully) would work roughly as follows:

- each offered MIKEY protocol option is identified by a specific KMID. One for MIKEY-PS, a different KMID for MIKEY-DH etc
- we take advantage of the fact that MIKEY strongly integrity protects the list of protocol IDs end-to-end; so the KMIDs are indeed authenticated.
- The responder can detect
a) an injected additional KMID offer or
b) an offer that is replaced by new KMID (and string) which is not supported
simply as the KMID doesn't show up within the protected protocol list.

Thus, the responder does not require implementation of all options to thwart bidding-down attacks. The responder can reject offers that he/she hasn't implemented or for some policy reason, doesn't want to support.

10) Isn't it obvious that allocation of a few KMIDs basically and a few checks upon them by the responder is a much cheaper mechanism to satisfy above requirement than brute-force verification of each and every received option by the receiver?

I put this proposal forward to discussion.

Cheers

Martin.



Hi again Martin,

Yes, we think we are almost in complete agreement
now. Final(?) thoughts below:

MEU:> yes I think so too, nearly. So let's clear up the remaining issue.


 >
 > New concern: Unfortunately, this comes at a high price; namely the
 >answerer MUST implement all MIKEY protocol variants even those that
 >MIKEY calls just optional to implement such as MIKEY-DHSIGN,  in order
 >to be able to protect against bidding-down attacks. Right?


No. The bidding down attack will only be *detected* if the Responder
supports the variant that was inserted (MIKEY-X in the above case).
MEU> yes under the condition that the responder has implemented the inserted option and first checks validity of the offer, before deciding further. On the latter we have clearly agreement that not doing so would be highly insecure.

If the Responder does not even support MIKEY-X, the *substitution* will
not be detected, but OTHO you cannot then even talk of "bidding down" in
this case, since it's simply not possible to fool the Responder into
using something he doesn't even support.


In summary, you are right in the sense that the modification cannot be
*detected* unless all variants are supported, but OTOH, you cannot
"bid down" to something that is not implemented. For instance, if you
don't support MIKEY-DH, how can I fool you into using it?

MEU:>ok, as above. However, bad implementation of the processing order in the responder could indeed lead to a successful bidding-down attack, where the attacker forces the responder to choose an implemented option of lower security level (than without attack), where the responder actually hasn't implemented even the injected option.

 >
 > So the answerer MUST first verify all received MIKEY protocol offers,
 >and just after that actually would decide which option to choose
 >(authenticate first then policy decide).
 >
 > In fact, doing it the other way round (i.e. 1st make protocol choice 
then verify that the choice is authentic) would be insecure, even if the 
choice is not the replayed protocol offer! See counter example:


This is in very true: you can in general not make secure decisions based 
on (possibly) faked information, agreed.

MEU: ok we are in agreement.




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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 Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: RTSP Announce Method
	Author(s)	: T. Zeng, G. Sherwood
	Filename	: draft-ietf-mmusic-rtsp-announce-00.txt
	Pages		: 0
	Date		: 2004-10-20
	
This memo describes an extension RTSP request, ANNOUNCE, which 
   extends the core RTSP protocol [RTSP_NEW] to allow one end to 
   push to the other end various types of information by RTSP
   means.

A URL for this Internet-Draft is:
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Date: Thu, 21 Oct 2004 15:15:38 -0700
From: lazzaro <lazzaro@eecs.berkeley.edu>
Subject: Re: [MMUSIC] maxptime
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On Oct 21, 2004, at 12:45 AM, Magnus Westerlund wrote:
> Hi John,

Hi Magnus!

	There are a few separable issues here, let's take
them one by one ... CC'd to AVT since these issues cross
boundaries ...

> I think the usage of maxptime=0 may be a bit problematic. A 
> interpretation is that no audio can be included in the packet, as no 
> audio duration is allowed.
>
> I see that you have made another interpretation in the MIDI draft 
> (draft-ietf-avt-rtp-midi-format-06):
>
> "0 ms is a reasonable media time value for MIDI packets.  In a packet
> with a 0 ms media time, all commands execute at the instant coded by 
> the
> packet timestamp.  Prohibitions in [13] against 0 ms ptime values are
> not relevant for MIDI streams, and may be ignored."
>
> An RTP timestamp tick is strictly not an instant, it has a duration 
> equal to 1/(RTP timestamp rate).
> However these is my interpretation.

Here's the underlying motivation for the choice I made.

Imagine playing the piano.

Play three notes simultaneously. In MIDI, this corresponds
to three separate NoteOn commands.  In RTP MIDI, each
has a timestamp, expressed in RTP timestamp units.

It's really important to the sonic integrity of the piece to code that
the notes happen simultaneously -- even one RTP tick matters.
The reason is, phase effects as the waveforms beat against each
other change the timbre.  Hand drummers on the list (conga, etc)
can attest to this -- the art of creating certain timbres on those
types of percussion instruments is based on fine timing of strokes
on the drum with the two hands.  Standard MIDI Files do in fact code
true simultaneity, which is why they can be used in studio work.
Musicians go crazy shifting around MIDI notes by tiny amounts inside
the DAW to get this sort of stuff right, and SMFs preserve that hard 
work
on export.

Pull back from the nanosecond level to the few-millisecond
level, and we move from timbre distortion to timing sloppiness.
MIDI as transmitted on a MIDI 1.0 DIN cable suffers from this:
at 300 microseconds/command, playing full chords with both
hands creates timing slews you can hear.  Part of the motivation
to move off of MIDI DIN cable technology is to make instrument
controllers that don't suffer this timing problem.

You might ask -- why can't just put all simultaneous notes
in the same packet, and use something other than RTP timestamps
to denote intra-packet timing?

To understand the answer, you need to understand that a core
use of RTP MIDI is on "media LANs" -- a stage or studio situation
that may consist of two nodes -- piano controller and laptop --
connected by an Ethernet cable as the Layer 2 for RTP MIDI.
These folks want absolute lowest latency -- so, they don't want
their piano controller to wait around after the first MIDI command
to see if a second command that is "simultaneous" is happening soon.
Instead, they want the piano to take each command, put it in its own RTP
packet, and send it off.  So, they want the ability to send two RTP 
packets
in a row, each coding one MIDI command, each with the same
RTP timestamp.  To do this, they need the RTP timestamp semantics
that say an RTP MIDI packet can have 0 ms media time.

Some instrument makers use Ethernet today.  They use their own UDP
solutions (some use OSC on top of UDP) and have their own code.  They
want to add support for RTP MIDI, but RTP MIDI needs to work with their
model, which is as I describe it above.

I would also not be surprised if the recovery journal semantics
failed if we moved to your RTP timestamp interpretation, although
I'd have to think about to say for certain.  Other things might break
too.  This is a 4-year-old project now, and the timestamp model is
a fundamental assumption that underlies RTP MIDI in subtle ways.

So, that's the reply to issue #1.

Issue #2:

> Is really your intention of using maxptime=0 that for each timestamp 
> tick there is a command(s), a new packet should be sent? I can 
> understand using ptime=0 to recommend that you do this behaviour but 
> using maxptime to enforce the behaviour?

Yes. Once again, we're talking about apps in the "media network"
world, a laptop and piano keyboard with a point-to-point 2-node
Ethernet as the Layer 2 for RTP MIDI.  The maxptime=0 parameter
says: all packets must either code no MIDI commands, or code one
or more MIDI commands with the same timestamp -- and thus, the
media time for all packets must be 0 ms.

This accommodates receivers (such as, currently, sfront!) that do not
implement a receiver buffer of any sort.  Instead, sfront grabs a 
packet,
and executes all MIDI commands in the packet simultaneously.
It relies on the constant nominal latency of the point-to-point Layer 2,
and the constant nominal latency of the keyboard sender, for timing
accuracy -- not RTP timestamps, which are ignored.

Thus, these receivers can't handle a packet that has a non-zero media
time -- all they know is "now".

If one is designing receivers exclusively for use on media LANs
as live performance instruments (in other words -- a musical
instrument synthesizers intended to only be played locally), this
"no buffer" model makes perfect sense -- it's how soft synths that
are driven by MIDI 1.0 cables work today.  Eventually, it would be
better if these synths incorporated small configurable buffers to
compensate for sub-ms jitter, but the conceptual hurdle to do so
would be too great for many synth authors to do.  In practice,
OS APIs will probably implement those buffers if/when they choose
to support RTP MIDI as a media LAN MIDI source.

So, we need a way for those soft synths to negotiation (via 
Offer/Answer)
to say "I can only handle 0 ms packets, do not send me any other
kind".

As an aside, RTP MIDI works fairly well Berkeley/Stanford/Berkeley using
the no-buffer sfront  -- it doesn't get down to the "hand percussion 
jitter"
I mention above, but with a few coding heuristics it produces a playable
instrument when I do loop-back (keyboard in Berkeley, RTP MIDI to
Stanford and back, which sfront then turns into audio).  Chris Chafe
sees jitter under the audio sample rate on his DAT-quality long-distance
streaming sessions on occassion!

Issue #3

> Especially any that is primarily to handle Offer/Answer, where a 
> normative
> MUST be larger than 0 is present. I think you should consider the 
> impact on interoperability.

Well, the easiest thing to do is invent my own SDP parameter as an
RTP MIDI replacement for maxptime -- this skirts the compatibility
issue.  But as I explained above, I think we have a legitimate need
for the negotiating functionality.  And I think the most elegant thing
to do is for maxptime to bend its definition to include 0 ms, just for
RTP MIDI, which is why I did it that way ... a compromise would be
to put interoperability warning language into the RTP MIDI I-D, but
keep the normative text as is.

> So lets try to sort this out by answering some questions:
>
> - How big is the need to be able to specify ptime=0 and maxptime=0, 
> and what is the effect is the smallest one can use is ptime=1?

Hopefully this is answered above.

> - Do any implementers of generic SDP parsers have any problems with 
> ptime or maxptime equals 0?
>
> - Why does RFC 3264 have a normative MUST against using ptime=0?
>
> - Any further considerations?

These questions seem aimed at others on the list ... however, I
should note that I didn't take the route I did on maxptime
lightly -- after several years of consensus building with the music
technology community, the approach to RTP timing I outline above
is the best way I can see to solve their problems.

---
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
lazzaro [at] cs [dot] berkeley [dot] edu
---


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From: "Sami Peltotalo" <sami.peltotalo@tut.fi>
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Subject: [MMUSIC] draft-walsh-mmusic-img-envelope-01
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Hi All,

An update to Internet-Draft "The IMG Envelope" is available.

http://www.ietf.org/internet-drafts/draft-walsh-mmusic-img-envelope-01.txt

All comments are again very welcome.

Regards,
Sami Peltotalo.


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Subject: Re: [MMUSIC] RTSP: The normative level for implementation of OPTIONS
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Hi,

sunil bhargo wrote:
> Magnus,
>        Actually there is some contradiction in section
> 11.1 and D.1 (minimal client implementation). 11.1
> says
> that OPTIONS is must for client but according to D.1
> this is not MUST probably this is what you are
> pointing
> at.

Yes, it is the MUST in 11.1 that I like to change to a SHOULD or 
RECOMMENDED. To be consistent that would also mean to make the necessary 
changes in Appendix D.

We have three options on this:

A. Make the Implementation of OPTIONS as a requester a MUST. There is no 
technical reason for this.

B. Make an recommendation that clients implement OPTIONS. This due to 
that it is the main method of learning the methods and feature supported 
for an resource. However really stripped down clients could skip to 
implement the method.

C. Do not say anything about implementation of OPTIONS at all.

I personally do favour B.

>        If so then shouldn't the server support for   
>    OPTIONS be also changed from MUST to should in D.2
> ?
>        

No, then you loose the possibility to use OPTIONS at all. You need to 
know that OPTIONS is supported at the server side. Otherwise you can't 
know for certain that it is possible to learn which methods or features 
that are supported either globally or for a specific resource.

Cheers

Magnus Westerlund

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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Subject: RE: [MMUSIC] draft-walsh-mmusic-img-envelope-01
Date: Fri, 22 Oct 2004 12:46:58 +0300
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The (sorry: long overdue) updated version of the IMG envelope deals with =
embedding of metadata fragments in a simple way which is also backwards =
compatible with the existing referencing approach. It also goes a long =
way in explaining the usage of IMG envelopes and provides the textual =
(XML) instance of the envelope.

The greatest omission is a description of which IMG requirements =
(http://www.ietf.org/internet-drafts/draft-ietf-mmusic-img-req-07.txt) =
are addressed by the IMG envelope and the extent to which they are =
fulfilled.

Anyone interested may wish to have a look at the 3GPP equivalent =
(ftp://ftp.3gpp.org/TSG_SA/WG4_CODEC/Ad-hoc_PSM/Docs/S4-AHP176.zip) =
which enables referencing (3GPP has agreed to enable embedding though =
the updated schema is not yet in the aggregated specification). Previous =
efforts on harmonisation mean that the referencing envelope is currently =
identical. Harmonisation on the embedded envelope is realistic so if you =
are interested in both IETF and 3GPP please check the schema if you have =
not previously got up to speed on this!

The authors do not plan to present this I-D in Washington (unfortunately =
we 3 are unable to attend). Thus the mailing list (and private emails =
for the shy) are the recommended mode review.

Cheers, Rod.

> -----Original Message-----
> From: mmusic-bounces@ietf.org=20
> [mailto:mmusic-bounces@ietf.org]On Behalf
> Of ext Sami Peltotalo
> Sent: Friday, October 22, 2004 10:21 AM
> To: Mmusic
> Subject: [MMUSIC] draft-walsh-mmusic-img-envelope-01
>=20
>=20
> Hi All,
>=20
> An update to Internet-Draft "The IMG Envelope" is available.
>=20
> =
http://www.ietf.org/internet-drafts/draft-walsh-mmusic-img-envelope-01.tx=
t
>=20
> All comments are again very welcome.
>=20
> Regards,
> Sami Peltotalo.
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic

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From: sunil bhargo <marx_bhargav@yahoo.com>
Subject: Re: [MMUSIC] RTSP: The normative level for implementation of OPTIONS
	request
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Hi Magnus,
            B makes sense, would it be better to also
recommend that the client uses OPTIONS at the start of
the session before any other request so that method
capability are known at the start of the session.

sunil bhargo

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

> Hi,
> 
> sunil bhargo wrote:
> > Magnus,
> >        Actually there is some contradiction in
> section
> > 11.1 and D.1 (minimal client implementation). 11.1
> > says
> > that OPTIONS is must for client but according to
> D.1
> > this is not MUST probably this is what you are
> > pointing
> > at.
> 
> Yes, it is the MUST in 11.1 that I like to change to
> a SHOULD or 
> RECOMMENDED. To be consistent that would also mean
> to make the necessary 
> changes in Appendix D.
> 
> We have three options on this:
> 
> A. Make the Implementation of OPTIONS as a requester
> a MUST. There is no 
> technical reason for this.
> 
> B. Make an recommendation that clients implement
> OPTIONS. This due to 
> that it is the main method of learning the methods
> and feature supported 
> for an resource. However really stripped down
> clients could skip to 
> implement the method.
> 
> C. Do not say anything about implementation of
> OPTIONS at all.
> 
> I personally do favour B.
> 
> >        If so then shouldn't the server support for
>   
> >    OPTIONS be also changed from MUST to should in
> D.2
> > ?
> >        
> 
> No, then you loose the possibility to use OPTIONS at
> all. You need to 
> know that OPTIONS is supported at the server side.
> Otherwise you can't 
> know for certain that it is possible to learn which
> methods or features 
> that are supported either globally or for a specific
> resource.
> 
> Cheers
> 
> Magnus Westerlund
> 
> 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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Hi John,

Please see below. I also have to point out that I haven't read your 
drafts during the last year or so. I have only looked at certain 
specific things in them.

lazzaro wrote:
> 
> On Oct 21, 2004, at 12:45 AM, Magnus Westerlund wrote:
>  > Hi John,
> 
> Hi Magnus!
> 
>         There are a few separable issues here, let's take
> them one by one ... CC'd to AVT since these issues cross
> boundaries ...
> 
>  > I think the usage of maxptime=0 may be a bit problematic. A
>  > interpretation is that no audio can be included in the packet, as no
>  > audio duration is allowed.
>  >
>  > I see that you have made another interpretation in the MIDI draft
>  > (draft-ietf-avt-rtp-midi-format-06):
>  >
>  > "0 ms is a reasonable media time value for MIDI packets.  In a packet
>  > with a 0 ms media time, all commands execute at the instant coded by
>  > the
>  > packet timestamp.  Prohibitions in [13] against 0 ms ptime values are
>  > not relevant for MIDI streams, and may be ignored."
>  >
>  > An RTP timestamp tick is strictly not an instant, it has a duration
>  > equal to 1/(RTP timestamp rate).
>  > However these is my interpretation.
> 

I probably should clarify myself as I have received comments that it 
seems like I am reinterpreting the timestamp model. The only point I was 
trying to make is that in a digital domain, compared to the analog an 
instant is not strictly a precise moment without any duration. Due to 
the analog to digital conversion the smallest point in time we can 
express is a single sample (RTP TS tick). That sample represent some 
duration, equal to 1/(sampling frequency), thus it is not truly 
instantaneous. To connect to John's example below, about the 
keyboardist. Even if he strikes the keys simultaneously on a timescale 
of 1/10000 part of second it is most likely not true when you go to a 
timescale of 1/10^9 of a second.

I will return to this later down.

> Here's the underlying motivation for the choice I made.
> 
> Imagine playing the piano.
> 
> Play three notes simultaneously. In MIDI, this corresponds
> to three separate NoteOn commands.  In RTP MIDI, each
> has a timestamp, expressed in RTP timestamp units.
> 
> It's really important to the sonic integrity of the piece to code that
> the notes happen simultaneously -- even one RTP tick matters.
> The reason is, phase effects as the waveforms beat against each
> other change the timbre.  Hand drummers on the list (conga, etc)
> can attest to this -- the art of creating certain timbres on those
> types of percussion instruments is based on fine timing of strokes
> on the drum with the two hands.  Standard MIDI Files do in fact code
> true simultaneity, which is why they can be used in studio work.
> Musicians go crazy shifting around MIDI notes by tiny amounts inside
> the DAW to get this sort of stuff right, and SMFs preserve that hard
> work
> on export.
> 
> Pull back from the nanosecond level to the few-millisecond
> level, and we move from timbre distortion to timing sloppiness.
> MIDI as transmitted on a MIDI 1.0 DIN cable suffers from this:
> at 300 microseconds/command, playing full chords with both
> hands creates timing slews you can hear.  Part of the motivation
> to move off of MIDI DIN cable technology is to make instrument
> controllers that don't suffer this timing problem.

I think I understand this, however the RTP timestamp rate as I 
understand it for MIDI would be equal to the sample rate for the audio 
produced? That would mean that it is reasonable to expect that we are 
working with a sample frequency of for example 48kHz. Or do you needed 
even higher resolution?

> 
> You might ask -- why can't just put all simultaneous notes
> in the same packet, and use something other than RTP timestamps
> to denote intra-packet timing?

No, I am not questioning this timing model at all. It seems appropriate 
for RTP based on your description. However one needs to be aware what it 
means. A MIDI command can't be correctly synchronized with a finer scale 
then one RTP TS tick. Thus the precision is depending on the timestamp 
rate.

> 
> To understand the answer, you need to understand that a core
> use of RTP MIDI is on "media LANs" -- a stage or studio situation
> that may consist of two nodes -- piano controller and laptop --
> connected by an Ethernet cable as the Layer 2 for RTP MIDI.
> These folks want absolute lowest latency -- so, they don't want
> their piano controller to wait around after the first MIDI command
> to see if a second command that is "simultaneous" is happening soon.
> Instead, they want the piano to take each command, put it in its own RTP
> packet, and send it off.  So, they want the ability to send two RTP
> packets
> in a row, each coding one MIDI command, each with the same
> RTP timestamp.  To do this, they need the RTP timestamp semantics
> that say an RTP MIDI packet can have 0 ms media time.

First, I can understand that due to the desire to keep the latency to a 
minimal even a short buffering is not desirable, however RTP payloads 
are normal gathered using some type of buffering, either at the encoding 
step, due to the need to gather sufficient amount of samples to perform 
the encoding, or at the packetization step to aggregate together 
sufficient amount of data to get the right efficiency in the 
transmission. Thus MIDI commands are a bit special.

Secondly I don't think we are talking media time really here. The 
command does not truly have a duration as you say. They takes affect the 
instant (on a specific sample) indicated by the RTP timestamp. The 
problem is really about indicating packetization strategies.

> 
> Some instrument makers use Ethernet today.  They use their own UDP
> solutions (some use OSC on top of UDP) and have their own code.  They
> want to add support for RTP MIDI, but RTP MIDI needs to work with their
> model, which is as I describe it above.
> 
> I would also not be surprised if the recovery journal semantics
> failed if we moved to your RTP timestamp interpretation, although
> I'd have to think about to say for certain.  Other things might break
> too.  This is a 4-year-old project now, and the timestamp model is
> a fundamental assumption that underlies RTP MIDI in subtle ways.
> 

I am not redefining anything. I think we simply have misunderstood each 
other.

> So, that's the reply to issue #1.
> 
> Issue #2:
> 
>  > Is really your intention of using maxptime=0 that for each timestamp
>  > tick there is a command(s), a new packet should be sent? I can
>  > understand using ptime=0 to recommend that you do this behaviour but
>  > using maxptime to enforce the behaviour?
> 
> Yes. Once again, we're talking about apps in the "media network"
> world, a laptop and piano keyboard with a point-to-point 2-node
> Ethernet as the Layer 2 for RTP MIDI.  The maxptime=0 parameter
> says: all packets must either code no MIDI commands, or code one
> or more MIDI commands with the same timestamp -- and thus, the
> media time for all packets must be 0 ms.
> 
> This accommodates receivers (such as, currently, sfront!) that do not
> implement a receiver buffer of any sort.  Instead, sfront grabs a
> packet,
> and executes all MIDI commands in the packet simultaneously.
> It relies on the constant nominal latency of the point-to-point Layer 2,
> and the constant nominal latency of the keyboard sender, for timing
> accuracy -- not RTP timestamps, which are ignored.

Then it is not really RTP you are using. You are ignoring the 
functionality the timestamp provides to place the different packets 
correctly on a timescale to combat any transport jitter. As it seems the 
local stage applications are relying on a consistent transport behavior.

> 
> Thus, these receivers can't handle a packet that has a non-zero media
> time -- all they know is "now".
> 

Then they are not RTP aware, a RTP receiver needs to have knowledge 
about how time flows.

> If one is designing receivers exclusively for use on media LANs
> as live performance instruments (in other words -- a musical
> instrument synthesizers intended to only be played locally), this
> "no buffer" model makes perfect sense -- it's how soft synths that
> are driven by MIDI 1.0 cables work today.  Eventually, it would be
> better if these synths incorporated small configurable buffers to
> compensate for sub-ms jitter, but the conceptual hurdle to do so
> would be too great for many synth authors to do.  In practice,
> OS APIs will probably implement those buffers if/when they choose
> to support RTP MIDI as a media LAN MIDI source.
> 
> So, we need a way for those soft synths to negotiation (via
> Offer/Answer)
> to say "I can only handle 0 ms packets, do not send me any other
> kind".

The problem is really to indicate the desired behavior. Which is not 
really what ptime and maxptime is about. They give the recommend and 
maximal amount of media duration for a packet.

> 
> Issue #3
> 
>  > Especially any that is primarily to handle Offer/Answer, where a
>  > normative
>  > MUST be larger than 0 is present. I think you should consider the
>  > impact on interoperability.
> 
> Well, the easiest thing to do is invent my own SDP parameter as an
> RTP MIDI replacement for maxptime -- this skirts the compatibility
> issue.  But as I explained above, I think we have a legitimate need
> for the negotiating functionality.  And I think the most elegant thing
> to do is for maxptime to bend its definition to include 0 ms, just for
> RTP MIDI, which is why I did it that way ... a compromise would be
> to put interoperability warning language into the RTP MIDI I-D, but
> keep the normative text as is.

The basic problem is that you are using ptime and maxptime way beyond 
what was ever intended by the authors of these parameters. I am certain 
of this when it comes to maxptime, as I am the one that defined that one 
with some help.

Therefore I think one needs to look at the different use cases for RTP 
midi. I think there are several.

1. Local play, where most of RTP's timing model seems to be ignored, as 
it can't be excepted to buffer in either end of the chain.

2. Transmitting a live session to another point, but in which cases it 
is acceptable to introduce some small delay. This is done over a network 
that do not have the jitter behavior necessary to ignore the timestamp. 
Which means a receiver must follow the timestamp to give the right inter 
command time spacing.

3. Streaming of midi content.

In 1, I think you need something new to indicate the intended behavior 
of minimal transmission delay and local live play. However I do hope 
that a sender at least correctly timestamp the RTP packets sent. Thus 
the RTP receiver can in the future start using the timing model correctly.

In 2, I think the usage of ptime and maxptime could be reasonable to 
instruct a sender to gather 10, 50 or 100 samples or 1, 5, 20 ms of 
commands into each packet. However if you like to have finer resolution 
than 1 ms then it is necessary to have a new parameter.

In 3, theses parameters are not used at all. The amount of commands in 
the packets are pre determined for best performance at the receiver.

In conclusion I would suggest that you use another MIME parameter than 
ptime and maxptime to indicate the desired behavior.

I also think that the format drafts needs to be more explicit on 
considerations for the RTP timestamp. What rates are suitable, what are 
the impact on playback timing. There seems to be a bit on this in the 
guidelines, but I haven't had time to read it fully.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: sunil bhargo <marx_bhargav@yahoo.com>
Subject: Re: [MMUSIC] RTSP: The normative level for implementation of OPTIONS
	request
References: <20041022123744.44902.qmail@web53708.mail.yahoo.com>
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Hi,

sunil bhargo wrote:

> Hi Magnus,
>             B makes sense, would it be better to also
> recommend that the client uses OPTIONS at the start of
> the session before any other request so that method
> capability are known at the start of the session.

I would say that a recommendation depends on what type of session you 
are going to do. If you are going to do something plan vanilla, then you 
can simple ensure these things are true with Require header. Also in 
certain cases, where it is desirable to try to achieve a quick setup, 
without the extra round trip time, then one can use Require. If one is 
going to try something very special, not likely to be supported, then 
the usage of OPTIONS are welcome.

This is really a implementation choice for a client.

Cheers

Magnus Westerlund

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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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>,
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Subject: Re: [MMUSIC] ICE Issue: mid-session connectivity checks
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The reason why I think we should keep the checks, is that it can be useful 
to choose between different candidates with the same priority. An endpoint 
can allocate a port on more than one TURN server - on different countries 
or state. The checks can be used to measure the latency and the packet loss 
and so dynamically switch to a better route in the middle of a call.

Marc Petit-Huguenin wrote:
> I would prefer to keep the checks - choice 2.
> 
> Jonathan Rosenberg wrote:
> 
>> Folks,
>>
>> One of the big changes in the -02 ICE spec was the addition of 
>> mid-session connectivity checks. These checks served several purposes:
>>
>> 1. detection of a network failure that causes an interface to become 
>> unreachable
>>
>> 2. in the case of a failure of a nat in the middle of the network, if 
>> the nat recovers, or a backup takes its place, a new binding will get 
>> assigned, and it can be discovered
>>
>>
>> The particular motivating use case for 2 is an enterprise that is 
>> multihomed. It has a NAT on each of its two interfaces to its 
>> providers. The enterprise uses the address prefix provided from 
>> provider A, and advertises reachability to this prefix into provider 
>> B. A client within the enterprise makes a call. The NAT and/or 
>> connection to provider A fails. Routing now causes outbound UDP 
>> packets to go over the interface to provider B. So, outbound UDP 
>> packets from the client now go through this other NAT. It assigns a 
>> new binding. With mid-session checks, the new binding can be learned 
>> by the peer, and communications established using it.
>>
>> Now, the trick is that, if we are interested in VoIP as a driving use 
>> case for this, the amount of time to discover this has to be 
>> relatively short. If it takes 30 seconds, the users in the call would 
>> have already hung up. That requires the connectivity checks to be sent 
>> at a rapid clip. Furthermore, in the case of such a failure, signaling 
>> would fail too in all likelihood (depending where the servers are), 
>> which would require some SIP operations.
>>
>> So, the issue is, is it worth it? I have gotten a few complaints about 
>> the complexity introduced by the mid-session checks. Thus, we have 
>> three choices as I see it:
>>
>> 1. remove the checks completely
>>
>> ICE runs only on setup, to do an initial check. RTCP would, for 
>> example, allow you to detect some of these failure situations 
>> mid-call. Recovery would require re-establishing the call.
>>
>> 2. keep the checks
>>
>> THe draft stays as is
>>
>> 3. Allow for the usage of the checks to be negotiated
>>
>> In case a client doesn't want to implement support for the mid-session 
>> checks, we make them optional and allow it to be signaled.
>>
>>
>> I'd prefer NOT to do 3, in the interest of avoiding protocol options. 
>> The argument in favor of 1 is primarily one of simplification, and 
>> that you will have to make these connectivity checks really fast in 
>> order to be responsive enough to really recover from a failure, which 
>> could be a problem.
>>
>> As I constantly hear these days that "ICE is too complicated", in the 
>> interests of simplification, I'd be game to remove the feature. It 
>> could be added back in as an extension later if needed.
>>
>> Comments?
>>
>> -Jonathan R.
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 
> 


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Subject: Re: [MMUSIC] RTSP: The normative level for implementation of OPTIONS
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Magnus Westerlund (KI/EAB) wrote:

> Hi,
> 
> sunil bhargo wrote:
> 
>  > Hi Magnus,
>  >             B makes sense, would it be better to also
>  > recommend that the client uses OPTIONS at the start of
>  > the session before any other request so that method
>  > capability are known at the start of the session.
> 
> I would say that a recommendation depends on what type of session you
> are going to do. If you are going to do something plan vanilla, then you
> can simple ensure these things are true with Require header. Also in
> certain cases, where it is desirable to try to achieve a quick setup,
> without the extra round trip time, then one can use Require. If one is
> going to try something very special, not likely to be supported, then
> the usage of OPTIONS are welcome.
> 
> This is really a implementation choice for a client.
> 

Clarification: If you need to learn what methods are supported for a 
resource, then you need to use OPTIONS.

Due to the many different needs around capability discovery, I am 
hesitant to give any recommendation, more than explaining the usability 
of each method.

Cheers

Magnus Westerlund

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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On Oct 22, 2004, at 5:55 AM, Magnus Westerlund wrote:
> I think I understand this, however the RTP timestamp rate
> as I understand it for MIDI would be equal to the sample rate for the 
> audio produced?

Yes -- for best results, the MIDI timestamp resolution should
equal the audio sampling rate (44.1, 48, 96, or 192 KHz).

> However I do hope that a sender at least correctly timestamp the RTP 
> packets sent.
> Thus the RTP receiver can in the future start using the timing model 
> correctly.

Sender time stamping is a MUST.

> In conclusion I would suggest that you use another MIME parameter than
> ptime and maxptime to indicate the desired behavior.

OK, I'll redo this part of the I-D, and create new MIME parameters
to partially convey what the I-D previously coded using ptime and 
maxptime.

Unless a miracle happens, I won't make the Monday deadline with
a new I-D pair ... so it will have to wait post-D.C.  But I really hope 
we can
make it to Last Call for the two I-Ds by the holidays ...

> I also think that the format drafts needs to be more explicit on 
> considerations for the RTP timestamp. What rates are suitable, what 
> are the impact on playback timing. There seems to be a bit on this in 
> the guidelines, but I haven't had time to read it fully.

See C.6.3 in the normative I-D.  Part of Appendix C.6, the
"SDP Interoperability Section", it might be easier to read all of C.6.

Also, in the guide I-D, Section 4 discusses sender timing,
Section 6 discusses receiver timing.

> No, I am not questioning this timing model at all. It seems appropriate
> for RTP based on your description.

Great -- thinking it over last night, I'm pretty sure that if I was 
forced
to define the media time for a packet with a single MIDI command
whose command timestamp equals the RTP timestamp to be 1 RTP
timestamp tick long (and not 0 ticks as it is now), the recovery journal
would break.

---
John Lazzaro
http://www.cs.berkeley.edu/~lazzaro
lazzaro [at] cs [dot] berkeley [dot] edu
---


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I agree with the direction Jon is suggesting here, but think it is part 
of a bigger activity.

Without some kind of recovery mechanism, what usually happens is that 
one or both parties on a call notice that something is wrong with the 
call - either the media malfunction in some way, or there is some 
indication that signaling isn't working properly. At that time, the 
users take remedial action - usually attempting to reestablish the call.

I think there should be an analogous process that the endpoints perform 
on their own, so that the users aren't required to do anything other 
than tolerate a potential glitch. If this were well specified it 
wouldn't suffer from the problem that often happen when users do it - 
such as glare resulting in both getting a busy signal.

To achieve this, something needs to monitor the session and identify 
when there may be a potential problem. A lot of this is monitoring media 
sessions. How this is done is inherently media specific.

For two way RTP based audio media this may, as Jon suggests, only 
require checking that packets are received periodically, and helping the 
other end by always sending packets periodically. For one way or idle 
media this may be a bit more tricky. Sending STUN packets is perhaps one 
answer, but maybe it is ok to send rtp idle packets (did those get 
approved?) if it is agreed they can be sent even on a RECVONLY or 
INACTIVE connection.

A similar approach would presumably work for video. Session mode IM will 
require something different, but the concept is the same.

Once there is a hint of a problem, testing and recovery can commence. 
Part of this is to test the signaling path. This can be any mid-dialog 
request, but could easily be a reinvite or update that does a new 
offer/answer. Exactly what kind of offer/answer to do isn't entirely 
clear. If I think there is a problem, but don't think I am at fault, the 
"obvious" thing for me to do is to again offer the same SDP I used last. 
However, that may well solicit an answer the same as last time, and 
result in absolutely no change in the media.

There really needs to be some kind of hint in the reinvite or the 
accompanying offer that there is a problem to be diagnosed or recovered 
from. Maybe this would be a Reason header in the reinvite, or something 
(new preconditions?) in the offer. This should either result in 
initiating diagnostic actions on one or more media streams, or 
replacement of resources (ports, relays, etc.) in the path of one or 
more media streams, or both.

There are of course lots of failure modes to consider. In some cases it 
may be the entire peer UA that has failed, but that has a backup, or it 
could be a turn server, msrp relay, or NAT that has failed. In any case, 
the problem extends beyond the scope of what ICE deals with.

If this kind of recovery process were supported by every endpoint, then 
it would be much easier to design and build redundant servers.

	Paul

Jon Ringle wrote:
> Alternate suggestion inline.
> 
> On Wednesday 20 October 2004 08:46 am, Jonathan Rosenberg wrote:
> 
>>Folks,
>>
>>One of the big changes in the -02 ICE spec was the addition of
>>mid-session connectivity checks. These checks served several purposes:
>>
>>1. detection of a network failure that causes an interface to become
>>unreachable
>>
>>2. in the case of a failure of a nat in the middle of the network, if
>>the nat recovers, or a backup takes its place, a new binding will get
>>assigned, and it can be discovered
>>
>>
>>The particular motivating use case for 2 is an enterprise that is
>>multihomed. It has a NAT on each of its two interfaces to its providers.
>>The enterprise uses the address prefix provided from provider A, and
>>advertises reachability to this prefix into provider B. A client within
>>the enterprise makes a call. The NAT and/or connection to provider A
>>fails. Routing now causes outbound UDP packets to go over the interface
>>to provider B. So, outbound UDP packets from the client now go through
>>this other NAT. It assigns a new binding. With mid-session checks, the
>>new binding can be learned by the peer, and communications established
>>using it.
>>
>>Now, the trick is that, if we are interested in VoIP as a driving use
>>case for this, the amount of time to discover this has to be relatively
>>short. If it takes 30 seconds, the users in the call would have already
>>hung up. That requires the connectivity checks to be sent at a rapid
>>clip. Furthermore, in the case of such a failure, signaling would fail
>>too in all likelihood (depending where the servers are), which would
>>require some SIP operations.
>>
>>So, the issue is, is it worth it? I have gotten a few complaints about
>>the complexity introduced by the mid-session checks. Thus, we have three
>>choices as I see it:
>>
>>1. remove the checks completely
>>
>>ICE runs only on setup, to do an initial check. RTCP would, for example,
>>allow you to detect some of these failure situations mid-call. Recovery
>>would require re-establishing the call.
>>
>>2. keep the checks
>>
>>THe draft stays as is
>>
>>3. Allow for the usage of the checks to be negotiated
>>
>>In case a client doesn't want to implement support for the mid-session
>>checks, we make them optional and allow it to be signaled.
> 
> 
> I offer another suggestion:
> 
> Remove the STUN mid-call connectivity checks (or keep them to a minimum only 
> to refresh bindings). A timer could be placed based on the reception of RTP 
> packets (since ICE is demuxing STUN/RTP it is in a position to do this). If 
> the timer fires due to no RTP packets received within a specified time (codec 
> dependant), and the session has not been terminated yet, then this endpoint 
> knows that there is a possible problem with the peer endpoint being able to 
> reach this endpoint. This endpoint could then perform STUN connectivity 
> checks. When the peer receives a STUN bind request mid-call, it can then 
> assume that the reason it is receiving a STUN bind request is because it's 
> RTP media is not reaching it's peer. The peer endpoint can then perform it's 
> own connectivity checks again.
> 
> Jon
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 


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Dan,

Old message. New data.

The new anat draft makes more explicit reference to how it works with 
ICE, and the new ICE draft makes more explicit reference to how it works 
with ANAT.

-Jonathan R.

Dan Wing wrote:

> I'd like ICE's "alt" parameter and draft-ietf-mmusic-anat-01.txt to
> merge and be one.  Or else I'd like to see how -anat- and ICE would
> work together -- such an SDP would be difficult to parse at best
> (which I believe relates to your issue #1, "v6trans Applicability").
> 
> -d
> 
> 
>>-----Original Message-----
>>From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]On Behalf
>>Of Jonathan Rosenberg
>>Sent: Monday, July 26, 2004 12:12 PM
>>To: IETF MMUSIC WG
>>Subject: [MMUSIC] ICE Open Issue 5: alt parameter
>>
>>
>>At a previous meeting, it was mentioned that 3gpp had defined their own 
>>SDP parameter called "alt", creating an overlap in the specifications. 
>>One of the specs needs to change, as a result. However, I am not sure 
>>what 3gpp spec it was that used "alt", and whether it was still 
>>proceeding. Indeed, in such a conflict, its not clear who should move 
>>off the parameter name. I would think that IETF would get priority in 
>>such cases.
>>
>>Anyway, the issue is whether we should change the name or not. I'm 
>>hoping that someone can provide a reference for the 3gpp spec, and/or 
>>can provide an update on its status.
>>
>>I'd prefer to not change the name; there are already multiple 
>>implementations of ICE out there..
>>
>>-Jonathan R.
>>-- 
>>Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
>>Chief Technology Officer                    Parsippany, NJ 07054-2711
>>dynamicsoft
>>jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
>>http://www.jdrosen.net                      PHONE: (973) 952-5000
>>http://www.dynamicsoft.com
>>
>>
>>_______________________________________________
>>mmusic mailing list
>>mmusic@ietf.org
>>https://www1.ietf.org/mailman/listinfo/mmusic
> 
> 

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

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Magnus,

Thanks for these great comments, most were incorporated in the -03 draft 
which you can find at:

http://www.jdrosen.net/papers/draft-ietf-mmusic-ice-03.txt

inline for more.

Magnus Westerlund wrote:

> Hi Jonathan,
> 
> I have reviewed your latest ICE proposal and have some comments that I 
> don't think are in direct relation to the open issues you have presented.
> 
> 1. Section 4. page 8, third paragraph:
> "This would
>    typically be a TURN derived transport address, as it is most likely
>    to work with unknown non-ICE peers."
> 
> Is this really true. I would say it may be true for a client in a 
> private network. I don't think it is appropriate to make this statement. 
> I would say that it is highly dependent on the situation. One may have 
> public IP addresses, like most of us Swedes, but still need to run ICE 
> to allow others to get their connection correct.

It is indeed problematic. More text has been added to discuss this. 
There is no cure-all here; the TURN public address is still your best 
bet, but it won't always work.


> 
> 2. Section 5.1.1, fourth paragraph:
> "Each
>    alternate element conveys a transport address for RTP, one for RTCP,
>    a STUN username fragment and STUN password. "
> 
> What is the reason to not have separate user and passwords for RTP and 
> RTCP. They are separate channels, and one needs to ensure that a RTP 
> channel is not mapped to RTCP. If my understanding of ICE is correct, 
> the only way to ensure this is to use different usernames.

Yes. This was a bug. Usernames and passwords are required for both. The 
syntax now expresses that, and there is text explaining this.

> 
> 3. Section 5.3.1: Fourth paragraph:
> "These protocols work by having the client send, from a specific local
>    transport address, some kind of message to a server."
> 
> I think it might be clear if one replace "These" with "UNSAF".

fixed.

> 
> 4. Section 5.3.2: I am not comfortable with the notion to use the STUN 
> username to determine that it is STUN request. This will make all other 
> packets ending up in media protocol.

I don't follow this. We've discusssed this issue a bunch. The resolution 
proposed in this draft is that the transport address first listen for 
STUN, and then once an RTP packet is received, it switches to RTP.

> 
> 5. Section 5.3.2: page 14:
>   "The username fragment MUST be globally unique, so that no other host
>    will select a username with the same value."
> 
> My question is how you ensure this. You haven't specified a mechanism to 
> ensure that all clients will produce globally unique user names. I think 
> it is clear that one can produce randomly selected usernames that are 
> extremely likely to be unique, however to guarantee it seems impossible.

A proposed algorithm is provided, and the requirements say it needs to 
be unique with high probability.

> 
> 6. Section 5.3.3: Page 15,
> "If a user is a telecommuter, and therefore connected to their
>    corporate network and a local home network, they may prefer their
>    voice traffic to be routed over the VPN in order to keep it on the
>    local network when communicating within the enterprise, but use the
>    local network when communicating with users outside of the
>    enterprise."
> 
> I think the "local" on third line from the end, should be "corporate" 
> instead.

fixed.

> 
> 7. Congestion control:
> I think we will need to work out how congestion control should be 
> handled. In fact I think the initial full mesh STUN check can actually 
> be sufficient to force some low bit-rate networks into congestion. For 
> example, if I would connect my laptop with all its interfaces, and get 
> all sorts of different IP address that I could. I would think I would 
> end up with 4*4=16 different sourceable interfaces, and that is without 
> using any UNSAF mechanism, only the two IP versions and VPNs. Point is 
> that a peer can easily have around 10 announce addresses. If the other 
> peer then are going to try them in parallel and totally ignores to limit 
> the request rate, all 10 STUN request could go in a very short time. 
> This could easily congest the network, if the interface used is for 
> example a GPRS link, or something similar low bit-rate.
> 
> I do also agree with your later concerns over the timers. I think there 
> is need to figure out how to ensure that this scales and does not cause 
> congestion. Both in the simple case of the retry timer, and the quick 
> recovery method.

The timers have mostly been removed; there are no mid-session checks.

However, there is still an initial burst of connectivity checks. This 
seems like it would be very hard to do otherwise. We haven't sent 
anything on the network yet, so we don't know its transport 
characteristics to do any congestion control. If we try a slow 
exponential ramp up, it has a huge, and IMHO unacceptable impact on call 
setup. As such, I think we should leave it as is.

> 
> 8. Section 5.3.4, page 19: What is FSM? You don't spell it out.

Finite State Machine. Now expanded.

> 
> 9. Section 5.3.4, Figure 3:
> In the error/set retry timer transition from HANDSHAKING to BAD, isn't 
> the most likely case, that the giveup timer expires?

yes

> 
> 10. Section 5.3.4: page 20, second paragraph: Have you forgotten to 
> consider packet loss. I would think that packet loss would be likely 
> reason for the failure.

yes, though it would have to be over an extended duration - 9s for the 
stun transaction to timeout.

> 
> 11. Section 5.3.4: page 20, last sentence: Do you really intended the 
> refresh timer to be 1 second. That seems all to frequent to keep the 
> bindings alive, and ensuring that connectivity.

This timer is gone.

> 
> 12. Section 5.3.4: page 21, first paragraph: I think there is some need 
> to some consideration period before switching from Good to BAD. With the 
> current proposal it seems this will happen on the first packet loss the 
> STUN traffic encounters. I think it needs to be a bit more robust. I 
> think one can also to this state consider to take input from the 
> transport protocol sending media over the address combination.

This transition is gone now.

> 
> 13. Section 5.3.5, "Assuming the USERNAME is valid, the client MUST 
> generate a STUN response per RFC 3489."
> 
> Isn't a valid password also required?

yes. Thi s paragraph has been reworded.

> 
> 14. Section 6.1: last sentence:
>  "However, once one of the clients begins receiving media
>    on its STUN derived address, it can deallocate its TURN resources."
> 
> Is the above really a positive thing in ICE. If a peer has announced its 
> TURN address, shouldn't it keep it around. As I understand it can't be 
> deallocated by the other peer, so it will keep trying it. So the turn 
> address would always be in BAD state. Also having it around provides 
> fail over possibilities.

The spec says you SHOULD keep them, but often there is no need. The case 
yo are worried about won't happen. They can only be de-allocated when a 
higher priority transport was contacted and used. Unless you do 
re-invites in the sip case, you won't ever receive packets on them.

> 
> 15. section 9: I think the SDP attribute name could be better selected. 
> I don't think "alt" is really a good name for ICE address elements. Even 
> a=ice would seem better.

It is now a=candidate.


> 
> 16. Section 9. The syntax for the attribute seems broken in two ways:
> 
> There is a missing separator between the RTP and RTCP address, I think 
> it should read:
> unicast-address SP port [SP unicast-address SP port]
> 
> Secondly the syntax element:
> derived-from  = ":" / id
> 
> is not used.

fixed.

> 
> 17. Section 9:
>   "If the ICE message indicates an RTCP address and port that are not
>    one higher than that of the RTP, the SDP RTCP attribute [2] MUST be
>    used to convey them."
> 
> Shouldn't this include the word "default" and read like this:
> 
> If the ICE message indicates an default RTCP address and port that are 
> not one higher than that of the RTP, the SDP RTCP attribute [2] MUST be 
> used to convey them.

I don't think default is needed; there is no default defined in ICE.

> 
> 
> 18. Section 9. In regards to my above comment number 2. I think the 
> attribute should clear separate RTP and RTCP addresses and allow for 
> individual username and passwords.

done.

> 
> 19. Reference 12, need to include the standard number STD 64.

xml2rfc does what it does. I can't fix this.

Thanks,
Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@dynamicsoft.com                        FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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Jon,

Good question. Basically, this was a bug in ICE.

Each local transport address has to have its own username and password. 
Now, in -03, each local transport address is either used in the 
initiate/accept mesage, or else is associated with a single derived 
transport address. This means that if I have a single local interface 
and can use TURN and STUN unilateral allocations, I'll bind to *three* 
local interfaces. One of them gets sent directly to the peer. One is 
used for the STUN allocations. One is used for the TURN allocations. A 
stun server is set to run locally on all three sockets, and each of the 
three has a unique username/password.

If you don't do this, you run into a nasty problem which is there is no 
way to reliably know that your peer has suceeded in its connectivity 
checks. That is needed to allow you to turn off the stun server and 
accept only media packets. This also simplified the protocol in several 
places.

The drawback (compared to having multiple derived transport addresses 
from a single local transport address) is that it increases your port 
usage. I was hesitatnt to go this route at first because of this, but 
the alternative (not being able to turn off the stun servers) seemed 
worse based on group consensus on this point.

The problem is probably only a big deal in servers like gateways and 
conference bridges. However, if those do ICE, I wouldn't expect them to 
do turn or stun, in which case there is no additional cost in terms of 
socket resources.

HTH,
Jonathan R.

Jon Ringle wrote:
> I'm a little confused regarding the use of username fragments in ICE. The 
> following statement is in draft-ietf-mmusic-ice-02, Section 5.3.2:
> 
>    For each local transport address where a STUN server is running, the
>    client MUST choose a username fragment and a password.  The username
>    fragment created by the client will be concatenated with the fragment
>    created by its peer.  The result will serve as the username provided
>    by its peer in STUN requests.  By creating the username as a
>    combination of information from each side of a call, it allows a
>    client to correlate the source of the request with a candidate
>    transport address.  This is discussed further below.
> 
>    The username fragment MUST be globally unique, so that no other host
>    will select a username with the same value.  This username fragment
>    and password will be passed to its peer in an initiate or accept
>    message.  As such, the process described in this section will
>    associate, with each local transport address, a username fragment and
>    password.  The client also associates this same username fragment and
>    password with any transport addresses derived from the local
>    transport address.
> 
> Suppose phone-A has only one local transport address on eth0 and derives a 
> STUN and TURN address from the eth0 local transport address. From what I read 
> above, it seems like only a username fragment and password is chosen only for 
> the eth0 address, and that the derived STUN/TURN addresses use this same 
> username/password. So, this would result in alt's in SDP in the initiate 
> message from phone-A to phone-B like this:
> 
> a=alt:1 1.0 : PhoneAeth0user password 172.16.1.102 1234
> a=alt:2 0.8 : PhoneAeth0user password 1.1.1.4 8801
> a=alt:3 0.4 : PhoneAeth0user password 1.1.1.5 3002
> 
> This seems to go against the examples that are shown in 
> draft-rosenberg-sipping-nat-scenarios-03, where a separate username fragment 
> is shown for each local and derived addresses.
> 
> It seems to me that separate username fragments would be needed for each 
> gathered address for correlation to the candidate table.
> 
> I feel like I am missing a key concept here. Could someone please explain.
> 
> Thanks,
> 
> Jon
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@dynamicsoft.com                        FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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Paul,

I've come around to this line of thinking too, and this is what is 
reflected in -03, now available at:

http://www.jdrosen.net/papers/draft-ietf-mmusic-ice-03.txt

more inline.

Paul Kyzivat wrote:
> I agree with the direction Jon is suggesting here, but think it is part 
> of a bigger activity.
> 
> Without some kind of recovery mechanism, what usually happens is that 
> one or both parties on a call notice that something is wrong with the 
> call - either the media malfunction in some way, or there is some 
> indication that signaling isn't working properly. At that time, the 
> users take remedial action - usually attempting to reestablish the call.
> 
> I think there should be an analogous process that the endpoints perform 
> on their own, so that the users aren't required to do anything other 
> than tolerate a potential glitch. If this were well specified it 
> wouldn't suffer from the problem that often happen when users do it - 
> such as glare resulting in both getting a busy signal.

Right.

> 
> To achieve this, something needs to monitor the session and identify 
> when there may be a potential problem. A lot of this is monitoring media 
> sessions. How this is done is inherently media specific.

I agree.

> 
> For two way RTP based audio media this may, as Jon suggests, only 
> require checking that packets are received periodically, and helping the 
> other end by always sending packets periodically. For one way or idle 
> media this may be a bit more tricky. Sending STUN packets is perhaps one 
> answer, but maybe it is ok to send rtp idle packets (did those get 
> approved?) if it is agreed they can be sent even on a RECVONLY or 
> INACTIVE connection.

If you want to monitor quality, I think RTCP does the job. I don't think 
you need to send additional packets to generate data to measure. In any 
case, determining that there has been a quality failure is definitely 
outside the scope of ICE.

In -03, STUN is now only used for the initial checks. mid-call 
keepalives are handled at the application layer, and there is text on 
doing that for RTP, which talks about this music-on-hold problem. That 
is differnt still from detecting failed media to do something about it, 
which is not addressed at all, beyond a hook I left.

The hook that's in there is the treatment of re-invites. If you get one, 
and nothing has changed, don't do anything. That is, a re-invite in and 
of itself does not trigger you to restart connectivity checks. But, if 
the re-invite adds a new candidate address that hasn't been seen before, 
connecitivyt checks are sent to it. "never seen before" is determined 
based on boht the transport address AND the username fragment. This 
means that a client can use a transport address, and then if it thinks 
there are problems, start up the stun server again, change the username, 
and do a reinvite to trigger a recheck.


> 
> A similar approach would presumably work for video. Session mode IM will 
> require something different, but the concept is the same.
> 
> Once there is a hint of a problem, testing and recovery can commence. 
> Part of this is to test the signaling path. This can be any mid-dialog 
> request, but could easily be a reinvite or update that does a new 
> offer/answer. Exactly what kind of offer/answer to do isn't entirely 
> clear. If I think there is a problem, but don't think I am at fault, the 
> "obvious" thing for me to do is to again offer the same SDP I used last. 
> However, that may well solicit an answer the same as last time, and 
> result in absolutely no change in the media.
> 
> There really needs to be some kind of hint in the reinvite or the 
> accompanying offer that there is a problem to be diagnosed or recovered 
> from. Maybe this would be a Reason header in the reinvite, or something 
> (new preconditions?) in the offer. This should either result in 
> initiating diagnostic actions on one or more media streams, or 
> replacement of resources (ports, relays, etc.) in the path of one or 
> more media streams, or both.

Exactly what to do needs futher consideration. I've left just a hook for 
now.

> 
> There are of course lots of failure modes to consider. In some cases it 
> may be the entire peer UA that has failed, but that has a backup, or it 
> could be a turn server, msrp relay, or NAT that has failed. In any case, 
> the problem extends beyond the scope of what ICE deals with.

Yup.

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
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Cisco Systems
jdrosen@dynamicsoft.com                        FAX:   (973) 952-5050
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Folks,

Last August, we had some threads on open issues in ICE. This note
summarizes the resolution to each of these in the recently updated -03
document.

Issue 1: v6trans applicability
> Currently, the ICE spec describes its usage in facilitating the 
> transition to v6. It points out two possible usages. The first is for
>  dual-stack hosts, to allow them to advertise addresses for both v4 
> and v6, and then use the one which has connectivity. Another one is 
> for hosts with both v6 native and 6to4 addresses, to detect if native
>  v6 connectivity exists.
> 
> The open issue is, are these valid and useful usages? Are there other
>  ones? This isn't so much as an open issue, as it is a "todo", where 
> the "todo" is to vet this with some folks from v6ops. As such, any 
> thoughts or inputs here are welcome. I will try to seek some input 
> while I'm at ietf as well.

I havent gotten feedback yet, but I believe the applicability is correct.

Issue 2: processing STUN requests before the accept
> Consider the following scenario.
> 
> User A sends an ICE initiate message (i.e., SIP INVITE with SDP). It
> contains the username fragment contributed by user A. THis arrrives
> at the called party, user B. User B sends an ICE accept message
> (i.e., a 200 OK with an SDP answer). This contains the username
> fragment contributed by user B. User B, in parallel, sends a STUN
> request to user A. THat request has a username, and the username is
> the concatenation of the two username fragments. Most likely, this
> STUN message will beat the accept message (the 200 OK) back to user
> A, and be rejected. This case is handled in the spec by having user B
> retry (although this is far from obvious and needs to be clarified).
> 
> However, another behavior is possible. Instead of rejecting it, user
> A could accept these STUN requests, so long as the first half of the
> USERNAME matches the fragment sent by user A in its initiate message.
> However, until the accept message is received, user A won't be able
> to correlate the source of that stun request with the IP/port/STUN
> username fragment/password of its peer, and thus won't be able to
> generate its own stun request back to the source of the incoming stun
> request. It could cache the source IP/port of the stun request if it
> were new, and then gather the information later when the accept
> message arrives. However, this increases complexity.
> 
> What do we gain by having the user accept the stun request? Slightly
> faster operation, since the called party (user B) may be able to do
> its stun connectivity checks even before the 200 OK reaches the
> caller.
> 
> So, the open issue is, do folks want to allow user A to  accept these
> STUN requests before the accept message?
> 
> I'm on the fence somewhat, but probably would vote for no.

There seemed to be agreement not to allow this. -03 disallows it.

Issue 3: timer settings
> Currently, the ICE spec uses a number of timers to control how
> frequently a STUN transaction should be re-attempted to verify the
> current state. These timers are:
> 
> Refresh timer: While in the GOOD state, when to try again to verify
> that its still GOOD? Default: 5s
> 
> Rapid Retry timer: While in the HANDSHAKE state, when to try again
> while you wait for your peer to get the accept message and generate
> its STUN packets to open permissions? Default: 1s
> 
> Giveup Timer: While in the HANDSHAKE state, when to give up, and
> assume the address is no good. Default: 50s
> 
> Retry Timer: while in the BAD state, when to retry to make sure the
> address is still BAD. Default: 60s
> 
> 
> 
> The timer that needs the most work is the refresh timer. I think that
> this should probably be a fraction of the bandwidth of the media
> traffic, as a lower bound, just like RTCP.
> 
> The real issue is what to do in failure cases. THe spec currently
> says that, if the client should find that its addresses are all in
> the BAD state all of a suddent, to push them all into the INIT state
> and try to recover. This is meant to deal with a failure of a nat in
> the middle of a call that wipes out all of the bindings. Now, whilst
> a rapid recovery behavior is good here, it can lead to real problems
> if the cause of the failure was NOT a NAT failure, but congestion. In
> that case, the response to the congestion loss is to send more
> packets - i..e, STUN requests. Thus, some kind of careful retry and
> backoff approach is needed. The alternative is to simply not do this.
> 
> 
> To summarize, the open issues are:
> 
> 1. Are the timers properly set, and should we define a lower bound on
> the refresh timer as a fraction of the media bandwidth (I say yes)
> 
> 2. How should we handle the case where all of the addresses enter the
> BAD state? Should we support this reconnect mode?

This issue is mostly resolved by removal of the mid-session checks.
There are still two timers. One controls the gap between STUN
transctions in the HANDSHAKE state, which is 50ms (this is NOT the
interval between packets - it is the interval between transaction
attempts), and a timer for giving up in the handshake state, which is
60s. I think these are fine.

Issue 4: TCP applicability
> Currently, ICE is mostly worried about media streams over UDP. For
> TCP, the problem is less severe, but it does exist. The connectivity
> checks are certainly much easier in the TCP case; the ability to set
> up a connection represents a valid connectivity check. Its not clear
> that STUn should be sent within the connection. The framing and demux
> are more complex for TCP than UDP.
> 
> So, the question is, what, if anything, should ICE say about TCP?
> 
> Here is my proposal:
> 
> ICE should be applicable to TCP. In particular, an initiator or
> responder can send messages that include alternates for TCP streams.
> However, the connectivity checks would not be done using STUN.
> Rather, they would be done using something thats media specific. The
> initial check is easy - the successful establishment of the
> connection. Subsequent checks would be done using something that is
> specific to the media. It *could* be stun, but that would be
> specified by someone else.
> 
> ICE also specifies that, when a client receives a request, it checks
> the source IP/port and considers that a candidate too. I don't think
> that behavior really makes sense for TCP. If I receive a connection
> setup (i.e., SYN), the resulting connection could be used to send
> data to my peer. There is no point in trying to establish a
> connection to that source IP/port.
> 
> I think that this works well with comedia, where a passive or actpass
> endpoint would list alternates where it can be reached. Once the
> connection is established, one cannot anyway open one in the reverse
> direction.
> 
> Do folks agree with this proposal?

There was mixed opinions on this. Colin made what i felt was the most
convincing argument - the industry need is for UDP. Lets solve that
first and worry about other things later. Since TCP clearly differs, it
would complicate an already complicated protocol even further. So, TCP
is not treated in -03.

Issue 5: alt parameter
> At a previous meeting, it was mentioned that 3gpp had defined their
> own SDP parameter called "alt", creating an overlap in the
> specifications. One of the specs needs to change, as a result.
> However, I am not sure what 3gpp spec it was that used "alt", and
> whether it was still proceeding. Indeed, in such a conflict, its not
> clear who should move off the parameter name. I would think that IETF
> would get priority in such cases.
> 
> Anyway, the issue is whether we should change the name or not. I'm
> hoping that someone can provide a reference for the 3gpp spec, and/or
> can provide an update on its status.
> 
> I'd prefer to not change the name; there are already multiple
> implementations of ICE out there..

Folks wanted to change it for the most part. So, it is now "candidate".
This results in a complete break in compatibility with past revs of the
draft :(

Issue 6: non-STUN connectivity checks
> The ICE specification uses STUN, multiplexed within the media stream,
> as a form of connectivity check. STUN was chosen originally since, in
> addition to providing a check, it also told the client about its
> source IP/port as seen by the server. However, in this latest rev of
> ICE, the client does not need to learn its IP/port as seen by the
> server. The STUN server itself (which is embedded in the peer in the
> call) uses the source IP of the STUN request as a new potential media
> destination.
> 
> Since the primary feature provided by STUN is no longer required, it
> begs the question about what features we do need, and whether STUN
> remains the ideal choice. Here are the functions we need from any
> connectivity check protocol:
> 
> 1. One side sends a packet to the other which generates an immediate
> response
> 
> 2. The response gets back to the originator with high probability,
> informing the sender that their original request was received
> 
> 3. The request conveys the source IP/port to the recipient (pretty
> much any IP protocol gives us this)
> 
> 4. The request can contain some kind of correlation tag, allowing the
> recipient to correlate it as coming from an IP/port advertised in an
> SDP offer or answer
> 
> 5. The request can contain a unique identifier, provided by the
> client, so it can be sure that the connectivity request was generated
> in response to the SDP offer or answer it sent.
> 
> 
> Clearly, STUN meets these requirements. (1) is met because it
> generates a response immediately, (2) is met because STUN is reliable
> due to client retransmits, and the response goes back to the source
> IP/port, (3) is met easily, (4) is met by using one half of the
> USERNAME, and (5) is met by using the other half.
> 
> Now, it is possible for other things to work. In particular, RTP
> itself might work:
> 
> (1) We could make use of an immediate RTCP packet sent in response to
> an RTP packet. I believe the RTP no-op proposal had a flag for
> signaling this. However, RTCP might fail because its sent to a
> different IP/port, thus it is not as reliable as STUN.
> 
> (2) We could implement a retransmit strategy ontop of RTP to get
> reliability
> 
> (3) is met
> 
> (4) We could use the approach described in
> http://www.ietf.org/internet-drafts/draft-peterson-rosenberg-avt-rtp-ssrc-demux-00.txt,
> which allows each side to signal its own half of the SSRC, and then
> use one half for the sender correlator
> 
> (5) the other half provides the unique identifier
> 
> 
> So, what to do here? The options I see are:
> 
> 1. keep it STUN only as specified, 2. allow for multiple mechanisms,
> define STUN now, mandate STUN as baseline, and see what happens to
> the ssrc demux mechanism, specifying it as an alternative down the
> road 3. drop STUN completely, move to RTP with the immediate RTCP
> capability and SSRC demux (those would need to be specified within
> AVT)
> 
> I am inclined towards (2). I don't like (3) since it will
> substantially delay ICE, since it would depend on two mechanisms in
> AVT which don't have consensus or clear direction at all, and RTP
> doesnt work as well. I think (2) is better than (1) since the
> multiplexing of STUN and RTP on the same port is ugly, and it would
> be nice to have an out down the road if something better can be
> specified. However, we absolutely need a baseline mechanism to have
> interoperability, and STUN is that.

The current draft removes mid-session connectivity checks. Initial
checks are with STUN. Keepalvies are application dependent and done with
RTP for RTP applications. There is no way to negotiate something besides
stun in this document. I think the need for an alternative is reduced by
eliminating the mid-session checks.


Thanks,
Jonathan R.

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


_______________________________________________
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From mmusic-bounces@ietf.org  Tue Oct 26 02:04:25 2004
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Subject: [MMUSIC] updated ICE spec
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Folks,

I submitted an updated ICE spec yesterday. Until it appears in the 
archives, you can pick up a copy at:

http://www.jdrosen.net/papers/draft-ietf-mmusic-ice-03.txt

This is a fairly sizeable revision, but much of it is cleanup and 
addition of further clarification material. A lot was also the "dotting 
of the i's and crossing of the t's" - completion of IANA registrations, 
adding diagrams, added explanatory material for call flows, etc.

We're currently debating on the list what I think the main issues are - 
how/when to do the checks - STUN vs. other things, mid-call vs. not. -03 
proposes a specific resolution which is consistent I think with some of 
the recent threads, but we should debate that furhter. Beyond that, I 
believe we have resolved issues discussed in the past. I've sent a 
summary of how past open issues were resolved, including responding to 
several older emails with comments and issues.

Please raise any furher issues or comments to the list.

Below is a detailed list of diffs from -02 to -03:

* changed name of "alt" SDP attribute to "candidate" to avoid conflict with
3gpp attribute of the same name.

* removed derived-from SDP attribute, as it is no longer used

* changed username SDP attribute component to user-frag

* updated XML schema for ICE messages to allow both an IPv4 and IPv6
default.

* updated the mapping procedures for ICE to SDP based on the new
schema and SDP attribute definitions

* noted how performing the address gathering while the user is being
alerted, or upon entry of the user into the address book or buddy
list, can reduce call setup times

* changed TURN and STUN for the purposes of unilateral allocation to
SHOULD implement from MUST. Added a long-winded and ranting paragraph
on why you really, really to think hard before violating that SHOULD.

* allow, at MAY strength, other protocols for creating derived
transport addresses

* added requirement that password have 128 bits of randomness, and
that protocols that do ICE must provide and should use encryption and
integrity services. In the case of SIP, made sips usage a SHOULD,
SMIME a MAY. Passwords have to differ for each media stream for each
session. Usernames are just unique per transport address.

* added text describing Pete's scenario where a stun connectivity
check is received by the wrong host.

* removed mid-session connectivity checks, resulting in adjusted FSM.

* STUN service needs to only run until a media packet is received; at
that point you can safely switch to media-only operation

* added discussion on selecting of default address. Note that
TURN-based address from a public relay won't always work. Describe a
case where it won't, but still RECOMMEND to use such an address. Add a
note on deployment considerations to obviate this, by deploying ICE to
communities in whcih joint communications is common (i.e., do the
whole enterprise).

* adjusted algorithm that runs on receipt of stun requests,
simplifying the case where a new candidate address is learned

* added pictures to describe turn derived stun addresses

* added basic ice model call flow for tutorial purposes

* previously, a client obtained local transport addresses and then
both used those in its SDP, in addition to deriving perhaps several
derived transport addresses from it. No longer. No, there is a single
local transport address for each derived transport address, and that
local transport address is never advertised in ICE. This turned out to
be needed to figure out whether the local stun service could be turned
off when an RTP packet is received. Unfortunately it also required me
to introduce more terminology to describe.

* added a section on application layer keepalive issues. Namely, how
to deal with "RTP on hold". This basically replaces the STUN-based
connectivity checks

* removed concept of sourceable addresses; it fell away with the above
change where different local transport addresses are used to obtain
each derived transport addresses.

* if a gathered transport address is equal to a local transport
address, use that local transport address

* rewrote security considerations section, which was wrong. Focus on
showing how the attacks present in "regular stun" are completely
eliminated with ICE

* updated and simplified example and filled in textual description

* added discussion on handling of re-invites in SIP; the technique
allows for connectivity checks to be redone by changing the username
fragment. Otherwise, a re-invite where the SDP has not changed has no
effect.

* filled in details of IANA registration of the candidate
parameter. Added registrations of the XML namespace and schema.

* fixed incorrect assumptions on 6to4 behavior in IAB considerations

* moved TURN to a normative reference

* added missing space between RTP and RTCP information in SDP
attribute grammar

* removed stale derived-from SDP field

* RTCP information is optional in ICE now

* provided sample algorithm for selecting username fragments

* justify the inclusion of the id component of the candidate attribute

* specify what to put in the CHANGED-ADDRESS element in a STUN
BindingResponse.


Thanks,
Jonathan R.

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


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- What are the market drivers for VoWLAN? 
- How will VoWLAN systems prioritize and deliver data packets? 
- What are the planning and engineering issues? 
- What about security and privacy? 

At the WiFi Voice Conference to be held in Paris on 10 to 13 May, 2005.
distinguished experts and key players in the field 

will address these questions with detailed and technical explanations. 

A call for papers is online at:

 <http://www.upperside.fr/wifi05/wifivoice2005intro.htm>
http://www.upperside.fr/wifi05/wifivoice2005intro.htm

 


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<p class=3DMsoNormal><font size=3D1 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
9.0pt;font-family:Arial'>- What are the market drivers for VoWLAN? <br>
- How will VoWLAN systems prioritize and deliver data packets? <br>
- What are the planning and engineering issues?&nbsp;<br>
- What about security and privacy?&nbsp;<br>
<br>
At the WiFi Voice Conference to be held in <st1:place =
w:st=3D"on"><st1:City
 w:st=3D"on">Paris</st1:City></st1:place> on <strong><b><font =
color=3D"#6495be"
face=3DArial><span style=3D'font-family:Arial;color:#6495BE'>10 to 13 =
May, 2005.</span></font></b></strong>
distinguished experts and key players in the field =
<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
9.0pt;font-family:Arial'>will address these questions with detailed and
technical explanations. <br>
<br>
<span class=3Dtextebold>A call for papers </span><span class=3Dtexte>is =
online at:</span><o:p></o:p></span></font></p>

<p class=3DMsoNormal><span class=3Dtexte><font size=3D1 =
face=3DArial><span
style=3D'font-size:9.0pt;font-family:Arial'><a
href=3D"http://www.upperside.fr/wifi05/wifivoice2005intro.htm"
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- What are the market drivers for VoWLAN? 
- How will VoWLAN systems prioritize and deliver data packets? 
- What are the planning and engineering issues? 
- What about security and privacy? 

At the WiFi Voice Conference to be held in Paris on 10 to 13 May, 2005.
distinguished experts 

and key players in the field will address these questions with detailed and
technical explanations. 

A call for papers is online at:

 <http://www.upperside.fr/wifi05/wifivoice2005intro.htm>
http://www.upperside.fr/wifi05/wifivoice2005intro.htm

 


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<p class=3DMsoNormal><font size=3D1 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
9.0pt;font-family:Arial'>- What are the market drivers for VoWLAN? <br>
- How will VoWLAN systems prioritize and deliver data packets? <br>
- What are the planning and engineering issues?&nbsp;<br>
- What about security and privacy?&nbsp;<br>
<br>
At the WiFi Voice Conference to be held in <st1:place =
w:st=3D"on"><st1:City
 w:st=3D"on">Paris</st1:City></st1:place> on <strong><b><font =
face=3DArial><span
style=3D'font-family:Arial'>10 to 13 May, =
2005.</span></font></b></strong>
distinguished experts <o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D1 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
9.0pt;font-family:Arial'>and key players in the field will address these
questions with detailed and technical explanations. <br>
<br>
<span class=3Dtextebold>A call for papers </span><span class=3Dtexte>is =
online at:</span><o:p></o:p></span></font></p>

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title=3D"http://www.upperside.fr/wifi05/wifivoice2005intro.htm"><font
color=3Dblack><span lang=3DEN-GB =
style=3D'color:windowtext'>http://www.upperside.fr/wifi05/wifivoice2005in=
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size=3D1 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:9.0pt;font-family:Arial'><o:p></o:p></span></font></p>=


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From mmusic-bounces@ietf.org  Tue Oct 26 20:32:14 2004
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Jonathan Rosenberg wrote:

>
> * previously, a client obtained local transport addresses and then
> both used those in its SDP, in addition to deriving perhaps several
> derived transport addresses from it. No longer. No, there is a single
> local transport address for each derived transport address, and that
> local transport address is never advertised in ICE. This turned out to
> be needed to figure out whether the local stun service could be turned
> off when an RTP packet is received. Unfortunately it also required me
> to introduce more terminology to describe.
>
If I understand correctly to fulfill this requirement a device that uses 
both TURN and STUN and is starting an RTP/RTCP session will open six 
sockets(3 for RTP and 3 for RTCP) and run a STUN server on each of them

The advantage of this approach as you mentioned is that the STUN service 
can be stopped when an RTP packet is received. What will be the downside 
of keeping the STUN service running?

Thanks,
Ali Awais


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Subject: RE: [MMUSIC] Confusion regarding ICE username fragment
Date: Tue, 26 Oct 2004 17:09:05 -0700
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Thread-Topic: [MMUSIC] Confusion regarding ICE username fragment
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One comment inlined.=20

> -----Original Message-----
> From: mmusic-bounces@ietf.org=20
> [mailto:mmusic-bounces@ietf.org] On Behalf Of Jonathan Rosenberg
> Sent: Monday, October 25, 2004 10:36 PM
> To: ml-mmusic@ringle.org
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] Confusion regarding ICE username fragment
>=20
> Jon,
>=20
> Good question. Basically, this was a bug in ICE.
>=20
> Each local transport address has to have its own username and=20
> password.=20
> Now, in -03, each local transport address is either used in=20
> the initiate/accept mesage, or else is associated with a=20
> single derived transport address. This means that if I have a=20
> single local interface and can use TURN and STUN unilateral=20
> allocations, I'll bind to *three* local interfaces. One of=20
> them gets sent directly to the peer. One is used for the STUN=20
> allocations. One is used for the TURN allocations. A stun=20
> server is set to run locally on all three sockets, and each=20
> of the three has a unique username/password.
>=20
> If you don't do this, you run into a nasty problem which is=20
> there is no way to reliably know that your peer has suceeded=20
> in its connectivity checks. That is needed to allow you to=20
> turn off the stun server and accept only media packets. This=20
> also simplified the protocol in several places.

I'd like to point out that there are use cases for keeping STUN server
running on the chosen endpoint (i.e., <address, port>) even after
connectivity checks have succeeded.=20

For RTSP where media packets flow one way from the RTSP server to its
client, it is useful to keep periodic STUN exchanges to keep alive the
NAT bindings. It is known in the real world that cerntain NAT bindings
will be deleted if there is no traffic going a specific direction
(normally from outside in, if I remember correctly).

I do agree though the other STUN servers can be closed after the
connectivity phase.


>=20
> The drawback (compared to having multiple derived transport=20
> addresses from a single local transport address) is that it=20
> increases your port usage. I was hesitatnt to go this route=20
> at first because of this, but the alternative (not being able=20
> to turn off the stun servers) seemed worse based on group=20
> consensus on this point.
>=20
> The problem is probably only a big deal in servers like=20
> gateways and conference bridges. However, if those do ICE, I=20
> wouldn't expect them to do turn or stun, in which case there=20
> is no additional cost in terms of socket resources.
>=20
> HTH,
> Jonathan R.
>=20
> Jon Ringle wrote:
> > I'm a little confused regarding the use of username=20
> fragments in ICE.=20
> > The following statement is in draft-ietf-mmusic-ice-02,=20
> Section 5.3.2:
> >=20
> >    For each local transport address where a STUN server is=20
> running, the
> >    client MUST choose a username fragment and a password. =20
> The username
> >    fragment created by the client will be concatenated with=20
> the fragment
> >    created by its peer.  The result will serve as the=20
> username provided
> >    by its peer in STUN requests.  By creating the username as a
> >    combination of information from each side of a call, it allows a
> >    client to correlate the source of the request with a candidate
> >    transport address.  This is discussed further below.
> >=20
> >    The username fragment MUST be globally unique, so that=20
> no other host
> >    will select a username with the same value.  This=20
> username fragment
> >    and password will be passed to its peer in an initiate or accept
> >    message.  As such, the process described in this section will
> >    associate, with each local transport address, a username=20
> fragment and
> >    password.  The client also associates this same username=20
> fragment and
> >    password with any transport addresses derived from the local
> >    transport address.
> >=20
> > Suppose phone-A has only one local transport address on eth0 and=20
> > derives a STUN and TURN address from the eth0 local=20
> transport address.=20
> > From what I read above, it seems like only a username fragment and=20
> > password is chosen only for the eth0 address, and that the derived=20
> > STUN/TURN addresses use this same username/password. So, this would=20
> > result in alt's in SDP in the initiate message from phone-A=20
> to phone-B like this:
> >=20
> > a=3Dalt:1 1.0 : PhoneAeth0user password 172.16.1.102 1234
> > a=3Dalt:2 0.8 : PhoneAeth0user password 1.1.1.4 8801
> > a=3Dalt:3 0.4 : PhoneAeth0user password 1.1.1.5 3002
> >=20
> > This seems to go against the examples that are shown in=20
> > draft-rosenberg-sipping-nat-scenarios-03, where a separate username=20
> > fragment is shown for each local and derived addresses.
> >=20
> > It seems to me that separate username fragments would be needed for=20
> > each gathered address for correlation to the candidate table.
> >=20
> > I feel like I am missing a key concept here. Could someone=20
> please explain.
> >=20
> > Thanks,
> >=20
> > Jon
> >=20
> >=20
> >=20
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >=20
>=20
> --=20
> Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
> Director, Service Provider VoIP Architecture   Parsippany, NJ=20
> 07054-2711
> Cisco Systems
> jdrosen@dynamicsoft.com                        FAX:   (973) 952-5050
> http://www.jdrosen.net                         PHONE: (973) 952-5000
> http://www.cisco.com
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

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To: Euchner Martin <martin.euchner@siemens.com>
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Cc: "Karl Norrman \(KI/EAB\)" <karl.norrman@ericsson.com>,
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Subject: [MMUSIC] Re: draft-ietf-mmusic-kmgmt-ext-11.txt: KMID? IANA
 considerations section
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Hello Martin,


Euchner Martin wrote:

 > Hello Mats and MIKEY gang,
 >
 > Thanks once more for the helpful responses. I've added inline below my
 > comments to the exchange before. I believe we are getting very close now.

Yes it seems that on almost all *detail* we do agree, but we draw
different conclusions in the end from these details. You argue
there is a bidding-down attack that can/should be mitigated, we
argue that is really a side-effect of something else that cannot
be solved in the way you propose.

In this mail, we will argue again that there is no bidding-down
attack issue (the issue is on another level), and furthermore,
the countermeasure you suggest is not protecting against it.

 >
 > So to summarize the discussion and where we stand:
 > 1) There is nothing wrong with MIKEY; all known MIKEY protocol variants
 > provide sufficient and good security to protect against manipulation
 > attacks on the SDP key management phase.
 >
 > 2) I don't see kmgtext-11 protocol being flawed either.
 > 3) I'm a bit concerned that the overall security very much depends on a
 > correct implementation of the responder, in particular the right
 > processing order of the steps. My concern is fostered by the feeling
 > that while for security experts, the correct and secure processing order
 > is obvious, I'm not so certain, if it is also that obvious for
 > non-security experts; especially, since I could not find any advice/or
 > caution anywhere in the kmgtext document.
 >
 > So I would like to ask the WG:
 > Do you see a need to add some advice or caution (possibly in the
 > security consideration section) that explains what to do on the
 > responder side?
 >
 > I would highly appreciate to see such a text as some generally useful
 > non-obvious but important detail.
 >
 > 4) We clarified that the KMID is on the key management protocol level,
 > the current KMID scheme does not reflect the MIKEY protocol variant 
level.
 >
 > 5) We are in agreement that the bidding-down attack can be mitigated at
 > the responder as long as the responder implements all possible MIKEY
 > protocol options, even those that are not of interest to the responder,
 > and of course that the responder verifies each received option to detect
 > if it is an injection/replacement/replay etc.

Here, it is important to be precise. The bidding down can be
*detected* if all variants are implmented, but there cannot even be
a bidding down to something that is not supported. In either
case, it's a non-issue in our opinion.

Also, this seems to contradict your concern about the security
relying on implementation errors at the Responder:
suppose there *is* a really weak variant, which is used to launch
the (bidding down) attack. Wouldn't it be better NOT to implement
that variant, completely avoiding bidding-down attacks, rather
than implementing it just for the sake of *detecting* an attack, and
at the same time opening up for an implementation error that
by accident actually *accepts* the weak offer?

(We argue more below why you cannot detect the attack without
implementing all variants.)

 >
 > 6) We are also in agreement that we are concerned about bidding-down
 > attacks on the protocol level, but not on the algorithm/parameter level;
 > the latter would assume a very insecure MIKEY protocol anyhow; we can
 > de-scope this easily from our discussion.

OK, but why then the concern in 5, which is based on this insecure
MIKEY option?

 >
 > 7) I believe that we are further in agreement that the current mechanism
 > operating on a key management protocol level and the procedures are
 > secure but come at a high price (number of implemented options,
 > performance).
 >
 > I am clearly concerned about the consequences, namely that the existing
 > mechanism is very costly. Do we really want this?? Or can we try to 
improve?
 >

Why costly? We have made *provisions* for offering multiple key mgmt
protocol, but you don't need to implement more than MIKEY (or some
other, single, protocol). We do not mandate multiple protocols, not
even all MIKEY variants are mandatory.

The mechanism is to be used for those (exceptional) cases
where

1. the Initator support more than one protocol, but,
2. has no clue which the Responder supports.


 > 8) Where could we improve it? Which bidding-down attacks are worth it?
 > This is a requirements question:
 >
 > I would like to see
 > a) better (i.e. cheaper) yet equally secure protection mechanism in
 > place that quickly detects a maliciously injected/substituted MIKEY
 > protocol option that the responder has not implemented and/or doesn't
 > want to support.

Why is it considered a "malicious" act to inject something that
the reponder doesn't want to use or perhaps doesn't even support?
This is DoS, and could be achieved much simpler, e.g. deleting all
key mgt offers, or just garbling them.

Given that

- we want to open up the possibility for more than one offer,
- only the Responder can detect a substitution, and,
- he can only verify those options which he/she "understands",

we do not see how it can be made simpler, yet still secure
(more on that below).


 >
 > b) Possibility for the responder to quickly decide without full
 > verification of all offers not to consider further a particular offer.

This is impossible if the arguments above are correct. We have
just concluded (or at least we think we can convince you, see below)
that to *detect* the attack, you may in the worst case
need to implement all variants/offered protocols. From this (if it is
correct) it seems to follow that you cannot omit the
verification of any single offer, or?

 >
 > c) The responder being able to quickly sort out attacks from a
 > bidding-down attempt.
 >
 > So in essence I believe I'm requiring that the MIKEY protocol identifier
 > becomes an authenticated one.

We still argue it doesn't solve the real issue (should such exist).

 >
 > Imagine an example with a PKI enabled environment where MIKEY-DHSIGN or
 > MIKEY-PK are possible but MIKEY-PS is also available and implemented and
 > where the security policy of the responder would always prefer a PKI
 > option unless such one is not offered.
 >
 > In this example, the responder upon reception of (DHSIGN, PK, PS) offer
 > would just decide which of both DHSIGN or PK to take, and then verify
 > the offered and chosen MIKEY protocol string. The responder does not
 > need to verify the other offered MIKEY strings. So there is only one
 > MIKEY instantiation instead of three. Wouldn't that be worth it?
 >
 > In the example above where the responder receives an attacked offer of
 > (DHSIGN, PK; PW) and PW is not supported or implemented, the PW option
 > can instantly be sorted out and would not be processed any further.

 > Imagine an example with a PKI enabled environment where MIKEY-DHSIGN
or MIKEY-PK are possible but MIKEY-PS is also available and implemented
and where the security policy of the responder would always prefer a PKI
option unless such one is not offered.
 >
 > In this example, the responder upon reception of (DHSIGN, PK, PS)
offer would just decide which of both DHSIGN or PK to take, and then
verify the offered and chosen MIKEY protocol string. The responder does
not need to verify the other offered MIKEY strings. So there is only one
MIKEY instantiation instead of three. Wouldn't that be worth it?
 >
 > In the example above where the responder receives an attacked offer
of (DHSIGN, PK; PW) and PW is not supported or implemented, the PW
option can instantly be sorted out and would not be processed any
further. This


Yes, it can also be sorted out by the simple fact that it's not
implemented. You seem to suggest a method that can detect certain
bidding down attacks in the case when there is a bid-down to something
that would anyway never have been chosen by the Responder, becuase he
does not support it. We do not consider this an attack.

This argument can be turned around, and in doing so, it serves as
a counterexample, showing that extending the authenticated info
to also include teh "variant" does not help:

Suppose the Responder only supports only PW or only PK and that a
(PK, PS) offer is replaced (by an attacker) into (PK, PW). Also
assume that PK is very strong but PW is weak. The
original authenticator in the PK-line would say "MIKEY-PK MIKEY-PS".
Now, considering that the attacker managed
to fake the PW line, it is likely that it can be made to say
"MIKEY-PK MIKEY-PW". Now, a Responder that implements only MIKEY-PK,
will detect: "aha, the PS offer has been replaced by a PW offer". But,
consider now an attack towards a Responder who only supports PW.
In addition to faking the authenticator of the PW part (to say
"MIKEY-PK MIKEY-PW", as would be expected), the attacker modifies
the PK offer's authenticator to say "MIKEY-PK MIKEY-PW" too. Now,
he can't fake the strong signature on this, but he doesn't care
since he (correctly) hopes the Responder will not support PK.
When the Respoder receives this offer, he will think all is fine.
Had he implemented (and used) also the PK variant he would have seen
the attack on the PK line, since teh signature wouild be wrong,
but now it goes by unnoticed.

In summary, to *detect* this attack, the Responder needs to
implement (and use) at least one of the "secure" variants. In
the worst case, this could be *all* variants (if we are really
paranoid). Therefore, adding
the the "type" of MIKEY-offer to the authenticated info does not
remove the need to verify (possibly) all offers. Note though that
(again) we argue that the real issue is an insecure MIKEY
variant (PW) that could be used on it's own to cause harm, and
therefore these bidding-down attacks are purely artificial
examples that do not show the real problem.



 > would even improve resistance against certain DoS attacks, but that's a
 > different issue not of my concern here.
 >
 > 9) I wouldn't have argued so long, without offering a solution that
 > (hopefully) would work roughly as follows:
 >
 > - each offered MIKEY protocol option is identified by a specific KMID.
 > One for MIKEY-PS, a different KMID for MIKEY-DH etc
 >
 > - we take advantage of the fact that MIKEY strongly integrity protects
 > the list of protocol IDs end-to-end; so the KMIDs are indeed 
authenticated.
 >
 > - The responder can detect
 > a) an injected additional KMID offer or
 > b) an offer that is replaced by new KMID (and string) which is not
 > supported
 > simply as the KMID doesn't show up within the protected protocol list.
 >
 > Thus, the responder does not require implementation of all options to
 > thwart bidding-down attacks. The responder can reject offers that he/she
 > hasn't implemented or for some policy reason, doesn't want to support.
 >
 > 10) Isn't it obvious that allocation of a few KMIDs basically and a few
 > checks upon them by the responder is a much cheaper mechanism to satisfy
 > above requirement than brute-force verification of each and every
 > received option by the receiver?
 >

In summary, your main concern seem to boil down into the scenario
where you try to negotiate different MIKEY types (DH, PK, PS) at
the same time. The key management negotiation was intended for
negotiation of different key management protocols and not
negotiation of one and the same protocol. This is a position we
still hold. Therefore, we do not see the need for a specific KMID
indicating key management protocol's specific offer types.

Best,

/Mats and MIKEY team



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Subject: Re: [MMUSIC] Confusion regarding ICE username fragment
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inline.

Thomas Zeng wrote:


>>If you don't do this, you run into a nasty problem which is 
>>there is no way to reliably know that your peer has suceeded 
>>in its connectivity checks. That is needed to allow you to 
>>turn off the stun server and accept only media packets. This 
>>also simplified the protocol in several places.
> 
> 
> I'd like to point out that there are use cases for keeping STUN server
> running on the chosen endpoint (i.e., <address, port>) even after
> connectivity checks have succeeded. 
> 
> For RTSP where media packets flow one way from the RTSP server to its
> client, it is useful to keep periodic STUN exchanges to keep alive the
> NAT bindings. It is known in the real world that cerntain NAT bindings
> will be deleted if there is no traffic going a specific direction
> (normally from outside in, if I remember correctly).

Right. ICE does currently say that this is an application problem, and 
in this context, it would mean using things like comfort noise of 
perhaps the rtp no-op format from the client to the RTSP server. Have 
you considered that?

If we want to use STUN for keepalives, the problem you run into is how 
the client knows whether the stun server is still running or not. If its 
not, the client shouldnt send the STUn packets; if it is, it can. The 
-03 draft solves this by using the transmission and receipt of RTP 
packets onto that address/port as a means for synchronizing this 
transition. If you leave it flexible as to whether it stays running or 
not, now you need to signal and syncrhonize on it, adding additional 
protocol options (a bad thing in general) and complexity.

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

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To: Awais Ali <aali@i2telecom.com>
Subject: Re: [MMUSIC] updated ICE spec
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inline.

Awais Ali wrote:
> Jonathan Rosenberg wrote:
> 
>>
>> * previously, a client obtained local transport addresses and then
>> both used those in its SDP, in addition to deriving perhaps several
>> derived transport addresses from it. No longer. No, there is a single
>> local transport address for each derived transport address, and that
>> local transport address is never advertised in ICE. This turned out to
>> be needed to figure out whether the local stun service could be turned
>> off when an RTP packet is received. Unfortunately it also required me
>> to introduce more terminology to describe.
>>
> If I understand correctly to fulfill this requirement a device that uses 
> both TURN and STUN and is starting an RTP/RTCP session will open six 
> sockets(3 for RTP and 3 for RTCP) and run a STUN server on each of them

Yes.

> 
> The advantage of this approach as you mentioned is that the STUN service 
> can be stopped when an RTP packet is received. 

Right. Let me explain why you need this.

Lets say I have a local socket S1 listening for UDP packets. I use STUN 
and TURN to derive two additional address/port pairs for that socket. I 
send them out in an invite. THere is one stun server listening on S1. My 
peer in the call will perform connectivity checks to *all three* 
address/port pairs, all three of which will be stun requests against the 
single server on S1. Now, S1 gets an RTP packet. Presumably, this is 
because at least one connectivity check has succeeded for the peer. 
However, it has three connectivity checks to do, all three of which are 
against the same socket, all three of whicih, if successful, could 
result in an RTP packet being received on that socket. Unfortunately, I 
cannot tell from the RTP packet alone *which* of these three checks has 
succeeded and thus generated an RTP packet. As a result, there is no way 
for me to know if I may still get valid STUn requests which need to be 
checked in order for connectivity checks to continue.

With TURN, you can sort-of guess by looking at the source address of the 
packet, and if that is the TURN server, you know that the RTP was sent 
to the TURN derived address. But, with STUN derived addresses this won't 
work.

By maintaining a single socket, single server, single username, single 
password, single advertised address model, this problem went away 
(receipt of an RTP packet on a socket means that connectivity was 
established to the one transport address advertised for that scoket). It 
also simplified other parts of the protocol.

Now, I well recognize that there is a cost to this in terms of socket 
resources. I concluded that in practice it didnt look like an issue, but 
it is something to take note of, and I welcome broader feedback on this 
point.


What will be the downside
> of keeping the STUN service running?

If you keep it running, you need to specify what happens when these 
checks fail, specify the periodicity of the checks, worry about 
congestion control for them. Its interesting to note that these are very 
much application layer (i.e., rtp and payload format) kinds of issues 
generally. Furthermore, folks have complained about the difficultty in 
demuxing RTP/STUN perpetually. With this approach, a socket starts stun 
only but waits for an RTP packet, and when one arrives, the socket 
becomes RTP only. There were some on the list who favored this.

If you ake it optional about whether it runs or not, you need to solve 
some negotiation problems as I described in another note.

-Jonathan R.


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

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The -03 ICE draft in Figure 2, shows the Accept message occuring before the 
STUN checks are performed. I believe that this is because the STUN checks 
can't be performed until both the Initiator and Responder have created their 
connection tables. However the Accept message seems to imply that the user 
has accepted the call. The media won't flow until the after the connectivity 
checks has selected a usable transport address. This gives me concern in that 
normally when a user Accepts a call, the user speaks a greeting (ie. 
"Hello"). The greeting could be dropped because media from the Responder 
hasn't begun yet when it is spoken.

Might I suggest that the Accept message shown as step (4) be replaced with a 
Provisional message (which for SIP could map to a 183 Early Media response). 
The connectivity checks could then be done before the user Accepts the call.

Jon

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Jonathan Rosenberg wrote:
> Paul,
> 
> I've come around to this line of thinking too, and this is what is 
> reflected in -03, now available at:
> 
> http://www.jdrosen.net/papers/draft-ietf-mmusic-ice-03.txt
> 
> more inline.
> 

[snip]

>>
>> For two way RTP based audio media this may, as Jon suggests, only 
>> require checking that packets are received periodically, and helping 
>> the other end by always sending packets periodically. For one way or 
>> idle media this may be a bit more tricky. Sending STUN packets is 
>> perhaps one answer, but maybe it is ok to send rtp idle packets (did 
>> those get approved?) if it is agreed they can be sent even on a 
>> RECVONLY or INACTIVE connection.
> 
> 
> If you want to monitor quality, I think RTCP does the job. I don't think 
> you need to send additional packets to generate data to measure. In any 
> case, determining that there has been a quality failure is definitely 
> outside the scope of ICE.

Using RTCP to monitor the quality of the current best connection is OK, but 
what about the other routes? The advantage of the connectivity checks with 
STUN, is that it is possible to use them to measure dynamically the 
latency/packet lost in the middle of a media session, and so to eventually 
choose a different route. I understand that this is outside ICE scope, but 
having this connectivite checks had some interesting usages, and I do not 
see by what it is replaced.

> 
> In -03, STUN is now only used for the initial checks. mid-call 
> keepalives are handled at the application layer, and there is text on 
> doing that for RTP, which talks about this music-on-hold problem. That 
> is differnt still from detecting failed media to do something about it, 
> which is not addressed at all, beyond a hook I left.
> 
> The hook that's in there is the treatment of re-invites. If you get one, 
> and nothing has changed, don't do anything. That is, a re-invite in and 
> of itself does not trigger you to restart connectivity checks. But, if 
> the re-invite adds a new candidate address that hasn't been seen before, 
> connecitivyt checks are sent to it. "never seen before" is determined 
> based on boht the transport address AND the username fragment. This 
> means that a client can use a transport address, and then if it thinks 
> there are problems, start up the stun server again, change the username, 
> and do a reinvite to trigger a recheck.
> 
> 
>>
> -Jonathan R.


-- 
Marc Petit-Huguenin
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Work: marc@8x8.com

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Jonathan Rosenberg wrote:

>>
>> The advantage of this approach as you mentioned is that the STUN 
>> service can be stopped when an RTP packet is received. 
>
>
> Right. Let me explain why you need this.
>
> Lets say I have a local socket S1 listening for UDP packets. I use 
> STUN and TURN to derive two additional address/port pairs for that 
> socket. I send them out in an invite. THere is one stun server 
> listening on S1. My peer in the call will perform connectivity checks 
> to *all three* address/port pairs, all three of which will be stun 
> requests against the single server on S1. Now, S1 gets an RTP packet. 
> Presumably, this is because at least one connectivity check has 
> succeeded for the peer. However, it has three connectivity checks to 
> do, all three of which are against the same socket, all three of 
> whicih, if successful, could result in an RTP packet being received on 
> that socket. Unfortunately, I cannot tell from the RTP packet alone 
> *which* of these three checks has succeeded and thus generated an RTP 
> packet. As a result, there is no way for me to know if I may still get 
> valid STUn requests which need to be checked in order for connectivity 
> checks to continue.
>
Ok I understand that it is not possible to determine from the RTP packet 
which connection was used. Why can we not use the Giveup timer to 
determine when the client has stopped its connectivity checks?. If a 
maximum Giveup-time is specified, a peer can stop its STUN server, 
Giveup-time after receiving the first RTP packet.

Ali


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From mmusic-bounces@ietf.org  Wed Oct 27 20:09:56 2004
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From: Awais Ali <aali@i2telecom.com>
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Subject: [MMUSIC] Clarification about ICE Management of Resources
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I would like to verify if my understanding of the Management of 
Resources section in the ICE draft is correct.

Let us assume device A and device B want to establish an RTP/RTCP 
session. A and B have 3 addresses each, as follows:

AL: local address of A
AS: Stun derived address for A
AT: Turn derived address for A

BL: local address of B
BS: Stun derived address for B
BT: Turn derived address for B

After A completes its connectivity checks, A's connection table for RTP 
will look like this

1. AS     BS   ... ... ... 0.2    GOOD
2. AS     BT   ... ... ... 0.1    GOOD
3. AS     BL   ... ... ... 0.4    BAD
4. AT     BS   ... ... ... 0.2    GOOD
5. AT     BT   ... ... ... 0.1    GOOD
6. AT     BL   ... ... ... 0.4    BAD
7. AL     BS   ... ... ... 0.2    GOOD
8. AL     BT   ... ... ... 0.1    GOOD
9. AL     BL   ... ... ... 0.4    BAD

Lets say A selects Line 7 to send RTP packets, then A will keep the 
other five connections open by sending keep-alive packets on them. 
Similarly there will be 5 RTCP channels. So in total A has to keep-alive 
10 connections.

B will also have a simillar connnection table. If B also selects the BL 
AS line for sending Media packets, then in the above exampe, we will 
have the situation where A is sending from a different port than it is 
receiving on. In this case then A must send keep alive packets from AS 
to keep the NAT binding open, even if it decides not to keep the other 
connections as backup.


Thanks
Ali








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Subject: RE: [MMUSIC] ICE Issue: mid-session connectivity checks
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> Using RTCP to monitor the quality of the current best connection is
> OK, but what about the other routes?  The advantage of the
> connectivity checks with STUN, is that it is possible to use them to
> measure dynamically the latency/packet lost in the middle of a media
> session, and so to eventually choose a different route.  I understand
> that this is outside ICE scope, but having this connectivite checks
> had some interesting usages, and I do not see by what it is replaced.

An endpoint is still free to probe other paths if it wants to do so.

Changing the path during an active call results in out-of-order packet
delivery, the possibility of lost packets, and a high likelyhood that
the delay of the new path will be different than the delay of the
old path.  In may situations, these changes cause an adverse effect
on realtime, interactive media (audio or video).  As networks are
designed to reduce these very same adverse effects, it's not desirable
for a NAT optimization technique to continually probe and potentially
switch media paths during an active media session.

-d


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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 Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: Interactive Connectivity Establishment (ICE): 
			  A Methodology for Network Address Translator (NAT) 
			  Traversal for Multimedia Session Establishment Protocols
	Author(s)	: J. Rosenberg
	Filename	: draft-ietf-mmusic-ice-03.txt
	Pages		: 45
	Date		: 2004-10-27
	
This document describes a methodology for Network Address Translator
   (NAT) traversal for multimedia session signaling protocols, such as
   the Session Initiation Protocol (SIP).  This methodology is called
   Interactive Connectivity Establishment (ICE).  ICE makes use of
   existing protocols, such as Simple Traversal of UDP Through NAT
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	Title		: Real Time Streaming Protocol (RTSP)
	Author(s)	: H. Schulzrinne, et al.
	Filename	: draft-ietf-mmusic-rfc2326bis-08.txt
	Pages		: 176
	Date		: 2004-10-27
	
This memorandum is a revision of RFC 2326, which is currently a
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   The Real Time Streaming Protocol, or RTSP, is an application-level
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: SDP: Session Description Protocol
	Author(s)	: M. Handley, et al.
	Filename	: draft-ietf-mmusic-sdp-new-21.txt
	Pages		: 46
	Date		: 2004-10-27
	
This memo defines the Session Description Protocol (SDP). SDP is
intended for describing multimedia sessions for the purposes of
session announcement, session invitation, and other forms of
multimedia session initiation.

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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: The Alternative Network Address Types Semantics (ANAT)
			  for theSession Description Protocol (SDP) Grouping
			  Framework
	Author(s)	: G. Camarillo, J. Rosenberg
	Filename	: draft-ietf-mmusic-anat-02.txt
	Pages		: 8
	Date		: 2004-10-27
	
This document defines the Alternative Network Address Types (ANAT)
semantics for the SDP grouping framework. The ANAT semantics allow
offering alternative types of network addresses to establish a
particular media stream.

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Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-21.txt
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On 28 Oct 2004, at 16:02, Internet-Drafts@ietf.org wrote:
> A New Internet-Draft is available from the on-line Internet-Drafts 
> directories.
> This draft is a work item of the Multiparty Multimedia Session Control 
> Working Group of the IETF.
>
> 	Title		: SDP: Session Description Protocol
> 	Author(s)	: M. Handley, et al.
> 	Filename	: draft-ietf-mmusic-sdp-new-21.txt
> 	Pages		: 46
> 	Date		: 2004-10-27
> 	
> This memo defines the Session Description Protocol (SDP). SDP is
> intended for describing multimedia sessions for the purposes of
> session announcement, session invitation, and other forms of
> multimedia session initiation.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-new-21.txt

Changes in this version are:
- In section 4.1, clarify that the transport port for unicast IP 
sessions
   is the remote transport port.
- In section 5.1, clarify that the media description is optional.
- In section 5.7, clarify that "this memo only defines IP4 and IP6", 
rather
   than "currently only IP4 and IP6 are defined".
- In section 5.7, note that TTL scoping is NOT RECOMMENDED for 
multicast,
   and one should use admin-scoped addresses instead.
- Rename section 6 from "Suggested Attributes" to "SDP Attributes"
- In section 5.8, clarify that the bandwidth is in kilobits per second
   unless otherwise specified in the definition of a new <bwtype> 
modifier.
- In section 5.9, clarify that the end time SHOULD be specified unless a
   session is unbounded.
- In section 5.12, remove example of SIP-over-TLS, since that's not an 
end
   to end connection.
- In section 5.13, remove example of use of attributes.
- In section 5.14, clarify behaviour when two RTP sessions share the 
same
   transport address.
- Section 5.14 was missing a reference to the "application" top-level 
MIME
   type for the "udp" proto.
- In section 6, generalise "a=orient" slightly. Remove references to the
   "wb" application, which is of historical interest only.
- In section 6, specify that "a=maxptime" is an integer multiple of the
   frame size for frame based codecs
- In section 7, reference SIP and offer/answer security considerations.
- In section 7, note that firewall holes should only be opened when the
   user can be appropriately authenticated and is authorized to do so.
- In ABNF: a comment suggested that UDP ports should either be 0 or in
   the range 1024-65535. This conflicts with existing usage (e.g. port 9 
is
   used by comedia) and wasn't mandated by the text. Remove the comment, 
to
   allow any UDP port to be used.
- In ABNF: allow DNS names for multicast groups
- In ABNF: a comment in the definition of "sess-version" noted that 0
   indicates a new session. This conflicted with section 5.2, which said
   that an NTP-format timestamp SHOULD be used. Accordingly, the comment
   has been removed.
- In ABNF: allow 127.0.0.1 as a valid IPv4 address

Colin


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Index: sdp.xml
===================================================================
RCS file: /Users/csp/CVS/sdp/sdp.xml,v
retrieving revision 1.37
retrieving revision 1.66
diff -u -r1.37 -r1.66
--- sdp.xml	2004/09/19 09:02:11	1.37
+++ sdp.xml	2004/10/25 12:36:10	1.66
@@ -6,7 +6,7 @@
 <?rfc subcompact="no"?>
 <!DOCTYPE rfc SYSTEM "rfc2629.dtd">
 <rfc obsoletes="2327, 3266" category="std" ipr="full3667"
-     docName="draft-ietf-mmusic-sdp-new-20.txt">
+     docName="draft-ietf-mmusic-sdp-new-21.txt">
 <front>
 <title abbrev="SDP">SDP: Session Description Protocol</title>
   <author initials="M." surname="Handley" fullname="Mark Handley">
@@ -48,7 +48,7 @@
       <email>csp@csperkins.org</email>
     </address>
   </author>
-  <date day="19" month="September" year="2004" />
+  <date day="25" month="October" year="2004" />
   <abstract>
   <t> This memo defines the Session Description Protocol (SDP).  SDP is
       intended for describing multimedia sessions for the purposes of
@@ -228,7 +228,7 @@
     <t> For unicast IP sessions, the following are conveyed:</t>
     <t> <list style="symbols">
         <t>  The remote address for media</t>
-        <t>  The transport port for media</t>
+        <t>  The remote transport port for media</t>
         </list></t>
 
     <t> The semantics of this address and port depend on the media and
@@ -351,7 +351,7 @@
 <artwork>
    Session description
       v=  (protocol version)
-      o=  (owner/creator and session identifier).
+      o=  (owner/creator and session identifier)
       s=  (session name)
       i=* (session information)
       u=* (URI of description)
@@ -370,7 +370,7 @@
       t=  (time the session is active)
       r=* (zero or more repeat times)
 
-   Media description
+   Media description, if present
       m=  (media name and transport address)
       i=* (media title)
       c=* (connection information - optional if included at
@@ -415,9 +415,8 @@
    t=2873397496 2873404696
    a=recvonly
    m=audio 49170 RTP/AVP 0
-   m=video 51372 RTP/AVP 31
-   m=application 32416 udp wb
-   a=orient:portrait
+   m=video 51372 RTP/AVP 99
+   a=rtpmap:99 h263-1998/90000
 </artwork>
 </figure>
 <t> Text fields such as the session name and information are octet
@@ -525,7 +524,7 @@
     </figure>
 
     <t> The "i=" field provides textual information about the session.
-	There may be at most one session-level "i=" field per session
+	There MUST be at most one session-level "i=" field per session
 	description, and at most one "i=" field per media.  If the
 	"a=charset" attribute is present, it specifies the character
 	set used in the "i=" field.  If the "a=charset" attribute is
@@ -554,7 +553,7 @@
     <t> A URI is a Universal Resource Identifier as used by WWW clients
         <xref target="RFC2396"/>, <xref target="RFC2732"/>.
         The URI should be a pointer to additional information about the
-        conference.  This field is OPTIONAL, but if it is present it MUST
+        session.  This field is OPTIONAL, but if it is present it MUST
         be specified before the first media field.  No more than one URI
         field is allowed per session description.</t>
     </section>
@@ -640,7 +639,7 @@
 
     <t> The second sub-field ("&lt;addrtype>") is the address type.
         This allows SDP to be used for sessions that are not IP based.
-	Currently only IP4 and IP6 are defined, but other values MAY be
+	This memo only defines IP4 and IP6, but other values MAY be
 	registered in the future (see Section 8).</t>
 
     <t> The third sub-field ("&lt;connection-address>") is the connection
@@ -660,12 +659,14 @@
             description that is communicated by a multicast announcement,
             though this is not prohibited.</t>
 
-        <t> Conferences using an IPv4 multicast connection address MUST
+        <t> Sessions using an IPv4 multicast connection address MUST
             also have a time to live (TTL) value present in addition to
             the multicast address.  The TTL and the address together
             define the scope with which multicast packets sent in this
             conference will be sent.  TTL values MUST be in the range
-            0-255.</t>
+            0-255. While the TTL MUST be specified, its use to scope 
+	    multicast traffic is deprecated; applications SHOULD use 
+	    an adminstratively scoped address instead.</t>
         </list></t>
 
     <t> The TTL for the session is appended to the address using a slash
@@ -768,7 +769,8 @@
 	If the bandwidth of a session or media in a session is different
 	from the bandwidth implicit from the scope, a "b=CT:..." line
 	SHOULD be supplied for the session giving the proposed upper limit
-	to the bandwidth used.  The primary purpose of this is to give an
+	to the bandwidth used (the "conference total" bandwidth).  The 
+	primary purpose of this is to give an
 	approximate idea as to whether two or more sessions can co-exist
 	simultaneously.  When using the CT modifier with RTP, if several RTP
 	sessions are part of the conference, the conference total refers to
@@ -804,9 +806,10 @@
         given, they are recommended to be short.</t>
     </section>
 
-    <t> The &lt;bandwidth> is in kilobits per second by default.
-        Modifiers MAY specify that alternative units are to be used (the
-        modifiers defined in this memo use the default units).</t>
+    <t> The &lt;bandwidth> is interpreted as kilobits per second by default.
+        The definition of a new &lt;bwtype> modifier MAY specify that the 
+	bandwidth is to be interpreted in some alternative unit (the "CT" 
+	and "AS" modifiers defined in this memo use the default units).</t>
 
     <section title="Timing (&quot;t=&quot;)">
     <figure>
@@ -850,7 +853,7 @@
         sessions that have not timed out to the user, that an unbounded
         session will only be active until half an hour from the current
         time or the session start time, whichever is the later.  If
-        behaviour other than this is required, an end-time should be given
+        behaviour other than this is required, an end-time SHOULD be given
         and modified as appropriate when new information becomes available
         about when the session should really end.</t>
 
@@ -1039,7 +1042,7 @@
     <t> The key field MUST NOT be used unless it can be guaranteed that
         the SDP is conveyed over a secure and trusted channel.  An example
         of such a channel might be SDP embedded inside an S/MIME message
-	or a TLS protected HTTP or SIP session.  It is important to ensure
+	or a TLS-protected HTTP session.  It is important to ensure
 	that the secure channel is with the party that is authorized to
 	join the session, not an intermediary: if a caching proxy server is
 	used, it is important to ensure that the proxy is either trusted or
@@ -1066,8 +1069,7 @@
         stream.  Attribute fields can also be added before the first media
         field; these "session-level" attributes convey additional
         information that applies to the conference as a whole rather than
-        to individual media; an example might be the conference's floor
-        control policy.</t>
+        to individual media.</t>
 
     <t> Attribute fields may be of two forms:</t>
 
@@ -1096,7 +1098,7 @@
         values are NOT normally affected by the "charset" attribute as
         this would make comparisons against known values problematic.
         However, when an attribute is defined, it can be defined to be
-        charset-dependent, in which case it's value should be interpreted
+        charset-dependent, in which case its value should be interpreted
         in the session charset rather than in ISO-10646.</t>
 
     <t> Attributes MUST be registered with IANA (see Section 8).  If an
@@ -1178,6 +1180,36 @@
         <t> would imply that address 224.2.1.1 is used with ports 49170 and
             49171, and address 224.2.1.2 is used with ports 49172 and 49173.</t>
 
+	<t> The combination of ports specified in "m=" lines and IP addresses
+	    specified in "c=" lines MUST comply with the following rules for
+	    RTP-based media streams (other protocols SHOULD define similar
+	    rules):</t>
+
+        <t> <list style="numbers">
+	    <t> If two media sessions have the same transport address (i.e.
+		identical IP address and port numbers), the associated
+		payload types (e.g. given in the "a=rtpmap:" attribute) 
+		MUST NOT be in conflict, i.e. the same payload type MUST
+		NOT be mapped to different media types.</t>
+
+	    <t> If two media sessions have the same transport address, they
+	        MUST use compatible media (e.g. both audio or both video).</t>
+
+ 	    <t> If two media sessions have the same transport address, they
+	        SHOULD operate under the same RTP profile. The sessions MAY
+		use two different RTP profiles only if those profiles are 
+		specifically designed to be compatible.</t>
+
+	    <t> If two media sessions have the same RTP transport address,
+	        they MUST also use the same RTCP address and vice versa.</t>
+	</list></t>
+
+	<t> Two media sessions with the same transport address indicate
+	    alternatives for the same media stream, i.e. all profiles,
+	    media types, and payload types provided in any of the "m=" 
+	    lines are valid.</t>
+
+
         <t hangText="&lt;proto>"> is the transport protocol. The meaning of
             the transport protocol is dependent on the address type field in 
             the relevant "c=" field.  Thus a "c=" field of IP4 indicates that 
@@ -1201,7 +1233,7 @@
             the media format is that the same standard media formats may be
             carried over different transport protocols even when the network
             protocol is the same - a historical example is vat PCM audio and
-            RTP PCM audio.  In addition, relays and monitoring tools that are
+            RTP PCM audio, another might be TCP/RTP PCM audio.  In addition, relays and monitoring tools that are
             transport-protocol-specific but format-independent are possible.</t>
 
 	<t hangText="&lt;fmt>"> is a media format description.  The fourth
@@ -1222,9 +1254,9 @@
 
         <t> If the &lt;proto> sub-field is "udp" the &lt;fmt> sub-fields
             MUST reference a media type describing the format under the 
-            "audio", "text" and "video" top-level MIME types. The media
-            type registration SHOULD define the packetization format for
-            use with UDP transport. </t>
+            "audio", "video", "text" and "application" top-level MIME types. 
+	    The media type registration SHOULD define the packetization 
+	    format for use with UDP transport. </t>
 
         <t> For media using other transport protocols, the &lt;fmt> field is 
             protocol specific.  Rules for interpretation of the &lt;fmt> 
@@ -1235,7 +1267,7 @@
     </section>
 </section>
 <!---------------------------------------------------------------------------->
-<section title="Suggested Attributes">
+<section title="SDP Attributes">
 
 <t> The following attributes are defined.  Since application
     writers may add new attributes as they are required, this list
@@ -1283,15 +1315,14 @@
       The maximum amount of media which can be encapsulated in each
       packet, expressed as time in milliseconds.  The time SHALL be
       calculated as the sum of the time the media present in the
-      packet represents.  The time SHOULD be a multiple of the frame
-      size.  This attribute is probably only meaningful for audio
-      data, but may be used with other media types if it makes
-      sense.  It is a media attribute, and is not dependent on
-      charset.  Note that this attribute was introduced after RFC
-      2327, and non updated implementations will ignore this
-      attribute.
+      packet represents.  For frame based codecs, the time SHOULD 
+      be an integer multiple of the frame size.  This attribute is 
+      probably only meaningful for audio data, but may be used with 
+      other media types if it makes sense.  It is a media attribute, 
+      and is not dependent on charset.  Note that this attribute was 
+      introduced after RFC 2327, and non updated implementations will 
+      ignore this attribute.
 
-
    a=rtpmap:&lt;payload type> &lt;encoding name>/&lt;clock rate>
      [/&lt;encoding parameters>]
 
@@ -1382,10 +1413,10 @@
       This specifies that the tools should be started in send-only
       mode.  An example may be where a different unicast address is
       to be used for a traffic destination than for a traffic
-      source.  In such a case, two media descriptions may be use,
+      source.  In such a case, two media descriptions may be used,
       one sendonly and one recvonly.  It can be either a session or
       media attribute, but would normally only be used as a media
-      attribute, and is not dependent on charset.  Note that sendonly
+      attribute. It is not dependent on charset.  Note that sendonly
       applies only to the media, and any associated control protocol
       (e.g. RTCP) SHOULD still be received and processed as normal.
 
@@ -1398,13 +1429,13 @@
       still send RTCP, even if started inactive.  It can be either a
       session or media attribute, and is not dependent on charset.
 
-   a=orient:&lt;whiteboard orientation>
+   a=orient:&lt;orientation>
 
-      Normally this is only used in a whiteboard media specification.
-      It specifies the orientation of a the whiteboard on the screen.
-      It is a media attribute.  Permitted values are "portrait",
-      "landscape" and "seascape" (upside down landscape).  It is not
-      dependent on charset.
+      Normally this is only used for a whiteboard or presentation 
+      tool.  It specifies the orientation of a the workspace on 
+      the screen.  It is a media attribute.  Permitted values are
+      "portrait", "landscape" and "seascape" (upside down landscape).
+      It is not dependent on charset.
 
    a=type:&lt;conference type>
 
@@ -1546,6 +1577,11 @@
 <!---------------------------------------------------------------------------->
 <section title="Security Considerations">
 
+<t> SDP is frequently used with the <xref target="RFC3261">Session Initiation
+    Protocol</xref> using the <xref target="RFC3264">offer/answer model</xref>
+    to agree parameters for unicast sessions. When used in this manner, the
+    security considerations of those protocols apply.</t>
+
 <t> SDP is a session description format that describes multimedia
     sessions.  A session description SHOULD NOT be trusted unless it has
     been obtained by an authenticated transport protocol from a trusted
@@ -1592,14 +1628,12 @@
 
 <t> Session descriptions may be parsed at intermediate systems such as
     firewalls for the purposes of opening a hole in the firewall to allow
-    the participation in multimedia sessions.  It is considered
-    inappropriate for a firewall to open such holes for unicast data
-    streams unless the session description comes in a request from inside
-    the firewall.  For multicast sessions, it is likely that local
-    administrators will apply their own policies, but the exclusive use
-    of "local" or "site-local" administrative scope within the firewall
-    and the refusal of the firewall to open a hole for such scopes will
-    provide separation of global multicast sessions from local ones.</t>
+    participation in multimedia sessions.  This SHOULD NOT be done unless
+    the SDP is conveyed in a manner that allows proper authentication and
+    authorization checks to ensure that firewall holes are only opened in
+    accordance with applicable security policy.  SDP by itself does not
+    include sufficient information to enable these checks: they depend on
+    the encapsulating protocol (e.g. SIP or RTSP).</t>
 
 <t> Use of the "k=" field poses a significant security risk, since it
     conveys session encryption keys in the clear.  SDP MUST NOT be used
@@ -1969,7 +2003,6 @@
                          ;should be unique for this username/host
 
    sess-version =        1*DIGIT
-                         ;0 is a new session
 
    nettype =             token
                          ;typically "IN"
@@ -2055,15 +2088,12 @@
                          ;typically "RTP/AVP" or "udp"
 
    port =                1*DIGIT
-                         ;should be either "0" or in the range "1024"
-                         ;to "65535" inclusive for UDP based media
-                         ;(a value of "0" is used to signal special
-                         ;conditions in some uses of SDP)
 
    ; generic sub-rules: addressing
    unicast-address =     IP4-address / IP6-address / FQDN / extn-addr
 
-   multicast-address =   IP4-multicast / IP6-multicast / extn-addr
+   multicast-address =   IP4-multicast / IP6-multicast / FQDN 
+                         / extn-addr
 
    IP4-multicast =       m1 3( "." decimal-uchar )
                          "/" ttl [ "/" integer ]
@@ -2080,12 +2110,12 @@
 
    FQDN =                4*(alpha-numeric / "-" / ".")
                          ; fully qualified domain name as specified
-                         ; in RFC1035
+                         ; in RFC1035 (and updates)
 
-   IP4-address =         b1 3("." decimal-uchar) / "0.0.0.0"
+   IP4-address =         b1 3("." decimal-uchar) 
 
    b1 =                  decimal-uchar
-                         ; less than "224"; not "0" or "127"
+                         ; less than "224"
 
    ; The following is from RFC2373 Appendix B.  It is a direct copy.
    IP6-address =         hexpart [ ":" IP4-address ]

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From: =?EUC-KR?B?sejAurz3?= <eunah714@dreamwiz.com>
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I submitted a draft defines a set of SDP attributes 
for specifying and negotiating very basic conference membership policy.
XCON-WG is working on a specific conference policy, 
and those information can be exchanged by the specific protocol.
However, for users those who do not have such protocol, and
still want to set very basic policy of group session,
the SDP information with offer/answer model can be used.

The draft can be found at 
http://www.ietf.org/internet-drafts/draft-ekim-mmusic-sdp-membership-01.txt

I'll appreciate any comments and thoughts from other folks.

eunah






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<BR>I submitted a draft defines a set of SDP attributes=20
<BR>for specifying and negotiating very basic conference membership polic=
y.
<BR>XCON-WG is working on a specific conference policy,=20
<BR>and those information can be exchanged by the specific protocol.
<BR>However, for users those who do not have such protocol, and
<BR>still want to set very basic policy of group session,
<BR>the SDP information with offer/answer model can be used.
<BR>
<BR>The draft can be found at=20
<BR>http://www.ietf.org/internet-drafts/draft-ekim-mmusic-sdp-membership-=
01.txt
<BR>
<BR>I'll appreciate any comments and thoughts from other folks.
<BR>
<BR>eunah
<BR>
<BR>
<BR></BODY>
</HTML>

<BR>

<BR>
<BR>

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