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From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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Hi Thomas,

Comments below:

Thomas Zeng wrote:

>Magnus:
>
>I have a comment on 3.2 Symetric RTP.
>
>As Tom Marshall pointed out ealier, a RTSP/RTP based streaming server
>normally uses a single RTP port (let's call it X) to send RTP packets to
>many (potentially hundreds) of clients. Now Symetric RTP asks these clients
>to send an RTP packet with their individual client SSRC in the payload to
>this port X on the server host. Since each client randomly generates client
>SSRC, there is some chance (as high as 10^-4 according to RFC1889, section
>8.1) that two clients may end up with same SSRC, in which case the server
>may send RTP packets to wrong clients. I did study the SSRC collision
>detection scheme in RFC1889 and found it irrelevant because it only applies
>to multicast RTP sessions. I don't think destination IP can help here since
>prior to the establishment of binding  our server doesn't know clients
>public IPs. Besides, the two collided clients may well be behind the same
>NAT and hence probably share public IP.
>  
>

My first comment is that the easiest way of solving this problem, is for 
the server to send from different ports for each client. That way you 
reduce the probability of collisions and ensures maximum security in 
respect to hijacking.

>I hereby suggest that the binding RTPs to add, in a fashion defined below,
>client's RTSP sessionID in addition to client's SSRC in those binding RTP
>packets. Notice RTSP sessionIDs are generated by the server and hence can be
>made unique amongst all clients of that server.
>
>The session ID happens to provide other niceties:
>
>1. One client only has to make sure that its client SSRCs are unique amongst
>the RTP tracks it has in the session (if it happens to have more than one
>track)
>2. Since RTSP session ID can be 64 bytes or longer, it offers added
>protection against stream hijacking or brute force attacks (since now the
>hijacker/attacker must guess not only SSRC, but also SessionID).
>
>Careful thinking revealed to me that the fundamental problem here is that
>prior to the establishment of port mapping, there is no way for the server
>to determine RTP session using the traditional source and destination (IP,
>port-pair) as defined in RFC1889 section 3. Unfortunately, it means that
>what I said about RTP port mapping is also true to RTCP port mapping.
>
>Assuming you readers are with me so far, I then propose that the binding RTP
>packets include SessionID in the following fashion:
>
>1. Do NOT use PT=0, instead, use a dedicated PT for binding purposes (we
>probably need to reserve one).
>2. The only payload in this RTP is the session ID.
>3. This payload type shall be used for the binding packets the client send
>to sender's RTCP send port as well (in lieu of the normal RTCP packets in
>Magnus draft, section 3.2.3, page 8). 
>  
>

I can agree that a new payload for RTP would allow a server to save it 
from the need to perform real port multiplexing between its clients. 
However I don't see a need for solution only to this problem. If we 
would be design a new RTP payload format then we should make this for 
the general security problems related to RTSP.

1. Provide improved security for symmetric RTP by allowing for longer 
random tags.
2. Allow for a RTP/RTCP based challenge mechanism that can be used to 
ensure that a receiver accepts the data traffic, i.e. solve the 
destination security problem.

Such a solution should consider the following problems:
- Be usable for both RTP and RTCP.
- Allow for dynamic tag length.
- Be possible to authenticate in respect to both sender and receiver.
- Have room for response codes.

>
>Interestingly, there is another way to deal with client SSRC collision:
>rejecting the SETUP request containing a client SSRC already registered.
>But that breaks backwards compatibility in that current clients don't know
>to change SSRC and resend SETUP request. Besides, the nice security feature
>of sessionID is gone.
>  
>

This seem to be a bad idea. It creates backwards compatibility issues 
where other solutions exist.

>Hence I prefer the inclusion of SessionID.
>  
>

I prefer the solution that the servers take responsibility and actually 
use multiple ports. In the long run we could define a general challenge 
mechanism. This mechanism can then in the future be used to ensure 
better security in several ways.

Best Regards

Magnus Westerlund 

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



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--NextPart

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

	Title		: SDP Security Descriptions for Media Streams
	Author(s)	: M. Baugher, D. Wing
	Filename	: draft-ietf-mmusic-sdescriptions-00.txt
	Pages		: 21
	Date		: 2003-2-28
	
This Internet Draft gives a cryptographic attribute to Session 
Description Protocol (SDP) media streams.  The attribute describes a 
cryptographic key and other parameters, which serve to configure 
security for a media stream.  This draft also defines the SRTP 
parameters for the attribute.  The SDP crypto attribute requires the 
services of a data security protocol to secure the SDP message.

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I have submitted draft-ietf-mmusic-sdp-comedia-05.txt to the IETF in prep 
for the 56th meeting.  A draft copy is available now from:

http://www.rfdsoftware.com/ietf/draft-ietf-mmusic-sdp-comedia-05.txt

I have removed all references to the source address and the 
firewall-related security considerations.  What I did not have time to do 
is incorporate the security issues raised by comedia-fix, but I think those 
are worthy of at least a brief discussion at the 56th.

Please look over the draft and let me know whether there are any remaining 
concerns.

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From: Tom Marshall <tmarshall@real.com>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: Thomas Zeng <zeng@packetvideo.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-rtsp-nat-00.txt
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On Mon, Mar 03, 2003 at 10:13:52AM +0100, Magnus Westerlund wrote:
> Hi Thomas,
>=20
> Comments below:
>=20
> Thomas Zeng wrote:
>=20
> >Magnus:
> >
> >I have a comment on 3.2 Symetric RTP.
> >
> >As Tom Marshall pointed out ealier, a RTSP/RTP based streaming server
> >normally uses a single RTP port (let's call it X) to send RTP packets to
> >many (potentially hundreds) of clients. Now Symetric RTP asks these clie=
nts
> >to send an RTP packet with their individual client SSRC in the payload to
> >this port X on the server host. Since each client randomly generates cli=
ent
> >SSRC, there is some chance (as high as 10^-4 according to RFC1889, secti=
on
> >8.1) that two clients may end up with same SSRC, in which case the server
> >may send RTP packets to wrong clients. I did study the SSRC collision
> >detection scheme in RFC1889 and found it irrelevant because it only appl=
ies
> >to multicast RTP sessions. I don't think destination IP can help here si=
nce
> >prior to the establishment of binding  our server doesn't know clients
> >public IPs. Besides, the two collided clients may well be behind the same
> >NAT and hence probably share public IP.
> >=20
> >
>=20
> My first comment is that the easiest way of solving this problem, is for=
=20
> the server to send from different ports for each client. That way you=20
> reduce the probability of collisions and ensures maximum security in=20
> respect to hijacking.

This is a big resource drain on the server.  It effectively *triples* the
number of sockets you need to pass into select() or poll().  That is, for N
clients, the server must poll on 3*N sockets (one RTSP, RTP, and RTCP socket
per client) instead of 1*N+2 (one RTSP socket per client plus a fixed RTP
and RTP socket).

I can see an example server doing this, but I have doubts that commercial
code would be willing to take the hit.

--=20
"When competing for a section of road or a parking space, remember that the
vehicle in need of the most body work has the right-of-way."
        -- Massachusetts Driver Education Manual

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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Tom Marshall'" <tmarshall@real.com>,
        Magnus Westerlund
	 <magnus.westerlund@era.ericsson.se>
Cc: Thomas Zeng <zeng@PacketVideo.COM>, mmusic@ietf.org
Subject: RE: [MMUSIC] Comments on draft-ietf-mmusic-rtsp-nat-00.txt
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Magnus, Tom:

Another reason that ports are precious resource on the server is that, a
well designed
streaming server may be able to scale up to 10000 simultaneous clients
(which Microsoft's Windows Media server claims to support... I know
Microsoft today doesn't use RTSP but it may in the future). If each has 6
RTP tracks then there will not be enough ports to go around with Magnus
proposal.

Therefor I still think we must be prepared to resolve client SSRC conflict.
Someone suggested that we can put in the "contributing SSRC" field the
server's SSRC for that session. The combination of client SSRC and server
SSRC in the binding RTP packets would be certainly possible to ensure
uniqueness (since the server can choose its SSRC with no collision).

What do people think?

- thomas


-----Original Message-----
From: Tom Marshall [mailto:tmarshall@real.com]
Sent: Monday, March 03, 2003 9:52 AM
To: Magnus Westerlund
Cc: Thomas Zeng; mmusic@ietf.org
Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-rtsp-nat-00.txt


On Mon, Mar 03, 2003 at 10:13:52AM +0100, Magnus Westerlund wrote:
> Hi Thomas,
> 
> Comments below:
> 
> Thomas Zeng wrote:
> 
> >Magnus:
> >
> >I have a comment on 3.2 Symetric RTP.
> >
> >As Tom Marshall pointed out ealier, a RTSP/RTP based streaming server
> >normally uses a single RTP port (let's call it X) to send RTP packets to
> >many (potentially hundreds) of clients. Now Symetric RTP asks these
clients
> >to send an RTP packet with their individual client SSRC in the payload to
> >this port X on the server host. Since each client randomly generates
client
> >SSRC, there is some chance (as high as 10^-4 according to RFC1889,
section
> >8.1) that two clients may end up with same SSRC, in which case the server
> >may send RTP packets to wrong clients. I did study the SSRC collision
> >detection scheme in RFC1889 and found it irrelevant because it only
applies
> >to multicast RTP sessions. I don't think destination IP can help here
since
> >prior to the establishment of binding  our server doesn't know clients
> >public IPs. Besides, the two collided clients may well be behind the same
> >NAT and hence probably share public IP.
> > 
> >
> 
> My first comment is that the easiest way of solving this problem, is for 
> the server to send from different ports for each client. That way you 
> reduce the probability of collisions and ensures maximum security in 
> respect to hijacking.

This is a big resource drain on the server.  It effectively *triples* the
number of sockets you need to pass into select() or poll().  That is, for N
clients, the server must poll on 3*N sockets (one RTSP, RTP, and RTCP socket
per client) instead of 1*N+2 (one RTSP socket per client plus a fixed RTP
and RTP socket).

I can see an example server doing this, but I have doubts that commercial
code would be willing to take the hit.

-- 
"When competing for a section of road or a parking space, remember that the
vehicle in need of the most body work has the right-of-way."
        -- Massachusetts Driver Education Manual
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Hi Thomas and Tom,

I understand your performance considerations. I know we have discussed 
this before on the teleconference. One reason raised for poor port 
performance was the lack of optimized implementations. For example 
linear search that of course give works performance as the number of 
ports increase.

Should we fix this problem I think we should design something that are a 
general mechanism that actually help us in other cases. Therefore my 
suggestion is still to construct a RTP payload format and a RTCP packet 
type that can carries challenge and responses containing security tags.

Thomas Zeng wrote:

>Magnus, Tom:
>
>Another reason that ports are precious resource on the server is that, a
>well designed
>streaming server may be able to scale up to 10000 simultaneous clients
>(which Microsoft's Windows Media server claims to support... I know
>Microsoft today doesn't use RTSP but it may in the future). If each has 6
>RTP tracks then there will not be enough ports to go around with Magnus
>proposal.
>

At least not with only a single IP address.

>
>Therefor I still think we must be prepared to resolve client SSRC conflict.
>Someone suggested that we can put in the "contributing SSRC" field the
>server's SSRC for that session. The combination of client SSRC and server
>SSRC in the binding RTP packets would be certainly possible to ensure
>uniqueness (since the server can choose its SSRC with no collision).
>
>What do people think?
>
I think this last proposal is a bad idea. It may very well have 
implications on the future extensions that we deploy. It also changes 
the meaning of the SSRC and CSRC field to not be the standardized ones. 
The general problem is that RTP was not designed to have multiple 
sessions on the same port. This is simple something the manufacturers of 
streaming severs has done to solve an operations system performance 
problem.

Best Regards

Magnus Westerlund 

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



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Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-rtsp-nat-00.txt
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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Magnus,

Would not this look an awfull lot like STUN then ?

regards,


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com


                                                                                                                                              
                                                                                                                                              
                                                          To:   Thomas Zeng <zeng@packetvideo.com>                                            
                                                          cc:   "'Tom Marshall'" <tmarshall@real.com>                                         
                                                           mmusic@ietf.org                                                                    
                                                           (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                         
               Magnus Westerlund                          Subject:    Re: [MMUSIC] Comments on draft-ietf-mmusic-rtsp-nat-00.txt              
               <magnus.westerlund@era.ericsson                                                                                                
               .se>                                       Classification:                                                                     
                                                                                                                                              
               Sent by:                                                                                                                       
               mmusic-admin@ietf.org                                                                                                          
                                                                                                                                              
               04/03/2003 10:34                                                                                                               
                                                                                                                                              
                                                                                                                                              




Hi Thomas and Tom,

I understand your performance considerations. I know we have discussed
this before on the teleconference. One reason raised for poor port
performance was the lack of optimized implementations. For example
linear search that of course give works performance as the number of
ports increase.

Should we fix this problem I think we should design something that are a
general mechanism that actually help us in other cases. Therefore my
suggestion is still to construct a RTP payload format and a RTCP packet
type that can carries challenge and responses containing security tags.

Thomas Zeng wrote:

>Magnus, Tom:
>
>Another reason that ports are precious resource on the server is that, a
>well designed
>streaming server may be able to scale up to 10000 simultaneous clients
>(which Microsoft's Windows Media server claims to support... I know
>Microsoft today doesn't use RTSP but it may in the future). If each has 6
>RTP tracks then there will not be enough ports to go around with Magnus
>proposal.
>

At least not with only a single IP address.

>
>Therefor I still think we must be prepared to resolve client SSRC conflict.
>Someone suggested that we can put in the "contributing SSRC" field the
>server's SSRC for that session. The combination of client SSRC and server
>SSRC in the binding RTP packets would be certainly possible to ensure
>uniqueness (since the server can choose its SSRC with no collision).
>
>What do people think?
>
I think this last proposal is a bad idea. It may very well have
implications on the future extensions that we deploy. It also changes
the meaning of the SSRC and CSRC field to not be the standardized ones.
The general problem is that RTP was not designed to have multiple
sessions on the same port. This is simple something the manufacturers of
streaming severs has done to solve an operations system performance
problem.

Best Regards

Magnus Westerlund

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



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




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Hi, Phillipe,

My proposal:

Should we fix this problem I think we should design something that are a
general mechanism that actually help us in other cases. Therefore my
suggestion is still to construct a RTP payload format and a RTCP packet
type that can carries challenge and responses containing security tags.


philippe.gentric@philips.com wrote:

>Would not this look an awfull lot like STUN then ?
>  
>
In some sense yes, but mostly no. The goal of this mechanism is to 
identify the intention of sending media and authorizing it. What it 
shares with STUN is the possibility to use it for NAT traversal.

The comparison with STUN would give the following advantages:
- Being integrated into RTP/RTCP it will not have the multiplexing 
problems of STUN.
- It can be used for the general DESTINATION problem of RTSP.

Disadvantage:
- Not being general, only useful with RTP. To solve the multiplexing the 
solution needs to be part of the media protocol. Problem for any 
solution that handles symmetric NAT. 

What this solution will not provide that STUN has is:
- Mechanism to learn the external mapping.
- Mechanism to detect what type of NAT your behind.
- Mechanism to determine the NAT mapping time-out.

I hope you  see the differences that it is not awful lot like STUN. It is in fact another way of traversing NATs which has other advantages. 

Regards

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
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Torshamsgatan 23           | Fax   +46 8 7575550
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Subject: [MMUSIC] RTSP: Agenda telecon 5th March 18.00-19.30 CET.
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Hi,

Sorry being this late. Here is the proposed agenda for tomorrow's telecon.

1. Follow up:
    - draft-ietf-mmusic-rfc2326bis-03.txt
    - Interleave: Aravind to write text and post to MMUSIC to address 
all four bugs.
    - Everybody reads on offer/answer (RFC 3264) and meet in next IETF 
meeting to take next step.

2. What to discuss during the IETF and the MMUSIC session.
    -  What to take up to the whole WG:
        + The last changes
        + Open issues that need input?
    - Off line session
        + Offer Answer for RTSP?
        +
3. New Issues:
* Unclarities when doing resetup
  http://rtsp.org/bug513737

* Conference header underspecified
  http://rtsp.org/bug539040

* "Clarify Range Where t0 <= t1" & "How to use Range when Scale is 
negative"
  http://rtsp.org/bug533691
  http://rtsp.org/bug513880

* "When can REDIRECT be received?",
  http://rtsp.org/bug556061

* "Client issues PAUSE while server ready"
   http://rtsp.org/bug556295

Next meeting:
Proposal: 2nd April.

If you would like to participate please send a mail and you will receive 
the contact information.

Best Regards

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
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Hi,

Rob and I manage to submit a new version of the RTSP updated draft. Most 
of the given comments has found its way into the version. However due to 
lack of time I am not sure I managed to edit all the comments given on 
this draft. So please let me ask forgiveness already now. Please send 
any comments on the draft to the list. I am also sorry that the draft 
has managed to get even thicker. It is now 138 pages in the text version.

To fetch it before being announce, see:

http://rtsp.org/2003/drafts/draft03/draft-ietf-mmusic-rfc2326bis-03.txt
http://rtsp.org/2003/drafts/draft03/draft-ietf-mmusic-rfc2326bis-03.ps
http://rtsp.org/2003/drafts/draft03/draft-ietf-mmusic-rfc2326bis-03.pdf

Best Regards

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
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Subject: RE: [MMUSIC] Comments on draft-ietf-mmusic-rtsp-nat-00.txt
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Magnus:

You wrote:

>The general problem is that RTP was not designed to have multiple 
>sessions on the same port. This is simple something the manufacturers of 
>streaming severs has done to solve an operations system performance 
>problem.

Refering to section 3 of RFC1889, RTP session was defined by one
"source-destination" pair of (ip, rtp-port, rtcp-port) tuples.
It is VERY legal to have one source (ip, rtp-port, rtcp-port) serve traffic
to multiple destination (ip, rtp-port, rtcp-port)'s in unit-cast fashion.
Therefore us streaming server manufactures have not broken the law.

I'd like to point out it is not just OS performance we are talking about. It
is also the *reality* that there are only 64K ports per IP in IPV4, in
addition to the inclination of server Firewall administrators always wanting
to limit port range that the firewall (perhaps without address/port
translation) on the server side will let pass.

If we look at RFC1889 and RFC2326, RTSP servers do NOT have to open a RTP
send port per RTP session. So we are talking about
introducing new rules.

Any time a new restriction is imposed, it must be studied long and hard. In
this case, there exist alternatives. Let me summarize below:

-- Alternative A is to include a SessionID in the binding RTP packet. 
-- Alternative B is to have server suggest client SSRC in setup response, as
part of Transport header. Server and only server can make sure no collision
in suggested client SSRC.  To do so, a new transport parameter,  client_ssrc
will need to be added as optional parameter to Transport header in SETUP
response. For some reason,  "client_ssrc" parameter has already been
proposed in
http://rtsp.org/2003/drafts/draft03/draft-ietf-mmusic-rfc2326bis-03.txt, or
is that just a typo?  I double checked to make sure that client_ssrc is not
part of RFC2326.

- thomas
 

-----Original Message-----
From: Magnus Westerlund [mailto:magnus.westerlund@era.ericsson.se]
Sent: Tuesday, March 04, 2003 1:34 AM
To: Thomas Zeng
Cc: 'Tom Marshall'; mmusic@ietf.org
Subject: Re: [MMUSIC] Comments on draft-ietf-mmusic-rtsp-nat-00.txt


Hi Thomas and Tom,

I understand your performance considerations. I know we have discussed 
this before on the teleconference. One reason raised for poor port 
performance was the lack of optimized implementations. For example 
linear search that of course give works performance as the number of 
ports increase.

Should we fix this problem I think we should design something that are a 
general mechanism that actually help us in other cases. Therefore my 
suggestion is still to construct a RTP payload format and a RTCP packet 
type that can carries challenge and responses containing security tags.

Thomas Zeng wrote:

>Magnus, Tom:
>
>Another reason that ports are precious resource on the server is that, a
>well designed
>streaming server may be able to scale up to 10000 simultaneous clients
>(which Microsoft's Windows Media server claims to support... I know
>Microsoft today doesn't use RTSP but it may in the future). If each has 6
>RTP tracks then there will not be enough ports to go around with Magnus
>proposal.
>

At least not with only a single IP address.

>
>Therefor I still think we must be prepared to resolve client SSRC conflict.
>Someone suggested that we can put in the "contributing SSRC" field the
>server's SSRC for that session. The combination of client SSRC and server
>SSRC in the binding RTP packets would be certainly possible to ensure
>uniqueness (since the server can choose its SSRC with no collision).
>
>What do people think?
>
I think this last proposal is a bad idea. It may very well have 
implications on the future extensions that we deploy. It also changes 
the meaning of the SSRC and CSRC field to not be the standardized ones. 
The general problem is that RTP was not designed to have multiple 
sessions on the same port. This is simple something the manufacturers of 
streaming severs has done to solve an operations system performance 
problem.

Best Regards

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
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--NextPart

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

	Title		: 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-07.txt
	Pages		: 15
	Date		: 2003-3-4
	
This document defines general extensions for SDP and RTSP to carry
the security information needed 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, its use is meaningful only when it is completed by the key
management protocol in use.
General guidelines are also given on how the framework should be used
together with SIP and RTSP.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-kmgmt-ext-07.txt

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Hi Thomas,

Thomas Zeng wrote:

>Magnus:
>
>You wrote:
>
>  
>
>>The general problem is that RTP was not designed to have multiple 
>>sessions on the same port. This is simple something the manufacturers of 
>>streaming severs has done to solve an operations system performance 
>>problem.
>>    
>>
>
>Refering to section 3 of RFC1889, RTP session was defined by one
>"source-destination" pair of (ip, rtp-port, rtcp-port) tuples.
>It is VERY legal to have one source (ip, rtp-port, rtcp-port) serve traffic
>to multiple destination (ip, rtp-port, rtcp-port)'s in unit-cast fashion.
>Therefore us streaming server manufactures have not broken the law.
>  
>

This is in fact a multiplexing problem. One needs to establish the 
session and what it consist of. For unicast it is possible to define a 
session by defining the two end points and the port numbers it uses. 
This gives sufficient information for demultiplexing. This has so far 
been used in servers to save ports. Each session is defined by the 
clients IP/Ports.
For a multicast session the multicast address and port is used for 
multiplexing. If we would have a multi-unicast session one way of 
defining the session is to say all that I receive on a certain port 
belongs to this session, independly of source address.

As shown above how you can utilize the port depends on the 
demultiplexing requirements you have. For normal unicast sessions where 
you know the remote address it works fine. However in the case of 
symmetric RTP we don't know the external address. Therefore one solution 
is to use different server ports.

>I'd like to point out it is not just OS performance we are talking about. It
>is also the *reality* that there are only 64K ports per IP in IPV4, in
>addition to the inclination of server Firewall administrators always wanting
>to limit port range that the firewall (perhaps without address/port
>translation) on the server side will let pass.
>

Yes, there is a limitation to the number of ports per IP. However this 
is a fairly high limitation as it is sufficient for 15000 users each 
having two RTP sessions, i.e. 2 * 2 ports. I don't think using firewalls 
as an argument is a very valid one. If your having a streaming service 
you need to configure the FW to handle this server correctly. Doesn't 
the server in fact belong in the demilitarized zone in most cases.

>
>If we look at RFC1889 and RFC2326, RTSP servers do NOT have to open a RTP
>send port per RTP session. So we are talking about
>introducing new rules.
>

I haven't talking so much about introducing new rules as of actually on 
how you can implement things within the standard.

>
>Any time a new restriction is imposed, it must be studied long and hard. 
>
As this is no restriction just a implementation question.

>In
>this case, there exist alternatives. Let me summarize below:
>
>
>-- Alternative A is to include a SessionID in the binding RTP packet. 
>
The information used to id the session is not important, just that 
something is used. This requires a new RTP payload format and for RTCP 
new packet type. I think this is a good idea, especially when you are at 
it to define a format that can be used to handle destination 
verification of media streams. This solution does allow for much greater 
security by having longer id tags.

>-- Alternative B is to have server suggest client SSRC in setup response, as
>part of Transport header. Server and only server can make sure no collision
>in suggested client SSRC.  To do so, a new transport parameter,  client_ssrc
>will need to be added as optional parameter to Transport header in SETUP
>response. 
>
Yes, this could be one way of assigning the ssrc. However one other 
consideration to think of in regards to symmetric RTP is that if we have 
10000 clients on the same port it will be that many times easier to 
hijack a stream. If you multiplex using different port for each session 
you ensure that the whole number space of the SSRC is used.

>For some reason,  "client_ssrc" parameter has already been
>proposed in
>http://rtsp.org/2003/drafts/draft03/draft-ietf-mmusic-rfc2326bis-03.txt, or
>is that just a typo?  I double checked to make sure that client_ssrc is not
>part of RFC2326.
>
This was added by me in this version. My intention was to use it with 
symmetric RTP and be used by the client to tell which SSRC it will be 
using. This would of course rely on the server ensuring that no two 
users with the same SSRC will be sent to the same server side port.  So 
this is Alternative C as described by the RTSP & NAT draft. The 
parameter should perhaps belong to the RTSP & NAT draft.

There is a fourth alternative (D) as you have earlier described:
Putting this sessions server SSRC as a CSRC. However I still think this 
is a bad idea as it redefines the field. This may interfere with future 
services.

To summarize:
Solution A, good can solve another problem and gives best security. 
Requires more standards work.

Solution B, Requires minimal standards work. Its security might not be 
that good. Changes the assignment rules for SSRC.

Solution C,  Requires minimal standards work (same as B). Its security 
is slightly better than B. Uses more ports.

Solution D. Same standards works as B & C. Does not look future proof.

I would prefer A.


Best Regards

Magnus Westerlund 

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



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CC: Magnus Westerlund <magnus.westerlund@era.ericsson.se>, mmusic@ietf.org
Subject: Re: [MMUSIC] Relative URLs in RTP-Info field
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Aaron Colwell wrote:

>Thank you for the clarification. I was somewhat expecting that answer, but
>I just wanted to make sure.
>
>One other thing that I'd like to have clarified is that if absolute URLs
>are returned that they match what was sent in the SETUP request for that
>stream. I've run into a case where a server will return me an IP address
>in the hostname part of the URL instead of the hostname that I specified.
>It is things like this that make matching SETUP URLs with RTP-Info
>information not very easy. I'd like to not have to use heuristic methods
>for dealing with these types of situations. Perhaps the clarification for
>RTP-Info URLs should be something like
>
>"Any relative URL SHALL use the request URL as base URL. The resulting
> URL must match the URL used to SETUP that stream.
>
I'm not sure exactly what it means to say "the resulting URL must match 
the URL used to SETUP that stream."  In the case below, for example, the 
SETUP URLs are different than the PLAY URL.

OPTIONS rtsp://foo.com
SETUP rtsp://foo.com/audio.3gp/trackID=1
SETUP response
SETUP rtsp://foo.com/video.3gp/trackID=1
SETUP response
PLAY rtsp://foo.com/movie
PLAY response
RTP-Info: url=audio.3gp/trackID=1;seq=0;rtptime=0,
url=video.3gp/trackID=1;seq=234;rtptime=23487

The resulting full url is rtsp://foo.com/movie/audio.3gp/trackID=1 and rtsp://foo.com/movie/video.3gp/trackID=1, which are not the same as the SETUP urls.

> If an absolute URL is
> specified, it MUST match the URL used to SETUP that stream."
>
I think we need to clearer on the exact wording and probably provide a 
good example (similar to what's been give on this alias).

>  
>
Aravind


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To: Aravind Narasimhan <aravind.narasimhan@sun.com>
CC: Aaron Colwell <acolwell@real.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] Relative URLs in RTP-Info field
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Hi Aravind,

You are making a mistake here. You are not following the correct rules 
for how to resolve relative URLs.
See RFC 2396 for these rules.

Aravind Narasimhan wrote:

> PLAY rtsp://foo.com/movie
> PLAY response
> RTP-Info: url=audio.3gp/trackID=1;seq=0;rtptime=0,
> url=video.3gp/trackID=1;seq=234;rtptime=23487

>
> The resulting full url is rtsp://foo.com/movie/audio.3gp/trackID=1 and 
> rtsp://foo.com/movie/video.3gp/trackID=1, which are not the same as 
> the SETUP urls.


The error you made here is that you did not remove all that is after the 
last "/". The base URL is "rtsp://foo.com/movie" however when resolving 
this you take the part upto and including the last "/", i.e, 
"rtsp://foo.com/". To this you add the relative URL resulting in:
rtsp://foo.com/audio.3gp/trackID=1
rtsp://foo.com/video.3gp/trackID=1

>
>> If an absolute URL is
>> specified, it MUST match the URL used to SETUP that stream."
>>
> I think we need to clearer on the exact wording and probably provide a 
> good example (similar to what's been give on this alias). 


Sure, improving the text is always good. Do you have any proposal what 
would be clearer?

Best Regards.

Magnus Westerlund 

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



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Subject: Re: [MMUSIC] Relative URLs in RTP-Info field
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>
>
>>
>> The resulting full url is rtsp://foo.com/movie/audio.3gp/trackID=1 
>> and rtsp://foo.com/movie/video.3gp/trackID=1, which are not the same 
>> as the SETUP urls.
>
>
> The error you made here is that you did not remove all that is after 
> the last "/". The base URL is "rtsp://foo.com/movie" however when 
> resolving this you take the part upto and including the last "/", i.e, 
> "rtsp://foo.com/". To this you add the relative URL resulting in:
> rtsp://foo.com/audio.3gp/trackID=1
> rtsp://foo.com/video.3gp/trackID=1


oh yeah. ..  my bad.  We ran into a similar problem before for the same 
reason. .   so ignore my comment.

>>> If an absolute URL is
>>> specified, it MUST match the URL used to SETUP that stream."
>>
>> I think we need to clearer on the exact wording and probably provide 
>> a good example (similar to what's been give on this alias
>
> Sure, improving the text is always good. Do you have any proposal what 
> would be clearer?

I'll look through the current wording as well as our current email 
thread on this and propose some new wording within the next week.

Aravind

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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:

http://sf.net/projects/rtspspec

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



Today's Topics:

   1. [ rtspspec-Bugs-533691 ] Clarify Range Where t0 <= t1 (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 05 Mar 2003 09:53:53 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-533691 ] Clarify Range Where t0 <= t1

Bugs item #533691, was opened at 2002-03-22 17:36
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=533691&group_id=23194

Category: None
Group: Fix for DS - Need WG Input
Status: Open
Resolution: None
Priority: 5
Submitted By: Thomas M Zeng (zengtm)
Assigned to: Nobody/Anonymous (nobody)
Summary: Clarify Range Where t0 <= t1

Initial Comment:
If the server receives a PLAY range from t0-t1, where
t0 < t1, should it reject the PLAY, or should the 
server play backwards as if the timescale is -1?

If t0 = t1, should the server send any media data 
after responding 200 OK?  
We've seen different servers behave differently in 
this scenario.


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

Comment By: Thomas M Zeng (zengtm)
Date: 2003-03-05 17:53

Message:
Logged In: YES 
user_id=20397

If scale positive, t1 must >= t0.

Otherwise, t1 must <= t0

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

Comment By: Thomas M Zeng (zengtm)
Date: 2002-03-22 17:41

Message:
Logged In: YES 
user_id=20397

I just came across 513880, which is closely related.
That is to say, if timescale is positive, should the play 
range to be required to have t1 > t0?

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=533691&group_id=23194




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Hi everyone,

This is the summary message for the Wednesday, 19th March AVT WG
presentation:

Framing RTP over Connection-Oriented Transport      (Lazzaro)   15
       draft-lazzaro-avt-rtp-framing-contrans-00.txt

I'm CC'ing this to MMUSIC, given the topical overlap of the I-D.

This summary is somewhat long, because I describe the decisions that
will move the document forward, so that WG members who won't be in San
Francisco can voice their views on the lists by responding to this
post. My presentation in San Francisco will put the 3 open issues
described below up on the screen, to see if at least within the room a
consensus can be found.

---

This I-D began with an extended discussion on the WG:

http://www1.ietf.org/mail-archive/working-groups/avt/current/msg01847.html

Three requirements arose from this discussion:

  [1] A memo was needed to document the "classical" framing of RTP over
  TCP, to facilitate interoperability with applications that use it
  (several are mentioned in the linked WG thread).

  [2] There was an MMUSIC I-D that defined SDP conventions for setting
  up a TCP media session (draft-ietf-mmusic-sdp-comedia-04.txt, soon to
  be draft-ietf-mmusic-sdp-comedia-05.txt), but comedia defines neither
  the SDP fmt token to specify an RTP/AVP stream, nor the TCP framing 
  such a stream should use.

  [3] RTSP has support for interleaving RTP and RTCP media streams into
  the TCP stream used for RTSP control method traffic. However, although
  RTSP also has the syntactic power to specify non-interleaved TCP session
  in the SETUP method, there is (apparently) no semantics in common use
  for this syntax.

Requirement [1] is simple to address: write a new I-D to document
classical RTP TCP framing. Sections 2-3 of
draft-lazzaro-avt-rtp-framing-contrans-00.txt attempt to do this.

#####> Open Issue #1

Does draft-lazzaro-avt-rtp-framing-contrans-00.txt accurately capture
classical TCP framing, as used in practice? In specific, are the
details in Section 3 correct?

----

The second requirement is more controversial. Given that we try to
avoid "there's more than one way to do it", it would be best if there
was one RTP/AVP TCP framing method for use with
draft-ietf-mmusic-sdp-comedia-05.txt. The discussion in the WG thread
linked above did not reach consensus, but instead ended with two
different options:

   A. Define classical TCP framing as the framing method for RTP/AVP
      to use with draft-ietf-mmusic-sdp-comedia-05.txt.

   B. Write an AVT I-D that describes RTSP interleaved-mode
      RTP/AVP framing, as run over a stand-alone TCP stream.
      Define this framing method as the framing method to use
      with draft-ietf-mmusic-sdp-comedia-05.txt.

Section 4 of draft-lazzaro-avt-rtp-framing-contrans-00.txt implements
"A". Two more open issues:

#####> Open Issue #2

Is A or B above the right path to take? Or, should we define SDP fmt
codes for both A and B?

----

and

#####> Open Issue #3

If A is the right option to take, are the SDP details in Section 4 of
the draft-lazzaro-avt-rtp-framing-contrans-00.txt acceptable to
everyone?

----

The final requirement involves RTSP support for non-interleaved TCP.
Section 5 of draft-lazzaro-avt-rtp-framing-contrans-00.txt defines
candidate RTSP semantics to use the classical TCP framing in RTSP.

However, recent discussions in the Rtspspec-bugs transcripts seem to
indicate that the editors of the core RTSP draft would prefer to
handle this issue as a part of the RTSP draft document. So, unless I
hear otherwise, I would think that deleting Section 5, and all
mentions of RTSP, from draft-lazzaro-avt-rtp-framing-contrans-00.txt,
is appropriate.

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------
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Hi, Silvia.

I see the possibility of this proposal. However I have a number of 
comments on this proposal.

First is the possibility to specify this as a general mechanism. It 
seems that you want to have this to apply to all URI schemes. Is that 
really possible? How do you know that the use of @ will not collide with 
any usage? For example the rtsp scheme does not define how to interpret 
the fragment. So this is up to the server responding to a request. So 
this can potentially already be used by some implementations. Also does 
it make sense to be defined for all schemes. As this is currently 
related to media it seems that for the moment it applies to HTTP(S) and 
RTSP(U|S).

So a general question, what is the actually possibilities to standardize 
such a solution?

Second, is the usage of the fragment identifier. It seems that you tries 
to redefine the meaning of the fragment identifier. See below.

 From RFC 2396:

  "When a URI reference is used to perform a retrieval action on the
   identified resource, the optional fragment identifier, separated from
   the URI by a crosshatch ("#") character, consists of additional
   reference information to be interpreted by the user agent after the
   retrieval action has been successfully completed.  As such, it is not
   part of a URI, but is often used in conjunction with a URI.

      fragment      = *uric

   The semantics of a fragment identifier is a property of the data
   resulting from a retrieval action, regardless of the type of URI used
   in the reference.  Therefore, the format and interpretation of
   fragment identifiers is dependent on the media type [RFC2046] of the
   retrieval result.  The character restrictions described in Section 2
   for URI also apply to the fragment in a URI-reference.  Individual
   media types may define additional restrictions or structure within
   the fragment for specifying different types of "partial views" that
   can be identified within that media type."

Two things strikes me as not being taken into account by your draft in 
relation to this quote. The first is from the first paragraph of the 
quote. The fragment identifier is interpreted by the user agent. While 
in chapter 4 of your draft you clearly specifies that the server shall 
interpret the fragment. This redefines the behavior of the fragment 
identifier.

Secondly the interpretation of the fragment identifier is in relation to 
the data resulting in the request. So in fact it seems that you need to 
define the behavior on per media basis.

I think the use of fragment to specify where in a media resource is 
appropriate in some case. For example an RTSP URL that has a fragment of 
the proposed type is useful. However as RTSP works the server shall not 
interpret the fragment as this is pointless. Only the client needs to do 
this and use that to determine the range to include in the PLAY request. 
So again the usage rules in chapter 4 seems to be wrong.

It seems that the time specification should be complemented to allow 
specifying ranges. The result of a search or the part of your friends 
wedding video that you want to point out would in most cases only be a 
part of the video. Therefore both a start and stop time seems useful. 
Then it is also a very short step to having multiple ranges allow one to 
point out multiple parts of the same resource that are interesting.

Also I think you have a few strange thing in the definition. First you 
specify an attribute "start" that I think are pointless. Either you 
totally skip the fragment or you can use "npt=0". Secondly it the 
defaulting to "npt". This clashed with the general parameters. If you 
use start then you have actually wrote "npt=start" which isn't defined. 
"now" works as long as this is based on the RTSP definitions of NPT, 
however from your BNF definitions this is evidently not the case. So my 
proposal is to skip having any default time scheme if you are going to 
have general attributes that does not belong to a time scheme.

If you would like the server to take the range into account would not 
the use of query (?) be better? However I hope that somebody that are 
more will versed in URIs can give further suggestions.

I can see the usefulness of this proposal, however there is need to 
resolve a number of issues:
- A general standardization or target certain URI schemes?
- Applicability on different media types and protocols needs to better 
understood.
- Use of fragment or other URI syntax element, like parameter or query?
- Need for having ranges?
- The actual need for general parameters?

Regards

Magnus Westerlund 

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



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Date: Thu, 06 Mar 2003 15:12:12 +0200
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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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To: Louis Guindon <lguindon@mediatrix.com>
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Subject: [MMUSIC] Re: [Sipping] Media descriptor problem in SDP
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Louis,

I have removed the SIPPING mailing list from the cc and added MMUSIC,
since this is an MMUSIC related question.

If you want to provide different parameters for a particular codec, you
should define different dynamic payload types (e.g., 96 and 97).

Gonzalo


Louis Guindon wrote:
> 
> Hi,
> 
> We are receiving the following SDP with an INVITE...
>   m=audio 5004 RTP/AVP 8 8
>   a=fmtp:8 ptime:20
>   a=fmtp:8 ptime:30
>   a=sendrecv
>   a=ptime:20
> 
> After looking in the RFC2327 ABNF for those fields, I cannot find any problem with the grammar. However, it's doesn't make sense to me that we can put the media format twice on the media announcements line. Moreover, it doesn't make more sense to put two attribut for the same media format with different parameters.
> 
> Any idea where I can find about the validity/invadility of SDP packet?
> 
> Regards,
> 
> Louis Guindon
> Software Design Engineer
> Mediatrix Telecom, Inc.
> lguindon@mediatrix.com
> Tel :  819-829-8749 ext. 242
> Fax : 819-829-1973
> 4229, Garlock Street
> Sherbrooke, Qc J1L 2C8
> 
> This message and its attachment(s) contain protected proprietary and
> confidential information of Mediatrix.  It is intended solely for the
> named addressee(s).  If you are not an addressee indicated in this message
> (or responsible for delivery of the message to an addressee), you may not
> copy or deliver this message or its attachment(s) to anyone.  Rather, you
> should permanently delete this message and its attachment(s) and kindly
> notify the sender by email.  No warranty is made that the email or
> attachment(s) are free from computer virus or other defect.
> 
> Ce message et son(ses) fichier(s) adjoint(s) peuvent contenir de
> l'information légale privilégiée ou confidentielle de Mediatrix.  Il est
> destiné exclusivement au(x) destinataire(s) identifié(s).  Si vous n'êtes
> pas un destinataire indiqué dans ce message (ou responsable de la
> livraison du message au destinataire), vous ne pouvez copier ou livrer ce
> message ou son(ses) fichier(s) adjoint(s) à quiconque.  Vous devriez
> plutôt effacer de façon permanente ce message et son(ses) fichier(s)
> adjoint(s) et aimablement en avertir l'expéditeur par courriel.  Aucune
> garantie n'est donnée que ce courriel ou son(ses) fichier(s) adjoint(s) ne
> contiennent pas de virus informatique ou autre défectuosité.
> _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the application of SIP
> Use sip-implementors@cs.columbia.edu for questions on current sip
> Use sip@ietf.org for new developments of core SIP
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Subject: Re: [MMUSIC] Re: [Sipping] Media descriptor problem in SDP
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Hi Louis,

There are actually some errors in this SDP you present. It is however 
not a clear syntax error.

If you look in draft-ietf-rtp-mime-06.txt you will see that chapter 3 
tells how to map the ptime parameter to the SDP. And it shall not be 
part of the a=fmtp line but go into its own line. Unfortunately you 
can't specify these parameters on individual payload type basis. They 
apply in general to all payload formats that allow them in that media 
block.

Also there is an error using the same PT twice on the m= line. So if you 
of some reasons want to have two PTs for PCMA then your SDP should be 
written like this utilizing dynamic PTs as Gonzalo pointed out:

m=audio 5004 RTP/AVP 8 96
a=rtpmap:96 PCMA/8000
a=sendrecv
a=ptime:20

I hope this clarifies things. If the only thing you would like to 
accomplish is to have different frame lengths a single PT is enough. If 
you in the SDP likes to clarify that you will use 20 or 30 but never 
more than 30 ms then you could write this as:

m=audio 5004 RTP/AVP 8 96
a=sendrecv
a=ptime:20
a=maxptime:30


Regards

Magnus

PS. Please don't send confidentiality message like your initial message 
contained as the mailing list you sent to is an open list having a 
public archive.




>Louis Guindon wrote:
>  
>
>>Hi,
>>
>>We are receiving the following SDP with an INVITE...
>>  m=audio 5004 RTP/AVP 8 8
>>  a=fmtp:8 ptime:20
>>  a=fmtp:8 ptime:30
>>  a=sendrecv
>>  a=ptime:20
>>
>>After looking in the RFC2327 ABNF for those fields, I cannot find any problem with the grammar. However, it's doesn't make sense to me that we can put the media format twice on the media announcements line. Moreover, it doesn't make more sense to put two attribut for the same media format with different parameters.
>>    
>>

-- 

Magnus Westerlund 

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



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From mailnull@www1.ietf.org  Thu Mar  6 10:11:26 2003
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Subject: [MMUSIC] new I-D on temporal URI addresses
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Hi all,

we have submitted a proposal to specify temporal URI fragments on 
time-continuous Web resources (such as audio and video files/streams). I 
would like to ask for your brains on whether this is a problem that 
you've stumbled across, too, and whether you like the solution that we 
are proposing.

http://www.ietf.org/internet-drafts/draft-pfeiffer-temporal-fragments-00.txt

THE PROBLEM:
------------
We can see a need for the direct specification of points of interest in 
a URI for a/v. There are several uses for it:

1) Representation of search results:
When typing a search term into your favourite search engine, you would 
like it to search on a/v and bring up the link to an offset in a video 
where the search term matches. e.g. goal in last week's soccer matches

2) Favourite links:
When sending a link to a video to a friend, you'd really like to be able 
to specify the main part of your interest in it. e.g. look at my new 
cool sunglasses in my friend's wedding video


THE SOLUTION:
-------------
What we want is to address fragment offsets into videos. Therefore, the 
"#fragment" part of a URI naturally lends itself for including this 
address specification. For specification of the time points, we were 
inspired by the timestamp specification in RTSP. Therefore, we are 
proposing the following pattern:

#@npt=14.5

-> the "#" part signifies a fragment specification
-> the "@" tells us to look "at" that time specification
-> the "npt=" part is a time scheme just like in RTSP and we are also 
proposing npt, smpte, smpte-30-drop, smpte-25, and clock
-> the time format itself depends on the time scheme, this one 
signifying a 14.5 seconds offset

[for more details refer to the I-D]


THE USE:
--------
Here are examples for temporal URI fragment offsets:

http://foo/bar.mov#@smpte=00:00:14:15
(start downloading at 14 seconds and 15 frames into bar.mov)

rtsp://foo/bar.avi#@utc=20030308T143000.00Z
(start streaming at 8th March 2003 at half past 2pm)

http://foo/bar.mpg#@start
(start downloading from the start)

rtsp://foo/bar.rm#@now
(start streaming now)

---
We wanted to make it simple, intuitive and human readable and have 
therefore come up with this solution. Please let me know what you think.

Cheers,

Silvia.
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To: mmusic@ietf.org
cc: jo@tzi.uni-bremen.de
Date: Thu, 06 Mar 2003 22:57:52 -0500
From: Colin Perkins <csp@csperkins.org>
Subject: [MMUSIC] Working group last call: draft-ietf-mmusic-sdp-bwparam-01.txt
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This is to announce a working group last call on the Transport Independent
Bandwidth Modifier for SDP <draft-ietf-mmusic-sdp-bwparam-01.txt>.

Please send any final comments on this work to the mmusic@ietf.org mailing
list by 17th March. If no substantive comments are received by that time,
we propose to submit this draft to the IESG for publication as a proposed
standard RFC.

Joerg and Colin
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Subject: [MMUSIC] New draft on Stream Switching: draft-gentric-mmusic-stream-switching-00.txt
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Please find attached a new MMUSIC Internet Draft on stream switching

Abstract:

   Stream switching is a technique used to change the data rate of a
   media being streamed, typically for the purpose of adaptation to
   the effectively available bandwidth of the network. A backward
   compatible and independent RTSP "SWITCH" command is proposed in
   order to enable RTSP-based stream switching.

(See attached file: draft-gentric-mmusic-stream-switching-00.zip)

Comments are most welcome.

Regards,

PS: I know normally one should not post drafts on the reflector
but I sent it to internet-drafts@ietf.org on the 20th of february
and it has not surfaced since then(!?)

PS2: This document is also archived online at:
http://www.3gpp.org/ftp/tsg_sa/WG4_CODEC/TSGS4_25bis/Docs/S4-030195.zip


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com
--0__=4EBBE671DFB400C28f9e8a93df938690918c4EBBE671DFB400C2
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Subject: [MMUSIC] A Framework for Internet Media Guides
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Hi,

We have submitted a draft: "A Framework for Internet Media Guides"
It is available at the official website.
http://www.ietf.org/internet-drafts/draft-nomura-mmusic-img-framework-00.txt

This draft is an update to "A Framework for Internet Program Guides".

Abstract

   This memo provides a framework for Internet media guides (IMGs). An
   IMG is a set of meta-data describing the features of multimedia
   content in order to subscribe to, manage and exchange multimedia
   content. We present an architecture, a protocol model and IMG data
   model for several scenarios.

Feedback is appreciated.
----------
Yuji Nomura  E-mail: nom@flab.fujitsu.co.jp 
Network Architecture Lab., Fujitsu Laboratories Ltd.
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--NextPart

A New Internet-Draft is available from the on-line Internet-Drafts directories.


	Title		: XML Schema for Media Control
	Author(s)	: O. Levin, S. Olson, R. Even
	Filename	: draft-levin-mmusic-xml-media-control-01.txt
	Pages		: 9
	Date		: 2003-3-6
	
This document defines an XML Schema for Media Control in a tightly 
controlled environment. The current version includes commands for 
managing of video streams only. Implementation of this schema for 
interactive video applications in SIP environments significantly 
improves user experience.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-levin-mmusic-xml-media-control-01.txt

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

	Title		: SDP: Session Description Protocol
	Author(s)	: M. Handley, V. Jacobson, C. Perkins
	Filename	: draft-ietf-mmusic-sdp-new-12.txt
	Pages		: 46
	Date		: 2003-3-6
	
This memo defines the Session Description Protocol (SDP). SDP
is intended for describing multimedia sessions for the
purposes of session announcement, session invitation, and
other forms of multimedia session initiation.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-new-12.txt

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From mailnull@www1.ietf.org  Fri Mar  7 11:28:56 2003
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Date: Fri, 07 Mar 2003 17:28:48 +0100
From: Joerg Ott <jo@tzi.uni-bremen.de>
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Subject: [MMUSIC] MMUSIC at 56th IETF: Agenda and Draft Charter
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Please find a draft agenda for our meeting in SFO at:

     http://www.dmn.tzi.org/ietf/mmusic/56/agenda.html
     http://www.dmn.tzi.org/ietf/mmusic/56/agenda.txt

Please note that not all drafts have been posted yet and
therefore some of the links in the HTML page are invalid.
This will be fixed as soon as the drafts become available.

Please direct change requests directly to the chairs.

Also: Please find a first draft for a revised charter at

     http://www.dmn.tzi.org/ietf/mmusic/draft-charter.html

Please note that this is a first cut and has not yet been
agreed upon with the ADs or the IESG.

Also note that following discussions with our ADs in the past,
conference control is *not* part of the revised charter.  We
will need to sort out where this work can find a home in the
future.  SIPPING may surely be a suitable candidate here, but
this is entirely open to further debate.

Colin and Joerg



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--NextPart

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

	Title		: Connection-Oriented Media Transport in SDP
	Author(s)	: D. Yon
	Filename	: draft-ietf-mmusic-sdp-comedia-05.txt
	Pages		: 14
	Date		: 2003-3-7
	
This document describes how to express media transport over 
connection-oriented protocols using the Session Description Protocol 
(SDP).  It defines two new protocol identifiers: TCP and TLS.  It 
also defines the syntax and semantics for an SDP 'direction' 
attribute that describes the connection setup procedure.

A URL for this Internet-Draft is:
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-comedia-05.txt

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

	Title		: Real Time Streaming Protocol (RTSP)
	Author(s)	: H. Schulzrinne et. al.
	Filename	: draft-ietf-mmusic-rfc2326bis-03.txt,.ps,.pdf
	Pages		: 5
	Date		: 2003-3-7
	
This memorandum is a revision of RFC 2326, which is currently a Pro-
posed Standard.
The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time proper-
ties. RTSP provides an extensible framework to enable controlled, on-
demand delivery of real-time data, such as audio and video. Sources
of data can include both live data feeds and stored clips. This pro-
tocol is intended to control multiple data delivery sessions, provide
a means for choosing delivery channels such as UDP, multicast UDP and
TCP, and provide a means for choosing delivery mechanisms based upon
RTP (RFC 1889).

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--NextPart

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

	Title		: Session Description and Capability Negotiation
	Author(s)	: D. Kutscher, J. Ott, C. Bormann
	Filename	: draft-ietf-mmusic-sdpng-06.txt
	Pages		: 47
	Date		: 2003-3-7
	
This document defines a language for describing multimedia sessions
with respect to configuration parameters and capabilities of end
systems. 
This document is a product of the Multiparty Multimedia Session
Control (MMUSIC) working group of the Internet Engineering Task
Force. Comments are solicited and should be addressed to the working
group's mailing list at confctrl@isi.edu and/or the authors.

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	Title		: RTSP Stream Switching
	Author(s)	: P. Gentric
	Filename	: draft-gentric-mmusic-stream-switching-00.txt
	Pages		: 26
	Date		: 2003-3-10
	
Stream switching is a technique used to change the data rate of a 
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Hi,

I have some comments on the COMEDIA draft. I still reading up on this so you will not get one of these 20+ comment letters that some other drafts has got. 

1. The draft seems very directed at SIP usage only. It does not really describe how to handle comedia in the context of SAP or for that matter RTSP. In SAP or other uni-directional use cases it will be required that the SDP describes a passive side as the server can never connect as it does not know of the client. In regards to RTSP the port number MUST not be 0 comes into conflict. It seems relevant that a RTSP controlled server actually offers a media block which uses reliable connection oriented transport. SO this needs to be addressed I think. 

2. The security consideration chapter seems to be missing the important hijacking warning. If I don't misunderstand section 4.6 a passive side actually starts sending to the address of the first received packet. This is very unsecure and easy to hijack. Should definitely be addressed in the  security consideration.

3. IANA Considerations. This chapter needs to better clarify what exactly is registered in what registries. I also think that the definitions of the new attributes should be done better using ABNF of RFC 2234 and draft-ietf-mmusic-sdp-new. 

Regards 


Magnus Westerlund 

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



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Hi Magnus, all,

You wrote as a summary:
 > I can see the usefulness of this proposal, however there is need to
 > resolve a number of issues:
 > - A general standardization or target certain URI schemes?
 > - Applicability on different media types and protocols needs to better
 > understood.
 > - Use of fragment or other URI syntax element, like parameter or query?
 > - Need for having ranges?
 > - The actual need for general parameters?

Great to hear you liked the idea. Many thanks for your elaborate answer 
and well thought through comments, which I am very happy to address.


1) General mechanism questions, applicability to different media types 
and protocols:

> First is the possibility to specify this as a general mechanism. It 
> seems that you want to have this to apply to all URI schemes. Is that 
> really possible? 

The question that you are addressing goes beyond whether our mechanism 
applies to all URI schemes. In 
http://www.w3.org/DesignIssues/Fragment.html Tim Berners-Lee states as 
an axiom that "The fragment ID spec for a new MIME type should be part 
of the MIME type registration process". This implies that the meaning of 
a fragment specification depends on the MIME type of the addressed 
resource. Therefore, the interpretation of a fragment has to be 
specified for each MIME type, and then probably again for each MIME type 
on each different scheme. Our I-D can only propose a generic way to 
specify time-offsets for time-continuous data and we are proposing it 
for http and rtsp where they make sense.

As an example take QuickTime files. Apple registered the MIME type 
video/quicktime but the documentation does not have a specification of 
how to interpret fragment parts when the files are used in URIs. Our I-D 
would not automatically enable timed fragment offsets to be used with 
QuickTime files - it needs an active move from Apple to change their 
MIME type specification. The same holds true for RealMedia and 
WindowsMedia and any other time-continuous MIME type. In fact, I believe 
that no MIME type registration except for text/html includes a fragment 
interpretation section.

 > How do you know that the use of @ will not collide with
> any usage?

The "@" sign is a reserved character in the URI Generic Syntax 
(RFC2396). It is allowed as part of a fragment, but its use is reserved 
for special cases. No special case of the use of "@" is know to us or 
published with IETF and therefore we asume there will be no collisions.

> For example the rtsp scheme does not define how to interpret 
> the fragment. So this is up to the server responding to a request. So 
> this can potentially already be used by some implementations.

As this has never been suggested before, there won't be any existing 
implementations that use it. Streaming servers should however find it 
easy, to support this mechanism as they only have to copy the 
specification after the "@" sign into the appropriate rtsp timestamp field.

> So a general question, what is the actually possibilities to standardize 
> such a solution?

It needs some minor adjustments in the URI standard. As that standard is 
currently being changed/adapted, this could be the right time to get it 
adjusted. Then our mechanism can be provided as a Generic Syntax 
proposal and MIME type specifications for time-continuous data can elect 
to support it. That's my take.


2) Questions on "fragment" definition, ranges:

> Second, is the usage of the fragment identifier. It seems that you tries 
> to redefine the meaning of the fragment identifier. See below.
> 
>  From RFC 2396:
> 
>  "When a URI reference is used to perform a retrieval action on the
>   identified resource, the optional fragment identifier, separated from
>   the URI by a crosshatch ("#") character, consists of additional
>   reference information to be interpreted by the user agent after the
>   retrieval action has been successfully completed.  As such, it is not
>   part of a URI, but is often used in conjunction with a URI.
> 
>      fragment      = *uric
> 
>   The semantics of a fragment identifier is a property of the data
>   resulting from a retrieval action, regardless of the type of URI used
>   in the reference.  Therefore, the format and interpretation of
>   fragment identifiers is dependent on the media type [RFC2046] of the
>   retrieval result.  The character restrictions described in Section 2
>   for URI also apply to the fragment in a URI-reference.  Individual
>   media types may define additional restrictions or structure within
>   the fragment for specifying different types of "partial views" that
>   can be identified within that media type."
> 
> Two things strikes me as not being taken into account by your draft in 
> relation to this quote. The first is from the first paragraph of the 
> quote. The fragment identifier is interpreted by the user agent. While 
> in chapter 4 of your draft you clearly specifies that the server shall 
> interpret the fragment. This redefines the behavior of the fragment 
> identifier.

We are aware that with the current URI specification, fragments only get 
interpreted on the user agent side. One way to overcome this problem is 
to use the Query part of the URI ("?") instead, but this creates other 
problems, iin the least overloading the meaning of "querying". 
Inherently, a temporal offset is a specification of a "fragment" in its 
core meaning and therefore we would prefer to use the "fragment" part. 
It makes everything much simpler. If there is no burning need to 
interpret the fragment identifier at the user agent, a change of the 
standard may be adequate. As the URI standard is currently reworked, we 
may get a chance to just do that (see URIBOF at next IETF meeting).

> Secondly the interpretation of the fragment identifier is in relation to 
> the data resulting in the request. So in fact it seems that you need to 
> define the behavior on per media basis.

Agreed.

> I think the use of fragment to specify where in a media resource is 
> appropriate in some case. For example an RTSP URL that has a fragment of 
> the proposed type is useful. However as RTSP works the server shall not 
> interpret the fragment as this is pointless. Only the client needs to do 
> this and use that to determine the range to include in the PLAY request. 
> So again the usage rules in chapter 4 seems to be wrong.

Thanks. We'll take that into account when working over it.

> It seems that the time specification should be complemented to allow 
> specifying ranges. The result of a search or the part of your friends 
> wedding video that you want to point out would in most cases only be a 
> part of the video. Therefore both a start and stop time seems useful. 
> Then it is also a very short step to having multiple ranges allow one to 
> point out multiple parts of the same resource that are interesting.

As we've had several requests for that feature, we should probably work 
that as an option into the next version of the document.

> Also I think you have a few strange thing in the definition. First you 
> specify an attribute "start" that I think are pointless. Either you 
> totally skip the fragment or you can use "npt=0". 

Possibly. But sometimes you have a file that does not start at 0 but 
rather with a time offset (because e.g. it is mirroring a timecode from 
an analog band from where it got digitized). In this case, "start" makes 
sense and does not default to "npt=0".

 > Secondly it the defaulting to "npt".
 > This clashed with the general parameters. If you
> use start then you have actually wrote "npt=start" which isn't defined. 

That's a matter of parsing sequence and probably adjustment of the BNF 
that we wrote in the document. The use of "npt" as default came from our 
experience with using the mechanism and the short form added much usability.

> "now" works as long as this is based on the RTSP definitions of NPT, 
> however from your BNF definitions this is evidently not the case. So my 
> proposal is to skip having any default time scheme if you are going to 
> have general attributes that does not belong to a time scheme.

I guess, a change of the BNF definition is more appropriate. Also, 
"start" and "now" seem to be useful for streaming and thus should be 
covered in URIs, too.


Best Regards,

Silvia.
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Hi Silvia,

First I have a question on how you are going to go forward in IETF. Are 
you planning to take this work into any WG or do a individual submission?

Then more comments below:

Silvia.Pfeiffer@csiro.au wrote:

> 1) General mechanism questions, applicability to different media types 
> and protocols:
>
>> First is the possibility to specify this as a general mechanism. It 
>> seems that you want to have this to apply to all URI schemes. Is that 
>> really possible? 
>
>
> The question that you are addressing goes beyond whether our mechanism 
> applies to all URI schemes. In 
> http://www.w3.org/DesignIssues/Fragment.html Tim Berners-Lee states as 
> an axiom that "The fragment ID spec for a new MIME type should be part 
> of the MIME type registration process". This implies that the meaning 
> of a fragment specification depends on the MIME type of the addressed 
> resource. Therefore, the interpretation of a fragment has to be 
> specified for each MIME type, and then probably again for each MIME 
> type on each different scheme. Our I-D can only propose a generic way 
> to specify time-offsets for time-continuous data and we are proposing 
> it for http and rtsp where they make sense.
>
> As an example take QuickTime files. Apple registered the MIME type 
> video/quicktime but the documentation does not have a specification of 
> how to interpret fragment parts when the files are used in URIs. Our 
> I-D would not automatically enable timed fragment offsets to be used 
> with QuickTime files - it needs an active move from Apple to change 
> their MIME type specification. The same holds true for RealMedia and 
> WindowsMedia and any other time-continuous MIME type. In fact, I 
> believe that no MIME type registration except for text/html includes a 
> fragment interpretation section.


Thanks, for the explanation, it greatly clarifies this. However to my 
understanding the usage of MIME types is actually depending on the URI 
scheme. One simple example is to compare HTTP and RTSP. In HTTP you: 1) 
download the object specified by the URI. 2) Look at the MIME type of 
the downloaded object and sends it to its parser together with the 
fragment. 3) The parser takes the fragment into account when displaying 
the content.

However for RTSP URI with a fragment one possible interpretation is to 
setup the presentation indicated by the URI. This setup results in that 
a media description is retrieved. This description has of course a MIME 
type. Is it to this description the the fragment applies? Or is it to a 
the presentation in general which does not have a MIME type, or is it to 
the media streams which can contain one or more different encodings 
perhaps identified by MIME types.

My point with this example is that the schemes seems to be taken into 
account. For RTSP it seems in fact that it is time to start defining 
some fragment identifier behavior. 

>
> > How do you know that the use of @ will not collide with
>
>> any usage?
>
>
> The "@" sign is a reserved character in the URI Generic Syntax 
> (RFC2396). It is allowed as part of a fragment, but its use is 
> reserved for special cases. No special case of the use of "@" is know 
> to us or published with IETF and therefore we assume there will be no 
> collisions.

Ok.

>
>> For example the rtsp scheme does not define how to interpret the 
>> fragment. So this is up to the server responding to a request. So 
>> this can potentially already be used by some implementations.
>
>
> As this has never been suggested before, there won't be any existing 
> implementations that use it. Streaming servers should however find it 
> easy, to support this mechanism as they only have to copy the 
> specification after the "@" sign into the appropriate rtsp timestamp 
> field.


I don't think we need to consider servers in this case, rather the 
clients as fragments should have been treated in accordance with the URI 
specification saying that fragment is a user agent problem.

>
>> So a general question, what is the actually possibilities to 
>> standardize such a solution?
>
>
> It needs some minor adjustments in the URI standard. As that standard 
> is currently being changed/adapted, this could be the right time to 
> get it adjusted. Then our mechanism can be provided as a Generic 
> Syntax proposal and MIME type specifications for time-continuous data 
> can elect to support it. That's my take.


I am not fully aware of what is happening with this. However I have seen 
a draft for making URI generic syntax to standard. Is that what you have 
in mind? To my knowledge the changes performed on a specification when 
going from draft to full standard are only clarifications. Technical 
changes are very limited if allowed at all.

I think that any such extension as you are proposing will be a proposed 
standard. 

>
>
> 2) Questions on "fragment" definition, ranges:
>
>> Second, is the usage of the fragment identifier. It seems that you 
>> tries to redefine the meaning of the fragment identifier. See below.
>>
>>  From RFC 2396:
>>
>>  "When a URI reference is used to perform a retrieval action on the
>>   identified resource, the optional fragment identifier, separated from
>>   the URI by a crosshatch ("#") character, consists of additional
>>   reference information to be interpreted by the user agent after the
>>   retrieval action has been successfully completed.  As such, it is not
>>   part of a URI, but is often used in conjunction with a URI.
>>
>>      fragment      = *uric
>>
>>   The semantics of a fragment identifier is a property of the data
>>   resulting from a retrieval action, regardless of the type of URI used
>>   in the reference.  Therefore, the format and interpretation of
>>   fragment identifiers is dependent on the media type [RFC2046] of the
>>   retrieval result.  The character restrictions described in Section 2
>>   for URI also apply to the fragment in a URI-reference.  Individual
>>   media types may define additional restrictions or structure within
>>   the fragment for specifying different types of "partial views" that
>>   can be identified within that media type."
>>
>> Two things strikes me as not being taken into account by your draft 
>> in relation to this quote. The first is from the first paragraph of 
>> the quote. The fragment identifier is interpreted by the user agent. 
>> While in chapter 4 of your draft you clearly specifies that the 
>> server shall interpret the fragment. This redefines the behavior of 
>> the fragment identifier.
>
>
> We are aware that with the current URI specification, fragments only 
> get interpreted on the user agent side. One way to overcome this 
> problem is to use the Query part of the URI ("?") instead, but this 
> creates other problems, in the least overloading the meaning of 
> "querying". Inherently, a temporal offset is a specification of a 
> "fragment" in its core meaning and therefore we would prefer to use 
> the "fragment" part. It makes everything much simpler. If there is no 
> burning need to interpret the fragment identifier at the user agent, a 
> change of the standard may be adequate. As the URI standard is 
> currently reworked, we may get a chance to just do that (see URIBOF at 
> next IETF meeting).


For me it seems very bad step to redefine this. I think one should look 
at both use cases. There will still be plenty of reasons to use fragment 
on the client side. And as there is a large number of HTTP clients out 
there using the current definition it seems a bad move to redefine it.  
However one should also look into how to address this when one wants the 
server to perform the action. I can clearly understand the use case of 
having a server reduce large media files to only contain what one is 
interested of. 

I looked through the URIBOF agenda: 
http://www.ietf.org/ietf/03mar/uribof.txt . I could however not see that 
this issue would be part of the BOF. Are you planning to take up your 
proposal there?

Best Regards

Magnus Westerlund 

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



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All,

regarding:

56th IETF: Draft MMUSIC Agenda
==============================

Monday, 17 March 2003, 0900 - 1130, Continental 8/9
---------------------------------------------------


1120         RTSP Stream Switching                                             (Gentric, 10)
             draft-gentric-mmusic-stream-switching-00.txt

*****************

this presentation will:
* state that stream switching control is the missing building block
in order to achieve "Streaming with Rate and Congestion Control"
* skip as out of focus some associated issues(rate control, session description/negociation)
* rapidly describe stream switching control key issues
* outline the proposed solution
* ask for feedback regarding the suitability of this topic as a working group item.

*************************************************

In the meanwhile I will be glad to answer questions about the draft
which is available at:

http://www.ietf.org/internet-drafts/draft-gentric-mmusic-stream-switching-00.txt


regards,


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com

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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:

http://sf.net/projects/rtspspec

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



Today's Topics:

   1. [ rtspspec-Bugs-640848 ] umbiguity in aggregate/non-aggregate pla (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Wed, 12 Mar 2003 06:45:23 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-640848 ] umbiguity in aggregate/non-aggregate pla

Bugs item #640848, was opened at 2002-11-19 19:24
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=640848&group_id=23194

Category: None
Group: None
Status: Open
Resolution: None
Priority: 5
Submitted By: Go Hori (ghori)
Assigned to: Nobody/Anonymous (nobody)
Summary: umbiguity in aggregate/non-aggregate pla

Initial Comment:
It is not clear to me who's choice it is to pick
aggregate play vs. non-aggregate play.  


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

>Comment By: Magnus Westerlund (magwes)
Date: 2003-03-12 15:45

Message:
Logged In: YES 
user_id=302620

It is the clients. With the updated specification the client
chooses this by setting up multiple streams or not in a each
session. 

This has no impact on a server that supports aggregated
control. 

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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=640848&group_id=23194




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Subject: [MMUSIC] RTSP Slides for MMUSIC WG meeting & AD-hoc RTSP meeting
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Hi,

Here comes an initial version of the slides that I will use on monday. 
The slides are for both the RTSP Update and the RTSP & NAT draft, 
draft-ietf-mmusic-rfc2326bis-03.txt and 
draft-ietf-mmusic-rtsp-nat-00.txt. The slides can be retrieved at:

http://standards.ericsson.net/westerlund/RTSPatIETF56.pdf

Please look through them before the meeting because I will need to have 
a certain tempo in the presentation and will not go into much detail. So 
if you have any input into anything please raise them on the mailing 
list or interrupt me during the presentation.

We have also planned to have two RTSP ad-hoc meetings during the 
meeting. The first is already scheduled to Sunday (16/3) at 21.00 in the 
bar. The location is the hotel bar. If there are multiple ones, a 
clarifying message will be sent to the list as soon as we have managed 
to scout the location. We will also in that case put up a note on the 
message board. We will at that meeting schedule the second one and 
present the actual time at the WG meeting.

Best Regards

Magnus Westerlund 

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


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Hi,

If people have had problems with the below file please download it again.

> http://standards.ericsson.net/westerlund/RTSPatIETF56.pdf


An ASCII FTP transfer corrupted it. This is now fixed

Best Regards

Magnus Westerlund 

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



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Subject: [MMUSIC] Comments on: draft-gentric-mmusic-stream-switching-00.txt
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Hi,

I have some comments on this proposal. My general comment is that the need for such a solution as outline in the proposal is not very present.

The draft includes two different mechanisms. 
1. A way of selecting streams to switch between, i.e. SWITCHSETUP and SWITCHCLOSE. 
2. A way of signaling requests and notifications on happening or soon to happen switches, i.e. SWITCH and SWITCHSIGNAL. 

Functionality one. The basic problem here is actually one of capability exchange. A streaming server needs to now what capabilities a client has so the offered streams does not go outside what the client can handle. However this problem is much larger than simple selecting what streams to switch between. It also includes media codecs, signaling extensions that are supported, etc. If one has a general mechanism for capability exchange one can ensure before setting up the session that one's client can handle what the server will send. That way a server can perform switching without having to worry if the client can handle it. 

I think mechanisms to handle this general problem are starting to come into being. The first component is a more capable session description protocol which supports capability negotiation, i.e. SDP-NG. Then we will need to look at how we can use SDP-NG in the context of RTSP. I think it will be necessary to provide some extensions in functionality, perhaps as simple as providing a back channel for the SDP-NG response in SETUP. 

My conclusion is that making a narrow solution that only solves a little piece of the problem is no good. Lets fix this properly. Until then I think the current mechanisms work sufficiently well. What the draft misses to point out is that a client already today are very capable of handling adaptation of bit-rates for streams. As pointed out by the draft what it calls client transparent switches works very well. These changes that are within the defined payload format and codec configuration can in certain cases be very large and in other none. 

For the non-transparent cases, i.e. needs to use multiple RTP payload types to signal which configuration to use, is not a major problem from standards point. If one accepts to start a session that contains a SDP with multiple PTs one needs to handle this. I can understand that it puts some pressure on the client implementations to handle multiple codecs. However in most case I think this is simple a question of different configurations, for example AAC and MPEG4 Visual streams. I don't think as a general case that one may not perform announced and potential a codec change / reconfiguration without explicit approval from the client. The client accepted to handle this by setting up the session. 

Looking at functionality 2: The signaling of switching commands. Lets start with the service SWITCHSIGNAL provides. This is used to notify that a switch will take place. I don't see how signaling this in advance can help. First, in most case you can't determine that you are going to switch any significant time in advance. The reason of course being that switching is done to react to changes in the network. When changes happens you need to switch quickly, there is no time to wait two seconds after a RTSP command has gone out. So therefore its better to switch directly. A transparent switch does not effect client in any way. A non-transparent may effect the client if its built on dynamic loading of codecs, etc. In this case you will still receive some heads up notice. This due to the clients buffer. When the packets start arriving with a different RTP PT the client can detect this directly and start preparing. The time in advance will depend on the amount of buffering left at !
that time. Another aspect of SWITCHSIGNAL is that it may not arrive before the actual media packets that represent the switch. 

One argument that might be used for the usefulness of SWITCHSIGNAL is the synchronization information carried in it. However I don't see any need. As long as media playout are continuos the existing timeline will continue to exist and provide synchronization. 

Lets continue with SWITCH. This being a client to server command to change to another stream. In my work with the 3GPP PSS adaptation work I have come to the conclusion that server controlled adaptation is the best thing. Reasons are:

1. In a best effort network the sending side are responsible for what it sends out on the network. This is as pointed out the model used by all IETF congestion control mechanisms. Server controls allows for halting transmission if feedback stops. 

2. Server has the knowledge where it is suitable to switch. It simply has the media and all its meta data available and can therefore best determine when and how to switch.

3. There are fewer servers in both implementations and actual numbers. This makes bug fixes updates of algorithms etc. much easier. 

4. Server has knowledge when packets where sent. This allows it to take client feedback into that context. Doing it the other way seems to require more signaling. 

5. It can be deployed in a backwards compatible way. As soon as a server support adaptation it can start performing it. The basic feedback will be RTCP which hopefully most support and sends. However a adaptation solution based on client side requires both client and server to be updated. 

Of course in some cases it would be simpler if the client took the decision but from my point of view it is in favor for the server controlled solution. 

Another point is that it already exist RTSP ways to influence the server on what to send. The Bandwidth header can be included in PLAY, or SET_PARAMETER requests to tell the server what available bit-rate it sees. The server can then adapt its transmission to what the client claims and then verify it by looking at RTCP feedback. 

Further the adaptation is not as simple as switching between different encodings. The server can actually perform other actions that change the sent bit-rate. One good example is stream thinning. I could for example reduce a video sequence with 5% by simply removing some frames. Done in an optimal way it would not degrade the quality very much. Perhaps simply a little lower frame rate.

Finally I would like to make some explicit remarks about a few thing written in the draft.

1. Section 1.5 bullet 1: Assumed that bit-rate will be constant. I have not made that assumption in a long time. However given a certain bit-rate it is best to utilize it to the maximum, and that means producing constant bit-rate streams for that bit-rate. 

2. Section 2.1.2 The MUST NOT in first bullet: In some sense I would have to agree with you that the client must agree on the codecs/configuratins to use. However that is done by any client setting up a session which SDP file specify this. 

My point with this rather long mail is that I don't think the proposal is appropriate and better ways of handling the problems exist or are under development. Not all pieces may be in place and work may be needed. However I think that it should be performed in areas that have more general application and helps fix larger problems. 


Best Regards

Magnus Westerlund 

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



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Subject: [MMUSIC] RTSP and NAT - More issues found with symmetric RTP
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Hi,

When I wrote the slides for the presentation I identified one more 
problem in the RTSP and NAT draft. This concerns symmetric RTP.

What I found is that symmetric RTP has the same problems with different 
IP addresses that a STUN based solution would have. We must also enforce 
that media shall only be sent to an IP address that match the RTSP 
signaling connection. The attack that forces us to take this stand also 
for symmetric RTP looks like this:

This attack is how you can use the server to perform DDOS attacks. The 
attacker connects to the RTSP server and request that at session should 
determine source address based on symmetric RTP. Then the attacker 
knowing the shared secret sends RTP and RTCP binding packets to the 
server ports with spoofed IP addresses of the victim of  the DOS attack. 
Nice and simple and with minimum effort from attacker.

When we are on the topic of using RTSP to perform denial of service 
attacks: Couldn't RTCP be used to reduce the impact of these attacks. IF 
one requires to receive RTCP messages from the receiver one can limit 
which can perform this attack. If one needs to send RTCP about the media 
stream actually sent the attacker must be able to eavesdrop the media 
stream from server towards victim to be able to generate correct RTCP 
reports. The server would be responsible to cut the media stream if no 
RTCP receiver reports corresponding to recently sent packets are 
received. I know that this only provides partial security. Just fishing 
for possible solutions.

Best Regards

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
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Subject: Re: [MMUSIC] Comments on: draft-gentric-mmusic-stream-switching-00.txt
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: IETF MMUSIC WG <mmusic@ietf.org>
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Magnus,

thanks a lot for your feedback!

a long reply to a long comment ...

**************************************************

you write:

>The draft includes two different mechanisms.
>1. A way of selecting streams to switch between, i.e. SWITCHSETUP and SWITCHCLOSE.
> [......]
>Functionality one. The basic problem here is actually one of capability exchange.

There is a misunderstanding here!

Indeed there is a capability exchange issue associated with stream switching  (And I agree that SDP-NG could help here )
...but not only the draft does not pretend to cover this at all
it is also intended to propose a solution independently from these issues!

specifically neither SWITCHSETUP nor SWITCHCLOSE are for capability exchange!

* SWITCHSETUP is actually exactly like SETUP, as mentionned in the draft,
the point of introducing SWITCHSETUP is only
a) to make it explicit for alternate streams
b) to improve backward compatibility

* SWITCHCLOSE is for ressource garbage collection only.
(BTW it is a well know problem of RTSP that you can SETUP dynamically but
that TEARDOWN kills the whole session)

>Lets start with the service SWITCHSIGNAL provides.
>This is used to notify that a switch will take place.
>I don't see how signaling this in advance can help. First, in
>most case you can't determine that you are going to switch any
>significant time in advance. The reason of course
>being that switching is done to react to changes in the network.

I agree that for switching in *very* short notice SWITCHSIGNAL does not help...

this is in my view the difference between *rate* control and *congestion* control:
in an "emmergency" (congestion control) you want to switch down fast,
but when the network is stable you want to slowly explore "up" (TFRC, etc)
and you will have cases when the system is slowly drifting because the streaming rate
is only slitghtly too large for the network, and then you can have many seconds to react ...
(if you have a way to react!)

also remember that the target is *seamless* switching, in an emmergency (congestion control)
you want to preserve the network, seamless-ness is secondary, but outside of emmergencies
there are cases where you want to a really smooth playback and SWITCHSIGNAL is intended for that:

>One argument that might be used for the usefulness of
> SWITCHSIGNAL is the synchronization information carried in it.

YES!

> However I don't see any need. As long as media playout are
> continuos the existing timeline will continue to
>exist and provide synchronization.

Yes, true, but tricky and also with an fundamentaly important exception for audio
with multiple frames per packet, which makes SWITCHSIGNAL very useful:

In that case in order to synchronize the decoder needs to get *exact* RTP time stamp values
because 2 streams encoded from the same content at different bit rates will produce different
packetization, so that the decoder needs to discard a certain number of audio *frames* from
the first packet of the new streams in order to get perfect lip-sync ...

>Lets continue with SWITCH. This being a client to server command to
> change to another stream. In my work with the
>3GPP PSS adaptation work I have come to the conclusion
> that server controlled adaptation is the best thing.

First I completely agree that in some cases server control is best.
as explained above SWITCHSIGNAL is actually provided solely
as an (optional) helper for that case !

The point however is that in *some* cases client control is  also nice.

******One key case is when RTCP feedback is blocked***********
(unfortunately I forgot to mention that one in the draft!!)

You will have to recognize it is also *very* natural
from the RTSP "philosophy" point of view!

>1. In a best effort network the sending side are responsible for
> what it sends out on the network. This is as
>pointed out the model used by all IETF congestion control
> mechanisms. Server controls allows for halting
>transmission if feedback stops.

I understand that, and for that reason
a) SWITCH is OPTIONAL
b) servers can deny SWITCH requests (especially up)
c) in all cases servers *can* stop sending when the RTCP feedback (or absence of) indicates congestion

the point is rather that in less dramatic cases you want to provide a smooth transition
(a seamless switch).

>2. Server has the knowledge where it is suitable to switch.
>It simply has the media and all its meta data available
>and can therefore best determine when and how to switch.

true, the server is also a mission critical device which economics are to be able
for a given hardware to support as many concurrent sessions as possible,
for that reason there are people (like us) who prefer to have the client make
the rate control computation and heuristics for target stream selection
(a bit like web servers use cookie to save the context at the client side)

>3. There are fewer servers in both implementations and actual numbers.
> This makes bug fixes updates of algorithms etc. much easier.

Hum, maintenance and upgrades looks to me like a much more complicated issue than that...

anyway *everything* is optional in my proposal so ...
you can upgrade client first, or server first as you like;-)

>4. Server has knowledge when packets where sent.
> This allows it to take client feedback into that context.
> Doing it the other way seems to require more signaling.

again, maintaining a lot of context in the server may not be a good idea for everybody...

also I dont see the logic here since the first place where congestion is detected
is always the client (with one exception: TRIGTRAN, but that is yet another animal!)
plus the fact that the player buffer content is also an important parameter.

>5. It can be deployed in a backwards compatible way. As soon as a
> server support adaptation it can start performing it.

well again, same here !

> The basic feedback will be RTCP which hopefully most support and sends.

I am sorry hope is not enough for me:
actually THAT is a key motivation for client side command i.e. "SWITCH" command.

the possibility that RTCP feedback is blocked is a very real one.

> However a adaptation solution based on
>client side requires both client and server to be updated.

well, yes but by experience I am not so optimistic about a client which has not been designed
implemented and tested for it surviving a switch from a server!-)

Anyway one key point is that my proposal is 100% backward compatible:

* "old" servers that do not implement it will ignore SWITHSETUP and SWITCH ("501 not implemented")

* "old" clients that do not implement it will not send SWITHSETUP and therefore the server will *know*
that the client does not support Stream Switching

and *this* is in contrast with the solution you propose where a server would switch without even knowing if a client supports it!

****************************

you also write:

>Until then I think the current mechanisms work sufficiently well.
> What the draft misses to point out is that a client already today
> are very capable of handling adaptation of bit-rates for streams.
> As pointed out by the draft what it calls client transparent
> switches works very well.

I am very surprised you can say that "it works very well"!
Can you point out interroperability test reports that would have escaped my attention?-)

The draft does not say it works well it says it works "in theory" and I am
of the kind who beleives there is a large gap between theory and practice ;-)

But OK, the point of the draft is also to aim for the more general solution because
the bit rate ranges reachable with the client-transparent modes are *just not good enough.*

The aim is definitively to be able to switch from a few Mb/s to a few kb/s,
and for that you will have drastic configuration changes (codecs,
screen sizes, number of audio channels, you name it)

>Another point is that it already exist RTSP ways to influence the server
> on what to send. The Bandwidth header can be included in PLAY,
> or SET_PARAMETER requests to tell the server what available bit-rate it sees.

sorry but I like to call a cat a cat ;-)

if it is a switch why not call it "SWITCH" ?????????

also there are 2 dozens of different such tricks being proposed/used by now!

the *point* would be that everybody uses the same one !

> The server can actually perform other actions that change the sent
> bit-rate. One good example is stream thinning. I could for example
> reduce a video sequence with 5% by simply removing some frames.
> Done in an optimal way it would not degrade the quality very much.
> Perhaps simply a little lower frame rate.

yes, it works but it is transparent only by removing B frames, which covers a very narrow bandwidth range ...


anyway this is something, as you point out, for which no standardization is needed anyway ...


>1. Section 1.5 bullet 1: Assumed that bit-rate will be constant.
> I have not made that assumption in a long time.
>However given a certain bit-rate it is best to utilize
>it to the maximum, and that means producing constant
>bit-rate streams for that bit-rate.

AH But *that* is the point: to leverage on existing codec technology which has been
optimized for constant bit rate ! ... on networks with variable band width !

>2. Section 2.1.2 The MUST NOT in first bullet: In some sense I would
> have to agree with you that the client must agree on the codecs/configuratins
> to use. However that is done by any client setting up a session which
> SDP file specify this.

True,

however as explained I want to give some more flexibility here,
**just as SETUP can be performed at anytime**
I want to be able to SWITCHSETUP at anytime

By symmetry I also want to be able to SWITCHCLOSE at anytime
because the amount of ressources involved can be pretty large.

an example: a streaming service is nominal at 2 Mb/s, but
one client is apparently with a bad network, the session
has been rapidly switching down to 500 kb/s in the first
seconds of streaming and is now stable for several minutes:

I want to have the possibility that in the policy of the server
the server "decides" that this guy will not be able to get
2 Mb/s anymore (except maybe by restarting another session)
so my server garbage-collects the resources for streaming this
guy at 2 Mb/s (which involve a lot of saved space in cache etc)
plus it sends a SWITCHCLOSE to tell the client to do the same!
- which saves ressource on the client
- and if SWITCH is supported avoids it trying to switch up,
which would be denied anyway.


A probably more important example is a client running
low on ressources in a server-initiated-switch
scenario: the client sends a SWITCHCLOSE to the server to tell it
"dont switch to this bit rate,
not only would it waste BW
but playback would be interrupted,
because I had to garbage collect this codec."



> I don't think the proposal is appropriate and better ways of
> handling the problems exist or are under development.

I wish you could be more specific?

> Not all pieces may be in place and work may be needed.

oh yes !

> However I think that it should be performed in areas
> that have more general application and helps fix larger problems.

is this about SDP NG? (then I agree)
however again, my proposal is mostly orthogonal to session description/negociation...

Maybe I should have called it "stream switching control ?"

******************************************

regards,


Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com


                                                                                                                                              
                                                                                                                                              
                                                          To:   IETF MMUSIC WG <mmusic@ietf.org>                                              
                                                          cc:   (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                    
                                                          Subject:    [MMUSIC] Comments on: draft-gentric-mmusic-stream-switching-00.txt      
                                                                                                                                              
               Magnus Westerlund                          Classification:                                                                     
               <magnus.westerlund@era.ericsson                                                                                                
               .se>                                                                                                                           
                                                                                                                                              
               Sent by:                                                                                                                       
               mmusic-admin@ietf.org                                                                                                          
                                                                                                                                              
               13/03/2003 16:32                                                                                                               
                                                                                                                                              
                                                                                                                                              




Hi,

I have some comments on this proposal. My general comment is that the need for such a solution as outline in the proposal is not very present.

The draft includes two different mechanisms.
1. A way of selecting streams to switch between, i.e. SWITCHSETUP and SWITCHCLOSE.
2. A way of signaling requests and notifications on happening or soon to happen switches, i.e. SWITCH and SWITCHSIGNAL.

Functionality one. The basic problem here is actually one of capability exchange. A streaming server needs to now what capabilities a client has so the offered streams does not go outside what the client can handle. However this problem is much larger
than simple selecting what streams to switch between. It also includes media codecs, signaling extensions that are supported, etc. If one has a general mechanism for capability exchange one can ensure before setting up the session that one's client can
handle what the server will send. That way a server can perform switching without having to worry if the client can handle it.

I think mechanisms to handle this general problem are starting to come into being. The first component is a more capable session description protocol which supports capability negotiation, i.e. SDP-NG. Then we will need to look at how we can use SDP-NG in
the context of RTSP. I think it will be necessary to provide some extensions in functionality, perhaps as simple as providing a back channel for the SDP-NG response in SETUP.

My conclusion is that making a narrow solution that only solves a little piece of the problem is no good. Lets fix this properly. Until then I think the current mechanisms work sufficiently well. What the draft misses to point out is that a client already
today are very capable of handling adaptation of bit-rates for streams. As pointed out by the draft what it calls client transparent switches works very well. These changes that are within the defined payload format and codec configuration can in certain
cases be very large and in other none.

For the non-transparent cases, i.e. needs to use multiple RTP payload types to signal which configuration to use, is not a major problem from standards point. If one accepts to start a session that contains a SDP with multiple PTs one needs to handle
this. I can understand that it puts some pressure on the client implementations to handle multiple codecs. However in most case I think this is simple a question of different configurations, for example AAC and MPEG4 Visual streams. I don't think as a
general case that one may not perform announced and potential a codec change / reconfiguration without explicit approval from the client. The client accepted to handle this by setting up the session.

Looking at functionality 2: The signaling of switching commands. Lets start with the service SWITCHSIGNAL provides. This is used to notify that a switch will take place. I don't see how signaling this in advance can help. First, in most case you can't
determine that you are going to switch any significant time in advance. The reason of course being that switching is done to react to changes in the network. When changes happens you need to switch quickly, there is no time to wait two seconds after a
RTSP command has gone out. So therefore its better to switch directly. A transparent switch does not effect client in any way. A non-transparent may effect the client if its built on dynamic loading of codecs, etc. In this case you will still receive some
heads up notice. This due to the clients buffer. When the packets start arriving with a different RTP PT the client can detect this directly and start preparing. The time in advance will depend on the amount of buffering left at !
that time. Another aspect of SWITCHSIGNAL is that it may not arrive before the actual media packets that represent the switch.

One argument that might be used for the usefulness of SWITCHSIGNAL is the synchronization information carried in it. However I don't see any need. As long as media playout are continuos the existing timeline will continue to exist and provide
synchronization.

Lets continue with SWITCH. This being a client to server command to change to another stream. In my work with the 3GPP PSS adaptation work I have come to the conclusion that server controlled adaptation is the best thing. Reasons are:

1. In a best effort network the sending side are responsible for what it sends out on the network. This is as pointed out the model used by all IETF congestion control mechanisms. Server controls allows for halting transmission if feedback stops.

2. Server has the knowledge where it is suitable to switch. It simply has the media and all its meta data available and can therefore best determine when and how to switch.

3. There are fewer servers in both implementations and actual numbers. This makes bug fixes updates of algorithms etc. much easier.

4. Server has knowledge when packets where sent. This allows it to take client feedback into that context. Doing it the other way seems to require more signaling.

5. It can be deployed in a backwards compatible way. As soon as a server support adaptation it can start performing it. The basic feedback will be RTCP which hopefully most support and sends. However a adaptation solution based on client side requires
both client and server to be updated.

Of course in some cases it would be simpler if the client took the decision but from my point of view it is in favor for the server controlled solution.

Another point is that it already exist RTSP ways to influence the server on what to send. The Bandwidth header can be included in PLAY, or SET_PARAMETER requests to tell the server what available bit-rate it sees. The server can then adapt its
transmission to what the client claims and then verify it by looking at RTCP feedback.

Further the adaptation is not as simple as switching between different encodings. The server can actually perform other actions that change the sent bit-rate. One good example is stream thinning. I could for example reduce a video sequence with 5% by
simply removing some frames. Done in an optimal way it would not degrade the quality very much. Perhaps simply a little lower frame rate.

Finally I would like to make some explicit remarks about a few thing written in the draft.

1. Section 1.5 bullet 1: Assumed that bit-rate will be constant. I have not made that assumption in a long time. However given a certain bit-rate it is best to utilize it to the maximum, and that means producing constant bit-rate streams for that
bit-rate.

2. Section 2.1.2 The MUST NOT in first bullet: In some sense I would have to agree with you that the client must agree on the codecs/configuratins to use. However that is done by any client setting up a session which SDP file specify this.

My point with this rather long mail is that I don't think the proposal is appropriate and better ways of handling the problems exist or are under development. Not all pieces may be in place and work may be needed. However I think that it should be
performed in areas that have more general application and helps fix larger problems.


Best Regards

Magnus Westerlund

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



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> Magnus Westerlund <magnus.westerlund@era.ericsson.se> writes:
>
> if you have any input into anything please raise them on the mailing 
> list or interrupt me during the presentation.

Slide 17, bullet 2:

The problem does not exist for the control signaling over TCP,
but for the media streams.

--                                   

This would seem to imply that interleaved media over the TCP
control signalling channel is also not a problem. This would
seem to be the actual "competition" for these NAT techniques --
is the savings in server overhead + the quality improvement
of UDP media flow over TCP media flow worth the added expense
of implementing these RTSP NAT techniques? I hope the answer  
is "yes", of course ... but I think its a question the skeptical
potential implementor may ask ...                               

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John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
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Subject: [MMUSIC] Re: [AVT] new I-D on temporal URI addresses
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Hi Magnus,

> First I have a question on how you are going to go forward in IETF. Are 
> you planning to take this work into any WG or do a individual submission?

I am most happy to take it into a WG but am uncertain which one that 
would be.

> Thanks, for the explanation, it greatly clarifies this. However to my 
> understanding the usage of MIME types is actually depending on the URI 
> scheme. One simple example is to compare HTTP and RTSP. In HTTP you: 1) 
> download the object specified by the URI. 2) Look at the MIME type of 
> the downloaded object and sends it to its parser together with the 
> fragment. 3) The parser takes the fragment into account when displaying 
> the content.

Not necessarily. That's just the way current browsers work because of 
the recommendations for the use of the fragment. I know for a fact that 
the Apache Web server can receive, parse and handle fragment offsets.

> However for RTSP URI with a fragment one possible interpretation is to 
> setup the presentation indicated by the URI. This setup results in that 
> a media description is retrieved. This description has of course a MIME 
> type. Is it to this description the the fragment applies? Or is it to a 
> the presentation in general which does not have a MIME type, or is it to 
> the media streams which can contain one or more different encodings 
> perhaps identified by MIME types.

That should be discussed, but my take is that it applies to each one of 
the media streams that are part of the general presentation.

> My point with this example is that the schemes seems to be taken into 
> account. For RTSP it seems in fact that it is time to start defining 
> some fragment identifier behavior.

I agree, we definately need to take the schemes into account when 
describing the expected behaviour of servers and clients when 
encountering these temporal offsets in URIs.

> I don't think we need to consider servers in this case, rather the 
> clients as fragments should have been treated in accordance with the URI 
> specification saying that fragment is a user agent problem.

It is important to consider servers, too. I does not make sense to 
burden the Internet or the user with a huge amount of video data when 
all that the user requires is a clip 20 minutes into the video. I am 
aware that the current use of fragments does not conform to this idea 
and therefore suggested to discuss the possibilities of changing that 
usage specification.

>>> So a general question, what is the actually possibilities to 
>>> standardize such a solution?
>>
>> It needs some minor adjustments in the URI standard. As that standard 
>> is currently being changed/adapted, this could be the right time to 
>> get it adjusted. Then our mechanism can be provided as a Generic 
>> Syntax proposal and MIME type specifications for time-continuous data 
>> can elect to support it. That's my take.
> 
> I am not fully aware of what is happening with this. However I have seen 
> a draft for making URI generic syntax to standard. Is that what you have 
> in mind? To my knowledge the changes performed on a specification when 
> going from draft to full standard are only clarifications. Technical 
> changes are very limited if allowed at all.

As far as I understand it, there are more changes than just 
clarifications happening to RFC2396. But if you are right, we are 
probably forced to go with the query "?" solution rather than fragments, 
which makes it a lot less elegant. Why should we be using a different 
scheme for addressing offsets in audio/video than in html? Also, the use 
of query is already overloaded and therefore a sensible use of fragments 
for media would make sense.

> I think that any such extension as you are proposing will be a proposed 
> standard.

I'd be more than happy to have it become that and through an IETF WG.

> For me it seems very bad step to redefine this. I think one should look 
> at both use cases. There will still be plenty of reasons to use fragment 
> on the client side.

Could you please tell me some? I tend to disagree. I believe that as 
time-continuous media are usually high data rates, fragment addressing 
of such data only makes sense if we can avoid having to serve out the 
complete file before we can address the offset. By the way, even with 
HTML I have often been upset about having to wait (especially on a modem 
line) for the complete document to download when all I wanted was to 
read a specific paragraph in it.

> And as there is a large number of HTTP clients out 
> there using the current definition it seems a bad move to redefine it.

No. That would  not affect them. It is still possible to say that if the 
user (agent) wishes to interpret the fragment offset locally, he's free 
to strip off the fragment offset from the URI (as happening today) and 
not sending that to the server. However, if a server receives a fragment 
offset request, he requires to send data only from that offset onwards.

> However one should also look into how to address this when one wants the 
> server to perform the action. I can clearly understand the use case of 
> having a server reduce large media files to only contain what one is 
> interested of.
> I looked through the URIBOF agenda: 
> http://www.ietf.org/ietf/03mar/uribof.txt . I could however not see that 
> this issue would be part of the BOF. Are you planning to take up your 
> proposal there?

I only found out about the URIBOF agenda after having submitted the I-D. 
I have however sent the I-D to the relevant mailing lists and hope to 
get them to pick it up and discuss the issues. I cannot attend the 
meeting personally, unfortunately.

Best Regards,

Silvia.

---
Silvia Pfeiffer
Research Scientist
CSIRO Australia
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Hi John,

John Lazzaro wrote:

>>Magnus Westerlund <magnus.westerlund@era.ericsson.se> writes:
>>
>>if you have any input into anything please raise them on the mailing 
>>list or interrupt me during the presentation.
>>    
>>
>
>Slide 17, bullet 2:
>
>The problem does not exist for the control signaling over TCP,
>but for the media streams.
>
>--                                   
>
>This would seem to imply that interleaved media over the TCP
>control signalling channel is also not a problem. This would
>seem to be the actual "competition" for these NAT techniques --
>is the savings in server overhead + the quality improvement
>of UDP media flow over TCP media flow worth the added expense
>of implementing these RTSP NAT techniques? I hope the answer  
>is "yes", of course ... but I think its a question the skeptical
>potential implementor may ask ...                               
>  
>

Quoting from my draft regarding usage of TCP tunneling:

3.4.3. Deployment Considerations

   Advantage:

   - Works through all types of NATs.

   Disadvantage:

   - Functionality needs to be implemented on both server and client.
   - May not give real-time performance.

So also this usage needs to be implemented. However I hope that a 
implementor are aware that there is very potential problems with the 
real-time performance when TCP is used.

Best Regards

Magnus Westerlund 

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



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Subject: [MMUSIC] A proposal for comedia-06 security section
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I've run a number of scenarios for the connection-hijack concern raised in 
comedia-fix, modeling the possible levels of connectivity between Alice, 
Bob, and Carol.  I believe that in every scenario except for one, this can 
be downgraded from a hijacking concern to a DoS concern.  I have PowerPoint 
prepared for Monday, but here is the discussion laid out in detail.

The backdrop is that Bob is assumed to follow the rules in section 4.4 (b) 
and (c), which means that a connection originating from Carol does *not* 
prevent Bob from attempting a connection.  This argues for 4.4 to be 
upgraded from RECOMMENDED to MUST.

Something not clarified in comedia-05 is that it would be helpful for an 
endpoint to extend the listener lifetime such that it maintains its 
listener for a short period of time after it received an initial connection 
in order to detect the possibility a duplicate connection may occur.  If a 
double
connect occurs that correlates to the session under negotiation, the 
endpoint SHOULD consider itself under attack and terminate the session.  In 
the case of bidirectional RTP/AVP, an endpoint that starts receiving two 
RTP streams from different addresses to the same port pair SHOULD consider 
itself under attack and terminate the session.

On the other side of the timeline, Bob should be mindful of the order in 
which an incoming connection occurs relative to providing an SDP reply to 
Alice.  If a connection arrives ahead of this event, Bob should be 
suspicious of the presence of Carol attempting to guess the address/port he 
is about to advertise in his reply.  The connection can either be discarded 
and Bob can continue, or he can opt to terminate the session.

Here are the connection scenarios:

FULL CONNECTIVITY:

Alice         Carol        Bob

Offer ----------------->
       <------CONNECT
??    <------------------ CONNECT
       <------------------ Answer
              CONNECT---->

=====================================

BOB/CAROL CANNOT CONNECT TO ALICE:

Alice         Carol        Bob

Offer ----------------->
            X--CONNECT
                       X-- CONNECT
       <------------------ Answer
              CONNECT---->
CONNECT ---------------->  ??

=====================================

ALICE/CAROL CANNOT CONNECT TO BOB:

Alice         Carol        Bob

Offer ----------------->
       <------CONNECT
??    <------------------ CONNECT
       <------------------ Answer
              CONNECT--X

=====================================

BOB CANNOT CONNECT TO ALICE:

Alice         Carol        Bob

Offer ----------------->
       <-------CONNECT
                       X-- CONNECT
       <------------------ Answer
              CONNECT---->
              Hijack!

=====================================

The ?? shows that Alice or Bob has detected a duplicate media connection 
and therefore has enough information to infer a possible attack.

The failure case is a common client/server scenario where the client 
(Alice) is behind a NAT/firewall and Bob (the server) is on the public 
Internet.  In many cases Bob knows that he is fully routable, and therefore 
SHOULD opt to advertise "connection:passive" which will cause Alice to 
discard Carol's connection once Alice receives Bob's SDP reply.

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Is this really a problem for SAP?  Presumably the server would be publicly 
routable and therefore it would make sense to advertise passive.

Do you have any suggestions re: the RTSP port-zero conflict?

Security considerations will come back in comedia-06.  See my other 
posting, and this will be discussed at the meeting.

If someone wants to give me more guidance on the IANA and ABNF (starting 
with some group consensus that this is really an issue to be address), I'm 
all ears.

At 12:12 PM 3/11/2003, Magnus Westerlund wrote:
>Hi,
>
>I have some comments on the COMEDIA draft. I still reading up on this so 
>you will not get one of these 20+ comment letters that some other drafts 
>has got.
>1. The draft seems very directed at SIP usage only. It does not really 
>describe how to handle comedia in the context of SAP or for that matter 
>RTSP. In SAP or other uni-directional use cases it will be required that 
>the SDP describes a passive side as the server can never connect as it 
>does not know of the client. In regards to RTSP the port number MUST not 
>be 0 comes into conflict. It seems relevant that a RTSP controlled server 
>actually offers a media block which uses reliable connection oriented 
>transport. SO this needs to be addressed I think.
>2. The security consideration chapter seems to be missing the important 
>hijacking warning. If I don't misunderstand section 4.6 a passive side 
>actually starts sending to the address of the first received packet. This 
>is very unsecure and easy to hijack. Should definitely be addressed in 
>the  security consideration.
>
>3. IANA Considerations. This chapter needs to better clarify what exactly 
>is registered in what registries. I also think that the definitions of the 
>new attributes should be done better using ABNF of RFC 2234 and 
>draft-ietf-mmusic-sdp-new.
>Regards
>
>Magnus Westerlund
>Multimedia Technologies, Ericsson Research ERA/TVA/A
>----------------------------------------------------------------------
>Ericsson AB                | Phone +46 8 4048287
>Torshamsgatan 23           | Fax   +46 8 7575550
>S-164 80 Stockholm, Sweden | mailto: magnus.westerlund@era.ericsson.se
>
>
>
>_______________________________________________
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Hi,

I have a a number of comments on this draft.

1. The short copyright statement are missing on the title page.

2. I am missing a chapter that clearly explains what one must define and 
think of if defining extensions to the framework defined by this draft. 
One thing I don't think the draft has considered sufficiently in form of 
how the text is written is how to adapt it to other profiles. First one 
mentions using this for non RTP transports, secondly it seems obvious 
that we should in the near future define the combination of AVPF and SAVP.

3. Chapter 2.0: I find the usage of sub chapter 0 a bit strange. I am 
definitely used to having 2, 2.1 or 2.1.1 type of difference between 
chapter levels.

4. Chapter 2.0 The second paragraph: Introduction chapter are normally 
informative and therefore one avoids all normative language. But this 
contain a normative SHOULD.

5. Section 4.1 Crypto suites for SRTP. I think this section is badly 
written it could be much clearer. To my eyes it looks like any crypto 
suite must contain a certain number of parameters. These parameters 
should be defined and explained. Then each definition can contain a note 
on usage and a table with the values. Last the defining new crypto suit 
then is a piece of cake as it simple requires any defined crypto suit to 
define values for all parameters. Last it should also define that there 
is a repository with IANA for crypto suits and that they should be 
registered.

6. Section 4.2: What is the reasons that one must have SRTP even if one 
is only receiving media streams and only sends SRTCP? Please provide 
motivation.

7. Section 4.3.2 second paragraph: I think it is a bit unclear when one 
uses right as telling which bytes to truncate. I would think that using 
end or last bytes is better. Also the inclusion of little end here is a 
bit strange. Where is this defined? It is not a term that I normally 
encounter. If you are going to use it it should be defined in the terms 
section.

8. Section 4.3.2 third paragraph || is defined. please put this also in 
a the definitions section.

9. Section 4.4.1: This section does not take into account that a single 
sender/issuer may use multiple SSRCs. This case needs to be handled as 
it is for example one that can result from employing RTP retransmission 
with SSRC-multiplexing.

10. Section 4.4.2: Isn't each ROC connected to a SSRC. Shouldn't they be 
bound together and allowed to exist in multiple cases to handle comment 9.

11. Section 5.0:  I know that you defines a new attributes for SRTP. 
However if one runs a totally defaulted session is it still impossible 
to use the k= attribute? I have understanding that a more competent way 
is needed.

12. Section 5.1: First sentence: "On each SDP media line (m=) where the 
<transport> is "RTP/SAVP", the offerer MAY follow that media line with 
at least one a=crypto line." Two things. First I think using better 
terms on how a line is included in the SDP should be used. Secondly the 
I think one should be as clear as possible about if one or more is 
allowed. From my view one or more is clearer than at least, of course 
context must be considered. However my proposal would be: "Each media 
description that on its m= lines has a transport parameter of 
"RTP/SAVP"  MAY contain one or more a=crypto line."

13. Section 5.2 first paragraph: I think you have an offer answer error 
here. Media streams must be included in the answer independently if one 
is accepting or not, unless the whole session is refused. So the last 
otherwise seems to be wrong. A refusal of a stream would be performed by 
setting port to 0.

14. Section 5.3: I think it is badly pointed out that the offerer must 
be responsible for accepting or refusing the answered security level.

15. Section 5: What is the potentials for performing bid-downs attacks 
when using this solution. It seems dependent on the security of the SDP 
transport level.

16. Section 5.6: How does one know from which packet the new keys apply. 
The synchronization between SIP and RTP are only loose. Wouldn't there 
be need for including a a complete ROI || seq per SSRC one rekeys?

17. Section 6.2 Second paragraph: Shouldn't the SHOULD here be a SHALL? 
Any use of these attributes in a SDP that are unprotected END-to-END is 
a fatal security breach.

18. Section 7.0: I have a number of comments on the grammar:
a. The grammar should define already today how the extension mechanism 
looks. What values that are allowed when defining new usage's. For 
example "application" should allow an "app-ext" that is then defined. 
This need for extensions are also true when it comes to cipher defintions.
b. One of the defined cipher suits are missing in the grammar.
c. The SSRC/ROC must be resolved in the ABNF

19. Section 11. You are missing the IPR statement, see ID-nits.


-- 

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
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Subject: [MMUSIC] RTSP meeting tonight
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Hi,

The RTSP meeting taking place today sunday at 21.00 will be in lobby bar.

Well meet

Regards 

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
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Subject: [MMUSIC] Comments on draft-ietf-mmusic-sdp-srcfilter-03.txt
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Hi,

A number of comments on this draft.

1. The title page headers looks bad. I would suggest that the author 
list is right justified. Also the expires date should be moved, see 
ID-nits for the recommendations.

2. The text in the whole draft should be indented in a good way. Normal 
running text is normally indented 3 characters. This makes the draft 
easier to read and find headings.

3. The conventions section is normally placed in its own section after 
the abstract. Location in regards to introduction varies and depends on 
content.

4. The short copyright statement is missing, see ID-nits

5. Section 2. This section should early declare if the attribute is 
allowed on session or media level or both.

6. Section 2.1, paragraph three: Does it really work to imply the number 
of address part of this source filter. Is it totally unforeseeable that 
a multi-layered media stream using multiple multicast addresses actually 
has different filters? If not, the case where one source filter contains 
the first address and another the third is ambiguous. It can either be 
that the first filter apply to the second address or it is not 
specified. Please clarify this rule. It can become problematic if one 
does not want to specify any address as forbidden and still becomes 
required to include the attribute. It might just be easier to include 
the number of addresses and explicit about it.

7. Is it never interesting to define the filters based on ports or port 
ranges?

8. Section 4 & 5. It is customary to have the reference list last to be 
easy to find. I would appreciate if it is moved to just before 
copy-right and IPR statements.

9. Section 6: One important security problem for this is how easy it is 
to do DOS by specifying filters that exclude the correct source. 
Therefore there is strong need for authentication and also a general 
warning that SDPs not coming from the session source may not be trusted.

10. The formulation of the IANA section might be better. See 
draft-ietf-avt-rtcp-feedback for a good IANA section.

11. Appendix A. The syntax used does not match the reference. ABNF 
according to 2234 uses "/" as alternative markers. Also SP is not 
defined in RFC 2327. Also the IPv6 address is not defined in RFC 2327. 
There exist other reference that can be used. See rfc 3266 or you could 
make this draft dependent on the SDP new draft which contains all the 
missing ABNF tags I think.


Best Regards


Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
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Magnus,

Thanks for the feedback.  Addressing the substantive (i.e., non-syntactic) 
comments:

>6. Section 2.1, paragraph three: Does it really work to imply the number 
>of address part of this source filter. Is it totally unforeseeable that a 
>multi-layered media stream using multiple multicast addresses actually has 
>different filters? If not, the case where one source filter contains the 
>first address and another the third is ambiguous. It can either be that 
>the first filter apply to the second address or it is not specified. 
>Please clarify this rule.

Good point.  The issue here is how to specify source filters if the SDP 
'connection address' specifies multiple multicast addresses, e.g.
         c=IN IP4 224.2.1.1/127/3

There are a number of possible ways to deal with this:

1/ Specify that the source filter line must use the first multicast address 
only, but that it applies to all of the possible multicast addresses in the 
range.  (E.g., in the example above, it would not be possible to specify 
different source filters for 224.2.1.1 and 224.2.1.3.  Also, it would be 
illegal for the source filter line to use any multicast address other than 
224.2.1.1.)

2/ Specify that the source filter line contains individual multicast 
addresses only.  To specify source filters for a range of multicast 
addresses, there must be an individual source filter line for each one - e.g.,
         a=source-filter: incl IN IP4 224.2.1.1 192.168.9.10
         a=source-filter: incl IN IP4 224.2.1.3 192.168.9.42
Note that in this example, there would *not* be any source filter for 
224.2.1.3.2

3/ Change the source filter specification to include the 
"/<number-of-addresses>" field.  But then, for consistency, should we also 
require the "/<ttl>" field, even though it's useless?  Also, what would the 
following mean:
         a=source-filter: incl IN IP4 224.2.1.2/127/2 192.168.9.10
         a=source-filter: incl IN IP4 224.2.1.1/127/3 192.168.9.42
and would the meaning change if the order of these two lines were reversed?

Personally, I think that option 1 is too restrictive, and option 3 far too 
complicated, and too difficult to understand.  For this reason, I'd prefer 
option 2: Require individual source filters for each multicast address in a 
range.  Any comments?


>7. Is it never interesting to define the filters based on ports or port 
>ranges?

(Presumably you mean *source* ports here, right?)  Noone has ever expressed 
an interest in specifying this, but if someone wanted this, I guess they 
could define a separate attribute for this - e.g., "a=source-port-filter:" 
or something...


>9. Section 6: One important security problem for this is how easy it is to 
>do DOS by specifying filters that exclude the correct source. Therefore 
>there is strong need for authentication and also a general warning that 
>SDPs not coming from the session source may not be trusted.

If one could generate a fake SDP description that specified a bogus source 
filter, then they could have just as easily specified bogus *destination 
address(es)*.  I.e., a SDP description always needs to be 'trusted', and 
the presence of source filter lines doesn't make any this more (or less) 
important

         Ross.

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Hi all,

Below are the minutes for the RTSP telecon from March 5.  Sorry for the
delay.

Rob
-- 
Rob Lanphier, Helix Community Coordinator - RealNetworks
http://helixcommunity.org http://rtsp.org http://realnetworks.com



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= Attendees =
 * Magnus Westerlund
 * Sean Sheedy
 * Colin Perkins
 * Tom Marshall
 * Rob Lanphier
 * Aravind Narasimhan
 * Thomas Zeng
 * Anders Klemets
 * Andrea Basso

= Minutes =

== Summary from last telecon ==
 * Magnus: (summarizes status of draft 03)
 * Magnus: please review for minor issues, etc
 * Magnus: we should take one meeting post-IETF to review draft03
 * Magnus: did you have time to review interleaving spec?
 * Aravind: not yet
 * Magnus: we've got a number of things we need to mention at next IETF meeting
 * Magnus: one thing we should point out that the URL BNF now contains fragment identifier
 * Magnus: removal of ANNOUNCE/RECORD
 * Magnus: rtspu and rtsps removed
 * Magnus: ssrc changes
 * (discussion involving specifying non-contiguous ports for eg. STUN.  Comma is already used in Transport: header as a separator, so / was proposed)
 * Thomas: is port-port correct?
 * Magnus: yes
 * Thomas: did we agree to include comma in port spec?
 * Thomas: what about colon
 * Magnus: used in MIME spedc
 * Magnus: we need to prepare IETF presentation next week.  will take to list.
 * Magnus: need to discuss NAT strategy
 * Magnus: need to schedule offline meetings
 * (scheduled meeting for Sunday 9pm, March 16)

== Bug #513737 ==
 * http://rtsp.org/bug513737
 * Sean: easier for error recovery if string is left alone
 * Magnus: parameters need to be changed in atomic way
 * Anders: is there a contradiction here?
 * Magnus: someone want to write this up?
 * Magnus: resolution, go with alternative #1 from bug report (1. Keep the old setting and states as if the second SETUP never happend.)
 * Magnus: next issue is one that is already closed
 * (skipping http://rtsp.org/bug539040)

==  Bug #539040 and  Bug #539040 ==
 * http://rtsp.org/bug539040
 * http://rtsp.org/bug539040
 * Magnus: how to use range when scale is backwards
 * Aravind: should always go from lower to higher
 * Aravind: e.g. to go backwards from 45 sec, say "-45s" (assuming BNF supports it)
 * Magnus: BNF supports
 * Thomas: I agree
 * Magnus: problem is range is specified as half open interval
 * Sean: I think we should play from first number to second number (opposite of what I'm hearing as predominant view)
 * Anders: agreed
 * Aravind: ok, makes sense
 * Thomas: makes sense to me too
 * Magnus: anyway, we need to resolve this
 * Magnus: agree that range is always from first value to second value
 * Aravind: if scale not there, and range is not there, then its an error
 * Magnus: yes
 * Andrea: is offer answer to be discussed?
 * Magnus: no, that's at next IETF meeting
 * Magnus: resolved: range is always from first value to second value
 * Aravind: if scale is not there, and range is bigger to smaller, then its an error
 * Magnus: any volunteers to write up?
 * Aravind: I can take this on
 * Magnus: need examples
 * Aravind: ok, I can do that

== Bug #556061 ==
 * http://rtsp.org/bug556061
 * Magnus: lack of clarity in state machine
 * Magnus: Tom, is the new REDIRECT text clear enough?
 * Tom: I think you need to allow REDIRECT at any time.  If a client can disconnect at any time, then the server needs to be able to redir at any time
 * Tom: the phrase i was looking for is connectionless control
 * Magnus: need to add REDIRECT into INIT state
 * Magnus: need to add server to client redirect possibility here
 * Magnus: should add some sentences in the REDIRECT section to clarify that this can be sent at any time.  Any comments on that?
 * Thomas: makes sense
 * Magnus: resolution: add text to REDIRECT section, and add to state machine
 * Aravind: clarify whether full session reestablishment is necessary
 * Magnus: yes it is, seems to be covered in 11.9
 * Andrea: need to reestablish?
 * Magnus: yes..one may have buffer depth to pull it off
 * Andrea: any provision for continuous service?
 * Magnus: no guarantees...example is redirect for end of live session.  Another is power failure.
 * Andrea: there's an implicit cost, because full setup is needed
 * Magnus: yes

== Bug #556295 ==
 * http://rtsp.org/bug556295
 * Client issues PAUSE while server ready
 * If a client issues a PAUSE and the server acknowledges and enters ready state, what is proper server response.  If the player issues another PAUSE?  There appear to be two possible correct behaviors:
   * server responds with OK because it does not change state and does no harm.
   * server responds with error 455 because PAUSE does not make sense in this state. 
 * Sean: I think that issuing PAUSE is an error, but I don't see any big advantage to returning an error, either.  Don't want to break implementations
 * Magnus: we can say that it is ok
 * Magnus: what about when there's a range header involved
 * Aravind: what about using it to reset pause point?
 * Magnus: pause is something you send after you are in the play state
 * Aravind: I don't see the problem with setting the pause point using pause
 * Aravind: we could break backwards compatibility if we issue error when getting pause in ready state
 * Thomas: are you saying that range doesn't matter?
 * Magnus: yes
 * Thomas: how does client know whether or not it has made a mistake in sending PAUSE in ready?
 * Magnus: pause response actuallly includes pause point
 * Tom: behavior with pause should mirror behavior with play
 * Tom: if pause while paused is allowed, play while playing is allowed
 * Thomas: we don't have queued play
 * Rob: why is symmetry important?
 * Thomas: it would make the spec make more sense
 * Rob: the reason for asymmetry with PAUSE is that it PAUSE in READY doesn't introduce same problems as PLAY while PLAYING
 * Aravind: just need clarity, not symmetry
 * Magnus: timed PAUSE request does introduce problems
 * Aravind: what happens when you have a timed pause request, and then a normal pause request, what happens?
 * Thomas: I think the earliest of the PAUSE requests is effective
 * Magnus: my interpretation of the spec is that all should be interpreted
 * Magnus: maybe we should eliminate timed PAUSE altogether
 * Aravind: we need to sort out dependence
 * Rob: range response is ambiguous in the case of timed PAUSE
 * Thomas: timed PAUSE doesn't seem useful
 * Tom: any use cases?
 * Aravind: maybe some live streams
 * Thomas: even that can be handled with PLAY with Range:
 * Magnus: one question: shoudl we remove time from the Range header, or should we say you can't use it
 * Andrea: do you have limit on range?
 * Magnus: looks like we should look into removing time
 * Magnus: next meeting, April 2?

== RTSP NAT ==
 * Thomas: Concerned about performance issues with having separate port for each RTP session
 * Colin: There are ways to alleviate the performance concerns.  Linux has a hashtable that you can stick into the poll processing
 * Tom:  Not always have choice about platform
 * Tom:  Solaris has /dev/poll, NT has callbacks
 * Colin:  Don't want to settle one something that is incompatible with RTP
 * Colin:  All the proposal I'm hearing on the mailing list look to be non-backwards compatible
 * Thomas:  Symmetric RTP is not backwards compatible.  We are proposing that
 * Magnus:  Issues it that we need to add some field in RTSP to enable NAT traversal
 * Colin:  We can probably talk about this for another hour.  . . .
 * Aravind:  I'm not sure what else was talked about, but people can add their comments to the minutes.


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Hi Ross,

See below:

Ross Finlayson wrote:

> Magnus,
>
> Thanks for the feedback.  Addressing the substantive (i.e., 
> non-syntactic) comments:
>
>> 6. Section 2.1, paragraph three: Does it really work to imply the 
>> number of address part of this source filter. Is it totally 
>> unforeseeable that a multi-layered media stream using multiple 
>> multicast addresses actually has different filters? If not, the case 
>> where one source filter contains the first address and another the 
>> third is ambiguous. It can either be that the first filter apply to 
>> the second address or it is not specified. Please clarify this rule.
>
>
> Good point.  The issue here is how to specify source filters if the 
> SDP 'connection address' specifies multiple multicast addresses, e.g.
>         c=IN IP4 224.2.1.1/127/3
>
> There are a number of possible ways to deal with this:
>
> 1/ Specify that the source filter line must use the first multicast 
> address only, but that it applies to all of the possible multicast 
> addresses in the range.  (E.g., in the example above, it would not be 
> possible to specify different source filters for 224.2.1.1 and 
> 224.2.1.3.  Also, it would be illegal for the source filter line to 
> use any multicast address other than 224.2.1.1.)
>
> 2/ Specify that the source filter line contains individual multicast 
> addresses only.  To specify source filters for a range of multicast 
> addresses, there must be an individual source filter line for each one 
> - e.g.,
>         a=source-filter: incl IN IP4 224.2.1.1 192.168.9.10
>         a=source-filter: incl IN IP4 224.2.1.3 192.168.9.42
> Note that in this example, there would *not* be any source filter for 
> 224.2.1.3.2
>
> 3/ Change the source filter specification to include the 
> "/<number-of-addresses>" field.  But then, for consistency, should we 
> also require the "/<ttl>" field, even though it's useless?  Also, what 
> would the following mean:
>         a=source-filter: incl IN IP4 224.2.1.2/127/2 192.168.9.10
>         a=source-filter: incl IN IP4 224.2.1.1/127/3 192.168.9.42
> and would the meaning change if the order of these two lines were 
> reversed?
>
> Personally, I think that option 1 is too restrictive, and option 3 far 
> too complicated, and too difficult to understand.  For this reason, 
> I'd prefer option 2: Require individual source filters for each 
> multicast address in a range.  Any comments?

I am fine with two. It will be clear and simple at the small cost of 
possibly  having to duplicate lines in cases the same filter applies to 
several address in a range.

>
>
>> 7. Is it never interesting to define the filters based on ports or 
>> port ranges?
>
>
> (Presumably you mean *source* ports here, right?)  Noone has ever 
> expressed an interest in specifying this, but if someone wanted this, 
> I guess they could define a separate attribute for this - e.g., 
> "a=source-port-filter:" or something...
>
Actually, no. I was thinking of destination ports. It was around the 
same lines of the multiple addresses in one line. I think it is very 
common that a single multicast address is used to deliver both audio and 
video. In this case these streams may be produced on different entities 
and would still likes to be the only valid source. It is a corner case, 
however I think it should be addressed in someway. Either by 
acknowledging that it is not possible or fix so that it works. I have no 
real view on the usefulness of these, I just want to ensure that it as 
been thought through.

>
>> 9. Section 6: One important security problem for this is how easy it 
>> is to do DOS by specifying filters that exclude the correct source. 
>> Therefore there is strong need for authentication and also a general 
>> warning that SDPs not coming from the session source may not be trusted.
>
>
> If one could generate a fake SDP description that specified a bogus 
> source filter, then they could have just as easily specified bogus 
> *destination address(es)*.  I.e., a SDP description always needs to be 
> 'trusted', and the presence of source filter lines doesn't make any 
> this more (or less) important

Okay, you might be right that it is no different than changing the 
destination address to one that any attacker control. The point being 
that SDPs should be authenticated.

Best Regards


Magnus Westerlund 

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




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Jonathan brought up the issue of forked-INVITE WRT my examples and the 
ability of Alice or Bob to detect a hijack attempt.  Thinking this through, 
I'm not yet convinced this is a issue for the security issue, but rather 
the forked-INVITE scenario is not explicitly addressed in comedia-05.

Here's a normal forked-INVITE exchange without an attacker (Alice, Bob, and 
Dan, where Bob "wins" the forked-INVITE):

Alice           Proxy       Bob            Dan

Offer   ------->
                  Offer ---->
                        |
                        +------------------>

        <--------------------CONNECT
        <-----------------------------------CONNECT

                        <----Answer
        <---------Answer
                        <------------------Answer
                  Cancel------------------->
        <x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-DISCONNECT

So the problem that is not addressed in comedia is that there may be <n> 
connections arrive as a result of a forked-INVITE, that will eventually 
unwind back to a single connection.  The question is: when and how does 
Alice know which of the connections will be the actual connection used for 
the session?  This is a real problem (oh boy, another reason to delay 
comedia!), so I think the question needs to have an answer.

I would submit that if you can answer the above question, then the 
scenarios I presented at the meeting are still valid, albeit somewhat 
simplified.  I.e., *eventually* Alice will know that it should have one 
connection but has more, and therefore can detect the attack.

And to throw more wood on the fire, it starts to get non-trivial when you 
consider cases like Bob's connection setup being blocked but not 
Dan's.  For example:

Alice           Proxy       Bob            Dan

Offer   ------->
                  Offer ---->
                        |
                        +------------------>

                          X--CONNECT
        <-----------------------------------CONNECT

                        <----Answer
        <---------Answer
                        <------------------Answer
                  Cancel------------------->
        <x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-x-DISCONNECT
(Alice: now what do I do?)


If this was not a forked-INVITE, Alice would know that she needs to attempt 
a connection to Bob, as a connection hadn't arrived when the Answer arrives 
via the proxy.  But here we have a connect succeed from Dan, then a short 
time later it terminates.  So how does Alice differentiate this 
forked-INVITE scenario from a failed media setup scenario?

Are we having fun yet? :-)

Temporarily flipping to a SIP-centric worldview, wouldn't one answer be to 
map comedia section 4 (b/c) to the ACK message from Alice rather than the 
OK reply from Bob?  This doesn't answer the question for two-way 
handshakes, but it seems to for three-way.

At 10:28 PM 3/14/2003, you wrote:
>\
>FULL CONNECTIVITY:
>
>Alice         Carol        Bob
>
>Offer ----------------->
>       <------CONNECT
>??    <------------------ CONNECT
>       <------------------ Answer
>              CONNECT---->
>
>
>The ?? shows that Alice or Bob has detected a duplicate media connection 
>and therefore has enough information to infer a possible attack.

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From: "Dan Wing" <dwing@cisco.com>
To: "David Yon" <yon.ietf@rfdsoftware.com>, <mmusic@ietf.org>
Subject: RE: [MMUSIC] A proposal for comedia-06 security section
Date: Mon, 17 Mar 2003 10:26:28 -0800
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...
> If this was not a forked-INVITE, Alice would know that she needs
> to attempt
> a connection to Bob, as a connection hadn't arrived when the
> Answer arrives
> via the proxy.  But here we have a connect succeed from Dan, then a short
> time later it terminates.  So how does Alice differentiate this
> forked-INVITE scenario from a failed media setup scenario?

Likewise, an attacker could actively prevent the media from being setup.

-d

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Subject: RE: [MMUSIC] A proposal for comedia-06 security section
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Well, this maps to my scenario 4 (and all of my scenarios result in DoS as 
the best case), I don't see how forked-INVITE fundamentally changes the 
equation...

At 01:26 PM 3/17/2003, Dan Wing wrote:
>...
> > If this was not a forked-INVITE, Alice would know that she needs
> > to attempt
> > a connection to Bob, as a connection hadn't arrived when the
> > Answer arrives
> > via the proxy.  But here we have a connect succeed from Dan, then a short
> > time later it terminates.  So how does Alice differentiate this
> > forked-INVITE scenario from a failed media setup scenario?
>
>Likewise, an attacker could actively prevent the media from being setup.
>
>-d

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After talking this over with some of the folks here, it's clear that 
perhaps that the three-way handshake is the Grand Simplifying 
Notion(tm).  SIP has this property, and to date it is SIP that is the only 
protocol causing grief with the forking-INVITE.

So, my assertion is that forking-INVITE ceases to be a problem if the 
condition described in section 4 of comedia is defined to be the ACK 
message from Alice rather than the 200 response from Bob.  Taking my 
earlier forking INVITE example, here's what happens:

Alice           Proxy       Bob            Dan

Offer   ------->
                  Offer ---->
                        |
                        +------------------>
                        <----Answer
        <---------Answer
                        <------------------Answer
                  Cancel------------------->

CONNECT ------------------->
ACK     ------------------->

By delaying the initiation of the connect-then-answer listed in section 
4(b/c) to the transmission of the ACK, we've allowed the fork to collapse 
back to a pair of endpoints before any connections are initiated.  So 
instead of dealing with the large permutations caused by the 1-to-n fan-out 
of Alice to possible destinations, we're back to just Alice and Bob (Bob 
being defined to always be the "winner" of the forked INVITE).

With this in mind, there are now five unique scenarios of Alice, Bob, and 
Carol:

1) FULL CONNECTIVITY (OR BOB CANNOT CONNECT TO ALICE):

Alice           Carol          Bob

Offer ------------------------>
       <-----------------------Answer
                CONNECT------->

CONNECT----------------------> ??
ACK   ----------------------->
       <--------CONNECT

=====================================

2) ALICE CANNOT CONNECT TO BOB:

Alice           Carol          Bob

Offer ------------------------>
       <-----------------------Answer
                CONNECT------->

CONNECT----X
ACK   ----------------------->
       <--------CONNECT
                Hijack!

=====================================

3) ALICE/CAROL CANNOT CONNECT TO BOB:

Alice           Carol          Bob

Offer ------------------------>
       <-----------------------Answer
                CONNECT--X

CONNECT----X
ACK   ----------------------->
       <--------CONNECT
??    <-----------------------CONNECT

=====================================

4) CAROL CANNOT CONNECT TO ALICE (OR CAROL/BOB CANNOT CONNECT TO ALICE):

Alice           Carol          Bob

Offer ------------------------>
       <-----------------------Answer
                CONNECT------->

CONNECT----------------------> ??
ACK   ----------------------->
             X--CONNECT

=====================================

5) CAROL CANNOT CONNECT TO BOB:

Alice           Carol          Bob

Offer ------------------------>
       <-----------------------Answer
                CONNECT--X

CONNECT---------------------->
ACK   ----------------------->
       <--------CONNECT

======================================

So there are two possible problems.  The first is #2, where the connection 
can be hijacked.  The second is #5, where the stakes are raised for Alice 
to choose correctly between the connection to Bob and the connection to 
Carol.

#2 has the same mitigation strategy as before---if Alice is on the public 
Internet and Bob is not, then Alice should use direction:passive.

#5 is trickier, any thoughts are welcomed.

Jonathan, you mentioned that there were some permutations of Offer/Answer 
that may upset this particular apple cart.  Now that this has been all laid 
out, what say you on this topic?

At 01:14 PM 3/17/2003, David Yon wrote:

>Temporarily flipping to a SIP-centric worldview, wouldn't one answer be to 
>map comedia section 4 (b/c) to the ACK message from Alice rather than the 
>OK reply from Bob?  This doesn't answer the question for two-way 
>handshakes, but it seems to for three-way.
>

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Hi,

Below is RTSP meeting notes from the sunday and monday. This is my 
interpretation of what was discussed. Please comment if something is 
missing or something is in error.

Best Regards

Magnus

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

Minutes of meetings for the two RTSP discussion meeting during IETF 56.

First meeting Sunday 21.00-23.15 16th April 2003

Participants

Sean Sheedy
Rob Lanphier
Tom Marshall
Andrea Basso
Anders Klements
Henning Schulzrinne
Colin Perkins
Joerg Ott
Thomas Zheng

Started with looking at the proposed presentation slides for the MMUSIC 
WG meeting presentation.

- Some short clarification on the usage of relative URL's for RTP-Info 
and having the Request URL as base for the this relative URL.

- A note that although we need support for Non-persistent connections a 
client is RECOMMENDED to keep the connection until it has finished all 
its sessions and signaling. The slides may not look like that they argue 
for persistent connection, but only as above. We need the 
non-persistentency to handle sessions in a reliable way.

Offer-Answer
------------

We discussed on how one could, and why one, would include a Offer Answer 
model into RTSP.

Use cases that where discussed were:

- Stream switching, one way of deciding which alternatives that can be 
used in switching would to use Offer-Answer to negotiate accepted 
alternatives.

- Capability exchange to determine if the a client can handle a stream 
and enable the server to tailor a presentation to fit the client.

A Important note in RTSP is that the capabilities are often restricted 
and can't be freely negotiated. For any on-demand multi-media 
presentation the capabilities are always dependent on the content 
creation step. Only what is available can be tailored. Content may also 
be restricted in what ways it can be adapted by the content creator. A 
possible way of having more latitude of capabilities are usage of 
transcoding functionality in servers.

It is also important to remember that RTSP uses session description in a 
uni-directional way which is different from normal SIP usage where the 
offer and answer is bi-directional.

Possible solutions.

A. Server if offerer and offers all that is available in regards to a 
stream. This could be done in DESCRIBE responses. The client is the 
Answerer and MAY then reduce the set of offered alternatives into an 
answer. A response could be included in the SETUP request. However this 
has some implications about binding answer to session and the SETUP 
requests. You can either send the answer in the first SETUP request and 
all future SETUP into that session uses the applied Answer. This has the 
implications that a aggregated session is much easier and does not 
require multiple sending of an Answer. Another alternative is to split 
an answer into answers for individual media streams.

The answer could also be done by a new method to establish State in the 
server that the SETUP can then bind to.

We will need a mechanism that could actually allow one to do multiple 
rounds of offer answer if the answer is not acceptable. However in most 
cases it seems possible that offer-answer should resolve in a single 
exchange.

B. Another scenario is that the client is the offerer. It sends the 
server its capabilities. Then the server would send an answer containing 
a matched presentation for the requested resource. A method would be 
DESCRIBE or similar. Then one must simply bind the offer answer 
agreement to a session. This is similar to what 3GPP PSS has available 
when one uses UAProf.

C. A half way case. A client retrieves a session description. The client 
creates an offer on what it can accept of this. The server would then 
answer with if this is suitable.


Considerations for selecting a possible solution is:

- What type of states must reside on the which sides?
- The amount of bandwidth that will be used by the offers and answers.


Resolution:
It seems interested and has some good use cases. However no one 
committed to create some ground breaking work in this area in form of 
use case and requirements draft.


RTSP IANA registration requirements
-----------------------------------

We discussed this a long time but could actually not come to any general 
conclusions.

- The was proposal that we could have permanent registrations that must 
be stable specifications. While proprietary registrations could be bound 
in time and if not renewed they would be removed from registry. However 
the problems are: The reclaiming mechanism. And what happens when a 
registered value are in wide use but the owner are no longer active in 
the are.

- It might be hard to swear to long from what they have defined in SIP & 
HTTP without having a good use case.

- The registration rules can be really customized and allows us to write 
rules for what a registration requires.

- Remember the x- problem that it might be better to allow for good 
names from the beginning. Vendor tree of registration would basically 
have the x- problem.

- Most of the registries are in fact unlimited. All having free width 
text fields has no limit. 

- Feature tags seems one that will be used in proprietary ways. However 
the question is if there is need for interoperability and collision 
avoidance that the registry supplies. Some discussion around having 
First come first served and a one paragraph description of the feature 
tag as minimum rules.

- Most agree that IETF standard tracks operation is suitable for 
Methods. Some voiced that restricting it to much would result in 
divergence. However this strictness may result in that all functionality 
is moved to SET_PARAMETER and headers.

- For headers there where discussion on splitting this into, a Vendor 
tree with tags that allow vendor registration, and a IETF tree which 
requires IETF standards track. However there was also other 
considerations that Public specification might be enough. Or even a 
lower level as there exist proprietary headers.

- The Transport header repositories for Protocol/profile/lower-transport 
to had as an triplet in a common registry. Public specification was 
agreed on. This can be defined by other standards body to allow them to 
adapt RTSP for there transport techniques.


Second meeting Monday 13.00-15.00 17 April 2003

Second meeting participants:
Magnus Westerlund
Rob Lanphier
Tom Marshall
Phillipe Gentric
Sean Sheedy
Anders Klemets
Thomas Zheng
Andrea Basso

* Stream-switching discussion

   We started by discussing around stream switching. Comments raised was:
   - We need to find good terminology. Stream switching are not really 
what we are talking about.
   - The goal is to have switching between considerable different streams.
   - We should consider the extreme cases to handle all normal cases.
   - An offer answer model could be used to select what alternative that 
are desired.
   - We have problem with caching proxies when we performs switches 
between different alternatives.  There was some discussion on different 
possibilities to mark packets on what alternative they belong to.
   -
  
   RESOLUTION: Phillipe Gentric writes a requirement draft with use cases.
  
* Ways to allow usage of Destination parameter
  The goal would be to find ways that give good enough security that a 
server can be allowed to send media traffic to a media receiver.
  - The easiest way of ensuring that a receiver accepts the media 
traffic is to ask it. Therefore the server could send a message to ask 
the receiver if it accepts to receive the traffic. The receiver 
responds. When the server receives a response it knows if it is allowed 
or not.
  - We could use the fallback rule that all client requesting media to 
the same address as there RTSP message connection comes from does not 
need to use this mechanism but MAY be used. => Gives backwards 
compatibility and makes it possible to deploy.
  - A server should or shall not send media until media acceptance 
challenge is responded.
  - A short term solution would be to require RTCP messages from other 
end-point to be able to send media.
  - RTCP RR are rather simple to spoof as the start RTP sequence number 
is known.
  - However it would still lower the force multiplier a RTSP server 
would give for performing a DOS attack. What I mean with force 
multiplier is the amount of work a attacker needs to do get a certain 
amount of affect at the target.
  - To get a guaranteed secure mechanism we would need to have way of 
source authenticating the media receiver target.
  - How do we handle multicast receiver addresses?
  - Should ask security expert for suggestions when we have described 
the problem well.
  - Write a problem description draft.
  - We should also look at having a general mechanism as this seem to be 
a problem all media transport can end up in. This is probably something 
that SIP has been ignoring and has not considered.
 
RESOLUTION: Phillipe Gentric will write a basic problem description draft.

* RTSP & NAT problems

Some minor discussion. Important comments:

TURN will not work unless we allow different IP address as the TCP RTSP 
will be directly from client while all media destinations will be the 
TURN servers IP.

The STUN vs. Symmetric RTP discussion.

- The port usage of symmetric RTP is a reality problem for large 
servers. May need to consider a solution that allows a trade off between 
port usage and security.

- Even if we haven't seen any actual case of multiple IP address NATs, 
it seems probable that wireless operators will deploy large scale NATs 
that will be simple and will result in one client streams end up on 
different IP addresses.

- We should include NAT implementor recommendations/requirements telling 
that a NAT should for any client only give a single IP on the public 
address.

- We discussed the need for ALG implementations recommendations. 
Especially the need for avoiding messing up the other methods. The 
recommendation would that any request containing a destination address 
shall not be touched by the ALG. This also effects the client behavior.

- Firewalls where discussed. It seems that there exist three basic types.
  1. A simple home NAT & FW box that simple opens ports and maps them to 
the machine that punched open the whole in the wall. This makes this 
equivalent with a Cone NAT.
  2. A UNIX/Linux software FW that actually are more restricted and 
looks at the source IP and decide if the private side has pounced a hole 
towards this address or not. Makes the FW equivalent to a symmetric NAT.
  3. Corporate FW. This ones is really paranoid and will block all 
unknown and not allowed traffic. Often make statefull inspection. To get 
RTSP and media transport on UDP through this types requires a policy 
allowing RTSP and requires ALG functionality to allow for UDP media traffic.
 
* Allowing for multiple SSRC in RTSP.

- The problem is that any media stream having multiple SSRCs can't be 
initially synchronized with RTP-Info. Fortunately RTCP SR provides a 
fall back for this. RTSP has SSRC issues in two places: 1. Transport 
header, 2. RTP-Info where RTP sequence number and RTP timestamp are 
related to a SSRC but not explicit present.
- The transport header may not need that the server informs the client 
on what SSRC it will use.
- The right thing would to redefine the RTP-Info header to allow for 
multiple SSRCs and their sequence number and timestamp.
- Will not be backwards compatible. This has implications on how it can 
be handled. Either simply redefine in the updated and forcing one to 
keep track of updated or not implementations. The other would be to 
ignore this problem.
- As fallback for old implementations is to use RTCP for this 
synchronization.

* Removal of Timed request

- Consensus to disallow if for now. The syntax may be left in for 
backwards completeness.

* How to distinguish the updated specification.

- Due to multiple SSRC discussion we revisited mechanisms to distinguish 
a updated server. One possible solution would to upgrade the version to 
1.1. We should consider this more in the future when we comes closer to 
a finished specification.
- Should look into more detail how this should work and if we adapt HTTP 
model?


* Warning header

- Could be good to provide more information about what happens in 
certain cases and for debugging.
- Needs to look into HTTP and SIP for more consideration how to use it.
- Any addition should be considered if this will delay the new updated 
draft.
- Needs more consideration.


-- 

Magnus Westerlund 

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


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From: Tom Marshall <tmarshall@real.com>
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Subject: [MMUSIC] What is symmetric RTP?
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I have been doing a bit of digging, trying to find an exact definition of
"Symmetric RTP" -- that is, what exactly is it, and how it should be used.=
=20
Ironically, the biggest reference to Symmetric RTP (according to google)
seems to be the recent discussions on this mailing list.

The draft draft-ietf-mmusic-sdp-comedia-04.txt was mentioned at one point.=
=20
It has now expired.  I found a cached copy and read it.  It does seem to
provide some brief explanation of how to do Symmetric RTP, but it does not
look detailed enough.  Section 8.5 is titled "Example: "Bidirectional" RTP
and RTCP" and it says only "... immediately sends RTP from ..." and "...
immediately sends RTCP from ..." with no explanation of the expected packet
contents.

I would like to see Symmetric RTP defined more precisely, both for our own
use and for comedia.  What do others think?  Would someone like to clarify
usage, provide examples, start a draft, etc?

--=20
Harrison's Postulate:
        For every action, there is an equal and opposite criticism.

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From mailnull@www1.ietf.org  Wed Mar 19 23:47:37 2003
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Hi Tom,

I agree that the term symmetric RTP is really not defined in any 
document. I have one view of this that is a result of following 
discussions around NATs in AVT, SIPPING, and MMUSIC. In my view people 
often mean two things when they are talking about symmetric RTP.

1. The sending and receiving of RTP from the same port. This was the 
case in the ancient days, :-),  of multicast usage and RAT . However 
then the SIP and some other people come around and distorted this view 
with not using the same port for receiving and sending.
2. The usage of a received message to determine the destination address. 
This is I think the main idea when performing NAT traversal. Of course 1 
must be used to make it work, however this is simple. It is the second 
usage that has some problems.

One possible place to better define symmetric RTP could be the RTSP NAT 
draft.

Best Regards

Magnus 

Tom Marshall wrote:

>I have been doing a bit of digging, trying to find an exact definition
>of
>"Symmetric RTP" -- that is, what exactly is it, and how it should be
>used. 
>Ironically, the biggest reference to Symmetric RTP (according to google)
>seems to be the recent discussions on this mailing list.
>
>The draft draft-ietf-mmusic-sdp-comedia-04.txt was mentioned at one
>point. 
>It has now expired.  I found a cached copy and read it.  It does seem to
>provide some brief explanation of how to do Symmetric RTP, but it does
>not
>look detailed enough.  Section 8.5 is titled "Example: "Bidirectional"
>RTP
>and RTCP" and it says only "... immediately sends RTP from ..." and "...
>immediately sends RTCP from ..." with no explanation of the expected
>packet
>contents.
>
>I would like to see Symmetric RTP defined more precisely, both for our
>own
>use and for comedia.  What do others think?  Would someone like to
>clarify
>usage, provide examples, start a draft, etc?
>

-- 

Magnus Westerlund 

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



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Hi,

I have a question about the legality/normality of the following use case:

A user agent sends a DESCRIBE for the following resource
rtsp://192.168.200.12/resource.mp4:
DESCRIBE rtsp://192.168.200.12/resource.mp4
CSeq: 1
User-agent: MyUa
Accept: application/sdp

The server replies:
RTSP/1.0 200 OK
CSeq: 1
Server: MyServer
Content-Location: rtsp://192.168.202.12/go_there_please.mp4
Content-Type: application/sdp
Content-Length: XXXX

v=0...
a=control:*
m=audio...
a=control:stuff/lady_in_red
m=video...
a=control:stuff/shae_marks

In this example, the server replies successfully but "redirects" the user
agent to another host for the subsequent requests. The RTSP spec (RFC 2326)
is very broad concerning Content-Location. However, I've ran into UAs that
seem to  expect Content-Location to mandatorily point to the SAME host
address (regardless of the URI path that can vary).
Hence my question: as far as the network authority is concerned, can
Content-Location point to a different host than the replying host??

Thanks,

Olivier



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From: "Vishal Verma" <vishal@sonim-india.com>
To: <mmusic@ietf.org>
Date: Fri, 21 Mar 2003 18:49:04 +0530
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Hello,
  I am looking into SDP protocol presently and I have some 
questions regarding SDP. My questions are :

1. SDP is used by SIP to inform about endpoint media capability 
in INVITE message .My question is ...if endpoint A initiated  
call to endpoint B and on receiving A's SDP ,B finds out A's SDP does 
not matches with A so B can not go ahead with call .In this 
situation what is the response code to indicate this failure ?
   If there is no response code for this so what exactly is 
method to communicate error to A.

2. Why SDP does not madates use of certain codecs(G.711) as done 
by ITU-T protocols(H.323)(I mean to ask what is the idea behind 
not suggesting mandatory codecs) .There will always be high 
probability of codec mismatch

regards, vishal

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inline.

Vishal Verma wrote:
> Hello,
>   I am looking into SDP protocol presently and I have some 
> questions regarding SDP. My questions are :
> 
> 1. SDP is used by SIP to inform about endpoint media capability 
> in INVITE message .My question is ...if endpoint A initiated  
> call to endpoint B and on receiving A's SDP ,B finds out A's SDP does 
> not matches with A so B can not go ahead with call .In this 
> situation what is the response code to indicate this failure ?

488

>    If there is no response code for this so what exactly is 
> method to communicate error to A.
> 
> 2. Why SDP does not madates use of certain codecs(G.711) as done 
> by ITU-T protocols(H.323)(I mean to ask what is the idea behind 
> not suggesting mandatory codecs) .There will always be high 
> probability of codec mismatch

Selecting a baseline codec is an IPR nightmare.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                72 Eagle Rock Ave.
Chief Scientist                             First Floor
dynamicsoft                                 East Hanover, NJ 07936
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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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:

http://sf.net/projects/rtspspec

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



Today's Topics:

   1. [ rtspspec-Bugs-707252 ] Update the RTSP URI schemes (SourceForge.net)

--__--__--

Message: 1
To: noreply@sourceforge.net
From: "SourceForge.net" <noreply@sourceforge.net>
Date: Thu, 20 Mar 2003 16:45:13 -0800
Subject: [rtspspec-bugs] [ rtspspec-Bugs-707252 ] Update the RTSP URI schemes

Bugs item #707252, was opened at 2003-03-21 01:45
You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=707252&group_id=23194

Category: General
Group: None
Status: Open
Resolution: None
Priority: 5
Submitted By: Magnus Westerlund (magwes)
Assigned to: Magnus Westerlund (magwes)
Summary: Update the RTSP URI schemes

Initial Comment:
The current RTSP schemes has been updated to allow 
for literall IPv6. However the URI scheme should also be 
updated to cover IRI (Internationalized Resource 
Identifiers). I (Magnus W) will try to get more information 
on this from the people having created the IRI. Hopefully 
this is little or no work.

For people to read up on URI scheme considerations 
and other things should look at:
RFC 2717 and 2718. I hade so far not had time to dig up 
any references to the IRI work. 


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

You can respond by visiting: 
https://sourceforge.net/tracker/?func=detail&atid=377744&aid=707252&group_id=23194




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Subject: [MMUSIC] question on draft-ietf-mmusic-sdp-new-12 ...
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Hi everyone,

        Quick 'wording' question on draft-ietf-mmusic-sdp-new-12,
now that maxptime exists as an SDP attribute, is there an
implied "typical" in the ptime attribute definition, i.e.

  This gives the typical length in milliseconds ...

        If maxptime and ptime are intended to be used together
in a session description, this "typical" would seem to be 
implied. However, if maxptime and ptime are intended to be
mutually exclusive (one or the other but not both are permitted
in a session description), the "typical" is probably not 
implied.   

        Thanks in advance ...

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
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From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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To: olivier.randriamanana.extern46@philips.com
CC: mmusic@ietf.org
Subject: Re: [MMUSIC] Content-Location in a 200 resp. leading to other server
References: <OFFAC6720C.6BD2E3A5-ONC1256CF0.00481323@diamond.philips.com>
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Hi Olivier,

To me this example seems legal. In fact from the language of RFC 2326 
the intention is to support use cases when even the different media 
streams are located on different servers. Also looking at the definition 
of Content Location I don't see anything that indicates the restriction 
you describe. So in my opinion a UA should definitely handle the case 
with different host address in Content-Location.

Best Regards

Magnus 

olivier.randriamanana.extern46@philips.com wrote:

>Hi,
>
>I have a question about the legality/normality of the following use case:
>
>A user agent sends a DESCRIBE for the following resource
>rtsp://192.168.200.12/resource.mp4:
>DESCRIBE rtsp://192.168.200.12/resource.mp4
>CSeq: 1
>User-agent: MyUa
>Accept: application/sdp
>
>The server replies:
>RTSP/1.0 200 OK
>CSeq: 1
>Server: MyServer
>Content-Location: rtsp://192.168.202.12/go_there_please.mp4
>Content-Type: application/sdp
>Content-Length: XXXX
>
>v=0...
>a=control:*
>m=audio...
>a=control:stuff/lady_in_red
>m=video...
>a=control:stuff/shae_marks
>
>In this example, the server replies successfully but "redirects" the user
>agent to another host for the subsequent requests. The RTSP spec (RFC 2326)
>is very broad concerning Content-Location. However, I've ran into UAs that
>seem to  expect Content-Location to mandatorily point to the SAME host
>address (regardless of the URI path that can vary).
>Hence my question: as far as the network authority is concerned, can
>Content-Location point to a different host than the replying host??
>
>Thanks,
>
>Olivier
>
>
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
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>
>  
>

-- 

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
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Subject: Re: [MMUSIC] question on draft-ietf-mmusic-sdp-new-12 ...
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Hi John,

The usage of ptime and maxptime is not mutually exclusive neither are 
they required or even recommended to be used together. The usage is 
dependent on application and can be seen as two attributes for 
controlling how the packetization/encoding work. For ptime the "typical" 
you imply does exist in my opinion. It seems reinforced by the following 
sentence from the ptime definition: "It should not be necessary to know 
ptime to decode RTP or vat audio, and it is intended as a recommendation 
for the encoding/packetisation of audio."

when looking at the version 12 draft I think the definition of maxptime 
needs to be addressed. The following sentence:
"The time SHOULD be a multiple of the frame size." This text comes 
directly from AMR definition that has a frame structure but in the SDP 
specification this should be reformulated to cover the more general and 
take sample based codecs into consideration. My proposal for text is:
"The time SHOULD be a multiple of any frame size."

Best Regards

Magnus

John Lazzaro wrote:

>Hi everyone,
>
>        Quick 'wording' question on draft-ietf-mmusic-sdp-new-12,
>now that maxptime exists as an SDP attribute, is there an
>implied "typical" in the ptime attribute definition, i.e.
>
>  This gives the typical length in milliseconds ...
>
>        If maxptime and ptime are intended to be used together
>in a session description, this "typical" would seem to be 
>implied. However, if maxptime and ptime are intended to be
>mutually exclusive (one or the other but not both are permitted
>in a session description), the "typical" is probably not 
>implied.   
>
>        Thanks in advance ...
>
>-------------------------------------------------------------------------
>John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
>lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
>-------------------------------------------------------------------------
>_______________________________________________
>mmusic mailing list
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>  
>

-- 

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
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Subject: Re: [MMUSIC] What is symmetric RTP?
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Hi,

I would like to clarify some things in my previous mail.

Magnus Westerlund wrote:

> Hi Tom,
>
> I agree that the term symmetric RTP is really not defined in any 
> document. I have one view of this that is a result of following 
> discussions around NATs in AVT, SIPPING, and MMUSIC. In my view people 
> often mean two things when they are talking about symmetric RTP.
>
> 1. The sending and receiving of RTP from the same port. This was the 
> case in the ancient days, :-),  of multicast usage and RAT . However 
> then the SIP and some other people come around and distorted this view 
> with not using the same port for receiving and sending. 

Bad choice of language here. I don't want to blame the SIP people for 
the change. RTP has the flexibility to be used with different ports at 
different end points and sending from another port then receiving. It i 
s after all a matter of being able to signal.

>
> 2. The usage of a received message to determine the destination 
> address. This is I think the main idea when performing NAT traversal. 
> Of course 1 must be used to make it work, however this is simple. It 
> is the second usage that has some problems.

>
> One possible place to better define symmetric RTP could be the RTSP 
> NAT draft.


After some thought and discussion, I don't think the RTSP NAT draft is 
the right place anymore. If we would do a specification of what 
symmetric RTP is, it should be a stand alone document that can be 
referenced both by SIP and RTSP and any other signaling that has use of 
the functionality.

Best Regards

Magnus Westerlund 

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



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Magnus Westerlund wrote:
> Hi,
> 
> I would like to clarify some things in my previous mail.
> 
> Magnus Westerlund wrote:
> 
>> Hi Tom,
>>
>> I agree that the term symmetric RTP is really not defined in any 
>> document. I have one view of this that is a result of following 
>> discussions around NATs in AVT, SIPPING, and MMUSIC. In my view people 
>> often mean two things when they are talking about symmetric RTP.
>>
>> 1. The sending and receiving of RTP from the same port. This was the 
>> case in the ancient days, :-),  of multicast usage and RAT . However 
>> then the SIP and some other people come around and distorted this view 
>> with not using the same port for receiving and sending. 
> 
> 
> Bad choice of language here. I don't want to blame the SIP people for 
> the change. RTP has the flexibility to be used with different ports at 
> different end points and sending from another port then receiving. It i 
> s after all a matter of being able to signal.

Well, really the issue is that usage of RTP with unicast has been a 
source of constant confusion even to this day.

> 
>>
>> 2. The usage of a received message to determine the destination 
>> address. This is I think the main idea when performing NAT traversal. 
>> Of course 1 must be used to make it work, however this is simple. It 
>> is the second usage that has some problems.
> 
> 
>>
>> One possible place to better define symmetric RTP could be the RTSP 
>> NAT draft.
> 
> 
> 
> After some thought and discussion, I don't think the RTSP NAT draft is 
> the right place anymore. If we would do a specification of what 
> symmetric RTP is, it should be a stand alone document that can be 
> referenced both by SIP and RTSP and any other signaling that has use of 
> the functionality.

THe reason that this went into comedia is that, once you do this, UDP 
effectively looks "connection oriented". THe same issues exist as they 
do for TCP - who is active, who is passive? How is it negotiated? Thats 
what comedia does.

That said, I still believe that it is desirable to align RTSP NAT 
solutions with SIP. Symmetric RTP is nice, but it is not as general as 
the ICE-based approach I have proposed. There are many topologies where 
symmetric RTP will not work. Any topology where the server itself is 
behind a NAT. A good example is two enterprises. Lets say there is a 
manufacturer that makes widgets. They have customers in lots of 
enterprises. They want to run an RTSP server inside their enterprise, so 
users can connect and see videos providing customer support. If the 
customer enterprises are natted, and so is the widget manufacturer, 
symmetric RTP will not help. ICE would figure out the best approach. The 
only case in which ICE requires more than one exchange to work in this 
case is when both NATs are full-cone. In such a case, you can eliminate 
the second ICE exchange if you use a TURN relay.

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
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> Magnus Westerlund <magnus.westerlund@era.ericsson.se> writes
>
> The usage is 
> dependent on application and can be seen as two attributes for 
> controlling how the packetization/encoding work. For ptime the "typical" 
> you imply does exist in my opinion. 

Thanks -- I'm rewriting the media timing section of the MWPP
normative I-D, and wanted to make sure I was interpeting the
semantics correctly.

> when looking at the version 12 draft [...]

Yes, a tangentially related issue is that the value of ptime
and maxptime is neither constrained to be an integral value of
milliseconds, nor expressly permitted to have a fractional 
component. I would normally take that to mean that a fractional
part is permitted (" ... be liberal in what you accept"), except
that the definition of the framerate attribute explicitly allows
<integer>.<fractional> in its text, as if it was an exceptional
condition to an unstated norm ... 

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
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Subject: Re: [MMUSIC] What is symmetric RTP?
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Hi Jonathan,

See below:

Jonathan Rosenberg wrote:

>> After some thought and discussion, I don't think the RTSP NAT draft 
>> is the right place anymore. If we would do a specification of what 
>> symmetric RTP is, it should be a stand alone document that can be 
>> referenced both by SIP and RTSP and any other signaling that has use 
>> of the functionality.
>
>
> THe reason that this went into comedia is that, once you do this, UDP 
> effectively looks "connection oriented". THe same issues exist as they 
> do for TCP - who is active, who is passive? How is it negotiated? 
> Thats what comedia does.
>
> That said, I still believe that it is desirable to align RTSP NAT 
> solutions with SIP. Symmetric RTP is nice, but it is not as general as 
> the ICE-based approach I have proposed. There are many topologies 
> where symmetric RTP will not work. Any topology where the server 
> itself is behind a NAT. A good example is two enterprises. Lets say 
> there is a manufacturer that makes widgets. They have customers in 
> lots of enterprises. They want to run an RTSP server inside their 
> enterprise, so users can connect and see videos providing customer 
> support. If the customer enterprises are natted, and so is the widget 
> manufacturer, symmetric RTP will not help. ICE would figure out the 
> best approach. The only case in which ICE requires more than one 
> exchange to work in this case is when both NATs are full-cone. In such 
> a case, you can eliminate the second ICE exchange if you use a TURN 
> relay.
>
> -Jonathan R.
>
I can understand your desire to align RTSP and SIP. However when it 
comes to usage of ICE it definitely gets problematic. First RTSP does 
not have the SETUP exchange in place to accommodate something like ICE 
need for one or more exchange with parameters both for the STUN server 
and the given addresses. I don't say it is impossible however more work 
will be needed to look at how it could be included. Then one will see 
how many modifications it would requires. In the end it might be simpler 
to take on the SIP session initiation and only keep RTSP's media 
control.  However this is a radical proposal that would significantly 
change the protocol.

One question I would like to know is how big the envisioned problem will 
be with servers that reside in NATed networks. I can understand that SIP 
must consider all cases as it set up communication between users which 
are more likely to have NATed addresses. However looking at RTSP which 
has a clear client to server structure, much like HTTP. If you move a 
server to a NATed address you will also have problem setting up the TCP 
connection. You simple will require a proxy between the address domains 
to be able to communicate. Are there many that deploy HTTP servers in 
private address spaces and actually intend them to be used by other 
address domains? So far my view has been that as this is server, the 
server owner can ensure that the server has an IP address that makes it 
reachable by the intended clients. So in your widget manufacturer 
example above, why does not he deploy the server in the public IP space. 
I would assume that he has HTTP server that are also reachable. If you 
don't share my opinion on this please speak up and give some motivation 
why this is a useful example. 

Regards

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
Ericsson AB                | Phone +46 8 4048287
Torshamsgatan 23           | Fax   +46 8 7575550
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Subject: Re: [MMUSIC] Usage of maxptime and ptime
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Hi John,

John Lazzaro wrote:

>>when looking at the version 12 draft [...]
>>    
>>
>
>Yes, a tangentially related issue is that the value of ptime
>and maxptime is neither constrained to be an integral value of
>milliseconds, nor expressly permitted to have a fractional 
>component. I would normally take that to mean that a fractional
>part is permitted (" ... be liberal in what you accept"), except
>that the definition of the framerate attribute explicitly allows
><integer>.<fractional> in its text, as if it was an exceptional
>condition to an unstated norm ... 
>  
>

No, I have never seen ptime or maxptime being used with a fractional 
part. I would strongly discourage the use with fractional part as it 
will probably not interoperate. However the attributes seem to be 
lacking any formal notation or language to clarify that it is only 
integer. My intention for maxptime was never to have it fractional as I 
saw no need for AMR. I know that there exist some codecs that could 
result in fractional milli second values. Also I don't see a big 
problem. As maxptime provides a upper boundary when having a codec with 
frame length of 7.7 milli seconds it seems obvious that if you want to 
limit this to three frames one simple uses 24 ms in maxptime. If one at 
the same time want to recommend the usage of one frames ptime is given 
the value of 8.

Colin: why does not the attributes defined in the SDP new draft have 
ABNF definitions? Had ptime and maxptime had it the question would been 
clearly solved.

Best Regards

Magnus Westerlund 

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



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cc: John Lazzaro <lazzaro@CS.Berkeley.EDU>, mmusic@ietf.org
Subject: Re: [MMUSIC] Usage of maxptime and ptime 
In-Reply-To: Your message of "Thu, 27 Mar 2003 08:52:56 +0100."
             <3E82ADD8.8010005@era.ericsson.se> 
Date: Thu, 27 Mar 2003 11:28:31 -0500
From: Colin Perkins <csp@isi.edu>
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--> Magnus Westerlund writes:
>Hi John,
>
>John Lazzaro wrote:
>
>>>when looking at the version 12 draft [...]
>>>    
>>>
>>
>>Yes, a tangentially related issue is that the value of ptime
>>and maxptime is neither constrained to be an integral value of
>>milliseconds, nor expressly permitted to have a fractional 
>>component. I would normally take that to mean that a fractional
>>part is permitted (" ... be liberal in what you accept"), except
>>that the definition of the framerate attribute explicitly allows
>><integer>.<fractional> in its text, as if it was an exceptional
>>condition to an unstated norm ... 
>>  
>>
>
>No, I have never seen ptime or maxptime being used with a fractional 
>part. I would strongly discourage the use with fractional part as it 
>will probably not interoperate. However the attributes seem to be 
>lacking any formal notation or language to clarify that it is only 
>integer. My intention for maxptime was never to have it fractional as I 
>saw no need for AMR. I know that there exist some codecs that could 
>result in fractional milli second values. Also I don't see a big 
>problem. As maxptime provides a upper boundary when having a codec with 
>frame length of 7.7 milli seconds it seems obvious that if you want to 
>limit this to three frames one simple uses 24 ms in maxptime. If one at 
>the same time want to recommend the usage of one frames ptime is given 
>the value of 8.
>
>Colin: why does not the attributes defined in the SDP new draft have 
>ABNF definitions? Had ptime and maxptime had it the question would been 
>clearly solved.

History. I have no objection to including ABNF, if you want to suggest the
appropriate form.

Colin
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> Colin Perkins <csp@isi.edu> writes:
>
> History. I have no objection to including ABNF, if you want
> to suggest the appropriate form.

Here's one alternative, adapted from the MWPP ABNF ... 

                  ;
                  ; sub-rules of 'a='
                  ;

attribute-fields  =  *("a=" attribute CRLF)

                  ; ABNF for attributes defined in this memo

attribute         = "cat"      ":"  cat-fields

attribute        /= "keywds"   ":"  keyword-fields

attribute        /= "tool"     ":"  tool-fields

attribute        /= "ptime"    ":" four-octet-int

attribute        /= "maxptime" ":" four-octet-int

                  ;
                  ; todo: complete list of defined attributes 
                  ;
                  ; last entry (below) guides authors of new attributes

attribute        /= (att-field ":" att-value) / att-field

                 
                  ;
                  ; ABNF for specific attribute fields
                  ;

four-octet-int    = %d0-429496729
                  ; unsigned 32-bits integer

                  ; add cat-fields, keyword-fields ... below

att-field         =  token

att-value         =  byte-string

-------------------------------------------------------------------------
John Lazzaro -- Research Specialist -- CS Division -- EECS -- UC Berkeley
lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
-------------------------------------------------------------------------
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> Magnus Westerlund <magnus.westerlund@era.ericsson.se> writes:
>
>  Also I don't see a big 
> problem. As maxptime provides a upper boundary when having a codec with 
> frame length of 7.7 milli seconds it seems obvious that if you want to
> limit this to three frames one simple uses 24 ms in maxptime.

MWPP is an audio payload that is neither frame nor sample based;
the interactive norm may be ptime:0 -- i.e. all commands
in a payload execute at the same moment in time, and thus the 
interval of time coded in a packet is 0 ms.

But you can also use MWPP to do content-streaming type work, in
which case you'd probably specify ptime values in keeping with   
what MPEG audio codecs use ...

So, ptime and maxptime are the hints that will be used for 
senders and receivers to know which broad application class
the session falls into ... as well as to provide hints for
sender timing accuracy and receiver buffer sizing. This will
all be informative text tucked into an Appendix in the MWPP
normative memo ... 

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

>>Colin Perkins <csp@isi.edu> writes:
>>
>>History. I have no objection to including ABNF, if you want
>>to suggest the appropriate form.
>>    
>>
>
>Here's one alternative, adapted from the MWPP ABNF ... 
>
>
>attribute        /= "ptime"    ":" 1*DIGIT
>
>attribute        /= "maxptime" ":" 1*DIGIT
>
>  
>
I would change the value from a unsigned integer to the much more common 
DIGIT definition.

Best Regards


Magnus Westerlund 

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



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Magnus Westerlund wrote:

>> That said, I still believe that it is desirable to align RTSP NAT 
>> solutions with SIP. Symmetric RTP is nice, but it is not as general as 
>> the ICE-based approach I have proposed. There are many topologies 
>> where symmetric RTP will not work. Any topology where the server 
>> itself is behind a NAT. A good example is two enterprises. Lets say 
>> there is a manufacturer that makes widgets. They have customers in 
>> lots of enterprises. They want to run an RTSP server inside their 
>> enterprise, so users can connect and see videos providing customer 
>> support. If the customer enterprises are natted, and so is the widget 
>> manufacturer, symmetric RTP will not help. ICE would figure out the 
>> best approach. The only case in which ICE requires more than one 
>> exchange to work in this case is when both NATs are full-cone. In such 
>> a case, you can eliminate the second ICE exchange if you use a TURN 
>> relay.

> I can understand your desire to align RTSP and SIP. However when it 
> comes to usage of ICE it definitely gets problematic. First RTSP does 
> not have the SETUP exchange in place to accommodate something like ICE 
> need for one or more exchange with parameters both for the STUN server 
> and the given addresses. I don't say it is impossible however more work 
> will be needed to look at how it could be included. Then one will see 
> how many modifications it would requires. In the end it might be simpler 
> to take on the SIP session initiation and only keep RTSP's media 
> control.  However this is a radical proposal that would significantly 
> change the protocol.

Well, I suspect it would not change the protocol so much as only use a 
subset - the parts for control.

Regarding having RTSP itself support an offer/answer exchange with 
SETUP, generally only one exchange would be needed. So, its just an 
issue of putting more than one address in the SETUP.

> 
> One question I would like to know is how big the envisioned problem will 
> be with servers that reside in NATed networks. I can understand that SIP 
> must consider all cases as it set up communication between users which 
> are more likely to have NATed addresses. However looking at RTSP which 
> has a clear client to server structure, much like HTTP. If you move a 
> server to a NATed address you will also have problem setting up the TCP 
> connection. You simple will require a proxy between the address domains 
> to be able to communicate. Are there many that deploy HTTP servers in 
> private address spaces and actually intend them to be used by other 
> address domains? So far my view has been that as this is server, the 
> server owner can ensure that the server has an IP address that makes it 
> reachable by the intended clients. So in your widget manufacturer 
> example above, why does not he deploy the server in the public IP space. 
> I would assume that he has HTTP server that are also reachable. If you 
> don't share my opinion on this please speak up and give some motivation 
> why this is a useful example.

I think that you are making a fair point; if the widget manufacturer 
meant for this server to be reachable, they need to give it a public 
address.

I think the cases where this might not be true is where "server" becomes 
more loosely defined. You have in mind a big server meant for consumer 
streaming media applications. However, I can imagine using RTSP to do 
things like retrieve voicemails, and even control streams shared 
directly between two users (for example, one user calling another with 
sip and letting the called party hear an mp3 that the caller has on 
their PC). All of these are, IMHO, good uses for RTSP, and none of them 
will work with the symmetric-RTP only approach you are advocating.

So - are these places you want RTSP to work?

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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> Magnus Westerlund <magnus.westerlund@era.ericsson.se> writes
>
> I would change the value from a unsigned integer to the much more
> common DIGIT definition.

Yes, that definition was stolen from MWPP parameters that needed
32-bit explicit constraints ... not needed here though.

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lazzaro [at] cs [dot] berkeley [dot] edu     www.cs.berkeley.edu/~lazzaro
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To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: mmusic@ietf.org
Subject: A reason to put RTSP server behind NAT (was RE: [MMUSIC] What is 
	symmetric RTP?)
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Magnus:

I would like to chime in to add an use case for putting RTSP server behind a
firewall.

Imagine a startup company wanting to install a live camera in a daycare to
offer "nannycam" services. The live camer feeds audio and video to a RTSP
server (running on the same CPU) for paid clients to view  their kids  by
connectting a RTSP video player. It's well-known that some big companies
offer
streaming video in their company daycares to their employees. Now someone
just wants the same service in other kids' daycare (so as to bridge the 
digital gap, among other things :-)

The daycare has a combo cable modem / NAT box to run a private network of a
few computers, one of which hosts the live camer as well as the RTSP server.


This is a case where RTSP server is behind a server NAT, while clients may
or may not be behind another NAT. Clients out side of the server NAT can
connect because the NAT box can forward RTSP port to the RTSP server
computer (via, say, Linux ipchains commands).

Of course one can ask the daycare to put the RTSP server box in the open,
but that means additional cost (it costs $5 dollars/month to get even a
dynamic IP address from Time-Warner cable company here in San Diego), some
one time fee to change network topology -- not to mention security
degradation.

Do you think this application is a legitimate use of RTSP protocol?

- thomas

-----Original Message-----
From: Jonathan Rosenberg [mailto:jdrosen@dynamicsoft.com]
Sent: Friday, March 28, 2003 6:47 AM
To: Magnus Westerlund
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] What is symmetric RTP?




Magnus Westerlund wrote:

>> That said, I still believe that it is desirable to align RTSP NAT 
>> solutions with SIP. Symmetric RTP is nice, but it is not as general as 
>> the ICE-based approach I have proposed. There are many topologies 
>> where symmetric RTP will not work. Any topology where the server 
>> itself is behind a NAT. A good example is two enterprises. Lets say 
>> there is a manufacturer that makes widgets. They have customers in 
>> lots of enterprises. They want to run an RTSP server inside their 
>> enterprise, so users can connect and see videos providing customer 
>> support. If the customer enterprises are natted, and so is the widget 
>> manufacturer, symmetric RTP will not help. ICE would figure out the 
>> best approach. The only case in which ICE requires more than one 
>> exchange to work in this case is when both NATs are full-cone. In such 
>> a case, you can eliminate the second ICE exchange if you use a TURN 
>> relay.

> I can understand your desire to align RTSP and SIP. However when it 
> comes to usage of ICE it definitely gets problematic. First RTSP does 
> not have the SETUP exchange in place to accommodate something like ICE 
> need for one or more exchange with parameters both for the STUN server 
> and the given addresses. I don't say it is impossible however more work 
> will be needed to look at how it could be included. Then one will see 
> how many modifications it would requires. In the end it might be simpler 
> to take on the SIP session initiation and only keep RTSP's media 
> control.  However this is a radical proposal that would significantly 
> change the protocol.

Well, I suspect it would not change the protocol so much as only use a 
subset - the parts for control.

Regarding having RTSP itself support an offer/answer exchange with 
SETUP, generally only one exchange would be needed. So, its just an 
issue of putting more than one address in the SETUP.

> 
> One question I would like to know is how big the envisioned problem will 
> be with servers that reside in NATed networks. I can understand that SIP 
> must consider all cases as it set up communication between users which 
> are more likely to have NATed addresses. However looking at RTSP which 
> has a clear client to server structure, much like HTTP. If you move a 
> server to a NATed address you will also have problem setting up the TCP 
> connection. You simple will require a proxy between the address domains 
> to be able to communicate. Are there many that deploy HTTP servers in 
> private address spaces and actually intend them to be used by other 
> address domains? So far my view has been that as this is server, the 
> server owner can ensure that the server has an IP address that makes it 
> reachable by the intended clients. So in your widget manufacturer 
> example above, why does not he deploy the server in the public IP space. 
> I would assume that he has HTTP server that are also reachable. If you 
> don't share my opinion on this please speak up and give some motivation 
> why this is a useful example.

I think that you are making a fair point; if the widget manufacturer 
meant for this server to be reachable, they need to give it a public 
address.

I think the cases where this might not be true is where "server" becomes 
more loosely defined. You have in mind a big server meant for consumer 
streaming media applications. However, I can imagine using RTSP to do 
things like retrieve voicemails, and even control streams shared 
directly between two users (for example, one user calling another with 
sip and letting the called party hear an mp3 that the caller has on 
their PC). All of these are, IMHO, good uses for RTSP, and none of them 
will work with the symmetric-RTP only approach you are advocating.

[tzeng's comments start] 
   

[tzeng's comemnts end]

So - are these places you want RTSP to work?

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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Subject: Re: A reason to put RTSP server behind NAT (was RE: [MMUSIC] What
 is 	symmetric RTP?)
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Hi Thomas,

I think the application in itself is a valid one. However if one 
provides services that is assumed to be used from the the public net I 
still think it is a reasonable thing to assume that the service provider 
(nanny cam host) in this case uses a public IP address.

I think that Jonathan is right in his observation of the big question. 
Will private persons run RTSP servers to provide functionality in for 
example conversational calls. If the answer is yes, then we need to 
address the RTSP servers behind NAT problem. However looking at the 
current market it seems that the important question is how to solve the 
RTSP server in public network first. If someone is ready to start 
digging into solving this problem, please go ahead and provide input. 
The design of the solution for public IP RTSP servers may need to 
consider something out of a proposal for how to solve the private IP 
RTSP server problem. However it is obvious that such a solution must 
also address how to establish the control connection to the server.

Last I have one comment to what you write:

Thomas Zeng wrote:

>Of course one can ask the daycare to put the RTSP server box in the open,
>but that means additional cost (it costs $5 dollars/month to get even a
>dynamic IP address from Time-Warner cable company here in San Diego), some
>one time fee to change network topology -- not to mention security
>degradation.
>  
>
I kind of react to the last thing. Security degradation: NATs does not provide any security protection, it only provides some obscurity. A NATed machine may of course not be reachable and can therefore not be be directly attacked. However this a firewall can also provide. The major problem is the huge security gap that the deployment of a NAT creates. With NATs there is basically no guaranteed protection against hijacking or eavesdropping attacks as it is impossible to source authenticate. So saying that removing the NAT will reduce the security I think is wrong. 


Best Regards


Magnus Westerlund 

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



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Subject: [MMUSIC] RTSP teleconference, have or not?
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Hi,

It was planned to have a RTSP telecon on Wednesday (2 April). I am 
somewhat short of time but I are willing to have this meeting at 18.00 
CET, which after having changed to daylight savings are GMT+2. As I 
understand it US are changing to daylight saving next sunday and 
therefore the conference will be one hour earlier than usual. This I 
have heard is a bit difficult for the people on the US west cost. Due to 
these considerations I would like to ask your opinion on what to do. Are 
we going to:
1. Cancel the meeting and having it next week instead?
2. Have the meeting at the proposed time.

Please send me your view before 12.00 tuesday (CET).

Best Regards


Magnus Westerlund 

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


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Cc: mmusic@ietf.org, Thomas Zeng <zeng@packetvideo.com>
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Magnus, I concur with Thomas,

Since we have telephone handsets that can encode video in real time,
Since we have PCs at home with WebCams,
Since all these are behind NATs,
Since multicast (and SSM) will not be available/desirable/permitted in many cases,
We need to make sure that a  RTSP server can exist in these environements,
as RTSP is the obvious peer-to-peer video protocol to use.

I suspect the key issue is that we have to assume that both end points (peers)
are behind NATs and then I am afraid that the solution requires a "public" server in the middle.

At this moment I suspect that TURN is one possible candidate(?)

Some kind of a "reversed" RTSP relay could also do the trick;
From the second peer (receiver of streams) point on view that relay would be a plain RTSP server.
The special trick is that the first peer (origin of streams) would need to
contact this public gateway to "expose" that it has something to stream...
And here comes the "reversed" weird stuff:
That first peer acts as a RTSP server but it *initiates* the first TCP connection to the RTSP relay
and that relay acts as a RTSP client but it is listening on some TCP port...

It assumes that there is some way for the first peer to tell the second peer
that the streams are available at that relay server. Email could do that,
or a voice call, or pigeons ;-)

Not sure if/how  this is similar/different to what is intended with TURN ?

Jonathan can you explain what good TURN is supposed to do to RTSP ?


regards,

Philippe Gentric
Software Architect
Philips MP4Net
philippe.gentric@philips.com
http://www.platform4.philips.com


                                                                                                                                              
                                                                                                                                              
                                                          To:   Thomas Zeng <zeng@packetvideo.com>                                            
                                                          cc:   mmusic@ietf.org                                                               
                                                           (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                         
                                                          Subject:    Re: A reason to put RTSP server behind NAT (was RE: [MMUSIC] What is    
               Magnus Westerlund                           symmetric RTP?)                                                                    
               <magnus.westerlund@era.ericsson                                                                                                
               .se>                                       Classification:                                                                     
                                                                                                                                              
               Sent by:                                                                                                                       
               mmusic-admin@ietf.org                                                                                                          
                                                                                                                                              
               31/03/2003 10:14                                                                                                               
                                                                                                                                              
                                                                                                                                              




Hi Thomas,

I think the application in itself is a valid one. However if one
provides services that is assumed to be used from the the public net I
still think it is a reasonable thing to assume that the service provider
(nanny cam host) in this case uses a public IP address.

I think that Jonathan is right in his observation of the big question.
Will private persons run RTSP servers to provide functionality in for
example conversational calls. If the answer is yes, then we need to
address the RTSP servers behind NAT problem. However looking at the
current market it seems that the important question is how to solve the
RTSP server in public network first. If someone is ready to start
digging into solving this problem, please go ahead and provide input.
The design of the solution for public IP RTSP servers may need to
consider something out of a proposal for how to solve the private IP
RTSP server problem. However it is obvious that such a solution must
also address how to establish the control connection to the server.

Last I have one comment to what you write:

Thomas Zeng wrote:

>Of course one can ask the daycare to put the RTSP server box in the open,
>but that means additional cost (it costs $5 dollars/month to get even a
>dynamic IP address from Time-Warner cable company here in San Diego), some
>one time fee to change network topology -- not to mention security
>degradation.
>
>
I kind of react to the last thing. Security degradation: NATs does not provide any security protection, it only provides some obscurity. A NATed machine may of course not be reachable and can therefore not be be directly attacked. However this a firewall
can also provide. The major problem is the huge security gap that the deployment of a NAT creates. With NATs there is basically no guaranteed protection against hijacking or eavesdropping attacks as it is impossible to source authenticate. So saying that
removing the NAT will reduce the security I think is wrong.


Best Regards


Magnus Westerlund

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



_______________________________________________
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To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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Subject: Re: [MMUSIC] RTSP teleconference, have or not?
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I would not mind cancelling or postponing the call, as it would be nice to
be able to sleep in on my b'day.  :-)

On Mon, Mar 31, 2003 at 01:10:49PM +0200, Magnus Westerlund wrote:
> Hi,
>=20
> It was planned to have a RTSP telecon on Wednesday (2 April). I am=20
> somewhat short of time but I are willing to have this meeting at 18.00=20
> CET, which after having changed to daylight savings are GMT+2. As I=20
> understand it US are changing to daylight saving next sunday and=20
> therefore the conference will be one hour earlier than usual. This I=20
> have heard is a bit difficult for the people on the US west cost. Due to=
=20
> these considerations I would like to ask your opinion on what to do. Are=
=20
> we going to:
> 1. Cancel the meeting and having it next week instead?
> 2. Have the meeting at the proposed time.
>=20
> Please send me your view before 12.00 tuesday (CET).

--=20
Coding is easy; All you do is sit staring at a terminal until the drops of
blood form on your forehead.
        -- Anonymous

--gr/z0/N6AeWAPJVB
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From mailnull@www1.ietf.org  Mon Mar 31 13:55:49 2003
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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Tom Marshall'" <tmarshall@real.com>,
        Magnus Westerlund
	 <magnus.westerlund@era.ericsson.se>
Cc: IETF MMUSIC WG <mmusic@ietf.org>
Subject: RE: [MMUSIC] RTSP teleconference, have or not?
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Happy Birtheday, Tom.

I don't mind skipping this Wed's telecon, either. I am behind in RTSP
related work after SF meetings.

- thomas


-----Original Message-----
From: Tom Marshall [mailto:tmarshall@real.com]
Sent: Monday, March 31, 2003 10:18 AM
To: Magnus Westerlund
Cc: IETF MMUSIC WG
Subject: Re: [MMUSIC] RTSP teleconference, have or not?


I would not mind cancelling or postponing the call, as it would be nice to
be able to sleep in on my b'day.  :-)

On Mon, Mar 31, 2003 at 01:10:49PM +0200, Magnus Westerlund wrote:
> Hi,
> 
> It was planned to have a RTSP telecon on Wednesday (2 April). I am 
> somewhat short of time but I are willing to have this meeting at 18.00 
> CET, which after having changed to daylight savings are GMT+2. As I 
> understand it US are changing to daylight saving next sunday and 
> therefore the conference will be one hour earlier than usual. This I 
> have heard is a bit difficult for the people on the US west cost. Due to 
> these considerations I would like to ask your opinion on what to do. Are 
> we going to:
> 1. Cancel the meeting and having it next week instead?
> 2. Have the meeting at the proposed time.
> 
> Please send me your view before 12.00 tuesday (CET).

-- 
Coding is easy; All you do is sit staring at a terminal until the drops of
blood form on your forehead.
        -- Anonymous
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From mailnull@www1.ietf.org  Mon Mar 31 16:39:00 2003
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Date: Mon, 31 Mar 2003 16:39:40 -0500
From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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philippe.gentric@philips.com wrote:
 > Magnus, I concur with Thomas,
 >
 > Since we have telephone handsets that can encode video in real time,
 > Since we have PCs at home with WebCams, Since all these are behind
 > NATs, Since multicast (and SSM) will not be
 > available/desirable/permitted in many cases, We need to make sure
 > that a  RTSP server can exist in these environements, as RTSP is the
 > obvious peer-to-peer video protocol to use.
 >
 > I suspect the key issue is that we have to assume that both end
 > points (peers) are behind NATs and then I am afraid that the solution
 > requires a "public" server in the middle.
 >
 > At this moment I suspect that TURN is one possible candidate(?)

NATs on both sides does not always require a relay on the public 
Internet. It only does if both NATs are symmetric. If at least one is 
full cone, port restricted, etc., then the relay can be avoided. ICE 
will make this determination for you, and only use TURN if connectivity 
cannot be established directly.

 >
 > Some kind of a "reversed" RTSP relay could also do the trick;
 >> From the second peer (receiver of streams) point on view that relay
 >> would be a plain RTSP server.
 > The special trick is that the first peer (origin of streams) would
 > need to contact this public gateway to "expose" that it has something
 > to stream... And here comes the "reversed" weird stuff: That first
 > peer acts as a RTSP server but it *initiates* the first TCP
 > connection to the RTSP relay and that relay acts as a RTSP client but
 > it is listening on some TCP port...
 >
 > It assumes that there is some way for the first peer to tell the
 > second peer that the streams are available at that relay server.
 > Email could do that, or a voice call, or pigeons ;-)
 >
 > Not sure if/how  this is similar/different to what is intended with
 > TURN ?

TURN is similar, in that both sides need to rendezvous at a central 
server. However, only one side needs to know about TURN. The other will 
rendezvous by simply sending media to the TURN server.


 >
 > Jonathan can you explain what good TURN is supposed to do to RTSP ?

It would support the case of the RTSP client and server both behind 
separate, symmetric NAT.

To support it, you need only make sure that RTSP allows the client to 
specify the IP/port on which it will receive media. This is the same 
thing you would need to do to support STUN.

Getting back to this general issue of peer-to-peer RTSP, in my view this 
is the ideal case to consider the model where SIP is used to establish 
the session, and then RTSP is introduced for the control aspects. This 
way, RTSP does not have to reinvent all the work that is being done in 
SIP for nat/fw traversal, and all the applications where the media 
streaming is integrated with person to person communications is very easy.

I think that there are no changes to RTSP to support this. We would need 
some kind of SIP or SDP extensions to signal the usage of RTSP as a 
control for another media stream, and then use SDP extensions to pass 
any RTSP information needed to make this work (for example, the session ID).

Thoughts?

-Jonathan R.

-- 
Jonathan D. Rosenberg, Ph.D.                600 Lanidex Plaza
Chief Scientist                             Parsippany, NJ 07054-2711
dynamicsoft
jdrosen@dynamicsoft.com                     FAX:   (973) 952-5050
http://www.jdrosen.net                      PHONE: (973) 952-5000
http://www.dynamicsoft.com

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