Re: TSV-Dir Last Call review of draft-ietf-avt-rtp-cnames-02

2010-12-17 Thread Colin Perkins
rent versions of UUID that RFC 4122 covers:
> 
> + Version 1: generated from clock + MAC, or + MAC-format random number
> + Version 3: generated from namespace:name
> + Version 4: generated from "truly random or pseudo-random number"
> + Version 5: generated from hash of namespace:name
> 
> There needs to be some guidance because the use of the UUID Versions 3/5, 
> derived from namespace:name, has the same problem with non-uniqueness of 
> FQDNs as the RFC 3550 FQDN-based CNAMEs.  Perhaps this document should advise 
> against using 3/5.  In addition, right now the draft is silent on mitigating 
> identity exposure, but it could offer some guidance towards using the variant 
> of Version 1 that does not expose the MAC or towards using Version 4.

This guidance seems appropriate, thanks.

> 2) Short-term persistent CNAMEs  are allowed to use a pseudo-randomly 
> generated RFC 4941 IPv6 privacy address if the device is IPv6-capable, but 
> are required to exposetheir MACs if the device is IPv4-only.  Why not allow 
> IPv4-only endpoints to to generate pseudo-random MACs using the specification 
> of doing so provided for the Version 1 variant in RFC 4122?

That seems like a reasonable addition.

> 3) Per-session CNAMEs are generated using SHA1-HMAC over the concatenation of 
> the initially-used SSRC, the source and destination addresses and ports, and 
> "a randomly generated value".  RFC 4086 is given as a reference for the 
> randomly generated value.  I think an implementer would like to know how many 
> bytes of randomly generated value should be used; this doesn't come out 
> clearly from reading RFC 4086. Further, what are the requirements for the key 
> for HMAC here?  In reading RFC 2104, the reference for HMAC, I end up with 
> some ideas about the key length, but not about how long/where I can use the 
> same key or whether there are problematic keys - to mention two questions 
> that come to mind.  Keying would not normally be easy to under-specify this 
> way because a security application would call for some keying details, but in 
> this application there is no security use for the keys.  Either it would be 
> good to add some guidance or perhaps SHA1-HMAC could be replaced with an "inse
 cure hash."

We'll add some guidance in -03.

Thanks,
Colin

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Re: I-D Action:draft-zimmermann-avt-zrtp-20.txt

2010-05-14 Thread Colin Perkins

On 11 May 2010, at 22:15, internet-dra...@ietf.org wrote:
A New Internet-Draft is available from the on-line Internet-Drafts  
directories.


	Title   : ZRTP: Media Path Key Agreement for Unicast Secure  
RTP

Author(s)   : P. Zimmermann, et al.
Filename: draft-zimmermann-avt-zrtp-20.txt
Pages   : 111
Date: 2010-05-11

This document defines ZRTP, a protocol for media path Diffie-Hellman
exchange to agree on a session key and parameters for establishing
unicast Secure Real-time Transport Protocol (SRTP) sessions for VoIP
applications.  The ZRTP protocol is media path keying because it is
multiplexed on the same port as RTP and does not require support in
the signaling protocol.  ZRTP does not assume a Public Key
Infrastructure (PKI) or require the complexity of certificates in end
devices.  For the media session, ZRTP provides confidentiality,
protection against man-in-the-middle (MiTM) attacks, and, in cases
where the signaling protocol provides end-to-end integrity
protection, authentication.  ZRTP can utilize a Session Description
Protocol (SDP) attribute to provide discovery and authentication
through the signaling channel.  To provide best effort SRTP, ZRTP
utilizes normal RTP/AVP profiles.  ZRTP secures media sessions which
include a voice media stream, and can also secure media sessions
which do not include voice by using an optional digital signature.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-zimmermann-avt-zrtp-20.txt



This addresses some, but not all, of my comments on -17:

- “In terms of the RTP topologies defined in [RFC5117], ZRTP is  
designed for Point to Point topologies only” - if only Topo-Point-to- 
Point is supported, explicitly state that by shortcut name, otherwise  
also list the other topologies that are supported by shortcut name.


- “Other [RTP] profiles MAY also be used” - which ones? Need to be  
explicit, since not all conceivable RTP profiles would work with ZRTP.


- This version states that “SSRC collisions would be disruptive to  
ZRTP.  This may be avoided via the mechanisms in RFC 3550, Section 8.2  
[RFC3550]”. The mechanisms in section 8.2 of RFC 3550 explain how to  
resolve SSRC collisions, not how to avoid them, and say nothing about  
how to avoid disruption to ZRTP. Further clarification is needed - I’d  
guess that the right thing to do is to accept that SRRC collisions  
are  disruptive, and force a re-negotiation of the secure session. If  
so, the draft needs to state this explicitly.


- A ZRTP Error code of 0x91 “SSRC collision” is defined, but the text  
doesn’t mention how it’s used.


- The new discussion of timers for non-UDP transports in section 6 is  
much improved, but could still do with giving explicit recommendations  
for the lengthened timer intervals.


- The new text regarding the ZRTP header extension is better, but  
could still do with explicitly stating that not only does ZRTP not use  
the RFC 5285 mechanisms, it conflicts with them.


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Re: [AVT] Last Call: draft-zimmermann-avt-zrtp (ZRTP: Media Path Key Agreement for Secure RTP) to Informational RFC

2010-04-28 Thread Colin Perkins

Alan,

Thanks for your response - some comments inline.
Colin

On 23 Apr 2010, at 18:15, Alan Johnston wrote:


Colin,

Thank you for your detailed review of the draft.  See my comments  
below.


- Alan -

On 4/13/10 12:19 PM, Colin Perkins wrote:

On 17 Mar 2010, at 22:26, The IESG wrote:
The IESG has received a request from an individual submitter to  
consider the following document:


- 'ZRTP: Media Path Key Agreement for Secure RTP '
  as an Informational RFC

The IESG plans to make a decision in the next few weeks, and  
solicits final comments on this action.  Please send substantive  
comments to the ietf@ietf.org mailing lists by 2010-04-14.  
Exceptionally, comments may be sent to i...@ietf.org instead. In  
either case, please retain the beginning of the Subject line to  
allow automated sorting.



I've reviewed this draft, and have a number of comments. The main  
issue is the scope of the protocol: it claims to be "Media Path Key  
Agreement for Secure RTP", but may be better titled "Media Path Key  
Agreement for Unicast Voice Telephony Using Secure RTP". The  
majority of RTP sessions cannot be secured using ZRTP, but this  
draft is virtually silent on its limitations.


The draft does point out that it is for securing unicast RTP only.   
For

example, in Section 1:

"ZRTP is designed for unicast media sessions in which there is a voice
media stream."

Including unicast in the title does seem to be a reasonable thing to  
do

and we will do that.

However, scoping it to voice telephony is not accurate.  While the  
Short Authentication String is the preferred security mechanism, the  
protocol also works with digital signatures and can be authenticated  
using an integrity protected signaling channel. Neither of these  
methods require a voice channel.


Some details follow:


- RTP is explicitly a group communication protocol, that supports a  
wide range of topologies [RFC 5117], and a wide range of media  
types. ZRTP is defined for point-to-point audio calls or group  
audio conferences with a centralised conference bridge, and doesn't  
support the full range of RTP topologies or media formats. This is  
probably an acceptable limitation, but needs to be much more  
clearly documented than in the present draft.


We propose adding unicast to the title and having this as the first
sentence of the Abstract:

"  This document defines ZRTP, a protocol for media path Diffie- 
Hellman

  exchange to agree on a session key and parameters for establishing
  unicast Secure Real-time Transport Protocol (SRTP) sessions for VoIP
  applications."


That's fine, but I do think the draft should explicitly refer to RFC  
5117 and state which of the RTP topologies described therein are  
supported.


...
- Section 5 uses the SSRC to associate ZRTP messages with an RTP  
stream, but doesn't appear to consider the possibility of an SSRC  
collision [RFC 3550 section 8.2].


Sure - we should mention that if a VoIP call is forked to two  
endpoints, and media sessions are established with both, and they  
both choose the same 32 bit SSRC, then ZRTP will not be able to  
distinguish between them.


An SSRC collision is possible, if unlikely, for unicast sessions in  
the absence of forking. I believe the draft needs to explicitly  
address what should be done in the event of an SSRC collision  
(presumably you should re-negotiate the security, which should be  
acceptable since collisions are rare).


- RTP can run over non-UDP transports, such as TCP and DCCP.  
Section 6 of the draft defines retransmission timers for ZRTP that  
are reasonable for RTP/UDP/IP sessions, but not for RTP/TCP/IP or  
RTP/DCCP/IP sessions. The draft needs to discuss use of ZRTP over  
non-UDP transports.


If the media is transported end-to-end using a reliable transport,  
then

clearly ZRTP retransmissions are not needed.  However, there can be
cases involving relays where ZRTP could be sent over a reliable
transport part of the way, and an unreliable transport the rest of the
way.  We will add text recommending that if reliable transport is  
used,

then the retransmit timers should be lengthened.


It's not clear how such scenarios can be distinguished, but okay. Some  
guidance on the appropriate values for the lengthened timers would be  
helpful.


- Section 7.3 discusses relaying ZRTP through a PBX. There appears  
to be an assumption that the PBX is configured to provide point to  
multipoint service as an RTCP-Terminating MCU (Topo-RTCP- 
terminating-MCU in the terminology of RFC 5117). None of the other  
topologies defined in RFC 5117 are considered.


I couldn't find where you got this idea from.  A PBX is not going to  
act as an RTCP-Terminating MCU.  It will most likely act as a B2B  
for both the signaling and media, but it will still be 1:1.  We can  
clarify this. ZRTP does not apply to all the topologies of RFC 5117.



Re: Last Call: draft-zimmermann-avt-zrtp (ZRTP: Media Path Key Agreement for Secure RTP) to Informational RFC

2010-04-13 Thread Colin Perkins

On 17 Mar 2010, at 22:26, The IESG wrote:
The IESG has received a request from an individual submitter to  
consider

the following document:

- 'ZRTP: Media Path Key Agreement for Secure RTP '
   as an Informational RFC

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send substantive comments to  
the

ietf@ietf.org mailing lists by 2010-04-14. Exceptionally,
comments may be sent to i...@ietf.org instead. In either case, please
retain the beginning of the Subject line to allow automated sorting.



I've reviewed this draft, and have a number of comments. The main  
issue is the scope of the protocol: it claims to be "Media Path Key  
Agreement for Secure RTP", but may be better titled "Media Path Key  
Agreement for Unicast Voice Telephony Using Secure RTP". The majority  
of RTP sessions cannot be secured using ZRTP, but this draft is  
virtually silent on its limitations. Some details follow:


- RTP is explicitly a group communication protocol, that supports a  
wide range of topologies [RFC 5117], and a wide range of media types.  
ZRTP is defined for point-to-point audio calls or group audio  
conferences with a centralised conference bridge, and doesn't support  
the full range of RTP topologies or media formats. This is probably an  
acceptable limitation, but needs to be much more clearly documented  
than in the present draft.


- The draft claims it works with the RTP/AVP profile (although this is  
misspelt AVP/RTP in section 3). This is only one of several RTP  
profiles that can be used, but nothing is said about the other RTP  
profiles, such as RTP/AVPF. To avoid confusion, the draft should  
explicitly state which of the other RTP profiles are supported and  
which are not.


- Section 5 uses the SSRC to associate ZRTP messages with an RTP  
stream, but doesn't appear to consider the possibility of an SSRC  
collision [RFC 3550 section 8.2].


- RTP can run over non-UDP transports, such as TCP and DCCP. Section 6  
of the draft defines retransmission timers for ZRTP that are  
reasonable for RTP/UDP/IP sessions, but not for RTP/TCP/IP or RTP/DCCP/ 
IP sessions. The draft needs to discuss use of ZRTP over non-UDP  
transports.


- Section 7.3 discusses relaying ZRTP through a PBX. There appears to  
be an assumption that the PBX is configured to provide point to  
multipoint service as an RTCP-Terminating MCU (Topo-RTCP-terminating- 
MCU in the terminology of RFC 5117). None of the other topologies  
defined in RFC 5117 are considered.


- The draft does not specify how traffic RTCP is secured. This might  
be done using the key negotiated on the media path, or it might  
require a separate ZRTP negotiation using multistream mode, but the  
draft doesn't specify.


- SDES is a commonly used abbreviation in the RTP community for RTCP  
Source Description packets, yet this draft uses it for other purposes.  
To avoid confusion, please expand the acronym (e.g., in Section 8.2).


- The recommendations in Section 8.3 to avoid VBR traffic with secure  
calls are at odds with the recent discussion in the AVT working group  
(IETF 76).


- Section 10 assumes that the presence of an RTP mixer can be  
determined from the signalling. This assumption is false in general,  
although it's true for some SIP-initiated calls.


- The RTP header extension for ZRTP (Section 12) conflicts with the  
mechanism defined in RFC 5285. At minimum, this conflict should be  
documented, and it might make sense to update this draft to use the  
RFC 5285 mechanism.


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



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TSV review of draft-ietf-sipping-rtcp-summary-08

2010-02-17 Thread Colin Perkins
I've reviewed this document as part of the transport area  
directorate's ongoing effort to review key IETF documents. These  
comments were written primarily for the transport area directors, but  
are copied to the document's authors for their information and to  
allow them to address any issues raised. The authors should consider  
this review together with any other last-call comments they receive.  
Please always CC tsv-...@ietf.org if you reply to or forward this  
review.


This document defines a SIP event package to enable reporting some  
RTCP XR metrics, in particular those deemed relevant to some classes  
of VoIP application.  The metrics reported are a subset of those  
defined in RFC 3611, plus some extensions. This is a potentially  
useful specification, but I believe there are a number of problems  
with the way it interacts with RTP which I recommend be addressed  
before publication.


The applicability statement (section 1.1) references the list of RTP  
topologies in RFC 5117, and states that only the first (point-to- 
point) sessions is relevant to this document. The Usage Scenarios  
(section 1.2) and the rest of the document contradict this, discussing  
point-to-point sessions, conference calls using a central conferencing  
server, and multicast VoIP calls using SSM. RTP is inherently a group  
communication protocol, and this document needs to properly consider  
the full range of topologies defined in RFC 5117.


A large number of parameters are defined in Section 4.6:

The PayloadType parameter states that "IANA registered values SHOULD  
be used where possible", but the IANA registry only applies to the RTP/ 
AVP profile, and has been closed to new registrations for several  
years now. The implication of this document is that the payload type  
is sufficient to identify the codec, but this is not the case, and for  
this reason RFC 3551 recommends the use of media subtype names instead.


The PayloadDesc parameter is defined to provide a text description for  
the codec. It is stated that "The abbreviated standard name for the  
codec SHOULD be used (for example "G.729A")". The IETF has a standard  
namespace for codecs, in the form of media subtypes, registered  
according to RFC 4855. I believe it would be a mistake to introduce  
another, ill-defined, namespace in this document, rather than using  
the standard media type names as used for all other RTP-related  
documents.


Several parameters are defined relating to frame-based codecs (e.g.  
FrameDuration, FrameOctets, and FramesPerPacket). However, not all RTP  
audio payload types are frame based (see RFC 3551, section 4.3). It is  
not clear how sample based codecs are handled.


The FmtpOptions parameter is defined to convey "FMTP options from  
SDP". These options are parameters of media subtypes, and don't make  
sense unless the PayloadDesc is specified as a media subtype (and  
don't make any sense if just PayloadType is specified, since they're  
related to a media subtype name, not a payload type).


The MetricType parameter is ill-defined. What do the values specified  
refer to, and how are new values to be registered?


The LocalAddr and RemoteAddr parameters assume that STUN is used as a  
NAT traversal mechanism, if one is needed. While I agree with the  
principle of reporting the address on the "outside" of the NAT, STUN  
is not always available.


Section 4.6.2.9.5 expresses interarrival jitter as defined in RFC 3550  
in milliseconds, but RFC 3550 specifies it in RTP timestamp units.


Nit: In the last sentence of the introduction, "Real-Time Control  
Protocol" should be "RTP Control Protocol".


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Re: [Fecframe] Last Call: draft-ietf-fecframe-dvb-al-fec (DVB-IPTVApplication-Layer Hybrid FEC Protection) to Informational RFC

2010-01-07 Thread Colin Perkins

Thanks. The updated draft looks good.
Colin


On 14 Dec 2009, at 20:55, Ali C. Begen (abegen) wrote:

Sounds like a good suggestion. We will make the text change after  
the LC ends.


Thanks.
-acbegen


-Original Message-
From: fecframe-boun...@ietf.org [mailto:fecframe-boun...@ietf.org]  
On Behalf Of Colin

Perkins
Sent: Monday, December 14, 2009 11:48 AM
To: fecfr...@ietf.org
Cc: IETF Discussion Mailing List
Subject: Re: [Fecframe] Last Call: draft-ietf-fecframe-dvb-al-fec  
(DVB-IPTVApplication-

Layer Hybrid FEC Protection) to Informational RFC

On 30 Nov 2009, at 15:35, The IESG wrote:

The IESG has received a request from the FEC Framework WG (fecframe)
to
consider the following document:

- 'DVB-IPTV Application-Layer Hybrid FEC Protection '
  as an Informational RFC

The IESG plans to make a decision in the next few weeks, and  
solicits

final comments on this action.  Please send substantive comments to
the
ietf@ietf.org mailing lists by 2009-12-14. Exceptionally,
comments may be sent to i...@ietf.org instead. In either case,  
please

retain the beginning of the Subject line to allow automated sorting.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-fecframe-dvb-al-fec-03.txt


I understand, from previous conversations with the author, that the
aim of this draft is "to explain how one can use IETF specs to
implement AL-FEC protocol". That's a fine goal, and this draft does a
reasonable job in providing such an explanation.

However, I believe - as I previously stated in the working group -
that the relationship between this draft and the DVB AL-FEC
specification is not clearly defined.  While the abstract is
reasonably clear on the relationship, the title and introduction
suggest that this draft is the DVB AL-FEC protocol, rather than being
implementation guidelines for the DVB AL-FEC protocol. One way to
address this might be to change the title of the draft to "Guidelines
for Implementing DVB-IPTV Application-Layer Hybrid FEC Protection",
and to add a few clarifying sentences to the introduction.

I have no problems with the technical content of the draft.

--
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http://csperkins.org/



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Re: [Fecframe] Last Call: draft-ietf-fecframe-dvb-al-fec (DVB-IPTV Application-Layer Hybrid FEC Protection) to Informational RFC

2009-12-14 Thread Colin Perkins

On 30 Nov 2009, at 15:35, The IESG wrote:
The IESG has received a request from the FEC Framework WG (fecframe)  
to

consider the following document:

- 'DVB-IPTV Application-Layer Hybrid FEC Protection '
   as an Informational RFC

The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send substantive comments to  
the

ietf@ietf.org mailing lists by 2009-12-14. Exceptionally,
comments may be sent to i...@ietf.org instead. In either case, please
retain the beginning of the Subject line to allow automated sorting.

The file can be obtained via
http://www.ietf.org/internet-drafts/draft-ietf-fecframe-dvb-al-fec-03.txt


I understand, from previous conversations with the author, that the  
aim of this draft is "to explain how one can use IETF specs to  
implement AL-FEC protocol". That's a fine goal, and this draft does a  
reasonable job in providing such an explanation.


However, I believe - as I previously stated in the working group -  
that the relationship between this draft and the DVB AL-FEC  
specification is not clearly defined.  While the abstract is  
reasonably clear on the relationship, the title and introduction  
suggest that this draft is the DVB AL-FEC protocol, rather than being  
implementation guidelines for the DVB AL-FEC protocol. One way to  
address this might be to change the title of the draft to "Guidelines  
for Implementing DVB-IPTV Application-Layer Hybrid FEC Protection",  
and to add a few clarifying sentences to the introduction.


I have no problems with the technical content of the draft.

--
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Re: XML2RFC must die, was: Re: Two different threads - IETF Document Format

2009-07-05 Thread Colin Perkins

On 5 Jul 2009, at 20:52, James M. Polk wrote:

At 09:38 AM 7/5/2009, Colin Perkins wrote:

On 5 Jul 2009, at 14:24, Iljitsch van Beijnum wrote:

My apologies for the subject line. I'm very disappointed that the
silent majority of draft authors isn't speaking up. I can't imagine
that the vast majority of draft authors has absolutely no problems
with XML2RFC. So I'm assuming they've been ignoring the thread,
hopefully the new subject line will get some of them to chime in. If
that doesn't happen I'll shut up and try to figure out why I have so
much trouble with something that nobody else finds difficult.


I have no significant problems using xml2rfc, and find it easier to
write Internet-Drafts using xml2rfc than I did using nroff, LaTeX, or
Microsoft Word.

I also appreciate the added consistency in Internet-Draft formatting
that has resulted since xml2rfc has been widely adopted. This makes  
it

a lot easier to print drafts, since they have consistent page sizes
and form-feed characters.


huh?

the current boilerplate has the first page being 54 lines long, and  
subsequent pages being 57 lines long, but with the "footer" on the  
56th line, and not the 57th.


The "footer" on the first page isn't on the last line of the page  
either.


This isn't very uniform...  perhaps unless you use some special  
viewer to recreate this non-uniform display consistently.


It doesn't greatly offend me that the first page is three lines  
shorter than the other pages.  Nor does it confuse enscript, on those  
occasions when I want to print an internet-draft or RFC.


This is very much unlike the situation pre-xml2rfc, when the page  
length and width varied considerably for some drafts, and where form- 
feeds were often inconsistently applied.


--
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Re: XML2RFC must die, was: Re: Two different threads - IETF Document Format

2009-07-05 Thread Colin Perkins

On 5 Jul 2009, at 14:24, Iljitsch van Beijnum wrote:
My apologies for the subject line. I'm very disappointed that the  
silent majority of draft authors isn't speaking up. I can't imagine  
that the vast majority of draft authors has absolutely no problems  
with XML2RFC. So I'm assuming they've been ignoring the thread,  
hopefully the new subject line will get some of them to chime in. If  
that doesn't happen I'll shut up and try to figure out why I have so  
much trouble with something that nobody else finds difficult.


I have no significant problems using xml2rfc, and find it easier to  
write Internet-Drafts using xml2rfc than I did using nroff, LaTeX, or  
Microsoft Word.


I also appreciate the added consistency in Internet-Draft formatting  
that has resulted since xml2rfc has been widely adopted. This makes it  
a lot easier to print drafts, since they have consistent page sizes  
and form-feed characters.


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



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tsv-dir review of draft-ietf-sip-outbound-17

2009-05-25 Thread Colin Perkins
I've reviewed this document as part of the transport area  
directorate's ongoing effort to review key IETF documents. These  
comments were written primarily for the transport area directors, but  
are copied to the document's authors for their information and to  
allow them to address any issues raised. The authors should consider  
this review together with any other last-call comments they receive.  
Please always CC tsv-...@ietf.org if you reply to or forward this  
review.


Overall, this draft is basically ready for publication, but has some  
minor nits that should be fixed before publication.


The initial paragraph of section 3.5 can be interpreted as implying  
that a single lost ping is sufficient to declare a UDP flow dead.  
Section 4.4.2 is clear that this is not the case, but to avoid  
confusion, it might be appropriate to add a caveat or forward  
reference to section 3.5.


Section 4.4 defines a keep-alive mechanism using CRLF sequences  
multiplexed with the SIP messages if connection-oriented transports  
are used, and using STUN requests sent on the same port when  
connectionless transports are used. This makes sense for the examples  
of TCP, SCTP, and UDP transports. There's no mention of DCCP, though,  
which has somewhat different properties to the other connection  
oriented transports, and doesn't obviously work using a CRLF keep  
alive (e.g. RFC 3261 section 7.1 requires implementations that use  
stream-oriented transports to ignore an extra CRLF, but not those  
using unreliable datagram protocols). Some clarification would be  
useful, even if just to state that operation over DCCP is for future  
study.


Section 4.4.1 specifies keep-alive timer intervals for connection- 
oriented sessions for both battery operated and mains powered devices,  
while section 4.4.2 only provides a single time out value for  
connectionless sessions. It would be useful to explicitly mention the  
impact on battery powered devices in section 4.4.2, to be clear that  
the suggestion of only a single interval was intentional.


Section 4.4.1 requires a response to a keep alive to be received  
within 10 seconds of the request being sent. It's not clear why this  
value was chosen, especially since multi-second RTTs are not unheard  
of in some cases (c.f. the motivation for LEDBAT). While I don't think  
this is an inappropriate choice, some motivation and rationale would  
make it easier to judge why this value is sensible, and when it might  
be appropriate to use a different time out.


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Re: tsv-dir review of draft-ietf-mext-nemo-v4traversal-06.txt

2008-12-03 Thread Colin Perkins

On 2 Dec 2008, at 19:51, Jari Arkko wrote:

Thanks for your review, Colin.

I agree that fixes or better explanations of the background should  
be given for the points that you raised. It seems that you and  
Hesham are making progress on what needs to be added to the  
document, great!


The only item that I too am somewhat worried about in your list is  
the address replacement by broken NATs and the tricks used to  
prevent that. I feel that the best way to deal with that is to  
document this as a potential problem -- the protocol already has  
tools to provide a far better obfuscation technique than XOR --  
just turn on encryption if this turns out to be a problem and the  
device in question cannot be ripped out of the network.


Agreed.

Do we have any information about how widespread such NATs might be?  
RFC 5389 just says "some NATs".



Not really, but something that broken can't be common: it'd cause too  
many obvious problems.


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Re: [tsv-dir] tsv-dir review of draft-ietf-mext-nemo-v4traversal-06.txt

2008-12-01 Thread Colin Perkins

On 1 Dec 2008, at 11:06, Magnus Westerlund wrote:

Hesham Soliman skrev:

Thanks Colin,

I agree with your rationale but I wonder if we need to support  
every broken device out there. In any case, if we have to, I  
prefer to require encryption than to add the XORED address option.  
I'd like to hear what people in MEXT think about this, comments  
from MEXT?


I don't think we should go to far to correct this bug in
implementations. My understanding is that this have been observed  
but is
not a common behavior as these generic ALGs do break things  
randomly and
horribly. I would only do something about if it has no minimal  
impact on
the specification and any deployed implementations. Otherwise the  
fix is

likely to cause more grief than the few devices that are broken.



Right - the issue should be documented, and if there is a trivial fix  
it should be recommended, but this isn't a big issue.


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Re: tsv-dir review of draft-ietf-mext-nemo-v4traversal-06.txt

2008-12-01 Thread Colin Perkins

On 1 Dec 2008, at 09:00, Hesham Soliman wrote:

=> Well, I'm not sure how a NAT can do that. You mean the NAT will
parse the binding update message deep inside the IPv6 extension
header in the inner IP packet? This is where the original address
is preserved. To do that, a NAT would have to understand the
various MIPv6 options, and if it did, it would know not to do
that :) The inner header is IPv6, so a NAT should not touch that.


My understanding from the STUN work is that NATs have been observed
which rewrite any sequence of four aligned bytes matching the source
IP address, irrespective of its location within the packet (section
15.2 of RFC 5389).


=> Sounds freightning! May be we need to mandate encryption and  
hope that no 4-byte sequence matched the IP address? What do they  
do with encrypted packets? How do they know they're encrypted?


One would assume these broken devices will potentially corrupt  
encrypted packets, the same as they will potentially corrupt any  
other packet. Hiding the source address when it appears in the  
payload (either by encryption, or using some trivial obfuscation as  
does STUN) simply reduces the probability of such corruption so it's  
no longer 100% guaranteed that the payload will be mangled.


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Re: tsv-dir review of draft-ietf-mext-nemo-v4traversal-06.txt

2008-12-01 Thread Colin Perkins

Hesham,

Thanks - one minor follow-up inline.
Colin


On 1 Dec 2008, at 08:29, Hesham Soliman wrote:

Colin,

Thank you for the review. Please see some comments inline.



In summary, this draft seems to be on the right track but has some
open issues, as discussed below.

The NAT keep-alive interval (Section 5.3 and 7) is set to 110 seconds
by default, unless configured by the home agent in a binding
acknowledgement. While the default interval seems a safe choice, it's
not clear why it was chosen, or when it's appropriate for a home
agent to signal an alternative interval. The draft would benefit from
an explanation of this choice, and further discussion of the issues
inherent in choosing an appropriate keep-alive interval (Section 3.5
of RFC 5405 considers this point, and might be appropriate to
reference).


=> Thanks for the reference. I'll include that in the new rev. The  
interval was chosen based on recommendations from people with  
experience in NAT deployments and the WG agreed with it. But the  
reference will certainly help the reader.



The NAT detection mechanism specified in Section 5.2 relies on the
NAT rewriting the source address of the outer IPv4 header, but not
the IPv4 address carried in the binding update payload. I'm not
familiar with the way in which mobile IPv6 uses IPsec ESP, but if
it's used in a way that provides integrity protection but not
confidentiality, issues may arise with NAT devices that rewrite IPv4
addresses in the UDP payload, causing the integrity check to fail
(this is why STUN [RFC5389] provides an XOR-MAPPED-ADDRESS attribute
to obfuscate the IP address).


=> Well, I'm not sure how a NAT can do that. You mean the NAT will  
parse the binding update message deep inside the IPv6 extension  
header in the inner IP packet? This is where the original address  
is preserved. To do that, a NAT would have to understand the  
various MIPv6 options, and if it did, it would know not to do  
that :) The inner header is IPv6, so a NAT should not touch that.


My understanding from the STUN work is that NATs have been observed  
which rewrite any sequence of four aligned bytes matching the source  
IP address, irrespective of its location within the packet (section  
15.2 of RFC 5389).



It's not clear that the use with ECN is fully specified. Section
5.1.1 notes that it's "RECOMMENDED that the ECN information is copied
between the inner and outer header as defined in [RFC3168]" - is the
intent that the full-functionality option defined in RFC 3168 is
recommended? If so, it would be useful to state that explicitly.


=> Yes. I think we can explicitly mention that. This was done to  
follow the same rules which are done for IP in IP encapsulation, as  
per RFC 4213.


It

also seems that UDP encapsulation will prevent the use of ECN over
the tunnel, since the UDP API doesn't allow access to the ECN bits,
and cause different ECN (and hence congestion) behaviour depending on
the encapsulation method used. This should be called out in the  
draft.


=> Ok, although I think this might not be an issue with many  
implementations as UDP encapsulation may be done in the kernel with  
the rest of the MIPv6 implementations (typically anyway). I'll add  
your comment to the draft.



Processing of other IP header fields (e.g. the diffserv field and
TTL) when encapsulated in UDP tunnels also seems poorly defined. The
intent, presumably, is for the choice of tunnelling mechanism to be
hidden from higher-levels, but the exact en-/de-capsulation steps
needed to achieve this must be specified for the UDP encapsulation.
For example, when is the TTL of the encapsulated packet decremented?


=> It is decremented in the same way that it's done for IP in IP  
tunnelling. I can add that.



Finally, the use of multiple UDP tunnelling schemes (vanilla or with
TLV-header) seems undesirable, and it would be useful to document why
this is necessary, for those not familiar with the working group
history.


=> Well, I personally agree with you. However, there was a  
concensus call in the WG about this particular issue and it was  
decided (narrowly) that we should include support for this. Based  
on my understanding, some people wanted to be able to efficiently  
use GRE tunnelling. While this is not useful for MIPv6 mobile nodes  
(typically GRE tunnelling is done in the network), people wanted to  
be able to use this specification for PMIPv6, which uses MIPv6  
without the mobile node's involvement. Hence, the supposed need for  
GRE tunnelling. If you ask me why GRE tunnelling is needed at all  
in IPv6 then I'm not the right person to represent that opinion  
because I'm completely against it :)

I hope this helps.

Hesham


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tsv-dir review of draft-ietf-mext-nemo-v4traversal-06.txt

2008-11-30 Thread Colin Perkins
I've reviewed this document as part of the transport area  
directorate's ongoing effort to review key IETF documents. These  
comments were written primarily for the transport area directors, but  
are copied to the document's authors for their information and to  
allow them to address any issues raised. The authors should consider  
this review together with any other last-call comments they receive.  
Please always CC [EMAIL PROTECTED] if you reply to or forward this  
review.


In summary, this draft seems to be on the right track but has some  
open issues, as discussed below.


The NAT keep-alive interval (Section 5.3 and 7) is set to 110 seconds  
by default, unless configured by the home agent in a binding  
acknowledgement. While the default interval seems a safe choice, it's  
not clear why it was chosen, or when it's appropriate for a home  
agent to signal an alternative interval. The draft would benefit from  
an explanation of this choice, and further discussion of the issues  
inherent in choosing an appropriate keep-alive interval (Section 3.5  
of RFC 5405 considers this point, and might be appropriate to  
reference).


The NAT detection mechanism specified in Section 5.2 relies on the  
NAT rewriting the source address of the outer IPv4 header, but not  
the IPv4 address carried in the binding update payload. I'm not  
familiar with the way in which mobile IPv6 uses IPsec ESP, but if  
it's used in a way that provides integrity protection but not  
confidentiality, issues may arise with NAT devices that rewrite IPv4  
addresses in the UDP payload, causing the integrity check to fail  
(this is why STUN [RFC5389] provides an XOR-MAPPED-ADDRESS attribute  
to obfuscate the IP address).


It's not clear that the use with ECN is fully specified. Section  
5.1.1 notes that it's "RECOMMENDED that the ECN information is copied  
between the inner and outer header as defined in [RFC3168]" - is the  
intent that the full-functionality option defined in RFC 3168 is  
recommended? If so, it would be useful to state that explicitly. It  
also seems that UDP encapsulation will prevent the use of ECN over  
the tunnel, since the UDP API doesn't allow access to the ECN bits,  
and cause different ECN (and hence congestion) behaviour depending on  
the encapsulation method used. This should be called out in the draft.


Processing of other IP header fields (e.g. the diffserv field and  
TTL) when encapsulated in UDP tunnels also seems poorly defined. The  
intent, presumably, is for the choice of tunnelling mechanism to be  
hidden from higher-levels, but the exact en-/de-capsulation steps  
needed to achieve this must be specified for the UDP encapsulation.  
For example, when is the TTL of the encapsulated packet decremented?


Finally, the use of multiple UDP tunnelling schemes (vanilla or with  
TLV-header) seems undesirable, and it would be useful to document why  
this is necessary, for those not familiar with the working group  
history.


Regards,
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Re: [P2PSIP] P2PSIP diagnostics: PING discussion

2008-11-13 Thread Colin Perkins

On 13 Nov 2008, at 09:24, Roni Even wrote:
I am not sure what you mean NTP refresh, if the test was using the  
system clock than you would expect in XP that it will be updated  
every tick which is 10 or 15 msec which can account for the error.  
NTP is used in RTP (RFC 3550) for synchronization and the RTP time  
is using the system clock which may add skew but it is not because  
of NTP.





One minor clarification: RTP uses an NTP format timestamp, but it  
doesn't require the use of NTP.


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Re: SAFE BoF in Vancouver

2007-11-20 Thread Colin Perkins

On 21 Nov 2007, at 00:46, Ted Hardie wrote:

At 6:38 PM -0600 11/20/07, James M. Polk wrote:

But Ted, you're going to Geopriv?  So how does this work?


I'm trying to focus on the general problem (particularly the
apps open conflict), rather than my own ability to be there.  I'm  
conflicted either way, but lots of other APPs people would be free  
on Friday.  Similar for MEDIACTRL; some might still be conflicted,  
but a larger number would be free.  I understand from Mary Barnes  
that she would have a conflict with the Friday slot, so it may be  
unsolvable, but I hope there may still be a way.


There's also a desire to avoid BoFs late in the week, to give time  
for hallway discussions on the outcome (as Pete Resnick has just  
mentioned on the WG chairs list).


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Re: SAFE BoF in Vancouver

2007-11-20 Thread Colin Perkins

On 20 Nov 2007, at 19:07, Ted Hardie wrote:

At 1:59 PM + 11/16/07, Colin Perkins wrote:
The following BoF has been proposed for the Vancouver IETF. There  
is a mailing list <[EMAIL PROTECTED]> for discussion.


This seems to be scheduled against both the Applications area open  
meeting and a RAI group focused on media servers.  Both groups  
would have an interest in following this work and discussing where  
future IETF work in this area will happen. Is there still a  
possibility of adjusting this timing?  I understand, of course,  
that not every conflict can be resolved, but it would be useful to  
know whether this is still something that might be addressed.


We're aware of the conflicts, but this is the least bad scheduling  
we've been able to find. If you have suggestions for a better slot,  
we're open to ideas.


Regards,
Colin

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SAFE BoF in Vancouver

2007-11-16 Thread Colin Perkins
The following BoF has been proposed for the Vancouver IETF. There is  
a mailing list <[EMAIL PROTECTED]> for discussion.


Colin




SAFE - Self-Address Fixing Evolution BoF


Chairs:
  Colin Perkins  ([EMAIL PROTECTED])
  Markus Isomaki ([EMAIL PROTECTED])

Mailing list:
  https://www1.ietf.org/mailman/listinfo/safe

Various NAT hole-punching techniques such as IPsec NAT traversal,  
Teredo, and STUN/ICE send periodic UDP keep-alive messages to keep  
their NAT binding alive.


However, a drawback of these techniques is their chattiness which is  
a result of the host application not knowing the NAT's binding  
lifetime (IPsec NAT traversal, STUN/ICE) or because the application  
is unable to extend the lifetime of the NAT's binding (Teredo).  The  
endpoint has to send periodic packets which consume power on battery  
powered devices, consume network bandwidth, and place an unnecessary  
load on servers.


There are two approaches to resolve the problem of chattiness. The  
first is to interact directly with the NAT using a NAT control  
protocol. Several of these protocols exist which unfortunately have  
different drawbacks:


  * incremental deployment is not possible with MIDCOM, NSIS-NSLP,
UPnP IGD, or NAT-PMP.  With all of these protocols, both the NAT
and the endpoint have to support the same protocol
  * nested NATs are not possible with UPnP IGD or NAT-PMP
  * topology awareness is required of MIDCOM
  * security must be established between the controlling entity and
the NAT for MIDCOM and NSIS-NSLP
  * UPnP IGD uses broadcasts packets and NAT-PMP expects the NAT to be
the default gateway; neither work well on routed networks

The second approach is to empirically test the NAT's binding  
lifetime, as done by Teredo. This can optimise the keep-alive traffic  
based on the NAT's binding lifetime, but cannot extend the duration  
of the binding lifetime. Also, empirical testing does not always give  
reliable results due to varying behaviour of NAT and firewall  
implementations.


This BoF is intended to discuss a newly-proposed technique for using  
STUN to discover, query and control firewalls and NATs, that can  
eliminate UDP
keep-alive traffic. The BoF will review the problem space and  
existing work, and decide if there is a need for new work in the  
area, and if the IETF is an appropriate home for that work. The  
intent is not to form a new working group at this time, but to gauge  
interest in work in this area, and consider an appropriate future  
home for that work.



Agenda:
   Introduction .. (Chairs, 10)
   Problem statement and scope . (Wing, 15)
   Survey of existing work ... (Barnes, 30)
  draft-eggert-middlebox-control-survey-01.txt
   NAT/Firewall control with STUN .. (Wing, 15)
  draft-wing-behave-nat-control-stun-usage-05.txt
   Discussion  (20)
   Future directions . (Chairs, 30)


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Re: Gen-ART review of draft-ietf-avt-rtp-and-rtcp-mux-05.txt

2007-07-17 Thread Colin Perkins
 are updated as follows.  When assigning  
RTP RTCP
> Control Packet types, IANA is requested to assign unused  
values from
> the range 200-223 where possible.  If that range is fully  
occupied,
> values from the range 194-199 may be used, and then values  
from the
> ranges 1-191 and 224-254. This improves header validity  
checking of
> RTCP packets compared to RTP packets or other unrelated  
packets. The
> values 0 and 255 are avoided for improved validity checking  
relative to
> random packets since all-zeros and all-ones are common values.  




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Re: Gen-ART review of draft-ietf-avt-rtp-and-rtcp-mux-05.txt

2007-07-17 Thread Colin Perkins

On 17 Jul 2007, at 16:33, [EMAIL PROTECTED] wrote:

idnits 2.04.12 found a couple of reference nits:

   Checking references for intended status: Proposed Standard

-- 
--


 (See RFC 3967 for information about using normative references to
 lower-maturity documents in RFCs)
  ** Obsolete normative reference: RFC 2032 (ref. '3') (Obsoleted  
by RFC

4587)


This is intentional, and noted in the PROTO write-up in the tracker,  
since the aim is to reference the FIR and NACK packets defined in RFC  
2032, then removed when the H.261 payload format was updated.



  -- Possible downref: Normative reference to a draft: ref. '6'  (No
intended
 status found in state file of draft-ietf-avt-rtcpssm)


The draft-ietf-avt-rtcpssm-13.txt draft is intended for standards  
track, so this is not an issue.



  == Outdated reference: A later version (-17) exists of
 draft-ietf-mmusic-ice-15


ICE -16 and -17 were both submitted after this draft. The reference  
was current on submission.


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Re: Gen-ART review of draft-ietf-avt-rtp-and-rtcp-mux-05.txt

2007-07-17 Thread Colin Perkins

David,

Thanks for the review - some comments inline.

On 16 Jul 2007, at 22:35, [EMAIL PROTECTED] wrote:

I have been selected as the General Area Review Team (Gen-ART)
reviewer for this draft (for background on Gen-ART, please see
http://www.alvestrand.no/ietf/gen/art/gen-art-FAQ.html).

Please resolve these comments along with any other Last Call
comments you may receive.


Document: draft-ietf-avt-rtp-and-rtcp-mux-05.txt
Reviewer: David Black
Review Date: 16 July 2007
IETF LC End Date: 17 July 2007

Summary:
This draft is on the right track, but has open issues,
described in the review.

Comments:

The draft is generally in good shape, although one needs to
be an RTP expert to understand all the details.

The only "open issue" is the missing instructions to IANA
for RTCP packet type registration - from a technical standpoint,
this is a nit, but getting it right is of sufficient importance
to avoidance of future restrictions RTP/RTCP multiplexing that
I've flagged it as an open issue.

Everything else in this review is an editorial comment, although
the absence of the explanation of the mechanics of the RTP/RTCP
conflicts makes that section of the draft difficult to read.

Section 1 talks about NAT (Network Address Translation) as a
motivation, but the real motivation appears to be NAPT (Network
Address Port Translation).  This ought to be discussed, and I
strongly suggest an Informative reference to RFC 3022.  The
term "NAT pinhole" also needs to be explained here to connect
the problems caused by two UDP ports to NAPT usage, and it
may be useful to mention firewall pinholes as a related issue.


We can change "Network Address Translation (NAT)" to "Network Address  
Port Translation (NAPT)" in the Introduction, and add a reference to  
RFC 3022, sure. Sloppy terminology on my part.


Would changing "since opening multiple NAT pinholes can be costly" to  
"since maintaining multiple NAT bindings can be costly" address your  
other concern?



Section 4 should explain the mechanics of the RTP/RTCP conflict:
- The RTCP PT is bits 8-15.
- The RTP PT is bits 9-15.
- RTP uses bit 8 in that word for a M (Marker) bit that
may be on or off.
The latter item is the cause for needing to check for whether the
RTCP PT conflicts with either the RTP PT or the RTP PT + 128.


The draft hints at this already in section 4:

   This demultiplexing method works because the RTP payload type and
   RTCP packet type occupy the same position within the packet.

If you think more is needed, please suggest wording, and I'll add it.


The last 2 paragraphs in Section 4 before the final Note in that
section need attention:
- The paragraph on registration of new RTCP packet types needs to
instruct IANA on what rules to enforce.  The use of "SHOULD"
in the current text is not sufficient, instead this needs to
be restated as instructions to IANA on how to assign RTCP
packet types and noted in the IANA considerations section.


Makes sense. I suggest adding something like the following to the  
IANA considerations:


   The rules for assignment of RTP RTCP Control Packet Types in the RTP
   Parameters registry are updated as follows.  When assigning RTP RTCP
   Control Packet types, IANA is requested to assign unused values from
   the range 200-254 where possible.  If that range is fully occupied,
   values from the range 194-199 may be used, and then values from the
   range 1-191. This improves header validity checking of RTCP packets
   compared to RTP packets or other unrelated packets. The values 0 and
   255 are avoided for improved validity checking relative to random
   packets since all-zeros and all-ones are common values.


- In this text in the second paragraph:

   Given these constraints, it is RECOMMENDED to follow the guidelines
   in the RTP/AVP profile [7] for the choice of RTP payload type  
values,


Insert the word "dynamic" between "the choice of" and "RTP
payload type values" for clarity.


The guidelines in the RTP/AVP profile apply to both static and  
dynamic RTP payload types, so inserting "dynamic" here would be  
incorrect.


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Re: Last Call: draft-ietf-dccp-rtp (RTP and the DatagramCongestion Control Protocol (DCCP)) to Proposed Standard

2007-06-11 Thread Colin Perkins

Dan,

On 8 Jun 2007, at 18:38, Dan Wing wrote:

The draft is easy to read and understand.

1. Section 5.2, "Service Codes", refers to DCCP service codes.   
Several are
defined in this specification (SC:RTPA, SC:RTPV, SC:RTPT, and  
SC:RTPO).  The
draft needs to say something about the relationship between the SDP  
media
type field (the "audio" and "video" of m=audio and m=video) and the  
DCCP

service code -- I expect they SHOULD match.


Yes - I'll clarify.


2. I wonder if an optimization would be useful, where
"a=dccp-service-code:SC:RTPV" is assumed if the associated media  
type is
"video" (m=video); a similar optimization seems reasonable for  
audio (RTPA

is assumed when the media type is audio (m=audio)).


I tend to prefer explicit to implicit, so while this would reduce the  
size of the SDP slightly, I'm not sure it's worthwhile.


3. I believe the draft needs to define a value for its IANA- 
registered DCCP

service codes, as service codes appear to be sent in the DCCP packets
themselves according to RFC4340.


The service codes sent in the DCCP packets are the 4-character ASCII  
strings listed (e.g. RTPV).


4. I'm surprised a=rtcp (RFC3605) lacked normative language in  
section 5.1.
I know for UDP this is widely considered best practice.  I'm not  
confident
that we should expect DCCP will be able to avoid that quagmire.  I  
suggest

adding something like:

   If RTCP is to be sent on a separate DCCP connection
   to RTP, the RTCP connection SHOULD use the next higher destination
   port number, unless an alternative DCCP port is signalled using the
   "a=rtcp:" attribute [13].  For improved interoperability, "a=rtcp"
  ^^^
   SHOULD be used whenever an alternative DCCP port is used.
   ^^^^^^^^^


Makes sense - I'll add this.

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tsv-dir review of draft-shacham-sipping-session-mobility-03.txt

2007-05-08 Thread Colin Perkins
I've reviewed this document as part of the transport area  
directorate's ongoing effort to review key IETF documents. These  
comments were written primarily for the transport area directors, but  
are copied to the document's authors for their information and to  
allow them to address any issues raised.


The purpose of the mechanisms described is to transfer media streams,  
likely based on RTP, between devices. This is done using SLP to  
discover devices, followed by a SIP exchange to negotiate new (RTP)  
media streams. The use of SLP and SIP does not appear to offer  
concern from a transport perspective (I have not reviewed it for  
correctness from an SLP or SIP perspective).


The main transport concern with this draft is with how to ensure  
retargeted RTP streams don't congest the new network path. The draft  
touches on this issue in Section 6, hinting that application-level  
measures may be used to adjust, but it's not clear that is  
sufficient. An ideal response to ensure congestion safety would be to  
slow-start on the new path, but that's also problematic since it  
would unnecessarily disrupt the media on paths that can sustain the  
required rate. Given these conflicting constraints, the best solution  
we have available might be to signal the new link capacity in the SIP  
exchange if the bottleneck bandwidth is known, and be prepared to  
react to congestion within a small number of round-trip times (e.g.  
perhaps using early RTCP, or transport-level feedback to signal the  
need to adapt). The draft should include more discussion of these  
issues, and the potential solutions.


Section 5.2.1.2 implies that the connections for real-time media do  
not need to be established before use. This is not the case if  
comedia (RFC 4145) is used, e.g. for RTP streams running over TCP,  
TLS, or DCCP. The order of connection setup should be addressed.


The draft seems to ignore the presence of NAT boxes which might  
disrupt media connectivity. An ICE exchange will likely be needed to  
work around this during handover. At what point in the call flows  
should this be done? Similarly, the RTPSEC work is considering media- 
path keying for SRTP (e.g. DTLS or ZRTP), which also needs to be done  
during the handover, but is not mentioned.


None of these are not insurmountable problems, but they should be  
addressed before the draft is ready for publication.


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



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Re: RFC 4612 - historic status

2006-08-14 Thread Colin Perkins

John,

Without wishing to comment on the specifics of RFC 4612...

On 14 Aug 2006, at 09:13, John C Klensin wrote:

--On Monday, 14 August, 2006 03:41 -0400 "Paul E. Jones"
<[EMAIL PROTECTED]> wrote:


Brian,

The problem with using "image" is that it would mean that a
gateway would have to do one of:
1) Close the audio session and open an image session
2) Open a second "image" session during mid-call
3) Open both an audio session and image session at the outset

For a lot of reasons, none of those options are preferred.  In
the latter case, gateways just waste memory and CPU resources
servicing sockets that are never used.  In the first two
cases, there is a lot of extra signaling on the wire.  (For
SIP networks, and especially IMS-based SIP networks, this is
horrible due to the widespread use of UDP, slow timer
...


Paul,

I think there may be a fundamental misunderstanding, or at least
difference in understanding, about top-level media types implied
in the above.  From a process standpoint, if you are just
finding out about it now, something has, IMO, failed (see my
other note).  But, putting that aside...

The "text/" top-level type, and only the "text/" top-level type
has the property that one can reasonably expect to be able to
display the contents to the user without understanding the
subtype.  Whether that is realistic or not, and how narrowly
that requirement should (or must) be interpreted to make
something "text/" has been widely debated in the past.

But, for all other media types, one cannot interpret them
without both the top-level type and the subtype.  As far as
transmission issues are concerned, the type-subtype distinction
is largely irrelevant: there is not, meaningfully, an "audio
session" or an "image session" as far as the media type is
concerned.


This is not correct when it comes to RTP sessions signalled using  
media types in SIP/SDP. In those cases, Paul is correct that the  
session is negotiated to convey only one top level media type (e.g.  
only audio), and switching to an image format would require a  
separate session.


Discussion of why this is the case should probably happen on the AVT  
mailing list, since that is the working group responsible for RTP, or  
on rai-discuss.


Colin

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Re: Vancouver schedule

2005-11-10 Thread Colin Perkins

Fully agreed - the new schedule fits much better with my jet lag.

Also, I really liked the way some of the rooms had tables and power  
for the front two rows...


Colin



On 10 Nov 2005, at 06:13, Romascanu, Dan ((Dan)) wrote:
Actually I prefer this schedule because it does not call me for a  
night

session after dinner, and I suspect that many people coming from
European time zones may feel the same. As Ole points out, availability
of coffee during the afternoon sessions would help us make it easier
through the long afternoon meeting period.

Regards,

Dan





-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Behalf Of Aaron Falk
Sent: Thursday, November 10, 2005 9:40 AM
To: ietf@ietf.org
Subject: Vancouver schedule

Am I the only one dissatisfied with the meeting schedule?  I
find that the run of meetings from 1300 to 1930 is just too
long, especially the four hour period from 1300 - 1710.  I
would strongly prefer our 'traditional'
schedule over the current one.

Even if I can't know which continent I'll be meeting in, it
would be nice to know that the daily schedule is one I can
live with.  ;)

--aaron

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




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Re: RTP vs. MIME

2005-07-22 Thread Colin Perkins

On 22 Jul 2005, at 16:25, Bruce Lilly wrote:
...
To the extent that transport protocol overhead has been erroneously  
included in what is purported to be a media format specification,  
that is a problem with the registration and/or specification.


There is nothing in the MIME rules which requires an RFC to include  
only a MIME registration template. Despite your attempts to claim  
otherwise, it is entirely legitimate and appropriate for an RFC to  
include both a media type registration and some other specification.


And I would add that that causes problems for MIME reviewers and  
implementers, and said problems would be alleviated by a separate  
registry with separate rules, where RTP-specific registrations can  
conflate transport bits with format bits or not without causing  
problems for MIME reviewers and implementers.


The documents clearly separate "transport bits" from "format bits",  
so this is not an issue.  This separation has been repeatedly  
explained to you, and is documented in these particular  
specifications, and the RTP specification (RFC 3550 -- STD 64).



You are confusing the applicability statement which
describes RTP-specific processing with the description of the media
type.


I see nothing in 3267 labeled "applicability statement"; I do see
multiple "format" descriptions.


There is no requirement in RFC 2026 for the explicit label  
"applicability statement" to appear anywhere, so this cannot be your  
complaint. You will find several format descriptions in the RFC: two  
RTP payload formats, and two media types. The two RTP payload format  
definitions (section 4 of RFC 3267) describe how the two media types  
are conveyed within RTP, but don't define the media types. The two  
media types are described in section 5 of RFC 3267, with reference to  
the 3GPP specifications which define the bit patterns and semantics  
of the speech frames. The media type registrations are in section 8  
of RFC 3267. You are, as I previously stated, confusing the RTP  
specific processing (an "RTP Payload Format") with the description of  
the media type.



The formats are specified in the following:

[1]   3GPP TS 26.090, "Adaptive Multi-Rate (AMR) speech
transcoding",
  version 4.0.0 (2001-03), 3rd Generation Partnership Project
  (3GPP).

[2]   3GPP TS 26.101, "AMR Speech Codec Frame Structure", version
  4.1.0 (2001-06), 3rd Generation Partnership Project (3GPP).

[3]   3GPP TS 26.190 "AMR Wideband speech codec; Transcoding
  functions", version 5.0.0 (2001-03), 3rd Generation
Partnership
  Project (3GPP).

[4]   3GPP TS 26.201 "AMR Wideband speech codec; Frame  
Structure",

  version 5.0.0 (2001-03), 3rd Generation Partnership Project
  (3GPP).


26.090, according to http://www.3gpp.org/ftp/Specs/html-info/26090.htm
is "AMR speech Codec; Transcoding Functions".  Transcoding  
functions and

media formats are separate types of specifications.  "Frame structure"
specifications *might* be applicable, *if* the media format is  
identical

to said "frame structure", but I see nothing in the registration form
that says so, and several parts of section 5 imply otherwise; sections
5, 5.1, and 5.2 make no reference to what precisely comprises "speech
frames" or "speech bits" or indeed why two different terms are used.


The 3GPP documents are, without any doubt, both applicable and  
appropriate specifications for these formats.  There are several  
dozen independent and interoperable implementations based on the  
specifications, using the media formats both with RTP and in more  
traditional MIME contexts such as file downloads. These  
implementations are in daily use by tens of millions of users. By any  
measure, this meets the necessary standards of specification clarity  
for the community which is implementing the standard.


That is a very good reason for a separate registry.  There is no  
question that the proposed media type did not meet the  
requirements for registration in the text top-level MIME  
registry; it should never have been proposed to register it  
there.  With a separate RTP registry, those interested in RTP can  
establish whatever rules or lack of rules for things to be  
considered as RTP "text".


You seem to be complaining because we asked the advice of the MIME
community on whether the 3GPP timed text format should be registered
under text or video, and then took that advice. Quite how this can be
construed as a failure of the process, or as a statement that RTP is
not appropriate for use with MIME types is beyond me.


One might as well ask whether or not an elephant is eligible for a dog
license.  The question shouldn't even be asked.  And it wasn't a  
simple

matter of "took that advice", there was argument that even though the
media clearly was incompatible with text, that it should be registered
under text anyway, presumably merely because some RTPers would like it
that way wi

Re: RTP vs. MIME

2005-07-20 Thread Colin Perkins

On 20 Jul 2005, at 18:26, Bruce Lilly wrote:

 Date: 2005-07-12 15:58
 From: Colin Perkins <[EMAIL PROTECTED]>



On 12 Jul 2005, at 13:43, Bruce Lilly wrote:


  Re: Ietf Digest, Vol 15, Issue 19
 Date: 2005-07-11 19:13
 From: Colin Perkins <[EMAIL PROTECTED]>


[...]


As will I, since your message includes many misstatements of facts
relating to the RTP transport protocol and the way in which it
conveys media data.


I recall making no statements about RTP or the way it conveys data
save for the fact that that is as irrelevant to defining a media type
as are details of TCP or UDP etc.


You have made, and continue to make, numerous statements about RTP  
payload formats and their associated media type registrations which  
are factually incorrect. The most obvious of these is your insistence  
on treating the RTP transport protocol headers as part of the media  
data, as you repeatedly do in the message to which I am replying.



The audio/amr format contains identical media data, but the RTP
transport is defined to strip the initial magic number, which is
redundant with fields in the RTP protocol headers. The wide band
version of AMR, and the related EVRC formats are similar.


Audio/amr explicitly specifies two separate media formats; if it
were merely a matter of some RTP processing, such processing could
have been specified separately from the format.


The processing is specified separately from the format, so I don't
understand this comment.


Audio/amr is defined in RFC 3267. The registration forms state:
   This type is defined for transfer via both RTP (RFC  
1889)
   and stored-file methods as described in Sections 4  
and 5,

   respectively, of RFC 3267.
RFC 3267 sections 4 and 5 indeed define separate formats; the section
headings both refer to "Format" (plural in the case of section 4).


Since RFC 3267 registers both audio/amr and audio/amr-wb and defines  
their use with RTP and as file formats, section 4 correctly refers to  
"Formats" in the plural. For each of the two media types, section 4  
of RFC 3267 describes the RTP-specific process by which that format  
is transported in RTP, and section 5 describes the file format (a  
sequence of concatenated frames of the media format preceded by a  
magic number). You are confusing the applicability statement which  
describes RTP-specific processing with the description of the media  
type.



The registration form data for "Public [sic; should be 'Published']
specification" says only
   Please refer to Section 11 of RFC 3267.

RFC 3267 section 11 provides no format specification at all; it is the
RFC section listing references.


The formats are specified in the following:

   [1]   3GPP TS 26.090, "Adaptive Multi-Rate (AMR) speech  
transcoding",

 version 4.0.0 (2001-03), 3rd Generation Partnership Project
 (3GPP).

   [2]   3GPP TS 26.101, "AMR Speech Codec Frame Structure", version
 4.1.0 (2001-06), 3rd Generation Partnership Project (3GPP).

   [3]   3GPP TS 26.190 "AMR Wideband speech codec; Transcoding
 functions", version 5.0.0 (2001-03), 3rd Generation  
Partnership

 Project (3GPP).

   [4]   3GPP TS 26.201 "AMR Wideband speech codec; Frame Structure",
 version 5.0.0 (2001-03), 3rd Generation Partnership Project
 (3GPP).

There are references 1-4 from section 11 of RFC 3267. This is made  
clear in sections 3.1 and 3.2 of RFC 3267, but you are correct in  
noting that the registration template doesn't clearly reference the  
formats. We'll add this to the errata list for the RFC.



2) As has been mentioned previously, the text/t140 format, which
started this discussion,


No, the discussion started with an IESG I-D requesting  
discussion, and
that followed a failed attempt to register "text/3gpp-tt" in the  
MIME

registry despite many incompatibilities of the proposed format with
the requirements for registration under the "text" top-level type.


As mentioned previously, that format is now being registered under
video. I believe this is an example of the MIME review working well:
in the initial request for MIME review for the 3GPP-TT format, I
said: "I'm not sure we have the experience to decide this in the AVT
working group. The question is about draft-ietf-avt-rtp-3gpp-timed-
text-04.txt: is it appropriate to register the format under the video
or text top-level type?" [1]. There was considerable discussion, and
as a result of that discussion, the format was registered under the
video top-level type.


That is a very good reason for a separate registry.  There is no  
question
that the proposed media type did not meet the requirements for  
registration
in the text top-level MIME registry; it should never have been  
proposed to
register it there.  With a separate RTP re

Re: RTP vs. MIME (Was Re: Ietf Digest, Vol 15, Issue 35)

2005-07-12 Thread Colin Perkins

On 12 Jul 2005, at 13:43, Bruce Lilly wrote:

  Re: Ietf Digest, Vol 15, Issue 19
 Date: 2005-07-11 19:13
 From: Colin Perkins <[EMAIL PROTECTED]>
 To: iesg@ietf.org
 CC: ietf@ietf.org, Stephen Casner <[EMAIL PROTECTED]>

Now the initial internet-draft deadline has passed I want to respond
to the main points of this message. I don't believe that a point-by-
point rebuttal is useful, due to the length of the message and
repetition of arguments.


Indeed repetition is not helpful, particularly when it is a repetition
of a misstatement of position or of facts.  I'll limit my remarks to
those cases.


As will I, since your message includes many misstatements of facts  
relating to the RTP transport protocol and the way in which it  
conveys media data.



The audio/amr format contains identical media data, but the RTP
transport is defined to strip the initial magic number, which is
redundant with fields in the RTP protocol headers. The wide band
version of AMR, and the related EVRC formats are similar.


Audio/amr explicitly specifies two separate media formats; if it
were merely a matter of some RTP processing, such processing could  
have

been specified separately from the format.


The processing is specified separately from the format, so I don't  
understand this comment.



The thrust of most of the rest of this message seems to be that the
use of MIME type to identify RTP audio and video formats is
uncontroversial,


Not completely uncontroversial, just less damaging than incompatible
text registrations.


but you have a serious concern over use of text
formats with RTP, since you believe these will disrupt MIME
operations due to "leakage". This concern difficult to justify for
several reasons:


It's not about "leakage", it hasn't been about "leakage"; that is a
minor issue that has been turned into a strawman in order to
demonstrate how easily it can be knocked down.



1) These types have been in widespread use for several years now;
there have been no reports of actual operational problems, opposed to
the theoretical arguments being made here.


No, the proposed text/3gpp-tt hasn't been in use as a MIME type.  Of
course there are no operational problems; they have been nipped in the
bud.  To the extent that text/t140 hasn't caused operational problems,
that is because it has been largely ignored (not without cost) because
it is in fact not a valid MIME type.


There is no such thing as text/3gpp-tt; that format is being  
registered under the video top level type, as a result of comments  
during MIME review. More on this, and on text/t140, below.



The arguments are not "theoretical"; mixing disparate types of
registrations in the single MIME registry, some of which do not use
MIME mechanisms at all, makes extra work for implementers -- work
which yields no benefits, only wastes effort and creates confusion.

If our specifications and procedures do not benefit implementers --
indeed when they cause confusion and busy-work -- there is a problem
which needs to be addressed.  In this case, separate registries for
MIME and non-MIME content is an appropriate solution.



2) As has been mentioned previously, the text/t140 format, which
started this discussion,


No, the discussion started with an IESG I-D requesting discussion, and
that followed a failed attempt to register "text/3gpp-tt" in the MIME
registry despite many incompatibilities of the proposed format with
the requirements for registration under the "text" top-level type.


As mentioned previously, that format is now being registered under  
video. I believe this is an example of the MIME review working well:  
in the initial request for MIME review for the 3GPP-TT format, I  
said: "I'm not sure we have the experience to decide this in the AVT  
working group. The question is about draft-ietf-avt-rtp-3gpp-timed- 
text-04.txt: is it appropriate to register the format under the video  
or text top-level type?" [1]. There was considerable discussion, and  
as a result of that discussion, the format was registered under the  
video top-level type.



does comply with the MIME registration rules
for the text media type. The format is plain UTF-8 text with implicit
timing data, there are no embedded control characters, the CRLF
sequence is only used to identify line breaks, and a language tag is
available.


RFC 2793 (the AVT 2793bis draft is unavailable) specifies:


The 2793bis draft was published as RFC 4103.


   This is an example of a T140 RTP packet without redundancy.
0   1   2   3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |V=2|P|X| CC=0  |M|   T140 PT   |   sequence number 

Re: Ietf Digest, Vol 15, Issue 19

2005-07-11 Thread Colin Perkins
Now the initial internet-draft deadline has passed I want to respond  
to the main points of this message. I don't believe that a point-by- 
point rebuttal is useful, due to the length of the message and  
repetition of arguments.


Comments follow inline.

On 6 Jul 2005, at 18:01, Bruce Lilly wrote:

 Date: 2005-07-06 04:06
 From: Stephen Casner <[EMAIL PROTECTED]>


Stephen, you wrote:

That's not it.  Of course all the existing registrations would be
copied.  However, all of the many RTP-related RFCs that refer to MIME
registrations would become obsolete and need to be revised.  A fair
number of those RFCs are referenced by other SDO's documents.


Are any of those RFCs on the Standards Track?  Are they all at full
Standard?   Those which are Standards Track but not yet full Standard
need to be followed up on to advance on the Standards Track, and that
would be a good time to do a global replace of MIME with RTP and add
an IANA considerations section directing IANA to mark the type in the
MIME registry as obsolete.  Should take about 30 seconds to do that.

The MIME registrations will never go away (unfortunately for MIME),
so if a particular implementer goes ferreting through the MIME
registrations because he has an old copy of a document that says
MIME, he'll still find the type.

Some of the motivations for moving the RTP namespace into the MIME  
namespace were:


  - To avoid different names for the same media format, such as  
MIME's

audio/basic and RTP's PCMU.  More on this below.


To date, I know of no "same media format" for MIME and RTP; Magnus
specifically mentioned audio/amr, which has different formats, and
alluded to a tiny number of others but was not specific.


The simplest is audio/L16. For historical reasons, audio/basic and  
audio/pcmu are identical but have different names: eliminating  
duplicate names for such formats was one of the initial reasons for  
using the MIME namespace for RTP.


The audio/amr format contains identical media data, but the RTP  
transport is defined to strip the initial magic number, which is  
redundant with fields in the RTP protocol headers. The wide band  
version of AMR, and the related EVRC formats are similar.


...

The AVT working group did a lot of work between then and now to
specify the mechanism for the registrations and to create
registrations with what I believe was careful attention to detail.
This process has imposed very little load on the "MIME folks"; in
fact, it has received little notice until the recent consideration of
two subtypes under the text major type.


That is because of the way MIME works and has always worked; the text
type is for media containing human-readable (w/o software) text.  The
fallback is to text/plain, which is what the Internet Message Format
carries in the absence of MIME.  There are relatively few  
interoperability

considerations for audio and video media subtypes, so the attitude is
largely "ho hum, another subtype; who cares".  Not for text. More  
below.

...

There is no proposal to change the way email, html, and fax use MIME
types.


On the contrary, any registration of a format which requires software
for interpretation, contains binary content, etc. under the MIME text
type would require massive backwards compatibiliy- and  
interoperability-

breaking changes.  Given the mission-critical use of the affected
applications (e.g. IETF business such as this and other mailing lists)
that is simply unacceptable.


The thrust of most of the rest of this message seems to be that the  
use of MIME type to identify RTP audio and video formats is  
uncontroversial, but you have a serious concern over use of text  
formats with RTP, since you believe these will disrupt MIME  
operations due to "leakage". This concern difficult to justify for  
several reasons:


1) These types have been in widespread use for several years now;  
there have been no reports of actual operational problems, opposed to  
the theoretical arguments being made here.


2) As has been mentioned previously, the text/t140 format, which  
started this discussion, does comply with the MIME registration rules  
for the text media type. The format is plain UTF-8 text with implicit  
timing data, there are no embedded control characters, the CRLF  
sequence is only used to identify line breaks, and a language tag is  
available.


The claimed "massive backwards compatibility- and interoperability- 
breaking" changes you suggest would occur have not been needed in the  
five years since this format was registered; or as a result of the 8  
years worth of deployment of other RTP formats signalled using a  
range of MIME media types within SDP.


...
The simple rule -- necessary to ensure interoperability -- is "MIME  
registry,
MIME rules".   If the RTP specification writers were to conform to  
the MIME
rules -- which are in plain view in RFCs 2046 & 2049 -- there  
wouldn't be an
issue; mind you, the types might be of little interest o

Re: I-D ACTION:draft-iesg-media-type-00.txt

2005-07-01 Thread Colin Perkins
Considering the text/t140 media type (RFC 4103) which caused this  
controversy:


On 1 Jul 2005, at 22:49, Bruce Lilly wrote:
...

While the proposed RTP "text" types would present a small but non-zero
security risk, the essential properties of MIME text types of  
having the

characteristics above are not shared by the RTP types:
o clearly software which understands the format is required.  The  
format

  includes embedded binary formatting information


No, it does not. The format contains only plain UTF-8 text. You may  
be confused because the RFC describing the media type also explains  
how that type is to be transported within RTP, since several features  
of RTP are configured according to the media type parameters, and  
repeats the definition of the RTP headers.



o not readable without software; indeed as pointed out, the RTP types
  cannot even be save in a file format without conversion software


The contents of the RTP packets are plain UTF-8 text. It can  
trivially be saved as a text/plain file, but doing so will lose the  
timing information provided by the RTP headers (it is the presence of  
the additional timing parameters that require this to be a separate  
media type to text/plain for use with RTP). There are other media  
types, under audio/* and video/* where this is a more complex  
process, but that does not affect this case.


o it would be most unreasonable to try to present the RTP types  
without

  appropriate interpretation software


One clearly needs an implementation of a transport protocol to  
receive data sent in that protocol. Once the data has been received  
from the transport protocol, the contents are plain UTF-8 text with  
implicit timing information.


o CRLF octets may appear in the RTP types (as binary data 0x0D0A)  
where

  they do not represent a line ending.  Note the BCP 14 "MUST" keyword
  in the approved RFC 2046 text, indicating interoperability issues of
  a serious nature (refer to BCP 14 section 6). [BCP14]


The text/t140 format contains UTF-8 text. The CRLF octets are only  
used to represent a line ending, not in any other case, in  
conformance with these rules.


o the proposed RTP registrations provide no charset parameter,  
defaulting

  to US-ASCII as far as MIME implementations are concerned


The charset is always UTF-8 for text/t140. When signalled in SDP,  
there is also provision for a language tag.


o fallback treatment as text plain is a problem for the RTP type  
because

  of the incompatibilities above.


The content is plain UTF-8 text.

Colin

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Re: I-D ACTION:draft-iesg-media-type-00.txt

2005-07-01 Thread Colin Perkins

On 1 Jul 2005, at 14:02, Robert Elz wrote:

Date:Fri, 1 Jul 2005 12:57:15 +0100
From:Colin Perkins <[EMAIL PROTECTED]>
Message-ID:  <[EMAIL PROTECTED]>

  | I do not find this argument persuasive, since the media types in
  | question have been deliberately specified as framed types to avoid
  | this issue.

I believe that you're both seeing, and missing the point, at the
same time there.

I'd expect that much of the problem is that media types should
depend upon the data that is carried, and how that is to be  
represented,
and should not depend in any way upon the mechanism used to carry  
them.


Some of the objections may have been from people who simply assumed  
that,

and could not believe that a parameter that is intended to specify the
format of data was instead (or additionally) being used to specify a
format for the transport protocol used to carry the data, or being  
linked

to such a thing.


You're misunderstanding what is being done. The media type does  
specify the format of the data, not of the transport protocol. In  
addition to this, the specification defines how to populate the RTP  
framing based on the implicit properties of the media (e.g. timing).



For example, if I receive an RTP stream carrying a text/whatever, and
store the content (unchanged, but removed from its RTP transport)  
into a

file, what media type do I use to identify the file?   If it remains
text/whatever (same value) then it will appear as that when I send it
via e-mail, or a web browser fetches it.   If it isn't to be text/ 
whatever
then what on earth is it?   Something random and unidentified?  If  
there

is a media type for it, that is what should be being used by RTP.


The media type registrations explain what is to be done. There are  
three possibilities:


1) In some cases, you use the exact same media type for the file  
format as for the RTP payload format, since they contain the same  
data formatted in the same way. I don't believe this is in the least  
controversial.


2) In some cases, the media type registration says "defined only for  
transfer via RTP". This usually means no-one saw the need for a file  
format at the time the registration was done, and didn't spend the  
effort to decide how the media should be stored. This should probably  
be rectified at some point.


3) In some cases, there was a pre-existing file format defined in  
such a way that makes it impossible to use within RTP. This is  
usually the case when the file format requires reliable delivery.  
Since the media format differs in the two cases, they use different  
media types. This case is the controversial case, since it is where  
types may leak outside their domain of applicability.


The question becomes: will the leakage go away if we separate the  
registries? I do not believe it will. People will continue to ignore  
the specifications, and will continue to use names in the wrong  
context. The split will remove one problem (people use the wrong  
media type when converting between RTP and a stored file) and cause  
another (people use the RTP name instead of the MIME media type when  
converting between RTP and a stored file). It's a lot of work, for no  
benefit.


Colin

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Re: I-D ACTION:draft-iesg-media-type-00.txt

2005-07-01 Thread Colin Perkins

On 31 May 2005, at 20:50, [EMAIL PROTECTED] wrote:
A New Internet-Draft is available from the on-line Internet-Drafts  
directories.
This draft is a work item of the Internet Engineering Steering  
Group Working Group of the IETF.


Title: A question on Media Type Registrations
Author(s): T. Hardie
Filename: draft-iesg-media-type-00.txt
Pages: 9
Date: 2005-5-31

This document poses a question to the community on the future of the
   IANA Media Type Registry.  It presents three potential future  
courses

   of development and asks that feedback on these be provided to the
   IESG.


The idea of using the MIME media types registry to identify types  
that are carried predominantly by RTP was suggested at a meeting of  
the AVT working group, with participation from the applications area  
directors, in December 1997 [1]. The subject was discussed at several  
following meetings, with an initial internet-draft being submitted in  
February 1999 [2]. A later draft was reviewed on the [EMAIL PROTECTED]> mailing list [3], and eventually published as RFC  
3555. At present, there are 71 media types registered for use  
predominantly with RTP. These media types are widely used in the  
voice over IP and media streaming communities, signalled within SIP  
and RTSP.


There has recently been concern expressed in some quarters about the  
use of media under the "text" top level media type within RTP. In  
particular, the update to RFC 2793 for Draft Standard RFC status was  
contentious, although it used the same text media type within RTP  
framing as the original RFC. Much of the concern seems to stem from  
the use of framed text media types, where the plain text media data  
is embedded within some binary wrapper for transport, since there is  
a belief that such data might "leak" into areas which don't  
understand the transport protocol in which the data is framed.


I do not find this argument persuasive, since the media types in  
question have been deliberately specified as framed types to avoid  
this issue. The types are defined only in terms of transport within  
RTP, and rely heavily on RTP framing and knowledge of the transport  
protocol. An implementation that does not understand RTP will be  
unable to receive the data: the type is simply not defined in a  
manner that allows implementation independent of the RTP transport  
protocol. Such framed text media types have been in use for over 5  
years now, with no history of problems.


This does not, of course, preclude an incorrect implementation  
producing data in some unspecified format which it labels with the  
media type of one of these framed formats, and transmits within some  
other transport protocol. The receiver of such data would clearly  
have to be robust in the face of the malformed content, but that  
robustness is needed whether or not framed media types exist. Indeed,  
reliance on the property that unknown text formats can be treated as  
"text/plain" and displayed directly to the user would seem to be a  
security risk, unless those formats are carefully validated as  
actually containing plain text of an appropriate character set,  
without malicious control characters. The use of framed media types  
does not change this property.


With this in mind, I turn to the three possible registry futures  
presented in draft-iesg-media-type-00.txt. The second and third  
proposals in section 2 of the draft ("MIME handling is the model for  
other using protocols" and "Registry split") are both large and  
backwards incompatible changes. Either change would register the 71  
existing media types defined primarily for use with RTP framing  
obsolete. These changes would require significant amounts of new  
standards work, and would cause enormous confusion for users of RTP  
payload formats and SDP. It would also affect the extremely large  
deployed base of implementations. I see no benefit from either of  
these changes, and many disadvantages.


The other option, "All media type protocols may specify handling", is  
the status quo. I believe this is the appropriate direction for the  
registry, although clarifications should be made to the registration  
rules, perhaps updating RFC 3555, to better explain how framed types  
are used. There are well defined and tested procedures for  
registering media types, an extremely large deployed base, and no  
history of problems with this approach.


The media types registry works well as currently specified. We have a  
large installed base, and well developed procedures. I do not believe  
it would be appropriate to make fundamental changes to these  
procedures or the registry at this late stage.


Colin




[1] Minutes of AVT working group at the 40th IETF

[2] P. Hoschka, "MIME Type Registration of RTP Payload Types",  
February 1999, draft-hoschka-rtp-mime-00.txt.


[3] Steve Casner, "Registration of RTP payload format names", message 

Re: Last Call: 'Media Type Specifications and Registration Procedures' to BCP

2005-04-12 Thread Colin Perkins
On 12 Apr 2005, at 23:04, Robert Elz wrote:
Date:Tue, 12 Apr 2005 21:20:28 +0100
From:Colin Perkins <[EMAIL PROTECTED]>
Message-ID:  <[EMAIL PROTECTED]>
  | RFC 3555 allows media types to be defined for transport only via 
RTP.
  | The majority of these registrations are under the audio and video
  | top-level types, with a small number being under text. Is your
  | objection to any media type being defined only for transport via 
RTP,
  | or to text media types being defined only for transport via RTP?

Not to either of those being attempted, but to the expectation that the
"only via RTP" will, or can, ever be enforced.
That is, to your earlier statement ...
	Sure, but if the display agent is unaware of the restrictions, it 
won't
	ever be able to receive the media data.

You'd need to be able to show me how that can possibly be true, when I
can trivially easily send e-mail with text/t140 in the Content-Type 
header.
You can put anything you like in a Content-Type header, but that 
doesn't make the data meaningful without a specification for the 
content.

The restriction for "only via RTP" is made by only specifying the 
framing rules for RTP transport; the necessary information to convey 
the type via non-RTP transports is simply not specified. There are many 
(more than 50) such media types registered, and they are widely 
implemented without the problems at which you are hinting.

Colin
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Re: Last Call: 'Media Type Specifications and Registration Procedures' to BCP

2005-04-12 Thread Colin Perkins
On 12 Apr 2005, at 20:51, Robert Elz wrote:
Date:Tue, 12 Apr 2005 19:03:03 +0100
From:Colin Perkins <[EMAIL PROTECTED]>
Message-ID:  <[EMAIL PROTECTED]>
  | Sure, but if the display agent is unaware of the restrictions, it 
won't
  | ever be able to receive the media data. The example I have in mind 
in
  | "text/t140" which is UTF-8 text framed within RTP packets [RFC 
2793].
  | The media type is defined only for use with RTP,

I'm not sure I believe that works.   What prevents this message from
containing an alternative to the text/plain that is (currently) its
only body type, with a Content-Type: text/t140 field (and correctly
formatted for that, as much as it can be in a e-mailbody)  ?
Or what prevents a HTTP response with Content-Type: text/t140?
Sure, the rules for use of that content-type may have been violated,
but what is your MUA (or browser) supposed to do when it receives the 
message?
All it is likely to know is that it has received yet another odd 
text/unknown
content type.   And if it is to be displayed, treating it just like all
other text/weird types as the fallback (better than nothing) option ?

I think Ned's point (though he can certainly speak for himself) is that
if rfc3555 is allowing registrations like that, then rfc3555 needs to
be fixed, and any bogus registrations that have been made, need to be
corrected.
RFC 3555 allows media types to be defined for transport only via RTP. 
The majority of these registrations are under the audio and video 
top-level types, with a small number being under text. Is your 
objection to any media type being defined only for transport via RTP, 
or to text media types being defined only for transport via RTP?

Colin
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Re: Last Call: 'Media Type Specifications and Registration Procedures' to BCP

2005-04-12 Thread Colin Perkins
On 12 Apr 2005, at 19:45, Bruce Lilly wrote:
 Date: 2005-04-12 11:58
 From: Colin Perkins <[EMAIL PROTECTED]>

I have reviewed draft-freed-media-type-reg-03.txt, and have a number 
of
comments intended to align the registration procedures with the 
current
practice defined RFC 3555. These primarily arise due to the widespread
use
of media subtype names to identify formats that can be conveyed within
RTP.

The rules for display of text media types assume that such types are, 
to
some extent, readable without special purpose viewing software. This 
is
certainly true for most types, but some existing types have 
restrictions
on their use which are incompatible with this rule (e.g. the 
"text/t140"
type specified in RFC 2793 is a framed encoding defined only for
transfer
via RTP and cannot be directly displayed).
RFC 2793 states "COMMON" usage, not restricted, and makes no mention of
any sort of restrictions under Interoperability considerations!  RFC
2793 also says:
   Applications which use this media type:
 Text communication terminals and text conferencing tools.
The encoding considerations section of the registration limits this to 
RTP. The lack of clarity on where to specify restricted domains of 
applicability is one of the things I hope this update to the media type 
registration guidelines will fix.

Surely when some content is reassembled from the individual packets, it
can be presented without special purpose software (other than that
required for charset issues and local presentation conditions).  If
it cannot be presented, it's difficult to imagine how the media type
would be used in the stated intended applications.  If a substantial
amount (more than one MTU) of text/plain is transmitted via some
application-level protocol over TCP (rather than RTP), one of course
has to reassemble the content for the application layer for
presentation.  In what way does use of RTP instead of TCP (UDP, etc.)
at a transport protocol layer change that?
One might even ask in what sense -- at the application layer --
text/t140 is meant to be different from text/plain.  Perhaps the issue
is not so much with the registration procedure draft as with the
text/t140 registration...  or maybe RFC 2793 isn't a good example of
the issue you have in mind.
An alternative example would be draft-ietf-avt-text-red-05.txt which, 
due to the way SDP/SIP signalling works, needs to use the same top 
level media type as text/t140.

Colin
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Re: Last Call: 'Media Type Specifications and Registration Procedures' to BCP

2005-04-12 Thread Colin Perkins
On 12 Apr 2005, at 17:57, [EMAIL PROTECTED] wrote:
...
The rules for display of text media types assume that such types are, 
to
some extent, readable without special purpose viewing software. This 
is
certainly true for most types, but some existing types have 
restrictions
on their use which are incompatible with this rule (e.g. the 
"text/t140"
type specified in RFC 2793 is a framed encoding defined only for
transfer
via RTP and cannot be directly displayed). The following edit to 
Section
4.2.1 ("Text Media Types"), third paragraph, is one way to resolve 
this
inconsistency, by noting that subtypes which are defined with 
restricted
usage cannot necessarily be directly displayed:
...
NEW:

Beyond plain text, there are many formats for representing what 
might
be known as "rich text".  An interesting characteristic of many 
such
representations is that they are to some extent readable even 
without
the software that interprets them.  It is useful, then, to
distinguish them, at the highest level, from such unreadable data 
as
images, audio, or text represented in an unreadable form.  In the
absence of appropriate interpretation software, and provided the
subtype


is not defined for restricted usage, it is reasonable to show
subtypes

of "text" to the user, while it is not reasonable to do so with 
most
non
textual data.  Such formatted textual data should be represented
using
subtypes of "text".
I'm quite sympathetc to the underlying problem, but IMO this change is
unacceptable, in that in order to make it work the fact that a given 
subtype is intended for restricted usage would have to be known to the 
display agent. The whole idea of having top-level types is predicated 
on not needing this sort of exception information.
Sure, but if the display agent is unaware of the restrictions, it won't 
ever be able to receive the media data. The example I have in mind in 
"text/t140" which is UTF-8 text framed within RTP packets [RFC 2793]. 
The media type is defined only for use with RTP, and an agent that 
doesn't know how to decode RTP framing (i.e. a sequence of UDP packets 
with explicit time and ordering data) will never see the data.

Colin
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Re: Last Call: 'Media Type Specifications and Registration Procedures' to BCP

2005-04-12 Thread Colin Perkins
On 12 Apr 2005, at 18:30, [EMAIL PROTECTED] wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Behalf Of [EMAIL PROTECTED]
Sent: Tuesday, April 12, 2005 12:57 PM
To: Colin Perkins
Cc: iesg@ietf.org; ietf@ietf.org
Subject: Re: Last Call: 'Media Type Specifications and
Registration Procedures' to BCP

[snip]

I'm quite sympathetc to the underlying problem, but IMO this change 
is
unacceptable, in that in order to make it work the fact that
a given subtype is
intended for restricted usage would have to be known to the
display agent. The
whole idea of having top-level types is predicated on not
needing this sort of
exception information.

Ned, how would you reconcile the current text in your document with 
the
practice specified in RFC 3555?  It's been alleged that the documents 
are
not in alignment.
Assuming they really are out of alignment, I'd reconcile them by making
whatever changes are appropriate in a revision to RFC 3555. Changing
fundamental aspects of how media types are supposed to work and which 
vast
tracts of code depend on is just not an option.
The other changes I suggested were to align the draft with RFC 3555. 
This is a related change, primarily introduced because of the 
"text/t140 type" which is clearly textual data, but requires software 
to decode (and was registered under the RFC 3555 rules). Apologies if 
that wasn't clear.

Colin
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Re: Last Call: 'Media Type Specifications and Registration Procedures' to BCP

2005-04-12 Thread Colin Perkins
On 15 Mar 2005, at 21:25, The IESG wrote:
The IESG has received a request from an individual submitter to 
consider the following document:

- 'Media Type Specifications and Registration Procedures '
as a BCP
The IESG plans to make a decision in the next few weeks, and solicits
final comments on this action.  Please send any comments to the
iesg@ietf.org or ietf@ietf.org mailing lists by 2005-04-12.
I have reviewed draft-freed-media-type-reg-03.txt, and have a number of
comments intended to align the registration procedures with the current
practice defined RFC 3555. These primarily arise due to the widespread 
use
of media subtype names to identify formats that can be conveyed within 
RTP.

The rules for display of text media types assume that such types are, to
some extent, readable without special purpose viewing software. This is
certainly true for most types, but some existing types have restrictions
on their use which are incompatible with this rule (e.g. the "text/t140"
type specified in RFC 2793 is a framed encoding defined only for 
transfer
via RTP and cannot be directly displayed). The following edit to Section
4.2.1 ("Text Media Types"), third paragraph, is one way to resolve this
inconsistency, by noting that subtypes which are defined with restricted
usage cannot necessarily be directly displayed:

OLD:
   Beyond plain text, there are many formats for representing what might
   be known as "rich text".  An interesting characteristic of many such
   representations is that they are to some extent readable even without
   the software that interprets them.  It is useful, then, to
   distinguish them, at the highest level, from such unreadable data as
   images, audio, or text represented in an unreadable form.  In the
   absence of appropriate interpretation software, it is reasonable to
   show subtypes of "text" to the user, while it is not reasonable to do
   so with most non textual data.  Such formatted textual data should be
   represented using subtypes of "text".
NEW:
   Beyond plain text, there are many formats for representing what might
   be known as "rich text".  An interesting characteristic of many such
   representations is that they are to some extent readable even without
   the software that interprets them.  It is useful, then, to
   distinguish them, at the highest level, from such unreadable data as
   images, audio, or text represented in an unreadable form.  In the
   absence of appropriate interpretation software, and provided the 
subtype
   

   is not defined for restricted usage, it is reasonable to show 
subtypes
   
   of "text" to the user, while it is not reasonable to do so with most 
non
   textual data.  Such formatted textual data should be represented 
using
   subtypes of "text".

The addition of the "framed" encoding defined in section 4.8 is valuable
now that media types are being widely used outside the traditional email
environment. Since there are many media types defined to use RTP 
framing,
and since RFC 3555 imposes additional registration requirements on these
formats, it would be useful to reference it from within this draft. The
following change to section 4.8 is one possible such edit:

OLD:
   framed The content consists of a series of frames or packets without
  internal framing or alignment indicators.  Additional out of band
  information is needed to interpret the data properly, including
  but not necessarily limited to knowledge of the boundaries between
  successive frames.  Note that media types of this sort cannot
  simply be stored in a file or transported as a simple stream of
  octets, and therefore such media types are unsuitable for use in
  many traditional protocols.
   Additional restrictions on 7bit and 8bit are given in [RFC2046].
NEW:
   framed The content consists of a series of frames or packets without
  internal framing or alignment indicators.  Additional out of band
  information is needed to interpret the data properly, including
  but not necessarily limited to knowledge of the boundaries between
  successive frames and knowledge of the transport mechanism.  Note
   ^
  that media types of this sort cannot simply be stored in a file or
  transported as a simple stream of octets, and therefore such media
  types are unsuitable for use in many traditional protocols.
   Additional restrictions on 7bit and 8bit are given in [RFC2046].
   A commonly used transport with framed encoding is the Real-time
   ^^^
   Transport Protocol, RTP.  Additional rules for framed encodings
   ^^^
   defined for transport using RTP are given in [RFC3555].
   ^^

Re: What problems does the draft cut-off solve? (was: Re: MARID back from the grave?)

2005-03-02 Thread Colin Perkins
On 2 Mar 2005, at 12:39, Margaret Wasserman wrote:
I'd like to add-on to Spencer's point...
At 6:14 AM -0600 3/2/05, Spencer Dawkins wrote:
- Most important - we expect people to read the drafts before 
discussing them at face-to-face meetings, and thought that 
considering drafts submitted this morning didn't give working groups 
enough time to do necessary homework before having really confused 
and confusing discussions
Having the draft available two weeks in advance does very little good 
if the WGs don't have agendas posted that tell you what drafts to read 
in preparation for the meeting.

At this point, less than one week before the meeting, only 14 WGs (not 
counting BOFs) have agendas posted.  I'm at a loss for a suitable 
adjective.
You might start by asking the secretariat why all the agendas which 
have been submitted haven't been posted... I know of two working groups 
which have sent in their agenda that don't appear on the website, and I 
suspect they're not the only ones.

Colin
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Re: Internet-Draft cutoffs and getting work done

2004-10-20 Thread Colin Perkins
On 20 Oct 2004, at 09:45, Harald Tveit Alvestrand wrote:
--On onsdag, oktober 20, 2004 09:31:06 +0100 Colin Perkins 
<[EMAIL PROTECTED]> wrote:
that was what the procedure used to be - and someone had to keep 
track
of the pile of I-D submissions for which there was no response (yet)
from the WG chair. That extra load is what the secretariat has been
trying to avoid during the rush.
Can't we just require the working group chairs to send approvals 
before
the submission deadline? Much of the problem before was that there 
was no
definite cut-off date for approvals.
when was "before"?
there's been definite cut-off dates for approvals for at least 3 years 
(see the deadline announcement messages).

(I think the secretariat has accepted some late approvals, though...)
Checking back, I see you're right about the approval deadlines, 
although if I remember correctly the deadlines were a lot less firm in 
the past.

The main issue, though, is that having the approval deadline a week 
before the submission deadline causes problems, as John enumerated in 
his "rant". In an ideal world, authors would know the drafts they were 
planning to submit in plenty of time, and they'd tell the working group 
chairs so approval can be given. In practice, and despite several 
reminders sent to the working group lists, I've seen several cases 
where the early approval deadline caused drafts to be rejected. That 
doesn't help the IETF process.

Colin
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Re: Internet-Draft cutoffs and getting work done

2004-10-20 Thread Colin Perkins
On 19 Oct 2004, at 06:13, Harald Tveit Alvestrand wrote:
--On 18. oktober 2004 12:43 -0400 Michael Richardson 
<[EMAIL PROTECTED]> wrote:

  I wonder if it wouldn't just be simpler to have the WG chair
submit the -00 document themselves,
I've discussed this option with the secretariat, and they think this 
(having the WG chair submit or forward the document on behalf of the 
author was the thing suggested) is a very reasonable thing to do - for 
the spring IETF meeting.
Provided the criteria the secretariat use to check drafts for 
acceptability are clearly published, so chairs can check that drafts 
meet those criteria before accepting them for submission (and by this I 
mean something explicit like "run idnits v1.44 with no nits found and 
submit by , and we'll guarantee to post the draft").

As a chair, I don't want to be caught in the middle of an argument 
between an author and the secretariat about what constitutes correct 
boilerplate.

Colin
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Re: Internet-Draft cutoffs and getting work done

2004-10-20 Thread Colin Perkins
On 20 Oct 2004, at 06:13, Harald Tveit Alvestrand wrote:
--On tirsdag, oktober 19, 2004 18:39:49 -0700 Vijay Devarapalli 
<[EMAIL PROTECTED]> wrote:
this sometimes doesnt work. for example, I submitted a 00 version
working group draft on Oct 18 draft at 2 am (PST). I dont think
the WG chair could have stayed up that late to send out the draft
for me before the submissin deadline (6 am PST). :)
I prefer just cc'ing the WG chairs when I submit the draft,
and the WG chairs, as soon as possible, sending a mail confirming
that the document should be processed as a working group document.
can you ask the secretariat if they are okay with this? :)
that was what the procedure used to be - and someone had to keep track 
of the pile of I-D submissions for which there was no response (yet) 
from the WG chair. That extra load is what the secretariat has been 
trying to avoid during the rush.
Can't we just require the working group chairs to send approvals before 
the submission deadline? Much of the problem before was that there was 
no definite cut-off date for approvals.

Colin
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Re: Spammers using IETF meeting attendance lists

2003-09-09 Thread Colin Perkins
On Tuesday, Sep 9, 2003, at 17:48 Europe/London, Doug Royer wrote:
It means that at least one person that processed the registrations
was using a windows system that is now infected. That stupid
blaster virus then sent email with a forged From line as if it was
sent from you, then the spammers picked up that never released email
address and added it to their spam list.
Makes me wonder if the virus was build by someone wanting to find
a lot of valid email addresses that were not otherwise available.
A simpler explanation is that the proceedings on the IETF website 
include a list of attendees, with email addresses...
Colin




Re: Last Call: SDP attribute for qualifying Media Formats withGeneric Parameters to a Proposed Standard

2003-06-26 Thread Colin Perkins
--> [EMAIL PROTECTED] writes:
> > The IESG has received a request from Individual Submissions WG to
> > consider the following Internet-Draft(s) as a Proposed Standard. This
> > has been reviewed in the IETF but is not the product of an IETF Working
> > Group. o Individual Submissions (NONE): SDP attribute for qualifying
> > Media
> >   Formats with Generic Parameters
> > 
> > Proposed Standard
> 
> This looks to me like a substantive duplication of the media features
> facility defined in RFC 2533 and elaborated on in RFCs 2506, 2530, 2534,
> 2738, 2912, 2913, 2938, and 2987. Not only does this set of documents
> provide a full-featured media feature tagging and content negotiation
> facility, the relationship to MIME media types and parameters has also
> been worked out.

This draft adds a new namespace within the existing feature description
syntax (RFC 2327) and parameter negotiation (RFC 3264) model of SDP. It
is intentionally independent of MIME types, and their parameters, which
are mapped onto SDP in other ways (typically through RTP Payload Type
registrations).

> Although the media features facility defined in these various RFCs is
> perhaps too elaborate for SDP's needs, it should at least be possible to
> reuse the basic descriptive elements of media features and the
> corresponding media features registry.
> 
> At a minimum this draft needs to explain why the existing media features
> facility is unsuitable for SDP's use and another, similar facility needs
> to be defined.

That's an issue for SDP in general, which has a feature description and
negotiation mechanism incompatible with - and to large extent predating -
the RFCs you mention, but is out of scope of this draft. 

The SDPng work in MMUSIC references the RFCs you mention, and is looking to
reuse as much as possible from them. I think it's too late for SDP to adopt
them, because of the need to maintain backwards compatibility with a large
installed base.

Colin





Re: participation in IETF meetings

2001-10-24 Thread Colin Perkins

--> Aaron Falk writes:
>On Tue, Oct 23, 2001 at 11:23:06AM -0700, Paul Hoffman / IMC wrote:
>>
>> The advantage of multicast vs. tape-and-archive is the real-time
>> aspect for the viewer. However, this is rarely, rarely used. If it
>> turns out that switching from multicast to tape-and-archive can get
>> more camera operators in more rooms, that would be a win for more WGs.
>
>This is a great idea! I think I've attended about fifteen IETF's now
>and at each one I've been to many multicast sessions but I've never,
>NEVER heard a live question from a multicast viewer. Now, perhaps
>viewers have been posting questions to their repective mail lists
>(I've never seen that, either). But, it seems to me that history
>indicates the record is much more valuable than the live interaction.

It's not common, but I have seen questions passed from remote participants
to the meeting floor by the Mbone session operators, and I have had people
email me questions to be asked whilst I've been chairing a session (which
brings us back to the utility of wireless access in meeting rooms...)

But, I agree that having a recording of the session is often more useful
than the live multicast.

Cheers,
Colin




Re: presentation-prep as safety hazard

2001-03-21 Thread Colin Perkins

--> "Marshall T. Rose" writes:
>> Let me guess, I have the time to do research for you, that you are
>unwilling
>> or unable to do.  You say
>> that something is entirely anecdotal, I say that it isn't.  Do the
>research.
>
>you say the studies exist. i say i've never seen them. i'm not asking you to
>do any research. i'm asking you to prove what you say: that you've already
>seen them. there's quite a difference there.
>
>or perhaps when you say that the studies exist, you're saying you've heard
>the studies exist. i'd call that anecdotal.
>
>all i'm asking for is a url or a report number or something that can be
>researched. if someone can provide one, i'm happy to admit that i'm
>mistaken. in the interim, i think it reasonable for me to say that the
>"evidence" is all anecdotal...

Mobile phones rather than computers, but it's a starting point...
http://www.srg.caa.co.uk/news/Gsm_intf.PDF

Colin




Re: An alternative to TCP (part 1)

2001-02-06 Thread Colin Perkins

--> stanislav shalunov writes:
>Larry Foore <[EMAIL PROTECTED]> writes:
>
>> How often would a single TCP session have allocated to itself an
>> entire gigabit link?
>
>At SC2000 there was an application (non-interlaced HDTV) using
>1.5Gbps.

Sure, but it wasn't using TCP...  

Colin




Lost a secure ID card?

2000-12-12 Thread Colin Perkins

I found one after the CEOT BOF, contact me if you want it back.
Colin




Re: pay for using MPEG-4 standard

2000-08-07 Thread Colin Perkins

--> "Goel, Jiten" writes:
>Is this royalty applicable only to Audio/Video(multimedia) related
>standards or for any other standard under IETF working groups?

As I noted previously, there has been no IPR disclosed on the proposed RTP
payload formats for MPEG-4. 

Other parts of the MPEG-4 system are covered by IPR for which you may need
a license. I suggest you contact the MPEG committee for details, this has
nothing to do with the IETF.

Colin




Re: pay for using MPEG-4 standard

2000-08-07 Thread Colin Perkins

--> Jitendra Goel writes:
>Hi All,
>
>I came to know from one of my colleague that if 
>companies follows MPEG-4 standards during
>implementations then they have to pay royalty to
>*MPEG-4 Standardizing Committee* just because
>they are following the MPEG-4 standards.
>
>e.g. Supposing a company "A Corp" comes up with
>MPEG-4 solutions ( player/authoring tool/server ...)
>that are fully compliant with MPEG-4 standards.
>Does it mean, it needs to pay anything to MPEG-4 
>committee just because its products are MPEG-4 
>standard compilant.
>
>your comments are most welcome.

The MPEG-4 standard - like most other audio/video coding standards -
includes a number of features which are covered by patents and other 
IPR. It is likely that you will have to obtain a license to use that 
IPR before using (certain features of) the standard, and this might 
involve payment of royalties. You can find more information at:
http://www.cselt.it/mpeg/   MPEG home page
http://www.m4if.org/MPEG-4 industry forum

To make this discussion relevent to the IETF, I note that the audio/video
transport working group has not been notified of any IPR on the proposed
RTP payload formats for MPEG-4 media. Since this discussion includes the
joint MPEG/IETF mailing list developing such formats, I remind people that
section 10 of RFC 2026 discusses the requirements for IPR disclosure when
working on IETF standards.

Colin Perkins
IETF AVT co-chair