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In-Reply-To: <BGEKJNNFCPJKLLMMDMELAEKNFMAA.alan.d.clark@telchemy.com>
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From: Mark Baugher <mbaugher@cisco.com>
Date: Wed, 1 Dec 2004 08:54:51 -0800
To: "Alan Clark" <alan.d.clark@telchemy.com>
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Subject: [MMUSIC] Re: [AVT] RFC 3711 and RTCP
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Alan,

On Dec 1, 2004, at 4:08 AM, Alan Clark wrote:

> Mark
>
> To provide some background to the list - RFC3611 states the following:-
>
>    "Any encryption or filtering of XR report blocks entails a loss of
>    monitoring information to third parties.  For example, a network  
> that
>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>    information to the service providers traversed by the tunnel.  The
>    service providers cannot then monitor or respond to the quality of
>    the VoIP calls that they carry, potentially creating problems for  
> the
>    network's users.  As a default, XR packets should not be encrypted  
> or
>    filtered."

If this read "MUST NOT be encrypted" then we would not have a potential
ambiguity, which is a problem for the signaling.

Mark
>
> I can't conceive of any circumstance in which the contents of an XR  
> payload
> would be remotely useful to an eavesdropper and hence stating that XR
> payloads should always be left unencrypted would be (at least in my  
> opinion)
> ok.  If we wanted to make this even more specific we could narrow this
> requirement down to only XR VoIP Metrics payloads to avoid the  
> situation in
> which some future XR payload may be problematic.
>
> The availability of VoIP metrics payloads will be a huge issue for  
> service
> providers - as we move through 2005 RTCP XR will be a mainstay of VoIP
> performance management and is already a mandatory requirement within  
> certain
> sectors of the industry.  Making the VoIP metrics payload unencrypted  
> by
> default may seem a small issue today however it will have major  
> consequences
> to the industry as SRTP starts to roll out.
>
> Regards
>
> Alan
>
>
>
> -----Original Message-----
> From: Mark Baugher [mailto:mbaugher@cisco.com]
> Sent: Tuesday, November 30, 2004 6:45 PM
> To: Alan Clark
> Cc: mmusic@ietf.org
> Subject: Re: [AVT] RFC 3711 and RTCP
>
>
> Alan
>    I am copying MMUSIC (and not cross-posting to AVT) to further  
> discuss
> a possible change to SDP security descriptions, which has completed WG
> last call.  The change is needed for RFC 3611, and we also discovered
> that
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
> -07.txt does not address the SRTCP E bit properly. I offered the
> following text in a posting to AVT:
> "As the default behavior when UNENCRYPTED_SRTCP is signaled, all SRTCP
> packets MUST have the E bit set to zero and MUST NOT be encrypted.
> Otherwise, all SRTP packets MUST have the E bit set to one and MUST be
> encrypted by default.  Individual applications MAY override these
> default behaviors in their specifications.  In a session where SRTCP
> packets are encrypted, therefore, applications that use RFC 3611 MAY
> set the E bit to 0 for XR packets as recommended by RFC 3611."
>
> One of my co-authors pointed out that the last sentence is more than an
> editorial change.  We also don't like the fact that an offerer or
> answerer cannot explicitly signal when it wants the XR packet to be
> encrypted or unencrypted.  In other words, if ENCRYPTED_SRTCP is the
> default setting and RFC 3611 is also signaled, it is ambiguous as to
> whether the SRTCP sender will encrypt the XR packet or not.  It's
> solely up to the sender.  If the offerer or answerer want the XR packet
> to always be encrypted, there is no way to signal that.  Ditto for the
> unencrypted case.
>
> I think we need to put the first three sentences of the above-quoted
> paragraph into SDP security descriptions as an editorial change.  I'm
> proposing this to the working group.  I also propose not to include the
> last sentence.  Instead, we could add a new signaling parameter,
> UNENCRYPTED_XR, which implies that all SRTCP messages are to be
> encrypted except for the XR packets, which will have their E bit clear,
> of course.
>
> This new parameter could be placed in an RFC 3611bis or in a revision
> of draft-ietf-mmusic-sdescriptions-07.txt, which will then need to
> undergo another WG last call.  What do you (all) think about this?
>
> Mark
>
> On Nov 24, 2004, at 4:18 PM, Mark Baugher wrote:
>
>> Alan
>>   As Dave Oran has pointed out, the definition that you want is not
>> supported in the signaling, at present.  The current definition of
>> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
>> -07.txt treats SRTCP encryption as either on or off and does not
>> address the problem of setting the E bit in SRTCP packets.  This is a
>> bug that we need to fix.
>>
>>   MMUSIC has finished last call on
>> draft-ietf-mmusic-sdescriptions-07.txt, and the transport ADs have
>> reviewed it.  We'll be making some changes at Jon Peterson's request
>> and post another version in about a week.  This is an opportunity to
>> address the E bit, but the MMUSIC chairs or transport ADs may want
>> another WG last call.  Off the top of my head, I would offer something
>> like the following.
>>
>> "As the default behavior when UNENCRYPTED_SRTCP is signaled, all SRTCP
>> packets MUST have the E bit set to zero and MUST NOT be encrypted.
>> Otherwise, all SRTP packets MUST have the E bit set to one and MUST be
>> encrypted by default.  Individual applications MAY override these
>> default behaviors in their specifications.  In a session where SRTCP
>> packets are encrypted, therefore, applications that use RFC 3611 MAY
>> set the E bit to 0 for XR packets as recommended by RFC 3611."
>>
>> Mark
>>
>> On Nov 24, 2004, at 2:33 PM, Alan Clark wrote:
>>
>>>
>>>
>>> Mark
>>>
>>> We tried to flag this in RFC3611 with the following text in the
>>> "security
>>> considerations" section
>>>
>>>    Any encryption or filtering of XR report blocks entails a loss of
>>>    monitoring information to third parties.  For example, a network
>>> that
>>>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>>>    information to the service providers traversed by the tunnel.  The
>>>    service providers cannot then monitor or respond to the quality of
>>>    the VoIP calls that they carry, potentially creating problems for
>>> the
>>>    network's users.  As a default, XR packets should not be encrypted
>>> or
>>>    filtered.
>>>
>>> Could we potentially interpret this as being the definition of what  
>>> to
>>> selectively encrypt in the case of the RFC3611 specification?
>>>
>>> Regards
>>>
>>> Alan
>>>
>>>
>>> -----Original Message-----
>>> From: Mark Baugher [mailto:mbaugher@cisco.com]
>>> Sent: Wednesday, November 24, 2004 2:42 PM
>>> To: Alan Clark
>>> Cc: Oren Peleg; Cullen Jennings; avt@ietf.org; David R Oran; Michael
>>> A.
>>> Ramalho
>>> Subject: Re: [AVT] RFC 3711 and RTCP
>>>
>>>
>>> So, when the sender and receiver agree to encrypt SRTCP, this would
>>> mean
>>> only that some SRTCP messages would be encrypted and the sender would
>>> decide which ones to encrypt?  My personal opinion is that this is  
>>> not
>>> secure because it is unpredictable to the receiver, which may have a
>>> policy that all messages related to its session be encrypted.
>>>
>>> I can think of a few potential approaches to this problem.
>>>
>>> 1. Let individual SRTCP messages or a bundle of SRTCP messages
>>> be signaled to be unencrypted when SRTCP encryption is used.
>>>
>>> 2. Make a blanket policy that the sender determine what
>>> to selectively encrypt, as you suggest.
>>>
>>> 3. Define the policy of what to selectively encrypt in
>>> the particular specification.  In other words, a document could
>>> re-define the signaling of SRTCP encryption according to its
>>> needs.  The specification of foo, therefore, could define a new
>>> parameter such as foo_ENCRYPTED_SRTCP to be used in place of
>>> ENCRYPTED_SRTCP.  The foo document would define what gets
>>> encrypted and what does not get encrypted as per its needs.
>>>
>>> I commented on #2 already.  I don't like #1 because of its signaling
>>> complexity.  I like #3 because it limits the complexity to the
>>> specific needs of an application.  #1 would be more efficient than
>>> #3 if lots of applications had their own specific needs for
>>> selective encryption of SRTCP message.  But I don't think this is
>>> the case.
>>>
>>> Mark
>>>
>>> On Nov 24, 2004, at 10:58 AM, Alan Clark wrote:
>>>
>>>>
>>>> If the use of encryption in SRTCP is signalled by an explicit (E)  
>>>> bit
>>>> in the
>>>> SRTCP header then perhaps the sender could autonomously decide
>>>> whether
>>>> it
>>>> wants to encrypt?  Presumably since the sender is in the best
>>>> position
>>>> to
>>>> decide if the data is sensitive then this would be ok - and the
>>>> receiver is
>>>> able to just look at the E bit and hence recognize that decryption  
>>>> is
>>>> needed.
>>>>
>>>> Alan
>>>>
>>>> -----Original Message-----
>>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]On Behalf Of
>>>> Mark Baugher
>>>> Sent: Wednesday, November 24, 2004 1:33 PM
>>>> To: David R Oran
>>>> Cc: Cullen Jennings; Oren Peleg; Michael A. Ramalho; Alan Clark;
>>>> avt@ietf.org
>>>> Subject: Re: [AVT] RFC 3711 and RTCP
>>>>
>>>>
>>>>
>>>> On Nov 24, 2004, at 10:05 AM, David R Oran wrote:
>>>>
>>>>>
>>>>> On Nov 24, 2004, at 12:23 PM, Alan Clark wrote:
>>>>>
>>>>>>
>>>>>> Michael
>>>>>>
>>>>>> I don't see why this would be a problem unless you included SDES
>>>>>> and
>>>>>> (SR or
>>>>>> RR or XR) in the same RTCP packet.  SRTP (at least the protocol)  
>>>>>> is
>>>>>> apparently able to decided on a packet by packet basis whether to
>>>>>> encrypt
>>>>>> RTCP - hopefully SRTP implementations also permit this.
>>>>>>
>>>>> Sure, they can do this. But remember offer-answer. How does a
>>>>> receiver
>>>>> communicate such a granular policy to the transmitter? Right now if
>>>>> can say whether to auth or auth+encrypt RTP, and auth or
>>>>> auth+encrypt
>>>>> RTCP. There's no way to say auth+encrypt SDES but only auth SR/XR
>>>>> etc.
>>>>
>>>> Seems to me that this is a problem with SDP security descriptions.
>>>> That problem might be there because we forgot about the E bit.  This
>>>> seems very messy for the reason you mention above - granularity.  I
>>>> don't think that we want to negotiate the encryption policy for each
>>>> type SDP message, which is a problem with something like the E bit.
>>>>
>>>> Mark
>>>>>
>>>>>
>>>>>> Logically I'd send SDES on its own at the start of a call to
>>>>>> provide
>>>>>> some
>>>>>> user info, but would not send SR/RR/XR until at least 8-10 seconds
>>>>>> into the
>>>>>> call as I need some time to gather statistics to report.
>>>>> That's not exactly what the RTP specs say. RTCP timers don't have
>>>>> this
>>>>> kind of policy today. In fact, there are porposals on the table to
>>>>> use
>>>>> RTCP for early feedback to assess reception quality on some initial
>>>>> packets - e.g. the RTP-noop proposal. In any event that has no
>>>>> bearing
>>>>> on the present discussion.
>>>>>
>>>>>> After the
>>>>>> start-of-call exchange there would appear to be minimal need to
>>>>>> include
>>>>>> SDES. I'm sure we could construct some scenarios in which you  
>>>>>> "may"
>>>>>> need to
>>>>>> include SDES with SR/RR/XR however the vast majority of cases  
>>>>>> would
>>>>>> either
>>>>>> need only a single initial exchange or may not need SDES at all.
>>>>>>
>>>>> Any time you change SSRC you potentially need to send SDES.
>>>>> That said, I still don't see what bearing this has on the  
>>>>> discussion
>>>>> about whether we need the complexity of encrypting some RTCP  
>>>>> packets
>>>>> and not others for a single session.
>>>>>
>>>>> It's a tradeoff between extra complexity+less privacy versus
>>>>> allowing
>>>>> snoopers to gather RTCP stats.
>>>>>
>>>>> Dave.
>>>>>
>>>>>> Regards
>>>>>>
>>>>>> Alan
>>>>>>
>>>>>>
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: Michael A. Ramalho [mailto:mramalho@cisco.com]
>>>>>> Sent: Wednesday, November 24, 2004 9:39 AM
>>>>>> To: Alan Clark
>>>>>> Cc: Oren Peleg; avt@ietf.org
>>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>>
>>>>>>
>>>>>> At 07:54 AM 11/24/2004 -0500, Alan Clark wrote:
>>>>>>> Hi Oren
>>>>>>>
>>>>>>> Yes - encryption is intended to provide privacy however this
>>>>>>> relates
>>>>>>> primarily to the voice payloads and to information within RTCP
>>>>>>> that
>>>>>>> may be
>>>>>>> associated with individual users (e.g. SDES).
>>>>>>>
>>>>>>> SR, RR and XR contain only peformance statistics and there is no
>>>>>>> privacy
>>>>>>> consideration that I can think of related to this type of data.
>>>>>>>
>>>>>>> The message is - encrypt RTP, optionally encrypt RTCP SDES but
>>>>>>> don't
>>>>>> encrypt
>>>>>>> RTCP SR/RR/XR
>>>>>>
>>>>>> Hi Alan,
>>>>>>
>>>>>> I am wondering about implementations utilizing your advise
>>>>>> "optionally
>>>>>> encrypt
>>>>>> RTCP SDES but don't encrypt RTCP SR/RR/XR".
>>>>>>
>>>>>> Is there an easy way to tell SRTP to encrypt RTCP SDES, but not
>>>>>> RTCP
>>>>>> SR/RR/XR? It seems that all the RTCP blocks would otherwise be
>>>>>> highly
>>>>>> related to each other. For example (quote out of RFC 3611):
>>>>>>
>>>>>> "XR packets supplement the existing RTCP packets, and may be
>>>>>> stacked
>>>>>> with
>>>>>> other RTCP packets to form compound RTCP packets [9, Section 6]."
>>>>>>
>>>>>> Thus, RFC3611 explicitly admits the possibility of (but does not
>>>>>> mandate)
>>>>>> the piggybacking all the RTCP-XR block types.
>>>>>>
>>>>>> RTCP implementations would have to know the SRTP encryption rules
>>>>>> to
>>>>>> make sure they don't perform piggybacking with SDES ... but could
>>>>>> allow
>>>>>> (the more bandwidth efficient) piggybacking on the other RTCP-XR
>>>>>> blocks.
>>>>>>
>>>>>> This RTCP/SRTP interaction/dependency seems to add a burden to  
>>>>>> both
>>>>>> RTCP and SRTP implementations ... let alone the expressiveness
>>>>>> required
>>>>>> to signal your advise in existing SDP/SRTP mechanisms.
>>>>>>
>>>>>> Are you aware of any implementations that have actually followed
>>>>>> your
>>>>>> advise above? I rather go into this one with our eyes open ...
>>>>>> developers
>>>>>> highly dislike this cross-functional entity dependency (RTCP &  
>>>>>> SRTP
>>>>>> along
>>>>>> with expressing it in SDP/SRTP mechanisms).
>>>>>>
>>>>>> Regards,
>>>>>>
>>>>>> Michael Ramalho
>>>>>>
>>>>>>
>>>>>>
>>>>>>> Regards
>>>>>>>
>>>>>>> Alan
>>>>>>>
>>>>>>> -----Original Message-----
>>>>>>> From: Oren Peleg [mailto:OrenP@audiocodes.com]
>>>>>>> Sent: Wednesday, November 24, 2004 7:50 AM
>>>>>>> To: Alan Clark; avt@ietf.org
>>>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Isnt encryption is to prevent monitoring from the first place?
>>>>>>>
>>>>>>> -----Original Message-----
>>>>>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On  
>>>>>>> Behalf
>>>>>>> Of
>>>>>>> Alan Clark
>>>>>>> Sent: Tuesday, November 23, 2004 5:42 PM
>>>>>>> To: avt@ietf.org
>>>>>>> Subject: [AVT] RFC 3711 and RTCP
>>>>>>>
>>>>>>>
>>>>>>>> From the flurry of discussion on RFC3711 it looks as if there is
>>>>>>> considerable interest in SRTP implementation.  I'd like to add a
>>>>>>> request
>>>>>>> to
>>>>>>> SRTP implementors with regard to manageability.
>>>>>>>
>>>>>>> RTCP SR, RR and more particularly XR (e.g. VoIP Metrics report
>>>>>>> block)
>>>>>>> contain useful information for network monitoring, and are often
>>>>>>> used by
>>>>>>> probes and analyzers to support problem diagnosis.  RFC3711  
>>>>>>> allows
>>>>>>> the
>>>>>>> selective encryption of RTCP payloads and hence I would strongly
>>>>>>> recomment
>>>>>>> that RFC3711 implementors DO NOT encrypt RTCP SR, RR and XR
>>>>>>> payloads
>>>>>>> as
>>>>>>> this
>>>>>>> adversely impacts manageability.
>>>>>>>
>>>>>>> Regards
>>>>>>>
>>>>>>> Alan Clark
>>>>>>> Telchemy
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Audio/Video Transport Working Group
>>>>>>> avt@ietf.org
>>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>>
>>>>>>> ***************************************************************** 
>>>>>>> *
>>>>>>> **
>>>>>>> *
>>>>>>> ******
>>>>>> *
>>>>>>> ********************************
>>>>>>> This email and any files transmitted with it are confidential
>>>>>>> material.
>>>>>>> They are intended solely for the use of the designated individual
>>>>>>> or
>>>>>>> entity
>>>>>>> to whom they are addressed.
>>>>>>> If the reader of this message is not the intended recipient,
>>>>>>> you are hereby notified that any dissemination,use,
>>>>>>> distribution or copying of this communication is strictly
>>>>>>> prohibited
>>>>>>> and
>>>>>> may
>>>>>>> be unlawful.
>>>>>>>
>>>>>>> If you have received this email in error please notify
>>>>>>> postmaster@audiocodes.com and permanently delete the e-mail and
>>>>>>> files.
>>>>>>>
>>>>>>>
>>>>>>> ***************************************************************** 
>>>>>>> *
>>>>>>> **
>>>>>>> *
>>>>>>> ******
>>>>>> *
>>>>>>> *******************************
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Audio/Video Transport Working Group
>>>>>>> avt@ietf.org
>>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Audio/Video Transport Working Group
>>>>>> avt@ietf.org
>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>
>>>>> David R. Oran
>>>>> Cisco Fellow
>>>>> Cisco Systems
>>>>> 7 Ladyslipper Lane
>>>>> Acton, MA 01720 USA
>>>>> Tel: +1 978 264 2048
>>>>> Email: oran@cisco.com
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Audio/Video Transport Working Group
>>>>> avt@ietf.org
>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Audio/Video Transport Working Group
>>>> avt@ietf.org
>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>
>>>
>>
>


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From: "Alan Clark" <alan.d.clark@telchemy.com>
To: "Mark Baugher" <mbaugher@cisco.com>
Date: Wed, 1 Dec 2004 07:08:00 -0500
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Subject: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
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Mark

To provide some background to the list - RFC3611 states the following:-

   "Any encryption or filtering of XR report blocks entails a loss of
   monitoring information to third parties.  For example, a network that
   establishes a tunnel to encrypt VoIP Report Blocks denies that
   information to the service providers traversed by the tunnel.  The
   service providers cannot then monitor or respond to the quality of
   the VoIP calls that they carry, potentially creating problems for the
   network's users.  As a default, XR packets should not be encrypted or
   filtered."

I can't conceive of any circumstance in which the contents of an XR payload
would be remotely useful to an eavesdropper and hence stating that XR
payloads should always be left unencrypted would be (at least in my opinion)
ok.  If we wanted to make this even more specific we could narrow this
requirement down to only XR VoIP Metrics payloads to avoid the situation in
which some future XR payload may be problematic.

The availability of VoIP metrics payloads will be a huge issue for service
providers - as we move through 2005 RTCP XR will be a mainstay of VoIP
performance management and is already a mandatory requirement within certain
sectors of the industry.  Making the VoIP metrics payload unencrypted by
default may seem a small issue today however it will have major consequences
to the industry as SRTP starts to roll out.

Regards

Alan



-----Original Message-----
From: Mark Baugher [mailto:mbaugher@cisco.com]
Sent: Tuesday, November 30, 2004 6:45 PM
To: Alan Clark
Cc: mmusic@ietf.org
Subject: Re: [AVT] RFC 3711 and RTCP


Alan
   I am copying MMUSIC (and not cross-posting to AVT) to further discuss
a possible change to SDP security descriptions, which has completed WG
last call.  The change is needed for RFC 3611, and we also discovered
that
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
-07.txt does not address the SRTCP E bit properly. I offered the
following text in a posting to AVT:
"As the default behavior when UNENCRYPTED_SRTCP is signaled, all SRTCP
packets MUST have the E bit set to zero and MUST NOT be encrypted.
Otherwise, all SRTP packets MUST have the E bit set to one and MUST be
encrypted by default.  Individual applications MAY override these
default behaviors in their specifications.  In a session where SRTCP
packets are encrypted, therefore, applications that use RFC 3611 MAY
set the E bit to 0 for XR packets as recommended by RFC 3611."

One of my co-authors pointed out that the last sentence is more than an
editorial change.  We also don't like the fact that an offerer or
answerer cannot explicitly signal when it wants the XR packet to be
encrypted or unencrypted.  In other words, if ENCRYPTED_SRTCP is the
default setting and RFC 3611 is also signaled, it is ambiguous as to
whether the SRTCP sender will encrypt the XR packet or not.  It's
solely up to the sender.  If the offerer or answerer want the XR packet
to always be encrypted, there is no way to signal that.  Ditto for the
unencrypted case.

I think we need to put the first three sentences of the above-quoted
paragraph into SDP security descriptions as an editorial change.  I'm
proposing this to the working group.  I also propose not to include the
last sentence.  Instead, we could add a new signaling parameter,
UNENCRYPTED_XR, which implies that all SRTCP messages are to be
encrypted except for the XR packets, which will have their E bit clear,
of course.

This new parameter could be placed in an RFC 3611bis or in a revision
of draft-ietf-mmusic-sdescriptions-07.txt, which will then need to
undergo another WG last call.  What do you (all) think about this?

Mark

On Nov 24, 2004, at 4:18 PM, Mark Baugher wrote:

> Alan
>   As Dave Oran has pointed out, the definition that you want is not
> supported in the signaling, at present.  The current definition of
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
> -07.txt treats SRTCP encryption as either on or off and does not
> address the problem of setting the E bit in SRTCP packets.  This is a
> bug that we need to fix.
>
>   MMUSIC has finished last call on
> draft-ietf-mmusic-sdescriptions-07.txt, and the transport ADs have
> reviewed it.  We'll be making some changes at Jon Peterson's request
> and post another version in about a week.  This is an opportunity to
> address the E bit, but the MMUSIC chairs or transport ADs may want
> another WG last call.  Off the top of my head, I would offer something
> like the following.
>
> "As the default behavior when UNENCRYPTED_SRTCP is signaled, all SRTCP
> packets MUST have the E bit set to zero and MUST NOT be encrypted.
> Otherwise, all SRTP packets MUST have the E bit set to one and MUST be
> encrypted by default.  Individual applications MAY override these
> default behaviors in their specifications.  In a session where SRTCP
> packets are encrypted, therefore, applications that use RFC 3611 MAY
> set the E bit to 0 for XR packets as recommended by RFC 3611."
>
> Mark
>
> On Nov 24, 2004, at 2:33 PM, Alan Clark wrote:
>
>>
>>
>> Mark
>>
>> We tried to flag this in RFC3611 with the following text in the
>> "security
>> considerations" section
>>
>>    Any encryption or filtering of XR report blocks entails a loss of
>>    monitoring information to third parties.  For example, a network
>> that
>>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>>    information to the service providers traversed by the tunnel.  The
>>    service providers cannot then monitor or respond to the quality of
>>    the VoIP calls that they carry, potentially creating problems for
>> the
>>    network's users.  As a default, XR packets should not be encrypted
>> or
>>    filtered.
>>
>> Could we potentially interpret this as being the definition of what to
>> selectively encrypt in the case of the RFC3611 specification?
>>
>> Regards
>>
>> Alan
>>
>>
>> -----Original Message-----
>> From: Mark Baugher [mailto:mbaugher@cisco.com]
>> Sent: Wednesday, November 24, 2004 2:42 PM
>> To: Alan Clark
>> Cc: Oren Peleg; Cullen Jennings; avt@ietf.org; David R Oran; Michael
>> A.
>> Ramalho
>> Subject: Re: [AVT] RFC 3711 and RTCP
>>
>>
>> So, when the sender and receiver agree to encrypt SRTCP, this would
>> mean
>> only that some SRTCP messages would be encrypted and the sender would
>> decide which ones to encrypt?  My personal opinion is that this is not
>> secure because it is unpredictable to the receiver, which may have a
>> policy that all messages related to its session be encrypted.
>>
>> I can think of a few potential approaches to this problem.
>>
>> 1. Let individual SRTCP messages or a bundle of SRTCP messages
>> be signaled to be unencrypted when SRTCP encryption is used.
>>
>> 2. Make a blanket policy that the sender determine what
>> to selectively encrypt, as you suggest.
>>
>> 3. Define the policy of what to selectively encrypt in
>> the particular specification.  In other words, a document could
>> re-define the signaling of SRTCP encryption according to its
>> needs.  The specification of foo, therefore, could define a new
>> parameter such as foo_ENCRYPTED_SRTCP to be used in place of
>> ENCRYPTED_SRTCP.  The foo document would define what gets
>> encrypted and what does not get encrypted as per its needs.
>>
>> I commented on #2 already.  I don't like #1 because of its signaling
>> complexity.  I like #3 because it limits the complexity to the
>> specific needs of an application.  #1 would be more efficient than
>> #3 if lots of applications had their own specific needs for
>> selective encryption of SRTCP message.  But I don't think this is
>> the case.
>>
>> Mark
>>
>> On Nov 24, 2004, at 10:58 AM, Alan Clark wrote:
>>
>>>
>>> If the use of encryption in SRTCP is signalled by an explicit (E) bit
>>> in the
>>> SRTCP header then perhaps the sender could autonomously decide
>>> whether
>>> it
>>> wants to encrypt?  Presumably since the sender is in the best
>>> position
>>> to
>>> decide if the data is sensitive then this would be ok - and the
>>> receiver is
>>> able to just look at the E bit and hence recognize that decryption is
>>> needed.
>>>
>>> Alan
>>>
>>> -----Original Message-----
>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]On Behalf Of
>>> Mark Baugher
>>> Sent: Wednesday, November 24, 2004 1:33 PM
>>> To: David R Oran
>>> Cc: Cullen Jennings; Oren Peleg; Michael A. Ramalho; Alan Clark;
>>> avt@ietf.org
>>> Subject: Re: [AVT] RFC 3711 and RTCP
>>>
>>>
>>>
>>> On Nov 24, 2004, at 10:05 AM, David R Oran wrote:
>>>
>>>>
>>>> On Nov 24, 2004, at 12:23 PM, Alan Clark wrote:
>>>>
>>>>>
>>>>> Michael
>>>>>
>>>>> I don't see why this would be a problem unless you included SDES
>>>>> and
>>>>> (SR or
>>>>> RR or XR) in the same RTCP packet.  SRTP (at least the protocol) is
>>>>> apparently able to decided on a packet by packet basis whether to
>>>>> encrypt
>>>>> RTCP - hopefully SRTP implementations also permit this.
>>>>>
>>>> Sure, they can do this. But remember offer-answer. How does a
>>>> receiver
>>>> communicate such a granular policy to the transmitter? Right now if
>>>> can say whether to auth or auth+encrypt RTP, and auth or
>>>> auth+encrypt
>>>> RTCP. There's no way to say auth+encrypt SDES but only auth SR/XR
>>>> etc.
>>>
>>> Seems to me that this is a problem with SDP security descriptions.
>>> That problem might be there because we forgot about the E bit.  This
>>> seems very messy for the reason you mention above - granularity.  I
>>> don't think that we want to negotiate the encryption policy for each
>>> type SDP message, which is a problem with something like the E bit.
>>>
>>> Mark
>>>>
>>>>
>>>>> Logically I'd send SDES on its own at the start of a call to
>>>>> provide
>>>>> some
>>>>> user info, but would not send SR/RR/XR until at least 8-10 seconds
>>>>> into the
>>>>> call as I need some time to gather statistics to report.
>>>> That's not exactly what the RTP specs say. RTCP timers don't have
>>>> this
>>>> kind of policy today. In fact, there are porposals on the table to
>>>> use
>>>> RTCP for early feedback to assess reception quality on some initial
>>>> packets - e.g. the RTP-noop proposal. In any event that has no
>>>> bearing
>>>> on the present discussion.
>>>>
>>>>> After the
>>>>> start-of-call exchange there would appear to be minimal need to
>>>>> include
>>>>> SDES. I'm sure we could construct some scenarios in which you "may"
>>>>> need to
>>>>> include SDES with SR/RR/XR however the vast majority of cases would
>>>>> either
>>>>> need only a single initial exchange or may not need SDES at all.
>>>>>
>>>> Any time you change SSRC you potentially need to send SDES.
>>>> That said, I still don't see what bearing this has on the discussion
>>>> about whether we need the complexity of encrypting some RTCP packets
>>>> and not others for a single session.
>>>>
>>>> It's a tradeoff between extra complexity+less privacy versus
>>>> allowing
>>>> snoopers to gather RTCP stats.
>>>>
>>>> Dave.
>>>>
>>>>> Regards
>>>>>
>>>>> Alan
>>>>>
>>>>>
>>>>>
>>>>> -----Original Message-----
>>>>> From: Michael A. Ramalho [mailto:mramalho@cisco.com]
>>>>> Sent: Wednesday, November 24, 2004 9:39 AM
>>>>> To: Alan Clark
>>>>> Cc: Oren Peleg; avt@ietf.org
>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>
>>>>>
>>>>> At 07:54 AM 11/24/2004 -0500, Alan Clark wrote:
>>>>>> Hi Oren
>>>>>>
>>>>>> Yes - encryption is intended to provide privacy however this
>>>>>> relates
>>>>>> primarily to the voice payloads and to information within RTCP
>>>>>> that
>>>>>> may be
>>>>>> associated with individual users (e.g. SDES).
>>>>>>
>>>>>> SR, RR and XR contain only peformance statistics and there is no
>>>>>> privacy
>>>>>> consideration that I can think of related to this type of data.
>>>>>>
>>>>>> The message is - encrypt RTP, optionally encrypt RTCP SDES but
>>>>>> don't
>>>>> encrypt
>>>>>> RTCP SR/RR/XR
>>>>>
>>>>> Hi Alan,
>>>>>
>>>>> I am wondering about implementations utilizing your advise
>>>>> "optionally
>>>>> encrypt
>>>>> RTCP SDES but don't encrypt RTCP SR/RR/XR".
>>>>>
>>>>> Is there an easy way to tell SRTP to encrypt RTCP SDES, but not
>>>>> RTCP
>>>>> SR/RR/XR? It seems that all the RTCP blocks would otherwise be
>>>>> highly
>>>>> related to each other. For example (quote out of RFC 3611):
>>>>>
>>>>> "XR packets supplement the existing RTCP packets, and may be
>>>>> stacked
>>>>> with
>>>>> other RTCP packets to form compound RTCP packets [9, Section 6]."
>>>>>
>>>>> Thus, RFC3611 explicitly admits the possibility of (but does not
>>>>> mandate)
>>>>> the piggybacking all the RTCP-XR block types.
>>>>>
>>>>> RTCP implementations would have to know the SRTP encryption rules
>>>>> to
>>>>> make sure they don't perform piggybacking with SDES ... but could
>>>>> allow
>>>>> (the more bandwidth efficient) piggybacking on the other RTCP-XR
>>>>> blocks.
>>>>>
>>>>> This RTCP/SRTP interaction/dependency seems to add a burden to both
>>>>> RTCP and SRTP implementations ... let alone the expressiveness
>>>>> required
>>>>> to signal your advise in existing SDP/SRTP mechanisms.
>>>>>
>>>>> Are you aware of any implementations that have actually followed
>>>>> your
>>>>> advise above? I rather go into this one with our eyes open ...
>>>>> developers
>>>>> highly dislike this cross-functional entity dependency (RTCP & SRTP
>>>>> along
>>>>> with expressing it in SDP/SRTP mechanisms).
>>>>>
>>>>> Regards,
>>>>>
>>>>> Michael Ramalho
>>>>>
>>>>>
>>>>>
>>>>>> Regards
>>>>>>
>>>>>> Alan
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: Oren Peleg [mailto:OrenP@audiocodes.com]
>>>>>> Sent: Wednesday, November 24, 2004 7:50 AM
>>>>>> To: Alan Clark; avt@ietf.org
>>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>>
>>>>>>
>>>>>>
>>>>>> Isnt encryption is to prevent monitoring from the first place?
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On Behalf
>>>>>> Of
>>>>>> Alan Clark
>>>>>> Sent: Tuesday, November 23, 2004 5:42 PM
>>>>>> To: avt@ietf.org
>>>>>> Subject: [AVT] RFC 3711 and RTCP
>>>>>>
>>>>>>
>>>>>>> From the flurry of discussion on RFC3711 it looks as if there is
>>>>>> considerable interest in SRTP implementation.  I'd like to add a
>>>>>> request
>>>>>> to
>>>>>> SRTP implementors with regard to manageability.
>>>>>>
>>>>>> RTCP SR, RR and more particularly XR (e.g. VoIP Metrics report
>>>>>> block)
>>>>>> contain useful information for network monitoring, and are often
>>>>>> used by
>>>>>> probes and analyzers to support problem diagnosis.  RFC3711 allows
>>>>>> the
>>>>>> selective encryption of RTCP payloads and hence I would strongly
>>>>>> recomment
>>>>>> that RFC3711 implementors DO NOT encrypt RTCP SR, RR and XR
>>>>>> payloads
>>>>>> as
>>>>>> this
>>>>>> adversely impacts manageability.
>>>>>>
>>>>>> Regards
>>>>>>
>>>>>> Alan Clark
>>>>>> Telchemy
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Audio/Video Transport Working Group
>>>>>> avt@ietf.org
>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>
>>>>>> ******************************************************************
>>>>>> **
>>>>>> *
>>>>>> ******
>>>>> *
>>>>>> ********************************
>>>>>> This email and any files transmitted with it are confidential
>>>>>> material.
>>>>>> They are intended solely for the use of the designated individual
>>>>>> or
>>>>>> entity
>>>>>> to whom they are addressed.
>>>>>> If the reader of this message is not the intended recipient,
>>>>>> you are hereby notified that any dissemination,use,
>>>>>> distribution or copying of this communication is strictly
>>>>>> prohibited
>>>>>> and
>>>>> may
>>>>>> be unlawful.
>>>>>>
>>>>>> If you have received this email in error please notify
>>>>>> postmaster@audiocodes.com and permanently delete the e-mail and
>>>>>> files.
>>>>>>
>>>>>>
>>>>>> ******************************************************************
>>>>>> **
>>>>>> *
>>>>>> ******
>>>>> *
>>>>>> *******************************
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Audio/Video Transport Working Group
>>>>>> avt@ietf.org
>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Audio/Video Transport Working Group
>>>>> avt@ietf.org
>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>
>>>> David R. Oran
>>>> Cisco Fellow
>>>> Cisco Systems
>>>> 7 Ladyslipper Lane
>>>> Acton, MA 01720 USA
>>>> Tel: +1 978 264 2048
>>>> Email: oran@cisco.com
>>>>
>>>>
>>>> _______________________________________________
>>>> Audio/Video Transport Working Group
>>>> avt@ietf.org
>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>
>>>
>>>
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/avt
>>>
>>
>


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From mmusic-bounces@ietf.org  Thu Dec  2 07:35:17 2004
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Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Thu, 2 Dec 2004 13:45:34 +0200
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Thread-Topic: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
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From: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
To: "Alan Clark" <alan.d.clark@telchemy.com>,
        "Mark Baugher" <mbaugher@cisco.com>
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I would be cautious in saying that in no circumstances the contents of =
an XR payload would be remotely useful to an eavesdropper. We may =
encounter some very security aware environments where even the fact that =
a call happens between two IP phones, its duration, or QoS parameters =
need to be kept confidential. For this reason, while working to provide =
a strong recommendation for RTCP XR packets not to be encrypted, I do =
not believe that we can make this a MUST requirements. As an =
alternative, I suggest to use  RAQMON extensions to report the RTCP XR =
metrics to RAQMON collectors without the security risks that may be of =
concern in some environments.=20

Regards,

Dan



> -----Original Message-----
> From: mmusic-bounces@ietf.org=20
> [mailto:mmusic-bounces@ietf.org]On Behalf Of Alan Clark
> Sent: 01 December, 2004 2:08 PM
> To: Mark Baugher
> Cc: mmusic@ietf.org
> Subject: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>=20
>=20
> Mark
>=20
> To provide some background to the list - RFC3611 states the=20
> following:-
>=20
>    "Any encryption or filtering of XR report blocks entails a loss of
>    monitoring information to third parties.  For example, a=20
> network that
>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>    information to the service providers traversed by the tunnel.  The
>    service providers cannot then monitor or respond to the quality of
>    the VoIP calls that they carry, potentially creating=20
> problems for the
>    network's users.  As a default, XR packets should not be=20
> encrypted or
>    filtered."
>=20
> I can't conceive of any circumstance in which the contents of=20
> an XR payload
> would be remotely useful to an eavesdropper and hence stating that XR
> payloads should always be left unencrypted would be (at least=20
> in my opinion)
> ok.  If we wanted to make this even more specific we could narrow this
> requirement down to only XR VoIP Metrics payloads to avoid=20
> the situation in
> which some future XR payload may be problematic.
>=20
> The availability of VoIP metrics payloads will be a huge=20
> issue for service
> providers - as we move through 2005 RTCP XR will be a mainstay of VoIP
> performance management and is already a mandatory requirement=20
> within certain
> sectors of the industry.  Making the VoIP metrics payload=20
> unencrypted by
> default may seem a small issue today however it will have=20
> major consequences
> to the industry as SRTP starts to roll out.
>=20
> Regards
>=20
> Alan
>=20
>=20
>=20
> -----Original Message-----
> From: Mark Baugher [mailto:mbaugher@cisco.com]
> Sent: Tuesday, November 30, 2004 6:45 PM
> To: Alan Clark
> Cc: mmusic@ietf.org
> Subject: Re: [AVT] RFC 3711 and RTCP
>=20
>=20
> Alan
>    I am copying MMUSIC (and not cross-posting to AVT) to=20
> further discuss
> a possible change to SDP security descriptions, which has completed WG
> last call.  The change is needed for RFC 3611, and we also discovered
> that
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
> -07.txt does not address the SRTCP E bit properly. I offered the
> following text in a posting to AVT:
> "As the default behavior when UNENCRYPTED_SRTCP is signaled, all SRTCP
> packets MUST have the E bit set to zero and MUST NOT be encrypted.
> Otherwise, all SRTP packets MUST have the E bit set to one and MUST be
> encrypted by default.  Individual applications MAY override these
> default behaviors in their specifications.  In a session where SRTCP
> packets are encrypted, therefore, applications that use RFC 3611 MAY
> set the E bit to 0 for XR packets as recommended by RFC 3611."
>=20
> One of my co-authors pointed out that the last sentence is=20
> more than an
> editorial change.  We also don't like the fact that an offerer or
> answerer cannot explicitly signal when it wants the XR packet to be
> encrypted or unencrypted.  In other words, if ENCRYPTED_SRTCP is the
> default setting and RFC 3611 is also signaled, it is ambiguous as to
> whether the SRTCP sender will encrypt the XR packet or not.  It's
> solely up to the sender.  If the offerer or answerer want the=20
> XR packet
> to always be encrypted, there is no way to signal that.  Ditto for the
> unencrypted case.
>=20
> I think we need to put the first three sentences of the above-quoted
> paragraph into SDP security descriptions as an editorial change.  I'm
> proposing this to the working group.  I also propose not to=20
> include the
> last sentence.  Instead, we could add a new signaling parameter,
> UNENCRYPTED_XR, which implies that all SRTCP messages are to be
> encrypted except for the XR packets, which will have their E=20
> bit clear,
> of course.
>=20
> This new parameter could be placed in an RFC 3611bis or in a revision
> of draft-ietf-mmusic-sdescriptions-07.txt, which will then need to
> undergo another WG last call.  What do you (all) think about this?
>=20
> Mark
>=20
> On Nov 24, 2004, at 4:18 PM, Mark Baugher wrote:
>=20
> > Alan
> >   As Dave Oran has pointed out, the definition that you want is not
> > supported in the signaling, at present.  The current definition of
> > http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
> > -07.txt treats SRTCP encryption as either on or off and does not
> > address the problem of setting the E bit in SRTCP packets. =20
> This is a
> > bug that we need to fix.
> >
> >   MMUSIC has finished last call on
> > draft-ietf-mmusic-sdescriptions-07.txt, and the transport ADs have
> > reviewed it.  We'll be making some changes at Jon Peterson's request
> > and post another version in about a week.  This is an opportunity to
> > address the E bit, but the MMUSIC chairs or transport ADs may want
> > another WG last call.  Off the top of my head, I would=20
> offer something
> > like the following.
> >
> > "As the default behavior when UNENCRYPTED_SRTCP is=20
> signaled, all SRTCP
> > packets MUST have the E bit set to zero and MUST NOT be encrypted.
> > Otherwise, all SRTP packets MUST have the E bit set to one=20
> and MUST be
> > encrypted by default.  Individual applications MAY override these
> > default behaviors in their specifications.  In a session where SRTCP
> > packets are encrypted, therefore, applications that use RFC 3611 MAY
> > set the E bit to 0 for XR packets as recommended by RFC 3611."
> >
> > Mark
> >
> > On Nov 24, 2004, at 2:33 PM, Alan Clark wrote:
> >
> >>
> >>
> >> Mark
> >>
> >> We tried to flag this in RFC3611 with the following text in the
> >> "security
> >> considerations" section
> >>
> >>    Any encryption or filtering of XR report blocks entails=20
> a loss of
> >>    monitoring information to third parties.  For example, a network
> >> that
> >>    establishes a tunnel to encrypt VoIP Report Blocks denies that
> >>    information to the service providers traversed by the=20
> tunnel.  The
> >>    service providers cannot then monitor or respond to the=20
> quality of
> >>    the VoIP calls that they carry, potentially creating=20
> problems for
> >> the
> >>    network's users.  As a default, XR packets should not=20
> be encrypted
> >> or
> >>    filtered.
> >>
> >> Could we potentially interpret this as being the=20
> definition of what to
> >> selectively encrypt in the case of the RFC3611 specification?
> >>
> >> Regards
> >>
> >> Alan
> >>
> >>
> >> -----Original Message-----
> >> From: Mark Baugher [mailto:mbaugher@cisco.com]
> >> Sent: Wednesday, November 24, 2004 2:42 PM
> >> To: Alan Clark
> >> Cc: Oren Peleg; Cullen Jennings; avt@ietf.org; David R=20
> Oran; Michael
> >> A.
> >> Ramalho
> >> Subject: Re: [AVT] RFC 3711 and RTCP
> >>
> >>
> >> So, when the sender and receiver agree to encrypt SRTCP, this would
> >> mean
> >> only that some SRTCP messages would be encrypted and the=20
> sender would
> >> decide which ones to encrypt?  My personal opinion is that=20
> this is not
> >> secure because it is unpredictable to the receiver, which=20
> may have a
> >> policy that all messages related to its session be encrypted.
> >>
> >> I can think of a few potential approaches to this problem.
> >>
> >> 1. Let individual SRTCP messages or a bundle of SRTCP messages
> >> be signaled to be unencrypted when SRTCP encryption is used.
> >>
> >> 2. Make a blanket policy that the sender determine what
> >> to selectively encrypt, as you suggest.
> >>
> >> 3. Define the policy of what to selectively encrypt in
> >> the particular specification.  In other words, a document could
> >> re-define the signaling of SRTCP encryption according to its
> >> needs.  The specification of foo, therefore, could define a new
> >> parameter such as foo_ENCRYPTED_SRTCP to be used in place of
> >> ENCRYPTED_SRTCP.  The foo document would define what gets
> >> encrypted and what does not get encrypted as per its needs.
> >>
> >> I commented on #2 already.  I don't like #1 because of its=20
> signaling
> >> complexity.  I like #3 because it limits the complexity to the
> >> specific needs of an application.  #1 would be more efficient than
> >> #3 if lots of applications had their own specific needs for
> >> selective encryption of SRTCP message.  But I don't think this is
> >> the case.
> >>
> >> Mark
> >>
> >> On Nov 24, 2004, at 10:58 AM, Alan Clark wrote:
> >>
> >>>
> >>> If the use of encryption in SRTCP is signalled by an=20
> explicit (E) bit
> >>> in the
> >>> SRTCP header then perhaps the sender could autonomously decide
> >>> whether
> >>> it
> >>> wants to encrypt?  Presumably since the sender is in the best
> >>> position
> >>> to
> >>> decide if the data is sensitive then this would be ok - and the
> >>> receiver is
> >>> able to just look at the E bit and hence recognize that=20
> decryption is
> >>> needed.
> >>>
> >>> Alan
> >>>
> >>> -----Original Message-----
> >>> From: avt-bounces@ietf.org=20
> [mailto:avt-bounces@ietf.org]On Behalf Of
> >>> Mark Baugher
> >>> Sent: Wednesday, November 24, 2004 1:33 PM
> >>> To: David R Oran
> >>> Cc: Cullen Jennings; Oren Peleg; Michael A. Ramalho; Alan Clark;
> >>> avt@ietf.org
> >>> Subject: Re: [AVT] RFC 3711 and RTCP
> >>>
> >>>
> >>>
> >>> On Nov 24, 2004, at 10:05 AM, David R Oran wrote:
> >>>
> >>>>
> >>>> On Nov 24, 2004, at 12:23 PM, Alan Clark wrote:
> >>>>
> >>>>>
> >>>>> Michael
> >>>>>
> >>>>> I don't see why this would be a problem unless you included SDES
> >>>>> and
> >>>>> (SR or
> >>>>> RR or XR) in the same RTCP packet.  SRTP (at least the=20
> protocol) is
> >>>>> apparently able to decided on a packet by packet basis=20
> whether to
> >>>>> encrypt
> >>>>> RTCP - hopefully SRTP implementations also permit this.
> >>>>>
> >>>> Sure, they can do this. But remember offer-answer. How does a
> >>>> receiver
> >>>> communicate such a granular policy to the transmitter?=20
> Right now if
> >>>> can say whether to auth or auth+encrypt RTP, and auth or
> >>>> auth+encrypt
> >>>> RTCP. There's no way to say auth+encrypt SDES but only auth SR/XR
> >>>> etc.
> >>>
> >>> Seems to me that this is a problem with SDP security descriptions.
> >>> That problem might be there because we forgot about the E=20
> bit.  This
> >>> seems very messy for the reason you mention above -=20
> granularity.  I
> >>> don't think that we want to negotiate the encryption=20
> policy for each
> >>> type SDP message, which is a problem with something like=20
> the E bit.
> >>>
> >>> Mark
> >>>>
> >>>>
> >>>>> Logically I'd send SDES on its own at the start of a call to
> >>>>> provide
> >>>>> some
> >>>>> user info, but would not send SR/RR/XR until at least=20
> 8-10 seconds
> >>>>> into the
> >>>>> call as I need some time to gather statistics to report.
> >>>> That's not exactly what the RTP specs say. RTCP timers don't have
> >>>> this
> >>>> kind of policy today. In fact, there are porposals on=20
> the table to
> >>>> use
> >>>> RTCP for early feedback to assess reception quality on=20
> some initial
> >>>> packets - e.g. the RTP-noop proposal. In any event that has no
> >>>> bearing
> >>>> on the present discussion.
> >>>>
> >>>>> After the
> >>>>> start-of-call exchange there would appear to be minimal need to
> >>>>> include
> >>>>> SDES. I'm sure we could construct some scenarios in=20
> which you "may"
> >>>>> need to
> >>>>> include SDES with SR/RR/XR however the vast majority of=20
> cases would
> >>>>> either
> >>>>> need only a single initial exchange or may not need SDES at all.
> >>>>>
> >>>> Any time you change SSRC you potentially need to send SDES.
> >>>> That said, I still don't see what bearing this has on=20
> the discussion
> >>>> about whether we need the complexity of encrypting some=20
> RTCP packets
> >>>> and not others for a single session.
> >>>>
> >>>> It's a tradeoff between extra complexity+less privacy versus
> >>>> allowing
> >>>> snoopers to gather RTCP stats.
> >>>>
> >>>> Dave.
> >>>>
> >>>>> Regards
> >>>>>
> >>>>> Alan
> >>>>>
> >>>>>
> >>>>>
> >>>>> -----Original Message-----
> >>>>> From: Michael A. Ramalho [mailto:mramalho@cisco.com]
> >>>>> Sent: Wednesday, November 24, 2004 9:39 AM
> >>>>> To: Alan Clark
> >>>>> Cc: Oren Peleg; avt@ietf.org
> >>>>> Subject: RE: [AVT] RFC 3711 and RTCP
> >>>>>
> >>>>>
> >>>>> At 07:54 AM 11/24/2004 -0500, Alan Clark wrote:
> >>>>>> Hi Oren
> >>>>>>
> >>>>>> Yes - encryption is intended to provide privacy however this
> >>>>>> relates
> >>>>>> primarily to the voice payloads and to information within RTCP
> >>>>>> that
> >>>>>> may be
> >>>>>> associated with individual users (e.g. SDES).
> >>>>>>
> >>>>>> SR, RR and XR contain only peformance statistics and=20
> there is no
> >>>>>> privacy
> >>>>>> consideration that I can think of related to this type of data.
> >>>>>>
> >>>>>> The message is - encrypt RTP, optionally encrypt RTCP SDES but
> >>>>>> don't
> >>>>> encrypt
> >>>>>> RTCP SR/RR/XR
> >>>>>
> >>>>> Hi Alan,
> >>>>>
> >>>>> I am wondering about implementations utilizing your advise
> >>>>> "optionally
> >>>>> encrypt
> >>>>> RTCP SDES but don't encrypt RTCP SR/RR/XR".
> >>>>>
> >>>>> Is there an easy way to tell SRTP to encrypt RTCP SDES, but not
> >>>>> RTCP
> >>>>> SR/RR/XR? It seems that all the RTCP blocks would otherwise be
> >>>>> highly
> >>>>> related to each other. For example (quote out of RFC 3611):
> >>>>>
> >>>>> "XR packets supplement the existing RTCP packets, and may be
> >>>>> stacked
> >>>>> with
> >>>>> other RTCP packets to form compound RTCP packets [9,=20
> Section 6]."
> >>>>>
> >>>>> Thus, RFC3611 explicitly admits the possibility of (but does not
> >>>>> mandate)
> >>>>> the piggybacking all the RTCP-XR block types.
> >>>>>
> >>>>> RTCP implementations would have to know the SRTP=20
> encryption rules
> >>>>> to
> >>>>> make sure they don't perform piggybacking with SDES ...=20
> but could
> >>>>> allow
> >>>>> (the more bandwidth efficient) piggybacking on the other RTCP-XR
> >>>>> blocks.
> >>>>>
> >>>>> This RTCP/SRTP interaction/dependency seems to add a=20
> burden to both
> >>>>> RTCP and SRTP implementations ... let alone the expressiveness
> >>>>> required
> >>>>> to signal your advise in existing SDP/SRTP mechanisms.
> >>>>>
> >>>>> Are you aware of any implementations that have actually followed
> >>>>> your
> >>>>> advise above? I rather go into this one with our eyes open ...
> >>>>> developers
> >>>>> highly dislike this cross-functional entity dependency=20
> (RTCP & SRTP
> >>>>> along
> >>>>> with expressing it in SDP/SRTP mechanisms).
> >>>>>
> >>>>> Regards,
> >>>>>
> >>>>> Michael Ramalho
> >>>>>
> >>>>>
> >>>>>
> >>>>>> Regards
> >>>>>>
> >>>>>> Alan
> >>>>>>
> >>>>>> -----Original Message-----
> >>>>>> From: Oren Peleg [mailto:OrenP@audiocodes.com]
> >>>>>> Sent: Wednesday, November 24, 2004 7:50 AM
> >>>>>> To: Alan Clark; avt@ietf.org
> >>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> Isnt encryption is to prevent monitoring from the first place?
> >>>>>>
> >>>>>> -----Original Message-----
> >>>>>> From: avt-bounces@ietf.org=20
> [mailto:avt-bounces@ietf.org] On Behalf
> >>>>>> Of
> >>>>>> Alan Clark
> >>>>>> Sent: Tuesday, November 23, 2004 5:42 PM
> >>>>>> To: avt@ietf.org
> >>>>>> Subject: [AVT] RFC 3711 and RTCP
> >>>>>>
> >>>>>>
> >>>>>>> From the flurry of discussion on RFC3711 it looks as=20
> if there is
> >>>>>> considerable interest in SRTP implementation.  I'd=20
> like to add a
> >>>>>> request
> >>>>>> to
> >>>>>> SRTP implementors with regard to manageability.
> >>>>>>
> >>>>>> RTCP SR, RR and more particularly XR (e.g. VoIP Metrics report
> >>>>>> block)
> >>>>>> contain useful information for network monitoring, and=20
> are often
> >>>>>> used by
> >>>>>> probes and analyzers to support problem diagnosis. =20
> RFC3711 allows
> >>>>>> the
> >>>>>> selective encryption of RTCP payloads and hence I=20
> would strongly
> >>>>>> recomment
> >>>>>> that RFC3711 implementors DO NOT encrypt RTCP SR, RR and XR
> >>>>>> payloads
> >>>>>> as
> >>>>>> this
> >>>>>> adversely impacts manageability.
> >>>>>>
> >>>>>> Regards
> >>>>>>
> >>>>>> Alan Clark
> >>>>>> Telchemy
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> Audio/Video Transport Working Group
> >>>>>> avt@ietf.org
> >>>>>> https://www1.ietf.org/mailman/listinfo/avt
> >>>>>>
> >>>>>>=20
> ******************************************************************
> >>>>>> **
> >>>>>> *
> >>>>>> ******
> >>>>> *
> >>>>>> ********************************
> >>>>>> This email and any files transmitted with it are confidential
> >>>>>> material.
> >>>>>> They are intended solely for the use of the designated=20
> individual
> >>>>>> or
> >>>>>> entity
> >>>>>> to whom they are addressed.
> >>>>>> If the reader of this message is not the intended recipient,
> >>>>>> you are hereby notified that any dissemination,use,
> >>>>>> distribution or copying of this communication is strictly
> >>>>>> prohibited
> >>>>>> and
> >>>>> may
> >>>>>> be unlawful.
> >>>>>>
> >>>>>> If you have received this email in error please notify
> >>>>>> postmaster@audiocodes.com and permanently delete the e-mail and
> >>>>>> files.
> >>>>>>
> >>>>>>
> >>>>>>=20
> ******************************************************************
> >>>>>> **
> >>>>>> *
> >>>>>> ******
> >>>>> *
> >>>>>> *******************************
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> Audio/Video Transport Working Group
> >>>>>> avt@ietf.org
> >>>>>> https://www1.ietf.org/mailman/listinfo/avt
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> Audio/Video Transport Working Group
> >>>>> avt@ietf.org
> >>>>> https://www1.ietf.org/mailman/listinfo/avt
> >>>>>
> >>>> David R. Oran
> >>>> Cisco Fellow
> >>>> Cisco Systems
> >>>> 7 Ladyslipper Lane
> >>>> Acton, MA 01720 USA
> >>>> Tel: +1 978 264 2048
> >>>> Email: oran@cisco.com
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> Audio/Video Transport Working Group
> >>>> avt@ietf.org
> >>>> https://www1.ietf.org/mailman/listinfo/avt
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> Audio/Video Transport Working Group
> >>> avt@ietf.org
> >>> https://www1.ietf.org/mailman/listinfo/avt
> >>>
> >>
> >
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
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Thread-Topic: RFC 3312: Setting of send and recv tags for send/receive only
	media stream
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Subject: [MMUSIC] RFC 3312: Setting of send and recv tags for send/receive
	only media stream
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Hello,

I have a question on interpretation of the precondition RFC 3312.

The questions is on the following paragraph in RFC 3312:=09
<snip>The values of the tags "send", "recv", "local" and "remote" =
represent the point of view of the entity generating the SDP =
description. In an offer, "send" is the direction offerer->answerer and =
"local" is the offerer's access network. In an answer, "send" is the =
direction answerer->offerer and "local" is the answerer's access =
network.<snap>=20
How have the send/recv tags of preconditions to be set when a sendonly / =
receiveonly media stream from A to B is to be established? My =
understanding is as follows:

	INVITE 	(from A, who will be sending the stream)
	a=3Dcurr:qos local none
	a=3Dcurr:qos remote none
	a=3Ddes:qos mandatory local send
	a=3Ddes:qos none remote recv

	183 (from B, who will be receiving the stream)
	a=3Dcurr:qos remote none
 	a=3Dcurr:qos local none
 	a=3Ddes:qos mandatory remote send
 	a=3Ddes:qos mandatory local recv
 	a=3Dconf:qos remote send

Now the question was whether in the fourth line of the INVITE, it should =
be
	a=3Ddes:qos none remote recv
or
	a=3Ddes:qos none remote send

And the same question would apply to the third line of the 183, should =
it be=20
 	a=3Ddes:qos mandatory remote send
or
 	a=3Ddes:qos mandatory remote recv

-----------------------------------
http://www.imsbook.com
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From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Thu, 2 Dec 2004 09:02:33 -0800
To: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
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Dan
   I think use of the E bit needs to be explicitly signaled the SDP 
security descriptions.  That way, an offerer or answerer can express 
whether it wants to have all SRTCP messages encrypted or allow some to 
be in the clear.  I don't know if we need to call out the specific 
SRTCP messages that may be left in the clear or do make it "general 
purpose."  A general-purpose approach would be to signal that 
ENCRYPTED_SRTP can be overridden for any packet for which the sender 
chooses to clear the E bit.  The general-purpose mechanism might be 
suitable for going into draft-ietf-mmusic-sdescriptions-08.txt.  If we 
want to explicitly signal that the E bit can be clear just for XR 
packets when UNENCRYPTED_SRTCP is set, then I think that belongs in RFC 
3611bis.

Mark
On Dec 2, 2004, at 3:45 AM, Romascanu, Dan ((Dan)) wrote:

> I would be cautious in saying that in no circumstances the contents of 
> an XR payload would be remotely useful to an eavesdropper. We may 
> encounter some very security aware environments where even the fact 
> that a call happens between two IP phones, its duration, or QoS 
> parameters need to be kept confidential. For this reason, while 
> working to provide a strong recommendation for RTCP XR packets not to 
> be encrypted, I do not believe that we can make this a MUST 
> requirements. As an alternative, I suggest to use  RAQMON extensions 
> to report the RTCP XR metrics to RAQMON collectors without the 
> security risks that may be of concern in some environments.
>
> Regards,
>
> Dan
>
>
>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org
>> [mailto:mmusic-bounces@ietf.org]On Behalf Of Alan Clark
>> Sent: 01 December, 2004 2:08 PM
>> To: Mark Baugher
>> Cc: mmusic@ietf.org
>> Subject: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>>
>>
>> Mark
>>
>> To provide some background to the list - RFC3611 states the
>> following:-
>>
>>    "Any encryption or filtering of XR report blocks entails a loss of
>>    monitoring information to third parties.  For example, a
>> network that
>>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>>    information to the service providers traversed by the tunnel.  The
>>    service providers cannot then monitor or respond to the quality of
>>    the VoIP calls that they carry, potentially creating
>> problems for the
>>    network's users.  As a default, XR packets should not be
>> encrypted or
>>    filtered."
>>
>> I can't conceive of any circumstance in which the contents of
>> an XR payload
>> would be remotely useful to an eavesdropper and hence stating that XR
>> payloads should always be left unencrypted would be (at least
>> in my opinion)
>> ok.  If we wanted to make this even more specific we could narrow this
>> requirement down to only XR VoIP Metrics payloads to avoid
>> the situation in
>> which some future XR payload may be problematic.
>>
>> The availability of VoIP metrics payloads will be a huge
>> issue for service
>> providers - as we move through 2005 RTCP XR will be a mainstay of VoIP
>> performance management and is already a mandatory requirement
>> within certain
>> sectors of the industry.  Making the VoIP metrics payload
>> unencrypted by
>> default may seem a small issue today however it will have
>> major consequences
>> to the industry as SRTP starts to roll out.
>>
>> Regards
>>
>> Alan
>>
>>
>>
>> -----Original Message-----
>> From: Mark Baugher [mailto:mbaugher@cisco.com]
>> Sent: Tuesday, November 30, 2004 6:45 PM
>> To: Alan Clark
>> Cc: mmusic@ietf.org
>> Subject: Re: [AVT] RFC 3711 and RTCP
>>
>>
>> Alan
>>    I am copying MMUSIC (and not cross-posting to AVT) to
>> further discuss
>> a possible change to SDP security descriptions, which has completed WG
>> last call.  The change is needed for RFC 3611, and we also discovered
>> that
>> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
>> -07.txt does not address the SRTCP E bit properly. I offered the
>> following text in a posting to AVT:
>> "As the default behavior when UNENCRYPTED_SRTCP is signaled, all SRTCP
>> packets MUST have the E bit set to zero and MUST NOT be encrypted.
>> Otherwise, all SRTP packets MUST have the E bit set to one and MUST be
>> encrypted by default.  Individual applications MAY override these
>> default behaviors in their specifications.  In a session where SRTCP
>> packets are encrypted, therefore, applications that use RFC 3611 MAY
>> set the E bit to 0 for XR packets as recommended by RFC 3611."
>>
>> One of my co-authors pointed out that the last sentence is
>> more than an
>> editorial change.  We also don't like the fact that an offerer or
>> answerer cannot explicitly signal when it wants the XR packet to be
>> encrypted or unencrypted.  In other words, if ENCRYPTED_SRTCP is the
>> default setting and RFC 3611 is also signaled, it is ambiguous as to
>> whether the SRTCP sender will encrypt the XR packet or not.  It's
>> solely up to the sender.  If the offerer or answerer want the
>> XR packet
>> to always be encrypted, there is no way to signal that.  Ditto for the
>> unencrypted case.
>>
>> I think we need to put the first three sentences of the above-quoted
>> paragraph into SDP security descriptions as an editorial change.  I'm
>> proposing this to the working group.  I also propose not to
>> include the
>> last sentence.  Instead, we could add a new signaling parameter,
>> UNENCRYPTED_XR, which implies that all SRTCP messages are to be
>> encrypted except for the XR packets, which will have their E
>> bit clear,
>> of course.
>>
>> This new parameter could be placed in an RFC 3611bis or in a revision
>> of draft-ietf-mmusic-sdescriptions-07.txt, which will then need to
>> undergo another WG last call.  What do you (all) think about this?
>>
>> Mark
>>
>> On Nov 24, 2004, at 4:18 PM, Mark Baugher wrote:
>>
>>> Alan
>>>   As Dave Oran has pointed out, the definition that you want is not
>>> supported in the signaling, at present.  The current definition of
>>> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
>>> -07.txt treats SRTCP encryption as either on or off and does not
>>> address the problem of setting the E bit in SRTCP packets.
>> This is a
>>> bug that we need to fix.
>>>
>>>   MMUSIC has finished last call on
>>> draft-ietf-mmusic-sdescriptions-07.txt, and the transport ADs have
>>> reviewed it.  We'll be making some changes at Jon Peterson's request
>>> and post another version in about a week.  This is an opportunity to
>>> address the E bit, but the MMUSIC chairs or transport ADs may want
>>> another WG last call.  Off the top of my head, I would
>> offer something
>>> like the following.
>>>
>>> "As the default behavior when UNENCRYPTED_SRTCP is
>> signaled, all SRTCP
>>> packets MUST have the E bit set to zero and MUST NOT be encrypted.
>>> Otherwise, all SRTP packets MUST have the E bit set to one
>> and MUST be
>>> encrypted by default.  Individual applications MAY override these
>>> default behaviors in their specifications.  In a session where SRTCP
>>> packets are encrypted, therefore, applications that use RFC 3611 MAY
>>> set the E bit to 0 for XR packets as recommended by RFC 3611."
>>>
>>> Mark
>>>
>>> On Nov 24, 2004, at 2:33 PM, Alan Clark wrote:
>>>
>>>>
>>>>
>>>> Mark
>>>>
>>>> We tried to flag this in RFC3611 with the following text in the
>>>> "security
>>>> considerations" section
>>>>
>>>>    Any encryption or filtering of XR report blocks entails
>> a loss of
>>>>    monitoring information to third parties.  For example, a network
>>>> that
>>>>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>>>>    information to the service providers traversed by the
>> tunnel.  The
>>>>    service providers cannot then monitor or respond to the
>> quality of
>>>>    the VoIP calls that they carry, potentially creating
>> problems for
>>>> the
>>>>    network's users.  As a default, XR packets should not
>> be encrypted
>>>> or
>>>>    filtered.
>>>>
>>>> Could we potentially interpret this as being the
>> definition of what to
>>>> selectively encrypt in the case of the RFC3611 specification?
>>>>
>>>> Regards
>>>>
>>>> Alan
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: Mark Baugher [mailto:mbaugher@cisco.com]
>>>> Sent: Wednesday, November 24, 2004 2:42 PM
>>>> To: Alan Clark
>>>> Cc: Oren Peleg; Cullen Jennings; avt@ietf.org; David R
>> Oran; Michael
>>>> A.
>>>> Ramalho
>>>> Subject: Re: [AVT] RFC 3711 and RTCP
>>>>
>>>>
>>>> So, when the sender and receiver agree to encrypt SRTCP, this would
>>>> mean
>>>> only that some SRTCP messages would be encrypted and the
>> sender would
>>>> decide which ones to encrypt?  My personal opinion is that
>> this is not
>>>> secure because it is unpredictable to the receiver, which
>> may have a
>>>> policy that all messages related to its session be encrypted.
>>>>
>>>> I can think of a few potential approaches to this problem.
>>>>
>>>> 1. Let individual SRTCP messages or a bundle of SRTCP messages
>>>> be signaled to be unencrypted when SRTCP encryption is used.
>>>>
>>>> 2. Make a blanket policy that the sender determine what
>>>> to selectively encrypt, as you suggest.
>>>>
>>>> 3. Define the policy of what to selectively encrypt in
>>>> the particular specification.  In other words, a document could
>>>> re-define the signaling of SRTCP encryption according to its
>>>> needs.  The specification of foo, therefore, could define a new
>>>> parameter such as foo_ENCRYPTED_SRTCP to be used in place of
>>>> ENCRYPTED_SRTCP.  The foo document would define what gets
>>>> encrypted and what does not get encrypted as per its needs.
>>>>
>>>> I commented on #2 already.  I don't like #1 because of its
>> signaling
>>>> complexity.  I like #3 because it limits the complexity to the
>>>> specific needs of an application.  #1 would be more efficient than
>>>> #3 if lots of applications had their own specific needs for
>>>> selective encryption of SRTCP message.  But I don't think this is
>>>> the case.
>>>>
>>>> Mark
>>>>
>>>> On Nov 24, 2004, at 10:58 AM, Alan Clark wrote:
>>>>
>>>>>
>>>>> If the use of encryption in SRTCP is signalled by an
>> explicit (E) bit
>>>>> in the
>>>>> SRTCP header then perhaps the sender could autonomously decide
>>>>> whether
>>>>> it
>>>>> wants to encrypt?  Presumably since the sender is in the best
>>>>> position
>>>>> to
>>>>> decide if the data is sensitive then this would be ok - and the
>>>>> receiver is
>>>>> able to just look at the E bit and hence recognize that
>> decryption is
>>>>> needed.
>>>>>
>>>>> Alan
>>>>>
>>>>> -----Original Message-----
>>>>> From: avt-bounces@ietf.org
>> [mailto:avt-bounces@ietf.org]On Behalf Of
>>>>> Mark Baugher
>>>>> Sent: Wednesday, November 24, 2004 1:33 PM
>>>>> To: David R Oran
>>>>> Cc: Cullen Jennings; Oren Peleg; Michael A. Ramalho; Alan Clark;
>>>>> avt@ietf.org
>>>>> Subject: Re: [AVT] RFC 3711 and RTCP
>>>>>
>>>>>
>>>>>
>>>>> On Nov 24, 2004, at 10:05 AM, David R Oran wrote:
>>>>>
>>>>>>
>>>>>> On Nov 24, 2004, at 12:23 PM, Alan Clark wrote:
>>>>>>
>>>>>>>
>>>>>>> Michael
>>>>>>>
>>>>>>> I don't see why this would be a problem unless you included SDES
>>>>>>> and
>>>>>>> (SR or
>>>>>>> RR or XR) in the same RTCP packet.  SRTP (at least the
>> protocol) is
>>>>>>> apparently able to decided on a packet by packet basis
>> whether to
>>>>>>> encrypt
>>>>>>> RTCP - hopefully SRTP implementations also permit this.
>>>>>>>
>>>>>> Sure, they can do this. But remember offer-answer. How does a
>>>>>> receiver
>>>>>> communicate such a granular policy to the transmitter?
>> Right now if
>>>>>> can say whether to auth or auth+encrypt RTP, and auth or
>>>>>> auth+encrypt
>>>>>> RTCP. There's no way to say auth+encrypt SDES but only auth SR/XR
>>>>>> etc.
>>>>>
>>>>> Seems to me that this is a problem with SDP security descriptions.
>>>>> That problem might be there because we forgot about the E
>> bit.  This
>>>>> seems very messy for the reason you mention above -
>> granularity.  I
>>>>> don't think that we want to negotiate the encryption
>> policy for each
>>>>> type SDP message, which is a problem with something like
>> the E bit.
>>>>>
>>>>> Mark
>>>>>>
>>>>>>
>>>>>>> Logically I'd send SDES on its own at the start of a call to
>>>>>>> provide
>>>>>>> some
>>>>>>> user info, but would not send SR/RR/XR until at least
>> 8-10 seconds
>>>>>>> into the
>>>>>>> call as I need some time to gather statistics to report.
>>>>>> That's not exactly what the RTP specs say. RTCP timers don't have
>>>>>> this
>>>>>> kind of policy today. In fact, there are porposals on
>> the table to
>>>>>> use
>>>>>> RTCP for early feedback to assess reception quality on
>> some initial
>>>>>> packets - e.g. the RTP-noop proposal. In any event that has no
>>>>>> bearing
>>>>>> on the present discussion.
>>>>>>
>>>>>>> After the
>>>>>>> start-of-call exchange there would appear to be minimal need to
>>>>>>> include
>>>>>>> SDES. I'm sure we could construct some scenarios in
>> which you "may"
>>>>>>> need to
>>>>>>> include SDES with SR/RR/XR however the vast majority of
>> cases would
>>>>>>> either
>>>>>>> need only a single initial exchange or may not need SDES at all.
>>>>>>>
>>>>>> Any time you change SSRC you potentially need to send SDES.
>>>>>> That said, I still don't see what bearing this has on
>> the discussion
>>>>>> about whether we need the complexity of encrypting some
>> RTCP packets
>>>>>> and not others for a single session.
>>>>>>
>>>>>> It's a tradeoff between extra complexity+less privacy versus
>>>>>> allowing
>>>>>> snoopers to gather RTCP stats.
>>>>>>
>>>>>> Dave.
>>>>>>
>>>>>>> Regards
>>>>>>>
>>>>>>> Alan
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> -----Original Message-----
>>>>>>> From: Michael A. Ramalho [mailto:mramalho@cisco.com]
>>>>>>> Sent: Wednesday, November 24, 2004 9:39 AM
>>>>>>> To: Alan Clark
>>>>>>> Cc: Oren Peleg; avt@ietf.org
>>>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>>>
>>>>>>>
>>>>>>> At 07:54 AM 11/24/2004 -0500, Alan Clark wrote:
>>>>>>>> Hi Oren
>>>>>>>>
>>>>>>>> Yes - encryption is intended to provide privacy however this
>>>>>>>> relates
>>>>>>>> primarily to the voice payloads and to information within RTCP
>>>>>>>> that
>>>>>>>> may be
>>>>>>>> associated with individual users (e.g. SDES).
>>>>>>>>
>>>>>>>> SR, RR and XR contain only peformance statistics and
>> there is no
>>>>>>>> privacy
>>>>>>>> consideration that I can think of related to this type of data.
>>>>>>>>
>>>>>>>> The message is - encrypt RTP, optionally encrypt RTCP SDES but
>>>>>>>> don't
>>>>>>> encrypt
>>>>>>>> RTCP SR/RR/XR
>>>>>>>
>>>>>>> Hi Alan,
>>>>>>>
>>>>>>> I am wondering about implementations utilizing your advise
>>>>>>> "optionally
>>>>>>> encrypt
>>>>>>> RTCP SDES but don't encrypt RTCP SR/RR/XR".
>>>>>>>
>>>>>>> Is there an easy way to tell SRTP to encrypt RTCP SDES, but not
>>>>>>> RTCP
>>>>>>> SR/RR/XR? It seems that all the RTCP blocks would otherwise be
>>>>>>> highly
>>>>>>> related to each other. For example (quote out of RFC 3611):
>>>>>>>
>>>>>>> "XR packets supplement the existing RTCP packets, and may be
>>>>>>> stacked
>>>>>>> with
>>>>>>> other RTCP packets to form compound RTCP packets [9,
>> Section 6]."
>>>>>>>
>>>>>>> Thus, RFC3611 explicitly admits the possibility of (but does not
>>>>>>> mandate)
>>>>>>> the piggybacking all the RTCP-XR block types.
>>>>>>>
>>>>>>> RTCP implementations would have to know the SRTP
>> encryption rules
>>>>>>> to
>>>>>>> make sure they don't perform piggybacking with SDES ...
>> but could
>>>>>>> allow
>>>>>>> (the more bandwidth efficient) piggybacking on the other RTCP-XR
>>>>>>> blocks.
>>>>>>>
>>>>>>> This RTCP/SRTP interaction/dependency seems to add a
>> burden to both
>>>>>>> RTCP and SRTP implementations ... let alone the expressiveness
>>>>>>> required
>>>>>>> to signal your advise in existing SDP/SRTP mechanisms.
>>>>>>>
>>>>>>> Are you aware of any implementations that have actually followed
>>>>>>> your
>>>>>>> advise above? I rather go into this one with our eyes open ...
>>>>>>> developers
>>>>>>> highly dislike this cross-functional entity dependency
>> (RTCP & SRTP
>>>>>>> along
>>>>>>> with expressing it in SDP/SRTP mechanisms).
>>>>>>>
>>>>>>> Regards,
>>>>>>>
>>>>>>> Michael Ramalho
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> Regards
>>>>>>>>
>>>>>>>> Alan
>>>>>>>>
>>>>>>>> -----Original Message-----
>>>>>>>> From: Oren Peleg [mailto:OrenP@audiocodes.com]
>>>>>>>> Sent: Wednesday, November 24, 2004 7:50 AM
>>>>>>>> To: Alan Clark; avt@ietf.org
>>>>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Isnt encryption is to prevent monitoring from the first place?
>>>>>>>>
>>>>>>>> -----Original Message-----
>>>>>>>> From: avt-bounces@ietf.org
>> [mailto:avt-bounces@ietf.org] On Behalf
>>>>>>>> Of
>>>>>>>> Alan Clark
>>>>>>>> Sent: Tuesday, November 23, 2004 5:42 PM
>>>>>>>> To: avt@ietf.org
>>>>>>>> Subject: [AVT] RFC 3711 and RTCP
>>>>>>>>
>>>>>>>>
>>>>>>>>> From the flurry of discussion on RFC3711 it looks as
>> if there is
>>>>>>>> considerable interest in SRTP implementation.  I'd
>> like to add a
>>>>>>>> request
>>>>>>>> to
>>>>>>>> SRTP implementors with regard to manageability.
>>>>>>>>
>>>>>>>> RTCP SR, RR and more particularly XR (e.g. VoIP Metrics report
>>>>>>>> block)
>>>>>>>> contain useful information for network monitoring, and
>> are often
>>>>>>>> used by
>>>>>>>> probes and analyzers to support problem diagnosis.
>> RFC3711 allows
>>>>>>>> the
>>>>>>>> selective encryption of RTCP payloads and hence I
>> would strongly
>>>>>>>> recomment
>>>>>>>> that RFC3711 implementors DO NOT encrypt RTCP SR, RR and XR
>>>>>>>> payloads
>>>>>>>> as
>>>>>>>> this
>>>>>>>> adversely impacts manageability.
>>>>>>>>
>>>>>>>> Regards
>>>>>>>>
>>>>>>>> Alan Clark
>>>>>>>> Telchemy
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Audio/Video Transport Working Group
>>>>>>>> avt@ietf.org
>>>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>>>
>>>>>>>>
>> ******************************************************************
>>>>>>>> **
>>>>>>>> *
>>>>>>>> ******
>>>>>>> *
>>>>>>>> ********************************
>>>>>>>> This email and any files transmitted with it are confidential
>>>>>>>> material.
>>>>>>>> They are intended solely for the use of the designated
>> individual
>>>>>>>> or
>>>>>>>> entity
>>>>>>>> to whom they are addressed.
>>>>>>>> If the reader of this message is not the intended recipient,
>>>>>>>> you are hereby notified that any dissemination,use,
>>>>>>>> distribution or copying of this communication is strictly
>>>>>>>> prohibited
>>>>>>>> and
>>>>>>> may
>>>>>>>> be unlawful.
>>>>>>>>
>>>>>>>> If you have received this email in error please notify
>>>>>>>> postmaster@audiocodes.com and permanently delete the e-mail and
>>>>>>>> files.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>> ******************************************************************
>>>>>>>> **
>>>>>>>> *
>>>>>>>> ******
>>>>>>> *
>>>>>>>> *******************************
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Audio/Video Transport Working Group
>>>>>>>> avt@ietf.org
>>>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Audio/Video Transport Working Group
>>>>>>> avt@ietf.org
>>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>>
>>>>>> David R. Oran
>>>>>> Cisco Fellow
>>>>>> Cisco Systems
>>>>>> 7 Ladyslipper Lane
>>>>>> Acton, MA 01720 USA
>>>>>> Tel: +1 978 264 2048
>>>>>> Email: oran@cisco.com
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Audio/Video Transport Working Group
>>>>>> avt@ietf.org
>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Audio/Video Transport Working Group
>>>>> avt@ietf.org
>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>
>>>>
>>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>


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From mmusic-bounces@ietf.org  Thu Dec  2 19:17:42 2004
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From: "Alan Clark" <alan.d.clark@telchemy.com>
To: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>,
        "Mark Baugher" <mbaugher@cisco.com>
Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Thu, 2 Dec 2004 12:45:07 -0500
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Dan

I can understand your desire to create a RAQMON-centric reporting system
however the industry needs to keep its options open.  The RTCP XR based VoIP
performance management framework allows metrics to be reported via the media
path, signaling protocols or SNMP, and we need to give the implementor as
much flexibility as possible.  Reporting RTCP XR metrics to a RAQMON
collector is one option that should be available however this would not
address many practical scenarios.

Service providers and network managers make extensive use of probes and
analyzers to monitor key demarc points in the network and diagnose problems.
Encrypting everything can severely limit the functionality of these
mid-network systems and hence make it harder for the network manager to
manage the network.

If we had to trade the number of circumstances in which knowledge of packet
loss/ discard rates etc. were a security concern vs the number of
circumstances under which network managers needed improved tools to diagnose
problems then I'd be surprised if the former was even a minute fraction of
the latter.
It is also distinctly possible that if issues such as the duration of a call
was of concern then the IP endpoints would probably simply use an encrypted
IP tunnel - otherwise a mid-stream observer could examine the RTP stream.

Regards

Alan




-----Original Message-----
From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]On Behalf Of
Romascanu, Dan (Dan)
Sent: Thursday, December 02, 2004 6:46 AM
To: Alan Clark; Mark Baugher
Cc: rmonmib@ietf.org; mmusic@ietf.org; avt@ietf.org
Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP


I would be cautious in saying that in no circumstances the contents of an XR
payload would be remotely useful to an eavesdropper. We may encounter some
very security aware environments where even the fact that a call happens
between two IP phones, its duration, or QoS parameters need to be kept
confidential. For this reason, while working to provide a strong
recommendation for RTCP XR packets not to be encrypted, I do not believe
that we can make this a MUST requirements. As an alternative, I suggest to
use  RAQMON extensions to report the RTCP XR metrics to RAQMON collectors
without the security risks that may be of concern in some environments.

Regards,

Dan



> -----Original Message-----
> From: mmusic-bounces@ietf.org
> [mailto:mmusic-bounces@ietf.org]On Behalf Of Alan Clark
> Sent: 01 December, 2004 2:08 PM
> To: Mark Baugher
> Cc: mmusic@ietf.org
> Subject: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>
>
> Mark
>
> To provide some background to the list - RFC3611 states the
> following:-
>
>    "Any encryption or filtering of XR report blocks entails a loss of
>    monitoring information to third parties.  For example, a
> network that
>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>    information to the service providers traversed by the tunnel.  The
>    service providers cannot then monitor or respond to the quality of
>    the VoIP calls that they carry, potentially creating
> problems for the
>    network's users.  As a default, XR packets should not be
> encrypted or
>    filtered."
>
> I can't conceive of any circumstance in which the contents of
> an XR payload
> would be remotely useful to an eavesdropper and hence stating that XR
> payloads should always be left unencrypted would be (at least
> in my opinion)
> ok.  If we wanted to make this even more specific we could narrow this
> requirement down to only XR VoIP Metrics payloads to avoid
> the situation in
> which some future XR payload may be problematic.
>
> The availability of VoIP metrics payloads will be a huge
> issue for service
> providers - as we move through 2005 RTCP XR will be a mainstay of VoIP
> performance management and is already a mandatory requirement
> within certain
> sectors of the industry.  Making the VoIP metrics payload
> unencrypted by
> default may seem a small issue today however it will have
> major consequences
> to the industry as SRTP starts to roll out.
>
> Regards
>
> Alan
>
>
>
> -----Original Message-----
> From: Mark Baugher [mailto:mbaugher@cisco.com]
> Sent: Tuesday, November 30, 2004 6:45 PM
> To: Alan Clark
> Cc: mmusic@ietf.org
> Subject: Re: [AVT] RFC 3711 and RTCP
>
>
> Alan
>    I am copying MMUSIC (and not cross-posting to AVT) to
> further discuss
> a possible change to SDP security descriptions, which has completed WG
> last call.  The change is needed for RFC 3611, and we also discovered
> that
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
> -07.txt does not address the SRTCP E bit properly. I offered the
> following text in a posting to AVT:
> "As the default behavior when UNENCRYPTED_SRTCP is signaled, all SRTCP
> packets MUST have the E bit set to zero and MUST NOT be encrypted.
> Otherwise, all SRTP packets MUST have the E bit set to one and MUST be
> encrypted by default.  Individual applications MAY override these
> default behaviors in their specifications.  In a session where SRTCP
> packets are encrypted, therefore, applications that use RFC 3611 MAY
> set the E bit to 0 for XR packets as recommended by RFC 3611."
>
> One of my co-authors pointed out that the last sentence is
> more than an
> editorial change.  We also don't like the fact that an offerer or
> answerer cannot explicitly signal when it wants the XR packet to be
> encrypted or unencrypted.  In other words, if ENCRYPTED_SRTCP is the
> default setting and RFC 3611 is also signaled, it is ambiguous as to
> whether the SRTCP sender will encrypt the XR packet or not.  It's
> solely up to the sender.  If the offerer or answerer want the
> XR packet
> to always be encrypted, there is no way to signal that.  Ditto for the
> unencrypted case.
>
> I think we need to put the first three sentences of the above-quoted
> paragraph into SDP security descriptions as an editorial change.  I'm
> proposing this to the working group.  I also propose not to
> include the
> last sentence.  Instead, we could add a new signaling parameter,
> UNENCRYPTED_XR, which implies that all SRTCP messages are to be
> encrypted except for the XR packets, which will have their E
> bit clear,
> of course.
>
> This new parameter could be placed in an RFC 3611bis or in a revision
> of draft-ietf-mmusic-sdescriptions-07.txt, which will then need to
> undergo another WG last call.  What do you (all) think about this?
>
> Mark
>
> On Nov 24, 2004, at 4:18 PM, Mark Baugher wrote:
>
> > Alan
> >   As Dave Oran has pointed out, the definition that you want is not
> > supported in the signaling, at present.  The current definition of
> > http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
> > -07.txt treats SRTCP encryption as either on or off and does not
> > address the problem of setting the E bit in SRTCP packets.
> This is a
> > bug that we need to fix.
> >
> >   MMUSIC has finished last call on
> > draft-ietf-mmusic-sdescriptions-07.txt, and the transport ADs have
> > reviewed it.  We'll be making some changes at Jon Peterson's request
> > and post another version in about a week.  This is an opportunity to
> > address the E bit, but the MMUSIC chairs or transport ADs may want
> > another WG last call.  Off the top of my head, I would
> offer something
> > like the following.
> >
> > "As the default behavior when UNENCRYPTED_SRTCP is
> signaled, all SRTCP
> > packets MUST have the E bit set to zero and MUST NOT be encrypted.
> > Otherwise, all SRTP packets MUST have the E bit set to one
> and MUST be
> > encrypted by default.  Individual applications MAY override these
> > default behaviors in their specifications.  In a session where SRTCP
> > packets are encrypted, therefore, applications that use RFC 3611 MAY
> > set the E bit to 0 for XR packets as recommended by RFC 3611."
> >
> > Mark
> >
> > On Nov 24, 2004, at 2:33 PM, Alan Clark wrote:
> >
> >>
> >>
> >> Mark
> >>
> >> We tried to flag this in RFC3611 with the following text in the
> >> "security
> >> considerations" section
> >>
> >>    Any encryption or filtering of XR report blocks entails
> a loss of
> >>    monitoring information to third parties.  For example, a network
> >> that
> >>    establishes a tunnel to encrypt VoIP Report Blocks denies that
> >>    information to the service providers traversed by the
> tunnel.  The
> >>    service providers cannot then monitor or respond to the
> quality of
> >>    the VoIP calls that they carry, potentially creating
> problems for
> >> the
> >>    network's users.  As a default, XR packets should not
> be encrypted
> >> or
> >>    filtered.
> >>
> >> Could we potentially interpret this as being the
> definition of what to
> >> selectively encrypt in the case of the RFC3611 specification?
> >>
> >> Regards
> >>
> >> Alan
> >>
> >>
> >> -----Original Message-----
> >> From: Mark Baugher [mailto:mbaugher@cisco.com]
> >> Sent: Wednesday, November 24, 2004 2:42 PM
> >> To: Alan Clark
> >> Cc: Oren Peleg; Cullen Jennings; avt@ietf.org; David R
> Oran; Michael
> >> A.
> >> Ramalho
> >> Subject: Re: [AVT] RFC 3711 and RTCP
> >>
> >>
> >> So, when the sender and receiver agree to encrypt SRTCP, this would
> >> mean
> >> only that some SRTCP messages would be encrypted and the
> sender would
> >> decide which ones to encrypt?  My personal opinion is that
> this is not
> >> secure because it is unpredictable to the receiver, which
> may have a
> >> policy that all messages related to its session be encrypted.
> >>
> >> I can think of a few potential approaches to this problem.
> >>
> >> 1. Let individual SRTCP messages or a bundle of SRTCP messages
> >> be signaled to be unencrypted when SRTCP encryption is used.
> >>
> >> 2. Make a blanket policy that the sender determine what
> >> to selectively encrypt, as you suggest.
> >>
> >> 3. Define the policy of what to selectively encrypt in
> >> the particular specification.  In other words, a document could
> >> re-define the signaling of SRTCP encryption according to its
> >> needs.  The specification of foo, therefore, could define a new
> >> parameter such as foo_ENCRYPTED_SRTCP to be used in place of
> >> ENCRYPTED_SRTCP.  The foo document would define what gets
> >> encrypted and what does not get encrypted as per its needs.
> >>
> >> I commented on #2 already.  I don't like #1 because of its
> signaling
> >> complexity.  I like #3 because it limits the complexity to the
> >> specific needs of an application.  #1 would be more efficient than
> >> #3 if lots of applications had their own specific needs for
> >> selective encryption of SRTCP message.  But I don't think this is
> >> the case.
> >>
> >> Mark
> >>
> >> On Nov 24, 2004, at 10:58 AM, Alan Clark wrote:
> >>
> >>>
> >>> If the use of encryption in SRTCP is signalled by an
> explicit (E) bit
> >>> in the
> >>> SRTCP header then perhaps the sender could autonomously decide
> >>> whether
> >>> it
> >>> wants to encrypt?  Presumably since the sender is in the best
> >>> position
> >>> to
> >>> decide if the data is sensitive then this would be ok - and the
> >>> receiver is
> >>> able to just look at the E bit and hence recognize that
> decryption is
> >>> needed.
> >>>
> >>> Alan
> >>>
> >>> -----Original Message-----
> >>> From: avt-bounces@ietf.org
> [mailto:avt-bounces@ietf.org]On Behalf Of
> >>> Mark Baugher
> >>> Sent: Wednesday, November 24, 2004 1:33 PM
> >>> To: David R Oran
> >>> Cc: Cullen Jennings; Oren Peleg; Michael A. Ramalho; Alan Clark;
> >>> avt@ietf.org
> >>> Subject: Re: [AVT] RFC 3711 and RTCP
> >>>
> >>>
> >>>
> >>> On Nov 24, 2004, at 10:05 AM, David R Oran wrote:
> >>>
> >>>>
> >>>> On Nov 24, 2004, at 12:23 PM, Alan Clark wrote:
> >>>>
> >>>>>
> >>>>> Michael
> >>>>>
> >>>>> I don't see why this would be a problem unless you included SDES
> >>>>> and
> >>>>> (SR or
> >>>>> RR or XR) in the same RTCP packet.  SRTP (at least the
> protocol) is
> >>>>> apparently able to decided on a packet by packet basis
> whether to
> >>>>> encrypt
> >>>>> RTCP - hopefully SRTP implementations also permit this.
> >>>>>
> >>>> Sure, they can do this. But remember offer-answer. How does a
> >>>> receiver
> >>>> communicate such a granular policy to the transmitter?
> Right now if
> >>>> can say whether to auth or auth+encrypt RTP, and auth or
> >>>> auth+encrypt
> >>>> RTCP. There's no way to say auth+encrypt SDES but only auth SR/XR
> >>>> etc.
> >>>
> >>> Seems to me that this is a problem with SDP security descriptions.
> >>> That problem might be there because we forgot about the E
> bit.  This
> >>> seems very messy for the reason you mention above -
> granularity.  I
> >>> don't think that we want to negotiate the encryption
> policy for each
> >>> type SDP message, which is a problem with something like
> the E bit.
> >>>
> >>> Mark
> >>>>
> >>>>
> >>>>> Logically I'd send SDES on its own at the start of a call to
> >>>>> provide
> >>>>> some
> >>>>> user info, but would not send SR/RR/XR until at least
> 8-10 seconds
> >>>>> into the
> >>>>> call as I need some time to gather statistics to report.
> >>>> That's not exactly what the RTP specs say. RTCP timers don't have
> >>>> this
> >>>> kind of policy today. In fact, there are porposals on
> the table to
> >>>> use
> >>>> RTCP for early feedback to assess reception quality on
> some initial
> >>>> packets - e.g. the RTP-noop proposal. In any event that has no
> >>>> bearing
> >>>> on the present discussion.
> >>>>
> >>>>> After the
> >>>>> start-of-call exchange there would appear to be minimal need to
> >>>>> include
> >>>>> SDES. I'm sure we could construct some scenarios in
> which you "may"
> >>>>> need to
> >>>>> include SDES with SR/RR/XR however the vast majority of
> cases would
> >>>>> either
> >>>>> need only a single initial exchange or may not need SDES at all.
> >>>>>
> >>>> Any time you change SSRC you potentially need to send SDES.
> >>>> That said, I still don't see what bearing this has on
> the discussion
> >>>> about whether we need the complexity of encrypting some
> RTCP packets
> >>>> and not others for a single session.
> >>>>
> >>>> It's a tradeoff between extra complexity+less privacy versus
> >>>> allowing
> >>>> snoopers to gather RTCP stats.
> >>>>
> >>>> Dave.
> >>>>
> >>>>> Regards
> >>>>>
> >>>>> Alan
> >>>>>
> >>>>>
> >>>>>
> >>>>> -----Original Message-----
> >>>>> From: Michael A. Ramalho [mailto:mramalho@cisco.com]
> >>>>> Sent: Wednesday, November 24, 2004 9:39 AM
> >>>>> To: Alan Clark
> >>>>> Cc: Oren Peleg; avt@ietf.org
> >>>>> Subject: RE: [AVT] RFC 3711 and RTCP
> >>>>>
> >>>>>
> >>>>> At 07:54 AM 11/24/2004 -0500, Alan Clark wrote:
> >>>>>> Hi Oren
> >>>>>>
> >>>>>> Yes - encryption is intended to provide privacy however this
> >>>>>> relates
> >>>>>> primarily to the voice payloads and to information within RTCP
> >>>>>> that
> >>>>>> may be
> >>>>>> associated with individual users (e.g. SDES).
> >>>>>>
> >>>>>> SR, RR and XR contain only peformance statistics and
> there is no
> >>>>>> privacy
> >>>>>> consideration that I can think of related to this type of data.
> >>>>>>
> >>>>>> The message is - encrypt RTP, optionally encrypt RTCP SDES but
> >>>>>> don't
> >>>>> encrypt
> >>>>>> RTCP SR/RR/XR
> >>>>>
> >>>>> Hi Alan,
> >>>>>
> >>>>> I am wondering about implementations utilizing your advise
> >>>>> "optionally
> >>>>> encrypt
> >>>>> RTCP SDES but don't encrypt RTCP SR/RR/XR".
> >>>>>
> >>>>> Is there an easy way to tell SRTP to encrypt RTCP SDES, but not
> >>>>> RTCP
> >>>>> SR/RR/XR? It seems that all the RTCP blocks would otherwise be
> >>>>> highly
> >>>>> related to each other. For example (quote out of RFC 3611):
> >>>>>
> >>>>> "XR packets supplement the existing RTCP packets, and may be
> >>>>> stacked
> >>>>> with
> >>>>> other RTCP packets to form compound RTCP packets [9,
> Section 6]."
> >>>>>
> >>>>> Thus, RFC3611 explicitly admits the possibility of (but does not
> >>>>> mandate)
> >>>>> the piggybacking all the RTCP-XR block types.
> >>>>>
> >>>>> RTCP implementations would have to know the SRTP
> encryption rules
> >>>>> to
> >>>>> make sure they don't perform piggybacking with SDES ...
> but could
> >>>>> allow
> >>>>> (the more bandwidth efficient) piggybacking on the other RTCP-XR
> >>>>> blocks.
> >>>>>
> >>>>> This RTCP/SRTP interaction/dependency seems to add a
> burden to both
> >>>>> RTCP and SRTP implementations ... let alone the expressiveness
> >>>>> required
> >>>>> to signal your advise in existing SDP/SRTP mechanisms.
> >>>>>
> >>>>> Are you aware of any implementations that have actually followed
> >>>>> your
> >>>>> advise above? I rather go into this one with our eyes open ...
> >>>>> developers
> >>>>> highly dislike this cross-functional entity dependency
> (RTCP & SRTP
> >>>>> along
> >>>>> with expressing it in SDP/SRTP mechanisms).
> >>>>>
> >>>>> Regards,
> >>>>>
> >>>>> Michael Ramalho
> >>>>>
> >>>>>
> >>>>>
> >>>>>> Regards
> >>>>>>
> >>>>>> Alan
> >>>>>>
> >>>>>> -----Original Message-----
> >>>>>> From: Oren Peleg [mailto:OrenP@audiocodes.com]
> >>>>>> Sent: Wednesday, November 24, 2004 7:50 AM
> >>>>>> To: Alan Clark; avt@ietf.org
> >>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> Isnt encryption is to prevent monitoring from the first place?
> >>>>>>
> >>>>>> -----Original Message-----
> >>>>>> From: avt-bounces@ietf.org
> [mailto:avt-bounces@ietf.org] On Behalf
> >>>>>> Of
> >>>>>> Alan Clark
> >>>>>> Sent: Tuesday, November 23, 2004 5:42 PM
> >>>>>> To: avt@ietf.org
> >>>>>> Subject: [AVT] RFC 3711 and RTCP
> >>>>>>
> >>>>>>
> >>>>>>> From the flurry of discussion on RFC3711 it looks as
> if there is
> >>>>>> considerable interest in SRTP implementation.  I'd
> like to add a
> >>>>>> request
> >>>>>> to
> >>>>>> SRTP implementors with regard to manageability.
> >>>>>>
> >>>>>> RTCP SR, RR and more particularly XR (e.g. VoIP Metrics report
> >>>>>> block)
> >>>>>> contain useful information for network monitoring, and
> are often
> >>>>>> used by
> >>>>>> probes and analyzers to support problem diagnosis.
> RFC3711 allows
> >>>>>> the
> >>>>>> selective encryption of RTCP payloads and hence I
> would strongly
> >>>>>> recomment
> >>>>>> that RFC3711 implementors DO NOT encrypt RTCP SR, RR and XR
> >>>>>> payloads
> >>>>>> as
> >>>>>> this
> >>>>>> adversely impacts manageability.
> >>>>>>
> >>>>>> Regards
> >>>>>>
> >>>>>> Alan Clark
> >>>>>> Telchemy
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> Audio/Video Transport Working Group
> >>>>>> avt@ietf.org
> >>>>>> https://www1.ietf.org/mailman/listinfo/avt
> >>>>>>
> >>>>>>
> ******************************************************************
> >>>>>> **
> >>>>>> *
> >>>>>> ******
> >>>>> *
> >>>>>> ********************************
> >>>>>> This email and any files transmitted with it are confidential
> >>>>>> material.
> >>>>>> They are intended solely for the use of the designated
> individual
> >>>>>> or
> >>>>>> entity
> >>>>>> to whom they are addressed.
> >>>>>> If the reader of this message is not the intended recipient,
> >>>>>> you are hereby notified that any dissemination,use,
> >>>>>> distribution or copying of this communication is strictly
> >>>>>> prohibited
> >>>>>> and
> >>>>> may
> >>>>>> be unlawful.
> >>>>>>
> >>>>>> If you have received this email in error please notify
> >>>>>> postmaster@audiocodes.com and permanently delete the e-mail and
> >>>>>> files.
> >>>>>>
> >>>>>>
> >>>>>>
> ******************************************************************
> >>>>>> **
> >>>>>> *
> >>>>>> ******
> >>>>> *
> >>>>>> *******************************
> >>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> Audio/Video Transport Working Group
> >>>>>> avt@ietf.org
> >>>>>> https://www1.ietf.org/mailman/listinfo/avt
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> Audio/Video Transport Working Group
> >>>>> avt@ietf.org
> >>>>> https://www1.ietf.org/mailman/listinfo/avt
> >>>>>
> >>>> David R. Oran
> >>>> Cisco Fellow
> >>>> Cisco Systems
> >>>> 7 Ladyslipper Lane
> >>>> Acton, MA 01720 USA
> >>>> Tel: +1 978 264 2048
> >>>> Email: oran@cisco.com
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> Audio/Video Transport Working Group
> >>>> avt@ietf.org
> >>>> https://www1.ietf.org/mailman/listinfo/avt
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> Audio/Video Transport Working Group
> >>> avt@ietf.org
> >>> https://www1.ietf.org/mailman/listinfo/avt
> >>>
> >>
> >
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>

_______________________________________________
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From mmusic-bounces@ietf.org  Thu Dec  2 21:16:22 2004
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Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Thu, 2 Dec 2004 21:56:53 +0200
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Thread-Topic: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
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From: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
To: "Alan Clark" <alan.d.clark@telchemy.com>,
        "Mark Baugher" <mbaugher@cisco.com>
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Alan,

I believe that we are not in disagreement from a technical perspective. =
I am just suggesting that reporting RTCP XR metrics to a RAQMON =
collector be one of the options that should be available=20

This is not about pushing a RAQMON-centric technology to the industry or =
closing other options. This is about providing another way of =
transporting RTCP XR reports for these situations where the RTCP XR =
probes cannot be deployed . As the need for RTCP encryption will not run =
away completely, at least in some environments, and even the information =
carried by  RTCP XR may be considered confidential in some deployments, =
RAQMON reports carrying RTCP XR information can provide the missing =
functionality.=20

RAQMON Reports will also support the case of the encrypted IP tunnels, =
as well as any combination of media and signaling encryption.

Regards,

Dan



> -----Original Message-----
> From: Alan Clark [mailto:alan.d.clark@telchemy.com]
> Sent: 02 December, 2004 7:45 PM
> To: Romascanu, Dan (Dan); Mark Baugher
> Cc: rmonmib@ietf.org; mmusic@ietf.org; avt@ietf.org
> Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>=20
>=20
> Dan
>=20
> I can understand your desire to create a RAQMON-centric=20
> reporting system
> however the industry needs to keep its options open.  The=20
> RTCP XR based VoIP
> performance management framework allows metrics to be=20
> reported via the media
> path, signaling protocols or SNMP, and we need to give the=20
> implementor as
> much flexibility as possible.  Reporting RTCP XR metrics to a RAQMON
> collector is one option that should be available however this=20
> would not
> address many practical scenarios.
>=20
> Service providers and network managers make extensive use of=20
> probes and
> analyzers to monitor key demarc points in the network and=20
> diagnose problems.
> Encrypting everything can severely limit the functionality of these
> mid-network systems and hence make it harder for the network=20
> manager to
> manage the network.
>=20
> If we had to trade the number of circumstances in which=20
> knowledge of packet
> loss/ discard rates etc. were a security concern vs the number of
> circumstances under which network managers needed improved=20
> tools to diagnose
> problems then I'd be surprised if the former was even a=20
> minute fraction of
> the latter.
> It is also distinctly possible that if issues such as the=20
> duration of a call
> was of concern then the IP endpoints would probably simply=20
> use an encrypted
> IP tunnel - otherwise a mid-stream observer could examine the=20
> RTP stream.
>=20
> Regards
>=20
> Alan
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org]On Behalf Of
> Romascanu, Dan (Dan)
> Sent: Thursday, December 02, 2004 6:46 AM
> To: Alan Clark; Mark Baugher
> Cc: rmonmib@ietf.org; mmusic@ietf.org; avt@ietf.org
> Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>=20
>=20
> I would be cautious in saying that in no circumstances the=20
> contents of an XR
> payload would be remotely useful to an eavesdropper. We may=20
> encounter some
> very security aware environments where even the fact that a=20
> call happens
> between two IP phones, its duration, or QoS parameters need to be kept
> confidential. For this reason, while working to provide a strong
> recommendation for RTCP XR packets not to be encrypted, I do=20
> not believe
> that we can make this a MUST requirements. As an alternative,=20
> I suggest to
> use  RAQMON extensions to report the RTCP XR metrics to=20
> RAQMON collectors
> without the security risks that may be of concern in some=20
> environments.
>=20
> Regards,
>=20
> Dan
>=20
>=20
>=20
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org
> > [mailto:mmusic-bounces@ietf.org]On Behalf Of Alan Clark
> > Sent: 01 December, 2004 2:08 PM
> > To: Mark Baugher
> > Cc: mmusic@ietf.org
> > Subject: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
> >
> >
> > Mark
> >
> > To provide some background to the list - RFC3611 states the
> > following:-
> >
> >    "Any encryption or filtering of XR report blocks entails=20
> a loss of
> >    monitoring information to third parties.  For example, a
> > network that
> >    establishes a tunnel to encrypt VoIP Report Blocks denies that
> >    information to the service providers traversed by the=20
> tunnel.  The
> >    service providers cannot then monitor or respond to the=20
> quality of
> >    the VoIP calls that they carry, potentially creating
> > problems for the
> >    network's users.  As a default, XR packets should not be
> > encrypted or
> >    filtered."
> >
> > I can't conceive of any circumstance in which the contents of
> > an XR payload
> > would be remotely useful to an eavesdropper and hence=20
> stating that XR
> > payloads should always be left unencrypted would be (at least
> > in my opinion)
> > ok.  If we wanted to make this even more specific we could=20
> narrow this
> > requirement down to only XR VoIP Metrics payloads to avoid
> > the situation in
> > which some future XR payload may be problematic.
> >
> > The availability of VoIP metrics payloads will be a huge
> > issue for service
> > providers - as we move through 2005 RTCP XR will be a=20
> mainstay of VoIP
> > performance management and is already a mandatory requirement
> > within certain
> > sectors of the industry.  Making the VoIP metrics payload
> > unencrypted by
> > default may seem a small issue today however it will have
> > major consequences
> > to the industry as SRTP starts to roll out.
> >
> > Regards
> >
> > Alan
> >
> >
> >
> > -----Original Message-----
> > From: Mark Baugher [mailto:mbaugher@cisco.com]
> > Sent: Tuesday, November 30, 2004 6:45 PM
> > To: Alan Clark
> > Cc: mmusic@ietf.org
> > Subject: Re: [AVT] RFC 3711 and RTCP
> >
> >
> > Alan
> >    I am copying MMUSIC (and not cross-posting to AVT) to
> > further discuss
> > a possible change to SDP security descriptions, which has=20
> completed WG
> > last call.  The change is needed for RFC 3611, and we also=20
> discovered
> > that
> > http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
> > -07.txt does not address the SRTCP E bit properly. I offered the
> > following text in a posting to AVT:
> > "As the default behavior when UNENCRYPTED_SRTCP is=20
> signaled, all SRTCP
> > packets MUST have the E bit set to zero and MUST NOT be encrypted.
> > Otherwise, all SRTP packets MUST have the E bit set to one=20
> and MUST be
> > encrypted by default.  Individual applications MAY override these
> > default behaviors in their specifications.  In a session where SRTCP
> > packets are encrypted, therefore, applications that use RFC 3611 MAY
> > set the E bit to 0 for XR packets as recommended by RFC 3611."
> >
> > One of my co-authors pointed out that the last sentence is
> > more than an
> > editorial change.  We also don't like the fact that an offerer or
> > answerer cannot explicitly signal when it wants the XR packet to be
> > encrypted or unencrypted.  In other words, if ENCRYPTED_SRTCP is the
> > default setting and RFC 3611 is also signaled, it is ambiguous as to
> > whether the SRTCP sender will encrypt the XR packet or not.  It's
> > solely up to the sender.  If the offerer or answerer want the
> > XR packet
> > to always be encrypted, there is no way to signal that. =20
> Ditto for the
> > unencrypted case.
> >
> > I think we need to put the first three sentences of the above-quoted
> > paragraph into SDP security descriptions as an editorial=20
> change.  I'm
> > proposing this to the working group.  I also propose not to
> > include the
> > last sentence.  Instead, we could add a new signaling parameter,
> > UNENCRYPTED_XR, which implies that all SRTCP messages are to be
> > encrypted except for the XR packets, which will have their E
> > bit clear,
> > of course.
> >
> > This new parameter could be placed in an RFC 3611bis or in=20
> a revision
> > of draft-ietf-mmusic-sdescriptions-07.txt, which will then need to
> > undergo another WG last call.  What do you (all) think about this?
> >
> > Mark
> >
> > On Nov 24, 2004, at 4:18 PM, Mark Baugher wrote:
> >
> > > Alan
> > >   As Dave Oran has pointed out, the definition that you=20
> want is not
> > > supported in the signaling, at present.  The current definition of
> > >=20
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
> > > -07.txt treats SRTCP encryption as either on or off and does not
> > > address the problem of setting the E bit in SRTCP packets.
> > This is a
> > > bug that we need to fix.
> > >
> > >   MMUSIC has finished last call on
> > > draft-ietf-mmusic-sdescriptions-07.txt, and the transport ADs have
> > > reviewed it.  We'll be making some changes at Jon=20
> Peterson's request
> > > and post another version in about a week.  This is an=20
> opportunity to
> > > address the E bit, but the MMUSIC chairs or transport ADs may want
> > > another WG last call.  Off the top of my head, I would
> > offer something
> > > like the following.
> > >
> > > "As the default behavior when UNENCRYPTED_SRTCP is
> > signaled, all SRTCP
> > > packets MUST have the E bit set to zero and MUST NOT be encrypted.
> > > Otherwise, all SRTP packets MUST have the E bit set to one
> > and MUST be
> > > encrypted by default.  Individual applications MAY override these
> > > default behaviors in their specifications.  In a session=20
> where SRTCP
> > > packets are encrypted, therefore, applications that use=20
> RFC 3611 MAY
> > > set the E bit to 0 for XR packets as recommended by RFC 3611."
> > >
> > > Mark
> > >
> > > On Nov 24, 2004, at 2:33 PM, Alan Clark wrote:
> > >
> > >>
> > >>
> > >> Mark
> > >>
> > >> We tried to flag this in RFC3611 with the following text in the
> > >> "security
> > >> considerations" section
> > >>
> > >>    Any encryption or filtering of XR report blocks entails
> > a loss of
> > >>    monitoring information to third parties.  For=20
> example, a network
> > >> that
> > >>    establishes a tunnel to encrypt VoIP Report Blocks denies that
> > >>    information to the service providers traversed by the
> > tunnel.  The
> > >>    service providers cannot then monitor or respond to the
> > quality of
> > >>    the VoIP calls that they carry, potentially creating
> > problems for
> > >> the
> > >>    network's users.  As a default, XR packets should not
> > be encrypted
> > >> or
> > >>    filtered.
> > >>
> > >> Could we potentially interpret this as being the
> > definition of what to
> > >> selectively encrypt in the case of the RFC3611 specification?
> > >>
> > >> Regards
> > >>
> > >> Alan
> > >>
> > >>
> > >> -----Original Message-----
> > >> From: Mark Baugher [mailto:mbaugher@cisco.com]
> > >> Sent: Wednesday, November 24, 2004 2:42 PM
> > >> To: Alan Clark
> > >> Cc: Oren Peleg; Cullen Jennings; avt@ietf.org; David R
> > Oran; Michael
> > >> A.
> > >> Ramalho
> > >> Subject: Re: [AVT] RFC 3711 and RTCP
> > >>
> > >>
> > >> So, when the sender and receiver agree to encrypt SRTCP,=20
> this would
> > >> mean
> > >> only that some SRTCP messages would be encrypted and the
> > sender would
> > >> decide which ones to encrypt?  My personal opinion is that
> > this is not
> > >> secure because it is unpredictable to the receiver, which
> > may have a
> > >> policy that all messages related to its session be encrypted.
> > >>
> > >> I can think of a few potential approaches to this problem.
> > >>
> > >> 1. Let individual SRTCP messages or a bundle of SRTCP messages
> > >> be signaled to be unencrypted when SRTCP encryption is used.
> > >>
> > >> 2. Make a blanket policy that the sender determine what
> > >> to selectively encrypt, as you suggest.
> > >>
> > >> 3. Define the policy of what to selectively encrypt in
> > >> the particular specification.  In other words, a document could
> > >> re-define the signaling of SRTCP encryption according to its
> > >> needs.  The specification of foo, therefore, could define a new
> > >> parameter such as foo_ENCRYPTED_SRTCP to be used in place of
> > >> ENCRYPTED_SRTCP.  The foo document would define what gets
> > >> encrypted and what does not get encrypted as per its needs.
> > >>
> > >> I commented on #2 already.  I don't like #1 because of its
> > signaling
> > >> complexity.  I like #3 because it limits the complexity to the
> > >> specific needs of an application.  #1 would be more=20
> efficient than
> > >> #3 if lots of applications had their own specific needs for
> > >> selective encryption of SRTCP message.  But I don't think this is
> > >> the case.
> > >>
> > >> Mark
> > >>
> > >> On Nov 24, 2004, at 10:58 AM, Alan Clark wrote:
> > >>
> > >>>
> > >>> If the use of encryption in SRTCP is signalled by an
> > explicit (E) bit
> > >>> in the
> > >>> SRTCP header then perhaps the sender could autonomously decide
> > >>> whether
> > >>> it
> > >>> wants to encrypt?  Presumably since the sender is in the best
> > >>> position
> > >>> to
> > >>> decide if the data is sensitive then this would be ok - and the
> > >>> receiver is
> > >>> able to just look at the E bit and hence recognize that
> > decryption is
> > >>> needed.
> > >>>
> > >>> Alan
> > >>>
> > >>> -----Original Message-----
> > >>> From: avt-bounces@ietf.org
> > [mailto:avt-bounces@ietf.org]On Behalf Of
> > >>> Mark Baugher
> > >>> Sent: Wednesday, November 24, 2004 1:33 PM
> > >>> To: David R Oran
> > >>> Cc: Cullen Jennings; Oren Peleg; Michael A. Ramalho; Alan Clark;
> > >>> avt@ietf.org
> > >>> Subject: Re: [AVT] RFC 3711 and RTCP
> > >>>
> > >>>
> > >>>
> > >>> On Nov 24, 2004, at 10:05 AM, David R Oran wrote:
> > >>>
> > >>>>
> > >>>> On Nov 24, 2004, at 12:23 PM, Alan Clark wrote:
> > >>>>
> > >>>>>
> > >>>>> Michael
> > >>>>>
> > >>>>> I don't see why this would be a problem unless you=20
> included SDES
> > >>>>> and
> > >>>>> (SR or
> > >>>>> RR or XR) in the same RTCP packet.  SRTP (at least the
> > protocol) is
> > >>>>> apparently able to decided on a packet by packet basis
> > whether to
> > >>>>> encrypt
> > >>>>> RTCP - hopefully SRTP implementations also permit this.
> > >>>>>
> > >>>> Sure, they can do this. But remember offer-answer. How does a
> > >>>> receiver
> > >>>> communicate such a granular policy to the transmitter?
> > Right now if
> > >>>> can say whether to auth or auth+encrypt RTP, and auth or
> > >>>> auth+encrypt
> > >>>> RTCP. There's no way to say auth+encrypt SDES but only=20
> auth SR/XR
> > >>>> etc.
> > >>>
> > >>> Seems to me that this is a problem with SDP security=20
> descriptions.
> > >>> That problem might be there because we forgot about the E
> > bit.  This
> > >>> seems very messy for the reason you mention above -
> > granularity.  I
> > >>> don't think that we want to negotiate the encryption
> > policy for each
> > >>> type SDP message, which is a problem with something like
> > the E bit.
> > >>>
> > >>> Mark
> > >>>>
> > >>>>
> > >>>>> Logically I'd send SDES on its own at the start of a call to
> > >>>>> provide
> > >>>>> some
> > >>>>> user info, but would not send SR/RR/XR until at least
> > 8-10 seconds
> > >>>>> into the
> > >>>>> call as I need some time to gather statistics to report.
> > >>>> That's not exactly what the RTP specs say. RTCP timers=20
> don't have
> > >>>> this
> > >>>> kind of policy today. In fact, there are porposals on
> > the table to
> > >>>> use
> > >>>> RTCP for early feedback to assess reception quality on
> > some initial
> > >>>> packets - e.g. the RTP-noop proposal. In any event that has no
> > >>>> bearing
> > >>>> on the present discussion.
> > >>>>
> > >>>>> After the
> > >>>>> start-of-call exchange there would appear to be=20
> minimal need to
> > >>>>> include
> > >>>>> SDES. I'm sure we could construct some scenarios in
> > which you "may"
> > >>>>> need to
> > >>>>> include SDES with SR/RR/XR however the vast majority of
> > cases would
> > >>>>> either
> > >>>>> need only a single initial exchange or may not need=20
> SDES at all.
> > >>>>>
> > >>>> Any time you change SSRC you potentially need to send SDES.
> > >>>> That said, I still don't see what bearing this has on
> > the discussion
> > >>>> about whether we need the complexity of encrypting some
> > RTCP packets
> > >>>> and not others for a single session.
> > >>>>
> > >>>> It's a tradeoff between extra complexity+less privacy versus
> > >>>> allowing
> > >>>> snoopers to gather RTCP stats.
> > >>>>
> > >>>> Dave.
> > >>>>
> > >>>>> Regards
> > >>>>>
> > >>>>> Alan
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>> -----Original Message-----
> > >>>>> From: Michael A. Ramalho [mailto:mramalho@cisco.com]
> > >>>>> Sent: Wednesday, November 24, 2004 9:39 AM
> > >>>>> To: Alan Clark
> > >>>>> Cc: Oren Peleg; avt@ietf.org
> > >>>>> Subject: RE: [AVT] RFC 3711 and RTCP
> > >>>>>
> > >>>>>
> > >>>>> At 07:54 AM 11/24/2004 -0500, Alan Clark wrote:
> > >>>>>> Hi Oren
> > >>>>>>
> > >>>>>> Yes - encryption is intended to provide privacy however this
> > >>>>>> relates
> > >>>>>> primarily to the voice payloads and to information=20
> within RTCP
> > >>>>>> that
> > >>>>>> may be
> > >>>>>> associated with individual users (e.g. SDES).
> > >>>>>>
> > >>>>>> SR, RR and XR contain only peformance statistics and
> > there is no
> > >>>>>> privacy
> > >>>>>> consideration that I can think of related to this=20
> type of data.
> > >>>>>>
> > >>>>>> The message is - encrypt RTP, optionally encrypt=20
> RTCP SDES but
> > >>>>>> don't
> > >>>>> encrypt
> > >>>>>> RTCP SR/RR/XR
> > >>>>>
> > >>>>> Hi Alan,
> > >>>>>
> > >>>>> I am wondering about implementations utilizing your advise
> > >>>>> "optionally
> > >>>>> encrypt
> > >>>>> RTCP SDES but don't encrypt RTCP SR/RR/XR".
> > >>>>>
> > >>>>> Is there an easy way to tell SRTP to encrypt RTCP=20
> SDES, but not
> > >>>>> RTCP
> > >>>>> SR/RR/XR? It seems that all the RTCP blocks would otherwise be
> > >>>>> highly
> > >>>>> related to each other. For example (quote out of RFC 3611):
> > >>>>>
> > >>>>> "XR packets supplement the existing RTCP packets, and may be
> > >>>>> stacked
> > >>>>> with
> > >>>>> other RTCP packets to form compound RTCP packets [9,
> > Section 6]."
> > >>>>>
> > >>>>> Thus, RFC3611 explicitly admits the possibility of=20
> (but does not
> > >>>>> mandate)
> > >>>>> the piggybacking all the RTCP-XR block types.
> > >>>>>
> > >>>>> RTCP implementations would have to know the SRTP
> > encryption rules
> > >>>>> to
> > >>>>> make sure they don't perform piggybacking with SDES ...
> > but could
> > >>>>> allow
> > >>>>> (the more bandwidth efficient) piggybacking on the=20
> other RTCP-XR
> > >>>>> blocks.
> > >>>>>
> > >>>>> This RTCP/SRTP interaction/dependency seems to add a
> > burden to both
> > >>>>> RTCP and SRTP implementations ... let alone the expressiveness
> > >>>>> required
> > >>>>> to signal your advise in existing SDP/SRTP mechanisms.
> > >>>>>
> > >>>>> Are you aware of any implementations that have=20
> actually followed
> > >>>>> your
> > >>>>> advise above? I rather go into this one with our eyes open ...
> > >>>>> developers
> > >>>>> highly dislike this cross-functional entity dependency
> > (RTCP & SRTP
> > >>>>> along
> > >>>>> with expressing it in SDP/SRTP mechanisms).
> > >>>>>
> > >>>>> Regards,
> > >>>>>
> > >>>>> Michael Ramalho
> > >>>>>
> > >>>>>
> > >>>>>
> > >>>>>> Regards
> > >>>>>>
> > >>>>>> Alan
> > >>>>>>
> > >>>>>> -----Original Message-----
> > >>>>>> From: Oren Peleg [mailto:OrenP@audiocodes.com]
> > >>>>>> Sent: Wednesday, November 24, 2004 7:50 AM
> > >>>>>> To: Alan Clark; avt@ietf.org
> > >>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > >>>>>> Isnt encryption is to prevent monitoring from the=20
> first place?
> > >>>>>>
> > >>>>>> -----Original Message-----
> > >>>>>> From: avt-bounces@ietf.org
> > [mailto:avt-bounces@ietf.org] On Behalf
> > >>>>>> Of
> > >>>>>> Alan Clark
> > >>>>>> Sent: Tuesday, November 23, 2004 5:42 PM
> > >>>>>> To: avt@ietf.org
> > >>>>>> Subject: [AVT] RFC 3711 and RTCP
> > >>>>>>
> > >>>>>>
> > >>>>>>> From the flurry of discussion on RFC3711 it looks as
> > if there is
> > >>>>>> considerable interest in SRTP implementation.  I'd
> > like to add a
> > >>>>>> request
> > >>>>>> to
> > >>>>>> SRTP implementors with regard to manageability.
> > >>>>>>
> > >>>>>> RTCP SR, RR and more particularly XR (e.g. VoIP=20
> Metrics report
> > >>>>>> block)
> > >>>>>> contain useful information for network monitoring, and
> > are often
> > >>>>>> used by
> > >>>>>> probes and analyzers to support problem diagnosis.
> > RFC3711 allows
> > >>>>>> the
> > >>>>>> selective encryption of RTCP payloads and hence I
> > would strongly
> > >>>>>> recomment
> > >>>>>> that RFC3711 implementors DO NOT encrypt RTCP SR, RR and XR
> > >>>>>> payloads
> > >>>>>> as
> > >>>>>> this
> > >>>>>> adversely impacts manageability.
> > >>>>>>
> > >>>>>> Regards
> > >>>>>>
> > >>>>>> Alan Clark
> > >>>>>> Telchemy
> > >>>>>>
> > >>>>>>
> > >>>>>> _______________________________________________
> > >>>>>> Audio/Video Transport Working Group
> > >>>>>> avt@ietf.org
> > >>>>>> https://www1.ietf.org/mailman/listinfo/avt
> > >>>>>>
> > >>>>>>
> > ******************************************************************
> > >>>>>> **
> > >>>>>> *
> > >>>>>> ******
> > >>>>> *
> > >>>>>> ********************************
> > >>>>>> This email and any files transmitted with it are confidential
> > >>>>>> material.
> > >>>>>> They are intended solely for the use of the designated
> > individual
> > >>>>>> or
> > >>>>>> entity
> > >>>>>> to whom they are addressed.
> > >>>>>> If the reader of this message is not the intended recipient,
> > >>>>>> you are hereby notified that any dissemination,use,
> > >>>>>> distribution or copying of this communication is strictly
> > >>>>>> prohibited
> > >>>>>> and
> > >>>>> may
> > >>>>>> be unlawful.
> > >>>>>>
> > >>>>>> If you have received this email in error please notify
> > >>>>>> postmaster@audiocodes.com and permanently delete the=20
> e-mail and
> > >>>>>> files.
> > >>>>>>
> > >>>>>>
> > >>>>>>
> > ******************************************************************
> > >>>>>> **
> > >>>>>> *
> > >>>>>> ******
> > >>>>> *
> > >>>>>> *******************************
> > >>>>>>
> > >>>>>>
> > >>>>>> _______________________________________________
> > >>>>>> Audio/Video Transport Working Group
> > >>>>>> avt@ietf.org
> > >>>>>> https://www1.ietf.org/mailman/listinfo/avt
> > >>>>>
> > >>>>>
> > >>>>> _______________________________________________
> > >>>>> Audio/Video Transport Working Group
> > >>>>> avt@ietf.org
> > >>>>> https://www1.ietf.org/mailman/listinfo/avt
> > >>>>>
> > >>>> David R. Oran
> > >>>> Cisco Fellow
> > >>>> Cisco Systems
> > >>>> 7 Ladyslipper Lane
> > >>>> Acton, MA 01720 USA
> > >>>> Tel: +1 978 264 2048
> > >>>> Email: oran@cisco.com
> > >>>>
> > >>>>
> > >>>> _______________________________________________
> > >>>> Audio/Video Transport Working Group
> > >>>> avt@ietf.org
> > >>>> https://www1.ietf.org/mailman/listinfo/avt
> > >>>>
> > >>>
> > >>>
> > >>> _______________________________________________
> > >>> Audio/Video Transport Working Group
> > >>> avt@ietf.org
> > >>> https://www1.ietf.org/mailman/listinfo/avt
> > >>>
> > >>
> > >
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >
>=20
> _______________________________________________
> Audio/Video Transport Working Group
> avt@ietf.org
> https://www1.ietf.org/mailman/listinfo/avt
>=20
>=20

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From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Thu, 2 Dec 2004 17:31:55 -0800
To: Alan Clark <alan.d.clark@telchemy.com>,
        Dan ((Dan)) Romascanu <dromasca@avaya.com>
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Alan, Dan
   I don't know where the discussion is going.  The point of all this is 
to allow RFC 3611 to use SRTP, which is not of direct interest to 
rmonmib.  There is a problem signaling SRTP for RFC 3611 using SDP 
security descriptions, which is not of direct interest to AVT.  Because 
RFC 3611 uses a SHOULD for leaving SRTCP XR messages in the clear when 
other SRTCP messages are encrypted, we have an ambiguity in the 
signaling.  This ambiguity arises because SDP security descriptions 
doesn't address use of the SRTCP E bit, which XR needs packets need to 
use.  At least two of the authors of SDP security descriptions think 
that we need to remove the ambiguity.  So here are the choices that I 
see.

1. change RFC 3611 to describe how to do SDP security descriptions 
signaling
2. change draft-ietf-mmusic-sdescriptions to describe how to use the E 
bit.

I would not be surprised if you favored #2.  I like #1.  But I am 
willing to add a new parameter to sdescriptions and then run it through 
WG last call.  I'd like to resolve this and move forward.

Mark


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Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Fri, 3 Dec 2004 11:01:40 +0200
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From: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
To: "Mark Baugher" <mbaugher@cisco.com>,
        "Alan Clark" <alan.d.clark@telchemy.com>
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I do not really have a strong preference. My inclination is to leave RFC =
3611 as it is and to change draft-ietf-mmusic-sdescriptions to describe =
how to use the E bit. It looks like the right place to define SDP =
security descriptions, and has chances to run faster through the =
process.=20

Regards,

Dan



> -----Original Message-----
> From: mmusic-bounces@ietf.org=20
> [mailto:mmusic-bounces@ietf.org]On Behalf Of Mark Baugher
> Sent: 03 December, 2004 3:32 AM
> To: Alan Clark; Romascanu, Dan (Dan)
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>=20
>=20
> Alan, Dan
>    I don't know where the discussion is going.  The point of=20
> all this is=20
> to allow RFC 3611 to use SRTP, which is not of direct interest to=20
> rmonmib.  There is a problem signaling SRTP for RFC 3611 using SDP=20
> security descriptions, which is not of direct interest to=20
> AVT.  Because=20
> RFC 3611 uses a SHOULD for leaving SRTCP XR messages in the=20
> clear when=20
> other SRTCP messages are encrypted, we have an ambiguity in the=20
> signaling.  This ambiguity arises because SDP security descriptions=20
> doesn't address use of the SRTCP E bit, which XR needs=20
> packets need to=20
> use.  At least two of the authors of SDP security descriptions think=20
> that we need to remove the ambiguity.  So here are the choices that I=20
> see.
>=20
> 1. change RFC 3611 to describe how to do SDP security descriptions=20
> signaling
> 2. change draft-ietf-mmusic-sdescriptions to describe how to=20
> use the E=20
> bit.
>=20
> I would not be surprised if you favored #2.  I like #1.  But I am=20
> willing to add a new parameter to sdescriptions and then run=20
> it through=20
> WG last call.  I'd like to resolve this and move forward.
>=20
> Mark
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

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--NextPart

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

	Title		: Connection-Establishment Preconditions in the Session
                          Initiation Protocol (SIP)
	Author(s)	: G. Camarillo
	Filename	: draft-ietf-mmusic-connection-precon-01.txt
	Pages		: 8
	Date		: 2004-12-2
	
This document defines the connection-establishment precondition type
   for the SIP preconditions framework. Connection-establishment
   preconditions are met when a transport connection (e.g., a TCP
   connection) is successfully established between two endpoints.

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Gonzalo

This is a follow up question along the same lines to what I sent on 11/8.

I thought the idea of preconditions was that they need to be met prior to 
session establishment. I believe this is when the precondition state table 
is identical in both UAs. I do not believe your example in this ID shows 
this to be true prior to transmitting the 180. Shouldn't your UPDATE 
message have the a=curr line of:

         a=curr:conn e2e <something>
         a=des:conn mandatory e2e sendrecv

and not "none" as your examples has (maybe this should be "send" only since 
A has its radio up at this time)?

Further, shouldn't the 200 OK to that UPDATE (or some other message) have 
the following A lines:

         a=curr:conn e2e sendrecv
         a=des:conn mandatory e2e sendrecv

as this would show the precondition conn has been met in both directions?

My confusion is that your precondition is looking for the conn attribute to 
meet the current condition of "e2e sendrecv", and your example does not 
have this prior to showing everything is met and to progress to the 180 
(meaning preconditions have been met).

All this said, I understand you are changing the state in the A line:

         a=setup

attribute along the way. You just do not seem to be addressing the "a=curr" 
condition.

Or am I missing something here?


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>A New Internet-Draft is available from the on-line Internet-Drafts 
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>This draft is a work item of the Multiparty Multimedia Session Control 
>Working Group of the IETF.
>
>         Title           : Connection-Establishment Preconditions in the 
> Session
>                           Initiation Protocol (SIP)
>         Author(s)       : G. Camarillo
>         Filename        : draft-ietf-mmusic-connection-precon-01.txt
>         Pages           : 8
>         Date            : 2004-12-2
>
>This document defines the connection-establishment precondition type
>    for the SIP preconditions framework. Connection-establishment
>    preconditions are met when a transport connection (e.g., a TCP
>    connection) is successfully established between two endpoints.
>
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cheers,
James

                                *******************
                 Truth is not to be argued... it is to be presented


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It looks to me like the SDP syntax for "o=3D" is forcing users to =
divulge
information that could be considered sensitive, or private.

According to draft-ietf-mmusic-sdp-new-22.txt, the username field on the
"o=3D" line is supposed to reveal the user's login identity.  Divulging
this information to strangers is not a good security practice.

=20

The SDP spec allows the "username" field to be set to "-", but only if
the originating host does not support the concept of user IDs.  I think
the spec should be changed to allow the "username" field to be set to
"-" even when the host has a concept of user IDs.

=20

Current text:

=20

   <username> is the user's login on the originating host, or it is "-"
      if the originating host does not support the concept of user ids.
      The <username> MUST NOT contain spaces.


Proposed new text:

=20

   <username> is the user's login on the originating host.=20
      The <username> may be "-", or a random character string, as long
as
      the tuple of <username>, <sess-id>, <nettype>, <addrtype> and=20
      <unicast-address> form a globally unique identifier for the
session.
      The <username> MUST NOT contain spaces.

=20

There is a similar problem "unicast-address" parameter on the "o=3D" =
line.
For security reasons, some people may not want to divulge their IP
address or domain name, especially if the host is behind a NAT.  I think
we can fix it in a similar way.

=20

This what the text currently says:

=20

   <unicast-address> is the address of the machine from which the
      session was created.  For an address type of IP4, this is either
      the fully-qualified domain name of the machine, or the
      dotted-decimal representation of the IP version 4 address of the
      machine.  For an address type of IP6, this is either the
      fully-qualified domain name of the machine, or the compressed
      textual representation of the IP version 6 address of the machine.
      For both IP4 and IP6, the fully-qualified domain name is the form
      that SHOULD be given unless this is unavailable, in which case the
      globally unique address MAY be substituted.  A local IP address
      MUST NOT be used in any context where the SDP description might
      leave the scope in which the address is meaningful.


My proposal is to keep that text, but add this extra sentence at the
end:

=20

      A randomly generated character string MAY be used as
<unicast-address>
      as long as the tuple of <username>, <sess-id>, <nettype>,
<addrtype> and=20
      <unicast-address> form a globally unique identifier for the
session.
=20

=20

Comments?

=20

Anders

=20


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<div class=3DSection1>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>It looks to me like the SDP syntax for =
&#8220;o=3D&#8221; is
forcing users to divulge information that could be considered sensitive, =
or
private.<o:p></o:p></span></font></p>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>According to draft-ietf-mmusic-sdp-new-22.txt, the =
username
field on the &#8220;o=3D&#8221; line is supposed to reveal the =
user&#8217;s login
identity.&nbsp; Divulging this information to strangers is not a good =
security
practice.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>The SDP spec allows the &#8220;username&#8221; field =
to be
set to &#8220;-&#8220;, but only if the originating host does not =
support the
concept of user IDs. &nbsp;I think the spec should be changed to allow =
the &#8220;username&#8221;
field to be set to &#8220;-&#8220; even when the host has a concept of =
user
IDs.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Current text:<o:p></o:p></span></font></p>

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

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; &lt;username&gt; is the user's =
login on the originating host, or it is =
&quot;-&quot;<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; if the =
originating host does not support the concept of user =
ids.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The =
&lt;username&gt; MUST NOT contain spaces.<o:p></o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><br>
Proposed new text:<o:p></o:p></span></font></p>

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

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; &lt;username&gt; is the user's =
login on the originating host. <o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;The =
&lt;username&gt; may be &quot;-&quot;, or a random character string, as =
long as<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the tuple of =
&lt;username&gt;, &lt;sess-id&gt;, &lt;nettype&gt;, &lt;addrtype&gt; and =
<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;unicas=
t-address&gt; form a globally unique identifier for the =
session.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; The =
&lt;username&gt; MUST NOT contain spaces.<o:p></o:p></span></font></pre>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>There is a similar problem =
&#8220;unicast-address&#8221;
parameter on the &#8220;o=3D&#8221; line. &nbsp;For security reasons, =
some people
may not want to divulge their IP address or domain name, especially if =
the host
is behind a NAT.&nbsp; I think we can fix it in a similar =
way.<o:p></o:p></span></font></p>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>This what the text currently =
says:<o:p></o:p></span></font></p>

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

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp; &lt;unicast-address&gt; is the =
address of the machine from which =
the<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; session was =
created.&nbsp; For an address type of IP4, this is =
either<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the =
fully-qualified domain name of the machine, or =
the<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; dotted-decimal =
representation of the IP version 4 address of =
the<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; machine.&nbsp; =
For an address type of IP6, this is either =
the<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
fully-qualified domain name of the machine, or the =
compressed<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; textual =
representation of the IP version 6 address of the =
machine.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; For both IP4 =
and IP6, the fully-qualified domain name is the =
form<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; that SHOULD be =
given unless this is unavailable, in which case =
the<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; globally =
unique address MAY be substituted.&nbsp; A local IP =
address<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; MUST NOT be =
used in any context where the SDP description =
might<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; leave the =
scope in which the address is meaningful.<o:p></o:p></span></font></pre>

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'><br>
My proposal is to keep that text, but add this extra sentence at the =
end:<o:p></o:p></span></font></p>

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

<pre><font size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; A randomly =
generated character string MAY be used as =
&lt;unicast-address&gt;<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; as long as the =
tuple of &lt;username&gt;, &lt;sess-id&gt;, &lt;nettype&gt;, =
&lt;addrtype&gt; and <o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&lt;unicas=
t-address&gt; form a globally unique identifier for the =
session.<o:p></o:p></span></font></pre><pre><font
size=3D2 face=3D"Courier New"><span =
style=3D'font-size:10.0pt'><o:p>&nbsp;</o:p></span></font></pre>

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

<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
style=3D'font-size:10.0pt;
font-family:Arial'>Comments?<o:p></o:p></span></font></p>

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

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

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

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This is to announce a working group last call on the SDP Media Label 
draft (draft-ietf-mmusic-sdp-media-label-00.txt). Please send any final 
comments on this draft to the list by 17th December 2004. If no 
substantive comments are raised by that time, we will request this 
document be published as a proposed standard RFC.

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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] SDP security/privacy concerns
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Anders,

On 4 Dec 2004, at 19:39, Anders Klemets wrote:
> It looks to me like the SDP syntax for =93o=3D=94 is forcing users to=20=

> divulge information that could be considered sensitive, or private.
>
> According to draft-ietf-mmusic-sdp-new-22.txt, the username field on=20=

> the =93o=3D=94 line is supposed to reveal the user=92s login =
identity.=A0=20
> Divulging this information to strangers is not a good security=20
> practice.
>
> The SDP spec allows the =93username=94 field to be set to =93-=93, but =
only if=20
> the originating host does not support the concept of user IDs. =A0I=20
> think the spec should be changed to allow the =93username=94 field to =
be=20
> set to =93-=93 even when the host has a concept of user IDs.

Seems reasonable to me.

...
> There is a similar problem =93unicast-address=94 parameter on the =93o=3D=
=94=20
> line. =A0For security reasons, some people may not want to divulge =
their=20
> IP address or domain name, especially if the host is behind a NAT.=A0 =
I=20
> think we can fix it in a similar way.
>
> This what the text currently says:
> =A0
> =A0=A0 <unicast-address> is the address of the machine from which the
> =A0=A0=A0=A0=A0 session was created.=A0 For an address type of IP4, =
this is either
> =A0=A0=A0=A0=A0 the fully-qualified domain name of the machine, or the
> =A0=A0=A0=A0=A0 dotted-decimal representation of the IP version 4 =
address of the
> =A0=A0=A0=A0=A0 machine.=A0 For an address type of IP6, this is either =
the
> =A0=A0=A0=A0=A0 fully-qualified domain name of the machine, or the =
compressed
> =A0=A0=A0=A0=A0 textual representation of the IP version 6 address of =
the=20
> machine.
> =A0=A0=A0=A0=A0 For both IP4 and IP6, the fully-qualified domain name =
is the form
> =A0=A0=A0=A0=A0 that SHOULD be given unless this is unavailable, in =
which case=20
> the
> =A0=A0=A0=A0=A0 globally unique address MAY be substituted.=A0 A local =
IP address
> =A0=A0=A0=A0=A0 MUST NOT be used in any context where the SDP =
description might
> =A0=A0=A0=A0=A0 leave the scope in which the address is meaningful.
>
>
>  My proposal is to keep that text, but add this extra sentence at the=20=

> end:
> =A0
> =A0=A0=A0=A0=A0 A randomly generated character string MAY be used as=20=

> <unicast-address>
> =A0=A0=A0=A0=A0 as long as the tuple of <username>, <sess-id>, =
<nettype>,=20
> <addrtype> and
> =A0=A0=A0=A0=A0=A0<unicast-address> form a globally unique identifier =
for the=20
> session.

The <unicast-address> must  be consistent with the <nettype> and=20
<addrtype> fields, so I don't think it can be a randomly generated=20
string. I wouldn't object to implementations that use 127.0.0.1 though,=20=

provided the username was chosen to ensure uniqueness.

Colin=


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Hi,

I have reviewed the document and have some comments. I am sorry that I 
have not been active earlier and come this late:

1. Section 4: No ABNF definition of "token". I assume that you use that 
of SDP. However it needs to be either referenced or explicitly defined.

2. No reference to what BNF you are using. I guess a reference to RFC 
2234 is all that is needed. Otherwise, there would however be some 
uncertainties about spacing. Is any white spaces allowed between 
"a=label:" and the pointer. In RFC 2234 BNF the syntax would say no. In 
some other BNF the answer may be yes.

3. Section 5. I am a bit worried about the offer/answer section. It is 
very unclear on the handling of this attribute. Can we really trust 
applications to define such behavior? From my work with Offer/Answer I 
sometimes find it difficult to get the right behavior out of SDP in 
offer/answer usage. If this behavior is left to the application then 
there is need for a much more explicit section that explains the 
different choices. Also I think using application defined behavior will 
result in quite divergent behavior.

4. Reference [4] is only informative.

I do hope this quick comments make people believe they do not need to 
read the document in the WG last call.

Cheers

Magnus

Colin Perkins wrote:
> This is to announce a working group last call on the SDP Media Label
> draft (draft-ietf-mmusic-sdp-media-label-00.txt). Please send any final
> comments on this draft to the list by 17th December 2004. If no
> substantive comments are raised by that time, we will request this
> document be published as a proposed standard RFC.
> 
> Colin
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 


-- 

Magnus Westerlund

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

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From mmusic-bounces@ietf.org  Mon Dec  6 18:38:38 2004
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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Mon, 6 Dec 2004 23:21:35 +0000
To: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
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I'd also prefer this change to be made in sdescriptions (which needs 
another revision to address Jon Peterson's comments anyway, right?)

Colin



On 3 Dec 2004, at 09:01, Romascanu, Dan ((Dan)) wrote:
> I do not really have a strong preference. My inclination is to leave 
> RFC 3611 as it is and to change draft-ietf-mmusic-sdescriptions to 
> describe how to use the E bit. It looks like the right place to define 
> SDP security descriptions, and has chances to run faster through the 
> process.
>
> Regards,
>
> Dan
>
>
>
>> -----Original Message-----
>> From: mmusic-bounces@ietf.org
>> [mailto:mmusic-bounces@ietf.org]On Behalf Of Mark Baugher
>> Sent: 03 December, 2004 3:32 AM
>> To: Alan Clark; Romascanu, Dan (Dan)
>> Cc: mmusic@ietf.org
>> Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>>
>>
>> Alan, Dan
>>    I don't know where the discussion is going.  The point of
>> all this is
>> to allow RFC 3611 to use SRTP, which is not of direct interest to
>> rmonmib.  There is a problem signaling SRTP for RFC 3611 using SDP
>> security descriptions, which is not of direct interest to
>> AVT.  Because
>> RFC 3611 uses a SHOULD for leaving SRTCP XR messages in the
>> clear when
>> other SRTCP messages are encrypted, we have an ambiguity in the
>> signaling.  This ambiguity arises because SDP security descriptions
>> doesn't address use of the SRTCP E bit, which XR needs
>> packets need to
>> use.  At least two of the authors of SDP security descriptions think
>> that we need to remove the ambiguity.  So here are the choices that I
>> see.
>>
>> 1. change RFC 3611 to describe how to do SDP security descriptions
>> signaling
>> 2. change draft-ietf-mmusic-sdescriptions to describe how to
>> use the E
>> bit.
>>
>> I would not be surprised if you favored #2.  I like #1.  But I am
>> willing to add a new parameter to sdescriptions and then run
>> it through
>> WG last call.  I'd like to resolve this and move forward.
>>
>> Mark
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>
> _______________________________________________
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> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>


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From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Mon, 6 Dec 2004 15:32:28 -0800
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Yes it does.  There are two possible approaches.  We have an existing 
parameter called UNENCRYPTED_SRTCP, but the default is to encrypt the 
SRTCP.  We could add a parameter specific to RFC 3611 or SRTCP XR 
packets such as UNENCRYPTED_SRTCP_XR, which would mean that SRTCP is 
encrypted but the XR packets are to be left in the clear (and signaled 
using the E bit).  That's my current default if no one speaks up with 
an alternative preference.

I'd like to resolve this issue by around the middle of the week and 
generate a new version of sdescriptions.

Mark
On Dec 6, 2004, at 3:21 PM, Colin Perkins wrote:

> I'd also prefer this change to be made in sdescriptions (which needs 
> another revision to address Jon Peterson's comments anyway, right?)
>
> Colin
>
>
>
> On 3 Dec 2004, at 09:01, Romascanu, Dan ((Dan)) wrote:
>> I do not really have a strong preference. My inclination is to leave 
>> RFC 3611 as it is and to change draft-ietf-mmusic-sdescriptions to 
>> describe how to use the E bit. It looks like the right place to 
>> define SDP security descriptions, and has chances to run faster 
>> through the process.
>>
>> Regards,
>>
>> Dan
>>
>>
>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org
>>> [mailto:mmusic-bounces@ietf.org]On Behalf Of Mark Baugher
>>> Sent: 03 December, 2004 3:32 AM
>>> To: Alan Clark; Romascanu, Dan (Dan)
>>> Cc: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>>>
>>>
>>> Alan, Dan
>>>    I don't know where the discussion is going.  The point of
>>> all this is
>>> to allow RFC 3611 to use SRTP, which is not of direct interest to
>>> rmonmib.  There is a problem signaling SRTP for RFC 3611 using SDP
>>> security descriptions, which is not of direct interest to
>>> AVT.  Because
>>> RFC 3611 uses a SHOULD for leaving SRTCP XR messages in the
>>> clear when
>>> other SRTCP messages are encrypted, we have an ambiguity in the
>>> signaling.  This ambiguity arises because SDP security descriptions
>>> doesn't address use of the SRTCP E bit, which XR needs
>>> packets need to
>>> use.  At least two of the authors of SDP security descriptions think
>>> that we need to remove the ambiguity.  So here are the choices that I
>>> see.
>>>
>>> 1. change RFC 3611 to describe how to do SDP security descriptions
>>> signaling
>>> 2. change draft-ietf-mmusic-sdescriptions to describe how to
>>> use the E
>>> bit.
>>>
>>> I would not be surprised if you favored #2.  I like #1.  But I am
>>> willing to add a new parameter to sdescriptions and then run
>>> it through
>>> WG last call.  I'd like to resolve this and move forward.
>>>
>>> Mark
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>


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From: "Alan Clark" <alan.d.clark@telchemy.com>
To: "Mark Baugher" <mbaugher@cisco.com>, "Colin Perkins" <csp@csperkins.org>
Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Mon, 6 Dec 2004 19:27:26 -0500
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Mark

I support your UNENCRYPTED_SRTCP_XR solution, and thanks for resolving the
issue.

Regards

Alan Clark


-----Original Message-----
From: Mark Baugher [mailto:mbaugher@cisco.com]
Sent: Monday, December 06, 2004 6:32 PM
To: Colin Perkins
Cc: Romascanu, Dan (Dan); mmusic@ietf.org; Alan Clark
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP


Yes it does.  There are two possible approaches.  We have an existing
parameter called UNENCRYPTED_SRTCP, but the default is to encrypt the
SRTCP.  We could add a parameter specific to RFC 3611 or SRTCP XR
packets such as UNENCRYPTED_SRTCP_XR, which would mean that SRTCP is
encrypted but the XR packets are to be left in the clear (and signaled
using the E bit).  That's my current default if no one speaks up with
an alternative preference.

I'd like to resolve this issue by around the middle of the week and
generate a new version of sdescriptions.

Mark
On Dec 6, 2004, at 3:21 PM, Colin Perkins wrote:

> I'd also prefer this change to be made in sdescriptions (which needs
> another revision to address Jon Peterson's comments anyway, right?)
>
> Colin
>
>
>
> On 3 Dec 2004, at 09:01, Romascanu, Dan ((Dan)) wrote:
>> I do not really have a strong preference. My inclination is to leave
>> RFC 3611 as it is and to change draft-ietf-mmusic-sdescriptions to
>> describe how to use the E bit. It looks like the right place to
>> define SDP security descriptions, and has chances to run faster
>> through the process.
>>
>> Regards,
>>
>> Dan
>>
>>
>>
>>> -----Original Message-----
>>> From: mmusic-bounces@ietf.org
>>> [mailto:mmusic-bounces@ietf.org]On Behalf Of Mark Baugher
>>> Sent: 03 December, 2004 3:32 AM
>>> To: Alan Clark; Romascanu, Dan (Dan)
>>> Cc: mmusic@ietf.org
>>> Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>>>
>>>
>>> Alan, Dan
>>>    I don't know where the discussion is going.  The point of
>>> all this is
>>> to allow RFC 3611 to use SRTP, which is not of direct interest to
>>> rmonmib.  There is a problem signaling SRTP for RFC 3611 using SDP
>>> security descriptions, which is not of direct interest to
>>> AVT.  Because
>>> RFC 3611 uses a SHOULD for leaving SRTCP XR messages in the
>>> clear when
>>> other SRTCP messages are encrypted, we have an ambiguity in the
>>> signaling.  This ambiguity arises because SDP security descriptions
>>> doesn't address use of the SRTCP E bit, which XR needs
>>> packets need to
>>> use.  At least two of the authors of SDP security descriptions think
>>> that we need to remove the ambiguity.  So here are the choices that I
>>> see.
>>>
>>> 1. change RFC 3611 to describe how to do SDP security descriptions
>>> signaling
>>> 2. change draft-ietf-mmusic-sdescriptions to describe how to
>>> use the E
>>> bit.
>>>
>>> I would not be surprised if you favored #2.  I like #1.  But I am
>>> willing to add a new parameter to sdescriptions and then run
>>> it through
>>> WG last call.  I'd like to resolve this and move forward.
>>>
>>> Mark
>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>


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Subject: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-12.txt
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Hi, 

I have a 2 comments on 
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-kmgmt-ext-12.txt.


This first comment is for a clarification:

- Section 3.1.2 explains that SIP 488 is used for rejecting a key management
offer.
  488 allows for the inclusion of an SDP indicating the capabilities of the 
  answerer (the same way an SDP in a 200 OK sent in an OPTIONS request
indicates
  capabilities). It would be desireable to explain how this works with 
  kmgmt. I assume that this is what is meant by wording saying "Further
details
  about the cause of failure MAY be described in an included message from
the key
  management protocol." Perhaps a wording like the following would help:

	Further details about the media capabilities of the offerer MAY be
in
	included in 488 according to normal RFC 3261 rules. Further details
about
	the cause of the rejection of the key message itself MAY be
described in 
      the key message in the SDP in the 488, according to the rules
	of the key management protocol. The session is then ...

 ...or something along those lines.

 It would also be worthwile to describe how this relates to MIKEY in 
 particular (i.e, that the MIKEY message itself in the key-mgmt may have
 the key management rejection reason). It would also be helpful to have
 an Example of a failure case for SIP 488/MIKEY.
  
-----------------------

This next comment is concerning a potential issue with SIP forking. I'm
not sure we can fix it in MMUSIC, but I'll state it anyways.

- There is a somewhat unpleasant side-effect of using a SIP failure response

  for rejecting a key management message.

- Suppose that there is a SIP forking proxy forking the offer (in the
INVITE).
  Let's say the call is forked to A and B. A is a device for which no
  user is present at the time (say, an office phone). Suppose A rejects the 
  call with 401 Unauthorized because it needs authorization. Suppose the
user
  is at home (B), but rejects the offer with 488 (because it's using the key
  management protocol used for the corporation which is different than the
one
  used at home). The forking proxy will see both the 401 an the 488, but can
only
  forward one to the offerer. There is nothing saying which one should be
forwarded
  beyong the error response that indicates that there could be
retransmission (which
  applies to both codes in this case). Worst, in this particular example, if
the proxy
  follows the rule section 6/RFC 3261, chances are it will pick 401. This
means
  that you won't be able to call home because you'll never receive a 488.

- This is a weakness of the forking procedures in SIP, and perhaps it should
be
  addresses in the SIP working group.

- The other alternative would be to not consider the rejection of the key as
a
  rejection of the offer. Instead of sending a 488, you would accept the
  offer with an answer, and the key rejection would be included in the key
message
  itself (which could then trigger another offer/answer). In other words,
the 
  offer/answer is sucessful, but the key exchanged is not. This would have
the obvious
  advantage of the previous scenario working properly. However, if there is
no
  suitable key management protocol, we could run into the reverse problem
(when the
  user is at work (A) instead of (B). In this case, the proxy would forward
the
  200 with the answer with the key message, and drop the 401. There would be
another
  offer/answer round which would then fail. The caller in this case would
never get
  the 401.

I'm not sure how to deal with this forking issue.

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<P><FONT SIZE=3D2>Hi, </FONT>
</P>

<P><FONT SIZE=3D2>I have a 2 comments on </FONT>
<BR><FONT SIZE=3D2><A =
HREF=3D"http://www.ietf.org/internet-drafts/draft-ietf-mmusic-kmgmt-ext-=
12.txt" =
TARGET=3D"_blank">http://www.ietf.org/internet-drafts/draft-ietf-mmusic-=
kmgmt-ext-12.txt</A>.</FONT>
</P>
<BR>

<P><FONT SIZE=3D2>This first comment is for a clarification:</FONT>
</P>

<P><FONT SIZE=3D2>- Section 3.1.2 explains that SIP 488 is used for =
rejecting a key management offer.</FONT>
<BR><FONT SIZE=3D2>&nbsp; 488 allows for the inclusion of an SDP =
indicating the capabilities of the </FONT>
<BR><FONT SIZE=3D2>&nbsp; answerer (the same way an SDP in a 200 OK =
sent in an OPTIONS request indicates</FONT>
<BR><FONT SIZE=3D2>&nbsp; capabilities). It would be desireable to =
explain how this works with </FONT>
<BR><FONT SIZE=3D2>&nbsp; kmgmt. I assume that this is what is meant by =
wording saying &quot;Further details</FONT>
<BR><FONT SIZE=3D2>&nbsp; about the cause of failure MAY be described =
in an included message from the key</FONT>
<BR><FONT SIZE=3D2>&nbsp; management protocol.&quot; Perhaps a wording =
like the following would help:</FONT>
</P>

<P>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>Further =
details about the media capabilities of the offerer MAY be in</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>included =
in 488 according to normal RFC 3261 rules. Further details about</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>the cause =
of the rejection of the key message itself MAY be described in </FONT>
<BR><FONT SIZE=3D2>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; the key message in =
the SDP in the 488, according to the rules</FONT>
<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <FONT SIZE=3D2>of the =
key management protocol. The session is then ...</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;...or something along those lines.</FONT>
</P>

<P><FONT SIZE=3D2>&nbsp;It would also be worthwile to describe how this =
relates to MIKEY in </FONT>
<BR><FONT SIZE=3D2>&nbsp;particular (i.e, that the MIKEY message itself =
in the key-mgmt may have</FONT>
<BR><FONT SIZE=3D2>&nbsp;the key management rejection reason). It would =
also be helpful to have</FONT>
<BR><FONT SIZE=3D2>&nbsp;an Example of a failure case for SIP =
488/MIKEY.</FONT>
<BR><FONT SIZE=3D2>&nbsp; </FONT>
<BR><FONT SIZE=3D2>-----------------------</FONT>
</P>

<P><FONT SIZE=3D2>This next comment is concerning a potential issue =
with SIP forking. I'm</FONT>
<BR><FONT SIZE=3D2>not sure we can fix it in MMUSIC, but I'll state it =
anyways.</FONT>
</P>

<P><FONT SIZE=3D2>- There is a somewhat unpleasant side-effect of using =
a SIP failure response </FONT>
<BR><FONT SIZE=3D2>&nbsp; for rejecting a key management =
message.</FONT>
</P>

<P><FONT SIZE=3D2>- Suppose that there is a SIP forking proxy forking =
the offer (in the INVITE).</FONT>
<BR><FONT SIZE=3D2>&nbsp; Let's say the call is forked to A and B. A is =
a device for which no</FONT>
<BR><FONT SIZE=3D2>&nbsp; user is present at the time (say, an office =
phone). Suppose A rejects the </FONT>
<BR><FONT SIZE=3D2>&nbsp; call with 401 Unauthorized because it needs =
authorization. Suppose the user</FONT>
<BR><FONT SIZE=3D2>&nbsp; is at home (B), but rejects the offer with =
488 (because it's using the key</FONT>
<BR><FONT SIZE=3D2>&nbsp; management protocol used for the corporation =
which is different than the one</FONT>
<BR><FONT SIZE=3D2>&nbsp; used at home). The forking proxy will see =
both the 401 an the 488, but can only</FONT>
<BR><FONT SIZE=3D2>&nbsp; forward one to the offerer. There is nothing =
saying which one should be forwarded</FONT>
<BR><FONT SIZE=3D2>&nbsp; beyong the error response that indicates that =
there could be retransmission (which</FONT>
<BR><FONT SIZE=3D2>&nbsp; applies to both codes in this case). Worst, =
in this particular example, if the proxy</FONT>
<BR><FONT SIZE=3D2>&nbsp; follows the rule section 6/RFC 3261, chances =
are it will pick 401. This means</FONT>
<BR><FONT SIZE=3D2>&nbsp; that you won't be able to call home because =
you'll never receive a 488.</FONT>
</P>

<P><FONT SIZE=3D2>- This is a weakness of the forking procedures in =
SIP, and perhaps it should be</FONT>
<BR><FONT SIZE=3D2>&nbsp; addresses in the SIP working group.</FONT>
</P>

<P><FONT SIZE=3D2>- The other alternative would be to not consider the =
rejection of the key as a</FONT>
<BR><FONT SIZE=3D2>&nbsp; rejection of the offer. Instead of sending a =
488, you would accept the</FONT>
<BR><FONT SIZE=3D2>&nbsp; offer with an answer, and the key rejection =
would be included in the key message</FONT>
<BR><FONT SIZE=3D2>&nbsp; itself (which could then trigger another =
offer/answer). In other words, the </FONT>
<BR><FONT SIZE=3D2>&nbsp; offer/answer is sucessful, but the key =
exchanged is not. This would have the obvious</FONT>
<BR><FONT SIZE=3D2>&nbsp; advantage of the previous scenario working =
properly. However, if there is no</FONT>
<BR><FONT SIZE=3D2>&nbsp; suitable key management protocol, we could =
run into the reverse problem (when the</FONT>
<BR><FONT SIZE=3D2>&nbsp; user is at work (A) instead of (B). In this =
case, the proxy would forward the</FONT>
<BR><FONT SIZE=3D2>&nbsp; 200 with the answer with the key message, and =
drop the 401. There would be another</FONT>
<BR><FONT SIZE=3D2>&nbsp; offer/answer round which would then fail. The =
caller in this case would never get</FONT>
<BR><FONT SIZE=3D2>&nbsp; the 401.</FONT>
</P>

<P><FONT SIZE=3D2>I'm not sure how to deal with this forking =
issue.</FONT>
</P>

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To: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
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        Colin Perkins <csp@csperkins.org>, mmusic@ietf.org
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There are two issues here:
1. Relationship between the SRTCP E flag and the UNENCRYPTED_SRTCP 
sdescriptions parameter.
2. The ability to selectively encrypt parts of your SRTCP.

On the first issue, I agree the sdescriptions need to be clarified.

On the second issue, I understand, from a pragmatic point of view, the 
desire to leave RTCP XR in the clear, but I have some general concerns 
with the idea of specifying a security negotiation protocol that leaves 
certain information in the clear for the sole purpose of having 
intermediaries snoop on it. Considering that we have already been 
through WG last call and AD review of sdescriptions, I would like to 
hear from both the WG chairs and the ADs, that they consider such a 
mechanism reasonable before we proceed here.

Assuming that to be the case, we need to note that sdescriptions makes a 
distinction between negotiated and declarative session parameters (see 
Section 3.4). Is the suggestion here to make this new parameter 
negotiated (i.e. both sides have to agree on it, and it applies in both 
directions), or declarative (each side merely indicates what he will do 
with packets sent by him) ? Since UNENCRYPTED_SRTCP is a negotiated 
parameter, I assume UNENCRYPTED_SRTCP_XR would also be negotiated. 
Similarly, given the name of the parameter, I assume absence of it means 
that all of SRTCP will be encrypted (unless signaled otherwise). If so, 
this means that we will only leave SRTCP_XR in the clear if both sides 
explicitly negotiate support for that, which sounds like a reasonable 
suggestion to me.
 
-- Flemming


Mark Baugher wrote:

> Yes it does.  There are two possible approaches.  We have an existing 
> parameter called UNENCRYPTED_SRTCP, but the default is to encrypt the 
> SRTCP.  We could add a parameter specific to RFC 3611 or SRTCP XR 
> packets such as UNENCRYPTED_SRTCP_XR, which would mean that SRTCP is 
> encrypted but the XR packets are to be left in the clear (and signaled 
> using the E bit).  That's my current default if no one speaks up with 
> an alternative preference.
>
> I'd like to resolve this issue by around the middle of the week and 
> generate a new version of sdescriptions.
>
> Mark
> On Dec 6, 2004, at 3:21 PM, Colin Perkins wrote:
>
>> I'd also prefer this change to be made in sdescriptions (which needs 
>> another revision to address Jon Peterson's comments anyway, right?)
>>
>> Colin
>>
>>
>>
>> On 3 Dec 2004, at 09:01, Romascanu, Dan ((Dan)) wrote:
>>
>>> I do not really have a strong preference. My inclination is to leave 
>>> RFC 3611 as it is and to change draft-ietf-mmusic-sdescriptions to 
>>> describe how to use the E bit. It looks like the right place to 
>>> define SDP security descriptions, and has chances to run faster 
>>> through the process.
>>>
>>> Regards,
>>>
>>> Dan
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: mmusic-bounces@ietf.org
>>>> [mailto:mmusic-bounces@ietf.org]On Behalf Of Mark Baugher
>>>> Sent: 03 December, 2004 3:32 AM
>>>> To: Alan Clark; Romascanu, Dan (Dan)
>>>> Cc: mmusic@ietf.org
>>>> Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>>>>
>>>>
>>>> Alan, Dan
>>>>    I don't know where the discussion is going.  The point of
>>>> all this is
>>>> to allow RFC 3611 to use SRTP, which is not of direct interest to
>>>> rmonmib.  There is a problem signaling SRTP for RFC 3611 using SDP
>>>> security descriptions, which is not of direct interest to
>>>> AVT.  Because
>>>> RFC 3611 uses a SHOULD for leaving SRTCP XR messages in the
>>>> clear when
>>>> other SRTCP messages are encrypted, we have an ambiguity in the
>>>> signaling.  This ambiguity arises because SDP security descriptions
>>>> doesn't address use of the SRTCP E bit, which XR needs
>>>> packets need to
>>>> use.  At least two of the authors of SDP security descriptions think
>>>> that we need to remove the ambiguity.  So here are the choices that I
>>>> see.
>>>>
>>>> 1. change RFC 3611 to describe how to do SDP security descriptions
>>>> signaling
>>>> 2. change draft-ietf-mmusic-sdescriptions to describe how to
>>>> use the E
>>>> bit.
>>>>
>>>> I would not be surprised if you favored #2.  I like #1.  But I am
>>>> willing to add a new parameter to sdescriptions and then run
>>>> it through
>>>> WG last call.  I'd like to resolve this and move forward.
>>>>
>>>> Mark
>>>>
>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>>>
>>>
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>>
>>
>
>
> _______________________________________________
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To: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
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Hi Mark,

Mark Baugher wrote:
> Yes it does.  There are two possible approaches.  We have an existing
> parameter called UNENCRYPTED_SRTCP, but the default is to encrypt the
> SRTCP.  We could add a parameter specific to RFC 3611 or SRTCP XR
> packets such as UNENCRYPTED_SRTCP_XR, which would mean that SRTCP is
> encrypted but the XR packets are to be left in the clear (and signaled
> using the E bit).  That's my current default if no one speaks up with
> an alternative preference.
> 

I think one should consider that there actually may be need to negotiate 
what should be left unencrypted, or the other way around. Therefore my 
proposal would be to define a "PARTIALENCRYPTED_SRTCP" and then another 
negotiated attribute listing the RTCP packet types that should be 
protected or the other way around. One can also consider if the normal 
case is based on senders local policy.

Cheers

Magnus Westerlund

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

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From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Tue, 7 Dec 2004 07:06:16 -0800
To: Flemming Andreasen <fandreas@cisco.com>
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On Dec 6, 2004, at 6:38 PM, Flemming Andreasen wrote:

> There are two issues here:
> 1. Relationship between the SRTCP E flag and the UNENCRYPTED_SRTCP 
> sdescriptions parameter.
> 2. The ability to selectively encrypt parts of your SRTCP.
>
> On the first issue, I agree the sdescriptions need to be clarified.
>
> On the second issue, I understand, from a pragmatic point of view, the 
> desire to leave RTCP XR in the clear, but I have some general concerns 
> with the idea of specifying a security negotiation protocol that 
> leaves certain information in the clear for the sole purpose of having 
> intermediaries snoop on it.

They are free to negotiate whether or not all SRTCP messages are 
unencrypted so there is no security hole in allowing offerers and 
answerers to negotiate whether or not to leave one type of SRTCP 
message in the clear.

> Considering that we have already been through WG last call and AD 
> review of sdescriptions, I would like to hear from both the WG chairs 
> and the ADs, that they consider such a mechanism reasonable before we 
> proceed here.
>
> Assuming that to be the case, we need to note that sdescriptions makes 
> a distinction between negotiated and declarative session parameters 
> (see Section 3.4). Is the suggestion here to make this new parameter 
> negotiated (i.e. both sides have to agree on it, and it applies in 
> both directions), or declarative (each side merely indicates what he 
> will do with packets sent by him) ? Since UNENCRYPTED_SRTCP is a 
> negotiated parameter, I assume UNENCRYPTED_SRTCP_XR would also be 
> negotiated. Similarly, given the name of the parameter, I assume 
> absence of it means that all of SRTCP will be encrypted (unless 
> signaled otherwise).

Yes.
> If so, this means that we will only leave SRTCP_XR in the clear if 
> both sides explicitly negotiate support for that, which sounds like a 
> reasonable suggestion to me.

Ok.

Mark
> -- Flemming
>
>
> Mark Baugher wrote:
>
>> Yes it does.  There are two possible approaches.  We have an existing 
>> parameter called UNENCRYPTED_SRTCP, but the default is to encrypt the 
>> SRTCP.  We could add a parameter specific to RFC 3611 or SRTCP XR 
>> packets such as UNENCRYPTED_SRTCP_XR, which would mean that SRTCP is 
>> encrypted but the XR packets are to be left in the clear (and 
>> signaled using the E bit).  That's my current default if no one 
>> speaks up with an alternative preference.
>>
>> I'd like to resolve this issue by around the middle of the week and 
>> generate a new version of sdescriptions.
>>
>> Mark
>> On Dec 6, 2004, at 3:21 PM, Colin Perkins wrote:
>>
>>> I'd also prefer this change to be made in sdescriptions (which needs 
>>> another revision to address Jon Peterson's comments anyway, right?)
>>>
>>> Colin
>>>
>>>
>>>
>>> On 3 Dec 2004, at 09:01, Romascanu, Dan ((Dan)) wrote:
>>>
>>>> I do not really have a strong preference. My inclination is to 
>>>> leave RFC 3611 as it is and to change 
>>>> draft-ietf-mmusic-sdescriptions to describe how to use the E bit. 
>>>> It looks like the right place to define SDP security descriptions, 
>>>> and has chances to run faster through the process.
>>>>
>>>> Regards,
>>>>
>>>> Dan
>>>>
>>>>
>>>>
>>>>> -----Original Message-----
>>>>> From: mmusic-bounces@ietf.org
>>>>> [mailto:mmusic-bounces@ietf.org]On Behalf Of Mark Baugher
>>>>> Sent: 03 December, 2004 3:32 AM
>>>>> To: Alan Clark; Romascanu, Dan (Dan)
>>>>> Cc: mmusic@ietf.org
>>>>> Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>>>>>
>>>>>
>>>>> Alan, Dan
>>>>>    I don't know where the discussion is going.  The point of
>>>>> all this is
>>>>> to allow RFC 3611 to use SRTP, which is not of direct interest to
>>>>> rmonmib.  There is a problem signaling SRTP for RFC 3611 using SDP
>>>>> security descriptions, which is not of direct interest to
>>>>> AVT.  Because
>>>>> RFC 3611 uses a SHOULD for leaving SRTCP XR messages in the
>>>>> clear when
>>>>> other SRTCP messages are encrypted, we have an ambiguity in the
>>>>> signaling.  This ambiguity arises because SDP security descriptions
>>>>> doesn't address use of the SRTCP E bit, which XR needs
>>>>> packets need to
>>>>> use.  At least two of the authors of SDP security descriptions 
>>>>> think
>>>>> that we need to remove the ambiguity.  So here are the choices 
>>>>> that I
>>>>> see.
>>>>>
>>>>> 1. change RFC 3611 to describe how to do SDP security descriptions
>>>>> signaling
>>>>> 2. change draft-ietf-mmusic-sdescriptions to describe how to
>>>>> use the E
>>>>> bit.
>>>>>
>>>>> I would not be surprised if you favored #2.  I like #1.  But I am
>>>>> willing to add a new parameter to sdescriptions and then run
>>>>> it through
>>>>> WG last call.  I'd like to resolve this and move forward.
>>>>>
>>>>> Mark
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> mmusic mailing list
>>>>> mmusic@ietf.org
>>>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>>>>
>>>>
>>>> _______________________________________________
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>>>>
>>>
>>
>>
>> _______________________________________________
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>


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To: Magnus Westerlund <magnus.westerlund@ericsson.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
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Magnus Westerlund wrote:

> Hi Mark,
>
> Mark Baugher wrote:
>
>> Yes it does.  There are two possible approaches.  We have an existing
>> parameter called UNENCRYPTED_SRTCP, but the default is to encrypt the
>> SRTCP.  We could add a parameter specific to RFC 3611 or SRTCP XR
>> packets such as UNENCRYPTED_SRTCP_XR, which would mean that SRTCP is
>> encrypted but the XR packets are to be left in the clear (and signaled
>> using the E bit).  That's my current default if no one speaks up with
>> an alternative preference.
>>
>
> I think one should consider that there actually may be need to 
> negotiate what should be left unencrypted, or the other way around. 
> Therefore my proposal would be to define a "PARTIALENCRYPTED_SRTCP" 
> and then another negotiated attribute listing the RTCP packet types 
> that should be protected or the other way around. One can also 
> consider if the normal case is based on senders local policy.
>
Hmm, I think we are getting on a slippery slope here with an extended 
scope, which is part of my initial concern with the whole idea of 
partially encrypted streams for the benefit of intermediate snoopers. 
What about the RTP stream ? Should we allow different treatment of 
different RTP packets, e.g. based on payload type, etc. ? Taking a step 
back, what are the motivations behind all of this added complexity and 
is this really the right time to expand the scope this way ?

-- Flemming

>

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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Tue, 7 Dec 2004 16:53:06 +0000
To: Flemming Andreasen <fandreas@cisco.com>
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On 7 Dec 2004, at 15:23, Flemming Andreasen wrote:
> Magnus Westerlund wrote:
>> Mark Baugher wrote:
>>> Yes it does.  There are two possible approaches.  We have an existing
>>> parameter called UNENCRYPTED_SRTCP, but the default is to encrypt the
>>> SRTCP.  We could add a parameter specific to RFC 3611 or SRTCP XR
>>> packets such as UNENCRYPTED_SRTCP_XR, which would mean that SRTCP is
>>> encrypted but the XR packets are to be left in the clear (and 
>>> signaled
>>> using the E bit).  That's my current default if no one speaks up with
>>> an alternative preference.
>>
>> I think one should consider that there actually may be need to 
>> negotiate what should be left unencrypted, or the other way around. 
>> Therefore my proposal would be to define a "PARTIALENCRYPTED_SRTCP" 
>> and then another negotiated attribute listing the RTCP packet types 
>> that should be protected or the other way around. One can also 
>> consider if the normal case is based on senders local policy.
>>
> Hmm, I think we are getting on a slippery slope here with an extended 
> scope, which is part of my initial concern with the whole idea of 
> partially encrypted streams for the benefit of intermediate snoopers. 
> What about the RTP stream ? Should we allow different treatment of 
> different RTP packets, e.g. based on payload type, etc. ? Taking a 
> step back, what are the motivations behind all of this added 
> complexity and is this really the right time to expand the scope this 
> way ?

All good points. How difficult would it be to write this up as a 
separate draft? That would allow the "sdescriptions" work to progress 
quickly, and allow us time to analyse the requirements for partial 
encryption, write a considered specification of the various parameters, 
and enumerate the new security considerations that might arise.

Colin


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From mmusic-bounces@ietf.org  Tue Dec  7 18:57:13 2004
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Subject: [MMUSIC] MSRP: SDP usage concern
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I sent the attached to the SIMPLE list last week. I am reposting it to  
both the SIMPLE and MMUSIC lists, as the discussion is of interest to  
both groups. If you already responded to this on the SIMPLE list or in  
private, it would be helpful if you can re-post your response to this  
thread, so everyone sees the whole discussion.

Thanks!

Ben.

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


In the DC meeting, we discussed that MSRP's SDP usage should be  
reviewed in MMUSIC, or at least by people who are more expert in SDP.   
Colin Perkins and Joerg Ott looked at this, and suggested a change.  
(Many thanks to them for taking the time, and I hope they will jump in  
and correct me if I mischaracterize anything.)

The upshot of the proposal is that the fact that MSRP ignores the  
address in the sdp c-line, and the port and proto fields in the m-line.  
Instead, MSRP gets all of this from the first URL in the path a-line  
attribute. This deviates from the traditional usage in SDP, and causes  
confusion.

The proposed change is that the destination URL be broken into the  
appropriate parts and encoded into the c-line and m-line. For example  
(leaving out lines that don't change)

  c=IN IP4 alice.example.com
  m=message 9 msrp *
  a=path:msrp://a.example.com:7394/2s93i;tcp

would become:

  c=IN IP4 a.example.com
  m=message 7394 msrp/tcp *
  a=session:2s93i

The change would only be made for the destination URL. URLs for any  
relays would be expressed the same way they are now.

The advantage of this approach is that it more closely follows the  
spirit of SDP. This will be easier to understand for people who already  
know SDP. Existing SDP code could be reused more effectively.

The disadvantage of the proposed change is, it means the way an MSRP  
client determines the destination URL would be different depending on  
whether the URL represented the final destination or a relay. Instead  
of the current approach of simply copying the path attribute value into  
the To-Path header in MSRP messages,  a client would copy the path  
header value if exists, reconstitute the destination URL from the m and  
c lines, and append the destination URL to the path header value.

So, what do people think? (I note that we actually did something  
similar to this before we added the "path" attribute. )


Thanks!

Ben.



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From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Tue, 7 Dec 2004 09:22:38 -0800
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hi Magnus,

On Dec 7, 2004, at 4:04 AM, Magnus Westerlund wrote:

> Hi Mark,
>
> Mark Baugher wrote:
>> Yes it does.  There are two possible approaches.  We have an existing
>> parameter called UNENCRYPTED_SRTCP, but the default is to encrypt the
>> SRTCP.  We could add a parameter specific to RFC 3611 or SRTCP XR
>> packets such as UNENCRYPTED_SRTCP_XR, which would mean that SRTCP is
>> encrypted but the XR packets are to be left in the clear (and signaled
>> using the E bit).  That's my current default if no one speaks up with
>> an alternative preference.
>
> I think one should consider that there actually may be need to 
> negotiate what should be left unencrypted, or the other way around. 
> Therefore my proposal would be to define a "PARTIALENCRYPTED_SRTCP" 
> and then another negotiated attribute listing the RTCP packet types 
> that should be protected or the other way around. One can also 
> consider if the normal case is based on senders local policy.

I don't think there are any security concerns in doing what you suggest 
apart from complexity.  It is more secure to allow a particular SRTCP 
packet to be in the clear rather than having to keep all of them in the 
clear for the sake of monitoring one or a subset of the SRTCP.  This is 
the point of the E bit, I suppose.  The problem I have with negotiating 
which packet types would be encrypted or not is the complexity and the 
concomitant effort needed to test the protocol, which doubles the SRTCP 
test cases since each case needs to be run with each SRTCP message in 
the clear and encrypted using E bit settings.  It's actually much worse 
since we're dealing with subsets.  The signaling complexity is also 
huge:  For all practical purposes, implementations never test all 
possible subsets of SRTCP packets in a PARTIALENCRYPTED_SRTCP 
negotiation.  Given this level of complexity and the challenges of 
testing, the overall security of the implementations will be much 
weaker than if we kept it simple.

We would need to have a real strong motivation to take on this burden.  
So far, we have had one group ask to have one message optionally left 
in the clear when SRTCP is encrypted.  That's XR.  I think this is 
manageable.

Mark
>
> Cheers
>
> Magnus Westerlund
>
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>
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From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Tue, 7 Dec 2004 16:43:46 -0800
To: Colin Perkins <csp@csperkins.org>
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        Magnus Westerlund <magnus.westerlund@ericsson.com>, mmusic@ietf.org
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Colin
   Do we know of any other applications besides RFC 3611 that might need 
to selectively encrypt SRTCP packets?

   I am okay with putting this requirement aside pending a separate 
document.  But before someone goes to the trouble of writing the new 
draft, it might be good to consider if it's needed or not.

Mark
On Dec 7, 2004, at 8:53 AM, Colin Perkins wrote:

> On 7 Dec 2004, at 15:23, Flemming Andreasen wrote:
>> Magnus Westerlund wrote:
>>> Mark Baugher wrote:
>>>> Yes it does.  There are two possible approaches.  We have an 
>>>> existing
>>>> parameter called UNENCRYPTED_SRTCP, but the default is to encrypt 
>>>> the
>>>> SRTCP.  We could add a parameter specific to RFC 3611 or SRTCP XR
>>>> packets such as UNENCRYPTED_SRTCP_XR, which would mean that SRTCP is
>>>> encrypted but the XR packets are to be left in the clear (and 
>>>> signaled
>>>> using the E bit).  That's my current default if no one speaks up 
>>>> with
>>>> an alternative preference.
>>>
>>> I think one should consider that there actually may be need to 
>>> negotiate what should be left unencrypted, or the other way around. 
>>> Therefore my proposal would be to define a "PARTIALENCRYPTED_SRTCP" 
>>> and then another negotiated attribute listing the RTCP packet types 
>>> that should be protected or the other way around. One can also 
>>> consider if the normal case is based on senders local policy.
>>>
>> Hmm, I think we are getting on a slippery slope here with an extended 
>> scope, which is part of my initial concern with the whole idea of 
>> partially encrypted streams for the benefit of intermediate snoopers. 
>> What about the RTP stream ? Should we allow different treatment of 
>> different RTP packets, e.g. based on payload type, etc. ? Taking a 
>> step back, what are the motivations behind all of this added 
>> complexity and is this really the right time to expand the scope this 
>> way ?
>
> All good points. How difficult would it be to write this up as a 
> separate draft? That would allow the "sdescriptions" work to progress 
> quickly, and allow us time to analyse the requirements for partial 
> encryption, write a considered specification of the various 
> parameters, and enumerate the new security considerations that might 
> arise.
>
> Colin
>
>
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From mmusic-bounces@ietf.org  Tue Dec  7 23:20:04 2004
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Subject: [MMUSIC] ICE issue: RTP specificity of the candidate attribute
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Folks,

I got some private comments on the ICE-03 spec, and in the process of 
going through them, a few of them were significant enough that they 
require group discussion.

The issue that was raised here is that the SDP candidate attribute is 
currently very RTP specific. This would make it necessary to define new 
SDP attributes for other transport protocols that wish to use the ICE 
framework. It also means that its hard to handle cases like RFC 2733, 
where there is an IP/port conveyed in an fmtp line.

The suggestion is to change the syntax of the candidate attribute so 
that its not RTP specific. It would do this by specifying a potential 
candidate for a particular IP/port that is indicated explicitly. Thus:

  candidate-attribute   = "candidate" ":" id SP qvalue SP
                            user-frag SP password SP
                            unicast-address SP port
                            candidate-addr SP candidate-port


So, for example:
  v=0
  o=jdoe 2890844526 2890842807 IN IP4 10.47.16.5
  s=SDP Seminar
  c=IN IP4 1.2.3.4
  a=recvonly
  m=audio 49170 RTP/AVP 0
  a=candidate 1 0.4 adsasda 9as8dasd 1.2.3.4 49170 10.0.1.1 8700
  a=candidate 1 0.4 asf9fdf8 00d-ffas 1.2.3.4 49171 10.0.1.1 8765

In this example, it means that the RTP IP/port of 1.2.3.4:49170 has an 
alternate of 10.0.1.1:8700, and the RTCP IP/port of 1.2.3.4:49171 has an 
alternate of 10.0.1.1:8765.


The pros/cons of this proposal as I see them are:

Pros
   - makes ICE more RTP independent
   - works better with RFC 2733 and other protocols which might convey 
IP addresses in SDP in other places besides the c/m lines

Cons
   - increases SDP message size for the normal RTP case, since you 
always need to include to a=candidate lines for each media stream (one 
for RTP, one for RTCP)
   - yet another non-backwards compatible change in the grammar with the 
previous draft (not clear it matters)


I'm inclined to make the change. Comments?

-Jonathan R.
-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
http://www.jdrosen.net                         PHONE: (973) 952-5000
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From: Jonathan Rosenberg <jdrosen@cisco.com>
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Subject: [MMUSIC] ICE issue: obfuscating candidate addresses
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In ICE-03, the IP addresses in the a=candidate attribute are "in the 
clear". If the SDP is not protected (for example using SIP over TLS or 
S/MIME), a "helpful" NAT which sees the SDP may try to rewrite the IP 
addresses it sees in the payload. The STUN revision 
(draft-ietf-behave-rfc3489bis) addressed this by including an obfuscated 
version of the address in the STUN response, so these "helpful" nats 
couldn't rewrite them.

I got a private comment suggesting that we do the same for ICE, to 
prevent the same problem. Its less severe here, since SIP over TLS will 
eliminate the problem (you can't run STUN BindingRequests over TCP). 
However, for UDP it is a similar problem.

So, the proposal would be to encode the IP address/port in the SDP by 
xor'ing it with the lower bits of the username field.

The pros/cons of this approach are:

PROS

   - avoids incorrect ICE operation when "helpful" nats try to rewrite 
IP addresses they see in packets

CONS

   - debugging harder since IP/port are not directly readable
   - result of xor needs to be base64 encoded (or something similar) 
since the xor operation may not produce valid SDP characters
   - slightly larger SDP messages due to base64 encoding


I'm somewhat on the fence here since the right answer is really "use 
TLS". The costs are relatively low, but this type of NAT is not that 
common, so the risk is low as well. I'm probably inclined to not make 
the change, and just document the issue.

Comments?

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                   600 Lanidex Plaza
Director, Service Provider VoIP Architecture   Parsippany, NJ 07054-2711
Cisco Systems
jdrosen@cisco.com                              FAX:   (973) 952-5050
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Since ICE is not RTP-specific, we should make it general. Also, the 2733 
case shows that lack of generality may even be a problem for RTP 
streams. Comparing the -03 attribute definition with the proposal here, 
the overall size increase is not that great (nowhere near a doubling). 
While you could squeeze out a few bytes with some optimizations in the 
definition, I doubt it is worth it. In other words, I agree with the 
proposed change.

Btw, your example shows the same "id" value for the two candidates - is 
this intentional (implying some kind of grouping) ?

-- Flemming

Jonathan Rosenberg wrote:

> Folks,
>
> I got some private comments on the ICE-03 spec, and in the process of 
> going through them, a few of them were significant enough that they 
> require group discussion.
>
> The issue that was raised here is that the SDP candidate attribute is 
> currently very RTP specific. This would make it necessary to define 
> new SDP attributes for other transport protocols that wish to use the 
> ICE framework. It also means that its hard to handle cases like RFC 
> 2733, where there is an IP/port conveyed in an fmtp line.
>
> The suggestion is to change the syntax of the candidate attribute so 
> that its not RTP specific. It would do this by specifying a potential 
> candidate for a particular IP/port that is indicated explicitly. Thus:
>
>  candidate-attribute   = "candidate" ":" id SP qvalue SP
>                            user-frag SP password SP
>                            unicast-address SP port
>                            candidate-addr SP candidate-port
>
>
> So, for example:
>  v=0
>  o=jdoe 2890844526 2890842807 IN IP4 10.47.16.5
>  s=SDP Seminar
>  c=IN IP4 1.2.3.4
>  a=recvonly
>  m=audio 49170 RTP/AVP 0
>  a=candidate 1 0.4 adsasda 9as8dasd 1.2.3.4 49170 10.0.1.1 8700
>  a=candidate 1 0.4 asf9fdf8 00d-ffas 1.2.3.4 49171 10.0.1.1 8765
>
> In this example, it means that the RTP IP/port of 1.2.3.4:49170 has an 
> alternate of 10.0.1.1:8700, and the RTCP IP/port of 1.2.3.4:49171 has 
> an alternate of 10.0.1.1:8765.
>
>
> The pros/cons of this proposal as I see them are:
>
> Pros
>   - makes ICE more RTP independent
>   - works better with RFC 2733 and other protocols which might convey 
> IP addresses in SDP in other places besides the c/m lines
>
> Cons
>   - increases SDP message size for the normal RTP case, since you 
> always need to include to a=candidate lines for each media stream (one 
> for RTP, one for RTCP)
>   - yet another non-backwards compatible change in the grammar with 
> the previous draft (not clear it matters)
>
>
> I'm inclined to make the change. Comments?
>
> -Jonathan R.

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To: Jonathan Rosenberg <jdrosen@cisco.com>
Subject: Re: [MMUSIC] ICE issue: obfuscating candidate addresses
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I'm on the fence too, but UDP without S/MIME is out there, and if the 
problem warrants a solution in STUN, it begs the question of why it 
wouldn't warrant a similar solution here. I'm open either way.

-- Flemming

Jonathan Rosenberg wrote:

> In ICE-03, the IP addresses in the a=candidate attribute are "in the 
> clear". If the SDP is not protected (for example using SIP over TLS or 
> S/MIME), a "helpful" NAT which sees the SDP may try to rewrite the IP 
> addresses it sees in the payload. The STUN revision 
> (draft-ietf-behave-rfc3489bis) addressed this by including an 
> obfuscated version of the address in the STUN response, so these 
> "helpful" nats couldn't rewrite them.
>
> I got a private comment suggesting that we do the same for ICE, to 
> prevent the same problem. Its less severe here, since SIP over TLS 
> will eliminate the problem (you can't run STUN BindingRequests over 
> TCP). However, for UDP it is a similar problem.
>
> So, the proposal would be to encode the IP address/port in the SDP by 
> xor'ing it with the lower bits of the username field.
>
> The pros/cons of this approach are:
>
> PROS
>
>   - avoids incorrect ICE operation when "helpful" nats try to rewrite 
> IP addresses they see in packets
>
> CONS
>
>   - debugging harder since IP/port are not directly readable
>   - result of xor needs to be base64 encoded (or something similar) 
> since the xor operation may not produce valid SDP characters
>   - slightly larger SDP messages due to base64 encoding
>
>
> I'm somewhat on the fence here since the right answer is really "use 
> TLS". The costs are relatively low, but this type of NAT is not that 
> common, so the risk is low as well. I'm probably inclined to not make 
> the change, and just document the issue.
>
> Comments?
>
> -Jonathan R.
>

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Hi Jonathan,

I think this is the right direction. I do however wonder if it is the 
right solution. Your current solution indicates multiple lines that are 
part of the same address candidate. I think using a ABNF that kept each 
candidate on a single line might be better. As all address for the same 
candidate is needed, due to the protocols need, like both RTP and RTCP 
addresses for RTP. The question here is if one should use explicit 
binding as you propose, or rely on the order defined by the protocol 
when specifying address. However it might be easier to do the binding 
explicitly. My proposal for ABNF for a single line solution would be:

candidate-attribute   = "candidate" ":" id SP qvalue SP
                          user-frag SP password 1*(SP address-spec)
address-spec          =  (unicast-address "/" port ";"
                          candidate-addr "/" candidate-port )



Jonathan Rosenberg wrote:
> Folks,
> 
> I got some private comments on the ICE-03 spec, and in the process of
> going through them, a few of them were significant enough that they
> require group discussion.
> 
> The issue that was raised here is that the SDP candidate attribute is
> currently very RTP specific. This would make it necessary to define new
> SDP attributes for other transport protocols that wish to use the ICE
> framework. It also means that its hard to handle cases like RFC 2733,
> where there is an IP/port conveyed in an fmtp line.
> 

I would like to point out that I at least are trying to deprecate that 
type of usage, due to its unsuitability in SDP. As we have deployed a 
number of SDP extension, that applies to the media session, specifying 
another session inside FMTP lines looses all these extensions. Thus my 
personal opinion is that it is easier to remove the few special cases 
like RFC 2733 than to get the other mechanisms to work for the special 
cases. That is why I have argued in the ULP context to change the 
binding between original data and FEC data.


Cheers

Magnus Westerlund

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

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Hi Jonathan,

A few questions:

- Are these NATs that are helpfull common?

- How can these helpful NATs work at all. Even if I only replace bytes 
that matches the private source IP address of an outgoing packet. There 
must be significant risk that the packet contains that pattern in 
something that is not and adress. Do they only do this is certain 
identfiable protcols?

- Finnally if they only replace addresses related to the private 
adresses, would an replacement actually have any ill-effect on ICE? It 
would foul up the default address, however with a explicit syntax, only 
the default and binding adress woudl be replaced, not the candidate? 
Unless the candidate is the private source adress, which anyway becomes 
unvalid by the passing through the NAT.

So please consider a bit more if this really is useful.

Cheers

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
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	c:"From: Mark Baugher <mbaugher@cisco.com>";
	c:"Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP";
	c:"Date: Wed, 8 Dec 2004 12:30:58 -0800"
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Michael
   Here's my take on the security of all of this:  Endpoints can choose 
to negotiate UNENCRYPTED_SRTCP or refuse such an offer, but the goal is 
to not force use of UNENCRYPTED_SRTCP when only the XR packet needs to 
be in the clear.  If we enabled the E bit for XR packets, that would 
not prevent an endpoint from rejecting the offer or any offer in which 
the SRTCP is not encrypted.

   If one does not want to expose information in the XR packets, then 
reject UNENCRYPTED_SRTCP or any offer that allows use of the E bit (in 
the future).  I don't see any security problem apart from having no 
recourse except to offer UNENCRYPTED_SRTCP when all that is desired is 
to leave one SRTCP packet type in the clear.

   Besides that, all sequence numbers are always in the clear in SRTP.

Mark
On Dec 8, 2004, at 11:01 AM, Michael A. Ramalho wrote:

> At 09:02 AM 12/2/2004 -0800, Mark Baugher wrote:
>> Dan
>>   I think use of the E bit needs to be explicitly signaled the SDP 
>> security descriptions.  That way, an offerer or answerer can express 
>> whether it wants to have all SRTCP messages encrypted or allow some 
>> to be in the clear.  I don't know if we need to call out the specific 
>> SRTCP messages that may be left in the clear or do make it "general 
>> purpose."
>
> Mark,
>
> I was looking into this last week. I think it would be unwise to leave
> UNencrypted any RTCP block type that has RTP Sequence Numbers
> in it (as they would be in the clear).
>
> For example, the Loss RLE block (a RTCP-XR block) has information
> on the base RTP sequence numbers the Loss RLE block reports on.
> There are many such RTCP-XR block types (but the VoIP Metrics
> Block is not one of these :-)).
>
> This would expose a "guess-able range" for *part of* the packet index
> used in SRTP (as I remember it, the SRTP packet index uses the
> RTP sequence number for low order portion of the packet index).
> I think this would be a no-no for the cryptographic experts.
>
>> A general-purpose approach would be to signal that ENCRYPTED_SRTP can 
>> be overridden for any packet for which the sender chooses to clear 
>> the E bit.  The general-purpose mechanism might be suitable for going 
>> into draft-ietf-mmusic-sdescriptions-08.txt.  If we want to 
>> explicitly signal that the E bit can be clear just for XR packets 
>> when UNENCRYPTED_SRTCP is set, then I think that belongs in RFC 
>> 3611bis.
>
> If my assertion/guess above is correct, then I'd only recommend the
> option to leave in the clear only those RTCP-XR packets that do not
> expose information sensitive to the SRTP context. This would be a 
> trivial
> exercise (to look at the blocks and decide which ones you do not want
> sent in the clear) but does complicate the permutations/combinations of
> what would need to be signaled in SDP.
>
> And then, since RTPC blocks may be piggybacked, one must take
> care not to have*potentially unencrptable* blocks piggybacked with
> other "normally encrypted" blocks. I still don't like the potential 
> interaction
> between the "encryption" and "RTCP" layers here.
>
> Michael Ramalho
>
>
>> Mark
>> On Dec 2, 2004, at 3:45 AM, Romascanu, Dan ((Dan)) wrote:
>>
>>> I would be cautious in saying that in no circumstances the contents 
>>> of an XR payload would be remotely useful to an eavesdropper. We may 
>>> encounter some very security aware environments where even the fact 
>>> that a call happens between two IP phones, its duration, or QoS 
>>> parameters need to be kept confidential. For this reason, while 
>>> working to provide a strong recommendation for RTCP XR packets not 
>>> to be encrypted, I do not believe that we can make this a MUST 
>>> requirements. As an alternative, I suggest to use  RAQMON extensions 
>>> to report the RTCP XR metrics to RAQMON collectors without the 
>>> security risks that may be of concern in some environments.
>>>
>>> Regards,
>>>
>>> Dan
>>>
>>>
>>>
>>>> -----Original Message-----
>>>> From: mmusic-bounces@ietf.org
>>>> [mailto:mmusic-bounces@ietf.org]On Behalf Of Alan Clark
>>>> Sent: 01 December, 2004 2:08 PM
>>>> To: Mark Baugher
>>>> Cc: mmusic@ietf.org
>>>> Subject: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>>>>
>>>>
>>>> Mark
>>>>
>>>> To provide some background to the list - RFC3611 states the
>>>> following:-
>>>>
>>>>    "Any encryption or filtering of XR report blocks entails a loss 
>>>> of
>>>>    monitoring information to third parties.  For example, a
>>>> network that
>>>>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>>>>    information to the service providers traversed by the tunnel.  
>>>> The
>>>>    service providers cannot then monitor or respond to the quality 
>>>> of
>>>>    the VoIP calls that they carry, potentially creating
>>>> problems for the
>>>>    network's users.  As a default, XR packets should not be
>>>> encrypted or
>>>>    filtered."
>>>>
>>>> I can't conceive of any circumstance in which the contents of
>>>> an XR payload
>>>> would be remotely useful to an eavesdropper and hence stating that 
>>>> XR
>>>> payloads should always be left unencrypted would be (at least
>>>> in my opinion)
>>>> ok.  If we wanted to make this even more specific we could narrow 
>>>> this
>>>> requirement down to only XR VoIP Metrics payloads to avoid
>>>> the situation in
>>>> which some future XR payload may be problematic.
>>>>
>>>> The availability of VoIP metrics payloads will be a huge
>>>> issue for service
>>>> providers - as we move through 2005 RTCP XR will be a mainstay of 
>>>> VoIP
>>>> performance management and is already a mandatory requirement
>>>> within certain
>>>> sectors of the industry.  Making the VoIP metrics payload
>>>> unencrypted by
>>>> default may seem a small issue today however it will have
>>>> major consequences
>>>> to the industry as SRTP starts to roll out.
>>>>
>>>> Regards
>>>>
>>>> Alan
>>>>
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: Mark Baugher [mailto:mbaugher@cisco.com]
>>>> Sent: Tuesday, November 30, 2004 6:45 PM
>>>> To: Alan Clark
>>>> Cc: mmusic@ietf.org
>>>> Subject: Re: [AVT] RFC 3711 and RTCP
>>>>
>>>>
>>>> Alan
>>>>    I am copying MMUSIC (and not cross-posting to AVT) to
>>>> further discuss
>>>> a possible change to SDP security descriptions, which has completed 
>>>> WG
>>>> last call.  The change is needed for RFC 3611, and we also 
>>>> discovered
>>>> that
>>>> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
>>>> -07.txt does not address the SRTCP E bit properly. I offered the
>>>> following text in a posting to AVT:
>>>> "As the default behavior when UNENCRYPTED_SRTCP is signaled, all 
>>>> SRTCP
>>>> packets MUST have the E bit set to zero and MUST NOT be encrypted.
>>>> Otherwise, all SRTP packets MUST have the E bit set to one and MUST 
>>>> be
>>>> encrypted by default.  Individual applications MAY override these
>>>> default behaviors in their specifications.  In a session where SRTCP
>>>> packets are encrypted, therefore, applications that use RFC 3611 MAY
>>>> set the E bit to 0 for XR packets as recommended by RFC 3611."
>>>>
>>>> One of my co-authors pointed out that the last sentence is
>>>> more than an
>>>> editorial change.  We also don't like the fact that an offerer or
>>>> answerer cannot explicitly signal when it wants the XR packet to be
>>>> encrypted or unencrypted.  In other words, if ENCRYPTED_SRTCP is the
>>>> default setting and RFC 3611 is also signaled, it is ambiguous as to
>>>> whether the SRTCP sender will encrypt the XR packet or not.  It's
>>>> solely up to the sender.  If the offerer or answerer want the
>>>> XR packet
>>>> to always be encrypted, there is no way to signal that.  Ditto for 
>>>> the
>>>> unencrypted case.
>>>>
>>>> I think we need to put the first three sentences of the above-quoted
>>>> paragraph into SDP security descriptions as an editorial change.  
>>>> I'm
>>>> proposing this to the working group.  I also propose not to
>>>> include the
>>>> last sentence.  Instead, we could add a new signaling parameter,
>>>> UNENCRYPTED_XR, which implies that all SRTCP messages are to be
>>>> encrypted except for the XR packets, which will have their E
>>>> bit clear,
>>>> of course.
>>>>
>>>> This new parameter could be placed in an RFC 3611bis or in a 
>>>> revision
>>>> of draft-ietf-mmusic-sdescriptions-07.txt, which will then need to
>>>> undergo another WG last call.  What do you (all) think about this?
>>>>
>>>> Mark
>>>>
>>>> On Nov 24, 2004, at 4:18 PM, Mark Baugher wrote:
>>>>
>>>>> Alan
>>>>>   As Dave Oran has pointed out, the definition that you want is not
>>>>> supported in the signaling, at present.  The current definition of
>>>>> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
>>>>> -07.txt treats SRTCP encryption as either on or off and does not
>>>>> address the problem of setting the E bit in SRTCP packets.
>>>> This is a
>>>>> bug that we need to fix.
>>>>>
>>>>>   MMUSIC has finished last call on
>>>>> draft-ietf-mmusic-sdescriptions-07.txt, and the transport ADs have
>>>>> reviewed it.  We'll be making some changes at Jon Peterson's 
>>>>> request
>>>>> and post another version in about a week.  This is an opportunity 
>>>>> to
>>>>> address the E bit, but the MMUSIC chairs or transport ADs may want
>>>>> another WG last call.  Off the top of my head, I would
>>>> offer something
>>>>> like the following.
>>>>>
>>>>> "As the default behavior when UNENCRYPTED_SRTCP is
>>>> signaled, all SRTCP
>>>>> packets MUST have the E bit set to zero and MUST NOT be encrypted.
>>>>> Otherwise, all SRTP packets MUST have the E bit set to one
>>>> and MUST be
>>>>> encrypted by default.  Individual applications MAY override these
>>>>> default behaviors in their specifications.  In a session where 
>>>>> SRTCP
>>>>> packets are encrypted, therefore, applications that use RFC 3611 
>>>>> MAY
>>>>> set the E bit to 0 for XR packets as recommended by RFC 3611."
>>>>>
>>>>> Mark
>>>>>
>>>>> On Nov 24, 2004, at 2:33 PM, Alan Clark wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> Mark
>>>>>>
>>>>>> We tried to flag this in RFC3611 with the following text in the
>>>>>> "security
>>>>>> considerations" section
>>>>>>
>>>>>>    Any encryption or filtering of XR report blocks entails
>>>> a loss of
>>>>>>    monitoring information to third parties.  For example, a 
>>>>>> network
>>>>>> that
>>>>>>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>>>>>>    information to the service providers traversed by the
>>>> tunnel.  The
>>>>>>    service providers cannot then monitor or respond to the
>>>> quality of
>>>>>>    the VoIP calls that they carry, potentially creating
>>>> problems for
>>>>>> the
>>>>>>    network's users.  As a default, XR packets should not
>>>> be encrypted
>>>>>> or
>>>>>>    filtered.
>>>>>>
>>>>>> Could we potentially interpret this as being the
>>>> definition of what to
>>>>>> selectively encrypt in the case of the RFC3611 specification?
>>>>>>
>>>>>> Regards
>>>>>>
>>>>>> Alan
>>>>>>
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: Mark Baugher [mailto:mbaugher@cisco.com]
>>>>>> Sent: Wednesday, November 24, 2004 2:42 PM
>>>>>> To: Alan Clark
>>>>>> Cc: Oren Peleg; Cullen Jennings; avt@ietf.org; David R
>>>> Oran; Michael
>>>>>> A.
>>>>>> Ramalho
>>>>>> Subject: Re: [AVT] RFC 3711 and RTCP
>>>>>>
>>>>>>
>>>>>> So, when the sender and receiver agree to encrypt SRTCP, this 
>>>>>> would
>>>>>> mean
>>>>>> only that some SRTCP messages would be encrypted and the
>>>> sender would
>>>>>> decide which ones to encrypt?  My personal opinion is that
>>>> this is not
>>>>>> secure because it is unpredictable to the receiver, which
>>>> may have a
>>>>>> policy that all messages related to its session be encrypted.
>>>>>>
>>>>>> I can think of a few potential approaches to this problem.
>>>>>>
>>>>>> 1. Let individual SRTCP messages or a bundle of SRTCP messages
>>>>>> be signaled to be unencrypted when SRTCP encryption is used.
>>>>>>
>>>>>> 2. Make a blanket policy that the sender determine what
>>>>>> to selectively encrypt, as you suggest.
>>>>>>
>>>>>> 3. Define the policy of what to selectively encrypt in
>>>>>> the particular specification.  In other words, a document could
>>>>>> re-define the signaling of SRTCP encryption according to its
>>>>>> needs.  The specification of foo, therefore, could define a new
>>>>>> parameter such as foo_ENCRYPTED_SRTCP to be used in place of
>>>>>> ENCRYPTED_SRTCP.  The foo document would define what gets
>>>>>> encrypted and what does not get encrypted as per its needs.
>>>>>>
>>>>>> I commented on #2 already.  I don't like #1 because of its
>>>> signaling
>>>>>> complexity.  I like #3 because it limits the complexity to the
>>>>>> specific needs of an application.  #1 would be more efficient than
>>>>>> #3 if lots of applications had their own specific needs for
>>>>>> selective encryption of SRTCP message.  But I don't think this is
>>>>>> the case.
>>>>>>
>>>>>> Mark
>>>>>>
>>>>>> On Nov 24, 2004, at 10:58 AM, Alan Clark wrote:
>>>>>>
>>>>>>>
>>>>>>> If the use of encryption in SRTCP is signalled by an
>>>> explicit (E) bit
>>>>>>> in the
>>>>>>> SRTCP header then perhaps the sender could autonomously decide
>>>>>>> whether
>>>>>>> it
>>>>>>> wants to encrypt?  Presumably since the sender is in the best
>>>>>>> position
>>>>>>> to
>>>>>>> decide if the data is sensitive then this would be ok - and the
>>>>>>> receiver is
>>>>>>> able to just look at the E bit and hence recognize that
>>>> decryption is
>>>>>>> needed.
>>>>>>>
>>>>>>> Alan
>>>>>>>
>>>>>>> -----Original Message-----
>>>>>>> From: avt-bounces@ietf.org
>>>> [mailto:avt-bounces@ietf.org]On Behalf Of
>>>>>>> Mark Baugher
>>>>>>> Sent: Wednesday, November 24, 2004 1:33 PM
>>>>>>> To: David R Oran
>>>>>>> Cc: Cullen Jennings; Oren Peleg; Michael A. Ramalho; Alan Clark;
>>>>>>> avt@ietf.org
>>>>>>> Subject: Re: [AVT] RFC 3711 and RTCP
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Nov 24, 2004, at 10:05 AM, David R Oran wrote:
>>>>>>>
>>>>>>>>
>>>>>>>> On Nov 24, 2004, at 12:23 PM, Alan Clark wrote:
>>>>>>>>
>>>>>>>>>
>>>>>>>>> Michael
>>>>>>>>>
>>>>>>>>> I don't see why this would be a problem unless you included 
>>>>>>>>> SDES
>>>>>>>>> and
>>>>>>>>> (SR or
>>>>>>>>> RR or XR) in the same RTCP packet.  SRTP (at least the
>>>> protocol) is
>>>>>>>>> apparently able to decided on a packet by packet basis
>>>> whether to
>>>>>>>>> encrypt
>>>>>>>>> RTCP - hopefully SRTP implementations also permit this.
>>>>>>>> Sure, they can do this. But remember offer-answer. How does a
>>>>>>>> receiver
>>>>>>>> communicate such a granular policy to the transmitter?
>>>> Right now if
>>>>>>>> can say whether to auth or auth+encrypt RTP, and auth or
>>>>>>>> auth+encrypt
>>>>>>>> RTCP. There's no way to say auth+encrypt SDES but only auth 
>>>>>>>> SR/XR
>>>>>>>> etc.
>>>>>>>
>>>>>>> Seems to me that this is a problem with SDP security 
>>>>>>> descriptions.
>>>>>>> That problem might be there because we forgot about the E
>>>> bit.  This
>>>>>>> seems very messy for the reason you mention above -
>>>> granularity.  I
>>>>>>> don't think that we want to negotiate the encryption
>>>> policy for each
>>>>>>> type SDP message, which is a problem with something like
>>>> the E bit.
>>>>>>>
>>>>>>> Mark
>>>>>>>>
>>>>>>>>
>>>>>>>>> Logically I'd send SDES on its own at the start of a call to
>>>>>>>>> provide
>>>>>>>>> some
>>>>>>>>> user info, but would not send SR/RR/XR until at least
>>>> 8-10 seconds
>>>>>>>>> into the
>>>>>>>>> call as I need some time to gather statistics to report.
>>>>>>>> That's not exactly what the RTP specs say. RTCP timers don't 
>>>>>>>> have
>>>>>>>> this
>>>>>>>> kind of policy today. In fact, there are porposals on
>>>> the table to
>>>>>>>> use
>>>>>>>> RTCP for early feedback to assess reception quality on
>>>> some initial
>>>>>>>> packets - e.g. the RTP-noop proposal. In any event that has no
>>>>>>>> bearing
>>>>>>>> on the present discussion.
>>>>>>>>
>>>>>>>>> After the
>>>>>>>>> start-of-call exchange there would appear to be minimal need to
>>>>>>>>> include
>>>>>>>>> SDES. I'm sure we could construct some scenarios in
>>>> which you "may"
>>>>>>>>> need to
>>>>>>>>> include SDES with SR/RR/XR however the vast majority of
>>>> cases would
>>>>>>>>> either
>>>>>>>>> need only a single initial exchange or may not need SDES at 
>>>>>>>>> all.
>>>>>>>> Any time you change SSRC you potentially need to send SDES.
>>>>>>>> That said, I still don't see what bearing this has on
>>>> the discussion
>>>>>>>> about whether we need the complexity of encrypting some
>>>> RTCP packets
>>>>>>>> and not others for a single session.
>>>>>>>>
>>>>>>>> It's a tradeoff between extra complexity+less privacy versus
>>>>>>>> allowing
>>>>>>>> snoopers to gather RTCP stats.
>>>>>>>>
>>>>>>>> Dave.
>>>>>>>>
>>>>>>>>> Regards
>>>>>>>>>
>>>>>>>>> Alan
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> -----Original Message-----
>>>>>>>>> From: Michael A. Ramalho [mailto:mramalho@cisco.com]
>>>>>>>>> Sent: Wednesday, November 24, 2004 9:39 AM
>>>>>>>>> To: Alan Clark
>>>>>>>>> Cc: Oren Peleg; avt@ietf.org
>>>>>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> At 07:54 AM 11/24/2004 -0500, Alan Clark wrote:
>>>>>>>>>> Hi Oren
>>>>>>>>>>
>>>>>>>>>> Yes - encryption is intended to provide privacy however this
>>>>>>>>>> relates
>>>>>>>>>> primarily to the voice payloads and to information within RTCP
>>>>>>>>>> that
>>>>>>>>>> may be
>>>>>>>>>> associated with individual users (e.g. SDES).
>>>>>>>>>>
>>>>>>>>>> SR, RR and XR contain only peformance statistics and
>>>> there is no
>>>>>>>>>> privacy
>>>>>>>>>> consideration that I can think of related to this type of 
>>>>>>>>>> data.
>>>>>>>>>>
>>>>>>>>>> The message is - encrypt RTP, optionally encrypt RTCP SDES but
>>>>>>>>>> don't
>>>>>>>>> encrypt
>>>>>>>>>> RTCP SR/RR/XR
>>>>>>>>>
>>>>>>>>> Hi Alan,
>>>>>>>>>
>>>>>>>>> I am wondering about implementations utilizing your advise
>>>>>>>>> "optionally
>>>>>>>>> encrypt
>>>>>>>>> RTCP SDES but don't encrypt RTCP SR/RR/XR".
>>>>>>>>>
>>>>>>>>> Is there an easy way to tell SRTP to encrypt RTCP SDES, but not
>>>>>>>>> RTCP
>>>>>>>>> SR/RR/XR? It seems that all the RTCP blocks would otherwise be
>>>>>>>>> highly
>>>>>>>>> related to each other. For example (quote out of RFC 3611):
>>>>>>>>>
>>>>>>>>> "XR packets supplement the existing RTCP packets, and may be
>>>>>>>>> stacked
>>>>>>>>> with
>>>>>>>>> other RTCP packets to form compound RTCP packets [9,
>>>> Section 6]."
>>>>>>>>>
>>>>>>>>> Thus, RFC3611 explicitly admits the possibility of (but does 
>>>>>>>>> not
>>>>>>>>> mandate)
>>>>>>>>> the piggybacking all the RTCP-XR block types.
>>>>>>>>>
>>>>>>>>> RTCP implementations would have to know the SRTP
>>>> encryption rules
>>>>>>>>> to
>>>>>>>>> make sure they don't perform piggybacking with SDES ...
>>>> but could
>>>>>>>>> allow
>>>>>>>>> (the more bandwidth efficient) piggybacking on the other 
>>>>>>>>> RTCP-XR
>>>>>>>>> blocks.
>>>>>>>>>
>>>>>>>>> This RTCP/SRTP interaction/dependency seems to add a
>>>> burden to both
>>>>>>>>> RTCP and SRTP implementations ... let alone the expressiveness
>>>>>>>>> required
>>>>>>>>> to signal your advise in existing SDP/SRTP mechanisms.
>>>>>>>>>
>>>>>>>>> Are you aware of any implementations that have actually 
>>>>>>>>> followed
>>>>>>>>> your
>>>>>>>>> advise above? I rather go into this one with our eyes open ...
>>>>>>>>> developers
>>>>>>>>> highly dislike this cross-functional entity dependency
>>>> (RTCP & SRTP
>>>>>>>>> along
>>>>>>>>> with expressing it in SDP/SRTP mechanisms).
>>>>>>>>>
>>>>>>>>> Regards,
>>>>>>>>>
>>>>>>>>> Michael Ramalho
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> Regards
>>>>>>>>>>
>>>>>>>>>> Alan
>>>>>>>>>>
>>>>>>>>>> -----Original Message-----
>>>>>>>>>> From: Oren Peleg [mailto:OrenP@audiocodes.com]
>>>>>>>>>> Sent: Wednesday, November 24, 2004 7:50 AM
>>>>>>>>>> To: Alan Clark; avt@ietf.org
>>>>>>>>>> Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> Isnt encryption is to prevent monitoring from the first place?
>>>>>>>>>>
>>>>>>>>>> -----Original Message-----
>>>>>>>>>> From: avt-bounces@ietf.org
>>>> [mailto:avt-bounces@ietf.org] On Behalf
>>>>>>>>>> Of
>>>>>>>>>> Alan Clark
>>>>>>>>>> Sent: Tuesday, November 23, 2004 5:42 PM
>>>>>>>>>> To: avt@ietf.org
>>>>>>>>>> Subject: [AVT] RFC 3711 and RTCP
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> From the flurry of discussion on RFC3711 it looks as
>>>> if there is
>>>>>>>>>> considerable interest in SRTP implementation.  I'd
>>>> like to add a
>>>>>>>>>> request
>>>>>>>>>> to
>>>>>>>>>> SRTP implementors with regard to manageability.
>>>>>>>>>>
>>>>>>>>>> RTCP SR, RR and more particularly XR (e.g. VoIP Metrics report
>>>>>>>>>> block)
>>>>>>>>>> contain useful information for network monitoring, and
>>>> are often
>>>>>>>>>> used by
>>>>>>>>>> probes and analyzers to support problem diagnosis.
>>>> RFC3711 allows
>>>>>>>>>> the
>>>>>>>>>> selective encryption of RTCP payloads and hence I
>>>> would strongly
>>>>>>>>>> recomment
>>>>>>>>>> that RFC3711 implementors DO NOT encrypt RTCP SR, RR and XR
>>>>>>>>>> payloads
>>>>>>>>>> as
>>>>>>>>>> this
>>>>>>>>>> adversely impacts manageability.
>>>>>>>>>>
>>>>>>>>>> Regards
>>>>>>>>>>
>>>>>>>>>> Alan Clark
>>>>>>>>>> Telchemy
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Audio/Video Transport Working Group
>>>>>>>>>> avt@ietf.org
>>>>>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>>>>>
>>>> ******************************************************************
>>>>>>>>>> **
>>>>>>>>>> *
>>>>>>>>>> ******
>>>>>>>>> *
>>>>>>>>>> ********************************
>>>>>>>>>> This email and any files transmitted with it are confidential
>>>>>>>>>> material.
>>>>>>>>>> They are intended solely for the use of the designated
>>>> individual
>>>>>>>>>> or
>>>>>>>>>> entity
>>>>>>>>>> to whom they are addressed.
>>>>>>>>>> If the reader of this message is not the intended recipient,
>>>>>>>>>> you are hereby notified that any dissemination,use,
>>>>>>>>>> distribution or copying of this communication is strictly
>>>>>>>>>> prohibited
>>>>>>>>>> and
>>>>>>>>> may
>>>>>>>>>> be unlawful.
>>>>>>>>>>
>>>>>>>>>> If you have received this email in error please notify
>>>>>>>>>> postmaster@audiocodes.com and permanently delete the e-mail 
>>>>>>>>>> and
>>>>>>>>>> files.
>>>>>>>>>>
>>>>>>>>>>
>>>> ******************************************************************
>>>>>>>>>> **
>>>>>>>>>> *
>>>>>>>>>> ******
>>>>>>>>> *
>>>>>>>>>> *******************************
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Audio/Video Transport Working Group
>>>>>>>>>> avt@ietf.org
>>>>>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Audio/Video Transport Working Group
>>>>>>>>> avt@ietf.org
>>>>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>>> David R. Oran
>>>>>>>> Cisco Fellow
>>>>>>>> Cisco Systems
>>>>>>>> 7 Ladyslipper Lane
>>>>>>>> Acton, MA 01720 USA
>>>>>>>> Tel: +1 978 264 2048
>>>>>>>> Email: oran@cisco.com
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Audio/Video Transport Working Group
>>>>>>>> avt@ietf.org
>>>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Audio/Video Transport Working Group
>>>>>>> avt@ietf.org
>>>>>>> https://www1.ietf.org/mailman/listinfo/avt
>>>>
>>>>
>>>> _______________________________________________
>>>> mmusic mailing list
>>>> mmusic@ietf.org
>>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>>
>>> _______________________________________________
>>> Audio/Video Transport Working Group
>>> avt@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/avt
>>
>>
>> _______________________________________________
>> Audio/Video Transport Working Group
>> avt@ietf.org
>> https://www1.ietf.org/mailman/listinfo/avt
>

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To: Mark Baugher <mbaugher@cisco.com>
From: "Michael A. Ramalho" <mramalho@cisco.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
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At 09:02 AM 12/2/2004 -0800, Mark Baugher wrote:
>Dan
>   I think use of the E bit needs to be explicitly signaled the SDP 
> security descriptions.  That way, an offerer or answerer can express 
> whether it wants to have all SRTCP messages encrypted or allow some to be 
> in the clear.  I don't know if we need to call out the specific SRTCP 
> messages that may be left in the clear or do make it "general purpose."

Mark,

I was looking into this last week. I think it would be unwise to leave
UNencrypted any RTCP block type that has RTP Sequence Numbers
in it (as they would be in the clear).

For example, the Loss RLE block (a RTCP-XR block) has information
on the base RTP sequence numbers the Loss RLE block reports on.
There are many such RTCP-XR block types (but the VoIP Metrics
Block is not one of these :-)).

This would expose a "guess-able range" for *part of* the packet index
used in SRTP (as I remember it, the SRTP packet index uses the
RTP sequence number for low order portion of the packet index).
I think this would be a no-no for the cryptographic experts.

>A general-purpose approach would be to signal that ENCRYPTED_SRTP can be 
>overridden for any packet for which the sender chooses to clear the E 
>bit.  The general-purpose mechanism might be suitable for going into 
>draft-ietf-mmusic-sdescriptions-08.txt.  If we want to explicitly signal 
>that the E bit can be clear just for XR packets when UNENCRYPTED_SRTCP is 
>set, then I think that belongs in RFC 3611bis.

If my assertion/guess above is correct, then I'd only recommend the
option to leave in the clear only those RTCP-XR packets that do not
expose information sensitive to the SRTP context. This would be a trivial
exercise (to look at the blocks and decide which ones you do not want
sent in the clear) but does complicate the permutations/combinations of
what would need to be signaled in SDP.

And then, since RTPC blocks may be piggybacked, one must take
care not to have*potentially unencrptable* blocks piggybacked with
other "normally encrypted" blocks. I still don't like the potential interaction
between the "encryption" and "RTCP" layers here.

Michael Ramalho


>Mark
>On Dec 2, 2004, at 3:45 AM, Romascanu, Dan ((Dan)) wrote:
>
>>I would be cautious in saying that in no circumstances the contents of an 
>>XR payload would be remotely useful to an eavesdropper. We may encounter 
>>some very security aware environments where even the fact that a call 
>>happens between two IP phones, its duration, or QoS parameters need to be 
>>kept confidential. For this reason, while working to provide a strong 
>>recommendation for RTCP XR packets not to be encrypted, I do not believe 
>>that we can make this a MUST requirements. As an alternative, I suggest 
>>to use  RAQMON extensions to report the RTCP XR metrics to RAQMON 
>>collectors without the security risks that may be of concern in some 
>>environments.
>>
>>Regards,
>>
>>Dan
>>
>>
>>
>>>-----Original Message-----
>>>From: mmusic-bounces@ietf.org
>>>[mailto:mmusic-bounces@ietf.org]On Behalf Of Alan Clark
>>>Sent: 01 December, 2004 2:08 PM
>>>To: Mark Baugher
>>>Cc: mmusic@ietf.org
>>>Subject: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>>>
>>>
>>>Mark
>>>
>>>To provide some background to the list - RFC3611 states the
>>>following:-
>>>
>>>    "Any encryption or filtering of XR report blocks entails a loss of
>>>    monitoring information to third parties.  For example, a
>>>network that
>>>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>>>    information to the service providers traversed by the tunnel.  The
>>>    service providers cannot then monitor or respond to the quality of
>>>    the VoIP calls that they carry, potentially creating
>>>problems for the
>>>    network's users.  As a default, XR packets should not be
>>>encrypted or
>>>    filtered."
>>>
>>>I can't conceive of any circumstance in which the contents of
>>>an XR payload
>>>would be remotely useful to an eavesdropper and hence stating that XR
>>>payloads should always be left unencrypted would be (at least
>>>in my opinion)
>>>ok.  If we wanted to make this even more specific we could narrow this
>>>requirement down to only XR VoIP Metrics payloads to avoid
>>>the situation in
>>>which some future XR payload may be problematic.
>>>
>>>The availability of VoIP metrics payloads will be a huge
>>>issue for service
>>>providers - as we move through 2005 RTCP XR will be a mainstay of VoIP
>>>performance management and is already a mandatory requirement
>>>within certain
>>>sectors of the industry.  Making the VoIP metrics payload
>>>unencrypted by
>>>default may seem a small issue today however it will have
>>>major consequences
>>>to the industry as SRTP starts to roll out.
>>>
>>>Regards
>>>
>>>Alan
>>>
>>>
>>>
>>>-----Original Message-----
>>>From: Mark Baugher [mailto:mbaugher@cisco.com]
>>>Sent: Tuesday, November 30, 2004 6:45 PM
>>>To: Alan Clark
>>>Cc: mmusic@ietf.org
>>>Subject: Re: [AVT] RFC 3711 and RTCP
>>>
>>>
>>>Alan
>>>    I am copying MMUSIC (and not cross-posting to AVT) to
>>>further discuss
>>>a possible change to SDP security descriptions, which has completed WG
>>>last call.  The change is needed for RFC 3611, and we also discovered
>>>that
>>>http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
>>>-07.txt does not address the SRTCP E bit properly. I offered the
>>>following text in a posting to AVT:
>>>"As the default behavior when UNENCRYPTED_SRTCP is signaled, all SRTCP
>>>packets MUST have the E bit set to zero and MUST NOT be encrypted.
>>>Otherwise, all SRTP packets MUST have the E bit set to one and MUST be
>>>encrypted by default.  Individual applications MAY override these
>>>default behaviors in their specifications.  In a session where SRTCP
>>>packets are encrypted, therefore, applications that use RFC 3611 MAY
>>>set the E bit to 0 for XR packets as recommended by RFC 3611."
>>>
>>>One of my co-authors pointed out that the last sentence is
>>>more than an
>>>editorial change.  We also don't like the fact that an offerer or
>>>answerer cannot explicitly signal when it wants the XR packet to be
>>>encrypted or unencrypted.  In other words, if ENCRYPTED_SRTCP is the
>>>default setting and RFC 3611 is also signaled, it is ambiguous as to
>>>whether the SRTCP sender will encrypt the XR packet or not.  It's
>>>solely up to the sender.  If the offerer or answerer want the
>>>XR packet
>>>to always be encrypted, there is no way to signal that.  Ditto for the
>>>unencrypted case.
>>>
>>>I think we need to put the first three sentences of the above-quoted
>>>paragraph into SDP security descriptions as an editorial change.  I'm
>>>proposing this to the working group.  I also propose not to
>>>include the
>>>last sentence.  Instead, we could add a new signaling parameter,
>>>UNENCRYPTED_XR, which implies that all SRTCP messages are to be
>>>encrypted except for the XR packets, which will have their E
>>>bit clear,
>>>of course.
>>>
>>>This new parameter could be placed in an RFC 3611bis or in a revision
>>>of draft-ietf-mmusic-sdescriptions-07.txt, which will then need to
>>>undergo another WG last call.  What do you (all) think about this?
>>>
>>>Mark
>>>
>>>On Nov 24, 2004, at 4:18 PM, Mark Baugher wrote:
>>>
>>>>Alan
>>>>   As Dave Oran has pointed out, the definition that you want is not
>>>>supported in the signaling, at present.  The current definition of
>>>>http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdescriptions
>>>>-07.txt treats SRTCP encryption as either on or off and does not
>>>>address the problem of setting the E bit in SRTCP packets.
>>>This is a
>>>>bug that we need to fix.
>>>>
>>>>   MMUSIC has finished last call on
>>>>draft-ietf-mmusic-sdescriptions-07.txt, and the transport ADs have
>>>>reviewed it.  We'll be making some changes at Jon Peterson's request
>>>>and post another version in about a week.  This is an opportunity to
>>>>address the E bit, but the MMUSIC chairs or transport ADs may want
>>>>another WG last call.  Off the top of my head, I would
>>>offer something
>>>>like the following.
>>>>
>>>>"As the default behavior when UNENCRYPTED_SRTCP is
>>>signaled, all SRTCP
>>>>packets MUST have the E bit set to zero and MUST NOT be encrypted.
>>>>Otherwise, all SRTP packets MUST have the E bit set to one
>>>and MUST be
>>>>encrypted by default.  Individual applications MAY override these
>>>>default behaviors in their specifications.  In a session where SRTCP
>>>>packets are encrypted, therefore, applications that use RFC 3611 MAY
>>>>set the E bit to 0 for XR packets as recommended by RFC 3611."
>>>>
>>>>Mark
>>>>
>>>>On Nov 24, 2004, at 2:33 PM, Alan Clark wrote:
>>>>
>>>>>
>>>>>
>>>>>Mark
>>>>>
>>>>>We tried to flag this in RFC3611 with the following text in the
>>>>>"security
>>>>>considerations" section
>>>>>
>>>>>    Any encryption or filtering of XR report blocks entails
>>>a loss of
>>>>>    monitoring information to third parties.  For example, a network
>>>>>that
>>>>>    establishes a tunnel to encrypt VoIP Report Blocks denies that
>>>>>    information to the service providers traversed by the
>>>tunnel.  The
>>>>>    service providers cannot then monitor or respond to the
>>>quality of
>>>>>    the VoIP calls that they carry, potentially creating
>>>problems for
>>>>>the
>>>>>    network's users.  As a default, XR packets should not
>>>be encrypted
>>>>>or
>>>>>    filtered.
>>>>>
>>>>>Could we potentially interpret this as being the
>>>definition of what to
>>>>>selectively encrypt in the case of the RFC3611 specification?
>>>>>
>>>>>Regards
>>>>>
>>>>>Alan
>>>>>
>>>>>
>>>>>-----Original Message-----
>>>>>From: Mark Baugher [mailto:mbaugher@cisco.com]
>>>>>Sent: Wednesday, November 24, 2004 2:42 PM
>>>>>To: Alan Clark
>>>>>Cc: Oren Peleg; Cullen Jennings; avt@ietf.org; David R
>>>Oran; Michael
>>>>>A.
>>>>>Ramalho
>>>>>Subject: Re: [AVT] RFC 3711 and RTCP
>>>>>
>>>>>
>>>>>So, when the sender and receiver agree to encrypt SRTCP, this would
>>>>>mean
>>>>>only that some SRTCP messages would be encrypted and the
>>>sender would
>>>>>decide which ones to encrypt?  My personal opinion is that
>>>this is not
>>>>>secure because it is unpredictable to the receiver, which
>>>may have a
>>>>>policy that all messages related to its session be encrypted.
>>>>>
>>>>>I can think of a few potential approaches to this problem.
>>>>>
>>>>>1. Let individual SRTCP messages or a bundle of SRTCP messages
>>>>>be signaled to be unencrypted when SRTCP encryption is used.
>>>>>
>>>>>2. Make a blanket policy that the sender determine what
>>>>>to selectively encrypt, as you suggest.
>>>>>
>>>>>3. Define the policy of what to selectively encrypt in
>>>>>the particular specification.  In other words, a document could
>>>>>re-define the signaling of SRTCP encryption according to its
>>>>>needs.  The specification of foo, therefore, could define a new
>>>>>parameter such as foo_ENCRYPTED_SRTCP to be used in place of
>>>>>ENCRYPTED_SRTCP.  The foo document would define what gets
>>>>>encrypted and what does not get encrypted as per its needs.
>>>>>
>>>>>I commented on #2 already.  I don't like #1 because of its
>>>signaling
>>>>>complexity.  I like #3 because it limits the complexity to the
>>>>>specific needs of an application.  #1 would be more efficient than
>>>>>#3 if lots of applications had their own specific needs for
>>>>>selective encryption of SRTCP message.  But I don't think this is
>>>>>the case.
>>>>>
>>>>>Mark
>>>>>
>>>>>On Nov 24, 2004, at 10:58 AM, Alan Clark wrote:
>>>>>
>>>>>>
>>>>>>If the use of encryption in SRTCP is signalled by an
>>>explicit (E) bit
>>>>>>in the
>>>>>>SRTCP header then perhaps the sender could autonomously decide
>>>>>>whether
>>>>>>it
>>>>>>wants to encrypt?  Presumably since the sender is in the best
>>>>>>position
>>>>>>to
>>>>>>decide if the data is sensitive then this would be ok - and the
>>>>>>receiver is
>>>>>>able to just look at the E bit and hence recognize that
>>>decryption is
>>>>>>needed.
>>>>>>
>>>>>>Alan
>>>>>>
>>>>>>-----Original Message-----
>>>>>>From: avt-bounces@ietf.org
>>>[mailto:avt-bounces@ietf.org]On Behalf Of
>>>>>>Mark Baugher
>>>>>>Sent: Wednesday, November 24, 2004 1:33 PM
>>>>>>To: David R Oran
>>>>>>Cc: Cullen Jennings; Oren Peleg; Michael A. Ramalho; Alan Clark;
>>>>>>avt@ietf.org
>>>>>>Subject: Re: [AVT] RFC 3711 and RTCP
>>>>>>
>>>>>>
>>>>>>
>>>>>>On Nov 24, 2004, at 10:05 AM, David R Oran wrote:
>>>>>>
>>>>>>>
>>>>>>>On Nov 24, 2004, at 12:23 PM, Alan Clark wrote:
>>>>>>>
>>>>>>>>
>>>>>>>>Michael
>>>>>>>>
>>>>>>>>I don't see why this would be a problem unless you included SDES
>>>>>>>>and
>>>>>>>>(SR or
>>>>>>>>RR or XR) in the same RTCP packet.  SRTP (at least the
>>>protocol) is
>>>>>>>>apparently able to decided on a packet by packet basis
>>>whether to
>>>>>>>>encrypt
>>>>>>>>RTCP - hopefully SRTP implementations also permit this.
>>>>>>>Sure, they can do this. But remember offer-answer. How does a
>>>>>>>receiver
>>>>>>>communicate such a granular policy to the transmitter?
>>>Right now if
>>>>>>>can say whether to auth or auth+encrypt RTP, and auth or
>>>>>>>auth+encrypt
>>>>>>>RTCP. There's no way to say auth+encrypt SDES but only auth SR/XR
>>>>>>>etc.
>>>>>>
>>>>>>Seems to me that this is a problem with SDP security descriptions.
>>>>>>That problem might be there because we forgot about the E
>>>bit.  This
>>>>>>seems very messy for the reason you mention above -
>>>granularity.  I
>>>>>>don't think that we want to negotiate the encryption
>>>policy for each
>>>>>>type SDP message, which is a problem with something like
>>>the E bit.
>>>>>>
>>>>>>Mark
>>>>>>>
>>>>>>>
>>>>>>>>Logically I'd send SDES on its own at the start of a call to
>>>>>>>>provide
>>>>>>>>some
>>>>>>>>user info, but would not send SR/RR/XR until at least
>>>8-10 seconds
>>>>>>>>into the
>>>>>>>>call as I need some time to gather statistics to report.
>>>>>>>That's not exactly what the RTP specs say. RTCP timers don't have
>>>>>>>this
>>>>>>>kind of policy today. In fact, there are porposals on
>>>the table to
>>>>>>>use
>>>>>>>RTCP for early feedback to assess reception quality on
>>>some initial
>>>>>>>packets - e.g. the RTP-noop proposal. In any event that has no
>>>>>>>bearing
>>>>>>>on the present discussion.
>>>>>>>
>>>>>>>>After the
>>>>>>>>start-of-call exchange there would appear to be minimal need to
>>>>>>>>include
>>>>>>>>SDES. I'm sure we could construct some scenarios in
>>>which you "may"
>>>>>>>>need to
>>>>>>>>include SDES with SR/RR/XR however the vast majority of
>>>cases would
>>>>>>>>either
>>>>>>>>need only a single initial exchange or may not need SDES at all.
>>>>>>>Any time you change SSRC you potentially need to send SDES.
>>>>>>>That said, I still don't see what bearing this has on
>>>the discussion
>>>>>>>about whether we need the complexity of encrypting some
>>>RTCP packets
>>>>>>>and not others for a single session.
>>>>>>>
>>>>>>>It's a tradeoff between extra complexity+less privacy versus
>>>>>>>allowing
>>>>>>>snoopers to gather RTCP stats.
>>>>>>>
>>>>>>>Dave.
>>>>>>>
>>>>>>>>Regards
>>>>>>>>
>>>>>>>>Alan
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>-----Original Message-----
>>>>>>>>From: Michael A. Ramalho [mailto:mramalho@cisco.com]
>>>>>>>>Sent: Wednesday, November 24, 2004 9:39 AM
>>>>>>>>To: Alan Clark
>>>>>>>>Cc: Oren Peleg; avt@ietf.org
>>>>>>>>Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>>>>
>>>>>>>>
>>>>>>>>At 07:54 AM 11/24/2004 -0500, Alan Clark wrote:
>>>>>>>>>Hi Oren
>>>>>>>>>
>>>>>>>>>Yes - encryption is intended to provide privacy however this
>>>>>>>>>relates
>>>>>>>>>primarily to the voice payloads and to information within RTCP
>>>>>>>>>that
>>>>>>>>>may be
>>>>>>>>>associated with individual users (e.g. SDES).
>>>>>>>>>
>>>>>>>>>SR, RR and XR contain only peformance statistics and
>>>there is no
>>>>>>>>>privacy
>>>>>>>>>consideration that I can think of related to this type of data.
>>>>>>>>>
>>>>>>>>>The message is - encrypt RTP, optionally encrypt RTCP SDES but
>>>>>>>>>don't
>>>>>>>>encrypt
>>>>>>>>>RTCP SR/RR/XR
>>>>>>>>
>>>>>>>>Hi Alan,
>>>>>>>>
>>>>>>>>I am wondering about implementations utilizing your advise
>>>>>>>>"optionally
>>>>>>>>encrypt
>>>>>>>>RTCP SDES but don't encrypt RTCP SR/RR/XR".
>>>>>>>>
>>>>>>>>Is there an easy way to tell SRTP to encrypt RTCP SDES, but not
>>>>>>>>RTCP
>>>>>>>>SR/RR/XR? It seems that all the RTCP blocks would otherwise be
>>>>>>>>highly
>>>>>>>>related to each other. For example (quote out of RFC 3611):
>>>>>>>>
>>>>>>>>"XR packets supplement the existing RTCP packets, and may be
>>>>>>>>stacked
>>>>>>>>with
>>>>>>>>other RTCP packets to form compound RTCP packets [9,
>>>Section 6]."
>>>>>>>>
>>>>>>>>Thus, RFC3611 explicitly admits the possibility of (but does not
>>>>>>>>mandate)
>>>>>>>>the piggybacking all the RTCP-XR block types.
>>>>>>>>
>>>>>>>>RTCP implementations would have to know the SRTP
>>>encryption rules
>>>>>>>>to
>>>>>>>>make sure they don't perform piggybacking with SDES ...
>>>but could
>>>>>>>>allow
>>>>>>>>(the more bandwidth efficient) piggybacking on the other RTCP-XR
>>>>>>>>blocks.
>>>>>>>>
>>>>>>>>This RTCP/SRTP interaction/dependency seems to add a
>>>burden to both
>>>>>>>>RTCP and SRTP implementations ... let alone the expressiveness
>>>>>>>>required
>>>>>>>>to signal your advise in existing SDP/SRTP mechanisms.
>>>>>>>>
>>>>>>>>Are you aware of any implementations that have actually followed
>>>>>>>>your
>>>>>>>>advise above? I rather go into this one with our eyes open ...
>>>>>>>>developers
>>>>>>>>highly dislike this cross-functional entity dependency
>>>(RTCP & SRTP
>>>>>>>>along
>>>>>>>>with expressing it in SDP/SRTP mechanisms).
>>>>>>>>
>>>>>>>>Regards,
>>>>>>>>
>>>>>>>>Michael Ramalho
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>>Regards
>>>>>>>>>
>>>>>>>>>Alan
>>>>>>>>>
>>>>>>>>>-----Original Message-----
>>>>>>>>>From: Oren Peleg [mailto:OrenP@audiocodes.com]
>>>>>>>>>Sent: Wednesday, November 24, 2004 7:50 AM
>>>>>>>>>To: Alan Clark; avt@ietf.org
>>>>>>>>>Subject: RE: [AVT] RFC 3711 and RTCP
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>Isnt encryption is to prevent monitoring from the first place?
>>>>>>>>>
>>>>>>>>>-----Original Message-----
>>>>>>>>>From: avt-bounces@ietf.org
>>>[mailto:avt-bounces@ietf.org] On Behalf
>>>>>>>>>Of
>>>>>>>>>Alan Clark
>>>>>>>>>Sent: Tuesday, November 23, 2004 5:42 PM
>>>>>>>>>To: avt@ietf.org
>>>>>>>>>Subject: [AVT] RFC 3711 and RTCP
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> From the flurry of discussion on RFC3711 it looks as
>>>if there is
>>>>>>>>>considerable interest in SRTP implementation.  I'd
>>>like to add a
>>>>>>>>>request
>>>>>>>>>to
>>>>>>>>>SRTP implementors with regard to manageability.
>>>>>>>>>
>>>>>>>>>RTCP SR, RR and more particularly XR (e.g. VoIP Metrics report
>>>>>>>>>block)
>>>>>>>>>contain useful information for network monitoring, and
>>>are often
>>>>>>>>>used by
>>>>>>>>>probes and analyzers to support problem diagnosis.
>>>RFC3711 allows
>>>>>>>>>the
>>>>>>>>>selective encryption of RTCP payloads and hence I
>>>would strongly
>>>>>>>>>recomment
>>>>>>>>>that RFC3711 implementors DO NOT encrypt RTCP SR, RR and XR
>>>>>>>>>payloads
>>>>>>>>>as
>>>>>>>>>this
>>>>>>>>>adversely impacts manageability.
>>>>>>>>>
>>>>>>>>>Regards
>>>>>>>>>
>>>>>>>>>Alan Clark
>>>>>>>>>Telchemy
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>_______________________________________________
>>>>>>>>>Audio/Video Transport Working Group
>>>>>>>>>avt@ietf.org
>>>>>>>>>https://www1.ietf.org/mailman/listinfo/avt
>>>>>>>>>
>>>******************************************************************
>>>>>>>>>**
>>>>>>>>>*
>>>>>>>>>******
>>>>>>>>*
>>>>>>>>>********************************
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>>>>>>>>>They are intended solely for the use of the designated
>>>individual
>>>>>>>>>or
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>>>>>>>>>to whom they are addressed.
>>>>>>>>>If the reader of this message is not the intended recipient,
>>>>>>>>>you are hereby notified that any dissemination,use,
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>>>>>>>>>and
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>>>>>>>>>be unlawful.
>>>>>>>>>
>>>>>>>>>If you have received this email in error please notify
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>>>>>>>>>
>>>>>>>>>
>>>******************************************************************
>>>>>>>>>**
>>>>>>>>>*
>>>>>>>>>******
>>>>>>>>*
>>>>>>>>>*******************************
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>_______________________________________________
>>>>>>>>>Audio/Video Transport Working Group
>>>>>>>>>avt@ietf.org
>>>>>>>>>https://www1.ietf.org/mailman/listinfo/avt
>>>>>>>>
>>>>>>>>
>>>>>>>>_______________________________________________
>>>>>>>>Audio/Video Transport Working Group
>>>>>>>>avt@ietf.org
>>>>>>>>https://www1.ietf.org/mailman/listinfo/avt
>>>>>>>David R. Oran
>>>>>>>Cisco Fellow
>>>>>>>Cisco Systems
>>>>>>>7 Ladyslipper Lane
>>>>>>>Acton, MA 01720 USA
>>>>>>>Tel: +1 978 264 2048
>>>>>>>Email: oran@cisco.com
>>>>>>>
>>>>>>>
>>>>>>>_______________________________________________
>>>>>>>Audio/Video Transport Working Group
>>>>>>>avt@ietf.org
>>>>>>>https://www1.ietf.org/mailman/listinfo/avt
>>>>>>
>>>>>>
>>>>>>_______________________________________________
>>>>>>Audio/Video Transport Working Group
>>>>>>avt@ietf.org
>>>>>>https://www1.ietf.org/mailman/listinfo/avt
>>>
>>>
>>>_______________________________________________
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Subject: RE: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-12.txt
Date: Thu, 9 Dec 2004 12:02:36 +0100
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Thread-Topic: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-12.txt
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From: "Elisabetta Carrara \(KI/EAB\)" <elisabetta.carrara@ericsson.com>
To: "Francois Audet" <audet@nortelnetworks.com>, <mmusic@ietf.org>
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Cc: "Jari Arkko \(JO/LMF\)" <jari.arkko@ericsson.com>
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Hi Francois,
thanks for your comments. See in-line.

>-----Original Message-----
>From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]
>On Behalf Of Francois Audet
>Sent: den 7 december 2004 03:07
>To: 'mmusic@ietf.org'
>Cc: 'elisabetta.carrara@ericasson.com'; Mats N=E4slund (KI/EAB);=20
>Jari Arkko (JO/LMF); 'Karl.norrmann@ericsson.com'
>Subject: [MMUSIC] Comments on draft-ietf-mmusic-kmgmt-ext-12.txt
>
>
>Hi,=20
>I have a 2 comments on=20
>http://www.ietf.org/internet-drafts/draft-ietf-mmusic-kmgmt-
>ext-12.txt.=20
>
>
>This first comment is for a clarification:=20
>- Section 3.1.2 explains that SIP 488 is used for rejecting a=20
>key management offer.=20
> 488 allows for the inclusion of an SDP indicating the=20
>capabilities of the answerer (the same way an SDP in a 200 OK=20
>sent in an OPTIONS request indicates capabilities). It would be=20
>desireable to explain how this works with kmgmt. I assume that=20
>this is what is meant by wording saying "Further details about=20
>the cause of failure MAY be described in an included message=20
>from the key management protocol." Perhaps a wording like the=20
>following would help:=20
>Further details about the media capabilities of the offerer=20
>MAY be in included in 488 according to normal RFC 3261 rules.=20
>Further details about the cause of the rejection of the key=20
>message itself MAY be described in the key message in the SDP=20
>in the 488, according to the rules of the key management=20
>protocol. The session is then ...=20
>...or something along those lines.=20
>It would also be worthwile to describe how this relates to=20
>MIKEY in particular (i.e, that the MIKEY message itself in=20
>the key-mgmt may have the key management rejection reason).=20
>It would also be helpful to have an Example of a failure case=20
>for SIP 488/MIKEY.=20

Ok.=20
The draft has undergone the AD's review and has been=20
recently resubmitted to IESG. Hence, since what you suggest=20
is wording for better clarification, we will try to take=20
the chance and add this during IESG review process.=20

>=20
>-----------------------=20
>This next comment is concerning a potential issue with SIP=20
>forking. I'm not sure we can fix it in MMUSIC, but I'll state=20
>it anyways.=20
>- There is a somewhat unpleasant side-effect of using a SIP=20
>failure response for rejecting a key management message.=20
>- Suppose that there is a SIP forking proxy forking the offer=20
>(in the INVITE). Let's say the call is forked to A and B. A=20
>is a device for which no user is present at the time (say, an=20
>office phone). Suppose A rejects the call with 401 Unauthorized=20
>because it needs authorization. Suppose the user is at home (B),=20
>but rejects the offer with 488 (because it's using the key=20
>management protocol used for the corporation which is different=20
>than the one used at home). The forking proxy will see both the=20
>401 an the 488, but can only forward one to the offerer.=20
>There is nothing saying which one should be forwarded beyong=20
>the error response that indicates that there could be=20
>retransmission (which applies to both codes in this case).=20
>Worst, in this particular example, if the proxy follows the=20
>rule section 6/RFC 3261, chances are it will pick 401. This=20
>means that you won't be able to call home because you'll never=20
>receive a 488.=20
>- This is a weakness of the forking procedures in SIP, and=20
>perhaps it should be addresses in the SIP working group.

yes, that seems the appropriate place.

Cheers
/E

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Folks,

attached are the draft minutes of the MMUSIC meeting in DC.
Please review them and send any comments to the list.

Thanks,
Joerg


--------------030001050801000605050507
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Multiparty Multimedia Session Control (MMUSIC) Working Group Minutes
61st IETF meeting, Washington DC, November 2004

Reported by Colin Perkins

   The MMUSIC working group met once at the 61st IETF in Washington
   DC.  The meeting was attended by some 80 participants.  Subjects
   under discussion included application sharing, ICE, Internet Media
   Guides, SDP and various SDP extensions (for media loopback, BFCP,
   labels and connection establishment, security and connectivity
   preconditions).


Introduction and Status Update

   The meeting started with an introduction and status update from the
   chairs. The working group has had one RFC published since the last
   meeting (RFC 3890 on the Session Bandwidth Modifier) and has several
   drafts are under IESG review: the IMG requirements and framework, SDP
   source filter extensions and SDP connection oriented media extensions.
   The draft on Alternative Network Address Types Semantics (ANAT) for the
   SDP Grouping Framework has been updated in response to IESG comments and
   is back under review. The SDP key management and security descriptions
   drafts are being updated in response to review comments from the IESG.

   Several drafts have been discussed on the mailing list, but are not on
   the agenda for the meeting. Remote camera control will be discussed in
   AVT (draft-even-avt-h224-00.txt). RTSP (draft-ietf-mmusic-rfc2326bis-08.txt)
   and the ANNOUNCE extension (draft-ietf-mmusic-rtsp-announce-00.txt) are 
   in progress, but delayed due to lack of review and editing manpower --
   help was solicited. The main connection-oriented media specification
   (draft-ietf-mmusic-sdp-comedia-09.txt) is under IESG review, the TLS
   extension to comedia (draft-ietf-mmusic-comedia-tls-02.txt) is under
   development. Symmetric RTP (draft-wing-mmusic-symmetric-rtprtcp-01.txt)
   is clearly useful, but the chairs suggested it would be better suited 
   to the AVT working group. The draft on tight membership support in SDP
   (draft-ekim-mmusic-sdp-membership-01.txt) conflicts with work in XCON,
   and should be discussed there.

   The Internet Media Guide (IMG) requirements and framework are complete
   and under review by the IESG. We have a milestone to revise our charter 
   to accomodate future work in this area. Work to be considered includes:
   IMG ANNOUNCE via FLUTE, IMG QUERY/RESOLVE via HTTP, IMG SUBSCRIBE/NOTIFY 
   via SIP, and possibly IMG MIME types and URI/URL formats. The chairs,
   area director, and parties interested in IMGs will work to develop new
   milestones are charter text to cover this work, and bring it back to
   the working group for review.


ICE: A Methodology for NAT Traversal

   Jonathan Rosenberg presented ICE (draft-ietf-mmusic-ice-03.txt) outlining
   the changes since the previous version and discussing open issues. Major 
   changes include: 

    - In the ABNF, rename the "alt" attribute to "candidate" to avoid a
      conflict with 3GPP specs, remove the remnents of "derived-from",
      and change the "username" attribute to "user-frag" to more closely
      reflect its semantics.

    - The ICE XML schema has been updated to support both IPv4 and IPv6
      (the implementation of ICE using ANAT supports this already, this
      change brings the schema in line with reality).

    - Added implementation hints on masking allocation latency.

    - Updated discussion of use of TURN as an address allocation protocol,
      making it "SHOULD" strength rather than "MUST" strength, to reflect
      concerns expressed previously. Added text to explain why TURN is a 
      good thing to implement for address allocation. Noted that other
      allocation protocol can be used.

    - Clarified that username and password are different for each transport
      address.

    - Discussed the misrouting scenario raised by Pete Cordell, where STUN
      keep-alives can be sent to a third party that isn't expecting them.
      This is a fundamental issue with the non-uniqueness of IP addresses
      across NAT-realms, not something that can be solved in ICE.

    - Removed mid-session checks. STUN now runs until a media packet is
      received, then you revert to sending only media packets (and using
      media-specific means, rather than STUN, as a keep-alive). Enforces
      a clear separation between the STUN server and the media tool.
      Added a section on application-layer keep-alives.

    - Add discussion on how to choose a default address. 

    - Now use one local transport address for each usable address. This
      gives a one-to-one mapping between the local address/port pair and
      the address that goes in SDP, simplifying the protocol and fixing 
      some bugs, but at the cost of opening more sockets. 

    - Rewrote the description of re-invite handling. Cullen Jennings noted 
      an issue with this, since reverting to an old address can fail if the
      NAT binding for that address has timed out. There may be a need to
      redo the negotiation with a new username and password. 

    - TURN has been promoted to a normative reference, however Cullen
      Jennings is worried that TURN is late and a normative reference may
      delay ICE. It was agreed to revert this to a reference to "some media
      relay protocol" using TURN as an example of a protocol with the
      required properties, to avoid this potential delay.  Considerable
      discussion ensued, covering a number of these issues.

    - Made RTCP information optional in SDP, since many devices don't
      implement RTCP. Magnus Westerlund suggested it is necessary to set
      the RTCP bandwidth parameter to zero, in addition to avoiding the ICE
      check, to inform the other end-point that RTCP isn't used. Jonathan
      agreed that setting the RTCP bandwidth to zero would be a good thing,
      but that implementations didn't do it. Rohan Mahy noted that we can
      say "if you don't do RTCP, you MUST do this with the signalling".
      After much discussion, the consensus seemed to be to make inclusion
      of the RTCP address/port optional, but mandate that implementations
      MUST include a "b=" field of zero if RTCP is not used.

    - Rewrote the security and IANA considerations, and made a number of
      editorial fixes.

   There are a number of open issues relating to mid-session checks, use
   of STUN vs. STUN-over-RTP, forcing symmetry, and local-to-derived
   mapping.

   Mid-session STUN checks serve the purpose of keeping alive NAT bindings,
   detecting interface and NAT failures, and probing for bandwidth. If they
   are included, we need to specify frequency of transmission, a definition 
   of failure and protection against flapping. There are two choices for a
   mid-session keepalive: STUN based and application based. STUN based mid-
   session checks have the advantage of being codec/transport independent, 
   but require applications to demultiplex STUN and media data, make it hard 
   to specify timers and only provide a partial failure detection/recovery 
   solution. Application based mid-session checks provide a clean handoff
   between STUN and the media, separating problems and limiting the scope
   of ICE, and move timers and failure detection to the application.

   Jonathan proposed to keep with the solution in the current, with STUN
   being used for the initial keepalive check, and application specific
   solutions being used after the initial check. He noted that his slide 
   saying "Recommentation: keep stun keepalives" is wrong. Joerg Ott and
   Dave Oran agreed with Jonathan.

   The next open issue was the use of STUN vs. STUN-over-RTP for initial 
   connectivity checks (i.e. the 1st RTP packet's payload data is a STUN
   packet). The disadvantages of using STUN over RTP are that it's ugly,
   the request/response nature of STUN isn't well matched with RTP
   semantics, the RTP timestamp would jump since the STUN packets aren't
   really media, it doesn't work with SRTP, and it makes ICE specific to
   RTP. The only advantage is that is avoids a STUN/RTP demultiplex --
   although that comes at the cost of embedding STUN into RTP. Dave Oran
   agreed with all these, provided this is done for RTCP. For RTCP, the
   issues are not as bad since RTCP APP packets can be used. Magnus
   Westerlund noted that there are other protocols and making ICE RTP
   specific is a mistake. The consensus of the room was that STUN-over-RTP
   is a mistake, especially since we now have a clean handoff between STUN
   and the media.

   Regarding forced symmetry, the problem is that the current algorithm can
   result in you sending on a different port than that on which you receive
   media. This doesn't work if one is depending on application-level keep
   alives. The suggested fix is MUST send on the port on which you last
   received a valid RTP packet. Magnus Westerlund noted that there's a
   potential for flapping, but Jonathan countered that the protocol will
   settle to a stable state, and there are further rules to ensure one
   SHOULD use the highest priority address on which you got a STUN request. 

   Regarding local to derived mapping, previous versions of the draft
   allowed a single socket to be used for it's local address, and as a
   source of STUN and TURN requests used to determine derived addresses.
   This caused confusion with username-to-address mapping for the STUN and
   TURN requests, made it hard to determine when to stop the STUN server,
   and made it difficult to assess which address was used. The latest
   version of the draft uses distinct ports for the local address, and to
   run STUN and TURN to derive addresses. This avoids all the multiplexing
   issues, but at the expense of additional local port usage. Jonathan is
   worried about the port footprint, and is considering going back to the
   multiplexing solution (one local address for each derived address), and 
   defining appropriate timeouts for each protocol. Rohan Mahy noted that 
   the folks that have port issues will likely be those who have problem
   multiplexing, so this may not solve the problem. Dan Wing suggested that
   the problem is that we're lacking a message to say "STUN packet received"
   so why not create a stun message to say exactly that? Jonathan considered
   that, but thought that while it helps you turn off the STUN listener, it
   doesn't solve the demultiplexing issues. Cullen Jennings asked why we
   can't do both? The choice of one-to-one or many-to-one mappings between
   local and derived addresses is a local implementation choice, and which
   doesn't affect remote clients. Why not document the issue, and let the
   implementor choose? Jonathan agreed that this is a good way forward.

   Magnus Westerlund noted that ICE could generate a lot of packets, and
   may cause congestion. It may be better to have some initial delay and to
   send messages in priority order. Jonathan agreed that this could reduce
   the amount of data sent, but increased the post-dial delay (which is a
   hard requirment for some applications). Joerg Ott noted that it's
   unlikely there are many addresses, so this may not be such a problem:
   maybe just add a warning note? Jonathan agreed that a discussion of the
   issue would be appropriate.

   Magnus also worried about switching media between different addresses;
   we should highlight the potential congestion control issue when changing 
   connection addresses.

   Jonathan committed to revising draft immediately after the meeting, to
   aim for working group last call before Christmas.

Internet Media Guides

   Yuji Nomura discussed Internet Media Guides. The IMG requirements and
   framework are with the IESG for review, so it is now time to identify
   the specific protocol components that need to be developed. The main
   items are an IMG transfer envelope, a multicast delivery method, and a
   subscribe/notify delivery method.

   The groundwork for the unicast subscribe/notify mechanism is defined
   in draft-nomura-mmusic-img-notify-00.txt. Yuji noted that using HTTP 
   doesn't meet the requirements, so the working assumption is that SIP
   events will be used for subscribe/notify, and there is a need to define
   an event package for IMG. Multicast delivery will be via FLUTE: an
   intial draft defining this has previously been submitted, and will
   be updated shortly.

   In addition to the transport protocols, and envelope is needed for the
   data. An initial proposal (draft-walsh-mmusic-img-envelope-01.txt)
   exists for this and provide a common minimal set of information to
   manage transfer of IMG data, independent of the transport protocol.
   The working assumption here is to use an XML format.

   Open issues include the need to define an IMG URI (e.g. img://...)
   and naming IMG data for encapsulation ("application/envelope+xml"?).
   It will also be necessary to define versioning and delta descriptions
   for the IMG data, to support partial transmission.

   Henning Schulzrinne asked if the IMG URI is intended to identify the
   content or the transport, and wondered how it should be used? There
   was some confusion on this point, and it isn't clear how to translate
   from the URI to the retrieval mechanism. Clarifiation is needed.


Application Sharing

   Henning Schulzrinne discussed draft-schulzrinne-mmusic-sharing-00.txt on
   requirements for application sharing. This is speculative new work, and
   it's not clear if it belongs in MMUSIC. Our conferencing architecture is 
   deficient in support for shared applications, and Henning has considered
   how this can be resolved without reinventing T.120. The presentation and 
   draft outline the problem space and the requirements. 

   Rohan Mahy expressed concern that the scope being addressed here is
   significantly larger than he had imagined. We may want to solve the
   problem for named categories of application, splitting the problem space
   rather than "trying to boil the ocean". Dave Oran, Jonathan Rosenberg
   and Roni Even and also expressed similar concern regarding the scope of
   the work, compatibility with other systems, the form of the data to be
   shared, etc. Henning noted that he's raising possible requirements, not
   suggesting that we'll solve them all immediately. Steve Casner noted
   that we already have limited RTP payload formats for pointers and text
   events, which may feed into this work. 

   Joerg Ott concluded by noted that this is an intersting problem, and
   noted that while the negotiation of application sharing sessions is
   clearly in scope for MMUSIC, the application sharing protocol itself 
   isn't. There doesn't seem to be a natural home for this work in the 
   IETF, although we can think about it here for now.


SDP

   Colin Perkins outlined the changes to draft-ietf-mmusic-sdp-new-21.txt
   since the last meeting (see slides for details). There are open issues
   regarding support for internationalized DNS, layered coding, direction
   attributes and the placement of the spec in context.
   
   Internationalized DNS names did not exist when the original SDP was
   written, so the draft doesn't indicate how they should be represented.
   Advice was solicited from the working group and Bernie Hoeneisen gave
   the suggestion that internationalized DNS names should be conveyed in 
   the ASCII Compatible Encoding (ACE) form defined in RFC 3490, rather
   than in UTF-8 or some other encoding. This requires no changes to SDP,
   other than to note that this form should be used.

   Regarding layered coding, the issues are that allocation of consecutive
   multicast addresses is problematic, and that the "a=rtcp:" attribute is 
   not fully specified for layered encoding. Colin suggested that, since
   this part of the spec is not widely implemented, it should be left as-is
   for now, and updated if layered multicast transport becomes popular. No
   comments were received.

   There was a suggestion that interactions between the direction attributes 
   ("a=recvonly", etc.) and the offer/answer spec should be documented, and
   offer/answer referenced. More generally, is there need to include more
   context on how SDP is used with, for example, SIP? Jonathan Rosenberg
   and Keith Drage suggested that these interactions are defined in other 
   drafts, so there's no need to add anything to the SDP spec. 


SDP Extension for Media Loopback

   Kaynam Hedayat discussed draft-hedayat-media-loopback-01.txt. The goal
   of this draft is to enable SIP endpoint to initiate calls with media
   loopback. It introduces two SDP media attributes, giving the type and
   mode of the loopback and is based on the offer/answer model.

   Colin Perkins and Dave Oran were uncomfortable with the requirement that
   endpoints MUST support RTCP-XR, and suggested that support for RTCP-XR
   be OPTIONAL. The loopback attribute is more general than voice over IP
   applications, so there is no way a particular subset of RTCP-XR support 
   can be mandated. 

   Magnus Westerlund expressed concern that the security aspects of this
   new attribute haven't been fully considered, and that it may introduce
   potential packet bombing attacks. This issue needs to be documented.

   The consensus of the room was that this a good idea, and should be
   adopted as an MMUSIC work item.


SDP Extension for BFCP

   Gonzalo Camarillo presented (draft-camarillo-mmusic-sdp-bfcp-00.txt)
   on the SDP extensions needed to support the XCON binary floor control
   protocol. BFCP is defined under the "application/bfcp" media type and
   runs over TCP. It fits into SDP as a transport protocol with connection
   establishment is handled by the comedia framework. The "k=" line can be
   used to convey the shared secret needed by BFCP, and attributes are
   defined to convey the confid and userid. Mapping between floor IDs and
   streams can be done using the SDP label attribute. There is an open
   issue regarding the need for a nonce attribute, which will be discussed
   in XCON.

   The consensus of the room was to accept this as an MMUSIC work item.


SDP Label Attribute

   Gonzalo Camarillo presented (draft-ietf-mmusic-sdp-media-label-00.txt)
   on the SDP label attribute. The main issue at the last meeting was why
   label media streams, rather than referring to the n'th media stream. It
   has been clarified that this is so static documents can be created that
   refer to specific media streams within future SDP files, without having
   to revise those static documents if the type or order of the media in
   those future sessions changes. 

   The authors asked for the draft to go to working group last call. Colin
   Perkins raised an issue that the label is defined as "any octet string"
   but needs to be restricted to octet strings that can be represented in
   SDP (i.e. printable ASCII). The working group last call will be issued
   after the meeting, with this comment being fixed as an editorial
   clarification after the last call.


SDP Connection Establishment Precondition

   Gonzalo Camarillo presented (draft-ietf-mmusic-connection-precon-00.txt)
   on SDP connection establishment preconditions, briefly noting that the
   SIP working group is performing an expert review of the draft. 


SDP Security Precondition

   Flemming Andreasen gave an overview of the SDP security precondition
   draft (draft-andreasen-mmusic-securityprecondition-02.txt). The problem
   that it solves is when an offer includes several security descriptions
   from which the answerer picks one and sends secure media. The media can
   start to arrive before the answer, but cannot be be decoded since the
   offerer doesn't know which security parameters were chosen. Hence a new
   precondition is defined to delay completion of the call until all
   parties have agred on the security parameters. 

   Changes in this version are to clarify the definition of "send" and
   "recv" security preconditions, add more detail in the example flow, and
   add security considerations. This is needed by SIP, and there are no
   known open issues.

   Colin Perkins asked that an example be included using the kgmnt-ext form
   of SDP signalling, in addition to the sdescriptions examples, but
   otherwise agreed that the draft was in good shape. The authors were
   requested to make this change, and submit the draft as a working group
   draft, then a working group last call could start.


SDP Connectivity Precondition

   Flemming Andreasen also presented the SDP connectivity preconditions
   (draft-andreasen-mmusic-connectivityprecondition-01.txt). This draft
   defines a precondition to ensure there is media stream connectivity
   between offerer and answerer prior to proceeding with session
   establishement. It doesn't mandate a particular mechanism to verify
   connectivity.

   Changes in this version are to clarify the definition of "send" and
   "recv" preconditions, add more detail in the example flow, and add 
   security considerations. Open issues include adding an explanation
   of the relationship between this and ICE, and explaining how it will
   interact with SIP forking.

   Colin Perkins asked that this draft be clearer on the difference between
   it and the connection precondition, in particular to be careful about
   the precondition names (i.e. "con" vs. "conn"). Paul Kyzivat noted that
   we have many preconditions, and worried about use cases and feature
   interaction. It may be appropriate to consider documenting possible
   interactions? Henning Schulzrinne raised a general concern that the
   precondition framework may be covered by IPR?

   It was agreed that this draft should become an MMUSIC work item.


				  -- + --


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From mmusic-bounces@ietf.org  Fri Dec 10 06:58:38 2004
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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Fri, 10 Dec 2004 11:47:35 +0000
To: Mark Baugher <mbaugher@cisco.com>
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Mark,

On 8 Dec 2004, at 00:43, Mark Baugher wrote:
>   Do we know of any other applications besides RFC 3611 that might 
> need to selectively encrypt SRTCP packets?

I don't, but I can imagine future RTCP extensions which might require 
this feature (and applications that might want to encrypt various 
subsets of the RFC 3611 data).

>   I am okay with putting this requirement aside pending a separate 
> document.  But before someone goes to the trouble of writing the new 
> draft, it might be good to consider if it's needed or not.

Even if RTCP XR is the only driver, it might be appropriate to make 
that a separate short draft. There's been enough discussion on the list 
that I'm not entirely comfortable slotting it into sdescriptions this 
late in the process.

Colin


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From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Fri, 10 Dec 2004 08:02:32 -0800
To: Colin Perkins <csp@csperkins.org>
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Colin,

On Dec 10, 2004, at 3:47 AM, Colin Perkins wrote:

> Mark,
>
> On 8 Dec 2004, at 00:43, Mark Baugher wrote:
>>   Do we know of any other applications besides RFC 3611 that might 
>> need to selectively encrypt SRTCP packets?
>
> I don't, but I can imagine future RTCP extensions which might require 
> this feature (and applications that might want to encrypt various 
> subsets of the RFC 3611 data).

I don't see a large downside in waiting until we see such extensions 
for real.
>
>>   I am okay with putting this requirement aside pending a separate 
>> document.  But before someone goes to the trouble of writing the new 
>> draft, it might be good to consider if it's needed or not.
>
> Even if RTCP XR is the only driver, it might be appropriate to make 
> that a separate short draft. There's been enough discussion on the 
> list that I'm not entirely comfortable slotting it into sdescriptions 
> this late in the process.

Ok.  If there are no objections, I'll generate a new version and post 
it early next week.

Mark
>
> Colin
>

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From mmusic-bounces@ietf.org  Fri Dec 10 16:33:08 2004
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From: "Alan Clark" <alan.d.clark@telchemy.com>
To: "Mark Baugher" <mbaugher@cisco.com>, "Colin Perkins" <csp@csperkins.org>
Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Fri, 10 Dec 2004 16:05:12 -0500
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Mark, Colin

Where does this leave the issue?  Some sizeable network and telecom
equipment vendors are in the process of integrating support for the RFC3611
VoIP metrics, due in a large part to service provider and enterprise
customer demand.  The demand exists because customers need to manage their
VoIP deployments, hence they see RFC3611 as a key element in their
management infrastructure.  The accessibility of RFC3611 VoIP metrics is
therefore an issue of concern today.

If it's proposed to deal with this in a separate draft then what is the
expected timeframe - going through the regular I-D, WG draft, IESG process
seem to take a year or so. Is there some mechanism by which this can be
accelerated?

If the proposal is to address the issue in the draft currently being
finalized then my apologies if this has caused any last minute delays
however the issue is important enough to warrant this (at least in my view).
We hoped that the comment in the security considerations section of RFC3611
was strong enough to address the issue however it appears not to have been.

Regards

Alan


-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]On Behalf
Of Mark Baugher
Sent: Friday, December 10, 2004 11:03 AM
To: Colin Perkins
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP


Colin,

On Dec 10, 2004, at 3:47 AM, Colin Perkins wrote:

> Mark,
>
> On 8 Dec 2004, at 00:43, Mark Baugher wrote:
>>   Do we know of any other applications besides RFC 3611 that might
>> need to selectively encrypt SRTCP packets?
>
> I don't, but I can imagine future RTCP extensions which might require
> this feature (and applications that might want to encrypt various
> subsets of the RFC 3611 data).

I don't see a large downside in waiting until we see such extensions
for real.
>
>>   I am okay with putting this requirement aside pending a separate
>> document.  But before someone goes to the trouble of writing the new
>> draft, it might be good to consider if it's needed or not.
>
> Even if RTCP XR is the only driver, it might be appropriate to make
> that a separate short draft. There's been enough discussion on the
> list that I'm not entirely comfortable slotting it into sdescriptions
> this late in the process.

Ok.  If there are no objections, I'll generate a new version and post
it early next week.

Mark
>
> Colin
>

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From: Ben Campbell <ben@nostrum.com>
Date: Fri, 10 Dec 2004 15:41:12 -0600
To: Ben Campbell <ben@nostrum.com>
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Cc: Simple WG <simple@ietf.org>, mmusic@ietf.org
Subject: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
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I really need to get closure on this, so I'd appreciate people agreeing  
or disagreeing.

I have had off-list correspondence indicating both positions. In  
summary:

In favor of the change, I had a comment to the effect that changing SDP  
semantics to ignore the C line and the port field on the M line really  
changes it to something that is no longer SDP, and that such deviations  
are in general bad for interop.

In favor of leaving things as currently specified, I had two comments  
that decomposing URLs into the C and M lines is ugly, and that having a  
different mechanism to communicate endpoint addresses vs relay  
addresses is really ugly.

So, do people agree with one or the other position? Something  
completely different? (Hopefully the people who sent off-list  
correspondence will reply on-list if they think I mischaracterized  
either position. Actually, I hope they will repost on-list even if they  
like my summaries)

I have so far avoided interjecting my personal opinion so that I could  
do a fair summary. But, I actually tend to agree with the  
leave-things-as-specified position.

While I am sensitive to the issue of deviating from the spirit of SDP,  
I don't really buy the interoperability argument. The currently defined  
usage _does_ put "msrp" in the M line--so only consumers that  
understand what that means will be able to act on it. I assume that any  
endpoint that understands what "msrp" means will also understand how to  
use the "path" attribute.

The place that interoperability might suffer is if any non-endpoint  
needs to interpret things--which is generally not a goal. I guess can  
also see debug or monitoring tools who don't know about MSRP getting  
things wrong.

My biggest technical objection is that using a different mechanism for  
determining where to connect depending on whether you are connecting  
peer to peer or through a relay, increases the chances of implementors  
getting things wrong. And I cannot personally think of a way to handle  
relay addresses that is more in the spirit of SDP than what we  
currently have.

But in any case, I will be happy to document consensus, even if it  
disagrees with my position. But we need to close on it asap.

Thanks!

Ben.

On Dec 7, 2004, at 5:20 PM, Ben Campbell wrote:

> I sent the attached to the SIMPLE list last week. I am reposting it to  
> both the SIMPLE and MMUSIC lists, as the discussion is of interest to  
> both groups. If you already responded to this on the SIMPLE list or in  
> private, it would be helpful if you can re-post your response to this  
> thread, so everyone sees the whole discussion.
>
> Thanks!
>
> Ben.
>
> ----------------------------------------------------------------------- 
> -----------------------------------------
>
>
> In the DC meeting, we discussed that MSRP's SDP usage should be  
> reviewed in MMUSIC, or at least by people who are more expert in SDP.   
> Colin Perkins and Joerg Ott looked at this, and suggested a change.  
> (Many thanks to them for taking the time, and I hope they will jump in  
> and correct me if I mischaracterize anything.)
>
> The upshot of the proposal is that the fact that MSRP ignores the  
> address in the sdp c-line, and the port and proto fields in the  
> m-line. Instead, MSRP gets all of this from the first URL in the path  
> a-line attribute. This deviates from the traditional usage in SDP, and  
> causes confusion.
>
> The proposed change is that the destination URL be broken into the  
> appropriate parts and encoded into the c-line and m-line. For example  
> (leaving out lines that don't change)
>
>  c=IN IP4 alice.example.com
>  m=message 9 msrp *
>  a=path:msrp://a.example.com:7394/2s93i;tcp
>
> would become:
>
>  c=IN IP4 a.example.com
>  m=message 7394 msrp/tcp *
>  a=session:2s93i
>
> The change would only be made for the destination URL. URLs for any  
> relays would be expressed the same way they are now.
>
> The advantage of this approach is that it more closely follows the  
> spirit of SDP. This will be easier to understand for people who  
> already know SDP. Existing SDP code could be reused more effectively.
>
> The disadvantage of the proposed change is, it means the way an MSRP  
> client determines the destination URL would be different depending on  
> whether the URL represented the final destination or a relay. Instead  
> of the current approach of simply copying the path attribute value  
> into the To-Path header in MSRP messages,  a client would copy the  
> path header value if exists, reconstitute the destination URL from the  
> m and c lines, and append the destination URL to the path header  
> value.
>
> So, what do people think? (I note that we actually did something  
> similar to this before we added the "path" attribute. )
>
>
> Thanks!
>
> Ben.
>
>
>
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>


_______________________________________________
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Cc: Simple WG <simple@ietf.org>, mmusic@ietf.org
Subject: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
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Ben,

I agree with what you say below, and have no more to add now.

	Paul

Ben Campbell wrote:
> I really need to get closure on this, so I'd appreciate people agreeing  
> or disagreeing.
> 
> I have had off-list correspondence indicating both positions. In  summary:
> 
> In favor of the change, I had a comment to the effect that changing SDP  
> semantics to ignore the C line and the port field on the M line really  
> changes it to something that is no longer SDP, and that such deviations  
> are in general bad for interop.
> 
> In favor of leaving things as currently specified, I had two comments  
> that decomposing URLs into the C and M lines is ugly, and that having a  
> different mechanism to communicate endpoint addresses vs relay  
> addresses is really ugly.
> 
> So, do people agree with one or the other position? Something  
> completely different? (Hopefully the people who sent off-list  
> correspondence will reply on-list if they think I mischaracterized  
> either position. Actually, I hope they will repost on-list even if they  
> like my summaries)
> 
> I have so far avoided interjecting my personal opinion so that I could  
> do a fair summary. But, I actually tend to agree with the  
> leave-things-as-specified position.
> 
> While I am sensitive to the issue of deviating from the spirit of SDP,  
> I don't really buy the interoperability argument. The currently defined  
> usage _does_ put "msrp" in the M line--so only consumers that  
> understand what that means will be able to act on it. I assume that any  
> endpoint that understands what "msrp" means will also understand how to  
> use the "path" attribute.
> 
> The place that interoperability might suffer is if any non-endpoint  
> needs to interpret things--which is generally not a goal. I guess can  
> also see debug or monitoring tools who don't know about MSRP getting  
> things wrong.
> 
> My biggest technical objection is that using a different mechanism for  
> determining where to connect depending on whether you are connecting  
> peer to peer or through a relay, increases the chances of implementors  
> getting things wrong. And I cannot personally think of a way to handle  
> relay addresses that is more in the spirit of SDP than what we  
> currently have.
> 
> But in any case, I will be happy to document consensus, even if it  
> disagrees with my position. But we need to close on it asap.
> 
> Thanks!
> 
> Ben.
> 
> On Dec 7, 2004, at 5:20 PM, Ben Campbell wrote:
> 
>> I sent the attached to the SIMPLE list last week. I am reposting it 
>> to  both the SIMPLE and MMUSIC lists, as the discussion is of interest 
>> to  both groups. If you already responded to this on the SIMPLE list 
>> or in  private, it would be helpful if you can re-post your response 
>> to this  thread, so everyone sees the whole discussion.
>>
>> Thanks!
>>
>> Ben.
>>
>> ----------------------------------------------------------------------- 
>> -----------------------------------------
>>
>>
>> In the DC meeting, we discussed that MSRP's SDP usage should be  
>> reviewed in MMUSIC, or at least by people who are more expert in 
>> SDP.   Colin Perkins and Joerg Ott looked at this, and suggested a 
>> change.  (Many thanks to them for taking the time, and I hope they 
>> will jump in  and correct me if I mischaracterize anything.)
>>
>> The upshot of the proposal is that the fact that MSRP ignores the  
>> address in the sdp c-line, and the port and proto fields in the  
>> m-line. Instead, MSRP gets all of this from the first URL in the path  
>> a-line attribute. This deviates from the traditional usage in SDP, 
>> and  causes confusion.
>>
>> The proposed change is that the destination URL be broken into the  
>> appropriate parts and encoded into the c-line and m-line. For example  
>> (leaving out lines that don't change)
>>
>>  c=IN IP4 alice.example.com
>>  m=message 9 msrp *
>>  a=path:msrp://a.example.com:7394/2s93i;tcp
>>
>> would become:
>>
>>  c=IN IP4 a.example.com
>>  m=message 7394 msrp/tcp *
>>  a=session:2s93i
>>
>> The change would only be made for the destination URL. URLs for any  
>> relays would be expressed the same way they are now.
>>
>> The advantage of this approach is that it more closely follows the  
>> spirit of SDP. This will be easier to understand for people who  
>> already know SDP. Existing SDP code could be reused more effectively.
>>
>> The disadvantage of the proposed change is, it means the way an MSRP  
>> client determines the destination URL would be different depending on  
>> whether the URL represented the final destination or a relay. Instead  
>> of the current approach of simply copying the path attribute value  
>> into the To-Path header in MSRP messages,  a client would copy the  
>> path header value if exists, reconstitute the destination URL from 
>> the  m and c lines, and append the destination URL to the path header  
>> value.
>>
>> So, what do people think? (I note that we actually did something  
>> similar to this before we added the "path" attribute. )
>>
>>
>> Thanks!
>>
>> Ben.
>>
>>
>>
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple
>>
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 

_______________________________________________
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https://www1.ietf.org/mailman/listinfo/mmusic


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From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Fri, 10 Dec 2004 15:38:58 -0800
To: "Alan Clark" <alan.d.clark@telchemy.com>
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Cc: mmusic@ietf.org, Colin Perkins <csp@csperkins.org>
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Alan,
   I'm finishing sdescriptions-08 without any change to accommodate XR 
packets to use the E bit.  I don't yet know who is going to do the work 
on the I-D that Colin suggested to address Magnus' issues with E-bit 
usage.  As it stands now, UNENCRYPTED_SRTCP will need to be signaled in 
order to keep XR packets in the clear.

Mark
On Dec 10, 2004, at 1:05 PM, Alan Clark wrote:

> Mark, Colin
>
> Where does this leave the issue?  Some sizeable network and telecom
> equipment vendors are in the process of integrating support for the 
> RFC3611
> VoIP metrics, due in a large part to service provider and enterprise
> customer demand.  The demand exists because customers need to manage 
> their
> VoIP deployments, hence they see RFC3611 as a key element in their
> management infrastructure.  The accessibility of RFC3611 VoIP metrics 
> is
> therefore an issue of concern today.
>
> If it's proposed to deal with this in a separate draft then what is the
> expected timeframe - going through the regular I-D, WG draft, IESG 
> process
> seem to take a year or so. Is there some mechanism by which this can be
> accelerated?
>
> If the proposal is to address the issue in the draft currently being
> finalized then my apologies if this has caused any last minute delays
> however the issue is important enough to warrant this (at least in my 
> view).
> We hoped that the comment in the security considerations section of 
> RFC3611
> was strong enough to address the issue however it appears not to have 
> been.
>
> Regards
>
> Alan
>
>
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]On Behalf
> Of Mark Baugher
> Sent: Friday, December 10, 2004 11:03 AM
> To: Colin Perkins
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>
>
> Colin,
>
> On Dec 10, 2004, at 3:47 AM, Colin Perkins wrote:
>
>> Mark,
>>
>> On 8 Dec 2004, at 00:43, Mark Baugher wrote:
>>>   Do we know of any other applications besides RFC 3611 that might
>>> need to selectively encrypt SRTCP packets?
>>
>> I don't, but I can imagine future RTCP extensions which might require
>> this feature (and applications that might want to encrypt various
>> subsets of the RFC 3611 data).
>
> I don't see a large downside in waiting until we see such extensions
> for real.
>>
>>>   I am okay with putting this requirement aside pending a separate
>>> document.  But before someone goes to the trouble of writing the new
>>> draft, it might be good to consider if it's needed or not.
>>
>> Even if RTCP XR is the only driver, it might be appropriate to make
>> that a separate short draft. There's been enough discussion on the
>> list that I'm not entirely comfortable slotting it into sdescriptions
>> this late in the process.
>
> Ok.  If there are no objections, I'll generate a new version and post
> it early next week.
>
> Mark
>>
>> Colin
>>
>
> _______________________________________________
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>
>
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From: Marc Manthey <marc@let.de>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Sat, 11 Dec 2004 04:09:03 +0100
To: IETF MMUSIC WG <mmusic@ietf.org>
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<snip>...This document defines the Extended Report (XR) packet type for 
the
    RTP Control Protocol (RTCP), and defines how the use of XR packets
    can be signaled by an application if it employs the Session
    Description Protocol (SDP).
</snip>
a protocol that protocols a protocol....

*SCNR*


leave it like it is;)

cheers

marc
-- 
    "Fragmentation is like classful addressing , an interesting early
    architectural error that shows how much experimentation was going
    on while IP was being designed."

            http://www.multicasttech.com/status/msdp.html


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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Sat, 11 Dec 2004 12:23:21 +0000
To: "Alan Clark" <alan.d.clark@telchemy.com>
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Alan,

On 10 Dec 2004, at 21:05, Alan Clark wrote:
> Where does this leave the issue?  Some sizeable network and telecom
> equipment vendors are in the process of integrating support for the 
> RFC3611
> VoIP metrics, due in a large part to service provider and enterprise
> customer demand.  The demand exists because customers need to manage 
> their
> VoIP deployments, hence they see RFC3611 as a key element in their
> management infrastructure.  The accessibility of RFC3611 VoIP metrics 
> is
> therefore an issue of concern today.

Understood. I note that the sdescriptions work is also eagerly awaited 
by the industry.

> If it's proposed to deal with this in a separate draft then what is the
> expected timeframe - going through the regular I-D, WG draft, IESG 
> process
> seem to take a year or so. Is there some mechanism by which this can be
> accelerated?

We have little influence on the IESG process, but the working group 
doesn't have to take a long time. If this issue is straightforward, we 
should be able to document a solution that can be progressed to the 
IESG very quickly. Would you be willing to write such a draft?

Colin


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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
Date: Sat, 11 Dec 2004 12:30:12 +0000
To: Ben Campbell <ben@nostrum.com>
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On 10 Dec 2004, at 21:41, Ben Campbell wrote:
> I really need to get closure on this, so I'd appreciate people 
> agreeing or disagreeing.
>
> I have had off-list correspondence indicating both positions. In 
> summary:
>
> In favor of the change, I had a comment to the effect that changing 
> SDP semantics to ignore the C line and the port field on the M line 
> really changes it to something that is no longer SDP, and that such 
> deviations are in general bad for interop.
>
> In favor of leaving things as currently specified, I had two comments 
> that decomposing URLs into the C and M lines is ugly, and that having 
> a different mechanism to communicate endpoint addresses vs relay 
> addresses is really ugly.
>
> So, do people agree with one or the other position?

Decomposing the URL into the "c=" and "m=" lines is very ugly. 
Unfortunately, that ugliness is required if you want to use SDP. This 
is why the SDPng work was started, to try to move to something less 
horrible...

Colin


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From: "Alan Clark" <alan.d.clark@telchemy.com>
To: "Colin Perkins" <csp@csperkins.org>
Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Sat, 11 Dec 2004 07:58:34 -0500
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Colin

I would of course be willing to write a suitable draft if this will expedite
things

Regards

Alan


-----Original Message-----
From: Colin Perkins [mailto:csp@csperkins.org]
Sent: Saturday, December 11, 2004 7:23 AM
To: Alan Clark
Cc: mmusic@ietf.org; Mark Baugher
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP


Alan,

On 10 Dec 2004, at 21:05, Alan Clark wrote:
> Where does this leave the issue?  Some sizeable network and telecom
> equipment vendors are in the process of integrating support for the
> RFC3611
> VoIP metrics, due in a large part to service provider and enterprise
> customer demand.  The demand exists because customers need to manage
> their
> VoIP deployments, hence they see RFC3611 as a key element in their
> management infrastructure.  The accessibility of RFC3611 VoIP metrics
> is
> therefore an issue of concern today.

Understood. I note that the sdescriptions work is also eagerly awaited
by the industry.

> If it's proposed to deal with this in a separate draft then what is the
> expected timeframe - going through the regular I-D, WG draft, IESG
> process
> seem to take a year or so. Is there some mechanism by which this can be
> accelerated?

We have little influence on the IESG process, but the working group
doesn't have to take a long time. If this issue is straightforward, we
should be able to document a solution that can be progressed to the
IESG very quickly. Would you be willing to write such a draft?

Colin


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From mmusic-bounces@ietf.org  Sat Dec 11 09:52:37 2004
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From: "Alan Clark" <alan.d.clark@telchemy.com>
To: "Mark Baugher" <mbaugher@cisco.com>, "Colin Perkins" <csp@csperkins.org>,
        "Magnus Westerlund" <magnus.westerlund@ericsson.com>
Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
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It would seem that an additional draft needs to state roughly the following:

(i) Scope - provides method for selective encryption/ unencryption of
certain RTCP payload types

(ii) Approach - allow for SRTCP encryption to be set ON by default but
overridden for certain payload types

(iii) New parameter

SELECTIVE_SRTCP_UNENCRYPTED means that by default RTCP frames are encrypted
with the exception of the listed payload types

List of RTCP payload types .. SDES, BYE, RR, SR, XR, XR_VOIPMETRICS, XR_...

Some compound types (for convenience) .. ALL_INFORMATIONAL_TYPES
where ALL_INFORMATIONAL_TYPES = SR, RR as defined in RFC3550 and XR payloads
as defined in RFC3611

(iv) Examples

 ....., SELECTIVE_SRTCP_UNENCRYPTED:ALL_INFORMATIONAL_TYPES

or

 ....., SELECTIVE_SRTCP_UNENCRYPTED:RR,SR,XR_VOIPMETRICS

Any comments on the general approach?

Regards

Alan


-----Original Message-----
From: Mark Baugher [mailto:mbaugher@cisco.com]
Sent: Friday, December 10, 2004 6:39 PM
To: Alan Clark
Cc: mmusic@ietf.org; Colin Perkins
Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP


Alan,
   I'm finishing sdescriptions-08 without any change to accommodate XR
packets to use the E bit.  I don't yet know who is going to do the work
on the I-D that Colin suggested to address Magnus' issues with E-bit
usage.  As it stands now, UNENCRYPTED_SRTCP will need to be signaled in
order to keep XR packets in the clear.

Mark
On Dec 10, 2004, at 1:05 PM, Alan Clark wrote:

> Mark, Colin
>
> Where does this leave the issue?  Some sizeable network and telecom
> equipment vendors are in the process of integrating support for the
> RFC3611
> VoIP metrics, due in a large part to service provider and enterprise
> customer demand.  The demand exists because customers need to manage
> their
> VoIP deployments, hence they see RFC3611 as a key element in their
> management infrastructure.  The accessibility of RFC3611 VoIP metrics
> is
> therefore an issue of concern today.
>
> If it's proposed to deal with this in a separate draft then what is the
> expected timeframe - going through the regular I-D, WG draft, IESG
> process
> seem to take a year or so. Is there some mechanism by which this can be
> accelerated?
>
> If the proposal is to address the issue in the draft currently being
> finalized then my apologies if this has caused any last minute delays
> however the issue is important enough to warrant this (at least in my
> view).
> We hoped that the comment in the security considerations section of
> RFC3611
> was strong enough to address the issue however it appears not to have
> been.
>
> Regards
>
> Alan
>
>
> -----Original Message-----
> From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org]On Behalf
> Of Mark Baugher
> Sent: Friday, December 10, 2004 11:03 AM
> To: Colin Perkins
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>
>
> Colin,
>
> On Dec 10, 2004, at 3:47 AM, Colin Perkins wrote:
>
>> Mark,
>>
>> On 8 Dec 2004, at 00:43, Mark Baugher wrote:
>>>   Do we know of any other applications besides RFC 3611 that might
>>> need to selectively encrypt SRTCP packets?
>>
>> I don't, but I can imagine future RTCP extensions which might require
>> this feature (and applications that might want to encrypt various
>> subsets of the RFC 3611 data).
>
> I don't see a large downside in waiting until we see such extensions
> for real.
>>
>>>   I am okay with putting this requirement aside pending a separate
>>> document.  But before someone goes to the trouble of writing the new
>>> draft, it might be good to consider if it's needed or not.
>>
>> Even if RTCP XR is the only driver, it might be appropriate to make
>> that a separate short draft. There's been enough discussion on the
>> list that I'm not entirely comfortable slotting it into sdescriptions
>> this late in the process.
>
> Ok.  If there are no objections, I'll generate a new version and post
> it early next week.
>
> Mark
>>
>> Colin
>>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>


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Date: Sat, 11 Dec 2004 03:47:49 -0800
Subject: Re: [MMUSIC] ICE issue: obfuscating candidate addresses
From: Cullen Jennings <fluffy@cisco.com>
To: Magnus Westerlund <magnus.westerlund@ericsson.com>,
        Jonathan Rosenberg <jdrosen@cisco.com>
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On 12/8/04 8:58 AM, "Magnus Westerlund" <magnus.westerlund@ericsson.com>
wrote:

> 
> - Are these NATs that are helpfull common?

I've been trying to figure this out for awhile and I really don't think
anyone knows the answer to that questions. I have software that detects in
but have not yet run it on a ton of NATs. It is clear that 1) the vast
majority of NATs don't do this 2) there is more than 1 vendor that at one
time or another has shipped one that does.

"helpful" now there's a new word for them :-)



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Subject: Re: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
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Colin Perkins wrote:
> 
> Decomposing the URL into the "c=" and "m=" lines is very ugly. 
> Unfortunately, that ugliness is required if you want to use SDP. This is 
> why the SDPng work was started, to try to move to something less 
> horrible...

You state that as if it was a hard and fast written law. Is it?

It was one thing to decompose these parts of the communication address 
one time for RTP. But there was no URL for that. Now times have moved 
on, and URLs are common and useful, and they don't fit so well into this 
paradigm. Something has to adapt, but why aren't we free to have an open 
mind about where the adaptation is applied?

	Paul

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Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Sun, 12 Dec 2004 12:34:35 +0200
Message-ID: <AAB4B3D3CF0F454F98272CBE187FDE2F038A9F99@is0004avexu1.global.avaya.com>
Thread-Topic: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Thread-Index: AcTfkTb9mNFHmcAlRheGOR5N5s3u+wAo4niw
From: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
To: "Alan Clark" <alan.d.clark@telchemy.com>,
        "Mark Baugher" <mbaugher@cisco.com>,
        "Colin Perkins" <csp@csperkins.org>,
        "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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The short poll run a few days ago was indicating that the RFC 3611 =
packets seem to be the only ones candidates right now for selective =
encryption. There also were concerns about making a negotiation =
mechanism too complex. To keep things simple by now, but flexible for =
the future I would suggest that the document does not include the list =
of the to the candidate payload types that have a demand for running =
non-encrypted in a SRTP deployment, but rather refer to a list external =
to the document and maintained by IANA. This list will include only XR =
to begin with, and can be extended later without changing this document, =
or creating new ones.=20

Regards,

Dan



> -----Original Message-----
> From: mmusic-bounces@ietf.org=20
> [mailto:mmusic-bounces@ietf.org]On Behalf Of Alan Clark
> Sent: 11 December, 2004 4:50 PM
> To: Mark Baugher; Colin Perkins; Magnus Westerlund
> Cc: mmusic@ietf.org
> Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>=20
>=20
>=20
> It would seem that an additional draft needs to state roughly=20
> the following:
>=20
> (i) Scope - provides method for selective encryption/ unencryption of
> certain RTCP payload types
>=20
> (ii) Approach - allow for SRTCP encryption to be set ON by default but
> overridden for certain payload types
>=20
> (iii) New parameter
>=20
> SELECTIVE_SRTCP_UNENCRYPTED means that by default RTCP frames=20
> are encrypted
> with the exception of the listed payload types
>=20
> List of RTCP payload types .. SDES, BYE, RR, SR, XR,=20
> XR_VOIPMETRICS, XR_...
>=20
> Some compound types (for convenience) .. ALL_INFORMATIONAL_TYPES
> where ALL_INFORMATIONAL_TYPES =3D SR, RR as defined in RFC3550=20
> and XR payloads
> as defined in RFC3611
>=20
> (iv) Examples
>=20
>  ....., SELECTIVE_SRTCP_UNENCRYPTED:ALL_INFORMATIONAL_TYPES
>=20
> or
>=20
>  ....., SELECTIVE_SRTCP_UNENCRYPTED:RR,SR,XR_VOIPMETRICS
>=20
> Any comments on the general approach?
>=20
> Regards
>=20
> Alan
>=20
>=20
> -----Original Message-----
> From: Mark Baugher [mailto:mbaugher@cisco.com]
> Sent: Friday, December 10, 2004 6:39 PM
> To: Alan Clark
> Cc: mmusic@ietf.org; Colin Perkins
> Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>=20
>=20
> Alan,
>    I'm finishing sdescriptions-08 without any change to accommodate XR
> packets to use the E bit.  I don't yet know who is going to=20
> do the work
> on the I-D that Colin suggested to address Magnus' issues with E-bit
> usage.  As it stands now, UNENCRYPTED_SRTCP will need to be=20
> signaled in
> order to keep XR packets in the clear.
>=20
> Mark
> On Dec 10, 2004, at 1:05 PM, Alan Clark wrote:
>=20
> > Mark, Colin
> >
> > Where does this leave the issue?  Some sizeable network and telecom
> > equipment vendors are in the process of integrating support for the
> > RFC3611
> > VoIP metrics, due in a large part to service provider and enterprise
> > customer demand.  The demand exists because customers need to manage
> > their
> > VoIP deployments, hence they see RFC3611 as a key element in their
> > management infrastructure.  The accessibility of RFC3611=20
> VoIP metrics
> > is
> > therefore an issue of concern today.
> >
> > If it's proposed to deal with this in a separate draft then=20
> what is the
> > expected timeframe - going through the regular I-D, WG draft, IESG
> > process
> > seem to take a year or so. Is there some mechanism by which=20
> this can be
> > accelerated?
> >
> > If the proposal is to address the issue in the draft currently being
> > finalized then my apologies if this has caused any last=20
> minute delays
> > however the issue is important enough to warrant this (at=20
> least in my
> > view).
> > We hoped that the comment in the security considerations section of
> > RFC3611
> > was strong enough to address the issue however it appears=20
> not to have
> > been.
> >
> > Regards
> >
> > Alan
> >
> >
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org=20
> [mailto:mmusic-bounces@ietf.org]On Behalf
> > Of Mark Baugher
> > Sent: Friday, December 10, 2004 11:03 AM
> > To: Colin Perkins
> > Cc: mmusic@ietf.org
> > Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
> >
> >
> > Colin,
> >
> > On Dec 10, 2004, at 3:47 AM, Colin Perkins wrote:
> >
> >> Mark,
> >>
> >> On 8 Dec 2004, at 00:43, Mark Baugher wrote:
> >>>   Do we know of any other applications besides RFC 3611 that might
> >>> need to selectively encrypt SRTCP packets?
> >>
> >> I don't, but I can imagine future RTCP extensions which=20
> might require
> >> this feature (and applications that might want to encrypt various
> >> subsets of the RFC 3611 data).
> >
> > I don't see a large downside in waiting until we see such extensions
> > for real.
> >>
> >>>   I am okay with putting this requirement aside pending a separate
> >>> document.  But before someone goes to the trouble of=20
> writing the new
> >>> draft, it might be good to consider if it's needed or not.
> >>
> >> Even if RTCP XR is the only driver, it might be appropriate to make
> >> that a separate short draft. There's been enough discussion on the
> >> list that I'm not entirely comfortable slotting it into=20
> sdescriptions
> >> this late in the process.
> >
> > Ok.  If there are no objections, I'll generate a new=20
> version and post
> > it early next week.
> >
> > Mark
> >>
> >> Colin
> >>
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >
>=20
>=20
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>=20

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Cc: Simple WG <simple@ietf.org>, mmusic@ietf.org
Subject: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
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I also agree with Ben here. I don't see interoperability problems, since 
MSRP implementations will need to add extra support for SDP.

I also think SDP is requiring a general mechanism to be able to indicate 
a URI as a connection address, rather than the traditional IP address + 
port number... but that is another story.

Let's keep the SDP usage in MSRP as it is now.

- Miguel

Ben Campbell wrote:

> I really need to get closure on this, so I'd appreciate people agreeing  
> or disagreeing.
> 
> I have had off-list correspondence indicating both positions. In  summary:
> 
> In favor of the change, I had a comment to the effect that changing SDP  
> semantics to ignore the C line and the port field on the M line really  
> changes it to something that is no longer SDP, and that such deviations  
> are in general bad for interop.
> 
> In favor of leaving things as currently specified, I had two comments  
> that decomposing URLs into the C and M lines is ugly, and that having a  
> different mechanism to communicate endpoint addresses vs relay  
> addresses is really ugly.
> 
> So, do people agree with one or the other position? Something  
> completely different? (Hopefully the people who sent off-list  
> correspondence will reply on-list if they think I mischaracterized  
> either position. Actually, I hope they will repost on-list even if they  
> like my summaries)
> 
> I have so far avoided interjecting my personal opinion so that I could  
> do a fair summary. But, I actually tend to agree with the  
> leave-things-as-specified position.
> 
> While I am sensitive to the issue of deviating from the spirit of SDP,  
> I don't really buy the interoperability argument. The currently defined  
> usage _does_ put "msrp" in the M line--so only consumers that  
> understand what that means will be able to act on it. I assume that any  
> endpoint that understands what "msrp" means will also understand how to  
> use the "path" attribute.
> 
> The place that interoperability might suffer is if any non-endpoint  
> needs to interpret things--which is generally not a goal. I guess can  
> also see debug or monitoring tools who don't know about MSRP getting  
> things wrong.
> 
> My biggest technical objection is that using a different mechanism for  
> determining where to connect depending on whether you are connecting  
> peer to peer or through a relay, increases the chances of implementors  
> getting things wrong. And I cannot personally think of a way to handle  
> relay addresses that is more in the spirit of SDP than what we  
> currently have.
> 
> But in any case, I will be happy to document consensus, even if it  
> disagrees with my position. But we need to close on it asap.
> 
> Thanks!
> 
> Ben.
> 
> On Dec 7, 2004, at 5:20 PM, Ben Campbell wrote:
> 
>> I sent the attached to the SIMPLE list last week. I am reposting it 
>> to  both the SIMPLE and MMUSIC lists, as the discussion is of interest 
>> to  both groups. If you already responded to this on the SIMPLE list 
>> or in  private, it would be helpful if you can re-post your response 
>> to this  thread, so everyone sees the whole discussion.
>>
>> Thanks!
>>
>> Ben.
>>
>> ----------------------------------------------------------------------- 
>> -----------------------------------------
>>
>>
>> In the DC meeting, we discussed that MSRP's SDP usage should be  
>> reviewed in MMUSIC, or at least by people who are more expert in 
>> SDP.   Colin Perkins and Joerg Ott looked at this, and suggested a 
>> change.  (Many thanks to them for taking the time, and I hope they 
>> will jump in  and correct me if I mischaracterize anything.)
>>
>> The upshot of the proposal is that the fact that MSRP ignores the  
>> address in the sdp c-line, and the port and proto fields in the  
>> m-line. Instead, MSRP gets all of this from the first URL in the path  
>> a-line attribute. This deviates from the traditional usage in SDP, 
>> and  causes confusion.
>>
>> The proposed change is that the destination URL be broken into the  
>> appropriate parts and encoded into the c-line and m-line. For example  
>> (leaving out lines that don't change)
>>
>>  c=IN IP4 alice.example.com
>>  m=message 9 msrp *
>>  a=path:msrp://a.example.com:7394/2s93i;tcp
>>
>> would become:
>>
>>  c=IN IP4 a.example.com
>>  m=message 7394 msrp/tcp *
>>  a=session:2s93i
>>
>> The change would only be made for the destination URL. URLs for any  
>> relays would be expressed the same way they are now.
>>
>> The advantage of this approach is that it more closely follows the  
>> spirit of SDP. This will be easier to understand for people who  
>> already know SDP. Existing SDP code could be reused more effectively.
>>
>> The disadvantage of the proposed change is, it means the way an MSRP  
>> client determines the destination URL would be different depending on  
>> whether the URL represented the final destination or a relay. Instead  
>> of the current approach of simply copying the path attribute value  
>> into the To-Path header in MSRP messages,  a client would copy the  
>> path header value if exists, reconstitute the destination URL from 
>> the  m and c lines, and append the destination URL to the path header  
>> value.
>>
>> So, what do people think? (I note that we actually did something  
>> similar to this before we added the "path" attribute. )
>>
>>
>> Thanks!
>>
>> Ben.
>>
>>
>>
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple
>>
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 

-- 
Miguel A. Garcia           tel:+358-50-4804586
Nokia Research Center      Helsinki, Finland

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I agree with Colin on the ugliness issue of decomposing URLs into
c= and m= lines.  But this is the way SDP has worked and works today.

The present discussion we are having here is about changing SDP to
for address/URL handling in one particular case.  This does not
strike me as something MSRP should be concerned with.

I would like to make one thing clear: we are either using existing tools
in another document or we are building new ones.  There is nothing much
in between.

Changing c=/m= semantics means changing SDP.  Just because the syntax
still looks the same does not mean that it is still the same protocol.
Any entity or generic protocol engine used to dealing with SDP m=/c=
would no longer do the right thing.

We see this kind of approach quite often -- by far too often -- over
the past years (just think the many "uses" of SIP defined elsewhere,
usually resulting in something "a little but not entirely unlike" SIP).

What I would like avoid is having special treatment of SDP depending
on what the media type is (or other attributes signify).

Therefore, I strongly agree with Colin's conclusion: take SDP as is
or leave it altogether and pursue a different path.

Joerg


Colin Perkins wrote:

> On 10 Dec 2004, at 21:41, Ben Campbell wrote:
> 
>> I really need to get closure on this, so I'd appreciate people 
>> agreeing or disagreeing.
>>
>> I have had off-list correspondence indicating both positions. In summary:
>>
>> In favor of the change, I had a comment to the effect that changing 
>> SDP semantics to ignore the C line and the port field on the M line 
>> really changes it to something that is no longer SDP, and that such 
>> deviations are in general bad for interop.
>>
>> In favor of leaving things as currently specified, I had two comments 
>> that decomposing URLs into the C and M lines is ugly, and that having 
>> a different mechanism to communicate endpoint addresses vs relay 
>> addresses is really ugly.
>>
>> So, do people agree with one or the other position?
> 
> 
> Decomposing the URL into the "c=" and "m=" lines is very ugly. 
> Unfortunately, that ugliness is required if you want to use SDP. This is 
> why the SDPng work was started, to try to move to something less 
> horrible...
> 
> Colin
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 



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Joerg Ott wrote:
> I agree with Colin on the ugliness issue of decomposing URLs into
> c= and m= lines.  But this is the way SDP has worked and works today.
> 
> The present discussion we are having here is about changing SDP to
> for address/URL handling in one particular case.  This does not
> strike me as something MSRP should be concerned with.
> 
> I would like to make one thing clear: we are either using existing tools
> in another document or we are building new ones.  There is nothing much
> in between.
> 
> Changing c=/m= semantics means changing SDP.  Just because the syntax
> still looks the same does not mean that it is still the same protocol.
> Any entity or generic protocol engine used to dealing with SDP m=/c=
> would no longer do the right thing.

Can you explain what sort of generic processing of SDP that would break 
as a result of this change, but would not break as a result of adding 
MSRP in a syntax you have in mind?

There is really only a minimal amount of generic processing that can be 
done for SDP as it stands. That doesn't include actual management of 
sockets, since that is all dependent on "transport". I have worked with 
an SDP "stack", and it isn't harmed in any way by the usage proposed in 
MSRP.

The work that will be needed to support MSRP will only be made more 
complex by tearing info out of a URL and pasting it into the c- and m- 
lines.

> We see this kind of approach quite often -- by far too often -- over
> the past years (just think the many "uses" of SIP defined elsewhere,
> usually resulting in something "a little but not entirely unlike" SIP).
> 
> What I would like avoid is having special treatment of SDP depending
> on what the media type is (or other attributes signify).

Then why not just restrict SDP to use with RTP transports? That covers 
what it seems to have been designed for. Half the stuff defined in SDP 
is only applicable in that case. It has also been a real stretch to 
apply SDP to the offer/answer needs of SIP. Maybe that was an abuse too.

> Therefore, I strongly agree with Colin's conclusion: take SDP as is
> or leave it altogether and pursue a different path.

Right! Given that the only alternative in sight is SDPng, and apparently 
nobody as any belief that it will be adopted in our lifetimes.

	Paul

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From: "Alan Clark" <alan.d.clark@telchemy.com>
To: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>,
        "Mark Baugher" <mbaugher@cisco.com>,
        "Colin Perkins" <csp@csperkins.org>,
        "Magnus Westerlund" <magnus.westerlund@ericsson.com>
Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Sun, 12 Dec 2004 18:41:33 -0500
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There is already an IANA numbering system for the various report types and
presumably we should just use this

IANA already knows about:-

   FIR        full INTRA-frame request 192      [RFC2032]
   NACK       negative acknowledgement 193      [RFC2032]
   SR         sender report            200      [RFC3551]
   RR         receiver report          201      [RFC3551]
   SDES       source description       202      [RFC3551]
   BYE        goodbye                  203      [RFC3551]
   APP        application-defined      204      [RFC3551]
   XR         extended report          207      [RFC3611]

and for RTCP XR

      BT  name
      --  ----
       1  Loss RLE Report Block
       2  Duplicate RLE Report Block
       3  Packet Receipt Times Report Block
       4  Receiver Reference Time Report Block
       5  DLRR Report Block
       6  Statistics Summary Report Block
       7  VoIP Metrics Report Block

This would result in syntax along the lines of

  ....., SELECTIVE_SRTCP_UNENCRYPTED:RR,SR,XR[7],BYE

or

  ....., SELECTIVE_SRTCP_UNENCRYPTED:RR,SR,XR[6,7]

or presumably

  ....., SELECTIVE_SRTCP_UNENCRYPTED:200,201,207[6,7]

This looks like quite a short draft!!!!!  We only need to add one new
parameter to the sdescriptions to give much more flexibility.

Regards

Alan





-----Original Message-----
From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
Sent: Sunday, December 12, 2004 5:35 AM
To: Alan Clark; Mark Baugher; Colin Perkins; Magnus Westerlund
Cc: mmusic@ietf.org
Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP


The short poll run a few days ago was indicating that the RFC 3611 packets
seem to be the only ones candidates right now for selective encryption.
There also were concerns about making a negotiation mechanism too complex.
To keep things simple by now, but flexible for the future I would suggest
that the document does not include the list of the to the candidate payload
types that have a demand for running non-encrypted in a SRTP deployment, but
rather refer to a list external to the document and maintained by IANA. This
list will include only XR to begin with, and can be extended later without
changing this document, or creating new ones.

Regards,

Dan



> -----Original Message-----
> From: mmusic-bounces@ietf.org
> [mailto:mmusic-bounces@ietf.org]On Behalf Of Alan Clark
> Sent: 11 December, 2004 4:50 PM
> To: Mark Baugher; Colin Perkins; Magnus Westerlund
> Cc: mmusic@ietf.org
> Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>
>
>
> It would seem that an additional draft needs to state roughly
> the following:
>
> (i) Scope - provides method for selective encryption/ unencryption of
> certain RTCP payload types
>
> (ii) Approach - allow for SRTCP encryption to be set ON by default but
> overridden for certain payload types
>
> (iii) New parameter
>
> SELECTIVE_SRTCP_UNENCRYPTED means that by default RTCP frames
> are encrypted
> with the exception of the listed payload types
>
> List of RTCP payload types .. SDES, BYE, RR, SR, XR,
> XR_VOIPMETRICS, XR_...
>
> Some compound types (for convenience) .. ALL_INFORMATIONAL_TYPES
> where ALL_INFORMATIONAL_TYPES = SR, RR as defined in RFC3550
> and XR payloads
> as defined in RFC3611
>
> (iv) Examples
>
>  ....., SELECTIVE_SRTCP_UNENCRYPTED:ALL_INFORMATIONAL_TYPES
>
> or
>
>  ....., SELECTIVE_SRTCP_UNENCRYPTED:RR,SR,XR_VOIPMETRICS
>
> Any comments on the general approach?
>
> Regards
>
> Alan
>
>
> -----Original Message-----
> From: Mark Baugher [mailto:mbaugher@cisco.com]
> Sent: Friday, December 10, 2004 6:39 PM
> To: Alan Clark
> Cc: mmusic@ietf.org; Colin Perkins
> Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>
>
> Alan,
>    I'm finishing sdescriptions-08 without any change to accommodate XR
> packets to use the E bit.  I don't yet know who is going to
> do the work
> on the I-D that Colin suggested to address Magnus' issues with E-bit
> usage.  As it stands now, UNENCRYPTED_SRTCP will need to be
> signaled in
> order to keep XR packets in the clear.
>
> Mark
> On Dec 10, 2004, at 1:05 PM, Alan Clark wrote:
>
> > Mark, Colin
> >
> > Where does this leave the issue?  Some sizeable network and telecom
> > equipment vendors are in the process of integrating support for the
> > RFC3611
> > VoIP metrics, due in a large part to service provider and enterprise
> > customer demand.  The demand exists because customers need to manage
> > their
> > VoIP deployments, hence they see RFC3611 as a key element in their
> > management infrastructure.  The accessibility of RFC3611
> VoIP metrics
> > is
> > therefore an issue of concern today.
> >
> > If it's proposed to deal with this in a separate draft then
> what is the
> > expected timeframe - going through the regular I-D, WG draft, IESG
> > process
> > seem to take a year or so. Is there some mechanism by which
> this can be
> > accelerated?
> >
> > If the proposal is to address the issue in the draft currently being
> > finalized then my apologies if this has caused any last
> minute delays
> > however the issue is important enough to warrant this (at
> least in my
> > view).
> > We hoped that the comment in the security considerations section of
> > RFC3611
> > was strong enough to address the issue however it appears
> not to have
> > been.
> >
> > Regards
> >
> > Alan
> >
> >
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org
> [mailto:mmusic-bounces@ietf.org]On Behalf
> > Of Mark Baugher
> > Sent: Friday, December 10, 2004 11:03 AM
> > To: Colin Perkins
> > Cc: mmusic@ietf.org
> > Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
> >
> >
> > Colin,
> >
> > On Dec 10, 2004, at 3:47 AM, Colin Perkins wrote:
> >
> >> Mark,
> >>
> >> On 8 Dec 2004, at 00:43, Mark Baugher wrote:
> >>>   Do we know of any other applications besides RFC 3611 that might
> >>> need to selectively encrypt SRTCP packets?
> >>
> >> I don't, but I can imagine future RTCP extensions which
> might require
> >> this feature (and applications that might want to encrypt various
> >> subsets of the RFC 3611 data).
> >
> > I don't see a large downside in waiting until we see such extensions
> > for real.
> >>
> >>>   I am okay with putting this requirement aside pending a separate
> >>> document.  But before someone goes to the trouble of
> writing the new
> >>> draft, it might be good to consider if it's needed or not.
> >>
> >> Even if RTCP XR is the only driver, it might be appropriate to make
> >> that a separate short draft. There's been enough discussion on the
> >> list that I'm not entirely comfortable slotting it into
> sdescriptions
> >> this late in the process.
> >
> > Ok.  If there are no objections, I'll generate a new
> version and post
> > it early next week.
> >
> > Mark
> >>
> >> Colin
> >>
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >
>
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>


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From mmusic-bounces@ietf.org  Mon Dec 13 04:25:38 2004
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From: Hisham Khartabil <hisham.khartabil@telio.no>
Subject: Re: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
Date: Mon, 13 Dec 2004 10:10:06 +0100
To: Joerg Ott <jo@tzi.uni-bremen.de>
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(not as chair, but as an MSRP protocol contributor)

The objection to using the SDP c and m lines is not purely a cosmetic 
objection. We are not saying that SDP the way it is defined today is 
ugly so we want to rearrange it to look better.

We have a fundamental objection: having 2 ways to construct a next hop 
URI that depends on the entity that is the next hop is not acceptable. 
Endpoints should not be required  to have 2 engines implemented. Unless 
we are given an alternative that allows endpoint implementers to 
implement one piece of code for constructing an next hop MSRP URI, I'm 
afraid we have no choice but to keep what we have today in MSRP as is.

You telling us to go find another protocol if we don't like SDP the way 
it is today is hardly contructive. SDPng is not an option and we call 
know it.

Let me add that comedia also violates what you claim to be a 
fundamental piece of the protocol that if broken will cause the SDP 
protocol not to be SDP any longer. An argument that I don't buy. We 
have violated things before:

    - t= line does not mean anything in SIP and is set to 0 0 (which in 
SDP means that the session is permanent and unbounded).
    - s= line carries session name. It is set to - when used with SIP.

Also, quoting the SDP draft:

"For unicast IP sessions, the following are conveyed:

    o  The remote address for media

    o  The remote transport port for media

    The semantics of this address and port depend on the media and
    transport protocol defined.  By default, this SHOULD be the remote
    address and remote port to which data is sent.  Some media types MAY
    redefine this behaviour, but this is NOT RECOMMENDED."

We are changing the semantics. As far as I understand from the quoted 
paragraph above, we are not violating anything, even-though it is not 
recommended.

I have a question to SDP stack implementers that plan to implement 
MSRP: Does the current way MSRP specifies the use of SDP (ignore the c 
line address and the m line port and instead use the attribute a=path) 
cause problems to you and requires you to do some hacking in your SDP 
stack? If so, is that acceptable or do you prefer for your MSRP 
implementation to implement two ways of contructing an MSRP URL 
depending on the what the next hop is?

Regards,
Hisham

On Dec 12, 2004, at 12:58 PM, Joerg Ott wrote:

> I agree with Colin on the ugliness issue of decomposing URLs into
> c= and m= lines.  But this is the way SDP has worked and works today.
>
> The present discussion we are having here is about changing SDP to
> for address/URL handling in one particular case.  This does not
> strike me as something MSRP should be concerned with.
>
> I would like to make one thing clear: we are either using existing 
> tools
> in another document or we are building new ones.  There is nothing much
> in between.
>
> Changing c=/m= semantics means changing SDP.  Just because the syntax
> still looks the same does not mean that it is still the same protocol.
> Any entity or generic protocol engine used to dealing with SDP m=/c=
> would no longer do the right thing.
>
> We see this kind of approach quite often -- by far too often -- over
> the past years (just think the many "uses" of SIP defined elsewhere,
> usually resulting in something "a little but not entirely unlike" SIP).
>
> What I would like avoid is having special treatment of SDP depending
> on what the media type is (or other attributes signify).
>
> Therefore, I strongly agree with Colin's conclusion: take SDP as is
> or leave it altogether and pursue a different path.
>
> Joerg
>
>
> Colin Perkins wrote:
>
>> On 10 Dec 2004, at 21:41, Ben Campbell wrote:
>>> I really need to get closure on this, so I'd appreciate people 
>>> agreeing or disagreeing.
>>>
>>> I have had off-list correspondence indicating both positions. In 
>>> summary:
>>>
>>> In favor of the change, I had a comment to the effect that changing 
>>> SDP semantics to ignore the C line and the port field on the M line 
>>> really changes it to something that is no longer SDP, and that such 
>>> deviations are in general bad for interop.
>>>
>>> In favor of leaving things as currently specified, I had two 
>>> comments that decomposing URLs into the C and M lines is ugly, and 
>>> that having a different mechanism to communicate endpoint addresses 
>>> vs relay addresses is really ugly.
>>>
>>> So, do people agree with one or the other position?
>> Decomposing the URL into the "c=" and "m=" lines is very ugly. 
>> Unfortunately, that ugliness is required if you want to use SDP. This 
>> is why the SDPng work was started, to try to move to something less 
>> horrible...
>> Colin
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>
>
>
> _______________________________________________
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> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>


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Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Mon, 13 Dec 2004 11:51:29 +0200
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From: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
To: "Alan Clark" <alan.d.clark@telchemy.com>,
        "Mark Baugher" <mbaugher@cisco.com>,
        "Colin Perkins" <csp@csperkins.org>,
        "Magnus Westerlund" <magnus.westerlund@ericsson.com>
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Previous discussions revealed concerns about extensive complexity, =
especially in implementation and testing, in the situation that today we =
know that only there is a need only for the 3611 report type to be =
transmitted as selectively unencrypted. My suggestion is that we define =
in IANA the subset of the report types that need to be transmitted =
unencrypted. The list would include only XR extended reports 207 by now =
and may be extended in the future. The draft remains similarly short, =
but the syntax is to be defined in a manner generic enough to cover by =
reference future extensions.=20

Regards,

Dan



> -----Original Message-----
> From: Alan Clark [mailto:alan.d.clark@telchemy.com]
> Sent: 13 December, 2004 1:42 AM
> To: Romascanu, Dan (Dan); Mark Baugher; Colin Perkins; Magnus=20
> Westerlund
> Cc: mmusic@ietf.org
> Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>=20
>=20
>=20
> There is already an IANA numbering system for the various=20
> report types and
> presumably we should just use this
>=20
> IANA already knows about:-
>=20
>    FIR        full INTRA-frame request 192      [RFC2032]
>    NACK       negative acknowledgement 193      [RFC2032]
>    SR         sender report            200      [RFC3551]
>    RR         receiver report          201      [RFC3551]
>    SDES       source description       202      [RFC3551]
>    BYE        goodbye                  203      [RFC3551]
>    APP        application-defined      204      [RFC3551]
>    XR         extended report          207      [RFC3611]
>=20
> and for RTCP XR
>=20
>       BT  name
>       --  ----
>        1  Loss RLE Report Block
>        2  Duplicate RLE Report Block
>        3  Packet Receipt Times Report Block
>        4  Receiver Reference Time Report Block
>        5  DLRR Report Block
>        6  Statistics Summary Report Block
>        7  VoIP Metrics Report Block
>=20
> This would result in syntax along the lines of
>=20
>   ....., SELECTIVE_SRTCP_UNENCRYPTED:RR,SR,XR[7],BYE
>=20
> or
>=20
>   ....., SELECTIVE_SRTCP_UNENCRYPTED:RR,SR,XR[6,7]
>=20
> or presumably
>=20
>   ....., SELECTIVE_SRTCP_UNENCRYPTED:200,201,207[6,7]
>=20
> This looks like quite a short draft!!!!!  We only need to add one new
> parameter to the sdescriptions to give much more flexibility.
>=20
> Regards
>=20
> Alan
>=20
>=20
>=20
>=20
>=20
> -----Original Message-----
> From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
> Sent: Sunday, December 12, 2004 5:35 AM
> To: Alan Clark; Mark Baugher; Colin Perkins; Magnus Westerlund
> Cc: mmusic@ietf.org
> Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>=20
>=20
> The short poll run a few days ago was indicating that the RFC=20
> 3611 packets
> seem to be the only ones candidates right now for selective=20
> encryption.
> There also were concerns about making a negotiation mechanism=20
> too complex.
> To keep things simple by now, but flexible for the future I=20
> would suggest
> that the document does not include the list of the to the=20
> candidate payload
> types that have a demand for running non-encrypted in a SRTP=20
> deployment, but
> rather refer to a list external to the document and=20
> maintained by IANA. This
> list will include only XR to begin with, and can be extended=20
> later without
> changing this document, or creating new ones.
>=20
> Regards,
>=20
> Dan
>=20
>=20
>=20
> > -----Original Message-----
> > From: mmusic-bounces@ietf.org
> > [mailto:mmusic-bounces@ietf.org]On Behalf Of Alan Clark
> > Sent: 11 December, 2004 4:50 PM
> > To: Mark Baugher; Colin Perkins; Magnus Westerlund
> > Cc: mmusic@ietf.org
> > Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
> >
> >
> >
> > It would seem that an additional draft needs to state roughly
> > the following:
> >
> > (i) Scope - provides method for selective encryption/=20
> unencryption of
> > certain RTCP payload types
> >
> > (ii) Approach - allow for SRTCP encryption to be set ON by=20
> default but
> > overridden for certain payload types
> >
> > (iii) New parameter
> >
> > SELECTIVE_SRTCP_UNENCRYPTED means that by default RTCP frames
> > are encrypted
> > with the exception of the listed payload types
> >
> > List of RTCP payload types .. SDES, BYE, RR, SR, XR,
> > XR_VOIPMETRICS, XR_...
> >
> > Some compound types (for convenience) .. ALL_INFORMATIONAL_TYPES
> > where ALL_INFORMATIONAL_TYPES =3D SR, RR as defined in RFC3550
> > and XR payloads
> > as defined in RFC3611
> >
> > (iv) Examples
> >
> >  ....., SELECTIVE_SRTCP_UNENCRYPTED:ALL_INFORMATIONAL_TYPES
> >
> > or
> >
> >  ....., SELECTIVE_SRTCP_UNENCRYPTED:RR,SR,XR_VOIPMETRICS
> >
> > Any comments on the general approach?
> >
> > Regards
> >
> > Alan
> >
> >
> > -----Original Message-----
> > From: Mark Baugher [mailto:mbaugher@cisco.com]
> > Sent: Friday, December 10, 2004 6:39 PM
> > To: Alan Clark
> > Cc: mmusic@ietf.org; Colin Perkins
> > Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
> >
> >
> > Alan,
> >    I'm finishing sdescriptions-08 without any change to=20
> accommodate XR
> > packets to use the E bit.  I don't yet know who is going to
> > do the work
> > on the I-D that Colin suggested to address Magnus' issues with E-bit
> > usage.  As it stands now, UNENCRYPTED_SRTCP will need to be
> > signaled in
> > order to keep XR packets in the clear.
> >
> > Mark
> > On Dec 10, 2004, at 1:05 PM, Alan Clark wrote:
> >
> > > Mark, Colin
> > >
> > > Where does this leave the issue?  Some sizeable network=20
> and telecom
> > > equipment vendors are in the process of integrating=20
> support for the
> > > RFC3611
> > > VoIP metrics, due in a large part to service provider and=20
> enterprise
> > > customer demand.  The demand exists because customers=20
> need to manage
> > > their
> > > VoIP deployments, hence they see RFC3611 as a key element in their
> > > management infrastructure.  The accessibility of RFC3611
> > VoIP metrics
> > > is
> > > therefore an issue of concern today.
> > >
> > > If it's proposed to deal with this in a separate draft then
> > what is the
> > > expected timeframe - going through the regular I-D, WG draft, IESG
> > > process
> > > seem to take a year or so. Is there some mechanism by which
> > this can be
> > > accelerated?
> > >
> > > If the proposal is to address the issue in the draft=20
> currently being
> > > finalized then my apologies if this has caused any last
> > minute delays
> > > however the issue is important enough to warrant this (at
> > least in my
> > > view).
> > > We hoped that the comment in the security considerations=20
> section of
> > > RFC3611
> > > was strong enough to address the issue however it appears
> > not to have
> > > been.
> > >
> > > Regards
> > >
> > > Alan
> > >
> > >
> > > -----Original Message-----
> > > From: mmusic-bounces@ietf.org
> > [mailto:mmusic-bounces@ietf.org]On Behalf
> > > Of Mark Baugher
> > > Sent: Friday, December 10, 2004 11:03 AM
> > > To: Colin Perkins
> > > Cc: mmusic@ietf.org
> > > Subject: Re: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
> > >
> > >
> > > Colin,
> > >
> > > On Dec 10, 2004, at 3:47 AM, Colin Perkins wrote:
> > >
> > >> Mark,
> > >>
> > >> On 8 Dec 2004, at 00:43, Mark Baugher wrote:
> > >>>   Do we know of any other applications besides RFC 3611=20
> that might
> > >>> need to selectively encrypt SRTCP packets?
> > >>
> > >> I don't, but I can imagine future RTCP extensions which
> > might require
> > >> this feature (and applications that might want to encrypt various
> > >> subsets of the RFC 3611 data).
> > >
> > > I don't see a large downside in waiting until we see such=20
> extensions
> > > for real.
> > >>
> > >>>   I am okay with putting this requirement aside pending=20
> a separate
> > >>> document.  But before someone goes to the trouble of
> > writing the new
> > >>> draft, it might be good to consider if it's needed or not.
> > >>
> > >> Even if RTCP XR is the only driver, it might be=20
> appropriate to make
> > >> that a separate short draft. There's been enough=20
> discussion on the
> > >> list that I'm not entirely comfortable slotting it into
> > sdescriptions
> > >> this late in the process.
> > >
> > > Ok.  If there are no objections, I'll generate a new
> > version and post
> > > it early next week.
> > >
> > > Mark
> > >>
> > >> Colin
> > >>
> > >
> > > _______________________________________________
> > > mmusic mailing list
> > > mmusic@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/mmusic
> > >
> > >
> > > _______________________________________________
> > > mmusic mailing list
> > > mmusic@ietf.org
> > > https://www1.ietf.org/mailman/listinfo/mmusic
> > >
> >
> >
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >
>=20
>=20

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--NextPart

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

	Title		: SDP: Session Description Protocol
	Author(s)	: M. Handley, et al.
	Filename	: draft-ietf-mmusic-sdp-new-23.txt
	Pages		: 48
	Date		: 2004-12-13
	
This memo defines the Session Description Protocol (SDP). SDP is
intended for describing multimedia sessions for the purposes of
session announcement, session invitation, and other forms of
multimedia session initiation.

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From mmusic-bounces@ietf.org  Mon Dec 13 17:50:01 2004
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Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-new-23.txt
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On 13 Dec 2004, at 20:33, Internet-Drafts@ietf.org wrote:
> 	Title		: SDP: Session Description Protocol
> 	Author(s)	: M. Handley, et al.
> 	Filename	: draft-ietf-mmusic-sdp-new-23.txt
> 	Pages		: 48
> 	Date		: 2004-12-13
> 	
> This memo defines the Session Description Protocol (SDP). SDP is
> intended for describing multimedia sessions for the purposes of
> session announcement, session invitation, and other forms of
> multimedia session initiation.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-new-23.txt

Changes in this version:
- Fix section references in IANA considerations; expand "Encoding
   Considerations"
- Note that the <username> and <unicast-address> in "o=" may be chosen 
to
   obfuscate the actual values, provided this does not compromise the 
global
   uniqueness of the field.

Colin


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Date: Mon, 13 Dec 2004 23:45:50 -0800 (PST)
From: Stephen Casner <casner@acm.org>
To: "Romascanu, Dan (Dan)" <dromasca@avaya.com>
Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
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On Mon, 13 Dec 2004, Romascanu, Dan (Dan) wrote:

> Previous discussions revealed concerns about extensive complexity,
> especially in implementation and testing, in the situation that
> today we know that only there is a need only for the 3611 report
> type to be transmitted as selectively unencrypted. My suggestion is
> that we define in IANA the subset of the report types that need to
> be transmitted unencrypted. The list would include only XR extended
> reports 207 by now and may be extended in the future.

This means I write my application today using the selectively
unencrypted RTCP option knowning that only XR is unencrypted, then
tomorrow my implementation talks to some new implementation that
knows the IANA registry has changed to include XR and FOO, so both
packet types are sent to me unencrypted when I don't expect it!

This is not feasible.  Security policies would be violated.

                                                        -- Steve

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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Hi James,

> I thought the idea of preconditions was that they need to be met prior 
> to session establishment. I believe this is when the precondition state 
> table is identical in both UAs.

The tables do not actually need to be the same in both UAs. The tables 
that need to be equal are the table showing the current status and the 
table showing the desired status of the preconditions. Furthermore, in 
our case, the tables need only to be equal at the UAS. That's why the 
UAC does not ask for updates on the status. Once the UAS knows that both 
tables are equal, it resumes session establishment without explicitly 
informing the UAC.

In this example, the UAS receives an INVITE that says: "do not alert the 
user until the TCP connection is established".

The rest of the SIP messages exchanged between both ends do not have 
anything to do with the precondition. They have to do with establishing 
the TCP connection.

Once the TCP connection is established, the precondition is met. Now the 
UAS could send a SIP message to the UAC saying "the current status is 
now sendrecv so I will now resume session establishment"... but the UAS 
does not actually need to do this, because the 180 Ringing will inform 
the UAC anyway that session establishment has resumed. That's why you do 
not see any message with "curr:conn e2e sendserv" in the flow.

Note that when dealing with this type of precondition, UAs usually get 
local information on the status of the connection. That is, the UAs know
that the TCP connection is established because they gets this 
information from their own socket. That's why we do not usually need to 
use SIP to inform the other end about this type of event.

In any case, if you want me to clarify something in the text let me know 
and I will do it.

Thanks,

Gonzalo


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From mmusic-bounces@ietf.org  Tue Dec 14 05:02:11 2004
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Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Tue, 14 Dec 2004 11:55:00 +0200
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Thread-Topic: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
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From: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
To: "Stephen Casner" <casner@acm.org>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, mmusic@ietf.org,
        Mark Baugher <mbaugher@cisco.com>, Colin Perkins <csp@csperkins.org>
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See in-line.=20

Regards,

Dan



> -----Original Message-----
> From: casner@packetdesign.com=20
> [mailto:casner@packetdesign.com]On Behalf Of Stephen Casner
> Sent: 14 December, 2004 9:46 AM
> To: Romascanu, Dan (Dan)
> Cc: Alan Clark; Mark Baugher; Colin Perkins; Magnus=20
> Westerlund; mmusic@ietf.org
> Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>=20
>=20
> On Mon, 13 Dec 2004, Romascanu, Dan (Dan) wrote:
>=20
> > Previous discussions revealed concerns about extensive complexity,
> > especially in implementation and testing, in the situation that
> > today we know that only there is a need only for the 3611 report
> > type to be transmitted as selectively unencrypted. My suggestion is
> > that we define in IANA the subset of the report types that need to
> > be transmitted unencrypted. The list would include only XR extended
> > reports 207 by now and may be extended in the future.
>=20
> This means I write my application today using the selectively
> unencrypted RTCP option knowning that only XR is unencrypted, then
> tomorrow my implementation talks to some new implementation that
> knows the IANA registry has changed to include XR and FOO, so both
> packet types are sent to me unencrypted when I don't expect it!
>=20
> This is not feasible.  Security policies would be violated.

The alternative is to define that only XR packets can run un-encrypted =
and every time new cases requiring other types to run un-encrypted show =
up re-write the standard. This looks to un-flexible to me.=20

Note that security policies may require in some cases that even XR be =
encrypted. In the enterprise space I can assure you that requiring end =
devices to transmit data unencrypted over Service Providers backbones =
may be a problem, because one is trading security for monitoring. That's =
why I believe that we need anyway some means of refusing to receive =
un-encrypted packets of a certain type - for example do not start =
sending un-encrypted packets until you received the information from the =
other end that it intends to do the same.=20

>=20
>                                                         -- Steve
>=20

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Hi Magnus,

> 1. Section 4: No ABNF definition of "token". I assume that you use that 
> of SDP. However it needs to be either referenced or explicitly defined.

yes, I got this comment from Colin before in a conversation we had. We 
will reference SDPnew and clarify that the attribute value is not 
arbitrary text but a token.

> 2. No reference to what BNF you are using. I guess a reference to RFC 
> 2234 is all that is needed. Otherwise, there would however be some 
> uncertainties about spacing. Is any white spaces allowed between 
> "a=label:" and the pointer. In RFC 2234 BNF the syntax would say no. In 
> some other BNF the answer may be yes.

We will reference RFC 2234.

> 3. Section 5. I am a bit worried about the offer/answer section. It is 
> very unclear on the handling of this attribute. Can we really trust 
> applications to define such behavior? From my work with Offer/Answer I 
> sometimes find it difficult to get the right behavior out of SDP in 
> offer/answer usage. If this behavior is left to the application then 
> there is need for a much more explicit section that explains the 
> different choices. Also I think using application defined behavior will 
> result in quite divergent behavior.

If this was an attribute that had any influence in the offer/answer 
behabior I would agree with you. But this attribute, from the 
offer/answer point of view, is just informational. That's why offerers 
and answerers can use it as the wish. It does not have any influence in 
the session being established.

The behavior we expect applications to define would be something like, 
"use the same label in the media line as in the XML document you are 
using to describe your conference". Note that this attribute is always 
for the remote's UA information, so, the values that this attribute 
takes in an answer is completely independent from the value it takes in 
the offer.


> 4. Reference [4] is only informative.

We'll fix it.

Thanks for the comments,

Gonzalo


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From mmusic-bounces@ietf.org  Tue Dec 14 11:32:51 2004
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From: "Alan Clark" <alan.d.clark@telchemy.com>
To: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>,
        "Stephen Casner" <casner@acm.org>
Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Tue, 14 Dec 2004 11:29:06 -0500
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Dan

The concept is to allow all RTCP packets to be sent in encrypted form if the
security policy requires it, and to allow either all RTCP packets or
selected types of payload to be sent unencrypted if that is acceptable to
the network operator/ enterprise/ service provider.  What's the problem?

Alan


-----Original Message-----
From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
Sent: Tuesday, December 14, 2004 4:55 AM
To: Stephen Casner
Cc: Alan Clark; Mark Baugher; Colin Perkins; Magnus Westerlund;
mmusic@ietf.org
Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP


See in-line.

Regards,

Dan



> -----Original Message-----
> From: casner@packetdesign.com
> [mailto:casner@packetdesign.com]On Behalf Of Stephen Casner
> Sent: 14 December, 2004 9:46 AM
> To: Romascanu, Dan (Dan)
> Cc: Alan Clark; Mark Baugher; Colin Perkins; Magnus
> Westerlund; mmusic@ietf.org
> Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>
>
> On Mon, 13 Dec 2004, Romascanu, Dan (Dan) wrote:
>
> > Previous discussions revealed concerns about extensive complexity,
> > especially in implementation and testing, in the situation that
> > today we know that only there is a need only for the 3611 report
> > type to be transmitted as selectively unencrypted. My suggestion is
> > that we define in IANA the subset of the report types that need to
> > be transmitted unencrypted. The list would include only XR extended
> > reports 207 by now and may be extended in the future.
>
> This means I write my application today using the selectively
> unencrypted RTCP option knowning that only XR is unencrypted, then
> tomorrow my implementation talks to some new implementation that
> knows the IANA registry has changed to include XR and FOO, so both
> packet types are sent to me unencrypted when I don't expect it!
>
> This is not feasible.  Security policies would be violated.

The alternative is to define that only XR packets can run un-encrypted and
every time new cases requiring other types to run un-encrypted show up
re-write the standard. This looks to un-flexible to me.

Note that security policies may require in some cases that even XR be
encrypted. In the enterprise space I can assure you that requiring end
devices to transmit data unencrypted over Service Providers backbones may be
a problem, because one is trading security for monitoring. That's why I
believe that we need anyway some means of refusing to receive un-encrypted
packets of a certain type - for example do not start sending un-encrypted
packets until you received the information from the other end that it
intends to do the same.

>
>                                                         -- Steve
>


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From mmusic-bounces@ietf.org  Tue Dec 14 11:47:03 2004
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Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Date: Tue, 14 Dec 2004 18:35:36 +0200
Message-ID: <AAB4B3D3CF0F454F98272CBE187FDE2F06E306B9@is0004avexu1.global.avaya.com>
Thread-Topic: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
Thread-Index: AcTh+hFac/4vFM79R0uAu29C0ezcXQAAEUOw
From: "Romascanu, Dan \(Dan\)" <dromasca@avaya.com>
To: "Alan Clark" <alan.d.clark@telchemy.com>,
        "Stephen Casner" <casner@acm.org>
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Cc: Magnus Westerlund <magnus.westerlund@ericsson.com>, mmusic@ietf.org,
        Mark Baugher <mbaugher@cisco.com>, Colin Perkins <csp@csperkins.org>
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Alan,

I understand the concept, and there is no problem.=20

The discussion is about how flexible we would like the mechanism =
described in the document to be. Stephen seems to favor a mechanism that =
designates only the XR packet types to be covered by the standard, I =
would prefer a more flexible mechanism, that refers a list maintained by =
IANA (subset of the full list) which may include other types than XR be =
added in the future.=20

Regards,

Dan



> -----Original Message-----
> From: Alan Clark [mailto:alan.d.clark@telchemy.com]
> Sent: 14 December, 2004 6:29 PM
> To: Romascanu, Dan (Dan); Stephen Casner
> Cc: Mark Baugher; Colin Perkins; Magnus Westerlund; mmusic@ietf.org
> Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>=20
>=20
> Dan
>=20
> The concept is to allow all RTCP packets to be sent in=20
> encrypted form if the
> security policy requires it, and to allow either all RTCP packets or
> selected types of payload to be sent unencrypted if that is=20
> acceptable to
> the network operator/ enterprise/ service provider.  What's=20
> the problem?
>=20
> Alan
>=20
>=20
> -----Original Message-----
> From: Romascanu, Dan (Dan) [mailto:dromasca@avaya.com]
> Sent: Tuesday, December 14, 2004 4:55 AM
> To: Stephen Casner
> Cc: Alan Clark; Mark Baugher; Colin Perkins; Magnus Westerlund;
> mmusic@ietf.org
> Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
>=20
>=20
> See in-line.
>=20
> Regards,
>=20
> Dan
>=20
>=20
>=20
> > -----Original Message-----
> > From: casner@packetdesign.com
> > [mailto:casner@packetdesign.com]On Behalf Of Stephen Casner
> > Sent: 14 December, 2004 9:46 AM
> > To: Romascanu, Dan (Dan)
> > Cc: Alan Clark; Mark Baugher; Colin Perkins; Magnus
> > Westerlund; mmusic@ietf.org
> > Subject: RE: [MMUSIC] RE: [AVT] RFC 3711 and RTCP
> >
> >
> > On Mon, 13 Dec 2004, Romascanu, Dan (Dan) wrote:
> >
> > > Previous discussions revealed concerns about extensive complexity,
> > > especially in implementation and testing, in the situation that
> > > today we know that only there is a need only for the 3611 report
> > > type to be transmitted as selectively unencrypted. My=20
> suggestion is
> > > that we define in IANA the subset of the report types that need to
> > > be transmitted unencrypted. The list would include only=20
> XR extended
> > > reports 207 by now and may be extended in the future.
> >
> > This means I write my application today using the selectively
> > unencrypted RTCP option knowning that only XR is unencrypted, then
> > tomorrow my implementation talks to some new implementation that
> > knows the IANA registry has changed to include XR and FOO, so both
> > packet types are sent to me unencrypted when I don't expect it!
> >
> > This is not feasible.  Security policies would be violated.
>=20
> The alternative is to define that only XR packets can run=20
> un-encrypted and
> every time new cases requiring other types to run un-encrypted show up
> re-write the standard. This looks to un-flexible to me.
>=20
> Note that security policies may require in some cases that even XR be
> encrypted. In the enterprise space I can assure you that requiring end
> devices to transmit data unencrypted over Service Providers=20
> backbones may be
> a problem, because one is trading security for monitoring.=20
> That's why I
> believe that we need anyway some means of refusing to receive=20
> un-encrypted
> packets of a certain type - for example do not start sending=20
> un-encrypted
> packets until you received the information from the other end that it
> intends to do the same.
>=20
> >
> >                                                         -- Steve
> >
>=20
>=20

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From mmusic-bounces@ietf.org  Tue Dec 14 18:45:25 2004
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Subject: [MMUSIC] FYI: Use port 8554 rather than port 7070 for RTSP
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Many current RTSP server implementations use port 7070 for RTSP (in 
addition to the default RTSP port 554).  Until recently, I had just assumed 
that 7070 was an 'official' alternate port number for RTSP.  However, 
looking at the official IANA registry
         <http://www.iana.org/assignments/port-numbers>
I discovered, to my surprise, that port 7070 is assigned to some other 
protocol: "ARCP".  (I don't know what protocol this is.)

Instead, port 8554 is assigned to RTSP:

     rtsp-alt   8554/tcp    RTSP Alternate (see port 554)
     rtsp-alt   8554/udp    RTSP Alternate (see port 554)

I thought I'd mention this on this ("mmusic") mailing list, because it does 
not seem to be common knowledge.

         Ross.

	Ross Finlayson
	LIVE.COM
	<http://www.live.com/>


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To: "Gonzalo Camarillo (JO/LMF)" <gonzalo.camarillo@ericsson.com>
Subject: Re: [MMUSIC] Working group last
	call:	draft-ietf-mmusic-sdp-media-label-00.txt
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Hi Gonzalo,

The proposed changes looks fine. See below about point 3.

Gonzalo Camarillo (JO/LMF) wrote:
>  > 3. Section 5. I am a bit worried about the offer/answer section. It is
>  > very unclear on the handling of this attribute. Can we really trust
>  > applications to define such behavior? From my work with Offer/Answer I
>  > sometimes find it difficult to get the right behavior out of SDP in
>  > offer/answer usage. If this behavior is left to the application then
>  > there is need for a much more explicit section that explains the
>  > different choices. Also I think using application defined behavior will
>  > result in quite divergent behavior.
> 
> If this was an attribute that had any influence in the offer/answer
> behabior I would agree with you. But this attribute, from the
> offer/answer point of view, is just informational. That's why offerers
> and answerers can use it as the wish. It does not have any influence in
> the session being established.
> 
> The behavior we expect applications to define would be something like,
> "use the same label in the media line as in the XML document you are
> using to describe your conference". Note that this attribute is always
> for the remote's UA information, so, the values that this attribute
> takes in an answer is completely independent from the value it takes in
> the offer.
> 

Okay, I do understand. I think it is okay with the current text.


Cheers

Magnus Westerlund

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

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

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

 

Get all details at:

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

 

 


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<p class=3DMsoNormal><font size=3D2 face=3DArial><span =
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font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><strong><b><font size=3D1 face=3DArial><span =
lang=3DEN-GB
style=3D'font-size:9.0pt;font-family:Arial'>&#8226; =
</span></font></b></strong><font
size=3D1 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:9.0pt;font-family:Arial'>What
is the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>business
model</span></font></b></strong> for WiMAX? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>What
do we learn from <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>earlier
deployments</span></font></b></strong>? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>What
about the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>future
extensions</span></font></b></strong> of the standard? <br>
<strong><b><font face=3DArial><span style=3D'font-family:Arial'>&#8226; =
</span></font></b></strong>How
is addressed the <strong><b><font face=3DArial><span =
style=3D'font-family:Arial'>interoperability</span></font></b></strong>
challenge? <br>
<br>
These questions, among others, will be addressed during the second =
edition of
the WiMAX Summit to be organised in <st1:place w:st=3D"on"><st1:City =
w:st=3D"on">Paris</st1:City></st1:place>
next 5-8 April 2005.<o:p></o:p></span></font></p>

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<p class=3DMsoNormal><font size=3D1 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
9.0pt;font-family:Arial'>Get all details =
at:<o:p></o:p></span></font></p>

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style=3D'font-size:9.0pt;
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title=3D"http://www.upperside.fr/wimax05/wimax2005intro.htm"><span =
lang=3DEN-GB><span
title=3D"http://www.upperside.fr/wimax05/wimax2005intro.htm">http://www.u=
pperside.fr/wimax05/wimax2005intro.htm</span></span></a></span></font><fo=
nt
size=3D1 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:9.0pt;font-family:Arial'><o:p></o:p></span></font></p>=


<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
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Subject: [MMUSIC] ICE-03 minor typos
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Some minor typos in the ICE-03 draft.

Jon

--- draft-ietf-mmusic-ice-03.txt        2004-11-09 10:49:40.000000000 -0500
+++ draft-ietf-mmusic-ice-03.txt.jon    2004-12-15 14:07:00.000000000 -0500
@@ -306,7 +306,7 @@

    Transport Address: The combination of an IP address and port.

-   Local Transport Address: A local transport address a transport
+   Local Transport Address: A local transport address is a transport
       address that has been allocated from the operating system on the
       host.  This includes transport addresses obtained through Virtual
       Private Networks (VPNs) and transport addresses obtained through
@@ -1054,7 +1054,7 @@
    (although that fact is not signaled in the message).  The usernames
    and passwords for these transport addresses are shown in Figure 3.
    The initiate message is sent to the responder.  The responder has a
-   local transport address (10.0.1.76:43443), and a a STUN derived
+   local transport address (10.0.1.76:43443), and a STUN derived
    transport address (192.0.2.64:54766) derived from (10.0.1.76:43444).
    Call these two local transport addresses X and Y respectively.  The

@@ -2077,7 +2077,7 @@

    Attribute Name: candidate

-   Long Form: candidiate
+   Long Form: candidate

    Type of Attribute: media level


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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] Working group last
	call:	draft-ietf-mmusic-sdp-media-label-00.txt
Date: Fri, 17 Dec 2004 13:13:13 +0000
To: Magnus Westerlund <magnus.westerlund@ericsson.com>
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This last call ends today. It looks like there are minor issues to be 
resolved in an updated draft. Once the revised draft is available, 
we'll allow a short time to check that all comments have been 
addressed, then advance the draft.

Colin



On 15 Dec 2004, at 12:59, Magnus Westerlund wrote:

> Hi Gonzalo,
>
> The proposed changes looks fine. See below about point 3.
>
> Gonzalo Camarillo (JO/LMF) wrote:
>>  > 3. Section 5. I am a bit worried about the offer/answer section. 
>> It is
>>  > very unclear on the handling of this attribute. Can we really trust
>>  > applications to define such behavior? From my work with 
>> Offer/Answer I
>>  > sometimes find it difficult to get the right behavior out of SDP in
>>  > offer/answer usage. If this behavior is left to the application 
>> then
>>  > there is need for a much more explicit section that explains the
>>  > different choices. Also I think using application defined behavior 
>> will
>>  > result in quite divergent behavior.
>> If this was an attribute that had any influence in the offer/answer
>> behabior I would agree with you. But this attribute, from the
>> offer/answer point of view, is just informational. That's why offerers
>> and answerers can use it as the wish. It does not have any influence 
>> in
>> the session being established.
>> The behavior we expect applications to define would be something like,
>> "use the same label in the media line as in the XML document you are
>> using to describe your conference". Note that this attribute is always
>> for the remote's UA information, so, the values that this attribute
>> takes in an answer is completely independent from the value it takes 
>> in
>> the offer.
>
> Okay, I do understand. I think it is okay with the current text.
>
>
> Cheers
>
> Magnus Westerlund
>
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com
>
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From: Ben Campbell <ben@nostrum.com>
Subject: Re: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
Date: Fri, 17 Dec 2004 16:17:58 -0600
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On Dec 12, 2004, at 5:58 AM, Joerg Ott wrote:

> I agree with Colin on the ugliness issue of decomposing URLs into
> c= and m= lines.  But this is the way SDP has worked and works today.
>
> The present discussion we are having here is about changing SDP to
> for address/URL handling in one particular case.  This does not
> strike me as something MSRP should be concerned with.
>
> I would like to make one thing clear: we are either using existing 
> tools
> in another document or we are building new ones.  There is nothing much
> in between.
>
> Changing c=/m= semantics means changing SDP.  Just because the syntax
> still looks the same does not mean that it is still the same protocol.
> Any entity or generic protocol engine used to dealing with SDP m=/c=
> would no longer do the right thing.
>
> We see this kind of approach quite often -- by far too often -- over
> the past years (just think the many "uses" of SIP defined elsewhere,
> usually resulting in something "a little but not entirely unlike" SIP).
>
> What I would like avoid is having special treatment of SDP depending
> on what the media type is (or other attributes signify).
>
> Therefore, I strongly agree with Colin's conclusion: take SDP as is
> or leave it altogether and pursue a different path.
>

I am sensitive to the philosophical desire to keep things pure, but on 
the pragmatic side this seems to me to be a false dichotomy. We extend 
protocols all the time, for good or ill. Can you describe specific 
interoperability problems that would be introduced by the MSRP usage as 
currently described?

How about this as a potential compromise:

We keep the path attribute as is, but _also_ copy the information for 
the endpoint into the m and c lines. This would seem to me to be 
consistent with the usage in the current ICE draft, in which ICE aware 
endpoints use the candidate attributes to determine the address and 
port, but the default address and port are copied into the m and c line 
for backwards compatibility? Considering that the ICE need for 
backwards compatibility seems much more clear than the MSRP need for 
the same?


> Joerg
>
>
> Colin Perkins wrote:
>
>> On 10 Dec 2004, at 21:41, Ben Campbell wrote:
>>> I really need to get closure on this, so I'd appreciate people 
>>> agreeing or disagreeing.
>>>
>>> I have had off-list correspondence indicating both positions. In 
>>> summary:
>>>
>>> In favor of the change, I had a comment to the effect that changing 
>>> SDP semantics to ignore the C line and the port field on the M line 
>>> really changes it to something that is no longer SDP, and that such 
>>> deviations are in general bad for interop.
>>>
>>> In favor of leaving things as currently specified, I had two 
>>> comments that decomposing URLs into the C and M lines is ugly, and 
>>> that having a different mechanism to communicate endpoint addresses 
>>> vs relay addresses is really ugly.
>>>
>>> So, do people agree with one or the other position?
>> Decomposing the URL into the "c=" and "m=" lines is very ugly. 
>> Unfortunately, that ugliness is required if you want to use SDP. This 
>> is why the SDPng work was started, to try to move to something less 
>> horrible...
>> Colin
>> _______________________________________________
>> mmusic mailing list
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>


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From: Ben Campbell <ben@nostrum.com>
Subject: Re: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
Date: Fri, 17 Dec 2004 16:19:31 -0600
To: Hisham Khartabil <hisham.khartabil@telio.no>
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On Dec 13, 2004, at 3:10 AM, Hisham Khartabil wrote:

> (not as chair, but as an MSRP protocol contributor)
>
> The objection to using the SDP c and m lines is not purely a cosmetic 
> objection. We are not saying that SDP the way it is defined today is 
> ugly so we want to rearrange it to look better.
>
> We have a fundamental objection: having 2 ways to construct a next hop 
> URI that depends on the entity that is the next hop is not acceptable. 
> Endpoints should not be required  to have 2 engines implemented. 
> Unless we are given an alternative that allows endpoint implementers 
> to implement one piece of code for constructing an next hop MSRP URI, 
> I'm afraid we have no choice but to keep what we have today in MSRP as 
> is.
>

I think Hisham expressed my concern much more precisely than I 
previously expressed it. This is exactly what I was trying to say.


> You telling us to go find another protocol if we don't like SDP the 
> way it is today is hardly contructive. SDPng is not an option and we 
> call know it.
>
> Let me add that comedia also violates what you claim to be a 
> fundamental piece of the protocol that if broken will cause the SDP 
> protocol not to be SDP any longer. An argument that I don't buy. We 
> have violated things before:
>
>    - t= line does not mean anything in SIP and is set to 0 0 (which in 
> SDP means that the session is permanent and unbounded).
>    - s= line carries session name. It is set to - when used with SIP.
>
> Also, quoting the SDP draft:
>
> "For unicast IP sessions, the following are conveyed:
>
>    o  The remote address for media
>
>    o  The remote transport port for media
>
>    The semantics of this address and port depend on the media and
>    transport protocol defined.  By default, this SHOULD be the remote
>    address and remote port to which data is sent.  Some media types MAY
>    redefine this behaviour, but this is NOT RECOMMENDED."
>
> We are changing the semantics. As far as I understand from the quoted 
> paragraph above, we are not violating anything, even-though it is not 
> recommended.
>
> I have a question to SDP stack implementers that plan to implement 
> MSRP: Does the current way MSRP specifies the use of SDP (ignore the c 
> line address and the m line port and instead use the attribute a=path) 
> cause problems to you and requires you to do some hacking in your SDP 
> stack? If so, is that acceptable or do you prefer for your MSRP 
> implementation to implement two ways of contructing an MSRP URL 
> depending on the what the next hop is?
>
> Regards,
> Hisham
>
> On Dec 12, 2004, at 12:58 PM, Joerg Ott wrote:
>
>> I agree with Colin on the ugliness issue of decomposing URLs into
>> c= and m= lines.  But this is the way SDP has worked and works today.
>>
>> The present discussion we are having here is about changing SDP to
>> for address/URL handling in one particular case.  This does not
>> strike me as something MSRP should be concerned with.
>>
>> I would like to make one thing clear: we are either using existing 
>> tools
>> in another document or we are building new ones.  There is nothing 
>> much
>> in between.
>>
>> Changing c=/m= semantics means changing SDP.  Just because the syntax
>> still looks the same does not mean that it is still the same protocol.
>> Any entity or generic protocol engine used to dealing with SDP m=/c=
>> would no longer do the right thing.
>>
>> We see this kind of approach quite often -- by far too often -- over
>> the past years (just think the many "uses" of SIP defined elsewhere,
>> usually resulting in something "a little but not entirely unlike" 
>> SIP).
>>
>> What I would like avoid is having special treatment of SDP depending
>> on what the media type is (or other attributes signify).
>>
>> Therefore, I strongly agree with Colin's conclusion: take SDP as is
>> or leave it altogether and pursue a different path.
>>
>> Joerg
>>
>>
>> Colin Perkins wrote:
>>
>>> On 10 Dec 2004, at 21:41, Ben Campbell wrote:
>>>> I really need to get closure on this, so I'd appreciate people 
>>>> agreeing or disagreeing.
>>>>
>>>> I have had off-list correspondence indicating both positions. In 
>>>> summary:
>>>>
>>>> In favor of the change, I had a comment to the effect that changing 
>>>> SDP semantics to ignore the C line and the port field on the M line 
>>>> really changes it to something that is no longer SDP, and that such 
>>>> deviations are in general bad for interop.
>>>>
>>>> In favor of leaving things as currently specified, I had two 
>>>> comments that decomposing URLs into the C and M lines is ugly, and 
>>>> that having a different mechanism to communicate endpoint addresses 
>>>> vs relay addresses is really ugly.
>>>>
>>>> So, do people agree with one or the other position?
>>> Decomposing the URL into the "c=" and "m=" lines is very ugly. 
>>> Unfortunately, that ugliness is required if you want to use SDP. 
>>> This is why the SDPng work was started, to try to move to something 
>>> less horrible...
>>> Colin
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>>
>
>
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
>


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Peter Ridler wrote:
> I also agree with Colin and Joerg.
> 
> I will also have two SDP parse engines (one for RTP and another for MSRP).

Do you currently support anything but RTP?
How much do you expect to share between the two in the best case?

I don't think you are really talking about the parser - that isn't what 
is different. I think you are talking about implementing the connection 
semantics represented by the SDP. No matter what you do, these are 
different for RTP and MSRP.

The SDP parser that I have worked with is quite capable of handling this 
either way. It is only involved with the *syntax* of SDP.

Note that the syntax of SDP is already quite a mallable thing. Much of 
the important syntax is hidden in a-lines. MSRP has its own a-lines, and 
these need to be parsed in any case.

> This will complicate the code.  Computers do not care about ugly formatting
> --- overall performance should be the concern (I think all text formatted
> protocols are a waste of network bandwidth and memory resources -- but I'm
> stuck with them). 
> 
> We should follow the standards as best we can.  If the standard does not
> meet your requirements then define/use another.

Can you suggest what you have in mind?

Its not like this is an independent decision for MSRP. MSRP is intended 
to be negotiated via SIP. And it isn't a matter of mutual exclusion - 
negotiate MSRP with sip and something other than SDP, negotiate RPT with 
sip and SDP - I expect to be negotiating *both* MSRP and RPT with the 
same sip offer/answer exchange.

It seems you must have SDPng in mind. So you are suggesting we can't 
finish MSRP until the migration to SDPng is complete?

> Another text based protocol is draft-ietf-sipping-toip-00.txt  This fits the
> SIP/SDP model nicely.

It fits because it runs over RTP. It doesn't meet our requirements.

It feels like you are trying to make the SDP tail wag the SIP dog.

	Paul

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This is to announce Working Group Last Call on the revised
SDP specification "SDP: Session Description Protocol"
recently published as

     draft-ietf-mmusic-sdp-new-23.txt

The document shall be advanced to Proposed Standard (PS).

The WGLC is to end on 9 January 2005.

Please review the latest draft carefully and direct all comments
to the MMUSIC list and the authors.

Joerg



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Date: Sun, 19 Dec 2004 14:38:22 -0800
Subject: Re: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
From: Cullen Jennings <fluffy@cisco.com>
To: Peter Ridler <ridler@softrac.ca>, "'Joerg Ott'" <jo@tzi.uni-bremen.de>,
        "'Colin Perkins'" <csp@csperkins.org>
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I think that the MSRP folks have not clearly explained what we want to do in
SDP. We want to set up a path of multiple addresses, not just a single
address for one far end. That the protocol uses TCP or that it transports
text instead of audio I don't think should cause it to be much different.


On 12/18/04 7:16 AM, "Peter Ridler" <ridler@softrac.ca> wrote:

> 
> I also agree with Colin and Joerg.
> 
> I will also have two SDP parse engines (one for RTP and another for MSRP).
> This will complicate the code.  Computers do not care about ugly formatting
> --- overall performance should be the concern (I think all text formatted
> protocols are a waste of network bandwidth and memory resources -- but I'm
> stuck with them).
> 
> We should follow the standards as best we can.  If the standard does not
> meet your requirements then define/use another.
> 
> Another text based protocol is draft-ietf-sipping-toip-00.txt  This fits the
> SIP/SDP model nicely.
> 
> Peter
>  
> 
>> -----Original Message-----
>> From: simple-bounces@ietf.org
>> [mailto:simple-bounces@ietf.org] On Behalf Of Joerg Ott
>> Sent: Sunday, December 12, 2004 6:58 AM
>> To: Colin Perkins
>> Cc: Simple WG; mmusic@ietf.org
>> Subject: Re: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
>> 
>> I agree with Colin on the ugliness issue of decomposing URLs
>> into c= and m= lines.  But this is the way SDP has worked and
>> works today.
>> 
>> The present discussion we are having here is about changing
>> SDP to for address/URL handling in one particular case.  This
>> does not strike me as something MSRP should be concerned with.
>> 
>> I would like to make one thing clear: we are either using
>> existing tools in another document or we are building new
>> ones.  There is nothing much in between.
>> 
>> Changing c=/m= semantics means changing SDP.  Just because
>> the syntax still looks the same does not mean that it is
>> still the same protocol.
>> Any entity or generic protocol engine used to dealing with
>> SDP m=/c= would no longer do the right thing.
>> 
>> We see this kind of approach quite often -- by far too often
>> -- over the past years (just think the many "uses" of SIP
>> defined elsewhere, usually resulting in something "a little
>> but not entirely unlike" SIP).
>> 
>> What I would like avoid is having special treatment of SDP
>> depending on what the media type is (or other attributes signify).
>> 
>> Therefore, I strongly agree with Colin's conclusion: take SDP
>> as is or leave it altogether and pursue a different path.
>> 
>> Joerg
>> 
>> 
>> Colin Perkins wrote:
>> 
>>> On 10 Dec 2004, at 21:41, Ben Campbell wrote:
>>> 
>>>> I really need to get closure on this, so I'd appreciate people
>>>> agreeing or disagreeing.
>>>> 
>>>> I have had off-list correspondence indicating both
>> positions. In summary:
>>>> 
>>>> In favor of the change, I had a comment to the effect that
>> changing 
>>>> SDP semantics to ignore the C line and the port field on
>> the M line 
>>>> really changes it to something that is no longer SDP, and
>> that such 
>>>> deviations are in general bad for interop.
>>>> 
>>>> In favor of leaving things as currently specified, I had
>> two comments 
>>>> that decomposing URLs into the C and M lines is ugly, and
>> that having 
>>>> a different mechanism to communicate endpoint addresses vs relay
>>>> addresses is really ugly.
>>>> 
>>>> So, do people agree with one or the other position?
>>> 
>>> 
>>> Decomposing the URL into the "c=" and "m=" lines is very ugly.
>>> Unfortunately, that ugliness is required if you want to use
>> SDP. This 
>>> is why the SDPng work was started, to try to move to something less
>>> horrible...
>>> 
>>> Colin
>>> 
>>> 
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>> 
>> 
>> 
>> 
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple
>> 
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple


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Subject: RE: [MMUSIC] WGLC: draft-ietf-mmusic-sdp-new-23.txt
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I would like to request a clarification in section 5.14 of 
draft-ietf-mmusic-sdp-new-23.

As we discovered in BEHAVE, certain SDP implementations are misinterpretting
RFC 3550 section 11 2nd paragraph. In a SIP Offer/Answer for example, these
implementations unilaterally change an odd RTP port indicated in an SDP
(from the 
other end) with to the even port that one lower. In other words, if the
other end
indicates an RTP port of 5201, these implemenations will send RTP to 5200,
which will
be a big problem.

Note that this issue is slightly orthogonal to the use of the rtcp attribute
of
RFC 3605. For example, if an application discovers (with STUN or ICE) that
its
RTP port is 5201, it may then discover that it's RTCP port is 5202 (which
would
satisfy the contiguity rule). This is very likely to happen because NATs
typically
assign ports sequentially. An application could (somewhat foolishlingly) not
use RFC 3605 in this case because it follows the contiguity rule (but not
the parity 
rule).

Another issue is for implementations that do not support RTCP in the first
place.

In order to avoid these problems, I propose that we explicitly state that 
the <port> for RTP indicates the actual port and MUST not be changed
regardless of 
parity (i.e., DO NOT substract 1 if it is odd).

I believe that would finally put to rest this parity issue for all SDP-based
protocols.

Also, on a similar note, would it make sense to incorporate RFC 3605 (the
directly into
sdp-new, instead of having it stand as a separate RFC) and make its support
mandatory?
The biggest problem with RFC 3605 is that I can do the best SIP/SDP
implementation possible, support STUN, ICE, 3605 and all, but if the other
end doesn't
support 3605 (because they don't even know of NATs), the problem above will
still occur.


> -----Original Message-----
> From: mmusic-bounces@ietf.org 
> [mailto:mmusic-bounces@ietf.org] On Behalf Of Joerg Ott
> Sent: Sunday, December 19, 2004 08:51
> To: mmusic@ietf.org
> Cc: Colin Perkins
> Subject: [MMUSIC] WGLC: draft-ietf-mmusic-sdp-new-23.txt
> 
> 
> This is to announce Working Group Last Call on the revised
> SDP specification "SDP: Session Description Protocol"
> recently published as
> 
>      draft-ietf-mmusic-sdp-new-23.txt
> 
> The document shall be advanced to Proposed Standard (PS).
> 
> The WGLC is to end on 9 January 2005.
> 
> Please review the latest draft carefully and direct all 
> comments to the MMUSIC list and the authors.
> 
> Joerg
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 
> 

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<P><FONT SIZE=2>I would like to request a clarification in section 5.14 of </FONT>
<BR><FONT SIZE=2>draft-ietf-mmusic-sdp-new-23.</FONT>
</P>

<P><FONT SIZE=2>As we discovered in BEHAVE, certain SDP implementations are misinterpretting</FONT>
<BR><FONT SIZE=2>RFC 3550 section 11 2nd paragraph. In a SIP Offer/Answer for example, these</FONT>
<BR><FONT SIZE=2>implementations unilaterally change an odd RTP port indicated in an SDP (from the </FONT>
<BR><FONT SIZE=2>other end) with to the even port that one lower. In other words, if the other end</FONT>
<BR><FONT SIZE=2>indicates an RTP port of 5201, these implemenations will send RTP to 5200, which will</FONT>
<BR><FONT SIZE=2>be a big problem.</FONT>
</P>

<P><FONT SIZE=2>Note that this issue is slightly orthogonal to the use of the rtcp attribute of</FONT>
<BR><FONT SIZE=2>RFC 3605. For example, if an application discovers (with STUN or ICE) that its</FONT>
<BR><FONT SIZE=2>RTP port is 5201, it may then discover that it's RTCP port is 5202 (which would</FONT>
<BR><FONT SIZE=2>satisfy the contiguity rule). This is very likely to happen because NATs typically</FONT>
<BR><FONT SIZE=2>assign ports sequentially. An application could (somewhat foolishlingly) not</FONT>
<BR><FONT SIZE=2>use RFC 3605 in this case because it follows the contiguity rule (but not the parity </FONT>
<BR><FONT SIZE=2>rule).</FONT>
</P>

<P><FONT SIZE=2>Another issue is for implementations that do not support RTCP in the first place.</FONT>
</P>

<P><FONT SIZE=2>In order to avoid these problems, I propose that we explicitly state that </FONT>
<BR><FONT SIZE=2>the &lt;port&gt; for RTP indicates the actual port and MUST not be changed regardless of </FONT>
<BR><FONT SIZE=2>parity (i.e., DO NOT substract 1 if it is odd).</FONT>
</P>

<P><FONT SIZE=2>I believe that would finally put to rest this parity issue for all SDP-based</FONT>
<BR><FONT SIZE=2>protocols.</FONT>
</P>

<P><FONT SIZE=2>Also, on a similar note, would it make sense to incorporate RFC 3605 (the&nbsp; directly into</FONT>
<BR><FONT SIZE=2>sdp-new, instead of having it stand as a separate RFC) and make its support mandatory?</FONT>
<BR><FONT SIZE=2>The biggest problem with RFC 3605 is that I can do the best SIP/SDP</FONT>
<BR><FONT SIZE=2>implementation possible, support STUN, ICE, 3605 and all, but if the other end doesn't</FONT>
<BR><FONT SIZE=2>support 3605 (because they don't even know of NATs), the problem above will still occur.</FONT>
</P>
<BR>

<P><FONT SIZE=2>&gt; -----Original Message-----</FONT>
<BR><FONT SIZE=2>&gt; From: mmusic-bounces@ietf.org </FONT>
<BR><FONT SIZE=2>&gt; [<A HREF="mailto:mmusic-bounces@ietf.org">mailto:mmusic-bounces@ietf.org</A>] On Behalf Of Joerg Ott</FONT>
<BR><FONT SIZE=2>&gt; Sent: Sunday, December 19, 2004 08:51</FONT>
<BR><FONT SIZE=2>&gt; To: mmusic@ietf.org</FONT>
<BR><FONT SIZE=2>&gt; Cc: Colin Perkins</FONT>
<BR><FONT SIZE=2>&gt; Subject: [MMUSIC] WGLC: draft-ietf-mmusic-sdp-new-23.txt</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; This is to announce Working Group Last Call on the revised</FONT>
<BR><FONT SIZE=2>&gt; SDP specification &quot;SDP: Session Description Protocol&quot;</FONT>
<BR><FONT SIZE=2>&gt; recently published as</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; draft-ietf-mmusic-sdp-new-23.txt</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; The document shall be advanced to Proposed Standard (PS).</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; The WGLC is to end on 9 January 2005.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Please review the latest draft carefully and direct all </FONT>
<BR><FONT SIZE=2>&gt; comments to the MMUSIC list and the authors.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Joerg</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; _______________________________________________</FONT>
<BR><FONT SIZE=2>&gt; mmusic mailing list</FONT>
<BR><FONT SIZE=2>&gt; mmusic@ietf.org</FONT>
<BR><FONT SIZE=2>&gt; <A HREF="https://www1.ietf.org/mailman/listinfo/mmusic" TARGET="_blank">https://www1.ietf.org/mailman/listinfo/mmusic</A></FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; </FONT>
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From: Henning Schulzrinne <hgs@cs.columbia.edu>
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Subject: [MMUSIC] Maximum non-scaled image size
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Do any video payload formats specify, as SDP format attributes or 
otherwise, the maximum image size supported natively (i.e., without 
downscaling)?

For a draft we're working on, I'd like to re-use any existing 
conventions. In general, it would seem useful to know this since there 
isn't much point in sending at a higher resolution than the screen 
resolution or the maximum size that an application can handle. This 
seems like a natural extension to a=framerate in SDP.

Henning


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Henning,
In the bis drafts for the H.263 and H.261 I added the support for
specifying the image size that the decoder can receive per frame rate.
For H.261 the support is for fixed resolutions for H.263 the support is
also for custom formats. See draft-ietf-avt-rfc2429-bis-03.txt and
draft-ietf-avt-rfc2032-bis-03.txt
In the H.264, see draft-ietf-avt-rtp-h264-11, the concept was that the
decoder, using the level parameter, specifies the number of macro blocks
per second it can process; The level as specified in H.264 gives a
maximum on the frame rate and resolution. The level parameters can be
overridden by the optional parameters specified in the draft.

The transformation from the decoder to the display is up to the
implementer. =20

As can be seen in H.261 and H.263 the SDP from the receiver can specify
an exact picture size(resolution)it can receive, but still the encoder
can send any allowed picture size that is bellow the decoder capability.
For example in H.261 if the decoder specifies it can receive a CIF
picture at 30 fps, the encoder can send a QCIF picture at 15 fps.=20
In H.264 there is no explicit resolution specified.

Roni Even



-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
Of Henning Schulzrinne
Sent: Monday, December 20, 2004 2:52 AM
To: avt@ietf.org; mmusic@ietf.org
Subject: [MMUSIC] Maximum non-scaled image size

Do any video payload formats specify, as SDP format attributes or=20
otherwise, the maximum image size supported natively (i.e., without=20
downscaling)?

For a draft we're working on, I'd like to re-use any existing=20
conventions. In general, it would seem useful to know this since there=20
isn't much point in sending at a higher resolution than the screen=20
resolution or the maximum size that an application can handle. This=20
seems like a natural extension to a=3Dframerate in SDP.

Henning


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

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Thread-Topic: [MMUSIC] Maximum non-scaled image size
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From: "Even, Roni" <roni.even@polycom.co.il>
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Henning,
In the bis drafts for the H.263 and H.261 I added the support for
specifying the image size that the decoder can receive per frame rate.
For H.261 the support is for fixed resolutions for H.263 the support is
also for custom formats. See draft-ietf-avt-rfc2429-bis-03.txt and
draft-ietf-avt-rfc2032-bis-03.txt
In the H.264, see draft-ietf-avt-rtp-h264-11, the concept was that the
decoder, using the level parameter, specifies the number of macro blocks
per second it can process; The level as specified in H.264 gives a
maximum on the frame rate and resolution. The level parameters can be
overridden by the optional parameters specified in the draft.

The transformation from the decoder to the display is up to the
implementer. =20

As can be seen in H.261 and H.263 the SDP from the receiver can specify
an exact picture size(resolution)it can receive, but still the encoder
can send any allowed picture size that is bellow the decoder capability.
For example in H.261 if the decoder specifies it can receive a CIF
picture at 30 fps, the encoder can send a QCIF picture at 15 fps.=20
In H.264 there is no explicit resolution specified.

Roni Even



-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
Of Henning Schulzrinne
Sent: Monday, December 20, 2004 2:52 AM
To: avt@ietf.org; mmusic@ietf.org
Subject: [MMUSIC] Maximum non-scaled image size

Do any video payload formats specify, as SDP format attributes or=20
otherwise, the maximum image size supported natively (i.e., without=20
downscaling)?

For a draft we're working on, I'd like to re-use any existing=20
conventions. In general, it would seem useful to know this since there=20
isn't much point in sending at a higher resolution than the screen=20
resolution or the maximum size that an application can handle. This=20
seems like a natural extension to a=3Dframerate in SDP.

Henning


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

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

_______________________________________________
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Henning,
In the bis drafts for the H.263 and H.261 I added the support for
specifying the image size that the decoder can receive per frame rate.
For H.261 the support is for fixed resolutions for H.263 the support is
also for custom formats. See draft-ietf-avt-rfc2429-bis-03.txt and
draft-ietf-avt-rfc2032-bis-03.txt
In the H.264, see draft-ietf-avt-rtp-h264-11, the concept was that the
decoder, using the level parameter, specifies the number of macro blocks
per second it can process; The level as specified in H.264 gives a
maximum on the frame rate and resolution. The level parameters can be
overridden by the optional parameters specified in the draft.

The transformation from the decoder to the display is up to the
implementer. =20

As can be seen in H.261 and H.263 the SDP from the receiver can specify
an exact picture size(resolution)it can receive, but still the encoder
can send any allowed picture size that is bellow the decoder capability.
For example in H.261 if the decoder specifies it can receive a CIF
picture at 30 fps, the encoder can send a QCIF picture at 15 fps.=20
In H.264 there is no explicit resolution specified.

Roni Even



-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
Of Henning Schulzrinne
Sent: Monday, December 20, 2004 2:52 AM
To: avt@ietf.org; mmusic@ietf.org
Subject: [MMUSIC] Maximum non-scaled image size

Do any video payload formats specify, as SDP format attributes or=20
otherwise, the maximum image size supported natively (i.e., without=20
downscaling)?

For a draft we're working on, I'd like to re-use any existing=20
conventions. In general, it would seem useful to know this since there=20
isn't much point in sending at a higher resolution than the screen=20
resolution or the maximum size that an application can handle. This=20
seems like a natural extension to a=3Dframerate in SDP.

Henning


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

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

_______________________________________________
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Henning Schulzrinne wrote:
 >Do any video payload formats specify, as SDP format attributes or
 >otherwise, the maximum image size supported natively (i.e., without
 >downscaling)?
-------------------------------------------------------------------------
Dear All,

The ITU-T standards (e.g. H.261, H.263, H.264, etc.) explicitly specify 
image sizes for the codecs. H.261 and H.263 specify image sizes 
according to ITU CIF standard in form of (height x width) resolution. 
H.264 also specifies max image sizes per used profile level in form of 
max bits per image, but without specifying aspect ratios. From the MPEG 
standards only MPEG-1 specifies max image size of 352x240 height x width 
resolution. All the other Video MPEGs are with arbitrary image sizes.  
Another possibility to specify image sizes are the VESA standards from 
www.vesa.org (e.g. VESA's VGA BIOS Extensions).  
Note, that some standards for video specify frame-rate, frame-size and 
colour-quality dependencies. Hence, using some standard definitions 
might influence all of these 3 parameters, i.e. the colour-quality 
(number of colours per image) might also influence the amount of network 
data for the video streams.
Note, that frame-size can be specified in 3 different manners:
- (height x width) resolution in (pixel x pixel)
- standard - i.e. CIF or VESA
- max number of bits + aspect ratio - e.g. commonly used aspect ratios 
are height: width = 4:3, 5:4, 11:9, 16:9, 16:10. These aspect ratios 
stem also from CIF and VESA
For the complete and unique video performance specification with SDP in 
case of arbitrary video image specification one needs to have 
specifications for frame-rate, frame-size and colour-quality.
Best Regards,
    Teodora Guenkova-Luy



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From: "Peter Ridler" <ridler@softrac.ca>
To: "'Joerg Ott'" <jo@tzi.uni-bremen.de>,
        "'Colin Perkins'" <csp@csperkins.org>
Subject: RE: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
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I also agree with Colin and Joerg.

I will also have two SDP parse engines (one for RTP and another for MSRP).
This will complicate the code.  Computers do not care about ugly formatting
--- overall performance should be the concern (I think all text formatted
protocols are a waste of network bandwidth and memory resources -- but I'm
stuck with them). 

We should follow the standards as best we can.  If the standard does not
meet your requirements then define/use another.

Another text based protocol is draft-ietf-sipping-toip-00.txt  This fits the
SIP/SDP model nicely.

Peter
 

> -----Original Message-----
> From: simple-bounces@ietf.org 
> [mailto:simple-bounces@ietf.org] On Behalf Of Joerg Ott
> Sent: Sunday, December 12, 2004 6:58 AM
> To: Colin Perkins
> Cc: Simple WG; mmusic@ietf.org
> Subject: Re: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
> 
> I agree with Colin on the ugliness issue of decomposing URLs 
> into c= and m= lines.  But this is the way SDP has worked and 
> works today.
> 
> The present discussion we are having here is about changing 
> SDP to for address/URL handling in one particular case.  This 
> does not strike me as something MSRP should be concerned with.
> 
> I would like to make one thing clear: we are either using 
> existing tools in another document or we are building new 
> ones.  There is nothing much in between.
> 
> Changing c=/m= semantics means changing SDP.  Just because 
> the syntax still looks the same does not mean that it is 
> still the same protocol.
> Any entity or generic protocol engine used to dealing with 
> SDP m=/c= would no longer do the right thing.
> 
> We see this kind of approach quite often -- by far too often 
> -- over the past years (just think the many "uses" of SIP 
> defined elsewhere, usually resulting in something "a little 
> but not entirely unlike" SIP).
> 
> What I would like avoid is having special treatment of SDP 
> depending on what the media type is (or other attributes signify).
> 
> Therefore, I strongly agree with Colin's conclusion: take SDP 
> as is or leave it altogether and pursue a different path.
> 
> Joerg
> 
> 
> Colin Perkins wrote:
> 
> > On 10 Dec 2004, at 21:41, Ben Campbell wrote:
> > 
> >> I really need to get closure on this, so I'd appreciate people 
> >> agreeing or disagreeing.
> >>
> >> I have had off-list correspondence indicating both 
> positions. In summary:
> >>
> >> In favor of the change, I had a comment to the effect that 
> changing 
> >> SDP semantics to ignore the C line and the port field on 
> the M line 
> >> really changes it to something that is no longer SDP, and 
> that such 
> >> deviations are in general bad for interop.
> >>
> >> In favor of leaving things as currently specified, I had 
> two comments 
> >> that decomposing URLs into the C and M lines is ugly, and 
> that having 
> >> a different mechanism to communicate endpoint addresses vs relay 
> >> addresses is really ugly.
> >>
> >> So, do people agree with one or the other position?
> > 
> > 
> > Decomposing the URL into the "c=" and "m=" lines is very ugly. 
> > Unfortunately, that ugliness is required if you want to use 
> SDP. This 
> > is why the SDPng work was started, to try to move to something less 
> > horrible...
> > 
> > Colin
> > 
> > 
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> > 
> 
> 
> 
> _______________________________________________
> Simple mailing list
> Simple@ietf.org
> https://www1.ietf.org/mailman/listinfo/simple
> 


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Thread-Topic: [AVT] Maximum non-scaled image size
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From: "Gary Sullivan" <garysull@windows.microsoft.com>
To: "Henning Schulzrinne" <hgs@cs.columbia.edu>, <avt@ietf.org>,
        <mmusic@ietf.org>
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Subject: [MMUSIC] RE: [AVT] Maximum non-scaled image size
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There are ways to declare an image size as part of a decoder capability
or a decoder preference indication in both H.245 and SDP (and perhaps
other places too, most likely including H.241 and what H.320 uses) -- I
would suggest checking the design in regards to H.263 as that has
probably been the most commonly used video codec in two-way
communication scenarios.

When specified as part of a declaration of decoder capability, an image
size would ordinarily be interpreted as the maximum that the decoder
could decode, rather than as the maximum it can display.  But if
specified as a decoder preference, its interpretation would be more akin
to what you mention.  That would be a declaration of what image size the
decoder would most like to receive, although perhaps not necessarily an
indication of exactly why.  In some sense, that might be a better thing
to design than what you seek.  (Why pick only one specific reason for
the decoder to indicate its preference for something?)

A decoder might also, of course, use censorship of its decoder
capability declaration to simply not declare capabilities beyond its
native display resolutions.

>From draft-ietf-avt-rfc2429-bis-03.txt (September 2004, section 8.1),
there is a way in SDP for an H.263 decoder to indicate an ordered list
of preferences for picture resolution and frame rate.  (That stuff is
also probably in the original RFC 2429 as well -- I just happen to have
checked the "bis" draft).

Best Regards,

Gary Sullivan


+> -----Original Message-----
+> From: avt-bounces@ietf.org [mailto:avt-bounces@ietf.org] On=20
+> Behalf Of Henning Schulzrinne
+> Sent: Sunday, December 19, 2004 4:52 PM
+> To: avt@ietf.org; mmusic@ietf.org
+> Subject: [AVT] Maximum non-scaled image size
+>=20
+> Do any video payload formats specify, as SDP format attributes or=20
+> otherwise, the maximum image size supported natively (i.e., without=20
+> downscaling)?
+>=20
+> For a draft we're working on, I'd like to re-use any existing=20
+> conventions. In general, it would seem useful to know this=20
+> since there=20
+> isn't much point in sending at a higher resolution than the screen=20
+> resolution or the maximum size that an application can handle. This=20
+> seems like a natural extension to a=3Dframerate in SDP.
+>=20
+> Henning
+>=20
+>=20
+> _______________________________________________
+> Audio/Video Transport Working Group
+> avt@ietf.org
+> https://www1.ietf.org/mailman/listinfo/avt
+>=20

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Date: Mon, 20 Dec 2004 19:51:28 +0100
From: Teodora Guenkova-Luy <guenkova@vs.informatik.uni-ulm.de>
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To: Henning Schulzrinne <hgs@cs.columbia.edu>, mmusic@ietf.org
Subject: Re: [MMUSIC] Maximum non-scaled image size
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Dear Henning Schulzrinne,
For some codecs the framerate is also a fix parameter of the codec=20
definition (e.g. the ITU-T H.26x codecs profiles and levels) and
for other codecs (e.g. MPEG-4) it is a generic parameter, which should=20
be defined using descriptions of the media like in SDP.
The framerate is the parameter of biggest influence for the network=20
traffic and IMHO that is why nobody has considered the framesize
and the colour quality as generic parameter in SDP up to now. But in=20
order to specify performance for codecs with arbitrary definitions
of framesize and the colour quality, you need their media definition in=20
SDP (or even better in XML format in SDPng) as well.

With respect to framesize, I would rather put the framesize information=20
as one line and would rather specify the following for SDP:

framesize-type =3D "a=3Dframesize:" sizetype SP width SP height
sizetype =3D ciftype | vesatype | "arbitrary"

Where the "ciftype" can be described considering the inputs of the=20
following table:
CIF Format        Resolution in pixels (width x height)
SQCIF               128 =D7 96
QCIF                 176 =D7 144
CIF                    352 =D7 288
4CIF                  704 =D7 576
9CIF                  1056 =D7 864
16CIF                1408 =D7 1152
TV                     720 =D7 576
SIF                    352 =D7 240
SIF (NTSC)      360 =D7 240
SIF (PAL)         360 =D7 288

and "vesatype" (AR is Aspect Ratio):

VESA Format    width    height    AR-16:10    AR-16:9    AR-4:3   =20
non-square pixel
QUXGA-Wide    3820    2400    x          =20
WUXGA            1920    1200    x          =20
WSXGA+            1680    1050    x          =20
WXGA+
 =20
WXGA                1440    900    x          =20
                            1280    854    x                         =20
                              3:2
                            1280    800    x          =20
                            1280    768    x                          =20
                             5:3
1080i/1080p        1920    1080                        x      =20
720i/720p            1280    720                          x      =20
QUXGA              3200    2400                                        x =
 =20
QXGA                 2048    1536                                       =20
x  =20
UXGA                1600    1200                                        =20
x  =20
SXGA+              1400    1050                                        =20
x  =20
SXGA                1280    1024                                        =20
x                    5:4
XGA                   1024    768                                       =20
  x  =20
SVGA                 800    600                                       =20
    x  =20
VGA                   640    480                                       =20
    x  =20
QVGA                320    240                                       =20
    x  =20


If specifying a "ciftype" or "vesatype" the SDP "width" and "height"=20
parameters could be optional, e.g.
a=3Dframesize:4CIF

If specifying an "arbitrary" framesize then the  "width" and "height"=20
parameters are mandatory and shall be expressed in pixel, e.g.
a=3Dframesize:arbitrary 300 200

Hope this helps,
Teodora Guenkova-Luy


Henning Schulzrinne wrote:

> Thanks for all the explanation. It seems a bit asymmetric that=20
> framerate is specified as a generic a=3D parameter in SDP, while=20
> resolution is specified, if at all, on a per-codec basis. Would it=20
> make sense to have an 'a=3Dwidth' and 'a=3Dheight' parameter as well?
>
> Teodora Guenkova-Luy wrote:
>
>> Henning Schulzrinne wrote:
>>  >Do any video payload formats specify, as SDP format attributes or
>>  >otherwise, the maximum image size supported natively (i.e., without
>>  >downscaling)?
>> ----------------------------------------------------------------------=
---=20
>>
>> Dear All,
>>
>> The ITU-T standards (e.g. H.261, H.263, H.264, etc.) explicitly=20
>> specify image sizes for the codecs. H.261 and H.263 specify image=20
>> sizes according to ITU CIF standard in form of (height x width)=20
>> resolution. H.264 also specifies max image sizes per used profile=20
>> level in form of max bits per image, but without specifying aspect=20
>> ratios. From the MPEG standards only MPEG-1 specifies max image size=20
>> of 352x240 height x width resolution. All the other Video MPEGs are=20
>> with arbitrary image sizes.  Another possibility to specify image=20
>> sizes are the VESA standards from www.vesa.org (e.g. VESA's VGA BIOS=20
>> Extensions).  Note, that some standards for video specify frame-rate,=20
>> frame-size and colour-quality dependencies. Hence, using some=20
>> standard definitions might influence all of these 3 parameters, i.e.=20
>> the colour-quality (number of colours per image) might also influence=20
>> the amount of network data for the video streams.
>> Note, that frame-size can be specified in 3 different manners:
>> - (height x width) resolution in (pixel x pixel)
>> - standard - i.e. CIF or VESA
>> - max number of bits + aspect ratio - e.g. commonly used aspect=20
>> ratios are height: width =3D 4:3, 5:4, 11:9, 16:9, 16:10. These aspect=
=20
>> ratios stem also from CIF and VESA
>> For the complete and unique video performance specification with SDP=20
>> in case of arbitrary video image specification one needs to have=20
>> specifications for frame-rate, frame-size and colour-quality.
>> Best Regards,
>>    Teodora Guenkova-Luy
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>
>
>
>

--=20
-----------------------------------------
Teodora Guenkova-Luy

Dept. Distributed Systems, University of Ulm,=20
Oberer Eselsberg, 89069 Ulm, Germany

Tel.: +49 (0731) 502-4148=20
FAX: +49 (0731) 502-4142

e-Mail: guenkova@vs.informatik.uni-ulm.de

Home Page: http://www-vs.informatik.uni-ulm.de/~Guenkova/=20

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



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Subject: RE: [MMUSIC] Maximum non-scaled image size
Date: Tue, 21 Dec 2004 16:00:21 +0200
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Thread-Topic: [MMUSIC] Maximum non-scaled image size
Thread-Index: AcTmxuZJ8tgEe2opSGKwZDuKSjYeHgAnZmdA
From: "Even, Roni" <roni.even@polycom.co.il>
To: "Teodora Guenkova-Luy" <guenkova@vs.informatik.uni-ulm.de>,
        "Henning Schulzrinne" <hgs@cs.columbia.edu>, <mmusic@ietf.org>
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Hi,
Just a small correction.
The frame rate is not a fixed parameter, for H.261 there are no profile =
and level. For H.263++ you can use profile and levels but the levels =
gives maximum on frame rate and on resolution. That is why we are adding =
in draft-ietf-avt-rfc2032-bis and draft-ietf-avt-rfc2429-bis the SDP =
parameters to specify the resolutions and the frame rates that can be =
used. The specific video codecs (H.261 and H.263) has a list of defined =
resolutions they support and not all the list specified. H.263 can =
support custom formats.

Roni Even=20

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf =
Of Teodora Guenkova-Luy
Sent: Monday, December 20, 2004 8:51 PM
To: Henning Schulzrinne; mmusic@ietf.org
Subject: Re: [MMUSIC] Maximum non-scaled image size

Dear Henning Schulzrinne,
For some codecs the framerate is also a fix parameter of the codec=20
definition (e.g. the ITU-T H.26x codecs profiles and levels) and
for other codecs (e.g. MPEG-4) it is a generic parameter, which should=20
be defined using descriptions of the media like in SDP.
The framerate is the parameter of biggest influence for the network=20
traffic and IMHO that is why nobody has considered the framesize
and the colour quality as generic parameter in SDP up to now. But in=20
order to specify performance for codecs with arbitrary definitions
of framesize and the colour quality, you need their media definition in=20
SDP (or even better in XML format in SDPng) as well.

With respect to framesize, I would rather put the framesize information=20
as one line and would rather specify the following for SDP:

framesize-type =3D "a=3Dframesize:" sizetype SP width SP height
sizetype =3D ciftype | vesatype | "arbitrary"

Where the "ciftype" can be described considering the inputs of the=20
following table:
CIF Format        Resolution in pixels (width x height)
SQCIF               128 =AA 96
QCIF                 176 =AA 144
CIF                    352 =AA 288
4CIF                  704 =AA 576
9CIF                  1056 =AA 864
16CIF                1408 =AA 1152
TV                     720 =AA 576
SIF                    352 =AA 240
SIF (NTSC)      360 =AA 240
SIF (PAL)         360 =AA 288

and "vesatype" (AR is Aspect Ratio):

VESA Format    width    height    AR-16:10    AR-16:9    AR-4:3   =20
non-square pixel
QUXGA-Wide    3820    2400    x          =20
WUXGA            1920    1200    x          =20
WSXGA+            1680    1050    x          =20
WXGA+
 =20
WXGA                1440    900    x          =20
                            1280    854    x                         =20
                              3:2
                            1280    800    x          =20
                            1280    768    x                          =20
                             5:3
1080i/1080p        1920    1080                        x      =20
720i/720p            1280    720                          x      =20
QUXGA              3200    2400                                        x =
 =20
QXGA                 2048    1536                                        =

x  =20
UXGA                1600    1200                                         =

x  =20
SXGA+              1400    1050                                        =20
x  =20
SXGA                1280    1024                                         =

x                    5:4
XGA                   1024    768                                        =

  x  =20
SVGA                 800    600                                       =20
    x  =20
VGA                   640    480                                       =20
    x  =20
QVGA                320    240                                       =20
    x  =20


If specifying a "ciftype" or "vesatype" the SDP "width" and "height"=20
parameters could be optional, e.g.
a=3Dframesize:4CIF

If specifying an "arbitrary" framesize then the  "width" and "height"=20
parameters are mandatory and shall be expressed in pixel, e.g.
a=3Dframesize:arbitrary 300 200

Hope this helps,
Teodora Guenkova-Luy


Henning Schulzrinne wrote:

> Thanks for all the explanation. It seems a bit asymmetric that=20
> framerate is specified as a generic a=3D parameter in SDP, while=20
> resolution is specified, if at all, on a per-codec basis. Would it=20
> make sense to have an 'a=3Dwidth' and 'a=3Dheight' parameter as well?
>
> Teodora Guenkova-Luy wrote:
>
>> Henning Schulzrinne wrote:
>>  >Do any video payload formats specify, as SDP format attributes or
>>  >otherwise, the maximum image size supported natively (i.e., without
>>  >downscaling)?
>> =
-------------------------------------------------------------------------=
=20
>>
>> Dear All,
>>
>> The ITU-T standards (e.g. H.261, H.263, H.264, etc.) explicitly=20
>> specify image sizes for the codecs. H.261 and H.263 specify image=20
>> sizes according to ITU CIF standard in form of (height x width)=20
>> resolution. H.264 also specifies max image sizes per used profile=20
>> level in form of max bits per image, but without specifying aspect=20
>> ratios. From the MPEG standards only MPEG-1 specifies max image size=20
>> of 352x240 height x width resolution. All the other Video MPEGs are=20
>> with arbitrary image sizes.  Another possibility to specify image=20
>> sizes are the VESA standards from www.vesa.org (e.g. VESA's VGA BIOS=20
>> Extensions).  Note, that some standards for video specify frame-rate, =

>> frame-size and colour-quality dependencies. Hence, using some=20
>> standard definitions might influence all of these 3 parameters, i.e.=20
>> the colour-quality (number of colours per image) might also influence =

>> the amount of network data for the video streams.
>> Note, that frame-size can be specified in 3 different manners:
>> - (height x width) resolution in (pixel x pixel)
>> - standard - i.e. CIF or VESA
>> - max number of bits + aspect ratio - e.g. commonly used aspect=20
>> ratios are height: width =3D 4:3, 5:4, 11:9, 16:9, 16:10. These =
aspect=20
>> ratios stem also from CIF and VESA
>> For the complete and unique video performance specification with SDP=20
>> in case of arbitrary video image specification one needs to have=20
>> specifications for frame-rate, frame-size and colour-quality.
>> Best Regards,
>>    Teodora Guenkova-Luy
>>
>>
>>
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic
>
>
>
>

--=20
-----------------------------------------
Teodora Guenkova-Luy

Dept. Distributed Systems, University of Ulm,=20
Oberer Eselsberg, 89069 Ulm, Germany

Tel.: +49 (0731) 502-4148=20
FAX: +49 (0731) 502-4142

e-Mail: guenkova@vs.informatik.uni-ulm.de

Home Page: http://www-vs.informatik.uni-ulm.de/~Guenkova/=20

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



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From mmusic-bounces@ietf.org  Tue Dec 21 18:20:55 2004
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Subject: [MMUSIC] Protocol Action: 'The Alternative Network Address Types 
 Semantics (ANAT) for theSession Description Protocol (SDP) Grouping 
 Framework' to Proposed Standard 
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The IESG has approved the following document:

- 'The Alternative Network Address Types Semantics (ANAT) for theSession 
   Description Protocol (SDP) Grouping Framework '
   <draft-ietf-mmusic-anat-02.txt> as a Proposed Standard

This document is the product of the Multiparty Multimedia Session Control 
Working Group. 

The IESG contact persons are Jon Peterson and Allison Mankin.

Technical Summary
 
This addition to the Session Description Protocol (SDP) allows an SDP offer to 
contain more than one network address associated with a single offered media 
stream. The motivation for this is IPv6 transition - it allows a dual-stack 
client to offer both an IPv6 and an IPv4 address from which the answerer can 
choose. It also allows these addresses to be prioritized, so an offerer can 
indicate that they would prefer to connect via IPv6, if possible. This has 
significant applicability to the Interactive Connectivity Establishment (ICE) 
mechanism being developed in MMUSIC.
 
Working Group Summary
 
The MMUSIC working group supported the advancement of this document. A concern 
was expressed that interaction of other media grouping semantics in SDP was 
insufficiently specified, but the media grouping framework was authored by some
f the same individuals who wrote this document, and they seem to feel that the 
distinction is clear enough.
 
Protocol Quality
 
This specification was reviewed for the IESG by Jon Peterson.


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From mmusic-bounces@ietf.org  Wed Dec 22 08:32:08 2004
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From: Vavilapalli Srikanth-A19563 <srikanthv@motorola.com>
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Cc: fandreas@cisco.com, dwing@cisco.com
Subject: [MMUSIC] SDP Support for exchanging different auth tag length for
	SRTP and SRTCP
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Hi

Ref: <draft-ietf-mmusic-sdescriptions-07.txt>

In this draft, I did not find a way to exchange the both SRTP and SRTCP
authentication tag lengths using SDP secure attributes. 

i.e. if a SRTP application wishes to use 32-bit weak authentication for SRTP
packets and 80-bit strong authentication for SRTCP packets and has chosen
the mechanism described in the above draft to communicate between sender and
receiver, How can a offerer/answerer include this information in the
a=crypto line of SDP offer or answer respectively?

Regards
Srikanth

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From: "Christer Holmberg (JO/LMF)" <christer.holmberg@ericsson.com>
To: "'Joerg Ott'" <jo@tzi.uni-bremen.de>, mmusic@ietf.org
Subject: RE: [MMUSIC] WGLC: draft-ietf-mmusic-sdp-new-23.txt
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Hi,

I have a small question.

The sdp-new draft says that "inactive" is used to put users on hold, while RFC3264(offer/answer) says that "sendonly" is used.

Now, in the original SDP RFC, which RFC3246 refers to, "inactive" was not defined, so I wonder if RFC3264 is to be updated?

Or, are both hold mechanisms "according to the standard"?

Regards,

Christer Holmberg
Ericsson Finland


> -----Original Message-----
> From: mmusic-bounces@ietf.org 
> [mailto:mmusic-bounces@ietf.org]On Behalf
> Of Joerg Ott
> Sent: 19. joulukuuta 2004 18:51
> To: mmusic@ietf.org
> Cc: Colin Perkins
> Subject: [MMUSIC] WGLC: draft-ietf-mmusic-sdp-new-23.txt
> 
> 
> This is to announce Working Group Last Call on the revised
> SDP specification "SDP: Session Description Protocol"
> recently published as
> 
>      draft-ietf-mmusic-sdp-new-23.txt
> 
> The document shall be advanced to Proposed Standard (PS).
> 
> The WGLC is to end on 9 January 2005.
> 
> Please review the latest draft carefully and direct all comments
> to the MMUSIC list and the authors.
> 
> Joerg
> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

_______________________________________________
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From: Mark Baugher <mbaugher@cisco.com>
Subject: Re: [MMUSIC] SDP Support for exchanging different auth tag length for
	SRTP and SRTCP
Date: Wed, 22 Dec 2004 06:22:18 -0800
To: Vavilapalli Srikanth-A19563 <srikanthv@motorola.com>
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	le" "ngth for SRTP and SRTCP";
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hi Srikanth,

On Dec 22, 2004, at 12:51 AM, Vavilapalli Srikanth-A19563 wrote:

> Hi
>
> Ref: <draft-ietf-mmusic-sdescriptions-07.txt>
>
> In this draft, I did not find a way to exchange the both SRTP and SRTCP
> authentication tag lengths using SDP secure attributes.

There is no way to specify a different cryptographic policy for SRTP 
and SRTCP in sdescriptions.  The reason for that is that I think it's a 
bad idea to have unnecessary complexity in a security protocol.
>
> i.e. if a SRTP application wishes to use 32-bit weak authentication 
> for SRTP
> packets and 80-bit strong authentication for SRTCP packets and has 
> chosen
> the mechanism described in the above draft to communicate between 
> sender and
> receiver, How can a offerer/answerer include this information in the
> a=crypto line of SDP offer or answer respectively?

They can't.  If someone wants to introduce this level of complexity 
into SRTP signaling, then they can write their own sdescriptions draft. 
  sdescriptions is extensible, but I think it's better to keep things as 
simple as they can be.  Allowing multiple policies within an SRTP 
session to me isn't simple.  SRTP as a protocol does admit separate 
policies for SRTP and SRTCP.  sdescriptions supports only a single 
policy per session.

Cheers, Mark
>
> Regards
> Srikanth
>
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
>

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Thread-Topic: Question about direction in
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From: "Pagani, Florence" <florence.pagani@hp.com>
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Hi all,
=20
I am wondering why in
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-offer-answer-examp
les-05.txt, there is an answer without any direction (i.e. the default =
=3D
sendrecv) to an offer with direction  sendonly. I would have put
recvonly instead... Did I miss something?
=20
Here is the example:
  _____ =20

3.2  Hold with Two Streams

   Alice offers two media streams, a bidirectional audio stream and a
   send only telephone event stream.  Bob accepts both streams.  Bob
   then puts Alice's audio stream on hold but not the tone stream.
   Alice responds with identical SDP to the initial offer.


    [Offer]

      v=3D0
      o=3Dalice 2890844526 2890844526 IN IP4 host.atlanta.example.com
      s=3D
      c=3DIN IP4 host.atlanta.example.com
      t=3D0 0
      m=3Daudio 49170 RTP/AVP 0 97
      a=3Drtpmap:0 PCMU/8000
      a=3Drtpmap:97 iLBC/8000
      m=3Daudio 49172 RTP/AVP 98
      a=3Drtpmap:98 telephone-event/8000
      a=3Dsendonly

   [Answer]

      v=3D0
      o=3Dbob 2808844564 2808844564 IN IP4 host.biloxi.example.com
      s=3D
      c=3DIN IP4 host.biloxi.example.com
      t=3D0 0
      m=3Daudio 49172 RTP/AVP 97
      a=3Drtpmap:97 iLBC/8000
      m=3Daudio 49174 RTP/AVP 98
      a=3Drtpmap:98 telephone-event/8000
      a=3Drecvonly

   [Second-Offer]

      v=3D0
      o=3Dbob 2808844564 2808844565 IN IP4 host.biloxi.example.com
      s=3D
      c=3DIN IP4 host.biloxi.example.com
      t=3D0 0
      m=3Daudio 49172 RTP/AVP 97
      a=3Drtpmap:97 iLBC/8000
      a=3Dsendonly
      m=3Daudio 49174 RTP/AVP 98
      a=3Drtpmap:98 telephone-event/8000
      a=3Drecvonly


   [Second-Answer]

      v=3D0
      o=3Dalice 2890844526 2890844527 IN IP4 host.atlanta.example.com
      s=3D
      c=3DIN IP4 host.atlanta.example.com
      t=3D0 0
      m=3Daudio 49170 RTP/AVP 0 97
      a=3Drtpmap:0 PCMU/8000
      a=3Drtpmap:97 iLBC/8000                    --> Why not recvonly =
????
      m=3Daudio 49172 RTP/AVP 98
      a=3Drtpmap:98 telephone-event/8000
      a=3Dsendonly

--
Florence PAGANI
HP OpenCall Business Unit - New Venture
+33 (0)4 76 14 57 53
florence.pagani@hp.com
=20

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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Dus-ascii">
<TITLE>Message</TITLE>

<META content=3D"MSHTML 6.00.2800.1479" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3D"Comic Sans MS" size=3D2><SPAN =
class=3D268261016-22122004>Hi=20
all,</SPAN></FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2><SPAN=20
class=3D268261016-22122004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2><SPAN =
class=3D268261016-22122004>I am=20
wondering why in <A=20
href=3D"http://www.ietf.org/internet-drafts/draft-ietf-mmusic-offer-answe=
r-examples-05.txt">http://www.ietf.org/internet-drafts/draft-ietf-mmusic-=
offer-answer-examples-05.txt</A>,=20
there is&nbsp;an answer without any&nbsp;direction (i.e. the default =3D =

<U>sendrecv</U>) to an offer with direction &nbsp;<U>sendonly</U>. I =
would have=20
put recvonly instead... Did I miss something?</SPAN></FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2><SPAN=20
class=3D268261016-22122004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2><SPAN =
class=3D268261016-22122004>Here is=20
the example:</SPAN></FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2><SPAN =
class=3D268261016-22122004>
<HR>
</SPAN></FONT></DIV>
<DIV><FONT face=3D"Comic Sans MS" size=3D2><SPAN =
class=3D268261016-22122004><SPAN=20
class=3D090461714-21122004><FONT size=3D2>3.2&nbsp; Hold with Two=20
Streams<BR><BR>&nbsp;&nbsp; Alice offers two media streams, a =
bidirectional=20
audio stream and a<BR>&nbsp;&nbsp; send only telephone event =
stream.&nbsp; Bob=20
accepts both streams.&nbsp; Bob<BR>&nbsp;&nbsp; then puts Alice's audio =
stream=20
on hold but not the tone stream.<BR>&nbsp;&nbsp; Alice responds with =
identical=20
SDP to the initial offer.<BR><BR><BR>&nbsp;&nbsp;&nbsp;=20
[Offer]<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
v=3D0<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; o=3Dalice 2890844526 2890844526 =
IN IP4=20
host.atlanta.example.com<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
s=3D<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; c=3DIN IP4=20
host.atlanta.example.com<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; t=3D0=20
0<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; m=3Daudio 49170 RTP/AVP 0=20
97<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drtpmap:0=20
PCMU/8000<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drtpmap:97=20
iLBC/8000<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; m=3Daudio 49172 RTP/AVP=20
98<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drtpmap:98=20
telephone-event/8000<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
a=3Dsendonly<BR><BR>&nbsp;&nbsp; =
[Answer]<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
v=3D0<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; o=3Dbob 2808844564 2808844564 IN =
IP4=20
host.biloxi.example.com<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
s=3D<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; c=3DIN IP4=20
host.biloxi.example.com<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; t=3D0=20
0<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; m=3Daudio 49172 RTP/AVP=20
97<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drtpmap:97=20
iLBC/8000<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; m=3Daudio 49174 RTP/AVP=20
98<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drtpmap:98=20
telephone-event/8000<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
a=3Drecvonly<BR><BR>&nbsp;&nbsp;=20
[Second-Offer]<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
v=3D0<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; o=3Dbob 2808844564 2808844565 IN =
IP4=20
host.biloxi.example.com<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
s=3D<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; c=3DIN IP4=20
host.biloxi.example.com<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; t=3D0=20
0<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; m=3Daudio 49172 RTP/AVP=20
97<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drtpmap:97=20
iLBC/8000<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3D<FONT=20
color=3D#ff00ff>sendonly</FONT><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
m=3Daudio 49174=20
RTP/AVP 98<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drtpmap:98=20
telephone-event/8000<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
a=3Drecvonly<BR><BR><BR>&nbsp;&nbsp;=20
[Second-Answer]<BR><BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
v=3D0<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; o=3Dalice 2890844526 2890844527 =
IN IP4=20
host.atlanta.example.com<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
s=3D<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; c=3DIN IP4=20
host.atlanta.example.com<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; t=3D0=20
0<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT><FONT color=3D#ff00ff =
size=3D2> m=3Daudio=20
49170 RTP/AVP 0 97<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drtpmap:0=20
PCMU/8000<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; a=3Drtpmap:97=20
iLBC/8000&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;</FONT><FONT size=3D2><FONT=20
color=3D#0000ff>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nb=
sp;&nbsp;&nbsp;&nbsp;&nbsp;=20
<EM>--&gt; Why not recvonly =
????</EM><BR></FONT>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
m=3Daudio 49172 RTP/AVP 98<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; =
a=3Drtpmap:98=20
telephone-event/8000<BR>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;=20
a=3Dsendonly<BR></FONT></SPAN></SPAN></FONT></DIV>
<DIV align=3Dleft><FONT face=3D"Comic Sans MS" size=3D2>--</FONT></DIV>
<DIV align=3Dleft><FONT face=3D"Comic Sans MS" size=3D2>Florence =
PAGANI</FONT></DIV>
<DIV align=3Dleft><FONT face=3D"Comic Sans MS"><FONT size=3D2><SPAN=20
class=3D268261016-22122004>HP </SPAN>OpenCall Business Unit - New=20
Venture</FONT></FONT></DIV>
<DIV align=3Dleft><FONT face=3D"Comic Sans MS" size=3D2>+33 (0)4 =
76&nbsp;14 57=20
53</FONT></DIV>
<DIV align=3Dleft><FONT face=3D"Comic Sans MS"=20
size=3D2>florence.pagani@hp.com</FONT></DIV>
<DIV>&nbsp;</DIV></BODY></HTML>

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Subject: RE: [MMUSIC] WGLC: draft-ietf-mmusic-sdp-new-23.txt
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> Hi,
> 
> I have a small question.
> 
> The sdp-new draft says that "inactive" is used to put users 
> on hold, while RFC3264(offer/answer) says that "sendonly" is used.
> 
> Now, in the original SDP RFC, which RFC3246 refers to, 
> "inactive" was not defined, so I wonder if RFC3264 is to be updated?
> 
> Or, are both hold mechanisms "according to the standard"?

Yes.

They are slighly different forms of "hold".

SIP/SDP doesn't really have a concept of "hold" in the traditional telephony
sense. 

sendonly means that you can not receive media.

inactive means that you can not receive media and that you are not sending
either.

They could both mean that you are "on hold". You can think of a Music On
Hold 
service,  versus a normal Hold service. But neither necessarily means "hold"
as per 
traditional telephony. You could send inactive or sendonly for reasons that
are
not a logical hold, but just plain media pausing.


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<P><FONT SIZE=2>&gt; Hi,</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; I have a small question.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; The sdp-new draft says that &quot;inactive&quot; is used to put users </FONT>
<BR><FONT SIZE=2>&gt; on hold, while RFC3264(offer/answer) says that &quot;sendonly&quot; is used.</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Now, in the original SDP RFC, which RFC3246 refers to, </FONT>
<BR><FONT SIZE=2>&gt; &quot;inactive&quot; was not defined, so I wonder if RFC3264 is to be updated?</FONT>
<BR><FONT SIZE=2>&gt; </FONT>
<BR><FONT SIZE=2>&gt; Or, are both hold mechanisms &quot;according to the standard&quot;?</FONT>
</P>

<P><FONT SIZE=2>Yes.</FONT>
</P>

<P><FONT SIZE=2>They are slighly different forms of &quot;hold&quot;.</FONT>
</P>

<P><FONT SIZE=2>SIP/SDP doesn't really have a concept of &quot;hold&quot; in the traditional telephony</FONT>
<BR><FONT SIZE=2>sense. </FONT>
</P>

<P><FONT SIZE=2>sendonly means that you can not receive media.</FONT>
</P>

<P><FONT SIZE=2>inactive means that you can not receive media and that you are not sending</FONT>
<BR><FONT SIZE=2>either.</FONT>
</P>

<P><FONT SIZE=2>They could both mean that you are &quot;on hold&quot;. You can think of a Music On Hold </FONT>
<BR><FONT SIZE=2>service,&nbsp; versus a normal Hold service. But neither necessarily means &quot;hold&quot; as per </FONT>
<BR><FONT SIZE=2>traditional telephony. You could send inactive or sendonly for reasons that are</FONT>
<BR><FONT SIZE=2>not a logical hold, but just plain media pausing.</FONT>
</P>

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From: "Moustafa A. Youssef" <moustafa@cs.umd.edu>
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Subject: [MMUSIC] CFP:  IEEE WirelessCom Symposium on Mobile Computing, 2005
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(Our apologies if you receive multiple copies of this CFP)

-------------------------------------------------------------
                Call for Papers
           Symposium on Mobile Computing
  http://www.csee.umbc.edu/~younis/WC_MobCom05.htm
                  as part of
             IEEE WirelessCom 2005
Sheraton Maui Resort, Kaanapali Beach, Maui, Hawaii, USA
http://www.ece.queensu.ca/hpages/faculty/safwat/wirelesscom/05/
                 June 13-16, 2005
  Technical Sponsorship: IEEE TCCC, TCPC, and SSCTC


Recent years have witnessed an enormous growth in the development
and applications of mobile computing environments and devices.
Many practitioners envision a future empowered with context and
location aware computation and communication infrastructure that
allow users to access data and receive service at any place and
any time. Compared to conventional platforms, the mobility of
computing nodes introduces many challenges to the design of almost
all systems components.

This symposium will foster a forum for discussing and presenting
recent research results on mobile computer systems. Original
papers are invited in the broader area of mobile computing. Papers
must report high-quality and previously unpublished work. Topics
of interest include, but are not limited to, the following:

-  Architecture and infrastructure support
-  Middleware and operating systems services
-  Development platforms and environments
-  Adaptive resource management schemes
-  Pervasive and Context-aware Computing
-  Performance metrics and studies
-  Power aware computation and communication
-  Data consistency and synchronization protocols
-  Fault-tolerant architectural and operation models
-  Wearable computing devices and systems
-  Time constrained mobile systems
-  Robust ad-hoc networking techniques
-  Efficient localization techniques
-  Wireless sensor networks
-  Mobile peer to peer systems
-  Quality of service provisioning
-  Context-conscious computing
-  Web caching and services
-  Emerging applications and systems
-  Mobile agents
-  Mobile commerce
-  Security and privacy issues


Submission Guidelines
---------------------
Prospective authors should submit their paper electronically to
younis@csee.umbc.edu.  All papers are limited to 6 pages in MS-Word, PS
or PDF using standard IEEE double-column format. The first page should
include title, authors' contact information, an abstract and five
keywords. Authors should attach the paper abstract to their message.

Important Dates
---------------
Paper Submission Deadline:      March 15, 2005
Notification of Acceptance:     April 15, 2005
Camera-Ready Papers:            May 1, 2005

Symposium Co-Chairs
-------------------
Mohamed Younis
Department of Computer Science and Electrical Engineering
University of Maryland Baltimore County,
E-mail: younis@csee.umbc.edu

Hesham El-Rewini
Department of Computer Science and Engineering
Southern Methodist University
E-mail: rewini@engr.smu.edu


Publicity Chair
----------------
Moustafa Youssef
Department of Computer Science
University of Maryland at College Park,
E-mail: moustafa@cs.umd.edu



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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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To: Teodora Guenkova-Luy <guenkova@vs.informatik.uni-ulm.de>
Subject: Re: [MMUSIC] Maximum non-scaled image size
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Hi,

I would like to point out that 3GPP has defined in TS 26.234 a SDP 
attribute a=framesize. It has the following definition:

The following media level SDP field is defined for PSS:

- "a=framesize:<payload type number> <width>-<height>"

This gives the largest video frame size of H.263 streams.
The frame size field in SDP is needed by the client in order to properly 
allocate frame buffer memory.  For MPEG-4 Visual streams, the frame size 
shall be extracted from the "config" information in the SDP. For H.264 
(AVC) streams, the frame size shall be extracted from the 
sprop-parameters-sets information in the SDP. For H.263 streams, a PSS 
server shall include the "a=framesize" field at the media level for each 
stream in SDP, and a PSS client should interpret this field, if present. 
Clients should be ready to receive SDP descriptions without this attribute.

If this attribute is present, the frame size parameters shall exactly 
match the largest frame size defined in the video stream. The width and 
height values shall be expressed in pixels.

However, to my understanding, this attribute is not what Henning is 
seeking. I only like to warn you at an early stage about the potential 
name conflict. I am sorry that I haven't had time yet to write the SDP 
registration draft for the PSS specification.

But on the point Henning is asking about, in the 3GPP Packet-based 
Streaming Service, the servers can use UAProf to determine the clients 
screen resolution.


Cheers

Magnus

-- 

Magnus Westerlund

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


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Date: Thu, 23 Dec 2004 13:03:49 +0200
From: "Even, Roni" <roni.even@polycom.co.il>
Subject: RE: [MMUSIC] Maximum non-scaled image size
To: Magnus Westerlund <magnus.westerlund@ericsson.com>,
        Teodora Guenkova-Luy <guenkova@vs.informatik.uni-ulm.de>
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Hi,
Hi,
As Magnus points out the maximum frame size is limited by the available
frame buffer in the decoder but another parameter that is important is
the frame rate. A decoder may have a large frame buffer or high
resolution display, but due to processing power may be able to use the
full size only for low frame rate.

As for screen resolution I am wondering if the parameter that Magnus
describes address the screen resolution or the available decoder frame
buffer, are they the same.

Roni Even 

-----Original Message-----
From: mmusic-bounces@ietf.org [mailto:mmusic-bounces@ietf.org] On Behalf
Of Magnus Westerlund
Sent: Wednesday, December 22, 2004 10:20 PM
To: Teodora Guenkova-Luy
Cc: mmusic@ietf.org; Henning Schulzrinne
Subject: Re: [MMUSIC] Maximum non-scaled image size

Hi,

I would like to point out that 3GPP has defined in TS 26.234 a SDP 
attribute a=framesize. It has the following definition:

The following media level SDP field is defined for PSS:

- "a=framesize:<payload type number> <width>-<height>"

This gives the largest video frame size of H.263 streams.
The frame size field in SDP is needed by the client in order to properly

allocate frame buffer memory.  For MPEG-4 Visual streams, the frame size

shall be extracted from the "config" information in the SDP. For H.264 
(AVC) streams, the frame size shall be extracted from the 
sprop-parameters-sets information in the SDP. For H.263 streams, a PSS 
server shall include the "a=framesize" field at the media level for each

stream in SDP, and a PSS client should interpret this field, if present.

Clients should be ready to receive SDP descriptions without this
attribute.

If this attribute is present, the frame size parameters shall exactly 
match the largest frame size defined in the video stream. The width and 
height values shall be expressed in pixels.

However, to my understanding, this attribute is not what Henning is 
seeking. I only like to warn you at an early stage about the potential 
name conflict. I am sorry that I haven't had time yet to write the SDP 
registration draft for the PSS specification.

But on the point Henning is asking about, in the 3GPP Packet-based 
Streaming Service, the servers can use UAProf to determine the clients 
screen resolution.


Cheers

Magnus

-- 

Magnus Westerlund

Multimedia Technologies, Ericsson Research EAB/TVA/A
----------------------------------------------------------------------
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Torshamsgatan 23           | Fax   +46 8 7575550
S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com


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Thread-Topic: Question on the Usage of Feature Tags in
	draft-ietf-mmusic-rfc2326bis-08.txt
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From: "Thomas Zeng" <zeng@pvnetsolutions.com>
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This is a multi-part message in MIME format.

--===============1775891231==
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------_=_NextPart_001_01C4E91E.07411FB0
Content-Type: text/plain;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

Hello Magnus:
=20
Merry Christmas.
=20
We have been looking at the subject of capability handling which led us =
to RFC2326bis-08, Chapter 10, where it is suggested that the server uses =
the Supported header to advertise the list of feature tags it is capable =
of handling.
=20
It is worth clarifying whether the list of tags in the Supported header =
should be a function of  the request URL or not.
=20
For instance, if a client issues a DESCRIBE on a audio only request URL, =
and the server in its 200 OK response includes the feature tag =
"com.A_company.video-bit-stream-switching". Is that appropriate?=20
=20
Clearly video-bit-stream-switching does not apply to an audio-only URL, =
but one could argue that video-bit-stream-switching is one of the =
server's capabilities and the client may choose to take advantage of it =
later on.=20
=20
What is your interpretation? What does the group think?
=20
Thanks,
-Thomas

------_=_NextPart_001_01C4E91E.07411FB0
Content-Type: text/html;
	charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; =
charset=3Diso-8859-1">


<META content=3D"MSHTML 6.00.2800.1479" name=3DGENERATOR></HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D460432518-23122004>Hello=20
Magnus:</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460432518-23122004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D460432518-23122004>Merry=20
Christmas.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460432518-23122004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D460432518-23122004>We =
have=20
been&nbsp;looking at the subject of capability handling&nbsp;which led =
us=20
to&nbsp;RFC2326bis-08, Chapter 10, where it is suggested that the server =
uses=20
the Supported header to advertise the list of feature tags it is capable =
of=20
handling.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460432518-23122004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D460432518-23122004>It is =
worth=20
clarifying whether the&nbsp;list of tags in&nbsp;the Supported=20
header&nbsp;should be a function of&nbsp;&nbsp;the request URL or=20
not.</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460432518-23122004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D460432518-23122004>For =
instance, if a=20
client&nbsp;issues a DESCRIBE on a audio only request URL, and the =
server in its=20
200 OK response includes the feature tag=20
"com.A_company.video-bit-stream-switching". Is that appropriate?=20
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460432518-23122004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN =
class=3D460432518-23122004>Clearly=20
video-bit-stream-switching does not apply to an audio-only URL, but one =
could=20
argue that video-bit-stream-switching is one of the server's =
capabilities and=20
the client may choose to take advantage of it later on. =
</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460432518-23122004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN class=3D460432518-23122004>What =
is your=20
interpretation? What does the group think?</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460432518-23122004></SPAN></FONT>&nbsp;</DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460432518-23122004>Thanks,</SPAN></FONT></DIV>
<DIV><FONT face=3DArial size=3D2><SPAN=20
class=3D460432518-23122004>-Thomas</SPAN></FONT></DIV></BODY></HTML>

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From mmusic-bounces@ietf.org  Fri Dec 24 12:28:45 2004
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From: sunil bhargo <marx_bhargav@yahoo.com>
Subject: Re: [MMUSIC] Question on the Usage of Feature Tags in
	draft-ietf-mmusic-rfc2326bis-08.txt
To: Thomas Zeng <zeng@pvnetsolutions.com>, mmusic@ietf.org,
        magnus.westerlund@ericsson.com
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Hi Thomas,

          Clients having the capability to maintain
the server capabilities might require server
capabilities and not url specific capabilities. If the
client and server have exchanged the capabilities
previously. In future if the clients receives the
meta-file having multiple urls for the same content
and the SDP is received Out of Band for these urls
then the client can use the previously determined
capabilities to select which url to use to have best
experience.

          But it also might be argued that if the
client has such capability then it should use the
OPTIONS with "*" to get the server capability. And the
server capabilities will only be sent if url is "*"
and if it is presentation url then the capabilities
pretaining to the media are sent. 

          But I would prefer that the Supported Header
gives the server capabilities instead of the
capabilities related to the url. If the server
capabilities are used then it is still easy for the
client to know what pretains to the current session
(as media capabilities will be subset of total
capabilities). 
          
sunil bhargo

--- Thomas Zeng <zeng@pvnetsolutions.com> wrote:

> Hello Magnus:
>  
> Merry Christmas.
>  
> We have been looking at the subject of capability
> handling which led us to RFC2326bis-08, Chapter 10,
> where it is suggested that the server uses the
> Supported header to advertise the list of feature
> tags it is capable of handling.
>  
> It is worth clarifying whether the list of tags in
> the Supported header should be a function of  the
> request URL or not.
>  
> For instance, if a client issues a DESCRIBE on a
> audio only request URL, and the server in its 200 OK
> response includes the feature tag
> "com.A_company.video-bit-stream-switching". Is that
> appropriate? 
>  
> Clearly video-bit-stream-switching does not apply to
> an audio-only URL, but one could argue that
> video-bit-stream-switching is one of the server's
> capabilities and the client may choose to take
> advantage of it later on. 
>  
> What is your interpretation? What does the group
> think?
>  
> Thanks,
> -Thomas
> > _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 


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Subject: [MMUSIC] Re: Question on the Usage of Feature Tags in
	draft-ietf-mmusic-rfc2326bis-08.txt
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Hi Thomas,

My intention with "Supported" is that these are not linked to the URI, 
and instead describe features supported by the client and server 
depending on the direction of the message.

In the case one has features that are media type specific, the ones 
understanding the features, knows that it is media specific. Also the 
definer of a feature must consider if further negotiation of the 
functionality is needed. The Supported mechanism is really only a 
mechanism for intial discovery of support of features. It has the 
meaning like: Hey, here is what I support, please send me what you can.

So I could see that one clarifies this in the next version, I would 
propose to make the following change in the section 14.43, see last 
sentencte of paragraph 2:

    The Supported header field enumerates all the extensions supported by
    the client or server using feature tags. The header carries the
    extensions supported by the message sending entity.  The Supported
    header MAY be included in any request.  When present in a request,
    the receiver MUST respond with its corresponding Supported header.
    Note, also in 4xx and 5xx responses is the supported header included.

    The Supported header field contains a list of feature-tags, described
    in Section 3.7, that are understood by the client or server. The list
    is not depending on the request URI context, and describes the client
    and server capabilities in general.

What do you say?

Merry Chrismtas and a Happy new Year

Magnus


Thomas Zeng wrote:
> Hello Magnus:
>  
> Merry Christmas.
>  
> We have been looking at the subject of capability handling which led us 
> to RFC2326bis-08, Chapter 10, where it is suggested that the server uses 
> the Supported header to advertise the list of feature tags it is capable 
> of handling.
>  
> It is worth clarifying whether the list of tags in the Supported 
> header should be a function of  the request URL or not.
>  
> For instance, if a client issues a DESCRIBE on a audio only request URL, 
> and the server in its 200 OK response includes the feature tag 
> "com.A_company.video-bit-stream-switching". Is that appropriate?
>  
> Clearly video-bit-stream-switching does not apply to an audio-only URL, 
> but one could argue that video-bit-stream-switching is one of the 
> server's capabilities and the client may choose to take advantage of it 
> later on.
>  
> What is your interpretation? What does the group think?
>  
> Thanks,
> -Thomas


-- 

Magnus Westerlund

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


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Subject: Re: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
From: Cullen Jennings <fluffy@cisco.com>
To: Peter Ridler <ridler@softrac.ca>, "'Joerg Ott'" <jo@tzi.uni-bremen.de>,
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What will your two SDP parse engines do differently?


On 12/18/04 7:16 AM, "Peter Ridler" <ridler@softrac.ca> wrote:

> 
> I also agree with Colin and Joerg.
> 
> I will also have two SDP parse engines (one for RTP and another for MSRP).
> This will complicate the code.  Computers do not care about ugly formatting
> --- overall performance should be the concern (I think all text formatted
> protocols are a waste of network bandwidth and memory resources -- but I'm
> stuck with them).
> 
> We should follow the standards as best we can.  If the standard does not
> meet your requirements then define/use another.
> 
> Another text based protocol is draft-ietf-sipping-toip-00.txt  This fits the
> SIP/SDP model nicely.
> 
> Peter
>  
> 
>> -----Original Message-----
>> From: simple-bounces@ietf.org
>> [mailto:simple-bounces@ietf.org] On Behalf Of Joerg Ott
>> Sent: Sunday, December 12, 2004 6:58 AM
>> To: Colin Perkins
>> Cc: Simple WG; mmusic@ietf.org
>> Subject: Re: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
>> 
>> I agree with Colin on the ugliness issue of decomposing URLs
>> into c= and m= lines.  But this is the way SDP has worked and
>> works today.
>> 
>> The present discussion we are having here is about changing
>> SDP to for address/URL handling in one particular case.  This
>> does not strike me as something MSRP should be concerned with.
>> 
>> I would like to make one thing clear: we are either using
>> existing tools in another document or we are building new
>> ones.  There is nothing much in between.
>> 
>> Changing c=/m= semantics means changing SDP.  Just because
>> the syntax still looks the same does not mean that it is
>> still the same protocol.
>> Any entity or generic protocol engine used to dealing with
>> SDP m=/c= would no longer do the right thing.
>> 
>> We see this kind of approach quite often -- by far too often
>> -- over the past years (just think the many "uses" of SIP
>> defined elsewhere, usually resulting in something "a little
>> but not entirely unlike" SIP).
>> 
>> What I would like avoid is having special treatment of SDP
>> depending on what the media type is (or other attributes signify).
>> 
>> Therefore, I strongly agree with Colin's conclusion: take SDP
>> as is or leave it altogether and pursue a different path.
>> 
>> Joerg
>> 
>> 
>> Colin Perkins wrote:
>> 
>>> On 10 Dec 2004, at 21:41, Ben Campbell wrote:
>>> 
>>>> I really need to get closure on this, so I'd appreciate people
>>>> agreeing or disagreeing.
>>>> 
>>>> I have had off-list correspondence indicating both
>> positions. In summary:
>>>> 
>>>> In favor of the change, I had a comment to the effect that
>> changing 
>>>> SDP semantics to ignore the C line and the port field on
>> the M line 
>>>> really changes it to something that is no longer SDP, and
>> that such 
>>>> deviations are in general bad for interop.
>>>> 
>>>> In favor of leaving things as currently specified, I had
>> two comments 
>>>> that decomposing URLs into the C and M lines is ugly, and
>> that having 
>>>> a different mechanism to communicate endpoint addresses vs relay
>>>> addresses is really ugly.
>>>> 
>>>> So, do people agree with one or the other position?
>>> 
>>> 
>>> Decomposing the URL into the "c=" and "m=" lines is very ugly.
>>> Unfortunately, that ugliness is required if you want to use
>> SDP. This 
>>> is why the SDPng work was started, to try to move to something less
>>> horrible...
>>> 
>>> Colin
>>> 
>>> 
>>> _______________________________________________
>>> mmusic mailing list
>>> mmusic@ietf.org
>>> https://www1.ietf.org/mailman/listinfo/mmusic
>>> 
>> 
>> 
>> 
>> _______________________________________________
>> Simple mailing list
>> Simple@ietf.org
>> https://www1.ietf.org/mailman/listinfo/simple
>> 
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic


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One basic question is whether this shouldn't be defined for all video 
codecs, rather than having each codec do this slightly differently.

Once this is clearer, we can then decide whether this is one parameter 
or possibly more. So far, all variations, whether motivated by frame 
buffer size or display size, seem to have the same effect: limit what 
the receiver is willing to receive or indicate what the sender will 
send, where the former seems somewhat more important.

Magnus, I'm not sure where you perceive the difference in semantics to 
my earlier description.

Magnus Westerlund wrote:
> Hi,
> 
> I would like to point out that 3GPP has defined in TS 26.234 a SDP 
> attribute a=framesize. It has the following definition:
> 
> The following media level SDP field is defined for PSS:
> 
> - "a=framesize:<payload type number> <width>-<height>"
> 
> This gives the largest video frame size of H.263 streams.
> The frame size field in SDP is needed by the client in order to properly 
> allocate frame buffer memory.  For MPEG-4 Visual streams, the frame size 
> shall be extracted from the "config" information in the SDP. For H.264 
> (AVC) streams, the frame size shall be extracted from the 
> sprop-parameters-sets information in the SDP. For H.263 streams, a PSS 
> server shall include the "a=framesize" field at the media level for each 
> stream in SDP, and a PSS client should interpret this field, if present. 
> Clients should be ready to receive SDP descriptions without this attribute.
> 
> If this attribute is present, the frame size parameters shall exactly 
> match the largest frame size defined in the video stream. The width and 
> height values shall be expressed in pixels.
> 
> However, to my understanding, this attribute is not what Henning is 
> seeking. I only like to warn you at an early stage about the potential 
> name conflict. I am sorry that I haven't had time yet to write the SDP 
> registration draft for the PSS specification.
> 
> But on the point Henning is asking about, in the 3GPP Packet-based 
> Streaming Service, the servers can use UAProf to determine the clients 
> screen resolution.
> 
> 
> Cheers
> 
> Magnus
> 

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This question has been answered, has it not?

Mark
On Dec 22, 2004, at 12:51 AM, Vavilapalli Srikanth-A19563 wrote:

> Hi
>
> Ref: <draft-ietf-mmusic-sdescriptions-07.txt>
>
> In this draft, I did not find a way to exchange the both SRTP and SRTCP
> authentication tag lengths using SDP secure attributes.
>
> i.e. if a SRTP application wishes to use 32-bit weak authentication 
> for SRTP
> packets and 80-bit strong authentication for SRTCP packets and has 
> chosen
> the mechanism described in the above draft to communicate between 
> sender and
> receiver, How can a offerer/answerer include this information in the
> a=crypto line of SDP offer or answer respectively?
>
> Regards
> Srikanth
>

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Hi Henning,

See my comments below. I have restructured your paragraphs to try to 
answer your question as well as possible.


 > Magnus, I'm not sure where you perceive the difference in semantics to
 > my earlier description.
 >

Based on the following Sentence:
"Do any video payload formats specify, as SDP format attributes or
otherwise, the maximum image size supported natively (i.e., without
downscaling)?"

It talks about native support of image size, which I interpreted to mean 
actual screen size. As the actual amount of macroblock, size of frame 
buffer, that a receiver is capable of decoding is different from what it 
natively can display. Thus I think there is clear reasons to separate 
the capability for decoding from that of displaying. However I don't 
necessarily think that it is appropriate to consider display resolution 
capabilities in SDP, and definitely not in any overlapping parameter. 
There are good reasons to differentiate the two capabilities.

However if your not interested in the terminal display capabilities, but 
rather decoding capabilities, then see below.

Henning Schulzrinne wrote:
> One basic question is whether this shouldn't be defined for all video 
> codecs, rather than having each codec do this slightly differently.
> 
The problem is that the video codecs do all provide some profiles and 
levels describing decoding capabilities. However of course the evolution 
of video encoding has resulted in that they are not described using a 
common profile and level description. Also as the profiles and levels 
describe more than simple image sizes, the profile and levels parameter 
would still be needed if a common frame size parameter existed.

So if you simple want to determine the decoders capability to handle a 
screen size, then I think the profile and level system included in the 
H.26x and MPEG codec are quite sufficient.

> Once this is clearer, we can then decide whether this is one parameter 
> or possibly more. So far, all variations, whether motivated by frame 
> buffer size or display size, seem to have the same effect: limit what 
> the receiver is willing to receive or indicate what the sender will 
> send, where the former seems somewhat more important.
> 

I think one needs to make clear separation between frame-rate and 
frame-size. However I do not think that this is sufficient at all. There 
are so many other parameters in relation to decoding capabilities that 
are to be considered. A few of these are: number of reference frames, 
bit-rate according to the buffer model, number of macro blocks possible 
to decode in a certain time interval, codec features.

Cheers

Magnus Westerlund

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

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> Based on the following Sentence:
> "Do any video payload formats specify, as SDP format attributes or
> otherwise, the maximum image size supported natively (i.e., without
> downscaling)?"
> 
> It talks about native support of image size, which I interpreted to mean 
> actual screen size. As the actual amount of macroblock, size of frame 
> buffer, that a receiver is capable of decoding is different from what it 
> natively can display. Thus I think there is clear reasons to separate 
> the capability for decoding from that of displaying. However I don't 
> necessarily think that it is appropriate to consider display resolution 
> capabilities in SDP, and definitely not in any overlapping parameter. 
> There are good reasons to differentiate the two capabilities.

Fully agreed. Among other differences, the display size is likely a 
"soft" limit, i.e., something that the sender can exceed if you don't 
care about wasting bits. The frame buffer size seems more of a hard limit.

> The problem is that the video codecs do all provide some profiles and 
> levels describing decoding capabilities. However of course the evolution 
> of video encoding has resulted in that they are not described using a 
> common profile and level description. Also as the profiles and levels 
> describe more than simple image sizes, the profile and levels parameter 
> would still be needed if a common frame size parameter existed.
> 
> So if you simple want to determine the decoders capability to handle a 
> screen size, then I think the profile and level system included in the 
> H.26x and MPEG codec are quite sufficient.
> 
>> Once this is clearer, we can then decide whether this is one parameter 
>> or possibly more. So far, all variations, whether motivated by frame 
>> buffer size or display size, seem to have the same effect: limit what 
>> the receiver is willing to receive or indicate what the sender will 
>> send, where the former seems somewhat more important.
>>
> 
> I think one needs to make clear separation between frame-rate and 
> frame-size. However I do not think that this is sufficient at all. There 
> are so many other parameters in relation to decoding capabilities that 
> are to be considered. A few of these are: number of reference frames, 
> bit-rate according to the buffer model, number of macro blocks possible 
> to decode in a certain time interval, codec features.

I'm trying to find a common, useful subset that has high impact on 
either interoperability or efficiency. Clearly, there will always be 
codec-specific parameters, such as the ones you mention. Maximum display 
size and maximum frame buffer size, however, seem pretty universal.


> 
> Cheers
> 
> Magnus Westerlund
> 
> Multimedia Technologies, Ericsson Research EAB/TVA/A
> ----------------------------------------------------------------------
> Ericsson AB                | Phone +46 8 4048287
> Torshamsgatan 23           | Fax   +46 8 7575550
> S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@ericsson.com

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--NextPart

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

	Title		: Session Description Protocol Security 
			  Descriptions for Media Streams
	Author(s)	: F. Andreasen, et al.
	Filename	: draft-ietf-mmusic-sdescriptions-08.txt
	Pages		: 39
	Date		: 2004-12-27
	
This document defines a Session Description Protocol (SDP) 
   cryptographic attribute for unicast media streams.  The attribute 
   describes a cryptographic key and other parameters, which serve to 
   configure security for a unicast media stream in either a single 
   message or a roundtrip exchange.  The attribute can be used with a 
   variety of SDP media transports and this document defines how to use 
   it for the Secure Real-time Transport Protocol (SRTP) unicast media 
   streams.  The SDP crypto attribute requires the services of a data 
   security protocol to secure the SDP message.

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From mmusic-bounces@ietf.org  Tue Dec 28 07:51:02 2004
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Subject: Re: [MMUSIC] Re: [Simple] MSRP: SDP usage concern
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Cc: Simple WG <simple@ietf.org>, Joerg Ott <jo@tzi.uni-bremen.de>,
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Hisham Khartabil wrote:

> I have a question to SDP stack implementers that plan to implement 
> MSRP: Does the current way MSRP specifies the use of SDP (ignore the c 
> line address and the m line port and instead use the attribute a=path) 
> cause problems to you and requires you to do some hacking in your SDP 
> stack? If so, is that acceptable or do you prefer for your MSRP 
> implementation to implement two ways of contructing an MSRP URL 
> depending on the what the next hop is?


Having implemented an SDP parser before, I can comfortably assert that 
the design I employed would require no modification at all to 
accommodate the usage proposed in MSRP.

Users of the SDP parser would need to know to look in the a= lines for 
address information -- but this requires no stack changes.

/a

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From: Gonzalo Camarillo <Gonzalo.Camarillo@ericsson.com>
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Cc: Joerg Ott <jo@tzi.uni-bremen.de>, Keith Drage <drage@lucent.com>
Subject: [MMUSIC] Floor-media stream mappings
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Folks,

I am working on a new version of the draft below, which will be an 
MMUSIC WG item as agreed in our last meeting in DC.

http://www.ietf.org/internet-drafts/draft-camarillo-mmusic-sdp-bfcp-00.txt

We need a means to provide clients with the Floor IDs to be used in the 
conference and with the mapping of  each Floor ID with one or more media 
streams. For example, floor ID 123 maps to the media stream whose media 
label is 456.

a=floorid:123 m-stream:456

We would like to get feedback on a couple of issues:

1) Do we want to include human readable information about the floors 
encoded in the SDP? (e.g., "request this floor if you want to make a 
comment on the issue being discussed" or "request this floor if you want 
to bring up a new issue to be discussed")

2) Do we want to allow the syntax of the 'floorid' attribute to express 
Floor IDs that do not map to any media stream? If the answer to both 1) 
and 2) is yes, we could use the floor control protocol to moderate 
face-to-face meetings where the media does not traverse any mixer, for 
example.

Of course, both 1) and 2) are interesting features that will probably 
need to be supported by the conference policy. The question here is 
whether or not we want systems using solely SDP to have access to these 
features.

Thanks,

Gonzalo


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Subject: [MMUSIC] I-D ACTION:draft-ietf-mmusic-media-loopback-00.txt
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A New Internet-Draft is available from the on-line Internet-Drafts directories.
This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: An Extension to the Session Description Protocol 
			  (SDP) for Media Loopback
	Author(s)	: K. Hedayat, et al.
	Filename	: draft-ietf-mmusic-media-loopback-00.txt
	Pages		: 14
	Date		: 2004-12-28
	
The wide deployment of VoIP and Video over IP services has 
    introduced new challenges in managing and maintaining voice/video 
    quality, reliability, and overall performance.  In particular, 
    media delivery is an area that needs attention.  One method of 
    meeting these challenges is monitoring the media delivery 
    performance by looping media back to the transmitter.  This is 
    typically referred to as "active monitoring" of services.   Media 
    loopback is especially popular in ensuring the quality of transport 
    to the edge of a given VoIP or Video over IP service.  Today in 
    networks that deliver real-time media, short of running ' ping' and 
    ' traceroute' to the edge, service providers are left without the 
    necessary tools to actively monitor, manage, and diagnose quality 
    issues with their service.  The extension defined herein adds new 
    SDP media attributes which enables establishment of media sessions 
    where the media is looped back to the transmitter. Such media 
    sessions will serve as monitoring and troubleshooting tools by 
    providing the means for measurement of more advanced VoIP and Video 
    Over IP performance metrics.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-media-loopback-00.txt

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Subject: Re: [MMUSIC] Floor-media stream mappings
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        "mmusic@ietf.org" <mmusic@ietf.org>
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On 12/28/04 11:00 AM, "Gonzalo Camarillo" <Gonzalo.Camarillo@ericsson.com>
wrote:

> Folks,
> 
> I am working on a new version of the draft below, which will be an
> MMUSIC WG item as agreed in our last meeting in DC.
> 
> http://www.ietf.org/internet-drafts/draft-camarillo-mmusic-sdp-bfcp-00.txt
> 
> We need a means to provide clients with the Floor IDs to be used in the
> conference and with the mapping of  each Floor ID with one or more media
> streams. For example, floor ID 123 maps to the media stream whose media
> label is 456.
> 
> a=floorid:123 m-stream:456
> 

Just to make sure I understand - I assume that several floors could map to
one media stream and a single floor could also map to multiple media stream
(like audio and video).

> We would like to get feedback on a couple of issues:
> 
> 1) Do we want to include human readable information about the floors
> encoded in the SDP? (e.g., "request this floor if you want to make a
> comment on the issue being discussed" or "request this floor if you want
> to bring up a new issue to be discussed")
> 

no - the human language issues make me not want to do this. I'm fine with a
a convention that if their is a floor, you probably need to request it if
you want anyone to pay intention to your media. Clearly your second example
would not follow this but I think there are many ways other that SDP that
users could get the idea of what to floor was used for. I don't find these
human readable things in SDP very useful in practice.

That said, I don't feel strongly about this - it the human readable
description is never used, it does no harm. It could also be added later if
it was needed. 

> 2) Do we want to allow the syntax of the 'floorid' attribute to express
> Floor IDs that do not map to any media stream? If the answer to both 1)
> and 2) is yes, we could use the floor control protocol to moderate
> face-to-face meetings where the media does not traverse any mixer, for
> example.
> 

yes 

And would also like to consider being able to specify a floor id when no
media stream exists but where the media stream and association to floor will
be added in a future SDP. It may be the requesting of the floor that causes
the creation of the media stream.

> Of course, both 1) and 2) are interesting features that will probably
> need to be supported by the conference policy. The question here is
> whether or not we want systems using solely SDP to have access to these
> features.
> 
> Thanks,
> 
> Gonzalo
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic


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