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Hi Pranjal,

I agree that the transfer of RTSP messages are important. To my 
understanding however there is nothing that prevents a client to use the 
available QoS mechanism, like INTServ or DIFFServ. Otherwise I think 
there is very little that you can do at RTSP level to mitigate such a 
problem depending on network congestion. However the server is 
responsible to ensure that it is not accepting more sessions than it can 
handle.

There are also certain things a client can do, for example the STOP 
scenario you are presenting should not be obvious to a user. When the 
user presses PAUSE/STOP the client can freeze local playback immediately 
and take note of the NPT time. If it takes some time to get the response 
from the server, only waste network and server resources. So it is in 
fact only the cases where the client like something to start that will 
be obvious. I would also like to point out that after the first PLAY 
message, it is in most cases possible to synchronize the media for the 
new play range based on the already established synch between multiple 
media streams. The only thing that will not be accurate in such a 
scenario is the NPT of a certain media TS.

Cheers

Magnus



pkdutta@vsnl.net wrote:
> Hi,
>           I have developed a RTSP Client to interact with a VoD Server
  and it is streaming MPEG-2 Videos quite well. Sometimes I am observing
delays in getting response for PLAY,PUASE,TEARDOWN messages from the
Server. Assuming that the Server is responding intact under heavy
loaded condition the source of the problem goes to the Network
congestion. Under heavy video traffic when the network is overloaded
there are chances that small RTSP Messages are getting lost(and so TCP
retransmission at the Server will try sending again and again) and so
there is a delay in getting response. Is there any already existing
mechanism to implement End to end QoS mechanism for giving priority to
  RTSP Messages. Is there any already established method to overcome such
problems. I feel RTSP Control messages are very important from a viewers
perspective.Just thinking about a viewer who is not able to STOP the
video for minutes!!! I will be very obliged with the valuable
suggestions/comments.

> 
> Thanks,
> Pranjal 
> 
> 
> _______________________________________________
> 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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Hi Geetha,

See inline

Geetha Srikantan wrote:

> Magnus,
> 
> I see your point of view on this, since you clarified your assumptions.
> 
> In previous offline discussions there were discussions about how a RTSP 
> server would
> maintain timing info for live streams. My confusion was centered around 
> that.
> 
> I believe RTCP SR-based synchronization is possible, as you pointed 
> out..however it does
> have some issues, as mentioned previously.
> 
> Replying inline to some of the points in this mesage...
> 
> 
> 
> Magnus Westerlund wrote:
> 
>>> (b) Lost packets between live source and server, imply that the server
>>>     cannot accurately predict the next packet's sequence number.
>>
>>
>>
>> Also here I can't see the problem. The RTP sequence number indicates 
>> the first sequence number for a stream that the given synchronization 
>> information and time range applies for. Thus if one receives a RTP 
>> packet with a higher sequence number one knows that it is for the new 
>> range.
>>
>> A relay server that has received at least one packet can give this 
>> information that is valid even if a several RTP packets is lost before 
>> one gets forward. The client can't determine if the loss is between 
>> the relay or before.
>>
>> And if the server really has not received either the information from 
>> a upstream source or any RTP packet when responding, this value can be 
>> excluded. The RTP-Info header only requires either of the values to be 
>> present.
> 
> 
> I did not realize that the RTP-Info header only requires either of the 
> values to be present.
> I don't see it in section 14.33 of the draft..am I missing something?
> 

Please review the ABNF, which specifies that at least one of the must be 
present.

[snip]

>>
>>> Also the main purpose of the Range header is to map the NPT to
>>> the RTP timestamp. As discussed above this mapping is not terribly
>>> accurate in several live streaming scenarios.
>>
>>
>>
>> As I don't think it is any less accurate then for on-demand media I 
>> don't understand what the problem is.
> 
> 
> I agree that RTCP SR-based mapping of NPT to TS is accurate..except that 
> this mapping
> is not possible until an SR is received for each stream of the RTSP 
> session.
> 

Sure, which depending on RTP profile, assigned RTCP SR bandwidth make 
vary significant from usage to usage.

>>> If the live streams are part of a multicast session then there would
>>> be an indication of the time when the session is active in the SDP,
>>> however, with clock skews etc, it is diffcult to accurately pinpoint
>>> the instantaneous time in the session.
>>
>>
>>
>> The instantaneous time is calculated based of the RTP timestamp for a 
>> given media and the mapping received by Range and RTP-Info between RTP 
>> timestamp and NPT at any point
>>
>> Example:
>>
>>  Sync point
>> -----|------------------------------------x------> t
>> TS   A                                    B
>>
>>
>> When one started the session one received by Range and RTP-Info the 
>> following information for RTP TS A.
>>
>> PLAY response:
>> Range: npt=56.97-
>> RTP-Info: url=<snip>; rtptime=164400
>>
>> Thus A = 1644000
>> The media timestamp rate is 90kHz.
>>
>> The NPT time for B=465357 is:
>>
>> (465357-164400)/90000 + 56.97 = 60,3139666 s
>>
> This is fine as long as a non-zero and numeric Range header value exists.
> In the case where the live source is not RTSP-aware, there is no Range 
> header available
> and sync point A is effectively NPT=0, assuming the server started 
> accessing the live
> session at A.

Sure, in the above case the server can set its internal calculation 
mechanism at point A to 0. Then it would probably be best to give a 
npt=now- answer as it is not in the loop to the better timing.

However even if a RTSP play has non numeric, i.e. npt=now- one can 
calculate the NPT from that point and forward. One can even use wall 
clock time and start counting it in relation to the start point.

> 
>>> If the start time of the live session is not known, then 'npt=now-'
>>> is the only reasonable value for the Range header in a PLAYresponse.
>>>
>>
>> I agree that if one truly does not know the time since session start 
>> it is a viable operation. 
> 
> 
> You mean that npt=now- is acceptable in the Range header in this case, 
> right?
> 

Yes.

>> However all functionality exist to know the time.
> 
> 
> Did you mean that the timestamp to NPT mapping is possible as you 
> described above?

What I mean is that one can build solutions that do not have any of 
these problems as long as one uses the right mechanism.

> 
>>> For this reason, it would be good to *not* require a Range header
>>> other than 'Range: npt=now-' for live streams.
>>>
>>
>> No, I think there are a lot of application that can benefit from 
>> presenting the original wall clock or time since session start.
> 
> 
> As long as there is an understanding that the session start may not be 
> an accurate value, as in cases
> where it is not known.

I think that a server should present an accurate time or use npt=now-.

> 
>>> Perhaps this could also be the mechanism to signal to the client that
>>> this is a live stream and that it needs to perform stream
>>> synchronization strictly via RTCP sender reports.
>>>
>>
>> There are already text in there to indicate that the session has live 
>> qualities. There are already reasons why a client must support 
>> synchronization by RTCP. For example sender side clock skew 
>> corrections, multiple sources using different SSRCs, the possibility 
>> to exclude RTP-Info header. However the RTP-Info+Range header is a 
>> reliable transferred information. Although not guaranteed to be 
>> timely, it has a rather high probability to reach the client prior to 
>> any RTCP message. Also in most case it should reach the client prior 
>> to it finishing initial buffering, thus providing it with good initial 
>> synchronization for all media streams.
> 
> 
> Yes, I agree that RTP-Info+Range can be helpful to the client since they 
> are transmitted over RTSP/TCP.
> My point is that if there are inaccuracies in either RTP-Info or Range 
> because of unknown quantities like
> session start time, missing packets, these need to be tolerated by clients.
> 

I don't see how missing packets effects this at all. Session start time 
is either correctly forwarded by, live source, or first hand RTSP 
server, or it will not be correctly given. In which case I would 
recommend that one uses the npt=now- to indicate this.


>>> Section 14.40, page 91:
>>> ssrc:
>>> The transport header in the server's response cannot populate the ssrc
>>> for each stream, accurately in certain scenarios.
>>> The source of the live stream may not be RTSP-aware, and the server 
>>> has to rely on RTP/RTCP to determine ssrc for each stream.
>>> The server may have joined a multicast live session and receives, 
>>> say, one of the streams.
>>> It also receives SETUP request from a client and needs to respond to 
>>> SETUP from a client,
>>> before it sees the first packet of each stream.
>>> Dropping the ssrc parameter from the Transport header in the SETUP 
>>> response would
>>> simplify this case.
>>
>>
>>
>> It is possible to skip the SSRC, however not recommend as it indicates 
>> for which SSRC the synchronization information in RTP-Info applies to. 
>> Thus I would recommend that server awaits at least the first RTP 
>> packet to determine what SSRC is used for media distribution.
> 
> 
> I am not convinced that the user experience is good when a server 
> refuses clients access to live sessions until
> it has access to all the stream synchronization information. If the 
> server were to send RTP and RTCP packets,
> without the RTP-Info or SSRC in RTSP responses, the client has an 
> opportunity to start decoding each stream
> independently and perform inter-stream synchronization as soon as the 
> client receivess one RTCP SR per stream.
> There is at least some media being rendered at the client, even if it is 
> not lip-sync'd etc.
> 

As I wrote in the other mail I don't think this will occur frequently. 
In a well designed system using RTSP the synch information will be 
distributed using TCP, and will basically arrive as soon as possible. If 
there are problems delivering the RTSP response.

A server waiting for the synchronization information can also use a 1xx 
code to tell that it is in fact waiting for the sync info.

I would like to get some feedback from client implementors, it they 
really are willing start without synch and then try to do graceful fix 
of the synchronization when the information arrives. Also without giving 
the RTP-Info TS + Range it will be impossible to present accurate 
session position information. A client will only be able to calculate 
time since media delivery started.

I think a lot of these discussion concerns issues that should be 
discussed in your streaming relay draft to achieve good performance and 
avoid these issues you are so concerned with. From a client perspective 
, I haven't seen a lot of need for clarifications. However I can look on 
how to further improve the discussion in the RTP appendix on synch. 
Otherwise I would recommend if you have specific concerns to present 
proposals for new text.

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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From: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Hi Geetha,

See inline.

Geetha Srikantan wrote:
> Magnus Westerlund wrote:
>>> (a) That all live sources are RTSP-aware?
>>
>>
>>
>> No, but that the there is a root-distributor that is a RTSP server. 
>> The root RTSP server can be synchronized using either standard RTP 
>> means or any proprietary solution as this interface is not 
>> standardized. However it will be this RTSP root server that will be 
>> responsible giving the stream a NPT time, unless that is also provided 
>> through the interface between RTSP and the live source.
> 
> 
> I was assuming the same thing.
> 
>>>
>>> (b) The server could maintain it's own RTP clock for each stream by 
>>> observing
>>>     the RTP timestamps of packets on each stream?
>>>
>>
>> I don't see how that this would be necessary. If one has a RTSP server 
>> that receives a RTP + RTCP media stream. It will after listened to 
>> this for a few seconds received all necessary synchronization 
>> information. If this is not the root server and is only a relay it can 
>> use RTSP to receive the synchronization information for the media 
>> streams it is forwarding directly in RTSP.
>>
> This is an important point of concern.
> I agree that an RTSP server, after it has received RTCP SR for each 
> stream, can map TS to NPT.
> There are some issues though:
> 
> 1. The server may not see RTCP SRs for all streams in the RTSP session 
> for several seconds,
>   because of (a) staggered transmission of RTCP SRs of each stream, (b) 
> lost SRs.
>   Since RTCP transmission intervals can be 5-10 seconds, this could be 
> 10-20 seconds or so.
> 
> Now the server could either
> (a) deny all client requests for the live session until it receives RTCP 
> SRs for all streams - even
> though the server has RTP packets for all streams. This implies that 
> clients won't have access
> to the live stream for several seconds after the session was requested.
> OR
> (b) respond to client requests without the RTP-Info header in PLAY 
> responses.
> 
> If (a) and (b) are accepted as likely scenarios, then the clients cannot 
> expect to see a RTP-Info header
> in PLAY responses, and should be able to handle the absence of the 
> RTP-Info header.

However I don't see it likely that a relay RTSP server would need to 
rely on RTCP to establish the NPT binding. A relay will use the live 
sessions root RTSP sever or another RTSP relay. As it is using RTSP it 
can receive the RTP-Info and Range header which gives it all necessary 
information, unless things are sent using multiple SSRC, in which case 
RTP-Info at the delivery server still can't be used for synch. If the 
relay is slow on responding, I would still recommend that one wait for 
the synch info. Delivering things without synch does not make people 
happy. In conclusion I don't see a good reason why the delivery server 
would not have access to the synchronization information.


> 
> 2. The server may not know when the live session started, for example if 
> it joins an ongoing
>    live session.
> One option is for the server to assume NPT=0 at the time it receives RTP 
> packets from the
> live source, and use that in synchronization mappings. So, the Range 
> header would have
> this server-interpreted version of NPT.
> 

But if the delivery server uses RTSP to request the media, it will 
receive this information.

I would also like to point out that any delay of getting the synch 
information will only be present for the first user of a certain stream. 
Any user requesting a stream already delivered to the delivery server 
will be able to get correct information directly.

>>> (c) The live source's transmission schedule for packets follows the 
>>> RTP timestamp clock?
>>>     That is, each succeeding packet's RTP timestamp will be equal to 
>>> or greater than all previous
>>>     packets, for that stream (modulo rollover)?
>>>
>>
>> I am only assuming that this is a normal RTP media stream. Thus it can 
>> have backwards jump and other non-monotonic behaviour in the timestamp 
>> field. However I do expect that the RTP packets are sent in order, but 
>> not received by any relay in order. Thus a relay will be required to 
>> take certain care against reordering in the RTP-Info sequence number 
>> field.
> 
> 
> Reordering and lost packets make it difficult to populate the sequence 
> number field of RTP-Info headers
> for live streams. This is not the case for stored content. I think this 
> is a point worth noting for live streaming.
> If the server has to guess about the next packet it is going to send, it 
> can't be a good thing.
> 

I don't see how reordering creates any real problem. When a new user 
joins one takes the last sequence number received. Then in the RTP-Info 
one includes this number plus one. Then one forwards the next packet, it 
this was reordered an is in fact an older packet than what one claims in 
the RTP-Info, the client is able to detect that this packet is prior to 
the start of the stream. If there is packet loss or reorder that results 
in that the first packet that arrives to the client has sequence number 
much higher than the indicated one, it is okay. Because the only thing 
the RTP sequence number in the RTP-Info indicates is from what sequence 
number the request media stream is delivered.


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,
           I have developed a RTSP Client to interact with a VoD Server and 
it is streaming MPEG-2 Videos quite well. Sometimes I am observing delays 
in getting response for PLAY,PUASE,TEARDOWN messages from the Server. 
Assuming that the Server is responding intact under heavy loaded condition 
the source of the problem goes to the Network congestion. Under heavy video 
traffic when the network is overloaded there are chances that small RTSP 
Messages are getting lost(and so TCP retransmission at the Server will try 
sending again and again) and so there is a delay in getting response. Is 
there any already existing mechanism to implement End to end QoS mechanism 
for giving priority to RTSP Messages. Is there any already established 
method to overcome such problems. I feel RTSP Control messages are very 
important from a viewers perspective.Just thinking about a viewer who is 
not able to STOP the video for minutes!!! I will be very obliged with the 
valuable suggestions/comments.

Thanks,
Pranjal 


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



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Hi,
           I have developed a RTSP Client to interact with a VoD Server and 
it is streaming MPEG-2 Videos quite well. Sometimes I am observing delays 
in getting response for PLAY,PUASE,TEARDOWN messages from the Server. 
Assuming that the Server is responding intact under heavy loaded condition 
the source of the problem goes to the Network congestion. Under heavy video 
traffic when the network is overloaded there are chances that small RTSP 
Messages are getting lost(and so TCP retransmission at the Server will try 
sending again and again) and so there is a delay in getting response. Is 
there any already existing mechanism to implement End to end QoS mechanism 
for giving priority to RTSP Messages. Is there any already established 
method to overcome such problems. I feel RTSP Control messages are very 
important from a viewers perspective.Just thinking about a viewer who is 
not able to STOP the video for minutes!!! I will be very obliged with the 
valuable suggestions/comments.

Thanks,
Pranjal


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



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Date: Thu, 1 Apr 2004 16:38:23 +0100
From: Colin Perkins <csp@csperkins.org>
To: "Barany, Pete" <pbarany@qualcomm.com>
Cc: mmusic@ietf.org, jo@tzi.uni-bremen.de
Subject: Re: [MMUSIC] WG last call: draft-ietf-mmusic-sdescriptions-03.txt
Message-Id: <20040401163823.6a6be5c3.csp@csperkins.org>
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We're discussing a few minor issues that came up in the working group last
call (mostly clarifying the IANA registration policies). You can expect an
update shortly, which we expect to submit to the IESG for publication.

Colin



--> "Barany, Pete" <pbarany@qualcomm.com> writes:
> Hi,
> 
> What is the status of this Internet draft? Thanks.
> 
> Regards,
> 
> Pete
> 
> -----Original Message-----
> From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org] On Behalf Of
> Colin Perkins
> Sent: Tuesday, March 02, 2004 9:28 PM
> To: mmusic@ietf.org
> Cc: jo@tzi.uni-bremen.de
> Subject: [MMUSIC] WG last call: draft-ietf-mmusic-sdescriptions-03.txt
> 
> This is to announce a working group last call on the SDP Security
> Descriptions for Media Streams <draft-ietf-mmusic-sdescriptions-03.txt>.
> Please send any final comments on this draft to the MMUSIC mailing list 
> by 19th March 2004.
> 
> If no substantive technical comments are received by that time, we will
> request that this draft be published as a proposed standard RFC. 
> 
> -- 
> Colin Perkins
> http://csperkins.org/

_______________________________________________
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Cc: Jae-Hwan Kim <kjh1905m@nextreaming.com>,
        "mmusic (E-mail)" <mmusic@ietf.org>
Subject: Re: [MMUSIC] ETag Definition in draft-rfc2326bis-06
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From: Pekka Pessi <Pekka.Pessi@nokia.com>
In-Reply-To: <4060728A.2010606@ericsson.com> (Magnus Westerlund's message of
 "Tue, 23 Mar 2004 18:23:22 +0100")
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Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
>Thanks for spotting this. I need some help with this as I have not a
>complete understanding of how ETag is supposed to work. I understand that
>they are server defined entities that only needs to follow the syntax.

>So what I understand there is need for a RTSP header "ETag:<tag>" that can
>be included in DESCRIBE and SETUP messages to be used to determine what
>content the client actually wants to SETUP or question if the description is
>up-to-date.

I *think* that the idea is when you get an SDP from HTTP server or
in response to DESCRIBE, you also get an a=etag. You then include
the tag in the If-Match header of the SETUP request. If the tag is
valid, the SETUP continues as usual. If the media resource has
changed (or the tag is not valid for any other reason) the server
should return 412. 

Now that ANNOUNCE/RECORD have been removed, I don't think we have
any use of If-Match with DESCRIBE.

According to RFC 2326, 304 is returned when a DESCRIBE or
SETUP has an If-Modified-Since header. The Last-Modified/
If-Modified-Since mechanism is borrowed from HTTP/1.0. The
description of 304 with ETag in bis-06, however, is pasted from
HTTP/1.1.

So, when you get a 304 response with ETag in HTTP/1.1? Lets
suppose you have done GET and got 200 with the resource contents
and ETag: "xyzzy". Later you send GET with If-None-Match: "xyzzy",
if the server has not changed the resource it could respond with
304 and the ETag "xyzzy".

When using the HTTP/1.1 entity tags there can multiple versions
of the same resource, say, Swedish and Finnish web pages. The
server could select a particular version based on the request (in
this case, Accept-Language header in it). It would include a ETag
and Content-Location headers in the response (probably also the
Content-Language). A client (or proxy) could cache all the
different versions. It can ask server if one of them is valid for
a particular request by including all the cached ETags in the
If-None-Match header of the GET request. The 304 response would
then contain the ETag and Content-Location for the correct web
page.

So, I think options for bis-07 are:

1) deprecate If-Modified-Since and introduce If-None-Match
2) ditch ETag header in 304 and continue to use the current mix of
   HTTP/1.0 and HTTP/1.1 mechanisms

>Having a ETag header in the describe response seem to have some
>possibilities to clash with the SDP usage. In session
>descriptions it seems much better to have the ETag in the
>description so one does not have to add it if one for example
>saves the SDP.

Yes. However, I could imagine a "particular session description
language" that does not have etag, so the ETag header would be
needed.

--Pekka

>Jae-Hwan Kim wrote:
>> Dear MMUSICers,
>> Please look at the page 60 mentioning ETag header field. Basically, "304
>> Not modified" response is always supposed to have this
>> header field.
>> However, where is ETag definition in this spec.?
>> In addition, when ETag is used in DESCRIBE response, should a client
>> ignore this field?
>> I found a commercial RTSP server had this field in the DESCRIBE
>> response.
>> TIA/BR, Jae-Hwan
>> _______________________________________________
>> mmusic mailing list
>> mmusic@ietf.org
>> https://www1.ietf.org/mailman/listinfo/mmusic

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Date: Thu, 01 Apr 2004 10:17:45 -0800
From: Andrew Large <Andrew.Large@Sun.COM>
Subject: RE: [MMUSIC] draft-rfc2326-06 comments wrt live streaming
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Please forgive me for jumping into the middle of this, but this area is
of considerable concern for us and I feel compelled to offer my $.02...

First, Magnus, do I correctly capture your assertions?

	1. The RTP-Info header need have only the "seq" or "rtptime"
clauses, but not both.

	2. For live streams, the server can provide "rtptime" values
that are meaningful to a client, even in the absence of a meaningful
Range header (e.g., when Range is "npt=now-").

	3. The server should always have access to stream
synchronization information, so having the raw data necessary for
generating the "rtptime" values shouldn't be an issue.

If I have correctly captured your position above, I have a few
questions/comments:

	a. [I apologize for not being on the list early enough to read
the entire dialog] Are you conceding the point that the server may not
know the sequence numbers of the first packets in each stream?

	b. When one populates the "seq" clause, how "close" does it have
to be to the actual value?  If it is acceptable to be off by some
amount, could you provide some concrete parameters?  Either by time or
by sequence number value?  (e.g., packet with given "seq" value must
have RTP timestamp within 500ms of that of the first packet actually
sent in that stream after the PLAY reply; or the first packet's sequence
number must differ by no more than N).

	c. When one populates the "Range" header with a real value, what
constraints are there (if any) on the relationship between that value
and the first packets actually sent in each stream?  For example,
several servers simply report the value requested by the client and then
start their streams with packets whose timestamps fall <= the value in
the range header.  In this case, the "rtptime" clause values in the
RTP-Info header (which must map to the Range header) will be >= the RTP
timestamps of the first packets actually received by the client.  Note
that this behavior is preferable to support (among other things), very
long duration media samples.  [If the reason for this is not clear, I'd
be happy to follow up with an example]

	d.  As a follow-up to [c], what constraints would there be on
the "implicit" value for the Range header in a live stream where it is
reported as "npt=now-", assuming that "rtptime" clauses are present in
the RTP-Info header.  For example, could the server pick some point,
say, 5 minutes in the past as the synchronization point it will use to
generate the "rtptime" clause values?

	e. One of the most common current scenarios for distributing a
live broadcast is to have a "broadcaster" component that is *not* an
RTSP server -- it simply generates an RTP stream to a pre-defined set of
broadcast or unicast UDP ports and produces a ".sdp" file that describes
the broadcast.  The SDP file is staged in the media root of a full
RTSP/RTP streaming server.  When a client requests a session from the
".sdp", the server parses the *.sdp (which it may only become aware of
at the point of the first client request) to figure out where it should
"listen" for the streams (there is no RTSP dialog between the server and
the broadcaster).  In such a case, there is simply no way to guarantee
that the server knows *either* sequence number *or* synchronization
information for the streams before it needs to respond to the client.
In theory, this is disallowed by the new draft of the spec, is it not?
In practice, we (and others) have been using this method for years, and
it works just fine, so long as the client does not insist on having the
RTP-Info header.  The client simply "resyncs" itself when the first SRs
actually arrive in the data stream.  I realize that we're working on a
new mechanism  (the relay work) that would have the broadcaster be an
RTSP client and thus pass along the synchronization information when it
"announces" the availability of the presentation, but do you really
think we should be breaking an existing facility at the same time as we
are introducing its replacement?

	f. Two editorial notes on the draft:

		+ The separation of header syntax and sub-value
discussion (ala NPT discussion) from the header descriptions is very
confusing.  It makes it quite difficult to get the full story on any one
attribute of the specification (you end up having to simply search for
keywords in the document, which is complicated in the case of RTP-Info
because it shows up in several examples).

		+ Maybe I just don't know how to parse the syntax
properly, but it actually looks to me like the syntax asserts that the
"seq" and "rtptime" values are in fact mutually exclusive -- you can
only have one or the other (or is 1* supposed to be interpreted as "one
or more"?)


-- 
Andrew Large
andrew.large@sun.com

> -----Original Message-----
> From: mmusic-admin@ietf.org [mailto:mmusic-admin@ietf.org] On 
> Behalf Of Magnus Westerlund
> Sent: Thursday, April 01, 2004 1:11 AM
> To: Geetha.Srikantan@sun.com
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] draft-rfc2326-06 comments wrt live streaming
> 
> 
> Hi Geetha,
> 
> See inline.
> 
> Geetha Srikantan wrote:
> > Magnus Westerlund wrote:
> >>> (a) That all live sources are RTSP-aware?
> >>
> >>
> >>
> >> No, but that the there is a root-distributor that is a 
> RTSP server. 
> >> The root RTSP server can be synchronized using either standard RTP 
> >> means or any proprietary solution as this interface is not 
> >> standardized. However it will be this RTSP root server 
> that will be 
> >> responsible giving the stream a NPT time, unless that is 
> also provided 
> >> through the interface between RTSP and the live source.
> > 
> > 
> > I was assuming the same thing.
> > 
> >>>
> >>> (b) The server could maintain it's own RTP clock for each 
> stream by 
> >>> observing
> >>>     the RTP timestamps of packets on each stream?
> >>>
> >>
> >> I don't see how that this would be necessary. If one has a 
> RTSP server 
> >> that receives a RTP + RTCP media stream. It will after listened to 
> >> this for a few seconds received all necessary synchronization 
> >> information. If this is not the root server and is only a 
> relay it can 
> >> use RTSP to receive the synchronization information for the media 
> >> streams it is forwarding directly in RTSP.
> >>
> > This is an important point of concern.
> > I agree that an RTSP server, after it has received RTCP SR for each 
> > stream, can map TS to NPT.
> > There are some issues though:
> > 
> > 1. The server may not see RTCP SRs for all streams in the 
> RTSP session 
> > for several seconds,
> >   because of (a) staggered transmission of RTCP SRs of each 
> stream, (b) 
> > lost SRs.
> >   Since RTCP transmission intervals can be 5-10 seconds, 
> this could be 
> > 10-20 seconds or so.
> > 
> > Now the server could either
> > (a) deny all client requests for the live session until it 
> receives RTCP 
> > SRs for all streams - even
> > though the server has RTP packets for all streams. This 
> implies that 
> > clients won't have access
> > to the live stream for several seconds after the session 
> was requested.
> > OR
> > (b) respond to client requests without the RTP-Info header in PLAY 
> > responses.
> > 
> > If (a) and (b) are accepted as likely scenarios, then the 
> clients cannot 
> > expect to see a RTP-Info header
> > in PLAY responses, and should be able to handle the absence of the 
> > RTP-Info header.
> 
> However I don't see it likely that a relay RTSP server would need to 
> rely on RTCP to establish the NPT binding. A relay will use the live 
> sessions root RTSP sever or another RTSP relay. As it is 
> using RTSP it 
> can receive the RTP-Info and Range header which gives it all 
> necessary 
> information, unless things are sent using multiple SSRC, in 
> which case 
> RTP-Info at the delivery server still can't be used for synch. If the 
> relay is slow on responding, I would still recommend that one 
> wait for 
> the synch info. Delivering things without synch does not make people 
> happy. In conclusion I don't see a good reason why the 
> delivery server 
> would not have access to the synchronization information.
> 
> 
> > 
> > 2. The server may not know when the live session started, 
> for example if 
> > it joins an ongoing
> >    live session.
> > One option is for the server to assume NPT=0 at the time it 
> receives RTP 
> > packets from the
> > live source, and use that in synchronization mappings. So, 
> the Range 
> > header would have
> > this server-interpreted version of NPT.
> > 
> 
> But if the delivery server uses RTSP to request the media, it will 
> receive this information.
> 
> I would also like to point out that any delay of getting the synch 
> information will only be present for the first user of a 
> certain stream. 
> Any user requesting a stream already delivered to the delivery server 
> will be able to get correct information directly.
> 
> >>> (c) The live source's transmission schedule for packets 
> follows the 
> >>> RTP timestamp clock?
> >>>     That is, each succeeding packet's RTP timestamp will 
> be equal to 
> >>> or greater than all previous
> >>>     packets, for that stream (modulo rollover)?
> >>>
> >>
> >> I am only assuming that this is a normal RTP media stream. 
> Thus it can 
> >> have backwards jump and other non-monotonic behaviour in 
> the timestamp 
> >> field. However I do expect that the RTP packets are sent 
> in order, but 
> >> not received by any relay in order. Thus a relay will be 
> required to 
> >> take certain care against reordering in the RTP-Info 
> sequence number 
> >> field.
> > 
> > 
> > Reordering and lost packets make it difficult to populate 
> the sequence 
> > number field of RTP-Info headers
> > for live streams. This is not the case for stored content. 
> I think this 
> > is a point worth noting for live streaming.
> > If the server has to guess about the next packet it is 
> going to send, it 
> > can't be a good thing.
> > 
> 
> I don't see how reordering creates any real problem. When a new user 
> joins one takes the last sequence number received. Then in 
> the RTP-Info 
> one includes this number plus one. Then one forwards the next 
> packet, it 
> this was reordered an is in fact an older packet than what 
> one claims in 
> the RTP-Info, the client is able to detect that this packet 
> is prior to 
> the start of the stream. If there is packet loss or reorder 
> that results 
> in that the first packet that arrives to the client has 
> sequence number 
> much higher than the indicated one, it is okay. Because the 
> only thing 
> the RTP sequence number in the RTP-Info indicates is from 
> what sequence 
> number the request media stream is delivered.
> 
> 
> 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
> 
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 


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Subject: [MMUSIC] Call for Papers: BASENETS '04
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Apologies if you receive multiple copies of this call!


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

                         CALL FOR PAPERS

      ********************************************************    
      *                                                      *
      *  1st Workshop on Broadband Advanced Sensor Networks  *
      *           (co-located with BroadNets 04)             *
      *             San Jose, CA, October 2004               *
      *                                                      *
      *                    BASENETS 04                       *
      *                                                      *
      ********************************************************
      *                                                      *
      *             http://www.basenets.org                  *
      *                                                      *
      ********************************************************

SCOPE

The emergence and adoption of broadband wireless access standards,
such as 802.16, 802.16a and 802.11a/g, offers exciting new
possibilities for high-bandwidth data communication over the "last
mile". A significant use of such high bandwidth may lie in the use of
a variety of multimodal sensors for monitoring and control of diverse
geographic areas. Such sensors may be used for remote video/audio
surveillance and active monitoring of telemetry data in city blocks,
highways, airports, hotels, campuses and a variety of other indoor and
outdoor environments.

This workshop is intended for the presentation and discussion of novel
architectures, algorithms and applications of sensor networks based on
the use of wireless broadband channels. Areas of interest include
problems related to energy efficiency, dynamic control and reliability
of such sensor network deployments, especially under the expected
resource constraints on the sensor devices. Of particular interest are
visions, practical case studies, demonstrations or pilots of
innovative techniques for building sensor networks using either
broadband local or metropolitan wireless networks.

Topics of interest include, but are not limited to:
* Multimedia Interfaces for Wireless Sensors
* Broadband vs. Narrowband Wireless Sensing
* Video and Audio Sensing Applications over Broadband Networks
* Distributed Multimedia Compression/Aggregation for Broadband Sensor Networks
* Scalability and Performance Bottlenecks for Wide-Area Broadband Sensing
* Adaptive Configuration of Broadband Sensor Networks
* Power and Spectrum Management Issues for Broadband Sensing
* Architectures for Broadband Sensor Networks
* Applications of Wireless Broadband Sensor Networks
* Security and Privacy for Broadband Sensor Networks
* Broadband Sensor Networks with Mobile Nodes
* Heterogeneous Wired-Wireless Broadband Sensor Networks

SUBMISSION

Papers should be written in English, should not exceed 20
double-spaced, single-column pages, including figures and references,
and should include an abstract, not to exceed 250 words.  Papers
should be submitted in PDF format via the EDAS system
(http://edas.info). All papers will be peer reviewed.

The best papers of this workshop will be considered for possible
publication in the ACM Transactions on Sensor Networks
(http://www.acm.org/tosn/).

IMPORTANT DATES

Full paper due:               June 1, 2004
Notification of acceptance:   July 12, 2004
Camera-ready copy:            Aug. 9, 2004

Workshop Chairs:
W. Heinzelman (U. of Rochester)
B. Krishnamachari (U. of Southern California)

Program Committee:
N. Abu-Ghazaleh (Binghamton U.)
J. Apostolopoulos (HP Labs)
A. Bestavros (Boston U.)
A. Campbell (Columbia U.)
M. El Zarki (U. of California at Irvine)
W.-C. Feng (OGI)
S. Gupta (Arizona State U.)
R. Han (U. of Colorado at Boulder)
A. Helmy (U. of Southern California)
S. Jha (U. of New South Wales)
H. Karl (TU Berlin)
M. Liu (U. of Michigan)
B. Nath (Rutgers U.)
M. Papadopouli (U. of North Carolina)
A. Safwat (Queens U.)
I. Stojmenovic (U. of Ottawa)
U. Tureli (Stevens U.)
M. van der Schaar (U. of California at Davis)
P. Varshney (Syracuse U.)

Publicity Chair:
R. Han (U. of Colorado at Boulder)
H. Karl (Technical University Berlin)


For more details, please visit http://www.basenets.org 
or contact basenets04-chairs@ece.rochester.edu


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        Geetha.Srikantan@Sun.COM
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Subject: RE: [MMUSIC] draft-rfc2326-06 comments wrt live streaming
Date: Wed, 7 Apr 2004 14:17:08 -0700 
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Hi Geetha and Magnus:

I have been reading this thread of discussion, and have some feedback for
your reference.

See inline.

Cheers,
-Thomas

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
Sent: Thursday, April 01, 2004 1:44 AM
To: Geetha.Srikantan@Sun.COM
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] draft-rfc2326-06 comments wrt live streaming


Hi Geetha,

See inline

Geetha Srikantan wrote:

> Magnus,
> 
> I see your point of view on this, since you clarified your assumptions.
> 
> In previous offline discussions there were discussions about how a RTSP 
> server would
> maintain timing info for live streams. My confusion was centered around 
> that.
> 
> I believe RTCP SR-based synchronization is possible, as you pointed 
> out..however it does
> have some issues, as mentioned previously.
> 
> Replying inline to some of the points in this mesage...
> 
[snip]


>>> Section 14.40, page 91:
>>> ssrc:
>>> The transport header in the server's response cannot populate the ssrc
>>> for each stream, accurately in certain scenarios.
>>> The source of the live stream may not be RTSP-aware, and the server 
>>> has to rely on RTP/RTCP to determine ssrc for each stream.
>>> The server may have joined a multicast live session and receives, 
>>> say, one of the streams.
>>> It also receives SETUP request from a client and needs to respond to 
>>> SETUP from a client,
>>> before it sees the first packet of each stream.
>>> Dropping the ssrc parameter from the Transport header in the SETUP 
>>> response would
>>> simplify this case.
>>
>>
>>
>> It is possible to skip the SSRC, however not recommend as it indicates 
>> for which SSRC the synchronization information in RTP-Info applies to. 
>> Thus I would recommend that server awaits at least the first RTP 
>> packet to determine what SSRC is used for media distribution.
> 
> 
> I am not convinced that the user experience is good when a server 
> refuses clients access to live sessions until
> it has access to all the stream synchronization information. If the 
> server were to send RTP and RTCP packets,
> without the RTP-Info or SSRC in RTSP responses,

[tz] Geetha: SSRC serves another purpose besides stream synchronization, and
therefore without SSRC some players may not function well. Hence I second
Magnus recommendation: server waits for the first RTP to get SSRC -- the
delay is not that bad, right? Here are some more details: in our player
(PacketVideo player) implementation, we use SSRC to filter out spurious
packets on the RTP and RTCP receive port. Here is how it works: we record
the SSRC in RTSP SETUP response; after we change to PLAYING state, we obtain
RTP packets from our RTP receive port. We then filter the RTP packets by the
expected SSRC.
[tz] Here is how it helps: in wireless environment, it is not uncommon to
have connections dropped due to bad channel conditions. Many a time player
connects back and start a new session, but the OS happens to assign the same
RTP receive port number. The server is not yet aware that previous session
has been dropped (it needs to go through the timeout period waiting for RRs
to come), and is sending both old session's RTP and new session's to the
player. Filtering by SSRC helps the player to ignore unwanted RTP packets.

> the client has an 
> opportunity to start decoding each stream
> independently and perform inter-stream synchronization as soon as the 
> client receivess one RTCP SR per stream.
> There is at least some media being rendered at the client, even if it is 
> not lip-sync'd etc.
> 

As I wrote in the other mail I don't think this will occur frequently. 
In a well designed system using RTSP the synch information will be 
distributed using TCP, and will basically arrive as soon as possible. If 
there are problems delivering the RTSP response.

A server waiting for the synchronization information can also use a 1xx 
code to tell that it is in fact waiting for the sync info.

I would like to get some feedback from client implementors, it they 
really are willing start without synch and then try to do graceful fix 
of the synchronization when the information arrives. 

[tz] In the early days we (PacketVideo) did have a version of player that
can start with neither RTP-Info information nor SR information. It assumes
the first received RTP packet in video track is in synch with the first RTP
packet in audio track, which is invalid if there are packets lost before the
first one received, or that the sender do not send them in sych. So I am
inclined to say that player can do it if the user can live with the degraded
user experience in the beginning. Note that the player always has to check
sychronization upon receiving  timing information in SRs.  

[tz] Now a more interesting problem is for random positioning (as you
touched upon below). It is well known that RTP-Info is required to handle
PLAY after PAUSE (see a discussion thread a few months ago in this mailing
list). Fortunately I have not seen customers asking for PAUSE in live
streaming. But others might have. I don't know.



Also without giving 
the RTP-Info TS + Range it will be impossible to present accurate 
session position information. A client will only be able to calculate 
time since media delivery started.

I think a lot of these discussion concerns issues that should be 
discussed in your streaming relay draft to achieve good performance and 
avoid these issues you are so concerned with. From a client perspective 
, I haven't seen a lot of need for clarifications. However I can look on 
how to further improve the discussion in the RTP appendix on synch. 
Otherwise I would recommend if you have specific concerns to present 
proposals for new text.

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: Wed, 07 Apr 2004 17:34:56 -0700
From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
Subject: Re: [MMUSIC] draft-rfc2326-06 comments wrt live streaming
In-reply-to: <72263E8E8622D611975C0002B32C19D805A180A1@misty.packetvideo.com>
To: Thomas Zeng <zeng@pvnetsolutions.com>, mmusic@ietf.org
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hi Thomas,

Thanks for the feedback and notes about client side issues.
Replying inline...

Thomas Zeng wrote:

>>>It is possible to skip the SSRC, however not recommend as it indicates 
>>>for which SSRC the synchronization information in RTP-Info applies to. 
>>>Thus I would recommend that server awaits at least the first RTP 
>>>packet to determine what SSRC is used for media distribution.
>>>      
>>>
>>I am not convinced that the user experience is good when a server 
>>refuses clients access to live sessions until
>>it has access to all the stream synchronization information. If the 
>>server were to send RTP and RTCP packets,
>>without the RTP-Info or SSRC in RTSP responses,
>>    
>>
>
>[tz] Geetha: SSRC serves another purpose besides stream synchronization, and
>therefore without SSRC some players may not function well. Hence I second
>Magnus recommendation: server waits for the first RTP to get SSRC -- the
>delay is not that bad, right? Here are some more details: in our player
>(PacketVideo player) implementation, we use SSRC to filter out spurious
>packets on the RTP and RTCP receive port. Here is how it works: we record
>the SSRC in RTSP SETUP response; after we change to PLAYING state, we obtain
>RTP packets from our RTP receive port. We then filter the RTP packets by the
>expected SSRC.
>[tz] Here is how it helps: in wireless environment, it is not uncommon to
>have connections dropped due to bad channel conditions. Many a time player
>connects back and start a new session, but the OS happens to assign the same
>RTP receive port number. The server is not yet aware that previous session
>has been dropped (it needs to go through the timeout period waiting for RRs
>to come), and is sending both old session's RTP and new session's to the
>player. Filtering by SSRC helps the player to ignore unwanted RTP packets.
>  
>

This looks like a good use case for having the SSRC in the SETUP 
response's Transport header.
Thanks for providing the details.
I'm less concerned about delays caused by the SETUP response, than by 
PLAY response.

>>the client has an 
>>opportunity to start decoding each stream
>>independently and perform inter-stream synchronization as soon as the 
>>client receivess one RTCP SR per stream.
>>There is at least some media being rendered at the client, even if it is 
>>not lip-sync'd etc.
>>
>>    
>>
>
>As I wrote in the other mail I don't think this will occur frequently. 
>In a well designed system using RTSP the synch information will be 
>distributed using TCP, and will basically arrive as soon as possible. If 
>there are problems delivering the RTSP response.
>
>A server waiting for the synchronization information can also use a 1xx 
>code to tell that it is in fact waiting for the sync info.
>
>I would like to get some feedback from client implementors, it they 
>really are willing start without synch and then try to do graceful fix 
>of the synchronization when the information arrives. 
>
>[tz] In the early days we (PacketVideo) did have a version of player that
>can start with neither RTP-Info information nor SR information. It assumes
>the first received RTP packet in video track is in synch with the first RTP
>packet in audio track, which is invalid if there are packets lost before the
>first one received, or that the sender do not send them in sych. So I am
>inclined to say that player can do it if the user can live with the degraded
>user experience in the beginning. Note that the player always has to check
>sychronization upon receiving  timing information in SRs.  
>  
>
Ok..

>[tz] Now a more interesting problem is for random positioning (as you
>touched upon below). It is well known that RTP-Info is required to handle
>PLAY after PAUSE (see a discussion thread a few months ago in this mailing
>list). Fortunately I have not seen customers asking for PAUSE in live
>streaming. But others might have. I don't know.
>
>  
>
Live sessions can't be PAUSED, right? So this shouldn't be an issue..

geetha


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Date: Wed, 07 Apr 2004 16:53:40 -0700
From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
Subject: Re: [MMUSIC] draft-rfc2326-06 comments wrt live streaming
In-reply-to: <406BDCAE.8090205@ericsson.com>
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hi Magnus,

Replying inline..

Magnus Westerlund wrote:

>> This is an important point of concern.
>> I agree that an RTSP server, after it has received RTCP SR for each 
>> stream, can map TS to NPT.
>> There are some issues though:
>>
>> 1. The server may not see RTCP SRs for all streams in the RTSP 
>> session for several seconds,
>>   because of (a) staggered transmission of RTCP SRs of each stream, 
>> (b) lost SRs.
>>   Since RTCP transmission intervals can be 5-10 seconds, this could 
>> be 10-20 seconds or so.
>>
>> Now the server could either
>> (a) deny all client requests for the live session until it receives 
>> RTCP SRs for all streams - even
>> though the server has RTP packets for all streams. This implies that 
>> clients won't have access
>> to the live stream for several seconds after the session was requested.
>> OR
>> (b) respond to client requests without the RTP-Info header in PLAY 
>> responses.
>>
>> If (a) and (b) are accepted as likely scenarios, then the clients 
>> cannot expect to see a RTP-Info header
>> in PLAY responses, and should be able to handle the absence of the 
>> RTP-Info header.
>
>
> However I don't see it likely that a relay RTSP server would need to 
> rely on RTCP to establish the NPT binding. A relay will use the live 
> sessions root RTSP sever or another RTSP relay. As it is using RTSP it 
> can receive the RTP-Info and Range header which gives it all necessary 
> information, unless things are sent using multiple SSRC, in which case 
> RTP-Info at the delivery server still can't be used for synch. If the 
> relay is slow on responding, I would still recommend that one wait for 
> the synch info. Delivering things without synch does not make people 
> happy. In conclusion I don't see a good reason why the delivery server 
> would not have access to the synchronization information.
>
Synchronization info is not always available when the origin/tier 1 RTSP 
server *starts* receiving
RTP streams from the live source. If the origin server is not to handle 
any requests for this live session,
then relay servers requesting this content will be stalled. That was my 
point.

I realize that when all servers have synchronization info available 
based on RTCP SRs or
RTP-Info/Range headers, a server can respond with the rtptime value for 
current wallclock time.

>> 2. The server may not know when the live session started, for example 
>> if it joins an ongoing
>>    live session.
>> One option is for the server to assume NPT=0 at the time it receives 
>> RTP packets from the
>> live source, and use that in synchronization mappings. So, the Range 
>> header would have
>> this server-interpreted version of NPT.
>>
>
> But if the delivery server uses RTSP to request the media, it will 
> receive this information.
>
> I would also like to point out that any delay of getting the synch 
> information will only be present for the first user of a certain 
> stream. Any user requesting a stream already delivered to the delivery 
> server will be able to get correct information directly.

 From my experience, in live sessions, there are hundreds of clients 
trying to connect and receive the live session
within a few seconds of each other. So if the streaming server joins a 
live session and waits
to receive synchronization info before it responds to any client, all of 
these clients
would have to wait.
It's a tradeoff between (a) clients waiting for RTSP PLAY response, (b) 
clients receiving PLAY response without
synchronization info, decoding RTP streams right away and performing 
delayed synchronization.
Many of us have opted for (b) in implementations (as Andrew Large 
mentions in his email).

>> Reordering and lost packets make it difficult to populate the 
>> sequence number field of RTP-Info headers
>> for live streams. This is not the case for stored content. I think 
>> this is a point worth noting for live streaming.
>> If the server has to guess about the next packet it is going to send, 
>> it can't be a good thing.
>>
>
> I don't see how reordering creates any real problem. When a new user 
> joins one takes the last sequence number received. Then in the 
> RTP-Info one includes this number plus one. Then one forwards the next 
> packet, it this was reordered an is in fact an older packet than what 
> one claims in the RTP-Info, the client is able to detect that this 
> packet is prior to the start of the stream. If there is packet loss or 
> reorder that results in that the first packet that arrives to the 
> client has sequence number much higher than the indicated one, it is 
> okay. Because the only thing the RTP sequence number in the RTP-Info 
> indicates is from what sequence number the request media stream is 
> delivered.

I can see that this would work, however it seems like you are 
recommending that a server 'guess' about
the sequence number of the next packet it sends, since the server cannot 
accurately pinpoint the
exact sequence number of the next packet it sends. It might be 
preferable to interpret 'seqno ' as the
sequence number of the packet last sent by the server, so the client 
knows to look for sequence numbers higher than that.

> Cheers
>
> Magnus Westerlund


geetha


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Date: Wed, 07 Apr 2004 17:13:36 -0700
From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
Subject: Re: [MMUSIC] draft-rfc2326-06 comments wrt live streaming
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hi Magnus,

Thanks for the pointer to the ABNF - yes it clarifies that RTP-Info can have
either seqno *or* rtptime.
I guess we agree that Range: npt=now- is acceptable in the PLAY response,
when accurate NPT value is not known.
My comment was about missing packets impacting seqno in the RTP-Info header
is covered in the previous email, so I'll skip it here.

Replying inline to the rest.


Magnus Westerlund wrote:

>> I am not convinced that the user experience is good when a server 
>> refuses clients access to live sessions until
>> it has access to all the stream synchronization information. If the 
>> server were to send RTP and RTCP packets,
>> without the RTP-Info or SSRC in RTSP responses, the client has an 
>> opportunity to start decoding each stream
>> independently and perform inter-stream synchronization as soon as the 
>> client receivess one RTCP SR per stream.
>> There is at least some media being rendered at the client, even if it 
>> is not lip-sync'd etc.
>>
>
> As I wrote in the other mail I don't think this will occur frequently. 
> In a well designed system using RTSP the synch information will be 
> distributed using TCP, and will basically arrive as soon as possible. 
> If there are problems delivering the RTSP response.
>
> A server waiting for the synchronization information can also use a 
> 1xx code to tell that it is in fact waiting for the sync info.
>
> I would like to get some feedback from client implementors, it they 
> really are willing start without synch and then try to do graceful fix 
> of the synchronization when the information arrives. Also without 
> giving the RTP-Info TS + Range it will be impossible to present 
> accurate session position information. A client will only be able to 
> calculate time since media delivery started.

As I mentioned in the previous email, it's a tradeoff between the client 
waiting for the RTSP handshake to complete and the client waiting for 
RTCP SRs, while it plays out individual streams. In the first case no 
media is rendered at the client until it completes the RTSP handshake 
and receives RTP.

> I think a lot of these discussion concerns issues that should be 
> discussed in your streaming relay draft to achieve good performance 
> and avoid these issues you are so concerned with. From a client 
> perspective , I haven't seen a lot of need for clarifications. However 
> I can look on how to further improve the discussion in the RTP 
> appendix on synch. Otherwise I would recommend if you have specific 
> concerns to present proposals for new text.

We are working on a revision of the relay draft and will cover these 
issues there. However, I'd like to
point out that the live streaming issues do occur with or without 
relays, so they need to be addressed
in the main spec. Sure, I will work on proposing some text to clarify 
these issues and send it to the alias.

> Cheers
>
> Magnus
>
thanks
geetha


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Andrew Large wrote:

> Please forgive me for jumping into the middle of this, but this area is
> of considerable concern for us and I feel compelled to offer my $.02...
> 
> First, Magnus, do I correctly capture your assertions?
> 
> 	1. The RTP-Info header need have only the "seq" or "rtptime"
> clauses, but not both.
> 

Yes, the header does not require both, however unless rtptime is used, 
synchronization is not achieved with this mechanism.

> 	2. For live streams, the server can provide "rtptime" values
> that are meaningful to a client, even in the absence of a meaningful
> Range header (e.g., when Range is "npt=now-").

Yes, however I have forgotten to point out that the case when using 
"npt=now-" will only achieve inter media synchronization when the media 
streams are under aggregated control.

> 
> 	3. The server should always have access to stream
> synchronization information, so having the raw data necessary for
> generating the "rtptime" values shouldn't be an issue.
> 

Yes.

> If I have correctly captured your position above, I have a few
> questions/comments:
> 
> 	a. [I apologize for not being on the list early enough to read
> the entire dialog] Are you conceding the point that the server may not
> know the sequence numbers of the first packets in each stream?

Yes, and this is the case for the first PLAY for a live stream to 
server. And in this case the lack of sequence number will not effect the 
client.

> 
> 	b. When one populates the "seq" clause, how "close" does it have
> to be to the actual value?  If it is acceptable to be off by some
> amount, could you provide some concrete parameters?  Either by time or
> by sequence number value?  (e.g., packet with given "seq" value must
> have RTP timestamp within 500ms of that of the first packet actually
> sent in that stream after the PLAY reply; or the first packet's sequence
> number must differ by no more than N).
> 

The point of sequence number in RTP-Info is to indicate the first RTP 
packet for which this synchronization information is valid. So it is not 
a matter of being able to be of with a certain amount. It is a matter of 
indicating from which packet the synch info is okay. So in a case where 
a server gives seq=C then the following observation can be done.

When the client receives an RTP packet with sequence number X the 
following applies:

if X<C:
This packet does not use the synchronization received in the RTP-Info + 
Range. Synchronization must be based on previous synch info. It might 
even be so that the most appropriate thing is to suppress this packet.

If X>=C:
This packet uses the synch info.

> 	c. When one populates the "Range" header with a real value, what
> constraints are there (if any) on the relationship between that value
> and the first packets actually sent in each stream?  For example,
> several servers simply report the value requested by the client and then
> start their streams with packets whose timestamps fall <= the value in
> the range header.  In this case, the "rtptime" clause values in the
> RTP-Info header (which must map to the Range header) will be >= the RTP
> timestamps of the first packets actually received by the client.  Note
> that this behavior is preferable to support (among other things), very
> long duration media samples.  [If the reason for this is not clear, I'd
> be happy to follow up with an example]
> 

 From draft version 6, chapter 11.4 PLAY

   "For a on-demand stream, the server MUST reply with the actual range
    that will be played back. This may differ from the requested range if
    alignment of the requested range to valid frame boundaries is
    required for the media source. If no range is specified in the
    request, the start position SHALL still be returned in the reply. If
    the medias that are part of an aggregate has different lengths, the
    PLAY request SHALL be performed as long as the given range is valid
    for any media, for example the longest media. Media will be sent
    whenever it is available for the given play-out point."

So I think it is clear that one should report what time interval one is 
actually playing. However it is not likely that one will be able to for 
all media stream send out RTP packets which matches that time exactly. 
Therefore the capability you are indicating is required to be supported 
by a client. However it might be necessary to further clarify this. An 
example like the following may be worth adding.

C->S:
PLAY
Range:npt=13.21-20

S->C
200 Ok
Range:npt=13.160-20
RTP-Info: url=audio;seq=50;rtptime=50000,url=video;seq=12;rtptime=123400

Then the RTP media arrives:

Audio (8kHz RTP timestamp rate):
S -> C: RTP packet - seq = 50, rtptime = 49830, NPT time = 13.15125 s

Video (90kHz RTP timestamp rate):
S -> C: RTP packet - seq = 10, rtptime = 1234000, NPT time= 13.160 s


So this indicate that it is expect that at least one media will have the 
same time as the indicate range in the response. However the client will 
need to handle that some media is prior than the requested and the 
indicated start time. What is currently not clear is if a server is 
allowed to send a audio packet with the following characteristics:

Audio (8kHz RTP timestamp rate):
S -> C: RTP packet - seq = 50, rtptime = 50100, NPT time = 13.1725 s

In other words a packet that is later than the indicated play time but 
prior to the request time.

Feedback on this is desired.

> 	d.  As a follow-up to [c], what constraints would there be on
> the "implicit" value for the Range header in a live stream where it is
> reported as "npt=now-", assuming that "rtptime" clauses are present in
> the RTP-Info header.  For example, could the server pick some point,
> say, 5 minutes in the past as the synchronization point it will use to
> generate the "rtptime" clause values?

In theory it could, but it would be less nice behaviour, but a client 
could experience this and should be capable of handling, because it is 
almost equal to a session where one losses 5 minutes of data. If you 
have the time 5 minutes ago, you only need to perform a multiplication + 
an addition per media to give a time value for the current time.

> 
> 	e. One of the most common current scenarios for distributing a
> live broadcast is to have a "broadcaster" component that is *not* an
> RTSP server -- it simply generates an RTP stream to a pre-defined set of
> broadcast or unicast UDP ports and produces a ".sdp" file that describes
> the broadcast.  The SDP file is staged in the media root of a full
> RTSP/RTP streaming server.  When a client requests a session from the
> ".sdp", the server parses the *.sdp (which it may only become aware of
> at the point of the first client request) to figure out where it should
> "listen" for the streams (there is no RTSP dialog between the server and
> the broadcaster).  In such a case, there is simply no way to guarantee
> that the server knows *either* sequence number *or* synchronization
> information for the streams before it needs to respond to the client.
> In theory, this is disallowed by the new draft of the spec, is it not?
> In practice, we (and others) have been using this method for years, and
> it works just fine, so long as the client does not insist on having the
> RTP-Info header.  The client simply "resyncs" itself when the first SRs
> actually arrive in the data stream.  I realize that we're working on a
> new mechanism  (the relay work) that would have the broadcaster be an
> RTSP client and thus pass along the synchronization information when it
> "announces" the availability of the presentation, but do you really
> think we should be breaking an existing facility at the same time as we
> are introducing its replacement?

Just to better understand the case: You are saying that the media root, 
is receiving the RTP stream from the broadcaster component, however it 
is totally ignoring it until a client requests it from the server?

If the server was not ignoring the stream, and instead parsed the RTCP 
information it would have the information.

Secondly, in the last revision we reduced the requirement on having 
RTP-Info from a MUST to a SHOULD, the current text says:

14.33 RTP-Info

    The RTP-Info response-header field is used to set RTP-specific
    parameters in the PLAY response. These parameters correspond to the
    synchronization source identified by the ssrc parameter of the
    Transport response header in the SETUP reponse. For streams using RTP
    as transport protocol the RTP-Info header SHOULD be part of a 200
    response to PLAY.


         The exclusion of the RTP-Info in a PLAY response for RTP
         transported media will result in that a client needs to
         synchronize the media streams using RTCP. This may have
         negative impact as the RTCP can be lost, and does not need
         to be particulary timely in their arrival. Also
         functionality as informing the client from which packet a
         seek has occurred is affected.


So the issue is simply if anyone is protesting against having this 
requirement on clients to handle media protocol (RTCP) for synch.


> 
> 	f. Two editorial notes on the draft:
> 
> 		+ The separation of header syntax and sub-value
> discussion (ala NPT discussion) from the header descriptions is very
> confusing.  It makes it quite difficult to get the full story on any one
> attribute of the specification (you end up having to simply search for
> keywords in the document, which is complicated in the case of RTP-Info
> because it shows up in several examples).

I think the main problem currently is that all headers description is 
lacking a bit of extra text to clarify syntax related questions. However 
I am not certain that I correctly understand you. Could you please 
better explain, and propose how to amend this problem.

> 
> 		+ Maybe I just don't know how to parse the syntax
> properly, but it actually looks to me like the syntax asserts that the
> "seq" and "rtptime" values are in fact mutually exclusive -- you can
> only have one or the other (or is 1* supposed to be interpreted as "one
> or more"?)

1* = one or more.


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: Magnus Westerlund <magnus.westerlund@ericsson.com>
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Subject: Re: [MMUSIC] ETag Definition in draft-rfc2326bis-06
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Thanks Pekka,

See some comments below.

Pekka Pessi wrote:
> Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
> 
>>Thanks for spotting this. I need some help with this as I have not a
>>complete understanding of how ETag is supposed to work. I understand that
>>they are server defined entities that only needs to follow the syntax.
> 
> 
>>So what I understand there is need for a RTSP header "ETag:<tag>" that can
>>be included in DESCRIBE and SETUP messages to be used to determine what
>>content the client actually wants to SETUP or question if the description is
>>up-to-date.
> 
> 
> I *think* that the idea is when you get an SDP from HTTP server or
> in response to DESCRIBE, you also get an a=etag. You then include
> the tag in the If-Match header of the SETUP request. If the tag is
> valid, the SETUP continues as usual. If the media resource has
> changed (or the tag is not valid for any other reason) the server
> should return 412. 
> 
> Now that ANNOUNCE/RECORD have been removed, I don't think we have
> any use of If-Match with DESCRIBE.

I agree that the usage of If-Match may not any longer be useful with 
DESCRIBE. However it is still useful for SETUP, where one can utilize in 
for example redirection to verify that one does SETUP on a resource 
where one has a valid session description.

> 
> According to RFC 2326, 304 is returned when a DESCRIBE or
> SETUP has an If-Modified-Since header. The Last-Modified/
> If-Modified-Since mechanism is borrowed from HTTP/1.0. The
> description of 304 with ETag in bis-06, however, is pasted from
> HTTP/1.1.
> 
> So, when you get a 304 response with ETag in HTTP/1.1? Lets
> suppose you have done GET and got 200 with the resource contents
> and ETag: "xyzzy". Later you send GET with If-None-Match: "xyzzy",
> if the server has not changed the resource it could respond with
> 304 and the ETag "xyzzy".
> 
> When using the HTTP/1.1 entity tags there can multiple versions
> of the same resource, say, Swedish and Finnish web pages. The
> server could select a particular version based on the request (in
> this case, Accept-Language header in it). It would include a ETag
> and Content-Location headers in the response (probably also the
> Content-Language). A client (or proxy) could cache all the
> different versions. It can ask server if one of them is valid for
> a particular request by including all the cached ETags in the
> If-None-Match header of the GET request. The 304 response would
> then contain the ETag and Content-Location for the correct web
> page.
> 
> So, I think options for bis-07 are:
> 
> 1) deprecate If-Modified-Since and introduce If-None-Match
> 2) ditch ETag header in 304 and continue to use the current mix of
>    HTTP/1.0 and HTTP/1.1 mechanisms


I don't think there are reasons to deprecated If-Modified-Since. It is 
still useful for cases when one likes to use HTTP dates. However adding 
If-None-Match to enabled conditional retrieves with DESCRIBE seems 
beneficial.

> 
> 
>>Having a ETag header in the describe response seem to have some
>>possibilities to clash with the SDP usage. In session
>>descriptions it seems much better to have the ETag in the
>>description so one does not have to add it if one for example
>>saves the SDP.
> 
> 
> Yes. However, I could imagine a "particular session description
> language" that does not have etag, so the ETag header would be
> needed.
> 

Agree, then the proposal is to include also the ETag header in RTSP. 
Comments on this.

Summary of proposed changes:

- Add ETag header.
- Add If-None-Match
- Clarify If-Match
- Clarify usage of ETags in general and their usage.

Comments?

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, 08 Apr 2004 03:46:45 -0700
From: Andrew Large <Andrew.Large@Sun.COM>
Subject: RE: [MMUSIC] draft-rfc2326-06 comments wrt live streaming
In-reply-to: <4072C5C6.5030107@ericsson.com>
To: "'Magnus Westerlund'" <magnus.westerlund@ericsson.com>
Cc: geetha.srikantan@Sun.COM, mmusic@ietf.org
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inline

-- 
Andrew Large
andrew.large@sun.com

> -----Original Message-----
> From: Magnus Westerlund [mailto:magnus.westerlund@ericsson.com] 
> Sent: Tuesday, April 06, 2004 7:59 AM
> To: Andrew Large
> Cc: geetha.srikantan@sun.com; mmusic@ietf.org
> Subject: Re: [MMUSIC] draft-rfc2326-06 comments wrt live streaming
> 
>
[snip]

> 
> > If I have correctly captured your position above, I have a few
> > questions/comments:
> > 
> > 	a. [I apologize for not being on the list early enough to read
> > the entire dialog] Are you conceding the point that the 
> server may not
> > know the sequence numbers of the first packets in each stream?
> 
> Yes, and this is the case for the first PLAY for a live stream to 
> server. And in this case the lack of sequence number will not 
> effect the 
> client.
> 
Okay, fair enough.  This can then probably be stated as "if you've got
it, you should send it; if you don't, don't worry about it".  One would
assume that a stream that was paused and resumed would have sequence
info.


> > 
> > 	b. When one populates the "seq" clause, how "close" does it have
> > to be to the actual value?  If it is acceptable to be off by some
> > amount, could you provide some concrete parameters?  Either 
> by time or
> > by sequence number value?  (e.g., packet with given "seq" value must
> > have RTP timestamp within 500ms of that of the first packet actually
> > sent in that stream after the PLAY reply; or the first 
> packet's sequence
> > number must differ by no more than N).
> > 
> 
> The point of sequence number in RTP-Info is to indicate the first RTP 
> packet for which this synchronization information is valid. 
> So it is not 
> a matter of being able to be of with a certain amount. It is 
> a matter of 
> indicating from which packet the synch info is okay. So in a 
> case where 
> a server gives seq=C then the following observation can be done.
> 
> When the client receives an RTP packet with sequence number X the 
> following applies:
> 
> if X<C:
> This packet does not use the synchronization received in the 
> RTP-Info + 
> Range. Synchronization must be based on previous synch info. It might 
> even be so that the most appropriate thing is to suppress this packet.
> 
> If X>=C:
> This packet uses the synch info.
>
So this sounds like "the seqno you send should be <= that of the first
packet the client receives from you".  So a client must be able to
handle the case where the first packet has a seqno that is maybe 2 or 3
values greater than what was put in the RTP-Info header, right?  If so,
that solves one of my concerns about the race condition implicit in
reflecting a live stream to several clients.




> 
> > 	c. When one populates the "Range" header with a real value, what
> > constraints are there (if any) on the relationship between 
> that value
> > and the first packets actually sent in each stream?  For example,
> > several servers simply report the value requested by the 
> client and then
> > start their streams with packets whose timestamps fall <= 
> the value in
> > the range header.  In this case, the "rtptime" clause values in the
> > RTP-Info header (which must map to the Range header) will 
> be >= the RTP
> > timestamps of the first packets actually received by the 
> client.  Note
> > that this behavior is preferable to support (among other 
> things), very
> > long duration media samples.  [If the reason for this is 
> not clear, I'd
> > be happy to follow up with an example]
> > 
> 
>  From draft version 6, chapter 11.4 PLAY
> 
>    "For a on-demand stream, the server MUST reply with the 
> actual range
>     that will be played back. This may differ from the 
> requested range if
>     alignment of the requested range to valid frame boundaries is
>     required for the media source. If no range is specified in the
>     request, the start position SHALL still be returned in 
> the reply. If
>     the medias that are part of an aggregate has different 
> lengths, the
>     PLAY request SHALL be performed as long as the given 
> range is valid
>     for any media, for example the longest media. Media will be sent
>     whenever it is available for the given play-out point."
> 
> So I think it is clear that one should report what time 
> interval one is 
> actually playing. However it is not likely that one will be 
> able to for 
> all media stream send out RTP packets which matches that time 
> exactly. 
> Therefore the capability you are indicating is required to be 
> supported 
> by a client. However it might be necessary to further clarify 
> this. An 
> example like the following may be worth adding.
> 
> C->S:
> PLAY
> Range:npt=13.21-20
> 
> S->C
> 200 Ok
> Range:npt=13.160-20
> RTP-Info: 
> url=audio;seq=50;rtptime=50000,url=video;seq=12;rtptime=123400
> 
> Then the RTP media arrives:
> 
> Audio (8kHz RTP timestamp rate):
> S -> C: RTP packet - seq = 50, rtptime = 49830, NPT time = 13.15125 s
> 
> Video (90kHz RTP timestamp rate):
> S -> C: RTP packet - seq = 10, rtptime = 1234000, NPT time= 13.160 s
> 
> 
> So this indicate that it is expect that at least one media 
> will have the 
> same time as the indicate range in the response. However the 
> client will 
> need to handle that some media is prior than the requested and the 
> indicated start time. What is currently not clear is if a server is 
> allowed to send a audio packet with the following characteristics:
> 
> Audio (8kHz RTP timestamp rate):
> S -> C: RTP packet - seq = 50, rtptime = 50100, NPT time = 13.1725 s
> 
> In other words a packet that is later than the indicated play 
> time but 
> prior to the request time.
> 
> Feedback on this is desired.
>
I guess that, if I had to, I could live with the constraint that (for
"on demand" streaming) at least one 'rtptime' value must exactly equal
the value of the first packet sent in one of the streams.  I do feel
quite strongly that this must be the "last" of the first packets (I
think we're in agreement here).

However, consider a hypothetical presentation that has a single MIDI
stream.  The MIDI stream has a duration of 30 seconds, broken into three
samples (one packet per sample) with durations like this:

	packet 1: NPT=0.0 through 10.0
	packet 2: NPT=10.0 through 20.0
	packet 3: NPT=20.0 through 30.0

Now suppose that a user (via the client) requests a seek to NPT=7.0.

By my original proposal, I'd set the Range header to "7.0", but send
packet 1.  The data sample within the packet would have its own internal
timing information, which would let the client 'skip' the first 7
seconds of the MIDI sample and start playing at npt=7.0.

With your constraint, I'd be forced to put either "0.0" or "10.0" in my
Range header (depending upon which packet I selected).  I submit that
neither of these would be very palatable from an end-user perspective.

Hopefully, its clear that I'm not arguing for an ability to start
streams at an arbitrary point earlier than the "Range" response.
Rather, I'm suggesting that the deviation from the Range header on *all*
tracks could be as much as one media sample's duration.

Just to be clear -- my example isn't all that contrived.  We've
definitely seen/streamed MIDI tracks, and there are efforts underway to
develop "timed text" specifications which may very well have the same
"long duration sample" properties.




> 
> > 	d.  As a follow-up to [c], what constraints would there be on
> > the "implicit" value for the Range header in a live stream 
> where it is
> > reported as "npt=now-", assuming that "rtptime" clauses are 
> present in
> > the RTP-Info header.  For example, could the server pick some point,
> > say, 5 minutes in the past as the synchronization point it 
> will use to
> > generate the "rtptime" clause values?
> 
> In theory it could, but it would be less nice behaviour, but a client 
> could experience this and should be capable of handling, 
> because it is 
> almost equal to a session where one losses 5 minutes of data. If you 
> have the time 5 minutes ago, you only need to perform a 
> multiplication + 
> an addition per media to give a time value for the current time.
>
Right, so this was really a follow-up to the above discussion.  I'm very
much in favor of insisting (for "on demand" streams) that the value in
the 'Range' header be >= the RTP timestamps of the first packets sent by
the server in all streams.  This gives a guarantee that a client has all
the media necessary to begin the presentation at that NPT.

However, in the case of a live stream, we aren't really saying anything
in the Range header about where the 'presentation start' point is -- nor
does a user/client select such a point.  Everything is basically "now"
(pause now, play now).  Hence I find it difficult to justify any sort of
required relationship between the 'RTP-Info' header 'rtptime' values and
the first RTP timestamps the client receives (except the obvious that
conversions from RTP timestamp to/from NPT must generate correct
answers).  This is one of my issues with requiring the RTP-Info header
for live streams -- I don't like having two different modes of
interpretation -- one for "on demand", and the other for "live".





> 
> > 
> > 	e. One of the most common current scenarios for distributing a
> > live broadcast is to have a "broadcaster" component that is *not* an
> > RTSP server -- it simply generates an RTP stream to a 
> pre-defined set of
> > broadcast or unicast UDP ports and produces a ".sdp" file 
> that describes
> > the broadcast.  The SDP file is staged in the media root of a full
> > RTSP/RTP streaming server.  When a client requests a 
> session from the
> > ".sdp", the server parses the *.sdp (which it may only 
> become aware of
> > at the point of the first client request) to figure out 
> where it should
> > "listen" for the streams (there is no RTSP dialog between 
> the server and
> > the broadcaster).  In such a case, there is simply no way 
> to guarantee
> > that the server knows *either* sequence number *or* synchronization
> > information for the streams before it needs to respond to 
> the client.
> > In theory, this is disallowed by the new draft of the spec, 
> is it not?
> > In practice, we (and others) have been using this method 
> for years, and
> > it works just fine, so long as the client does not insist 
> on having the
> > RTP-Info header.  The client simply "resyncs" itself when 
> the first SRs
> > actually arrive in the data stream.  I realize that we're 
> working on a
> > new mechanism  (the relay work) that would have the 
> broadcaster be an
> > RTSP client and thus pass along the synchronization 
> information when it
> > "announces" the availability of the presentation, but do you really
> > think we should be breaking an existing facility at the 
> same time as we
> > are introducing its replacement?
> 
> Just to better understand the case: You are saying that the 
> media root, 
> is receiving the RTP stream from the broadcaster component, 
> however it 
> is totally ignoring it until a client requests it from the server?
> 
> If the server was not ignoring the stream, and instead parsed 
> the RTCP 
> information it would have the information.
>
Almost, but not quite:

	* Broadcaster on host A is sending streams to host B, ports I
and J.  The individual who sets up the broadcast manually transfers a
".sdp" file into the content root on host B.

	* Server on host B has no idea that the broadcaster exists and
has no idea that it has been given a new ".sdp" file.

	* Client requests the ".sdp" file

	* In servicing the request, server notes that it isn't currently
receiving this broadcast and sets up listeners on ports I and J.

This is the way most live streaming we have seen is actually done today.

The notion that the server would hold up the PLAY reply until it
receives RTCP SRs is problematic in a ton of ways.  There is no
guaranteed upper bound on this wait period.  How long do you wait before
you time out and reject the request?  And what exactly are you gaining
by waiting?  You are making an end-user sit for (potentially) several
seconds before media buffering even begins.  How much do you lose by
simply omitting the RTP-Info header and beginning immediate packet
forwarding?  Consider two cases:

	* SRs come very soon: user will have very short period of
"slightly out of sync" presentation -- in fact, maybe none at all
(because the client may be able to process the sync while still doing
initial buffering).

	* SRs come after several secs: user will have long period of
"out of sync" presentation before the receipt of the SRs syncs things
up.  However, if you force the wait, the user will simply spend those
same seconds looking at a blank screen (and they'll never see that part
of the presentation).


> 
> Secondly, in the last revision we reduced the requirement on having 
> RTP-Info from a MUST to a SHOULD, the current text says:
> 
> 14.33 RTP-Info
> 
>     The RTP-Info response-header field is used to set RTP-specific
>     parameters in the PLAY response. These parameters 
> correspond to the
>     synchronization source identified by the ssrc parameter of the
>     Transport response header in the SETUP reponse. For 
> streams using RTP
>     as transport protocol the RTP-Info header SHOULD be part of a 200
>     response to PLAY.
> 
> 
>          The exclusion of the RTP-Info in a PLAY response for RTP
>          transported media will result in that a client needs to
>          synchronize the media streams using RTCP. This may have
>          negative impact as the RTCP can be lost, and does not need
>          to be particulary timely in their arrival. Also
>          functionality as informing the client from which packet a
>          seek has occurred is affected.
> 
> 
> So the issue is simply if anyone is protesting against having this 
> requirement on clients to handle media protocol (RTCP) for synch.
> 
> 
I guess my issue is that I think we need to bite the bullet and be clear
that the requirements are simply different for live streams and "on
demand" streams:

	* For reflected live streams, you may not be able to include
SSRC in SETUP response (because you haven't seen any packets); for "on
demand", you can.

	* For reflected live streams, the Range header is useless,
except as a mechanism to signal that this *is* a live stream (e.g.,
npt=now-).  For "on demand" streams, it is critical.

	* For reflected live streams, the RTP-Info header may simply not
be there (and maybe even *shouldn't* be there); if it is there, it is
interpreted a little differently than with "on demand" streams.  For "on
demand" streams, it should be there, fully populated, and deeply
understood.

> > 
> > 	f. Two editorial notes on the draft:
> > 
> > 		+ The separation of header syntax and sub-value
> > discussion (ala NPT discussion) from the header descriptions is very
> > confusing.  It makes it quite difficult to get the full 
> story on any one
> > attribute of the specification (you end up having to simply 
> search for
> > keywords in the document, which is complicated in the case 
> of RTP-Info
> > because it shows up in several examples).
> 
> I think the main problem currently is that all headers description is 
> lacking a bit of extra text to clarify syntax related 
> questions. However 
> I am not certain that I correctly understand you. Could you please 
> better explain, and propose how to amend this problem.
>
Maybe an example would clarify this.  I went to section 14.29 expecting
to get the full definition of the Range header and how it is used.  What
I did *not* find in 14.29:

	* The different ways of specifying the range (had to go to
sections 3.4, 3.5, and 3.6).

	* The "live streaming" correlation with "npt=now-" (section
11.4)

	* The actual syntax for the Range header (subtly buried in
section 17.2.3)

So what would be my proposal?  Well, to define/describe things (like
headers) in exactly one place in the spec and not split and distribute
definitions, e.g.:

	* Put the syntax with the header description (e.g. ditch 17.2.3
and pull it into the header descriptions)

	* Pull in all the context-related usage that is scattered
through other parts of the spec (suck 3.4, 3.5, 3.6 and relevant text in
11.4 into 14.29).

Now, I realize this is probably more than you're willing to bite off at
this point, but maybe there are some "low hanging fruit" things you
might think of doing if you recognize and agree with the source of my
frustration.

BTW, a little better description of when you use what error codes would
definitely be helpful.  For example, what error should a server return
if it attempts to parse a file requested for an "on demand" presentation
and finds that the file is corrupted in some fashion?

> 
> > 
> > 		+ Maybe I just don't know how to parse the syntax
> > properly, but it actually looks to me like the syntax 
> asserts that the
> > "seq" and "rtptime" values are in fact mutually exclusive -- you can
> > only have one or the other (or is 1* supposed to be 
> interpreted as "one
> > or more"?)
> 
> 1* = one or more.
> 
Yeah, figured that one out after spending a little more time here :-)



> 
> 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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        Jae-Hwan Kim <kjh1905m@nextreaming.com>,
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Subject: Re: [MMUSIC] ETag Definition in draft-rfc2326bis-06
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From: Pekka Pessi <Pekka.Pessi@nokia.com>
In-Reply-To: <40725865.7020904@ericsson.com> (Magnus Westerlund's message of
 "Tue, 06 Apr 2004 09:12:37 +0200")
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Date: Thu, 08 Apr 2004 15:43:54 +0300
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Hello Magnus,

Magnus Westerlund <magnus.westerlund@ericsson.com> writes:
>> So, I think options for bis-07 are:
>> 1) deprecate If-Modified-Since and introduce If-None-Match
>> 2) ditch ETag header in 304 and continue to use the current mix of
>>    HTTP/1.0 and HTTP/1.1 mechanisms

>I don't think there are reasons to deprecated If-Modified-Since. 
>It is still useful for cases when one likes to use HTTP dates. 
>However adding If-None-Match to enabled conditional retrieves
>with DESCRIBE seems beneficial.

Yes, you are right. By "deprecating" I meant that no-one should
build a new RTSP client that uses If-Modified-Since only.

>Agree, then the proposal is to include also the ETag header in RTSP. 
>Comments on this.

>Summary of proposed changes:

>- Add ETag header.
>- Add If-None-Match
>- Clarify If-Match
>- Clarify usage of ETags in general and their usage.

>Comments?

Also, a clarification of Accept-Language/Content-Language could be
in order. Accept-Language now states that 

   the language specified applies to the presentation description
   and any reason phrases, not the media content.

However, ETag effectively allows clients and proxies to cache
alternative contents for a particular resource. If we have two SDP
descriptions for URI in different languages, I'd expect that they
can refer to two different media resources. The actual media could
be in different languages, too, but the language must be chosen
before the SETUP is sent. For example, we can ask for Finnish SDP:

C->S: DESCRIBE rtsp://localhost/news RTSP/1.0
      Accept-Language: fi
      CSeq: 1

S->C: RTSP/1.0 200 OK
      CSeq: 1
      Content-Location: /news.fi/sdp
      Content-Language: fi
      Content-Type: application/sdp
      Last-Modified: Wed, 07 Apr 2004 23:15:22 GMT
      Date: Thu, 08 Apr 2004 14:44:22 GMT
      ETag: "1d387f94c4bf2b8aad21271c27cd456e"
      Content-Length: ...

      v=0
      o=news 32178463 8147637 IN IP6 ::1
      s=Nyytiset
      t=0 0
      a=range:npt=0-13:34
      m=audio 8468 RTP/AVP 96
      a=control:audio
      a=rtpmap:96 AMR-WB/16000

and swedish SDP:

C->S: DESCRIBE rtsp://localhost/news RTSP/1.0
      Accept-Language: se
      CSeq: 2

S->C: RTSP/1.0 200 OK
      CSeq: 2
      Content-Location: /news.se/sdp
      Content-Language: se
      Content-Type: application/sdp
      Last-Modified: Thu, 08 Apr 2004 14:44:22 GMT
      ETag: "140940f8c9aa3e2a9df426cd1d626f7a"
      Content-Length: ...

      v=0
      o=news 32178463 8147637 IN IP6 ::1
      s=Uuheter
      t=0 0
      a=range:npt=0-13:34
      m=audio 8468 RTP/AVP 96
      a=control:audio
      a=rtpmap:96 AMR-WB/16000

and then we can use the SDP (and Content-Location URL) to the
select the language:

C->S: SETUP rtsp//localhost/news.se/audio RTSP/1.0
      CSeq: 3
      Transport: RTP/AVP;unicast;client_port=5004-5005

S->C: RTSP/1.0 200 OK
      CSeq: 2
      Session: 9bc0d7421edc303a93ae2dacc7117150
      Transport: RTP/AVP;unicast;client_port=5004-5005;server_port=3056-3057

--Pekka

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Folks,

Draft minutes for the MMUSIC session at the 59th IETF in Seoul are 
attached. Please send any comments or corrections by Thursday 15th 
April, to meet the IETF publication deadline.

Colin

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Minutes of the Multiparty Multimedia Session Control (MMUSIC) WG
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=
=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D

Reported by:	J=F6rg Ott
Notes by:	Colin Perkins

General
-------
Colin Perkins, J=F6rg Ott

The MMUSIC WG met once at the 59th IETF.  The meeting was chaired by
Colin Perkins and J=F6rg Ott.  Some 100 people attended the meeting.

J=F6rg reviewed the agenda, with no changes being requested.  He also
provided a brief status update: Several WG documents are with the IESG
now, some of them waiting to be updated.  No RFCs were published since
the last IETF.


Internet Media Guides
---------------------
Rod Walsh, Juha-Pekka Luoma, Yuji Nomura
- draft-ietf-mmusic-img-req-03.txt
- draft-nomura-mmusic-img-framework-03.txt

Rod Walsh discussed the changes since the last version in the two
above WG documents.  The requirements document has seen various
clarifications and formal corrections.  The problem statement was
augmented to point to the holes existing IETF protocols leave in the
problem space.  The use cases were reordered, further cases covering
subscribe/notify requirements were added.  Delivery and description
requirements were clarified.  For the framework document, the
descriptions of SUBSCRIBE and NOTIFY operations were improved, the
draft was aligned with the requirements draft.  Finally, again, formal
and editorial corrections were made.

Rod then presented the conceivable next steps: the documents are
already in WGLC, so there are considered pretty much complete.  Within
the IMG framework, three potential new areas of work have been
identified: an IMG transfer envelope, a multicast delivery protocol,
and a subscribe notify mechanisms.  Thoughts on all areas exist as
individual drafts, and Rod requests they be added to the MMUSIC or
appropriate other WG charter (see below).  There is quite some
interest from other bodies (including 3GPP and DVB) so this work
should proceed.

In the following, Pekka and Yuji outlined the suggested work items,
starting from the framework draft and filling in the components
needed: The transfer envelope is supposed to be an XML definition for
a container that can carry arbitrary metadata as contents
(draft-luoma-mmusic-img-metadata-envelope-00.txt).  This could be done
in MMUSIC.  The multicast transport protocol should be designed as a
specific instantiation of the FLUTE protocol spec from the RMT WG, as
suggested in draft-luoma-mmusic-img-muppet-04.txt.  This work could be
in MMUSIC or in RMT.  The unicast subscribe/notifiy mechanisms could
be based upon the SIP event framework, but no draft exists here so
far.  This could be done in MMUSIC or SIPPING.

The presentation provoked some discussion on whether it would be
appropriate to think about using the SIP SUBSCRIBE/NOTIFY mechanism,
as no obvious relation can be seen to other SIP work.  It was noted
that this work is less about personal presence than about conveying
state changes in general, and it was requested that clear use cases and
requirement be drawn up and sent to the SIPPING list for discussion.
Furthermore, it was remarked that the XML encapsulation format will =
become
much easier if all data objects are labelled rather than allowing for
generic XML to be encapsulated, e.g. for diffs.  Yuji noted that a =
generic
encapsulation is necessary.

There were no objections raised against continuing this work.  Colin and
J=F6rg were tasked to talk to the ADs and the chairs of the other =
relevant
working groups (particularly SIPPING and RMT) to determine where which =
work
should go and to initiate re-chartering MMUSIC accordingly.


SDP Security Descriptions for Media Streams
-------------------------------------------
Flemming Andreasen
- draft-ietf-mmusic-sdescriptions-03.txt

Flemming briefly reviewed the overall structure of the draft and the
changes since the last revision: It consists of a core framework and
parameter definitions, a specific instantiation of this framework for=20
SRTP, procedures on how to operate with, and without, offer/answer
scheme, and a grammar.  Quite a few changes have been made to the draft
since -02: following consensus from the last IETF, support for multicast
and multipoint SRTP sessions was removed, so was the SRC parameter.
Rules for creating and removing crypto contexts are now more explicit.
Numerous additions were made to clarify the use with offer/answer.
Finally, the IANA registry structure was changed.

Flemming pointed out one minor issue with offer/answer: with o/a, an
initial crypto context is established.  However, subsequent o/a
exchanges may not just modify codec parameters but also peer addresses
(IP address and/or port).  In this case, the new entity may not have
information about the current ROC value.  The solution is to reset the
ROC value to zero whenever any part of an address changes (at either
end).

A question was raised which changes would affect security; this seems
to be restricted to the case of a transport address or port change at
either end.

The group felt the document was ready for WGLC.


SDP Security Preconditions
--------------------------
Flemming Andreasen
- draft-andreasen-mmusic-securityprecondition-01.txt

Flemming briefly introduced the problem to be addressed: during an
offer/answer exchange, the offerer and answerer exchange security
parameters.  When the answered has picked a scheme (possibly one out
of several offered) and the respective parameters, it can start
sending media right away.  The first packets of the secured media
stream may reach the offerer before the answer, so that the offerer
does not know which security description to use.  RFC 3312 provides a
solution by means of introducing preconditions which was suggested for
this problem at the last IETF.  In the meantime, no concerns were
raised and no issues found so that this work is believed to be useful
and also ready.

It was tentatively agreed to adopt this as a WG item, pending charter
discussion.


SDP Connectivity Preconditions
------------------------------
Flemming Andreasen
- draft-andreasen-mmusic-connectivityprecondition-00.txt

Flemming introduced the problem to be addressed: various circumstances
(e.g. deployment of NATs, firewalls, etc.) may interfere with packet
transmission and may prevent media streams from flowing in either or
both directions, even though call signaling setup succeeds.  The =
proposed
approach is to re-use the precondition mechanism defined in RFC 3312
with a new precondition tag to indicate that a call shouldn't proceed
until connectivity has been verified.  To test whether media stream
connectivity (which is only defined for RTP) is available, a mechanism
such as STUN or the proposed new RTP NOOP payload format can be used
(the draft doesn't mandate any particular connectivity check).

Significant discussion arose concerning the relationship between this
work and ICE (the latter being used to ensure that there is
connectivity).  Commenters found that the term "connectivity" needs a
more precise definition in the draft to be able to take a binary
decision of whether or not a connectivity precondition is satisfied.
A further issue was raised in conjunction with SIP forking, as the
initiator of a call may receive packets from multiple destinations and
may be unable to tell which branch satisfied the precondition.
Flemming believes the approach is workable and took up the task to
explain this in more detail in a revised version.

The chairs noted that there is clearly interest in the draft, but
chartering this as an MMUSIC work item again requires a charter
discussion first.


Interactive Connectivity Establishment (ICE)
--------------------------------------------
Jonathan Rosenberg
- draft-ietf-mmusic-ice-01.txt

Jonathan presented three issues and proposed solutions regarding the
current ICE draft.

Issue 1 is may occur with a port restricted flow if the called entity
does not respond soon enough, e.g. with a 200 OK in the case of SIP
(refer to the slides for details of the message flow).  In this case,
a STUN timeout may occur at the callee because the caller does not
receive the 200 OK and start sending STUN packets itself (thereby
creating the necessary binding in its local NAT) so that media
connectivity is not established.  This is a race condition between the
STUN timeout and the answer from the callee.

Jonathan has two solution proposals: solution 1 is to mandate an
immediate answer, this can be achieved in SIP e.g. with a 183 conveying
the necessary addressing information back to the caller, and to increase
the STUN timeout.  Solution 2 is to mandate a second ICE cycle to =
re-check=20
STUN-allocated addresses; this solution increases the signaling overhead
associated with session setup (not the setup delay!) but eliminates the
race condition in all cases.  Jonathan proposed solution 1 as the
appropriate solution.

A comment was made that a connectivity precondition may solve this
issue which Jonathan argues against.  A suggestion was made to avoid a
second cycle with solution two, based upon the priority order of the
connection type.  Jonathan feels that basing this decision on the NAT
type determination is too risky, since it is difficult to determine the
type of NAT that is present.  A question was raised whether or not this
use of 183 would interfere with "early media" via 183; this will be
checked.  It was also proposed to extend the timeout until the "200 Ok"
in solution 1, and there was some brief discussion of the implications
of this..=20

No consensus was reached on issue 1.  Jonathan will send mail to the
list summarizing the issue and soliciting feedback.

Issue 2 is on how to prioritize addressing information in ICE.  So far,
the intention is to minimize relaying and to prefer IPv6.  There are,
however, are other conceivable criteria, e.g. to maximize the number of
hops across security networks, minimize path latency, minimize number of
router hops, etc.  The issue is (a) whether one of these criteria needs
to be chosen and, if so, (b) which one.  Jonathan argues that this is in
essence a policy decision and ICE does not depend on any policy.  It
works as long as at least one uable address exists.  Nevertheless, there
may be many parties involved with different policies and preferences.
With the work of SIP/SIPPING on session policies (e.g. allowing service
providers to assert policies on media streams), the ICE address
preference appears simply as another policy.

Jonathan proposes to not fix address preferences but rather clarify, in
the ICE spec, that there are different axes of optimizations and,
document the existing algorithm.  Other specs shall be allowed to define
(informationally) other algorithms.  A reference session policy should
be described as a non-normative approach for finding out other policies.
This proposal was found reasonable.

Issue 3 addresses the interaction between ICE and RTP NOOP (a
presentation on RTP NOOP was given).  The NOOP format appears to have
some commonalities with STUN: sent to RTP port, requests an immediate
(RTCP) response, and checks for connectivity and QoS between endpoints.
The same (except for QoS checks) can be said for STUN used by ICE.
Jonathan briefly described the differences: NOOP uses RTP and hence is
cleaner for RTP sessions; but ICE works for other protocols, too.  NOOP
uses RTCP for responses which are sent to different port and hence may
not work through many NATs.  ICE (STUN) also provides address allocation
and uses username/password for validation while NOOP would require SRTP.

The obvious options, given this commonality, are to either use only
NOOP, to use only STUN, or to use both.  There was a lot of discussion,
trying to understand and clarify the differences between the drafts,
with no clear conclusion.  It is plain that clarification of use cases,
and of the relationship and overlap between the various drafts, is
needed.



RTSP Update
-----------
Magnus Westerlund
- draft-ietf-mmusic-rfc2326bis-06.txt
- draft-ietf-mmusic-rtsp-nat-02.txt
- draft-zeng-mmusic-map-ice-rtsp-00.txt

Magnus presented the current status of the RTSP core spec (see slides
for the details).  The spec has been progressing well.  Out of the
currently logged 40 bugs almost all are solved, with only six
unresolved ones.  The spec underwent major changes and restructuring:
the use of RTP was rewritten, the syntax is now compiled in a single
chapter, and a proposal on timing RTSP messages was included.  The
numerous minor updates are listed in the slides.

Magnus discussed the open issues and their current status (elaborated
on in detail on the slides).  There was some discussion of options for
authentication to prevent session hijacking, including the possibility
to mandate use of RTSP over TLS, as an alternative to cryptographically
random session IDs. It was also noted that we need to understand the use
cases for destination redirection, to determine how big an open issue
this is.

Finally, Magnus presented those resolved issues that did not yet make it
into the spec: RTSP proxies, use cases for RTSP, and the use of implicit
redirects.

Everyone is encouraged to review the revised spec in detail and
comment!  The editors will edit in the remaining issues.  The
teleconferences held for discussing and resolving issues shall be
resumed.  The goal is to finish the RTSP core spec in 2004.

Next, Magnus reported on the interactions between RTSP and NATs. =20
As per the discussion during IETF-58, a requirement section was=20
added and the usage of ICE is now in a separate draft, and solution=20
proposals (regarding protocol changing mechanisms) were removed.

The remaining open issues include to clarify the recommendations on
working with ALGs, with specific recommendation to be provided for
firewalls ALGs (which are different from NAT ALGs).  An appropriate
protocol modifying solution needs to be chosen.  Various high-level
proposals exist for the latter including using ICE (for a proposal
exists which is briefly presented), embedding STUN in an RTSP server,
and using symmetric RTP.  Interested parties are requested to write
drafts; these should be evaluated afterwards and a single solution
should be selected.


Temporal References within URIs
-------------------------------
Conrad Parker
- draft-pfeiffer-temporal-fragments-02.txt

Conrad Parker motivated the work by the need to define links to time
points and regions within a real-time media resource (e.g. a movie) so
that pieces of interest can be referred to directly.  The draft on
rfc2396bis defines "?query" and "#fragment" to enable referring to a
subpart of a resource.  This is integrated with HTTP and other
download in a straightforward fashion: the server interprets the URI
with the additional parameters and returns only the selected piece of
the resource.=20

Conrad notes that in RFC 2326 the handling of query identifiers and
fragments is undefined.  He suggests specific interpretations of queries
and fragments for handlng by the server and the client, respectively.
He raises various issues, particularly how the mechanism can be kept
extensible. It was noted that, while there is a clear use case for
fragment requests, it is less obvious why queries are useful. Some
clarification was requested.

Other short discussion suggests that -- rather than defining individual
specific extensions for RTSP URIs and extending those every now and then
-- it may be more appropriate if a generic header inclusion mechanism is
defined, as for SIP.

				 - * -


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Date: Thu, 08 Apr 2004 15:25:01 -0700
From: Geetha Srikantan <Geetha.Srikantan@Sun.COM>
Subject: Re: [MMUSIC] draft-rfc2326-06 comments wrt live streaming
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Geetha Srikantan wrote:

>> touched upon below). It is well known that RTP-Info is required to 
>> handle
>> PLAY after PAUSE (see a discussion thread a few months ago in this 
>> mailing
>> list). Fortunately I have not seen customers asking for PAUSE in live
>> streaming. But others might have. I don't know.
>>
>>  
>>
> [tz] Now a more interesting problem is for random positioning (as you
> Live sessions can't be PAUSED, right? So this shouldn't be an issue..
>
> geetha

I meant to say - live streams can't be paused in the same sense as on-demand
streams. PAUSE for on-demand streams, implies the capability to pick up
from the PAUSE point. For live, resuming after a PAUSE will resume at 
the *current*
position in the stream and not from the PAUSE point.

geetha


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From: "peter" <peter.lewis@upperside.fr>
To: <mmusic@ietf.org>
Date: Fri, 9 Apr 2004 15:24:11 +0200
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Subject: [MMUSIC] SSL VPN Conference
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. How to provide SSL-based remote access to a broad range of Web and
legacy applications? 
. What about application performance and requirements?
. Are encrypted application tunneling issues solved? 
. What differences with IPsec VPNs?

These questions, among others, will be tackled by the most recognised
experts in this field during the SSL VPN Conference to be held in Paris from
November 30th to the 3rd of December, 2004.

 

A call for proposals is online at:

 <http://www.upperside.fr/sslvpn04/sslvpn04intro.htm>
http://www.upperside.fr/sslvpn04/sslvpn04intro.htm

 


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<p class=3DMsoNormal><font size=3D3 face=3D"Times New Roman"><span =
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style=3D'font-size:12.0pt'><o:p>&nbsp;</o:p></span></font></p>

<p class=3DMsoNormal><strong><b><font size=3D3 color=3D"#e5a700" =
face=3DArial><span
lang=3DEN-GB =
style=3D'font-size:12.0pt;font-family:Arial;color:#E5A700'>&#8226; =
</span></font></b></strong><font
face=3DArial><span lang=3DEN-GB style=3D'font-family:Arial'>How to =
provide SSL-based
remote access to a broad range of Web and &nbsp;&nbsp;legacy =
applications? <br>
<strong><b><font color=3D"#e5a700" face=3DArial><span =
style=3D'font-family:Arial;
color:#E5A700'>&#8226; </span></font></b></strong>What about application
performance and requirements?<br>
<strong><b><font color=3D"#e5a700" face=3DArial><span =
style=3D'font-family:Arial;
color:#E5A700'>&#8226; </span></font></b></strong>Are encrypted =
application
tunneling issues solved? <br>
<strong><b><font color=3D"#e5a700" face=3DArial><span =
style=3D'font-family:Arial;
color:#E5A700'>&#8226; </span></font></b></strong>What differences with =
IPsec
VPNs?<br>
<br>
These questions, among others, will be tackled by the most recognised =
experts
in this field during the <strong><b><font color=3D"#015d6c" =
face=3DArial><span
style=3D'font-family:Arial;color:#015D6C'>SSL VPN =
Conference</span></font></b></strong>
to be held in <st1:place w:st=3D"on"><st1:City =
w:st=3D"on">Paris</st1:City></st1:place>
from November <span class=3Dtextebold>30th to the 3rd of December, =
2004</span>.</span></font><span
lang=3DEN-GB><o:p></o:p></span></p>

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<p class=3DMsoNormal><font size=3D3 face=3DArial><span lang=3DEN-GB =
style=3D'font-size:
12.0pt;font-family:Arial'>A call for proposals is online =
at:</span></font><span
lang=3DEN-GB><o:p></o:p></span></p>

<p class=3DMsoNormal><font size=3D3 face=3DArial><span =
style=3D'font-size:12.0pt;
font-family:Arial'><a =
href=3D"http://www.upperside.fr/sslvpn04/sslvpn04intro.htm"
title=3D"http://www.upperside.fr/sslvpn04/sslvpn04intro.htm"><span =
lang=3DEN-GB>http://www.upperside.fr/sslvpn04/sslvpn04intro.htm</span></a=
></span></font><span
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<p class=3DMsoNormal><font size=3D2 face=3DArial><span lang=3DEN-GB =
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</body>

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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		: Key Management Extensions for Session Description
                          Protocol (SDP) and Real Time Streaming Protocol (RTSP)
	Author(s)	: J. Arkko, et al.
	Filename	: draft-ietf-mmusic-kmgmt-ext-11.txt
	Pages		: 25
	Date		: 2004-4-9
	
This document defines general extensions for SDP and RTSP to carry
messages, as specified by a key management protocol, in order to
secure the media. These extensions are presented as a framework, to
be used by one or more key management protocols. As such, their use
is meaningful only when complemented by an appropriate key management
protocol.

General guidelines are also given on how the framework should be used
together with SIP and RTSP. The usage with the MIKEY key management
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This is to announce a working group last call on the key management 
extensions for SDP and RTSP (draft-ietf-mmusic-kmgmt-ext-11.txt).

Please send any final comments on this draft to the mailing list by 
Friday 23rd April. If no substantive technical comments are received by 
that time, we will request the IESG consider this draft for publication 
as a Proposed Standard RFC.

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


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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		: A Framework for the Usage of Internet Media Guides
	Author(s)	: R. Walsh, et al.
	Filename	: draft-ietf-mmusic-img-framework-04.txt
	Pages		: 23
	Date		: 2004-4-13
	
This document defines a framework for the delivery of Internet Media
   Guides (IMGs). An IMG is a structured collection of multimedia
   session descriptions expressed using SDP, SDPng or some similar
   session description format. This document describes a generalized
   model for IMG delivery mechanisms, and the use of existing protocol
   to create an IMG delivery infrastructure.

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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		: Protocol Requirements for Internet Media Guides
	Author(s)	: Y. Nomura, et al.
	Filename	: draft-ietf-mmusic-img-req-04.txt
	Pages		: 21
	Date		: 2004-4-13
	
This memo specifies requirements for a protocol for accessing and
updating Internet Media Guide (IMG) information for media-on-demand
and multicast applications. These requirements are designed to guide
development of an IMG protocol for efficient and scalable delivery.

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From exim@www1.ietf.org  Wed Apr 14 17:19:38 2004
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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		: A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP)
	Author(s)	: M. Westerlund
	Filename	: draft-ietf-mmusic-sdp-bwparam-06.txt
	Pages		: 21
	Date		: 2004-4-14
	
The existing Session Description Protocol (SDP) bandwidth modifiers
and their values include the bandwidth needed also for the transport
and IP layers. When using SDP with protocols like the Session
Announcement Protocol (SAP), the Session Initiation Protocol (SIP)
and the Real-Time Streaming Protocol (RTSP) and when the involved
hosts reside in networks running different IP versions, the
interpretation of what lower layer bandwidths are included is not
clear. This document defines a bandwidth modifier that does not
include transport overhead; instead an additional packet rate
attribute is defined. The transport independent bit-rate value
together with the maximum packet rate can then be used to calculate
the real bit-rate over the transport actually used.

A URL for this Internet-Draft is:
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From exim@www1.ietf.org  Wed Apr 28 14:18:09 2004
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Subject: [MMUSIC] Session Description for FLUTE Sessions (aka FLUTE SDP)
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Hello World

I would like to ask for ideas and opinions on a session description for =
FLUTE sessions. (FLUTE is the unidirectional file delivery protocol from =
the Reliable Multicast Transport WG - draft-ietf-rmt-flute-07.txt - like =
an RTP for files at a small stretch of the imagination).

In short, to use FLUTE certain out-of-band information is mandatory and =
optional. Although this can be sought by any number of proprietary and =
closed methods, it's likely to be a big help to have an open document on =
how to describe this in a way also suitable for the general Internet. =
Thus, I want to build up some initial expert (and novice) opinions on =
the semantics and syntax of what would be needed.

FLUTE and ALC provide a starter-kit of data items which can easily be =
turned into a semantic description. For syntax, it seems SDP is the =
natural choice and SDP already comes with a decent tool kit of =
attributes to reuse. (This is not to say SDPng and others would be =
inappropriate, just that to start the work now SDP seems to be the only =
show in town).

Some other relevant context is that 3GPP (MBMS/SA4) has identified the =
need for such a FLUTE session description, and the time constraints of =
3GPP release 6 (September freeze) offer a minimal window to do a 2 way =
swap of ideas and agreement before a standard is needed. I certainly =
have no experience in turning out RFC's in 5 months flat (anyone? :), =
but a serious attempt to work on the issue in the next few months would =
be an excellent way of ensuring a single interoperable method (and open =
standard). In addition, I am expecting ETSI/DVB to make a "call for =
technologies" in this area within a few weeks so the benefits of rapid =
progress should be pretty obvious (but don't shoot me for suggesting it =
:)

Who wants to get involved?

Cheers, Rod.

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Subject: [MMUSIC] SDP and SDP-new: control media type
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(if you're not an SDP guru, you may wish to check the relevant snips at =
the bottom first)

For sdp-new...

Since conference is a subset (example) of session, is there any good =
reason why the control media type is for "conference control channel for =
the session" as opposed to "any control channel for the session"?

My reason for asking is: if a media were entirely dedicated to control =
data, but the session were not a conference (e.g. a webcast) then the =
natural media type would still seem to be control even though the SDP =
spec seems to disallow this (in favour of data media type) - which seems =
odd.

If there are no great reasons for the limitation to conferece control, =
why not change the sdp text to say:
   "control" is used to specify an additional control channel for the =
session.

Cheers, Rod.



Note, relevant snips are:

sdp-new (and SDP) describe the "control" media type usage as:

   The first sub-field is the media type.  Currently defined media are
   "audio", "video", "application", "data" and "control" ...
   ... "control" is used to specify an additional
   conference control channel for the session.

(e.g. m=3Dcontrol 49234 H323 mc)

And also defines session and conference thus:

   Conference: A multimedia conference is a set of two or more
      communicating users along with the software they are using to
      communicate.

   Session: A multimedia session is a set of multimedia senders and
      receivers and the data streams flowing from senders to receivers.
      A multimedia conference is an example of a multimedia session

_______________________________________________
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One of the implications of wanting an SDP description for FLUTE =
(previous email) is that we will need a protocol ID for the m line. =
Although this seems trivial to me, I would like to know if anyone has =
any great objections to "UDP/FLUTE" style? ("RTP/AVT", "udp" and "TCP" =
are existing examples)

So for example, you'd get an media field looking like this:
   m=3Ddata 12345 UDP/FLUTE 0=20

(Any reason not to prefer "FLUTE/UDP"?)

Cheers, Rod.

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This is an automatic digest of bugs submitted and changed on the RTSP bug tracker.  To see a list of bugs, visit:

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When replying, please make your Subject line more specific than "Re: Rtspspec-bugs digest"



Today's Topics:

   1. [ rtspspec-Bugs-944002 ] Clarify port number usage (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 28 Apr 2004 14:06:00 -0700
Subject: [rtspspec-bugs] [ rtspspec-Bugs-944002 ] Clarify port number usage

Bugs item #944002, was opened at 2004-04-28 14:05
Message generated for change (Tracker Item Submitted) made by Item Submitter
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=944002&group_id=23194

Category: None
Group: None
Status: Open
Resolution: None
Priority: 5
Submitted By: Anders Klemets (klemets)
Assigned to: Nobody/Anonymous (nobody)
Summary: Clarify port number usage

Initial Comment:
In version 06 of the draft, there is some new text that 
says that "RTCP/UDP packets from the client to the 
server SHALL be sent to ... first odd port number ... in 
client_port range".  This sounds backwards to me.  

I think that if the client wants to send RTCP packets to 
the server, they should be sent to the "server_port" 
RTCP port, not the "client_port" RTCP port.  Surely there 
is no guarantee that the server would be listening for 
incoming packets on the "client_port" ports?

The next paragraph says, "The client SHOULD send its 
RTCP/UDP packets from the ... first odd port number 
specified in "server_port" parameter."
Shouldn't that be "client_port"?  The client may be 
unable to bind to the "server_port" ports.

There are similar problems a couple of paragraphs further 
down, in the section that talks about the new "src_addr" 
and "dest_addr" parameters.


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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=944002&group_id=23194




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Subject: Re: [MMUSIC] And one more SDP-quiz question on protocol IDs
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Hi Rod,

A few comments,

first it is RTP/AVP or RTP/SAVP, and in the near future also RTP/AVPF 
and RTP/SAVPF. I assume it was only a typo.

Personally I prefer FLUTE/UDP, which seems to be the more clear that the 
main protocol information is FLUTE, which in this case is transported 
over UDP. If anyone has another transport FOO that will work for FLUTE 
then they can specify FLUTE/FOO.

I don't think that using "data" as media type is appropriate at all. 
Although RFC 2327 mentions data I don't think it is appropriate at the 
current state. The media types that really is used to my knowledge is 
the ones using the names registered for MIME. See:
http://www.iana.org/assignments/media-types/index.html

If one looks at draft-ietf-mmusic-sdp-new-15.txt the following paragraph 
is present:

"9.2.1 Media types ("media")

    The set of media types is intended to be small and SHOULD NOT be
    extended except under rare circumstances.  The same rules should
    apply for media names as for top-level MIME content types, and where
    possible the same name should be registered for SDP as for MIME.  For
    media other than existing MIME top-level content types, a
    standards-track RFC MUST be produced for a new top-level content type
    to be registered, and the registration MUST provide good
    justification why no existing media name is appropriate (the
    "Standards Action" policy of RFC 2434 [5]."

In this respect I think that "application" is suitable for FLUTE. Can 
you please further motivate why data would be the right choice?

To avoid further confusion I am of the opinion that draft-ietf-sdp-new 
should be updated to move all unregistered media types into a note like:

Note: RFC 2326 mentions the media types "data" and "control", these are 
unregistered media types and should not be used. If these media types 
are deemed suitable in the future, they should be registered in 
accordance with section 9.2.1.

Cheers

Magnus


Rod.Walsh@nokia.com wrote:

> One of the implications of wanting an SDP description for FLUTE (previous email) is that we will need a protocol ID for the m line. Although this seems trivial to me, I would like to know if anyone has any great objections to "UDP/FLUTE" style? ("RTP/AVT", "udp" and "TCP" are existing examples)
> 
> So for example, you'd get an media field looking like this:
>    m=data 12345 UDP/FLUTE 0 
> 
> (Any reason not to prefer "FLUTE/UDP"?)
> 
> Cheers, Rod.
> 
> _______________________________________________
> 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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Subject: RE: [MMUSIC] And one more SDP-quiz question on protocol IDs
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Thread-Topic: [MMUSIC] And one more SDP-quiz question on protocol IDs
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To: <magnus.westerlund@ericsson.com>
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...and...

I prefer FLUTE/UDP as to protocol ID too, but am keeping an open mind =
for now.

Cheers, Rod.


> -----Original Message-----
> From: ext Magnus Westerlund [mailto:magnus.westerlund@ericsson.com]
> Sent: Thursday, April 29, 2004 10:33 AM
> To: Walsh Rod (Nokia-NRC/Tampere)
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] And one more SDP-quiz question on protocol IDs
>=20
>=20
> Hi Rod,
>=20
> A few comments,
>=20
> first it is RTP/AVP or RTP/SAVP, and in the near future also RTP/AVPF=20
> and RTP/SAVPF. I assume it was only a typo.
>=20
> Personally I prefer FLUTE/UDP, which seems to be the more=20
> clear that the=20
> main protocol information is FLUTE, which in this case is transported=20
> over UDP. If anyone has another transport FOO that will work=20
> for FLUTE=20
> then they can specify FLUTE/FOO.
>=20
> I don't think that using "data" as media type is appropriate at all.=20
> Although RFC 2327 mentions data I don't think it is=20
> appropriate at the=20
> current state. The media types that really is used to my knowledge is=20
> the ones using the names registered for MIME. See:
> http://www.iana.org/assignments/media-types/index.html
>=20
> If one looks at draft-ietf-mmusic-sdp-new-15.txt the=20
> following paragraph=20
> is present:
>=20
> "9.2.1 Media types ("media")
>=20
>     The set of media types is intended to be small and SHOULD NOT be
>     extended except under rare circumstances.  The same rules should
>     apply for media names as for top-level MIME content=20
> types, and where
>     possible the same name should be registered for SDP as=20
> for MIME.  For
>     media other than existing MIME top-level content types, a
>     standards-track RFC MUST be produced for a new top-level=20
> content type
>     to be registered, and the registration MUST provide good
>     justification why no existing media name is appropriate (the
>     "Standards Action" policy of RFC 2434 [5]."
>=20
> In this respect I think that "application" is suitable for FLUTE. Can=20
> you please further motivate why data would be the right choice?
>=20
> To avoid further confusion I am of the opinion that=20
> draft-ietf-sdp-new=20
> should be updated to move all unregistered media types into a=20
> note like:
>=20
> Note: RFC 2326 mentions the media types "data" and "control",=20
> these are=20
> unregistered media types and should not be used. If these media types=20
> are deemed suitable in the future, they should be registered in=20
> accordance with section 9.2.1.
>=20
> Cheers
>=20
> Magnus
>=20
>=20
> Rod.Walsh@nokia.com wrote:
>=20
> > One of the implications of wanting an SDP description for=20
> FLUTE (previous email) is that we will need a protocol ID for=20
> the m line. Although this seems trivial to me, I would like=20
> to know if anyone has any great objections to "UDP/FLUTE"=20
> style? ("RTP/AVT", "udp" and "TCP" are existing examples)
> >=20
> > So for example, you'd get an media field looking like this:
> >    m=3Ddata 12345 UDP/FLUTE 0=20
> >=20
> > (Any reason not to prefer "FLUTE/UDP"?)
> >=20
> > Cheers, Rod.
> >=20
> > _______________________________________________
> > mmusic mailing list
> > mmusic@ietf.org
> > https://www1.ietf.org/mailman/listinfo/mmusic
> >=20
>=20
> --=20
>=20
> Magnus Westerlund
>=20
> 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
>=20
>=20

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Subject: RE: [MMUSIC] And one more SDP-quiz question on protocol IDs
Date: Thu, 29 Apr 2004 14:13:35 +0300
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Thread-Topic: [MMUSIC] And one more SDP-quiz question on protocol IDs
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Thanks Magnus

One thing to make clear on the media type with FLUTE: I'm happy with =
anything that's simple and works. (I just want to check that the =
reasoning behind whatever media types are allowed for FLUTE usage are =
solid enough so we don't need to rake over the question many times.)

I Agree that "application" is the natural fit for "FLUTE applications". =
We do need to answer 2 questions though:
1. do we need to specify which media types are allowed? (if not, then we =
allow all)
2. do we need to exclude certain ones? (like the unregistered ones, and =
how does this impact the SDP spec?)

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

A quick trawl through RFC2045 found me:

     type :=3D discrete-type / composite-type
     discrete-type :=3D "text" / "image" / "audio" / "video" /
                      "application" / extension-token
     composite-type :=3D "message" / "multipart" / extension-token
     extension-token :=3D ietf-token / x-token
     ietf-token :=3D <An extension token defined by a
                    standards-track RFC and registered
                    with IANA.>
     x-token :=3D <The two characters "X-" or "x-" followed, with
                 no intervening white space, by any token>

RFC2046 kindly explains these media types: "text", "image", "audio", =
"video", "application", "multipart" and "message".

I'm coming to the opinion that we can allow any of the media types in a =
FLUTE "m line". Naturally any impementation would have to adhere to the =
media type definitions: a FLUTE channel (given by an m line) would only =
deliver files of the described media type (to ensure no impact on =
multi-channel congestion control schemes, I supect this is practically =
the same as saying the media type will be the same for the whole =
session).

So this would mean:
* you may use the application type for any file typess delivered and =
rendered to the user by the FLUTE receiver
* you may use the audio type if only audio files are delivered and =
rendered to the user by the FLUTE receiver
* you may use the video type if only video files are delivered and =
rendered to the user by the FLUTE receiver
etc...

So, for all but the most specialised FLUTE implementations, we'd expect =
the "application" media type. This may be as much as needs to be said in =
a "SDP for FLUTE" I-D.

--- this draws a line under my comments on "one or many media types =
allowed" ---

The issue about allowing the use of "data" and "control" is that SDP =
defines them so their use would be expected, but IANA has not registered =
them. Any idea why on earth these are not registered? Since SDP makes it =
clear that it's minimal set is to be rarely extended, surely the use of =
data and control is totally fine with SDP, regardless of the IANA =
registration?

The issue about allowing the use of "message" and "multipart" is that =
SDP does not define them but IANA has them registered. sdp-new text =
makes it clear that no further RFC is needed for these existing MIME top =
level types to be used - but what is actually needed (anything?) is =
unclear. (This is more academic interest than me having a real need).

--- now on when "data" might be prefered over "applciation" ---

SDP gave me the impression that "application" is suitable where 'the =
receiver renders the content to the user directly' and "data" is =
suitable where 'the receiver does not directly render the content to the =
user' (e.g. where the FLUTE receiver and media player are seperate =
applications):
   ...The difference between "application"
   and "data" is that the former is a media flow such as whiteboard
   information, and the latter is bulk-data transfer such as
   multicasting of program executables which will not typically be
   displayed to the user.=20

So the example of "m=3Ddata..." was based on trying to do what the spec =
asks for (not on any strong preference) and was not intended to exculude =
"m=3Dapplication...".

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

Since the quantity of text may have made this unclear, here's the bottom =
line:
* Practically anything if fine for me, including "application" only
* I want strong arguments for the way we go so that we don't need to =
re-enter this issue

Cheers, Rod.

PS Yep, AVT was a typo for AVP - oops


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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] And one more SDP-quiz question on protocol IDs
Date: Thu, 29 Apr 2004 13:09:05 +0100
To: Rod.Walsh@nokia.com
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Rod,

Can you remind us, briefly, what sort of data is conveyed in a FLUTE 
session? If it's always a single data format, could we not simply 
mandate that the MIME type and sub-type are signalled in the SDP "m=" 
line as the <media> and <fmt> respectively?

Colin



On 29 Apr 2004, at 12:13, Rod.Walsh@nokia.com wrote:

> Thanks Magnus
>
> One thing to make clear on the media type with FLUTE: I'm happy with 
> anything that's simple and works. (I just want to check that the 
> reasoning behind whatever media types are allowed for FLUTE usage are 
> solid enough so we don't need to rake over the question many times.)
>
> I Agree that "application" is the natural fit for "FLUTE 
> applications". We do need to answer 2 questions though:
> 1. do we need to specify which media types are allowed? (if not, then 
> we allow all)
> 2. do we need to exclude certain ones? (like the unregistered ones, 
> and how does this impact the SDP spec?)
>
> ---   ---   ---   ---
>
> A quick trawl through RFC2045 found me:
>
>      type := discrete-type / composite-type
>      discrete-type := "text" / "image" / "audio" / "video" /
>                       "application" / extension-token
>      composite-type := "message" / "multipart" / extension-token
>      extension-token := ietf-token / x-token
>      ietf-token := <An extension token defined by a
>                     standards-track RFC and registered
>                     with IANA.>
>      x-token := <The two characters "X-" or "x-" followed, with
>                  no intervening white space, by any token>
>
> RFC2046 kindly explains these media types: "text", "image", "audio", 
> "video", "application", "multipart" and "message".
>
> I'm coming to the opinion that we can allow any of the media types in 
> a FLUTE "m line". Naturally any impementation would have to adhere to 
> the media type definitions: a FLUTE channel (given by an m line) would 
> only deliver files of the described media type (to ensure no impact on 
> multi-channel congestion control schemes, I supect this is practically 
> the same as saying the media type will be the same for the whole 
> session).
>
> So this would mean:
> * you may use the application type for any file typess delivered and 
> rendered to the user by the FLUTE receiver
> * you may use the audio type if only audio files are delivered and 
> rendered to the user by the FLUTE receiver
> * you may use the video type if only video files are delivered and 
> rendered to the user by the FLUTE receiver
> etc...
>
> So, for all but the most specialised FLUTE implementations, we'd 
> expect the "application" media type. This may be as much as needs to 
> be said in a "SDP for FLUTE" I-D.
>
> --- this draws a line under my comments on "one or many media types 
> allowed" ---
>
> The issue about allowing the use of "data" and "control" is that SDP 
> defines them so their use would be expected, but IANA has not 
> registered them. Any idea why on earth these are not registered? Since 
> SDP makes it clear that it's minimal set is to be rarely extended, 
> surely the use of data and control is totally fine with SDP, 
> regardless of the IANA registration?
>
> The issue about allowing the use of "message" and "multipart" is that 
> SDP does not define them but IANA has them registered. sdp-new text 
> makes it clear that no further RFC is needed for these existing MIME 
> top level types to be used - but what is actually needed (anything?) 
> is unclear. (This is more academic interest than me having a real 
> need).
>
> --- now on when "data" might be prefered over "applciation" ---
>
> SDP gave me the impression that "application" is suitable where 'the 
> receiver renders the content to the user directly' and "data" is 
> suitable where 'the receiver does not directly render the content to 
> the user' (e.g. where the FLUTE receiver and media player are seperate 
> applications):
>    ...The difference between "application"
>    and "data" is that the former is a media flow such as whiteboard
>    information, and the latter is bulk-data transfer such as
>    multicasting of program executables which will not typically be
>    displayed to the user.
>
> So the example of "m=data..." was based on trying to do what the spec 
> asks for (not on any strong preference) and was not intended to 
> exculude "m=application...".
>
> ---   ---   ---   ---
>
> Since the quantity of text may have made this unclear, here's the 
> bottom line:
> * Practically anything if fine for me, including "application" only
> * I want strong arguments for the way we go so that we don't need to 
> re-enter this issue
>
> Cheers, Rod.
>
> PS Yep, AVT was a typo for AVP - oops


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Subject: RE: [MMUSIC] And one more SDP-quiz question on protocol IDs
Date: Thu, 29 Apr 2004 15:35:00 +0300
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Hi Colin

FLUTE transports "files". Potentially any file formats.

It is likely that some applications (server and client) will be used for =
homogeneous content where every file is of exactly the same type. (In =
this case, signalling file content type as the <media>+<fmt> is fine =
IMHO).

However, we expect that in most cases FLUTE will carry more than one =
(and sometimes very many) content/file types. So only "application" or =
"data" media types are generic enough for the bulk of cases. (And then =
signalling a list of file types would be done otherwise if it is =
necessary for capability or content selection or purchase - I daren't =
say more for fear of opening the data model can of worms :)

Apart from the IANA registration status of "data", the choice between =
"application" and "data" seems open to interpretation according to SDP =
(my earlier email said about as much as I have to say on why). Any =
opinions on data vs application vs either?

Cheers, Rod.


> -----Original Message-----
> From: ext Colin Perkins [mailto:csp@csperkins.org]
> Sent: Thursday, April 29, 2004 3:09 PM
> To: Walsh Rod (Nokia-NRC/Tampere)
> Cc: magnus.westerlund@ericsson.com; mmusic@ietf.org
> Subject: Re: [MMUSIC] And one more SDP-quiz question on protocol IDs
>=20
>=20
>=20
> Rod,
>=20
> Can you remind us, briefly, what sort of data is conveyed in a FLUTE=20
> session? If it's always a single data format, could we not simply=20
> mandate that the MIME type and sub-type are signalled in the SDP =
"m=3D"=20
> line as the <media> and <fmt> respectively?
>=20
> Colin
>=20
...

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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] And one more SDP-quiz question on protocol IDs
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Hi,

On 29 Apr 2004, at 13:35, <Rod.Walsh@nokia.com> wrote:
> FLUTE transports "files". Potentially any file formats.
>
> It is likely that some applications (server and client) will be used 
> for homogeneous content where every file is of exactly the same type. 
> (In this case, signalling file content type as the <media>+<fmt> is 
> fine IMHO).

Okay.

> However, we expect that in most cases FLUTE will carry more than one 
> (and sometimes very many) content/file types. So only "application" or 
> "data" media types are generic enough for the bulk of cases. (And then 
> signalling a list of file types would be done otherwise if it is 
> necessary for capability or content selection or purchase - I daren't 
> say more for fear of opening the data model can of worms :)

How does the receiver know the contents of the FLUTE session if more 
than one file type is transported? Are you trying to signal that list 
in SDP? Or is it picked up from the FLUTE session somehow?

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


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Hi,

After having thought of this, what I think for FLUTE is that it actually 
defines its own registry for applications. There will be need for 
different applications to be identified. So if someone is announcing a 
FLUTE delivery session one can also instantiate the correct receiver 
behaviour. Different applications may also go as far as redefining what 
it expects the FDT to contain.

So I think one should consider making definition of how the format 
specific part of the M line should be used. In that one can also 
consider if FLUTE is always one type of media, or if the it really is 
necessary to identify the media type already in the SDP. In my view I 
think we can consider any instantiation of the FLUTE delivery to be for 
the purpose of an application. Therefore use the "application" media 
type. Then the application itself is identified in the FMT part of the 
m= line. A registry for applications is then established so that one can 
determine the application. And it is the applications that defines what 
type of files are actually sent within the FLUTE delivery session.

How does that sound?


Cheers

Magnus

Colin Perkins wrote:

> Hi,
> 
> On 29 Apr 2004, at 13:35, <Rod.Walsh@nokia.com> wrote:
> 
>> FLUTE transports "files". Potentially any file formats.
>>
>> It is likely that some applications (server and client) will be used 
>> for homogeneous content where every file is of exactly the same type. 
>> (In this case, signalling file content type as the <media>+<fmt> is 
>> fine IMHO).
> 
> 
> Okay.
> 
>> However, we expect that in most cases FLUTE will carry more than one 
>> (and sometimes very many) content/file types. So only "application" or 
>> "data" media types are generic enough for the bulk of cases. (And then 
>> signalling a list of file types would be done otherwise if it is 
>> necessary for capability or content selection or purchase - I daren't 
>> say more for fear of opening the data model can of worms :)
> 
> 
> How does the receiver know the contents of the FLUTE session if more 
> than one file type is transported? Are you trying to signal that list in 
> SDP? Or is it picked up from the FLUTE session somehow?
> 

-- 

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] And one more SDP-quiz question on protocol IDs
Date: Thu, 29 Apr 2004 16:05:57 +0300
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> How does the receiver know the contents of the FLUTE session if more=20
> than one file type is transported? Are you trying to signal that list=20
> in SDP? Or is it picked up from the FLUTE session somehow?

It can be picked up from the FLUTE session. One or more FDT Instances =
describe the files and include the MIME type. FDT (File Decription =
Table) Instances are XML objects/files.

I not convinced that it is necessary to provide a list of several file =
types of the session for two reasons: the MIME types are available from =
FLUTE; and in some advanced media discrovery system (IMG stuff) the =
client would select the files, most likely already needing to knwo MIME =
types then (why what you can't use?). But I'm open to ideas!

Cheers, Rod.

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Date: Thu, 29 Apr 2004 08:23:13 -0700 (PDT)
From: Stephen Casner <casner@acm.org>
To: Rod.Walsh@nokia.com
cc: mmusic@ietf.org
Subject: Re: [MMUSIC] And one more SDP-quiz question on protocol IDs
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On Wed, 28 Apr 2004 Rod.Walsh@nokia.com wrote:

> One of the implications of wanting an SDP description for FLUTE
> (previous email) is that we will need a protocol ID for the m
> line. Although this seems trivial to me, I would like to know if
> anyone has any great objections to "UDP/FLUTE" style? ("RTP/AVT",
> "udp" and "TCP" are existing examples)
>
> So for example, you'd get an media field looking like this:
>    m=data 12345 UDP/FLUTE 0
>
> (Any reason not to prefer "FLUTE/UDP"?)

The order of RTP/AVP is the order of decreasing significance.  That
is, AVP is a particular specialization (profile) of RTP.  There are
multiple second components that follow the first, but it is unlikely
that the symbol "AVP" will follow anything other than "RTP" and still
have the same meaning.

In the case of UDP and FLUTE, FLUTE is one of several protocols that
can be layered after UDP in the packet.  FLUTE is a specialization of
the use of UDP.  So I prefer UDP/FLUTE.

(I also prefer IP/TCP, because it reflects layering order, but the
common terminology is TCP/IP because TCP is considered layered "over"
IP.  I suppose this is another big-endian / little-endian issue.)

Note that in the case of "RTP/AVP", the use of UDP is implied.  To use
RTP over another lower-layer protocol requires a new term, such as
DCCP/RTP/AVP.

If FLUTE would only be used with UDP in this context, then UDP could
be implicit for FLUTE as well.

                                                        -- Steve

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From: Colin Perkins <csp@csperkins.org>
Subject: Re: [MMUSIC] And one more SDP-quiz question on protocol IDs
Date: Thu, 29 Apr 2004 19:30:44 +0100
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On 29 Apr 2004, at 14:05, Rod.Walsh@nokia.com wrote:
>> How does the receiver know the contents of the FLUTE session if more
>> than one file type is transported? Are you trying to signal that list
>> in SDP? Or is it picked up from the FLUTE session somehow?
>
> It can be picked up from the FLUTE session. One or more FDT Instances 
> describe the files and include the MIME type. FDT (File Decription 
> Table) Instances are XML objects/files.
>
> I not convinced that it is necessary to provide a list of several file 
> types of the session for two reasons: the MIME types are available 
> from FLUTE; and in some advanced media discrovery system (IMG stuff) 
> the client would select the files, most likely already needing to knwo 
> MIME types then (why what you can't use?). But I'm open to ideas!

It's beginning to sound like "m=data 49000 FLUTE/UDP" would be the 
right way to go, letting the application pick up the media types from 
the FLUTE session.

Unfortunately, though, the <fmt> parameter is mandatory in SDP "m=" 
lines. Unless there is something specific to FLUTE that could be used, 
we might need a dummy value?

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


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Subject: RE: [MMUSIC] And one more SDP-quiz question on protocol IDs
Date: Fri, 30 Apr 2004 12:39:49 +0300
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> ... Therefore use the "application" media=20
> type. Then the application itself is identified in the FMT=20
> part of the=20
> m=3D line.=20
>> How does that sound?

Sounds good.

Naturally, the FMT could be "null" to indicate that the application is =
going to need some other means to work out session content types (e.g. =
FDT, "IMG service description", etc.)=20

> A registry for applications is then established so=20
> that one can=20
> determine the application. And it is the applications that=20
> defines what=20
> type of files are actually sent within the FLUTE delivery session.
>=20
> How does that sound?

Sounds like I need to be educated about how to deal with these =
registrations :)

Cheers, Rod.

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> It's beginning to sound like "m=3Ddata 49000 FLUTE/UDP" would be the=20
> right way to go, letting the application pick up the media types from=20
> the FLUTE session.

Sounds good. My lack of preference over application and data should be =
pretty clear now :)

I see that in the case of a data carousel kind of application (which =
user application request files from locally) "data" fits well. And in =
the media player application (where the player also includes the FLUTE =
protocol, as many already have HTTP/html) then "applciation" fits. Since =
these are receiver specific and since, in some situations, the sender =
may not have a good idea in advance that all receiver applications are =
of a certain sort - coming down hard on either only data or only =
application may be ticky. As far as I see "application" is great for my =
main use cases, but I suspect this may not be true of everyone.

> Unfortunately, though, the <fmt> parameter is mandatory in SDP "m=3D"=20
> lines. Unless there is something specific to FLUTE that could=20
> be used,=20
> we might need a dummy value?

Makes sense "dummy" =3D "null".

Cheers, Rod.

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Hi Steve

Thanks for the comment. I understand both arguments and am pretty sure =
either of FLUTE/UDP and UDP/FLUTE would do the job nicely

UDP/FLUTE seems closer to the RTP/AVP precedent. (Anyone know if there =
was a big-endian motivation behind this selection for SDP in the first =
place?)

FLUTE/UDP seems to show the right emphasis: FLUTE. For The receiver and =
application capability, the client is primarily concerned that it is =
using FLUTE (UDP or foo follows after that). This is also true for =
RTP/AVP, clients are first interested in RTP, then UDP and AVP comes =
next (since I've never implemented an RTP client, please put me straight =
if I blundered on this).

As long as we choose one, and not both ways, then I'm happy.

The reason for not just suggesting FLUTE (without the /UDP) is the =
potential for some theoretical future FLUTE to work over other than UDP =
(LCT and ALC bind us to UDP for now, but make it clear that this may not =
always be the case). Although FLUTE/TCP (foo/FLUTE, FLUTE/foo etc) would =
be contrary to the current LCT, it should not cause problems for FLUTE =
alone (FLUTE/TCP seems to fit into "congestion control scenario (c)" of =
=A74 as TCP would provide the CC which is "no more aggressive than TCP" =
:) I think this is more a discussion for LCT transition to PS (or LCTv2) =
in RMT, so let me put the lid back on that can of worms for now.

Cheers, Rod.


> -----Original Message-----
> From: casner@packetdesign.com=20
> [mailto:casner@packetdesign.com]On Behalf
> Of ext Stephen Casner
> Sent: Thursday, April 29, 2004 6:23 PM
> To: Walsh Rod (Nokia-NRC/Tampere)
> Cc: mmusic@ietf.org
> Subject: Re: [MMUSIC] And one more SDP-quiz question on protocol IDs
>=20
>=20
>=20
> On Wed, 28 Apr 2004 Rod.Walsh@nokia.com wrote:
>=20
> > One of the implications of wanting an SDP description for FLUTE
> > (previous email) is that we will need a protocol ID for the m
> > line. Although this seems trivial to me, I would like to know if
> > anyone has any great objections to "UDP/FLUTE" style? ("RTP/AVT",
> > "udp" and "TCP" are existing examples)
> >
> > So for example, you'd get an media field looking like this:
> >    m=3Ddata 12345 UDP/FLUTE 0
> >
> > (Any reason not to prefer "FLUTE/UDP"?)
>=20
> The order of RTP/AVP is the order of decreasing significance.  That
> is, AVP is a particular specialization (profile) of RTP.  There are
> multiple second components that follow the first, but it is unlikely
> that the symbol "AVP" will follow anything other than "RTP" and still
> have the same meaning.
>=20
> In the case of UDP and FLUTE, FLUTE is one of several protocols that
> can be layered after UDP in the packet.  FLUTE is a specialization of
> the use of UDP.  So I prefer UDP/FLUTE.
>=20
> (I also prefer IP/TCP, because it reflects layering order, but the
> common terminology is TCP/IP because TCP is considered layered "over"
> IP.  I suppose this is another big-endian / little-endian issue.)
>=20
> Note that in the case of "RTP/AVP", the use of UDP is implied.  To use
> RTP over another lower-layer protocol requires a new term, such as
> DCCP/RTP/AVP.
>=20
> If FLUTE would only be used with UDP in this context, then UDP could
> be implicit for FLUTE as well.
>=20
>                                                         -- Steve
>=20

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