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Hi,

Here is a proposal to the RTSP spec for how to handle NATs using STUN. 
Please read and comment. It is planned to discuss NAT's on wednessday's 
RTSP telecon.

Cheers

Magnus

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

15 RTSP through Firewalls & NATs

15.1 Introduction

   Today there is Network Address Translators (NAT)   [32]  everywhere
   and a protocol must make sure that it can work through them in some
   fashion. The problem with RTSP is that it carries information about
   network  addresses  and  ports  inside  itself. It is primarily the
   media streams that are being blocked by NAT, as the  given  address
   to send to is a local one.

   Firewalls are also middle boxes that needs to be  considered.  They
   are deployed to prevent non-desired traffic/protocols to be used or
   reach the protected network. RTSP is designed to allow  a  firewall
   to  let RTSP controlled streams through, if chosen to, with minimal
   implementation problems.

15.2 NAT

   Today there exist a number of different NAT types and usage  areas.
   The different NAT types, cited from STUN  [31]:

   Full Cone: A full cone NAT is one where all requests from the  same
        internal  IP  address and port are mapped to the same external
        IP address and port. Furthermore, any external host can send a
        packet to the internal host, by sending a packet to the mapped
        external address.

   Restricted Cone: A restricted cone NAT is one  where  all  requests
        from  the  same internal IP address and port are mapped to the
        same external IP address and port. Unlike a full cone NAT,  an
        external  host  (with  IP  address X) can send a packet to the
        internal host only if the internal host had previously sent  a
        packet to IP address X.

   Port Restricted Cone: A port restricted cone NAT  is  like  a  res-
        tricted  cone  NAT, but the restriction includes port numbers.
        Specifically, an external host can send a packet, with  source
        IP  address  X and source port P, to the internal host only if
        the internal host had previously sent a packet to IP address X
        and port P.

   Symmetric: A symmetric NAT is one where all requests from the  same
        internal  IP  address  and  port, to a specific destination IP
        address and port, are mapped to the same external  IP  address
        and port. If the same host sends a packet with the same source
        address and port, but to a different destination, a  different
        mapping  is  used.  Furthermore,  only  the external host that
        receives a packet can send a UDP packet back to  the  internal
        host.

   NAT's are used on both small and  large  scale.  The  normal  small
   scale  user is home user that has a NAT to allow multiple computers
   share the single IP address given by their  Internet  Service  Pro-
   vider  (ISP).  The  large scale users are the ISP's themselves that
   gives there users private addresses. This is done both for  control
   and lack of addresses.

15.2.1 STUN

   The STUN  [31] protocol allows a client to  discover  the  type  of
   NAT(s) he is behind and also discover a mappings public address and
   port number. The protocol also allows  discovery  of  the  mappings
   timeout period and can be used as keep alive mechanism.

   How useful this protocol is depends  on  the  type  of  NAT(s)  the
   client  is  behind.  If  the  user  is behind a full cone NAT, STUN
   allows the RTSP client to traverse the NAT with some simple  client
   side adaptations. For restricted cone NATs STUN is still useful but
   require some more adaptations. For  symmetric  NATs  STUN  requires
   such severe server adaptations that it is not practical to use.

   A RTSP client using RTP transport can use STUN to traverse  a  full
   cone NAT in the following way:

   1.   Use STUN to discover the type of NAT, if any, and the  timeout
        period  for any UDP mapping on the NAT. This is RECOMMENDED to
        be performed in the background as soon as IP  connectivity  is
        established.  If  this  is  performed  prior to the attempt to
        establish a streaming session the possible delays in the  ses-
        sion  establishment will be reduced. If no NAT is present, use
        the normal SETUP behavior.

   2.   The RTSP client determines the number of UDP ports  needed  by
        the  RTP  sessions  part of the multi-media presentation. This
        information is available in  the  media  description  protocol
        used,  e.g.  SDP. In general each RTP session will require two
        UDP ports, one for RTP, and one for RTCP. Ensure that the same
        public  IP  address  is used for each RTP/RTCP port pair esta-
        blished.

   3.   For each UDP port required, establish a mapping  and  discover
        the  public  IP  address  and port number with use of STUN. if
        successful this results in that the client now knows for  each
        port know the mapping: clients local address/port Leftrightar-
        row public address/port.

   4.   Perform the RTSP SETUP for each media. In the transport header
        the  following  parameters  SHOULD  be included with the given
        values:   "destination"   with   the   public   IP    address,
        "client_port"  with  the public port of the mapping determined
        to be used for RTP, "client_rtcp_port" with the port number of
        the   mapping   to   be   used   for   RTCP.   The   parameter
        "client_rtcp_port" needs to be  used  unless  the  client  has
        managed  to  establish  a mapping with two consecutive numbers
        starting with an even one.  To allow this to work servers MUST
        allow  a  client to setup the RTP stream on any port, not only
        even ports. The server SHALL respond with a  transport  header
        containing the source IP address of the media streams.

   5.   To keep the mappings alive the client SHOULD periodically send
        UDP traffic over all mappings needed for the session. STUN can
        be used to determine the timeout period of the NAT(s) UDP map-
        pings. For the mapping carrying RTCP traffic the periodic RTCP
        traffic may be sent frequently enough. If not or for RTP  car-
        rying  mappings,  empty  IP/UDP messages SHOULD be sent to the
        streaming servers discard port (port number 9).

   If a UDP mapping is lost above process is required to be  performed
   again and the media stream effected SETUP again.

   To allow the UDP packets to arrive from  the  server  to  a  client
   behind  a  restricted  NAT,  any  UDP  messages must be sent to the
   server. Therefore SHOULD the client, before  sending  a  RTSP  PLAY
   request  send  an  empty  UDP  message,  on each mapping, to the IP
   address given as the servers source address and  its  discard  port
   (port number 9). Otherwise no difference in procedure compared with
   a full cone NAT is needed.

   For a port restricted NAT the client  must  send  messages  to  the
   exact ports used by the server to send messages. This makes it pos-
   sible to use the above described process with the  following  addi-
   tions: For each port mapping, a UDP message needs to be sent to the
   servers source address/port. To minimize potential effects  on  the
   server  from  these messages the following type of messages MUST be
   sent. RTP: An empty UDP message with  out  any  payload.   RTCP:  A
   correctly  formed RTCP message. Unless enough bandwidth is assigned
   to RTCP it might not be possible to  keep  the  UDP  mapping  open.
   These  messages  SHOULD  be  sent before sending a PLAY request and
   then periodically according to the determined timeout time  of  the
   NAT.

   To be able to use STUN to traverse symmetric NATs the  STUN  server
   needs to be co-located with the streaming server media distribution
   ports. As this will create implementations difficulties and  possi-
   bly security problems this SHOULD NOT be done.

15.2.2 Application Level Gateways

   An Application Level Gateway (ALG) reads the application level mes-
   sages  and  perform necessary changes to allow the protocol to work
   through the middle box. However this behavior has some problems  in
   regards to RTSP:

   1.   Does not work when protocol is used with end-to-end  security.
        As the ALG can't inspect and change the application level mes-
        sages the protocol will fail over the middle box.

   2.   Needs to be updated if extensions to the protocol  are  added.
        Due  to  deployment  issues  with changing ALG's this may also
        break the end-to-end functionality of RTSP.

   Due to the above reasons it is RECOMMENDED to not use ALGs in NATs.
   This  is  especially important for NAT's targeted to home users and
   not ISPs. In that case it is recommended that NATs  are  full  cone
   and  that  the  RTSP client instead used STUN or the planned MIDCOM
   protocols.

   An ALG for the this specification would need to perform the follow-
   ing tasks and changes to RTSP:

   1.   Detect a SETUP request.  Create  UDP  mappings  (client  given
        IP/port  Leftrightarrow  public  IP/port) where needed for all
        possible transport specification in the  transport  header  of
        the request found in (1). Enter the public address and port(s)
        of these mappings  in  transport  header.  Mappings  SHALL  be
        created with consecutive public port number starting on a even
        number for RTP each stream.  Mappings SHOULD also be  given  a
        long timeout period, at least 5 minutes.

   2.   When the response is received from  the  server  the  ALG  MAY
        remove  the  unused UDP mappings, i.e. the ones not present in
        the transport header. The session ID SHOULD also be  bound  to
        the UDP mappings part of that session.

   3.   The ALG SHOULD keep alive the UDP mappings belonging to the an
        RTSP  session  as long as: RTSP messages with the session's ID
        has been sent in the last RTSP session  timeout  interval,  or
        UDP  messages  are  sent on any of the UDP mappings during the
        last RTSP timeout interval.

   4.   The ALG MAY remove a mapping as soon a TEARDOWN  response  has
        been received for that media stream.

15.2.3 TCP Tunneling

   TCP tunneling is the final fallback  to  get  any  RTSP  controlled
   streaming to work through a NAT. As it is the client that establish
   the TCP connection to the server it  will  automatically  create  a
   binding  through  the  NAT. This connection can then be used by the
   server to send media packets in.

   Using TCP tunneling has the drawback on real-time  properties  that
   TCP has with "head of line" blocking. It may also have large varia-
   tions in through put due to the congestion control algorithm.   TCP
   tunneling must also be supported by both server and client.

   See section  10.11 for how to perform TCP tunneling.

15.3 Firewalls

   Firewalls exist for a purpose to protect a network from traffic not
   desired by the firewall owner. Therefore it is a policy decision if
   a firewall will let RTSP and its media streams through or not. RTSP
   is  designed  to  be  as easy as possible to process for a firewall
   with a policy to let the traffic pass through.

   The firewall will need to allow the media streams associated with a
   RTSP  session  pass through it. Therefore the firewall will need an
   ALG that reads RTSP SETUP and TEARDOWN  messages.  By  reading  the
   SETUP message the firewall can determine what type of transport and
   from where the media streams will use. Very common will be the need
   to open UDP ports for RTP/RTCP. By looking at the source and desti-
   nation addresses and ports the  opening  in  the  firewall  can  be
   minimized  to  the least necessary. The opening in the firewall can
   be closed after a teardown message for that session or the  session
   itself timesout.

-- 

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: Mon, 3 Feb 2003 10:15:18 -0800
From: Tom Marshall <tmarshall@real.com>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP and NATs
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On Mon, Feb 03, 2003 at 05:26:22PM +0100, Magnus Westerlund wrote:
> Hi,
>=20
> Here is a proposal to the RTSP spec for how to handle NATs using STUN.=20
> Please read and comment. It is planned to discuss NAT's on wednessday's=
=20
> RTSP telecon.

I believe that a viable NAT traversal protocol should not require changes to
the protocols that it supports.  If it is desired that RTSP and STUN work
well together, I think the proper course of action is to modify the STUN
draft to allow requests for consecutive blocks of mapped UDP ports required
by RTSP and RTP.

Regardless of STUN, it may be useful to have a mechanism for non-contiguous
UDP port mappings.  I would argue in favor of a list based approach that has
general applicability instead of an approach that is specific to RTP.  For
example, the client_port attribute could use '/' to separate non-contiguous
ports:

Transport: rtp/avp/udp;unicast;client_port=3D7834/23478

Having said that, here are my thoughts on STUN in general:

I am not in favor of STUN.  There are other NAT traversal protocols[1] which
are already deployed and adding support for this one just muddies the waters
further.  If someone were to propose a simple and complete solution for all
NATs that everyone could get behind, that would be worth supporting.  STUN
is, by its own admission, not a complete solution[2].

The major focus of STUN seems to be the rapidly growing "NAT in a box" class
of router devices that have proliferated around broadband and wireless.=20
Given the current lack of support for RTSP, why would we assume that vendors
will be any more willing to support STUN?  If the vendors won't support it,
then adding more Transport attributes creates more implementation headaches
for ourselves with little or no benefit.  I think the best solution is to
find out how to persuade the vendors to properly support RTSP.  It is really
not much more difficult to support than, say, FTP.

[1] Another example (with heavy industry backing) is http://www.upnp.org/ .

[2] From STUN draft: "STUN does not enable incoming UDP packets through
    symmetric NATs ..., which are common in large enterprises."

--=20
Windows 2000: Designed for the Internet. The Internet: Designed for UNIX.

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inline.

Tom Marshall wrote:
> On Mon, Feb 03, 2003 at 05:26:22PM +0100, Magnus Westerlund wrote:
> 
>>Hi,
>>
>>Here is a proposal to the RTSP spec for how to handle NATs using STUN. 
>>Please read and comment. It is planned to discuss NAT's on wednessday's 
>>RTSP telecon.
> 
> 
> I believe that a viable NAT traversal protocol should not require changes to
> the protocols that it supports.  If it is desired that RTSP and STUN work
> well together, I think the proper course of action is to modify the STUN
> draft to allow requests for consecutive blocks of mapped UDP ports required
> by RTSP and RTP.

No, that is impossible. You have misunderstood how STUN works.

STUN does not allocate the addresses. It merely tells the client what a 
regular, off the shelf NAT has allocated to a user. Since regular, 
off-the-shelf NATs do not know anything about RTP port pairings, two 
separate STUN requests will result in a nat allocating two bindings 
which are most likely not consecutive (or follow the odd/even rule). 
THis is dealt with through an SDP extension that allows a client to 
provide a specific address and port for SDP 
(http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp4nat-03.txt).


> 
> Regardless of STUN, it may be useful to have a mechanism for non-contiguous
> UDP port mappings.  I would argue in favor of a list based approach that has
> general applicability instead of an approach that is specific to RTP.  For
> example, the client_port attribute could use '/' to separate non-contiguous
> ports:
> 
> Transport: rtp/avp/udp;unicast;client_port=7834/23478

Well, I will take this opportunity to once more complain, as I have done 
at IETF meetings in the past, of the continued cost of repeating 
protocol development related to session initiation twice - once for SIP, 
and once for RTSP. We already have a way to do this representation in 
SDP for SIP. Seems like we need to repeat such design now for RTSP.


> 
> Having said that, here are my thoughts on STUN in general:
> 
> I am not in favor of STUN.  There are other NAT traversal protocols[1] which
> are already deployed and adding support for this one just muddies the waters
> further. 

There are many topologies in which UPnP will not work. These icnlude:

* cases where the residential nat doesnt support it
* cases where there are multiple nats in tandem, such as cable operators 
that nat their networks
* enterprises with multiple access points to the public Internet


> If someone were to propose a simple and complete solution for all
> NATs that everyone could get behind, that would be worth supporting.  STUN
> is, by its own admission, not a complete solution[2].

Heh. How about world peace while you are at it.

The problem is this. Experience has shown that solutions which ALWAYS 
work tend to be way too expensive in situations that are far more 
simpler solution will do. There is therefore a cost/generality curve 
with a highly non-linear behavior, which results in solutions being 
defined across this curve. STUN happens to be extremely cheap to do in 
the cases where it works. THe most general solution, which are 
media-relay types of solutions, are very expensive because of the need 
to route media through intermediaries.


> 
> The major focus of STUN seems to be the rapidly growing "NAT in a box" class
> of router devices that have proliferated around broadband and wireless. 
> Given the current lack of support for RTSP, why would we assume that vendors
> will be any more willing to support STUN? 

You have again misunderstood how stun works. STUN requires no support in 
NATs. A provider that wants to provide streaming media services using 
rtsp would deploy the stun server, NOT the ISP or IP access provider of 
the end user.


> If the vendors won't support it,
> then adding more Transport attributes creates more implementation headaches
> for ourselves with little or no benefit.  I think the best solution is to
> find out how to persuade the vendors to properly support RTSP.  It is really
> not much more difficult to support than, say, FTP.

I sinceerely doubt that.

What incentive has Linksys, Netgear, and the REAMS of other home nat 
boxes to support ALGs for RTSP? Are those companies the experts on RTSP? 
Surely not! The need to separate out applications from NATs and 
firewalls is central to the midcom working group and much of the other 
work going on in IETF.


ANYWAY, all of that said (just responding to this assault on STUN), I do 
not think STUN by itself is the right solution to RTSP nat traversal. 
THere are two options that are better.

OPTION 1: Symmtric RTP. See 
http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-comedia-04.txt. 
Effectively, the IP address/ports provided by the RTSP client are 
ignored by the server. The client sends RTP/RTCP packets to the server, 
and the server sends media back to the client at the source ip/port that 
the RTP/RTCP came from. Requires no changes to the RTSP protocol, except 
to recommend this method. No STUN support needed. Works through all NAT 
types. It will require the client to send keepalive packets if there is 
a pause in media.

NB: This ONLY works if the RTSP server is on the public Internet!!!!

OPTION 2: Coming Soon
I am working on a draft (out before the -00 deadline end of this month) 
that finally unifies all of the mess of nat traversal together - STUN, 
TURN, symmetric RTP. It has the property that it always results in the 
optimal media path, lowest latency, lowest cost, and highest probability 
of working (i.e., you hear media), compared to any other technique. In 
the case of the RTSP server running on the public Internet, it would end 
up working a lot like OPTION 1 above. But, it would also work in cases 
where that was not true, for example, a client connecting to a media 
server in another enterprise that is behind a nat.

More to come soon.


In fact, one might say that the best approach for nat traversal is not 
to document it in the RTSP revision itself, as there are multiple 
solutions in any case. We have done that for sip, documenting solutions 
in a separate document:

http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-scenarios-00.txt

Then there are other thigns coming like nsis. So, keeping this separate 
from RTSP itself is probably a good thing.

-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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Date: Mon, 3 Feb 2003 14:39:39 -0800
From: Tom Marshall <tmarshall@real.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Magnus Westerlund <magnus.westerlund@era.ericsson.se>,
        IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP and NATs
Message-ID: <20030203223939.GB26777@real.com>
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On Mon, Feb 03, 2003 at 03:11:57PM -0500, Jonathan Rosenberg wrote:
> inline.
>=20
> Tom Marshall wrote:
> >On Mon, Feb 03, 2003 at 05:26:22PM +0100, Magnus Westerlund wrote:
> >
> >>Hi,
> >>
> >>Here is a proposal to the RTSP spec for how to handle NATs using STUN.=
=20
> >>Please read and comment. It is planned to discuss NAT's on wednessday's=
=20
> >>RTSP telecon.
> >
> >
> >I believe that a viable NAT traversal protocol should not require change=
s=20
> >to
> >the protocols that it supports.  If it is desired that RTSP and STUN work
> >well together, I think the proper course of action is to modify the STUN
> >draft to allow requests for consecutive blocks of mapped UDP ports requi=
red
> >by RTSP and RTP.
>=20
> No, that is impossible. You have misunderstood how STUN works.

Yes, you are correct.  On initial reading of the STUN draft, I assumed
"server" meant the NAT device.  My understanding now is that the STUN server
resides on a publicly accessible address outside and independent from any
NAT.

> STUN does not allocate the addresses. It merely tells the client what a=
=20
> regular, off the shelf NAT has allocated to a user. Since regular,=20
> off-the-shelf NATs do not know anything about RTP port pairings, two=20
> separate STUN requests will result in a nat allocating two bindings=20
> which are most likely not consecutive (or follow the odd/even rule).=20
> THis is dealt with through an SDP extension that allows a client to=20
> provide a specific address and port for SDP=20
> (http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp4nat-03.txt).

This draft again assumes that RTP is the only transport choice:

  rtcp-attribute =3D  "a=3Drtcp:" port

What if there were some other UDP based transport that required two UDP
ports (perhaps even "RTPng")?  You would need yet another draft to address
it.  Why not make the solution generic instead?

> >Regardless of STUN, it may be useful to have a mechanism for non-contigu=
ous
> >UDP port mappings.  I would argue in favor of a list based approach that=
=20
> >has
> >general applicability instead of an approach that is specific to RTP.  F=
or
> >example, the client_port attribute could use '/' to separate non-contigu=
ous
> >ports:
> >
> >Transport: rtp/avp/udp;unicast;client_port=3D7834/23478
>=20
> Well, I will take this opportunity to once more complain, as I have done=
=20
> at IETF meetings in the past, of the continued cost of repeating=20
> protocol development related to session initiation twice - once for SIP,=
=20
> and once for RTSP. We already have a way to do this representation in=20
> SDP for SIP. Seems like we need to repeat such design now for RTSP.

I am not familiar with SIP (although I will read up on it before the next
meeting).  My only concern is that any modification to RTSP does not limit
itself needlessly.  If SIP has done so (and it appears that at least one SDP
draft is doing so), then perhaps those decisions should be reconsidered.

> >If someone were to propose a simple and complete solution for all
> >NATs that everyone could get behind, that would be worth supporting.  ST=
UN
> >is, by its own admission, not a complete solution[2].
>=20
> Heh. How about world peace while you are at it.

And a complete IPv6 transition too, eh (reducing or eliminating NAT)? ;-)

> The problem is this. Experience has shown that solutions which ALWAYS=20
> work tend to be way too expensive in situations that are far more=20
> simpler solution will do. There is therefore a cost/generality curve=20
> with a highly non-linear behavior, which results in solutions being=20
> defined across this curve. STUN happens to be extremely cheap to do in=20
> the cases where it works. THe most general solution, which are=20
> media-relay types of solutions, are very expensive because of the need=20
> to route media through intermediaries.

I have just started looking at STUN but I have some questions:

1. Given a NAT that supports RTSP (eg. Linux 2.2 masquerading with an RTSP
   module loaded), will the client "detect" that it is behind a NAT?  If so,
   how does this affect the Transport: header as it passes through?  It
   seems to me that the RTSP enabled NAT will break the STUN-enabled
   Transport: header.  If this is the case, I think that STUN should be
   reconsidered (at least for RTSP) as would break an otherwise functional
   NAT setup.

2. Given a successful STUN setup, does the STUN client direct all UDP
   traffic to the STUN server as a relay point, or does the client assume
   that all UDP packets sent between the STUN mapped port and any outside
   address will have the same NAT mapping?  I will have to do some checking,
   but IIRC the Linux 'ipchains' NAT code does mappings with a '5-tuple':
   client ip, client port, nat port, server ip, server port.  If the latter
   case is assumed (no relay), I don't think that the STUN mappings would
   behave as described in the draft.  Packets sent to and from the appserver
   will have a different 5-tuple, and thus a different nat port, rendering
   STUN ineffective.  Has this case been considered?

> >If the vendors won't support it, then adding more Transport attributes
> >creates more implementation headaches for ourselves with little or no
> >benefit.  I think the best solution is to find out how to persuade the
> >vendors to properly support RTSP.  It is really not much more difficult
> >to support than, say, FTP.
>=20
> I sinceerely doubt that.

Which do you doubt?  The implementation headaches or that RTSP support is
not difficult in NAT?

> ANYWAY, all of that said (just responding to this assault on STUN), I do=
=20
> not think STUN by itself is the right solution to RTSP nat traversal.=20
> THere are two options that are better.
>=20
> OPTION 1: Symmtric RTP. See=20
> http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-comedia-04.txt.=
=20
> Effectively, the IP address/ports provided by the RTSP client are=20
> ignored by the server. The client sends RTP/RTCP packets to the server,=
=20
> and the server sends media back to the client at the source ip/port that=
=20
> the RTP/RTCP came from. Requires no changes to the RTSP protocol, except=
=20
> to recommend this method. No STUN support needed. Works through all NAT=
=20
> types. It will require the client to send keepalive packets if there is=
=20
> a pause in media.
>=20
> NB: This ONLY works if the RTSP server is on the public Internet!!!!

This is similar to a technique that we (RealNetworks) use for our internal
transport.  It is quite effective (but not 100% of course).  The biggest
problem I see is that client->server RTP traffic is not expected.  There is
some chance that an intermediate device or proxy may behave badly.  But so
far, it seems to be the best choice.

> OPTION 2: Coming Soon
> I am working on a draft (out before the -00 deadline end of this month)=
=20
> that finally unifies all of the mess of nat traversal together - STUN,=20
> TURN, symmetric RTP. It has the property that it always results in the=20
> optimal media path, lowest latency, lowest cost, and highest probability=
=20
> of working (i.e., you hear media), compared to any other technique. In=20
> the case of the RTSP server running on the public Internet, it would end=
=20
> up working a lot like OPTION 1 above. But, it would also work in cases=20
> where that was not true, for example, a client connecting to a media=20
> server in another enterprise that is behind a nat.
>=20
> More to come soon.

I will keep an eye out, thanks.

--=20
You can complain because roses have thorns, or you can rejoice because
thorns have roses.
        -- Abraham Lincoln

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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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        Magnus Westerlund <magnus.westerlund@era.ericsson.se>,
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Subject: Re: [MMUSIC] RTSP and NATs 
In-Reply-To: Your message of "Mon, 03 Feb 2003 15:11:57 EST."
             <3E3ECD0D.8010601@dynamicsoft.com> 
Date: Mon, 03 Feb 2003 18:04:02 -0500
From: Colin Perkins <csp@isi.edu>
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--> Jonathan Rosenberg writes:
...
>In fact, one might say that the best approach for nat traversal is not 
>to document it in the RTSP revision itself, as there are multiple 
>solutions in any case. We have done that for sip, documenting solutions 
>in a separate document:
>
>http://www.ietf.org/internet-drafts/draft-ietf-sipping-nat-scenarios-00.txt
>
>Then there are other thigns coming like nsis. So, keeping this separate 
>from RTSP itself is probably a good thing.

Perhaps, although the RTSP spec should probably address any issues related
to NAT/firewall traversal. Especially since RTSP supports interleaving a
media stream on the control channel, which has quite different properties
than separate media.

Colin
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Tom Marshall wrote:

>>STUN does not allocate the addresses. It merely tells the client what a 
>>regular, off the shelf NAT has allocated to a user. Since regular, 
>>off-the-shelf NATs do not know anything about RTP port pairings, two 
>>separate STUN requests will result in a nat allocating two bindings 
>>which are most likely not consecutive (or follow the odd/even rule). 
>>THis is dealt with through an SDP extension that allows a client to 
>>provide a specific address and port for SDP 
>>(http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp4nat-03.txt).
> 
> 
> This draft again assumes that RTP is the only transport choice:
> 
>   rtcp-attribute =  "a=rtcp:" port
> 
> What if there were some other UDP based transport that required two UDP
> ports (perhaps even "RTPng")?  You would need yet another draft to address
> it.  Why not make the solution generic instead?

Well, if thats so, I would have rather had such a debate in the context 
of the above draft. Anyway, I know of no other media stream that uses 
UDP that uses multiple ports AND that has an algorithmic relationship 
between those ports (i.e., has no way to express in the SDP which ports 
are needed). Seems like a non-problem.

> 
> 
>>>Regardless of STUN, it may be useful to have a mechanism for non-contiguous
>>>UDP port mappings.  I would argue in favor of a list based approach that 
>>>has
>>>general applicability instead of an approach that is specific to RTP.  For
>>>example, the client_port attribute could use '/' to separate non-contiguous
>>>ports:
>>>
>>>Transport: rtp/avp/udp;unicast;client_port=7834/23478
>>
>>Well, I will take this opportunity to once more complain, as I have done 
>>at IETF meetings in the past, of the continued cost of repeating 
>>protocol development related to session initiation twice - once for SIP, 
>>and once for RTSP. We already have a way to do this representation in 
>>SDP for SIP. Seems like we need to repeat such design now for RTSP.
> 
> 
> I am not familiar with SIP (although I will read up on it before the next
> meeting).  My only concern is that any modification to RTSP does not limit
> itself needlessly.  If SIP has done so (and it appears that at least one SDP
> draft is doing so), then perhaps those decisions should be reconsidered.

Ultimately, I'd like to see SIP used to set up sessions both 
point-to-point and to media servers, and if control is needed, RTSP be 
introduced in order to achieve that end. There are many benefits from 
using a single protocol to achieve setup. We've stumbled onto one of 
them (duplication of specification work). Another is a single 
call/communications routing infrastructure. I have been hoping to find 
the time to write that up in an I-D, but there are always higher 
priority ones than chasing a goal that no one else seems to care about...


> 
> 
>>>If someone were to propose a simple and complete solution for all
>>>NATs that everyone could get behind, that would be worth supporting.  STUN
>>>is, by its own admission, not a complete solution[2].
>>
>>Heh. How about world peace while you are at it.
> 
> 
> And a complete IPv6 transition too, eh (reducing or eliminating NAT)? ;-)

Dont get me started...

> 
> 
>>The problem is this. Experience has shown that solutions which ALWAYS 
>>work tend to be way too expensive in situations that are far more 
>>simpler solution will do. There is therefore a cost/generality curve 
>>with a highly non-linear behavior, which results in solutions being 
>>defined across this curve. STUN happens to be extremely cheap to do in 
>>the cases where it works. THe most general solution, which are 
>>media-relay types of solutions, are very expensive because of the need 
>>to route media through intermediaries.
> 
> 
> I have just started looking at STUN but I have some questions:
> 
> 1. Given a NAT that supports RTSP (eg. Linux 2.2 masquerading with an RTSP
>    module loaded), will the client "detect" that it is behind a NAT? 

Yes.

>    If so,
>    how does this affect the Transport: header as it passes through?  It
>    seems to me that the RTSP enabled NAT will break the STUN-enabled
>    Transport: header.  If this is the case, I think that STUN should be
>    reconsidered (at least for RTSP) as would break an otherwise functional
>    NAT setup.

A smart ALG won't. It should know not to muck with IP addresses inside 
of the RTSP message if they are not private addresses (if stun is used, 
those addresses will have already been changed to public ones by the 
RTSP client). Indeed, one can imagine usign RTSP to setup a streaming 
media session to some third party device outside of the nat, in which 
case the ALG has to know not to change the IP. I wonder if the linux 
code you cite supports that.

Which, of course, brings me to my beef with ALG - the wrong people write 
them in general. Only an RTSP expert would probably know to look for 
such nuances (even then...), and Linksys and Netgear to not hire such 
people - its not their business.


> 
> 2. Given a successful STUN setup, does the STUN client direct all UDP
>    traffic to the STUN server as a relay point,

No, the stun server is NOT a relay point.

>    or does the client assume
>    that all UDP packets sent between the STUN mapped port and any outside
>    address will have the same NAT mapping? 

STUn works a binding discovery mechanism ONLY if the client can assume 
that any client outside of the nat can send to that address.



> I will have to do some checking,
>    but IIRC the Linux 'ipchains' NAT code does mappings with a '5-tuple':
>    client ip, client port, nat port, server ip, server port.  If the latter
>    case is assumed (no relay), I don't think that the STUN mappings would
>    behave as described in the draft.  Packets sent to and from the appserver
>    will have a different 5-tuple, and thus a different nat port, rendering
>    STUN ineffective.  Has this case been considered?

Certainly. This is called "symmetric NAT". Its common but to date the 
full-cone variety, which maps bindings by private IP/port alone, is 
quite prevalent amongst the home NATs. The presenec of a symmtric NAT is 
discoverable by STUN, in which case you need somethign else to receive 
media. Symmetric RTP, for example, is an application-specific method. 
Another would be TURN 
(http://www.jdrosen.net/papers/draft-rosenberg-midcom-turn-00.txt) which 
does use a relay.

> 
> 
>>>If the vendors won't support it, then adding more Transport attributes
>>>creates more implementation headaches for ourselves with little or no
>>>benefit.  I think the best solution is to find out how to persuade the
>>>vendors to properly support RTSP.  It is really not much more difficult
>>>to support than, say, FTP.
>>
>>I sinceerely doubt that.
> 
> 
> Which do you doubt?  The implementation headaches or that RTSP support is
> not difficult in NAT?

That RTSP ALGs are likely to be supported widely in the vast array of 
$100 NAT boxes on the market. FTP *is* widely supported, but RTSP is a 
domain-specific protocol, and I suspect not on their radar screens.


> 
> 
>>ANYWAY, all of that said (just responding to this assault on STUN), I do 
>>not think STUN by itself is the right solution to RTSP nat traversal. 
>>THere are two options that are better.
>>
>>OPTION 1: Symmtric RTP. See 
>>http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-comedia-04.txt. 
>>Effectively, the IP address/ports provided by the RTSP client are 
>>ignored by the server. The client sends RTP/RTCP packets to the server, 
>>and the server sends media back to the client at the source ip/port that 
>>the RTP/RTCP came from. Requires no changes to the RTSP protocol, except 
>>to recommend this method. No STUN support needed. Works through all NAT 
>>types. It will require the client to send keepalive packets if there is 
>>a pause in media.
>>
>>NB: This ONLY works if the RTSP server is on the public Internet!!!!
> 
> 
> This is similar to a technique that we (RealNetworks) use for our internal
> transport.  It is quite effective (but not 100% of course). 

I am curious about what problems you have encountered.


-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
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Date: Mon, 3 Feb 2003 17:22:16 -0800
From: Tom Marshall <tmarshall@real.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: Magnus Westerlund <magnus.westerlund@era.ericsson.se>,
        IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP and NATs
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References: <3E3E982E.2010706@era.ericsson.se> <20030203181518.GA26315@real.com> <3E3ECD0D.8010601@dynamicsoft.com> <20030203223939.GB26777@real.com> <3E3EFF32.2090103@dynamicsoft.com>
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On Mon, Feb 03, 2003 at 06:45:54PM -0500, Jonathan Rosenberg wrote:
>=20
>=20
> Tom Marshall wrote:
>=20
> >>STUN does not allocate the addresses. It merely tells the client what a=
=20
> >>regular, off the shelf NAT has allocated to a user. Since regular,=20
> >>off-the-shelf NATs do not know anything about RTP port pairings, two=20
> >>separate STUN requests will result in a nat allocating two bindings=20
> >>which are most likely not consecutive (or follow the odd/even rule).=20
> >>THis is dealt with through an SDP extension that allows a client to=20
> >>provide a specific address and port for SDP=20
> >>(http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp4nat-03.txt).
> >
> >
> >This draft again assumes that RTP is the only transport choice:
> >
> >  rtcp-attribute =3D  "a=3Drtcp:" port
> >
> >What if there were some other UDP based transport that required two UDP
> >ports (perhaps even "RTPng")?  You would need yet another draft to addre=
ss
> >it.  Why not make the solution generic instead?
>=20
> Well, if thats so, I would have rather had such a debate in the context=
=20
> of the above draft. Anyway, I know of no other media stream that uses=20
> UDP that uses multiple ports AND that has an algorithmic relationship=20
> between those ports (i.e., has no way to express in the SDP which ports=
=20
> are needed). Seems like a non-problem.

Prior to IPv6 there was no known way to express a numeric IP address such
that it would not fit into a 32-bit integer.  Now there is, and it is one
source of poring issues.  Of course this is not a problem now.  Can you say
with certainty that it will never be a problem?

There is no need to deliberately limit functionality in this way.  Perhaps
SDP is committed to its course, although it appears to still be in draft
stage.  We don't have to do the same with RTSP.

> >I have just started looking at STUN but I have some questions:
> >
> >1. Given a NAT that supports RTSP (eg. Linux 2.2 masquerading with an RT=
SP
> >   module loaded), will the client "detect" that it is behind a NAT?=20
>=20
> Yes.
>=20
> >   If so,
> >   how does this affect the Transport: header as it passes through?  It
> >   seems to me that the RTSP enabled NAT will break the STUN-enabled
> >   Transport: header.  If this is the case, I think that STUN should be
> >   reconsidered (at least for RTSP) as would break an otherwise function=
al
> >   NAT setup.
>=20
> A smart ALG won't. It should know not to muck with IP addresses inside=20
> of the RTSP message if they are not private addresses (if stun is used,=
=20
> those addresses will have already been changed to public ones by the=20
> RTSP client). Indeed, one can imagine usign RTSP to setup a streaming=20
> media session to some third party device outside of the nat, in which=20
> case the ALG has to know not to change the IP. I wonder if the linux=20
> code you cite supports that.

All of the four or so RTSP modules that I am familiar with simply look at
client_port=3D and perform a mapping.  In order to detect that a transport
spec is using STUN, the module would also need to check for a destination=3D
attribute.

> Which, of course, brings me to my beef with ALG - the wrong people write=
=20
> them in general. Only an RTSP expert would probably know to look for=20
> such nuances (even then...),=20

Indeed.  I have written one of the modules referenced above and I did not
consider it.  On the other hand, the destination=3D parameter has always be=
en
a security risk at the server side and I am reluctant to advocate using it.=
=20

Unfortunately, the overlap of people who would check for the destination=3D
parameter and the people who know how to code ALGs is very very small.=20
Perhaps this argues for simpler protocol specs?

> and Linksys and Netgear to not hire such people - its not their business.

It certainly is their business if they make the equipment.  It's enough of
their business that they provide an ALG for FTP.  But I think it should also
be (or should have been) the business of the IETF to design a NAT traversal
protocol which allows the client to directly request mapped ports from the
NAT device.  Such a protocol would reduce or eliminate the need for STUN and
friends, and it would provide a single protocol for the vendors to implement
instead of a running treadmill of RFCs.

But the vendors have already been given a single protocol to implement:
UPnP.  It was created from the desire for NAT control at the client and
appears to have a lot of support from NAT vendors already.  And this makes
me wonder why we are continuing to propose more NAT traversal solutions.=20
Why not either recommend UPnP or design an equivalent?

> >I will have to do some checking,
> >   but IIRC the Linux 'ipchains' NAT code does mappings with a '5-tuple':
> >   client ip, client port, nat port, server ip, server port.  If the lat=
ter
> >   case is assumed (no relay), I don't think that the STUN mappings would
> >   behave as described in the draft.  Packets sent to and from the=20
> >   appserver
> >   will have a different 5-tuple, and thus a different nat port, renderi=
ng
> >   STUN ineffective.  Has this case been considered?
>=20
> Certainly. This is called "symmetric NAT". Its common but to date the=20
> full-cone variety, which maps bindings by private IP/port alone, is=20
> quite prevalent amongst the home NATs.

Interesting.  Where can I find out more about how many home networks use
which types of NAT?

> >>>If the vendors won't support it, then adding more Transport attributes
> >>>creates more implementation headaches for ourselves with little or no
> >>>benefit.  I think the best solution is to find out how to persuade the
> >>>vendors to properly support RTSP.  It is really not much more difficult
> >>>to support than, say, FTP.
> >>
> >>I sinceerely doubt that.
> >
> >
> >Which do you doubt?  The implementation headaches or that RTSP support is
> >not difficult in NAT?
>=20
> That RTSP ALGs are likely to be supported widely in the vast array of=20
> $100 NAT boxes on the market. FTP *is* widely supported, but RTSP is a=20
> domain-specific protocol, and I suspect not on their radar screens.

Unfortunately, I doubt any protocols requiring special handling besides FTP
are on their radar screens.  But UPnP compliance is, and together with the
ALGs in Linux and FreeBSD and Symmetric RTP, I suspect that a vast majority
of the home NATs are covered without STUN.

> >>OPTION 1: Symmtric RTP. See=20
> >>http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp-comedia-04.tx=
t.=20
> >
> >This is similar to a technique that we (RealNetworks) use for our intern=
al
> >transport.  It is quite effective (but not 100% of course).=20
>=20
> I am curious about what problems you have encountered.

I was not personally involved in the testing but I can probably find someone
who was.

--=20
Happiness isn't something you experience; it's something you remember.
        -- Oscar Levant

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inline.

David Yon wrote:
> At 09:02 PM 1/14/2003, Jonathan Rosenberg wrote:
> 
>> Responses inline.
>>
>> David Yon wrote:
>>
>>> comedia-fix again raises the argument against embedding the source 
>>> address/port information in the SDP.  That is something that I still 
>>> feel strongly about, and work has progressed that makes use of it, 
>>> although not as a fail-safe session correlation mechanism.  We've 
>>> found that it's useful as a DoS counter-measure, using the following 
>>> design:
>>>     - An intermediate proxy on our network acts as a STUN-like
>>>       inverse-NAT translator to at least recover the likely address
>>>       from where the media will originate, and rewrites the SDP
>>>       if necessary.
>>
>>
>> It is quite a mystery to me how such a thing could possibly work in a 
>> causal fashion. The endpoint won't actually open any connections until 
>> after the offer/answer exchange has completed. The opening of the TCP 
>> connection is the thing which causes the NAT to assign a binding. So, 
>> the binding is assigned after the o/a exchange, but your proxy needs 
>> to know the binding before the exchange completes. So, I don't get it.
> 
> 
> It works on ranking rather than absolute correlation. 

Ranking?

> You're correct 
> about the NPAT port binding not being known ahead of time.  However, 
> most NPAT scenarios involve a single IP Address, so a media connection 
> coming from that IP Address (regardless of source port) would be ranked 
> higher than one that comes an unknown address.

None of this makes sense to me. The problem remains that the proxy 
cannot know anything about the allocated nat binding until media is sent.

> 
> And let's not forget about connections that originate from dial-up 
> accounts (which accounts for the vast majority of home access), where 
> the IP Address and port are actually correct, in which case the ranking 
> is 100% accurate.

Still makes no sense to me.

Fact remains, that since you cannot know whether the source IP is 
accurate, it is simply unsafe to use.


> 
> 
>>>     - Our gateways, when under load, sort the incoming connection
>>>       pending queue based on source addresses that they expect
>>>       to arrive.
>>> The idea is that when under DoS attack, connections sourced from 
>>> uninvited addresses are not dealt with until the queue of invited 
>>> addresses has been serviced.  This is especially critical when the 
>>> media is carried over TLS, and in our case we do not commence with 
>>> the TLS handshake until either the connection has been correlated or 
>>> until servicing the connection does not overload the gateway.  The 
>>> side-effect of a failed/incorrect inverse-NAT rewrite only occurs 
>>> under heavy load, and is not fatal.  But without the source address 
>>> information, there is no way to implement such a scheme.
>>
>>
>> Since, practically speaking, I would expect the majority of users to 
>> connect to things from behind a NAT, this mechanism serves little 
>> purpose to prevent DOS attacks. I don't understand why you would not 
>> use more tried-and-true mechanisms, such as return routability, which 
>> validate a source IP address by sending it a cookie, and getting this 
>> cookie back. Such a mechanism actually works through NAT, and you can 
>> provide much more than just service prioritiziation based on it.
> 
> 
> Again, the scheme as described works in the presence of NAT (despite 
> assertions to the contrary), and allows the media to be 100% TLS without 
> any out-of-spec "enhancements".

You simply dismiss my argument, but do not address it.

My argument is, that when most users are behind NAT, the source IP of 
the TCP connection won't match that in the SDP, and the net result is 
that most if not all connections go in the low priority pool. This is 
equivalent to no priority. I have also asserted that there are more 
reliable, well-known techniques for DoS protection against these kinds 
of attacks, which work better.


> 
>> I still maintain that, given we are painfully aware of the trials and 
>> tribulations of distributing source IP addresses in protocols, and 
>> given the vast potential for misuse, and the potentially small (if not 
>> ZERO) deployable benefits of this mechanism, I would like to see it 
>> removed. It is one thing to include a mechanism that is useful in many 
>> cases, but might not work in some (and then documenting such) as 
>> compared to a mechanism that will most likely be misused, and whose 
>> real benefit is small or zero.
> 
> 
> It is OPTIONAL in the spec, and the limitations clearly outlined.  At 
> least one implementation finds it useful.  I don't know how to be more 
> clear than that, and why it causes such heartburn.

Because its unsafe. You have put a piece of information there, for which 
there is NO WAY to ever know whether it is reliable. Its not that it 
might be wrong, you don't know when its wrong and when its not. If I am 
building a proxy that wishes to look at the source IP in the SDP, and 
configure a pinhole in a firewall, instead of a cone, what advice can 
you give me about when I can use this source IP field? Your only answer 
is, you can only use it when you know its right because you control 
every piece of equipment on the network (the clients and all intervening 
nats). How can I be sure that this is the case? Someone all of a sudden 
installs a linksys and everything stops working, because the pinholes 
are not opened properly. That would be a fun customer call to get at 2 
in the morning.

Anyway, this has remained a largely one-on-one discussion. I think all 
that can be said has been said, on my part anyway. I would like to hear 
from our esteemed chairs on whether this feature stays or goes.

-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
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David Yon wrote:
> At 09:58 PM 1/14/2003, Jonathan Rosenberg wrote:
> 
>> David,
>>
>> Ben, Paul and I discussed your comments extensively over the last week 
>> or so. Our conclusion was that we will not use comedia for simple 
>> messaging sessions. At a high level, the conclusion was that we were 
>> looking at "message sessions over TCP" and that comedia was perhaps 
>> optimized for "streaming media sessions over TCP". As such, many of 
>> our requirements that were critical, or assumptions we could make 
>> (such as demux within a single TCP connection) don't necessarily apply 
>> to streaming media over TCP.
>>
>> Now, I still think that much of what we proposed is valid. However, it 
>> is no longer critical to get it changed. So, keep that in mind as you 
>> read my responses below.
> 
> 
> The the meta-question is whether comedia-fix will continue to block 
> comedia's progress.  On one hand you imply above that it won't, but then 
> again we have the rest of this email that argues that comedia is 
> severely flawed.  So, where are we going from here?

I still have a big beef with the source IP/port, per separate email. 
Beyond that, I believe that the limitations of comedia will restrict its 
utility; certainly, they limited the ability for me to use it. But since 
I am not using it, I am not going to complain further about things where 
  nothing is broken per se, its just not what I think is the right 
thing. Thus, the one port vs. many port debate, and connection lifetime 
issues, I won't pursue further.

However, the only other one is the security issue I have pointed out, 
where an eavesdropper can open connections allowing it to both send and 
receive media. You need to address it, both with protocol machinery and 
a note in security considerations. Your proposed solution of detecting a 
second connection on the same port, and then closing both and indicating 
an error or attack, is OK I believe, but it needs to be added.

-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
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inline.

Tom Marshall wrote:
> On Mon, Feb 03, 2003 at 06:45:54PM -0500, Jonathan Rosenberg wrote:
> 
>>
>>Tom Marshall wrote:
>>
>>
>>>>STUN does not allocate the addresses. It merely tells the client what a 
>>>>regular, off the shelf NAT has allocated to a user. Since regular, 
>>>>off-the-shelf NATs do not know anything about RTP port pairings, two 
>>>>separate STUN requests will result in a nat allocating two bindings 
>>>>which are most likely not consecutive (or follow the odd/even rule). 
>>>>THis is dealt with through an SDP extension that allows a client to 
>>>>provide a specific address and port for SDP 
>>>>(http://www.ietf.org/internet-drafts/draft-ietf-mmusic-sdp4nat-03.txt).
>>>
>>>
>>>This draft again assumes that RTP is the only transport choice:
>>>
>>> rtcp-attribute =  "a=rtcp:" port
>>>
>>>What if there were some other UDP based transport that required two UDP
>>>ports (perhaps even "RTPng")?  You would need yet another draft to address
>>>it.  Why not make the solution generic instead?
>>
>>Well, if thats so, I would have rather had such a debate in the context 
>>of the above draft. Anyway, I know of no other media stream that uses 
>>UDP that uses multiple ports AND that has an algorithmic relationship 
>>between those ports (i.e., has no way to express in the SDP which ports 
>>are needed). Seems like a non-problem.
> 
> 
> Prior to IPv6 there was no known way to express a numeric IP address such
> that it would not fit into a 32-bit integer.  Now there is, and it is one
> source of poring issues.  Of course this is not a problem now.  Can you say
> with certainty that it will never be a problem?

Seems like a bit of an over-exaggerated analogy. As I said, for any new 
protocol that needs multiple ports, it should simply make sure that its 
SDP parameter definitions include them. In such a case you will never 
have a problem.

Anyway, the point is moot, since, based on my limited RTSP knowledge, 
SDP is not used in the client addresses anyway. Thus, you can't use the 
SDP draft above.



>>A smart ALG won't. It should know not to muck with IP addresses inside 
>>of the RTSP message if they are not private addresses (if stun is used, 
>>those addresses will have already been changed to public ones by the 
>>RTSP client). Indeed, one can imagine usign RTSP to setup a streaming 
>>media session to some third party device outside of the nat, in which 
>>case the ALG has to know not to change the IP. I wonder if the linux 
>>code you cite supports that.
> 
> 
> All of the four or so RTSP modules that I am familiar with simply look at
> client_port= and perform a mapping.  In order to detect that a transport
> spec is using STUN, the module would also need to check for a destination=
> attribute.

Right. Although, I must admit confusion. If the client doesnt provide a 
destination attribute, to what IP address is the media sent? It seems to 
me that the destination parameter would normally be there. Or perhaps 
you are saying it is there, but its not looked at by the ALG?


> 
> 
>>Which, of course, brings me to my beef with ALG - the wrong people write 
>>them in general. Only an RTSP expert would probably know to look for 
>>such nuances (even then...), 
> 
> 
> Indeed.  I have written one of the modules referenced above and I did not
> consider it.  On the other hand, the destination= parameter has always been
> a security risk at the server side and I am reluctant to advocate using it. 

I guess I don't understand how media is delivered to the client then.


> It certainly is their business if they make the equipment.  It's enough of
> their business that they provide an ALG for FTP.  But I think it should also
> be (or should have been) the business of the IETF to design a NAT traversal
> protocol which allows the client to directly request mapped ports from the
> NAT device.  Such a protocol would reduce or eliminate the need for STUN and
> friends, and it would provide a single protocol for the vendors to implement
> instead of a running treadmill of RFCs.

Oh yes, no doubts there. midcom has been trying for years, and still no 
success. Even then, once its done, it doesn't work for many topoligies. 
The ultimate right answer is nsis, which would allow the client to 
directly ask for bindings through all nats that it will need to talk to. 
But, we are years and years away from such a thing seeing deployment.

> 
> But the vendors have already been given a single protocol to implement:
> UPnP.  It was created from the desire for NAT control at the client and
> appears to have a lot of support from NAT vendors already.  And this makes
> me wonder why we are continuing to propose more NAT traversal solutions. 
> Why not either recommend UPnP or design an equivalent?

I indicated in another mail several problems with upnp. Another 
practical one is that its not that widely deployed yet. So, when an 
media company wants to deploy, do they wait and hope that people go off 
and buy upgrades to their nat, or hope all the nat vendors fall in line? 
No way. They want to be able to deploy the service now, without being 
dependent on features placed into home routers.

Anyway, ietf is working on their version of upnp, which is the midcom 
protocol.

>>Certainly. This is called "symmetric NAT". Its common but to date the 
>>full-cone variety, which maps bindings by private IP/port alone, is 
>>quite prevalent amongst the home NATs.
> 
> 
> Interesting.  Where can I find out more about how many home networks use
> which types of NAT?

I've heard reports of studies done on the subject, but have not seen 
concrete or quotable reports. I think such an analysis, especially 
tracking it over time, would be a great masters project for some student 
somewhere.

-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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--zhXaljGHf11kAtnf
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> >>>This draft again assumes that RTP is the only transport choice:
> >>>
> >>>rtcp-attribute =3D  "a=3Drtcp:" port
> >>>
> >>>What if there were some other UDP based transport that required two UDP
> >>>ports (perhaps even "RTPng")?  You would need yet another draft to=20
> >>>address
> >>>it.  Why not make the solution generic instead?
> >>
> >>Well, if thats so, I would have rather had such a debate in the context=
=20
> >>of the above draft. Anyway, I know of no other media stream that uses=
=20
> >>UDP that uses multiple ports AND that has an algorithmic relationship=
=20
> >>between those ports (i.e., has no way to express in the SDP which ports=
=20
> >>are needed). Seems like a non-problem.
> >
> >
> >Prior to IPv6 there was no known way to express a numeric IP address such
> >that it would not fit into a 32-bit integer.  Now there is, and it is one
> >source of poring issues.  Of course this is not a problem now.  Can you =
say
> >with certainty that it will never be a problem?
>=20
> Seems like a bit of an over-exaggerated analogy. As I said, for any new=
=20
> protocol that needs multiple ports, it should simply make sure that its=
=20
> SDP parameter definitions include them. In such a case you will never=20
> have a problem.

How is it better to require more work down the road (someone has to write a
draft, etc.) than build an extensible specification?  It may not be likely
that anyone will need the functionality, but it really does not require any
more work to provide up front and you may reduce the work later on.

What reason is there for not making an extensible solution (besides "it is
unlilkely anyone would ever use it")?

> Anyway, the point is moot, since, based on my limited RTSP knowledge,=20
> SDP is not used in the client addresses anyway. Thus, you can't use the=
=20
> SDP draft above.

You were the one who mentioned it.  ;-)  It is analogous to the proposed
rtcp_client_port attribute in the RTSP Transport: header.

> >>A smart ALG won't. It should know not to muck with IP addresses inside=
=20
> >>of the RTSP message if they are not private addresses (if stun is used,=
=20
> >>those addresses will have already been changed to public ones by the=20
> >>RTSP client). Indeed, one can imagine usign RTSP to setup a streaming=
=20
> >>media session to some third party device outside of the nat, in which=
=20
> >>case the ALG has to know not to change the IP. I wonder if the linux=20
> >>code you cite supports that.
> >
> >
> >All of the four or so RTSP modules that I am familiar with simply look at
> >client_port=3D and perform a mapping.  In order to detect that a transpo=
rt
> >spec is using STUN, the module would also need to check for a destinatio=
n=3D
> >attribute.
>=20
> Right. Although, I must admit confusion. If the client doesnt provide a=
=20
> destination attribute, to what IP address is the media sent? It seems to=
=20
> me that the destination parameter would normally be there. Or perhaps=20
> you are saying it is there, but its not looked at by the ALG?

In the absence of a destination=3D attribute (which is far and away the most
common case as you may have inferred), the server sends to the same address
as the foreign end of the RTSP connection.  Disallowing the destination=3D
parameter pretty much eliminates the possibility for DoS abuse (hey server,
go flood this IP address with UDP data).  The possibility is more than idle
speculation -- a very similar DoS issue for online gaming servers was
recently discovered and published online (I forget which site but I can
probably find the article again if desired).  A few fat video stream makes
online game servers' response packets look like child's play.

--=20
There is nothing new under the sun, but there are lots of old things we
don't know yet.
        -- Ambrose Bierce

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


	Title		: The Alternative Semantics for the Session
                          Description Protocol Grouping Framework
	Author(s)	: G. Camarillo, J. Rosenberg
	Filename	: draft-camarillo-mmusic-alt-00.txt
	Pages		: 8
	Date		: 2003-2-3
	
This document defines the alternative (ALT) semantics for the SDP
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Hi,

I would like to try to summarize and give my comments to Tom's and
Jonathan's discussion. These are issues to continue to discuss here on
the list but which will also be brought up on tomorrow's RTSP
teleconference.

1. We might break out the NAT traversal part into a separate
specification. However we must look at how it relates to mechanisms that
will be in the base spec. I think it is important that as soon as
possible we outline these solutions.

2. The use of "client_rtcp_port" versus changing "client_port". Your are
correct that it might not be necessary to call it RTCP port. What we
will need for STUN support is to possibility to specify two explicit
port numbers without relation to each other. The general case would then
to allow to specify any set of port numbers in both directions. This
will cover any future transport protocol. However I went on to a new
parameter to avoid changing the already present parameters syntax. If
people find this acceptable then we can basically do anything with
"client_port".

3. Symmetric RTP.
- STUN over cone nats work without changing the servers, where symmetric
RTP requires server changes.
- Symmetric has hi-jacking potentials that will impossible to protect
against without RTP/RTCP based ID/authentication scheme.
- Needs binding mechanism for RTP/RTCP packets to the correct RTSP Session.
- Potential RTSP middle box (proxies, caches) problems. Should any way
be fixed for future RECORD servers.
- Otherwise of course a nice solution.

4. ALG's
- ALG: Deployment issues, especially for extensions
- ALG: Sub-standard implementations
- Prevent end-to-end security.
- Interactions between ALG's STUN is a potential problem.

5. UPnP or MIDCOM.
- Time to deploy these solutions.
- no real conflict, if UPnP/midcom is available use it.

6. Jonathan's other proposal
Please provide it so that we can discuss it.

7. Use of "Destination" parameter for NAT/ALG's
- Destination might be needed if NAT maps using several IP addresses to
same client.
- Needs client based authentication of that address/port in RTSP message
to work in secure 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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From mailnull@www1.ietf.org  Tue Feb  4 07:10:39 2003
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
From: David Yon <yon.ietf@rfdsoftware.com>
Subject: Re: [MMUSIC] comedia-fix and the whole source address/port 
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At 10:54 PM 2/3/2003, Jonathan Rosenberg wrote:
>inline.
>
>David Yon wrote:
>>At 09:02 PM 1/14/2003, Jonathan Rosenberg wrote:
>>>
>>>
>>>Since, practically speaking, I would expect the majority of users to 
>>>connect to things from behind a NAT, this mechanism serves little 
>>>purpose to prevent DOS attacks. I don't understand why you would not use 
>>>more tried-and-true mechanisms, such as return routability, which 
>>>validate a source IP address by sending it a cookie, and getting this 
>>>cookie back. Such a mechanism actually works through NAT, and you can 
>>>provide much more than just service prioritiziation based on it.
>>
>>Again, the scheme as described works in the presence of NAT (despite 
>>assertions to the contrary), and allows the media to be 100% TLS without 
>>any out-of-spec "enhancements".
>
>You simply dismiss my argument, but do not address it.
>
>My argument is, that when most users are behind NAT, the source IP of the 
>TCP connection won't match that in the SDP, and the net result is that 
>most if not all connections go in the low priority pool. This is 
>equivalent to no priority. I have also asserted that there are more 
>reliable, well-known techniques for DoS protection against these kinds of 
>attacks, which work better.
>

Well, one more try and then I'll agree to leave the topic where it is.

The piece of the puzzle you may be missing is that our system has the 
luxury of "owning" the front-line SIP proxies, and those proxies are 
outside of any client NAT boxes.  They can determine whether NATting is 
going on, and what the translation is FOR THE IP ADDRESS.  So if an INVITE 
is coming from an endpoint who's claiming a source address of 
192.168.0.128, but the INVITE message is sourced from the address 
216.153.163.173, it:

         (a) knows there's a NAT
         (b) knows with a high degree of certainty that the media
             will also appear to be sourced from 216.153.163.173.

It therefore rewrites the source address from 192.168.0.128 to 
216.153.163.173 for the downstream proxies (just like some poor tech at 
Netgear would have to do in their ALG :-)).  The vast majority of NAT's use 
a single public address, so this translation works.  Really.  We've seen 
it.  Existence proof hard at work.



>>>I still maintain that, given we are painfully aware of the trials and 
>>>tribulations of distributing source IP addresses in protocols, and given 
>>>the vast potential for misuse, and the potentially small (if not ZERO) 
>>>deployable benefits of this mechanism, I would like to see it removed. 
>>>It is one thing to include a mechanism that is useful in many cases, but 
>>>might not work in some (and then documenting such) as compared to a 
>>>mechanism that will most likely be misused, and whose real benefit is 
>>>small or zero.
>>
>>It is OPTIONAL in the spec, and the limitations clearly outlined.  At 
>>least one implementation finds it useful.  I don't know how to be more 
>>clear than that, and why it causes such heartburn.
>
>Because its unsafe. You have put a piece of information there, for which 
>there is NO WAY to ever know whether it is reliable. Its not that it might 
>be wrong, you don't know when its wrong and when its not. If I am building 
>a proxy that wishes to look at the source IP in the SDP, and configure a 
>pinhole in a firewall, instead of a cone, what advice can you give me 
>about when I can use this source IP field? Your only answer is, you can 
>only use it when you know its right because you control every piece of 
>equipment on the network (the clients and all intervening nats). How can I 
>be sure that this is the case? Someone all of a sudden installs a linksys 
>and everything stops working, because the pinholes are not opened 
>properly. That would be a fun customer call to get at 2 in the morning.

While there's room for argument here, I'll just leave this for now.


>Anyway, this has remained a largely one-on-one discussion. I think all 
>that can be said has been said, on my part anyway. I would like to hear 
>from our esteemed chairs on whether this feature stays or goes.

Yes, indeed, I would also like to hear from the chairs on this.


>-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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Hi,

In this revision of the Transport independent SDP bandwidth modifiers 
the following changes has been performed:

- Removed "avgprate" as it is not useful.
- Removed the RTCP rules present in the last proposal and replaced them 
with simpler ones. If 5% RTCP bandwidht that shall be based on the local 
transport dependent rate.
- Clarified the text.
- Brought the text up to ID-nits level, ready for publication.

As I think this document is ready and address the comments that I have 
received, I would like to ask for a WG last call on 
draft-ietf-mmusic-sdp-bwparam-00.txt.

Best Regards

Magnus Westerlund




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Subject: Re: [MMUSIC] RTSP and NATs
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: IETF MMUSIC WG <mmusic@ietf.org>
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Magnus,

I strongly support the recommendation to not use ALGs,
therefore STUN should be used instead but then you wrote:


 > To be able to use STUN to traverse symmetric NATs the  STUN  server
 >  needs to be co-located with the streaming server media distribution
 >  ports. As this will create implementations difficulties and  possi-
 > bly security problems this SHOULD NOT be done.

I am surprised, I would have proposed _on the contrary_ a "SHOULD" here
(since traversing symetric NATs is a key feature) !

also what do you mean by  "implementation difficulties" ? (I really cannot see any ?)
and could you explicit what type of _additional_ security issues this would cause
(i.e. issues that are not inherent to running either a RTSP or a STUN server in the first place ?)


regards


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


                                                                                                                                              
                                                                                                                                              
                                                          To:   IETF MMUSIC WG <mmusic@ietf.org>                                              
                                                          cc:   (bcc: Philippe Gentric/MP4-SUR/CE/PHILIPS)                                    
                                                          Subject:    [MMUSIC] RTSP and NATs                                                  
                                                                                                                                              
               Magnus Westerlund                          Classification:                                                                     
               <magnus.westerlund@era.ericsson                                                                                                
               .se>                                                                                                                           
                                                                                                                                              
               Sent by:                                                                                                                       
               mmusic-admin@ietf.org                                                                                                          
                                                                                                                                              
               03/02/2003 17:26                                                                                                               
                                                                                                                                              
                                                                                                                                              




Hi,

Here is a proposal to the RTSP spec for how to handle NATs using STUN.
Please read and comment. It is planned to discuss NAT's on wednessday's
RTSP telecon.

Cheers

Magnus

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

15 RTSP through Firewalls & NATs

15.1 Introduction

   Today there is Network Address Translators (NAT)   [32]  everywhere
   and a protocol must make sure that it can work through them in some
   fashion. The problem with RTSP is that it carries information about
   network  addresses  and  ports  inside  itself. It is primarily the
   media streams that are being blocked by NAT, as the  given  address
   to send to is a local one.

   Firewalls are also middle boxes that needs to be  considered.  They
   are deployed to prevent non-desired traffic/protocols to be used or
   reach the protected network. RTSP is designed to allow  a  firewall
   to  let RTSP controlled streams through, if chosen to, with minimal
   implementation problems.

15.2 NAT

   Today there exist a number of different NAT types and usage  areas.
   The different NAT types, cited from STUN  [31]:

   Full Cone: A full cone NAT is one where all requests from the  same
        internal  IP  address and port are mapped to the same external
        IP address and port. Furthermore, any external host can send a
        packet to the internal host, by sending a packet to the mapped
        external address.

   Restricted Cone: A restricted cone NAT is one  where  all  requests
        from  the  same internal IP address and port are mapped to the
        same external IP address and port. Unlike a full cone NAT,  an
        external  host  (with  IP  address X) can send a packet to the
        internal host only if the internal host had previously sent  a
        packet to IP address X.

   Port Restricted Cone: A port restricted cone NAT  is  like  a  res-
        tricted  cone  NAT, but the restriction includes port numbers.
        Specifically, an external host can send a packet, with  source
        IP  address  X and source port P, to the internal host only if
        the internal host had previously sent a packet to IP address X
        and port P.

   Symmetric: A symmetric NAT is one where all requests from the  same
        internal  IP  address  and  port, to a specific destination IP
        address and port, are mapped to the same external  IP  address
        and port. If the same host sends a packet with the same source
        address and port, but to a different destination, a  different
        mapping  is  used.  Furthermore,  only  the external host that
        receives a packet can send a UDP packet back to  the  internal
        host.

   NAT's are used on both small and  large  scale.  The  normal  small
   scale  user is home user that has a NAT to allow multiple computers
   share the single IP address given by their  Internet  Service  Pro-
   vider  (ISP).  The  large scale users are the ISP's themselves that
   gives there users private addresses. This is done both for  control
   and lack of addresses.

15.2.1 STUN

   The STUN  [31] protocol allows a client to  discover  the  type  of
   NAT(s) he is behind and also discover a mappings public address and
   port number. The protocol also allows  discovery  of  the  mappings
   timeout period and can be used as keep alive mechanism.

   How useful this protocol is depends  on  the  type  of  NAT(s)  the
   client  is  behind.  If  the  user  is behind a full cone NAT, STUN
   allows the RTSP client to traverse the NAT with some simple  client
   side adaptations. For restricted cone NATs STUN is still useful but
   require some more adaptations. For  symmetric  NATs  STUN  requires
   such severe server adaptations that it is not practical to use.

   A RTSP client using RTP transport can use STUN to traverse  a  full
   cone NAT in the following way:

   1.   Use STUN to discover the type of NAT, if any, and the  timeout
        period  for any UDP mapping on the NAT. This is RECOMMENDED to
        be performed in the background as soon as IP  connectivity  is
        established.  If  this  is  performed  prior to the attempt to
        establish a streaming session the possible delays in the  ses-
        sion  establishment will be reduced. If no NAT is present, use
        the normal SETUP behavior.

   2.   The RTSP client determines the number of UDP ports  needed  by
        the  RTP  sessions  part of the multi-media presentation. This
        information is available in  the  media  description  protocol
        used,  e.g.  SDP. In general each RTP session will require two
        UDP ports, one for RTP, and one for RTCP. Ensure that the same
        public  IP  address  is used for each RTP/RTCP port pair esta-
        blished.

   3.   For each UDP port required, establish a mapping  and  discover
        the  public  IP  address  and port number with use of STUN. if
        successful this results in that the client now knows for  each
        port know the mapping: clients local address/port Leftrightar-
        row public address/port.

   4.   Perform the RTSP SETUP for each media. In the transport header
        the  following  parameters  SHOULD  be included with the given
        values:   "destination"   with   the   public   IP    address,
        "client_port"  with  the public port of the mapping determined
        to be used for RTP, "client_rtcp_port" with the port number of
        the   mapping   to   be   used   for   RTCP.   The   parameter
        "client_rtcp_port" needs to be  used  unless  the  client  has
        managed  to  establish  a mapping with two consecutive numbers
        starting with an even one.  To allow this to work servers MUST
        allow  a  client to setup the RTP stream on any port, not only
        even ports. The server SHALL respond with a  transport  header
        containing the source IP address of the media streams.

   5.   To keep the mappings alive the client SHOULD periodically send
        UDP traffic over all mappings needed for the session. STUN can
        be used to determine the timeout period of the NAT(s) UDP map-
        pings. For the mapping carrying RTCP traffic the periodic RTCP
        traffic may be sent frequently enough. If not or for RTP  car-
        rying  mappings,  empty  IP/UDP messages SHOULD be sent to the
        streaming servers discard port (port number 9).

   If a UDP mapping is lost above process is required to be  performed
   again and the media stream effected SETUP again.

   To allow the UDP packets to arrive from  the  server  to  a  client
   behind  a  restricted  NAT,  any  UDP  messages must be sent to the
   server. Therefore SHOULD the client, before  sending  a  RTSP  PLAY
   request  send  an  empty  UDP  message,  on each mapping, to the IP
   address given as the servers source address and  its  discard  port
   (port number 9). Otherwise no difference in procedure compared with
   a full cone NAT is needed.

   For a port restricted NAT the client  must  send  messages  to  the
   exact ports used by the server to send messages. This makes it pos-
   sible to use the above described process with the  following  addi-
   tions: For each port mapping, a UDP message needs to be sent to the
   servers source address/port. To minimize potential effects  on  the
   server  from  these messages the following type of messages MUST be
   sent. RTP: An empty UDP message with  out  any  payload.   RTCP:  A
   correctly  formed RTCP message. Unless enough bandwidth is assigned
   to RTCP it might not be possible to  keep  the  UDP  mapping  open.
   These  messages  SHOULD  be  sent before sending a PLAY request and
   then periodically according to the determined timeout time  of  the
   NAT.

   To be able to use STUN to traverse symmetric NATs the  STUN  server
   needs to be co-located with the streaming server media distribution
   ports. As this will create implementations difficulties and  possi-
   bly security problems this SHOULD NOT be done.

15.2.2 Application Level Gateways

   An Application Level Gateway (ALG) reads the application level mes-
   sages  and  perform necessary changes to allow the protocol to work
   through the middle box. However this behavior has some problems  in
   regards to RTSP:

   1.   Does not work when protocol is used with end-to-end  security.
        As the ALG can't inspect and change the application level mes-
        sages the protocol will fail over the middle box.

   2.   Needs to be updated if extensions to the protocol  are  added.
        Due  to  deployment  issues  with changing ALG's this may also
        break the end-to-end functionality of RTSP.

   Due to the above reasons it is RECOMMENDED to not use ALGs in NATs.
   This  is  especially important for NAT's targeted to home users and
   not ISPs. In that case it is recommended that NATs  are  full  cone
   and  that  the  RTSP client instead used STUN or the planned MIDCOM
   protocols.

   An ALG for the this specification would need to perform the follow-
   ing tasks and changes to RTSP:

   1.   Detect a SETUP request.  Create  UDP  mappings  (client  given
        IP/port  Leftrightarrow  public  IP/port) where needed for all
        possible transport specification in the  transport  header  of
        the request found in (1). Enter the public address and port(s)
        of these mappings  in  transport  header.  Mappings  SHALL  be
        created with consecutive public port number starting on a even
        number for RTP each stream.  Mappings SHOULD also be  given  a
        long timeout period, at least 5 minutes.

   2.   When the response is received from  the  server  the  ALG  MAY
        remove  the  unused UDP mappings, i.e. the ones not present in
        the transport header. The session ID SHOULD also be  bound  to
        the UDP mappings part of that session.

   3.   The ALG SHOULD keep alive the UDP mappings belonging to the an
        RTSP  session  as long as: RTSP messages with the session's ID
        has been sent in the last RTSP session  timeout  interval,  or
        UDP  messages  are  sent on any of the UDP mappings during the
        last RTSP timeout interval.

   4.   The ALG MAY remove a mapping as soon a TEARDOWN  response  has
        been received for that media stream.

15.2.3 TCP Tunneling

   TCP tunneling is the final fallback  to  get  any  RTSP  controlled
   streaming to work through a NAT. As it is the client that establish
   the TCP connection to the server it  will  automatically  create  a
   binding  through  the  NAT. This connection can then be used by the
   server to send media packets in.

   Using TCP tunneling has the drawback on real-time  properties  that
   TCP has with "head of line" blocking. It may also have large varia-
   tions in through put due to the congestion control algorithm.   TCP
   tunneling must also be supported by both server and client.

   See section  10.11 for how to perform TCP tunneling.

15.3 Firewalls

   Firewalls exist for a purpose to protect a network from traffic not
   desired by the firewall owner. Therefore it is a policy decision if
   a firewall will let RTSP and its media streams through or not. RTSP
   is  designed  to  be  as easy as possible to process for a firewall
   with a policy to let the traffic pass through.

   The firewall will need to allow the media streams associated with a
   RTSP  session  pass through it. Therefore the firewall will need an
   ALG that reads RTSP SETUP and TEARDOWN  messages.  By  reading  the
   SETUP message the firewall can determine what type of transport and
   from where the media streams will use. Very common will be the need
   to open UDP ports for RTP/RTCP. By looking at the source and desti-
   nation addresses and ports the  opening  in  the  firewall  can  be
   minimized  to  the least necessary. The opening in the firewall can
   be closed after a teardown message for that session or the  session
   itself timesout.

--

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] RTSP and NATs
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Hi Philippe,

philippe.gentric@philips.com wrote:

>Magnus,
>
>I strongly support the recommendation to not use ALGs,
>therefore STUN should be used instead but then you wrote:
>
>
> > To be able to use STUN to traverse symmetric NATs the  STUN  server
> >  needs to be co-located with the streaming server media distribution
> >  ports. As this will create implementations difficulties and  possi-
> > bly security problems this SHOULD NOT be done.
>
>I am surprised, I would have proposed _on the contrary_ a "SHOULD" here
>(since traversing symetric NATs is a key feature) !
>  
>
>also what do you mean by  "implementation difficulties" ? (I really cannot see any ?)
>and could you explicit what type of _additional_ security issues this would cause
>(i.e. issues that are not inherent to running either a RTSP or a STUN server in the first place ?)
>  
>
The problem of running STUN for a symmetric NAT is that the STUN server 
must be located at the servers sending port. So using the same RTSP 
mechanism that are used for traversing a cone-nat the client would:
1. first contact a well known port with the server for each his media 
stream to get the mapping of the stream. The client can't use any other 
then the well known port because it doesn't know what port to send to.
2. In the SETUP ask the server to send from its STUN servers well known 
port. It also needs to receive RTCP on that port.
3. For keep alive on the RTP port the client needs to send periodically 
STUN messages to the STUN server.

This is a mess for a implementor. It need to have a STUN server 
receiving the STUN messages while the RTP/RTCP stack should receive all 
other messages. Also all the clients need to reside on the same port 
number which creates a multiplexing nightmare.

If you change the setup phase so that the streams are first SETUP to 
dummy ports then reconfigured with a later message, the STUN could be 
located at each media streams source address. However in this setup it 
is actually better to use symmetric RTP in all regards except perhaps 
hi-jacking of media streams. However with certain restrictions you can 
actually end up with at least the same level of security that RTSP 
provides today.

How are you using STUN to traverse a symmetric NAT?

Best Regards

Magnus
 

-- 

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] Deleted Streams in Grouping
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Hi,
      Suppose there exists a group which groups two media streams m1 and
m2.
If a stream gets deleted in the middle of the session(say m1), then do we
have to remove
the "Media Stream Identification tag"(mid) from the grouping??. The RFC
says we will
have to remove the mid's for rejected streams. There is no mention about
deleted streams.

If we dont do this then there might be a problem when we try to add a new
stream(say m3).

Assume that we have deleted m1 from the session and the *mid* is not
removed from
grouping. When we try to add a third stream m3 to the session, the slot of
the deleted stream
gets reused. Also the Media Stream Identification tag(mid) remains the same
.
This will cause this stream to be part of the grouping, which is not TRUE.

Please clarify.

Thanks and regds,
srikanth sundaragopalan
Hughes Software Systems.


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From: Gonzalo Camarillo <Gonzalo.Camarillo@lmf.ericsson.se>
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Hello,

in the offer/answer model there is not such a thing as a "deleted"
stream. You never delete m lines. You set the port number to zero, which
is reffered to as "rejecting" a media stream.

Therefore, if you have a session that lasts very long, and you have been
adding and "rejecting" streams, your SDPs will carry a lot of garbage...
ungly, but that's the way it is.

Regards,

Gonzalo

ssundaragopalan@hss.hns.com wrote:
> 
> Hi,
>       Suppose there exists a group which groups two media streams m1 and
> m2.
> If a stream gets deleted in the middle of the session(say m1), then do we
> have to remove
> the "Media Stream Identification tag"(mid) from the grouping??. The RFC
> says we will
> have to remove the mid's for rejected streams. There is no mention about
> deleted streams.
> 
> If we dont do this then there might be a problem when we try to add a new
> stream(say m3).
> 
> Assume that we have deleted m1 from the session and the *mid* is not
> removed from
> grouping. When we try to add a third stream m3 to the session, the slot of
> the deleted stream
> gets reused. Also the Media Stream Identification tag(mid) remains the same
> .
> This will cause this stream to be part of the grouping, which is not TRUE.
> 
> Please clarify.
> 
> Thanks and regds,
> srikanth sundaragopalan
> Hughes Software Systems.

-- 
Gonzalo Camarillo         Phone :  +358  9 299 33 71
Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
Telecom R&D               Fax   :  +358  9 299 30 52
FIN-02420 Jorvas          Email :  Gonzalo.Camarillo@ericsson.com
Finland                   http://www.hut.fi/~gonzalo
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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 SDP and RTSP
	Author(s)	: J. Arkko et al.
	Filename	: draft-ietf-mmusic-kmgmt-ext-06.txt
	Pages		: 12
	Date		: 2003-2-6
	
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.

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From: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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Subject: [MMUSIC] RTSP transport header, port parameter changes
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Hi,

At the wednesdays telecon we discussed on how to update the transport 
header to support STUN and symmetric RTP
for traversal of NATs. It was decided to allow the ports to be specified 
explicitly for RTP and RTCP. Therefore I have written up a proposal for 
new text for the transport header below:

We where discussing making the port attributes totally flexible. However 
as I started looking at the section I discovered that port, client_port, 
server_port are RTP-specific. So my proposed changes does therefore only 
solves what is necessary for RTP. An alternative to this change would be 
to reclassify the port attributes to be general attributes and then they 
should have total flexibility. That would also require some text on how 
to related to the RTP media transport setup.

I also included a client_ssrc parameter. This would be used for 
symmetric RTP to make the symmetric port opening more robust. I will try 
to write up a mail regarding using symmetric RTP for RTSP sessions.

Please read and comment

Magnus

--------------------------------------------------------
changed parts of section 12.40 Transport header in relation to 
draft-ietf-mmusic-rfc2326bis-02.txt.

RTP-specific:

   These parameters are only valid if the media transport protocol  is
   RTP.  The parameters containing port numbers are all subject to the
   following consideration:  All port numbers where prior specified as
   a value or range, see "old-port-spec" in ABNF below. To support NAT
   traversal RTP has been updated to allow non-continuous port specif-
   ication  not  following  the even/odd rule. As long as both the RTP
   and RTCP port is specified explicitly they are allow  to  take  any
   valid  port  number. Therefore the RTP specific port parameters has
   been updated to support this. The updated format  ("new-port-spec")
   SHOULD  only  be  used when required to specify non-continuous port
   numbers. Before using the "new-port-spec" format  the  client  MUST
   determine server support of the "play.basic" option tags.

   port: This parameter provides the RTP/RTCP port pair for  a  multi-
        cast  session.  This  parameter  SHOULD NOT use the "new-port-
        spec" format. It is should be  specified  as  a  range,  e.g.,
        port=3456-3457

   client_port: This parameter provides the unicast RTP/RTCP port pair
        on  the  client where media data and control information is to
        be sent. It is specified either as a range,  e.g.,  port=3456-
        3457 or as two explicit numbers port=3457/4976

   server_port: This parameter provides the unicast RTP/RTCP port pair
        on  the  server where media data and control information is to
        be sent. It is specified either as a range,  e.g.,  port=3456-
        3457 or as two explicit numbers port=3457/4976

   ssrc: The ssrc parameter indicates the RTP  SSRC  [23]  value  that
        should  be  (request)  or will be (response) used by the media
        server. This parameter is only valid for unicast transmission.
        It identifies the synchronization source to be associated with
        the media stream, and is expressed as an eight  digit  hexide-
        cimal value. In cases that a sender will use multiple SSRCs it
        SHOULD NOT use this parameter.

   client_ssrc: The client_ssrc parameter indicates the RTP SSRC  [23]
        value  that will be used by the client. This parameter is only
        valid for unicast transmission. It identifies the synchroniza-
        tion  source  to  be  associated with the media stream, and is
        expressed as an eight digit hexidecimal value. In cases that a
        client  will use multiple SSRCs it SHOULD NOT use this parame-
        ter.


   Transport                =  "Transport" ":" 1#transport-spec
   transport-spec           =  transport-id *parameter
   transport-id             =  transport-protocol "/" profile ["/" 
lower-transport]
                               ; no LWS is allowed inside transport-id
   transport-protocol       =  "RTP" / token
   profile                  =  "AVP" / token
   lower-transport          =  "TCP" / "UDP" / token
   parameter                =  ";" ( "unicast" / "multicast" )
                           /   ";" "source" "=" address
                           /   ";" "destination" [ "=" address ]
                           /   ";" "interleaved" "=" channel [ "-" channel ]
                           /   ";" "append"
                           /   ";" "ttl" "=" ttl
                           /   ";" "layers" "=" 1*DIGIT
                           /   ";" "port" "=" port-spec
                           /   ";" "client_port" "=" port-spec
                           /   ";" "server_port" "=" port-spec
                           /   ";" "ssrc" "=" ssrc
                           /   ";" "client_ssrc" "=" ssrc
                           /   ";" "mode" "=" mode-spec
                           /   ";" trn-parameter-extension
   port-spec                =  old-port-spec / new-port-spec
   old-port-spec            =  port [ "-" port ]
   new-port-spec            =  port "/" port ; RTP/RTCP port
   trn-parameter-extension  =  par-name "=" trn-par-value
   par-name                 =  token
   trn-par-value            =  *unreserved
   ttl                      =  1*3(DIGIT)
   port                     =  1*5(DIGIT)
   ssrc                     =  8*8(HEX)
   channel                  =  1*3(DIGIT)

   address                  =  host ;As defined in RFC 2732 [30]
   mode-spec                =  <"> 1#mode <"> / mode


-- 

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>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP transport header, port parameter changes
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> We where discussing making the port attributes totally flexible. However=
=20
> as I started looking at the section I discovered that port, client_port,=
=20
> server_port are RTP-specific.

Yes, 2326 does say that client_port and server_port are for RTP.  I would
like to propose that the wording be changed, as client_port is not specific
to RTP.  The RealNetworks client and server have been using the client_port
attribute with non-RTP transports since 2326 was written.

--=20
I used to think I was a child; now I think I am an adult -- not because
I no longer do childish things, but because those I call adults are no
more mature than I am.

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Subject: [MMUSIC] Range Header In Play Responses
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Hi:

During the Jan 22, 2003 RTSP telecon, we discussed Range header in PLAY
reponses in the context of option tag play.basic.
In current draft (draft-ietf-mmusic-rfc2326bis-02.txt), PLAY Range response
is required for on demand media (see section 10.5).
It was decided to make this an requirement for live streaming as well. 

Here is the proposed changes to section 10.5 and section 14 (examples) in
draft-ietf-mmusic-rfc2326bis-02.txt. Each change is separated by "-----"'s,
and instructions on how to make the changes are put in square brackets.

Your comments are always welcome.

Thanks
thomas

---------------------------------
[insert the following to Section 10.5, before the paragraph started with
"For a on-demand stream, the server MUST reply with the actual range...]

[new paragraph] Server MUST include a "Range" header in its Play response.
When playing live media, the Play response MUST use one, and only one, of
the two
Range headers below:

Range: now-

- or -

Range: 0-

Here in live mode "now" happens to mean "0"  in NPT time space for this
session.

----------------------------------------------------------------------------
--------------------------------------
[change section 14.2, add the following right before the begining of actual
RTSP DESCRIBE request]

In this example we assume the length of the presentation is 600 seconds.

[Also in section 14.2, Add a range header to PLAY response]

Range: 0-600


-------------------------------------------------------
[change section 14.3, add the following right before the begining of actual
RTSP DESCRIBE request]

In this example we assume the length of the presentation is 600 seconds.

[change the example RTSP conversation in section 14.3 to add the following
"Range" header to PLAY reponse]

Range: 0-600

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

------------------------------
[change section 14.4 Live Media example, add the following "Range" header to
PLAY reponse]

Range: now-

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



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From mailnull@www1.ietf.org  Sun Feb  9 17:48:56 2003
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To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
cc: mmusic@ietf.org
Subject: Re: [MMUSIC] I-D ACTION:draft-ietf-mmusic-sdp-bwparam-00.txt 
In-Reply-To: Your message of "Tue, 04 Feb 2003 13:24:01 +0100."
             <3E3FB0E1.5060402@era.ericsson.se> 
Date: Sun, 09 Feb 2003 17:47:47 -0500
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Magnus,

--> Magnus Westerlund writes:
>In this revision of the Transport independent SDP bandwidth modifiers 
>the following changes has been performed:
>
>- Removed "avgprate" as it is not useful.
>- Removed the RTCP rules present in the last proposal and replaced them 
>  with simpler ones. If 5% RTCP bandwidht that shall be based on the local 
>  transport dependent rate.
>- Clarified the text.
>- Brought the text up to ID-nits level, ready for publication.
>
>As I think this document is ready and address the comments that I have 
>received, I would like to ask for a WG last call on 
>draft-ietf-mmusic-sdp-bwparam-00.txt.

Does the TIAS bandwidth include the RTP payload header and/or any RTP
header extensions? What about profile specific additions, such as the
SRTP trailer?

As an editorial note, also, could you move the "Changes" section to be 
a final appendix, so that the RFC editor does not have to renumber the
draft?

Colin
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Hi Tom,

I am willing to make the client_port parameters to general parameters. 
It will make their BNF slightly more complicated as they should allow 
for any number of ports, both as ranges and lists. This change will also 
require us to write some text on usage of RTP with general parameter.

Best Regards

Magnus

Tom Marshall wrote:

>>We where discussing making the port attributes totally flexible.
>>    
>>
>However 
>  
>
>>as I started looking at the section I discovered that port,
>>    
>>
>client_port, 
>  
>
>>server_port are RTP-specific.
>>    
>>
>
>Yes, 2326 does say that client_port and server_port are for RTP.  I
>would
>like to propose that the wording be changed, as client_port is not
>specific
>to RTP.  The RealNetworks client and server have been using the
>client_port
>attribute with non-RTP transports since 2326 was written.
>
>  
>

-- 

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] Range Header In Play Responses
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Hi Thomas,

Some comments below.

Thomas Zeng wrote:

>Hi:
>
>During the Jan 22, 2003 RTSP telecon, we discussed Range header in PLAY
>reponses in the context of option tag play.basic.
>In current draft (draft-ietf-mmusic-rfc2326bis-02.txt), PLAY Range response
>is required for on demand media (see section 10.5).
>It was decided to make this an requirement for live streaming as well. 
>
>Here is the proposed changes to section 10.5 and section 14 (examples) in
>draft-ietf-mmusic-rfc2326bis-02.txt. Each change is separated by "-----"'s,
>and instructions on how to make the changes are put in square brackets.
>
>Your comments are always welcome.
>
>Thanks
>thomas
>
>---------------------------------
>[insert the following to Section 10.5, before the paragraph started with
>"For a on-demand stream, the server MUST reply with the actual range...]
>
>[new paragraph] Server MUST include a "Range" header in its Play response.
>When playing live media, the Play response MUST use one, and only one, of
>the two
>Range headers below:
>
>Range: now-
>
>- or -
>
>Range: 0-
>
>Here in live mode "now" happens to mean "0"  in NPT time space for this
>session.
>  
>

Actually I think at least one more range type attribute is allowed. That 
is the use of clock. It is perfectly all right for a client to ask a 
server to play a live session using the following range header:
Range:clock=20030213T143205Z-

Which means that the server should play out media starting at 13 Feb. 
2003 at 14:32:05. If this stream is live stream the play out can't start 
before that time. There are also a requirement that range responses uses 
the same sort of units as in the request. Therefore also clock responses 
must be allowed. And for being future proof I think the above proposal 
in text needs to be reformulated so it will be possible to use those 
formats. Also what values should be given in SMTP request streams.

--- My proposal for text would be:
Server MUST include a "Range" header in any PLAY response. The response 
MUST use the same format as the request's range header contained. If no 
Range header was in the request, the NPT time format SHOULD be used 
unless the client showed support for other formats. For a session with 
live media streams the Range header MUST also be given, containing a 
valid time indication. It is RECOMMENDED that either "npt=now-" or a UTC 
clock time for the corresponding time is given, i.e. 
"clock=20030213T143205Z-". The UTC clock format SHOULD only be used if 
client has shown support for it.
------ End of proposal ------
 
The examples must all be updated when we have a stable specification. 
Especially the ones in chapter 14 are out of date.

Best Regards

Magnus

>----------------------------------------------------------------------------
>--------------------------------------
>[change section 14.2, add the following right before the begining of actual
>RTSP DESCRIBE request]
>
>In this example we assume the length of the presentation is 600 seconds.
>
>[Also in section 14.2, Add a range header to PLAY response]
>
>Range: 0-600
>
>
>-------------------------------------------------------
>[change section 14.3, add the following right before the begining of actual
>RTSP DESCRIBE request]
>
>In this example we assume the length of the presentation is 600 seconds.
>
>[change the example RTSP conversation in section 14.3 to add the following
>"Range" header to PLAY reponse]
>
>Range: 0-600
>
>------------------------------
>
>------------------------------
>[change section 14.4 Live Media example, add the following "Range" header to
>PLAY reponse]
>
>Range: now-
>
>------------------------------
>
>
>
>_______________________________________________
>mmusic mailing list
>mmusic@ietf.org
>https://www1.ietf.org/mailman/listinfo/mmusic
>
>  
>

-- 

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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Folks,

we have submitted the draft below. It defines a new "semantics"
attribute for the SDP grouping framework. These semantics are intended
to be useful for NAT traversal.

There will be another draft that describes the NAT traversal problem and
that references, among others, this draft.

Regards,

Gonzalo


-------- Original Message --------
Subject: [MMUSIC] I-D ACTION:draft-camarillo-mmusic-alt-00.txt
Date: Tue, 04 Feb 2003 06:44:47 -0500
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CC: mmusic@ietf.org

--NextPart

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


	Title		: The Alternative Semantics for the Session
                          Description Protocol Grouping Framework
	Author(s)	: G. Camarillo, J. Rosenberg
	Filename	: draft-camarillo-mmusic-alt-00.txt
	Pages		: 8
	Date		: 2003-2-3
	
This document defines the alternative (ALT) semantics for the SDP
grouping framework. The ALT semantics allow offering alternative
media configurations to establish a particular media stream.

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

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Subject: [MMUSIC] RTSP & NAT: Symmetric RTP
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Hi,

During the last RTSP telecon we discussed symmetric RTP as a traversal 
mechanism for NATs, especially symmetric ones.

Basic idea behind symmetric RTSP: A for each media stream the server 
ports are defined as both send and receive ports. The client sends RTP 
and RTCP on respective port. The server upon reception of the IP/UDP/RTP 
packet takes that packets sender port and address as clients ports. 
Thereby ignoring what has been said in the signaling.

RTSP updates needed: Server support is needed and they must be able to 
indicate this. Therefore a client request symmetric RTP behavior through 
the use of a new transport header parameter. There are also need for a 
feature tag to be able to detect this feature prior to setup.

Security risks: Hi-jacking of media streams. The basic problem is that 
any RTP/RTCP traffic arriving to the sender port prior to the clients 
message will result in redirection of the media stream to the sent 
packets address and port. This allows an attacker to spoof the sender 
IP/port number to that of a target of an Distributed DOS attack. This 
can basically be done without being on the path between client and 
server by simple guessing port numbers.

Good Solution: Create a RTP payload format that carries session 
identifier and a sufficient large random tag that prevents guessing. 
However due to NATs and the setup this can never be ensured against 
attackers located on the path between client and server.

Maybe adequate solution: Only allow RTP messages from same IP as control 
session arrives from. The SSRC must also be correct as signaled through 
an new client_SSRC parameter in Transport header. This would force an 
attacker to: 1) be behind the same NAT and receive the same IP from the 
NAT as the legitimate client. 2) Require the attacker to be able to read 
the RTSP control traffic. Guessing will be at one per 2^31 probability 
unless bad random values are used.

The maybe adequate solution does have the restriction that a NAT must 
ensure to use the same IP address for both the RTSP control binding and 
the media flows bindings.

Problems: The client does normally not send RTP traffic to server. May 
create problems with RTSP aware proxies. They should however be 
corrected to not interfere as this will also be a problem for RECORD 
streams. More important is that there is lack of mechanism to negotiate 
what RTP payloads that will be used in direction client->server. This 
will be a problem independent if we use a authentication payload or any 
other payload. To allow the adequate security to work we need at least a 
RTP header in the upstream traffic on the RTP channel carrying a SSRC. A 
default behavior of reusing a downstream RTP payload type with an empty 
payload could be used. The RTCP streams has no problems of this kind.

Conclusion:
Symmetric RTP can be used with a minimal change to allow servers to 
create bindings based on incoming RTP/RTCP packets. The stream will 
always be possible to hijack by a man in the middle on the path between 
server and client.

basic security: Use SSRC as secret identifier of client. Gives 1/2^31 
probability of guessing. Requires extra transport header parameter 
"client_ssrc".

basic+ security: Requires media flow bindings IP address to have same 
address as RTSP connection. Both requires at least default rule on how 
to send RTP packets to server.

best possible: Create special payload for identification and binding of 
symmetric RTP ports. Can use any size of random tag. Creates problem 
with what RTP payload type to use. Or could be sent on plain UDP.

This presents my view on how symmetric RTP could be used with RTSP 
sessions.

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>
CC: IETF MMUSIC WG <mmusic@ietf.org>
Subject: Re: [MMUSIC] RTSP & NAT: Symmetric RTP
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I agree that symmetric RTP is clearly the best solution for RTSP as it 
is used today. I do not believe that relying on source IP being the same 
  between the RTSP and RTP sessions is a good idea. This won't work in 
enterprises which may be natting into a multiplicity of IP addresses.

I would also strongly advise not to mandate that as the one and only one 
solution for nat traversal. A client should be allowed to do other 
things, and you want to leave the door open for better solutions down 
the road.

As for the random identifiers, I will note that I recently proposed such 
a mechanism in support of connection oriented media. This draft also 
discusses the attack you are talking about:

http://www.ietf.org/internet-drafts/draft-rosenberg-mmusic-comedia-fix-00.txt

-Jonathan R.

Magnus Westerlund wrote:
> Hi,
> 
> During the last RTSP telecon we discussed symmetric RTP as a traversal 
> mechanism for NATs, especially symmetric ones.
> 
> Basic idea behind symmetric RTSP: A for each media stream the server 
> ports are defined as both send and receive ports. The client sends RTP 
> and RTCP on respective port. The server upon reception of the IP/UDP/RTP 
> packet takes that packets sender port and address as clients ports. 
> Thereby ignoring what has been said in the signaling.
> 
> RTSP updates needed: Server support is needed and they must be able to 
> indicate this. Therefore a client request symmetric RTP behavior through 
> the use of a new transport header parameter. There are also need for a 
> feature tag to be able to detect this feature prior to setup.
> 
> Security risks: Hi-jacking of media streams. The basic problem is that 
> any RTP/RTCP traffic arriving to the sender port prior to the clients 
> message will result in redirection of the media stream to the sent 
> packets address and port. This allows an attacker to spoof the sender 
> IP/port number to that of a target of an Distributed DOS attack. This 
> can basically be done without being on the path between client and 
> server by simple guessing port numbers.
> 
> Good Solution: Create a RTP payload format that carries session 
> identifier and a sufficient large random tag that prevents guessing. 
> However due to NATs and the setup this can never be ensured against 
> attackers located on the path between client and server.
> 
> Maybe adequate solution: Only allow RTP messages from same IP as control 
> session arrives from. The SSRC must also be correct as signaled through 
> an new client_SSRC parameter in Transport header. This would force an 
> attacker to: 1) be behind the same NAT and receive the same IP from the 
> NAT as the legitimate client. 2) Require the attacker to be able to read 
> the RTSP control traffic. Guessing will be at one per 2^31 probability 
> unless bad random values are used.
> 
> The maybe adequate solution does have the restriction that a NAT must 
> ensure to use the same IP address for both the RTSP control binding and 
> the media flows bindings.
> 
> Problems: The client does normally not send RTP traffic to server. May 
> create problems with RTSP aware proxies. They should however be 
> corrected to not interfere as this will also be a problem for RECORD 
> streams. More important is that there is lack of mechanism to negotiate 
> what RTP payloads that will be used in direction client->server. This 
> will be a problem independent if we use a authentication payload or any 
> other payload. To allow the adequate security to work we need at least a 
> RTP header in the upstream traffic on the RTP channel carrying a SSRC. A 
> default behavior of reusing a downstream RTP payload type with an empty 
> payload could be used. The RTCP streams has no problems of this kind.
> 
> Conclusion:
> Symmetric RTP can be used with a minimal change to allow servers to 
> create bindings based on incoming RTP/RTCP packets. The stream will 
> always be possible to hijack by a man in the middle on the path between 
> server and client.
> 
> basic security: Use SSRC as secret identifier of client. Gives 1/2^31 
> probability of guessing. Requires extra transport header parameter 
> "client_ssrc".
> 
> basic+ security: Requires media flow bindings IP address to have same 
> address as RTSP connection. Both requires at least default rule on how 
> to send RTP packets to server.
> 
> best possible: Create special payload for identification and binding of 
> symmetric RTP ports. Can use any size of random tag. Creates problem 
> with what RTP payload type to use. Or could be sent on plain UDP.
> 
> This presents my view on how symmetric RTP could be used with RTSP 
> sessions.
> 
> 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
> 

-- 
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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Subject: Re: [MMUSIC] RTSP transport header, port parameter changes
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Can a separate IP address be specified for RTCP, as the port can? 
draft-ietf-mmusic-sdp4nat has this capability, and its needed for cases 
where the network is natting into more than one IP address.

-Jonathan R.

Magnus Westerlund wrote:
> Hi,
> 
> At the wednesdays telecon we discussed on how to update the transport 
> header to support STUN and symmetric RTP
> for traversal of NATs. It was decided to allow the ports to be specified 
> explicitly for RTP and RTCP. Therefore I have written up a proposal for 
> new text for the transport header below:
> 
> We where discussing making the port attributes totally flexible. However 
> as I started looking at the section I discovered that port, client_port, 
> server_port are RTP-specific. So my proposed changes does therefore only 
> solves what is necessary for RTP. An alternative to this change would be 
> to reclassify the port attributes to be general attributes and then they 
> should have total flexibility. That would also require some text on how 
> to related to the RTP media transport setup.
> 
> I also included a client_ssrc parameter. This would be used for 
> symmetric RTP to make the symmetric port opening more robust. I will try 
> to write up a mail regarding using symmetric RTP for RTSP sessions.
> 
> Please read and comment
> 
> Magnus
> 
> --------------------------------------------------------
> changed parts of section 12.40 Transport header in relation to 
> draft-ietf-mmusic-rfc2326bis-02.txt.
> 
> RTP-specific:
> 
>   These parameters are only valid if the media transport protocol  is
>   RTP.  The parameters containing port numbers are all subject to the
>   following consideration:  All port numbers where prior specified as
>   a value or range, see "old-port-spec" in ABNF below. To support NAT
>   traversal RTP has been updated to allow non-continuous port specif-
>   ication  not  following  the even/odd rule. As long as both the RTP
>   and RTCP port is specified explicitly they are allow  to  take  any
>   valid  port  number. Therefore the RTP specific port parameters has
>   been updated to support this. The updated format  ("new-port-spec")
>   SHOULD  only  be  used when required to specify non-continuous port
>   numbers. Before using the "new-port-spec" format  the  client  MUST
>   determine server support of the "play.basic" option tags.
> 
>   port: This parameter provides the RTP/RTCP port pair for  a  multi-
>        cast  session.  This  parameter  SHOULD NOT use the "new-port-
>        spec" format. It is should be  specified  as  a  range,  e.g.,
>        port=3456-3457
> 
>   client_port: This parameter provides the unicast RTP/RTCP port pair
>        on  the  client where media data and control information is to
>        be sent. It is specified either as a range,  e.g.,  port=3456-
>        3457 or as two explicit numbers port=3457/4976
> 
>   server_port: This parameter provides the unicast RTP/RTCP port pair
>        on  the  server where media data and control information is to
>        be sent. It is specified either as a range,  e.g.,  port=3456-
>        3457 or as two explicit numbers port=3457/4976
> 
>   ssrc: The ssrc parameter indicates the RTP  SSRC  [23]  value  that
>        should  be  (request)  or will be (response) used by the media
>        server. This parameter is only valid for unicast transmission.
>        It identifies the synchronization source to be associated with
>        the media stream, and is expressed as an eight  digit  hexide-
>        cimal value. In cases that a sender will use multiple SSRCs it
>        SHOULD NOT use this parameter.
> 
>   client_ssrc: The client_ssrc parameter indicates the RTP SSRC  [23]
>        value  that will be used by the client. This parameter is only
>        valid for unicast transmission. It identifies the synchroniza-
>        tion  source  to  be  associated with the media stream, and is
>        expressed as an eight digit hexidecimal value. In cases that a
>        client  will use multiple SSRCs it SHOULD NOT use this parame-
>        ter.
> 
> 
>   Transport                =  "Transport" ":" 1#transport-spec
>   transport-spec           =  transport-id *parameter
>   transport-id             =  transport-protocol "/" profile ["/" 
> lower-transport]
>                               ; no LWS is allowed inside transport-id
>   transport-protocol       =  "RTP" / token
>   profile                  =  "AVP" / token
>   lower-transport          =  "TCP" / "UDP" / token
>   parameter                =  ";" ( "unicast" / "multicast" )
>                           /   ";" "source" "=" address
>                           /   ";" "destination" [ "=" address ]
>                           /   ";" "interleaved" "=" channel [ "-" channel ]
>                           /   ";" "append"
>                           /   ";" "ttl" "=" ttl
>                           /   ";" "layers" "=" 1*DIGIT
>                           /   ";" "port" "=" port-spec
>                           /   ";" "client_port" "=" port-spec
>                           /   ";" "server_port" "=" port-spec
>                           /   ";" "ssrc" "=" ssrc
>                           /   ";" "client_ssrc" "=" ssrc
>                           /   ";" "mode" "=" mode-spec
>                           /   ";" trn-parameter-extension
>   port-spec                =  old-port-spec / new-port-spec
>   old-port-spec            =  port [ "-" port ]
>   new-port-spec            =  port "/" port ; RTP/RTCP port
>   trn-parameter-extension  =  par-name "=" trn-par-value
>   par-name                 =  token
>   trn-par-value            =  *unreserved
>   ttl                      =  1*3(DIGIT)
>   port                     =  1*5(DIGIT)
>   ssrc                     =  8*8(HEX)
>   channel                  =  1*3(DIGIT)
> 
>   address                  =  host ;As defined in RFC 2732 [30]
>   mode-spec                =  <"> 1#mode <"> / mode
> 
> 

-- 
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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Hi Jonathan,

No, currently it is not possible to specify a separate IP address for 
the RTCP port in respect to the RTP port. I understand that this will 
create problems if a NAT would assign the RTP and RTCP udp flow mappings 
to different public addresses. Have you any knowledge how usual this 
behavior is for NATs?

It is of course possible to solve, even rather easy at least for RTP. 
However the general case of specifying n address/port for server 
respective client would require some more work. To allow for this 
behavior in a general case would be best solved with totally new 
parameters.

Best Regards

Magnus

Jonathan Rosenberg wrote:

> Can a separate IP address be specified for RTCP, as the port can? 
> draft-ietf-mmusic-sdp4nat has this capability, and its needed for 
> cases where the network is natting into more than one IP address.
>
> -Jonathan R.
>
> Magnus Westerlund wrote:
>
>> Hi,
>>
>> At the wednesdays telecon we discussed on how to update the transport 
>> header to support STUN and symmetric RTP
>> for traversal of NATs. It was decided to allow the ports to be 
>> specified explicitly for RTP and RTCP. Therefore I have written up a 
>> proposal for new text for the transport header below:
>>
>> We where discussing making the port attributes totally flexible. 
>> However as I started looking at the section I discovered that port, 
>> client_port, server_port are RTP-specific. So my proposed changes 
>> does therefore only solves what is necessary for RTP. An alternative 
>> to this change would be to reclassify the port attributes to be 
>> general attributes and then they should have total flexibility. That 
>> would also require some text on how to related to the RTP media 
>> transport setup.
>>
>> I also included a client_ssrc parameter. This would be used for 
>> symmetric RTP to make the symmetric port opening more robust. I will 
>> try to write up a mail regarding using symmetric RTP for RTSP sessions.
>>
>> Please read and comment
>>
>> Magnus
>>
>> --------------------------------------------------------
>> changed parts of section 12.40 Transport header in relation to 
>> draft-ietf-mmusic-rfc2326bis-02.txt.
>>
>> RTP-specific:
>>
>>   These parameters are only valid if the media transport protocol  is
>>   RTP.  The parameters containing port numbers are all subject to the
>>   following consideration:  All port numbers where prior specified as
>>   a value or range, see "old-port-spec" in ABNF below. To support NAT
>>   traversal RTP has been updated to allow non-continuous port specif-
>>   ication  not  following  the even/odd rule. As long as both the RTP
>>   and RTCP port is specified explicitly they are allow  to  take  any
>>   valid  port  number. Therefore the RTP specific port parameters has
>>   been updated to support this. The updated format  ("new-port-spec")
>>   SHOULD  only  be  used when required to specify non-continuous port
>>   numbers. Before using the "new-port-spec" format  the  client  MUST
>>   determine server support of the "play.basic" option tags.
>>
>>   port: This parameter provides the RTP/RTCP port pair for  a  multi-
>>        cast  session.  This  parameter  SHOULD NOT use the "new-port-
>>        spec" format. It is should be  specified  as  a  range,  e.g.,
>>        port=3456-3457
>>
>>   client_port: This parameter provides the unicast RTP/RTCP port pair
>>        on  the  client where media data and control information is to
>>        be sent. It is specified either as a range,  e.g.,  port=3456-
>>        3457 or as two explicit numbers port=3457/4976
>>
>>   server_port: This parameter provides the unicast RTP/RTCP port pair
>>        on  the  server where media data and control information is to
>>        be sent. It is specified either as a range,  e.g.,  port=3456-
>>        3457 or as two explicit numbers port=3457/4976
>>
>>   ssrc: The ssrc parameter indicates the RTP  SSRC  [23]  value  that
>>        should  be  (request)  or will be (response) used by the media
>>        server. This parameter is only valid for unicast transmission.
>>        It identifies the synchronization source to be associated with
>>        the media stream, and is expressed as an eight  digit  hexide-
>>        cimal value. In cases that a sender will use multiple SSRCs it
>>        SHOULD NOT use this parameter.
>>
>>   client_ssrc: The client_ssrc parameter indicates the RTP SSRC  [23]
>>        value  that will be used by the client. This parameter is only
>>        valid for unicast transmission. It identifies the synchroniza-
>>        tion  source  to  be  associated with the media stream, and is
>>        expressed as an eight digit hexidecimal value. In cases that a
>>        client  will use multiple SSRCs it SHOULD NOT use this parame-
>>        ter.
>>
>>
>>   Transport                =  "Transport" ":" 1#transport-spec
>>   transport-spec           =  transport-id *parameter
>>   transport-id             =  transport-protocol "/" profile ["/" 
>> lower-transport]
>>                               ; no LWS is allowed inside transport-id
>>   transport-protocol       =  "RTP" / token
>>   profile                  =  "AVP" / token
>>   lower-transport          =  "TCP" / "UDP" / token
>>   parameter                =  ";" ( "unicast" / "multicast" )
>>                           /   ";" "source" "=" address
>>                           /   ";" "destination" [ "=" address ]
>>                           /   ";" "interleaved" "=" channel [ "-" 
>> channel ]
>>                           /   ";" "append"
>>                           /   ";" "ttl" "=" ttl
>>                           /   ";" "layers" "=" 1*DIGIT
>>                           /   ";" "port" "=" port-spec
>>                           /   ";" "client_port" "=" port-spec
>>                           /   ";" "server_port" "=" port-spec
>>                           /   ";" "ssrc" "=" ssrc
>>                           /   ";" "client_ssrc" "=" ssrc
>>                           /   ";" "mode" "=" mode-spec
>>                           /   ";" trn-parameter-extension
>>   port-spec                =  old-port-spec / new-port-spec
>>   old-port-spec            =  port [ "-" port ]
>>   new-port-spec            =  port "/" port ; RTP/RTCP port
>>   trn-parameter-extension  =  par-name "=" trn-par-value
>>   par-name                 =  token
>>   trn-par-value            =  *unreserved
>>   ttl                      =  1*3(DIGIT)
>>   port                     =  1*5(DIGIT)
>>   ssrc                     =  8*8(HEX)
>>   channel                  =  1*3(DIGIT)
>>
>>   address                  =  host ;As defined in RFC 2732 [30]
>>   mode-spec                =  <"> 1#mode <"> / mode
>>
>>
>

-- 

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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The MMUSIC working group is scheduled for one slot at the 56th IETF meeting
in San Francisco. On the tentative agenda, this is from 9:00 to 11:30am on
Monday 17th March. This is subject to change - do not plan travel based on
this date.

Those wanting agenda time at the meeting should contact the working group
chairs by 3rd March. Submitting a draft does not automatically provide
agenda time. 

You are reminded that the cut-off date for -00 Internet Draft submissions
is Monday, 24th February at 9:00am EST (GMT -05:00). Unless you have prior
approval from the chairs, -00 submissions should have filenames with the
format draft-yourname-mmusic-...-00.txt. Be advised that updates to -00
drafts received after 24th February will not be accepted. 

All other Internet-Draft submissions must be made by Monday, 3rd March at
9:00am EST.

Drafts which are submitted late and do not make the cutoff date will not be
discussed at the meeting.

-- 
Colin Perkins
http://csperkins.org/
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To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
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Subject: Re: [MMUSIC] RTSP & NAT: Symmetric RTP
References: <3E479FFE.4030908@era.ericsson.se> <3E47A519.4050908@dynamicsoft.com>
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Hi Jonathan,

Comments:

Jonathan Rosenberg wrote:

> I agree that symmetric RTP is clearly the best solution for RTSP as it 
> is used today. I do not believe that relying on source IP being the 
> same  between the RTSP and RTP sessions is a good idea. This won't 
> work in enterprises which may be natting into a multiplicity of IP 
> addresses.

I agree that requiring IP address to be same may be to restrictive and 
create a lot of problems. I think that a sufficiently strong solution of 
identifying the end-point anyway is enough.

>
> I would also strongly advise not to mandate that as the one and only 
> one solution for nat traversal. A client should be allowed to do other 
> things, and you want to leave the door open for better solutions down 
> the road.

We are not planning to require any one solution. It is planned to make a 
draft about RTSP and NAT traversal. This will document and write up 
rules how to use the different mechanism and what to consider. Also what 
to think when writing firewall ALGs will be documented. The solution 
that the draft currently will discuss is:
1. STUN
2. Symmetric NAT
3. TCP tunneling
4. ALG (not recommended)

>
> As for the random identifiers, I will note that I recently proposed 
> such a mechanism in support of connection oriented media. This draft 
> also discusses the attack you are talking about:
>
> http://www.ietf.org/internet-drafts/draft-rosenberg-mmusic-comedia-fix-00.txt 
>

I will further discuss this in a reply to David Yon's reply.

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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From mailnull@www1.ietf.org  Mon Feb 10 10:40:31 2003
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Subject: Re: [MMUSIC] RTSP & NAT: Symmetric RTP
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As Jonathan points out in his reply, a similar attack is being discussed in 
the context of comedia/comedia-fix.  The proposed solution in comedia-fix 
is to window the protocol by always starting the stream with a unique ID 
prior to sending packets using the protocol that has been signalled in the 
SDP.  My discontent with this mechanism is well documented in that email 
thread.

If there is to be some sort of session correlation embedded in the media, I 
would greatly prefer the approach you are suggesting.  I.e., make the 
endpoint ID be something derived from the protocol proper, rather than 
inserting the ID out-of-band in relation to the signalled protocol.

As evidenced by this RTSP discussion, this need exists outside the scope of 
connection-based media, so perhaps comedia isn't the place to document a 
generic media/session correlation mechanism.

If one accepts the notion that connectionless media has this need for 
session correlation, then it becomes clear that the comedia-fix proposal is 
not equipped to address the problem in this larger scope.  What immediately 
comes to my mind is how to window the protocol when the media is carried 
over unreliable transport (i.e., UDP).  Since there is no guarantee of 
ordering or delivery, the sender cannot simply just send the first packet 
with the endpoint ID.  The packet may never arrive at the receiver, or it 
may arrive after other media packets.  So the receiver has no reliable way 
of parcelling out the endpoint ID from the in-band media, unless of course 
the endpoint ID is encoded within the framework of the media itself.

This is sounding more and more to me like endpoint ID is a standalone 
topic.  Perhaps a short I-D that documents the need and the SDP syntax, 
plus globbing together a few of the more obvious protocol extensions (i.e., 
the RTP suggestion discussed below).

At 07:50 AM 2/10/2003, Magnus Westerlund wrote:

>Security risks: Hi-jacking of media streams. The basic problem is that any 
>RTP/RTCP traffic arriving to the sender port prior to the clients message 
>will result in redirection of the media stream to the sent packets address 
>and port. This allows an attacker to spoof the sender IP/port number to 
>that of a target of an Distributed DOS attack. This can basically be done 
>without being on the path between client and server by simple guessing 
>port numbers.
>
>Good Solution: Create a RTP payload format that carries session identifier 
>and a sufficient large random tag that prevents guessing. However due to 
>NATs and the setup this can never be ensured against attackers located on 
>the path between client and server.
>
>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 David,

For RTSP and RTP usage it seems possible to make the whole process work 
using the RTP SSRC. In RTP flows that is present in the RTP header 
making it simple to handle. For RTCP it becomes even more important that 
it can be sent in the context of RTCP. RTCP traffic will flow through 
out the whole session. Therefore a special demultiplexing between RTCP 
and binding messages will be difficult.

Using SSRC requires a simple parameter addition to the transport header. 
The client simple tells the server what to use. However the big question 
is if a 32 bit SSRC value is unique enough to function well. A RTSP 
server at least has the advantage that it can use multiple sending 
ports. If a client SSRC proposal already is in use on some port this can 
simple be sent to a different sender port.

Using other methods of transporting this ID on top of UDP or other 
transport protocols create problems. Defining a general format that is 
simple used creates multiplexing problems for all use cases. For RTP it 
could be possible to define both a RTP payload format and a RTCP packet 
type that can be sent containing a general type of ID.

Best Regards

Magnus

David Yon wrote:

> As Jonathan points out in his reply, a similar attack is being 
> discussed in the context of comedia/comedia-fix.  The proposed 
> solution in comedia-fix is to window the protocol by always starting 
> the stream with a unique ID prior to sending packets using the 
> protocol that has been signalled in the SDP.  My discontent with this 
> mechanism is well documented in that email thread.
>
> If there is to be some sort of session correlation embedded in the 
> media, I would greatly prefer the approach you are suggesting.  I.e., 
> make the endpoint ID be something derived from the protocol proper, 
> rather than inserting the ID out-of-band in relation to the signalled 
> protocol.
>
> As evidenced by this RTSP discussion, this need exists outside the 
> scope of connection-based media, so perhaps comedia isn't the place to 
> document a generic media/session correlation mechanism.
>
> If one accepts the notion that connectionless media has this need for 
> session correlation, then it becomes clear that the comedia-fix 
> proposal is not equipped to address the problem in this larger scope.  
> What immediately comes to my mind is how to window the protocol when 
> the media is carried over unreliable transport (i.e., UDP).  Since 
> there is no guarantee of ordering or delivery, the sender cannot 
> simply just send the first packet with the endpoint ID.  The packet 
> may never arrive at the receiver, or it may arrive after other media 
> packets.  So the receiver has no reliable way of parcelling out the 
> endpoint ID from the in-band media, unless of course the endpoint ID 
> is encoded within the framework of the media itself.
>
> This is sounding more and more to me like endpoint ID is a standalone 
> topic.  Perhaps a short I-D that documents the need and the SDP 
> syntax, plus globbing together a few of the more obvious protocol 
> extensions (i.e., the RTP suggestion discussed below).
>
> At 07:50 AM 2/10/2003, Magnus Westerlund wrote:
>
>> Security risks: Hi-jacking of media streams. The basic problem is 
>> that any RTP/RTCP traffic arriving to the sender port prior to the 
>> clients message will result in redirection of the media stream to the 
>> sent packets address and port. This allows an attacker to spoof the 
>> sender IP/port number to that of a target of an Distributed DOS 
>> attack. This can basically be done without being on the path between 
>> client and server by simple guessing port numbers.
>>
>> Good Solution: Create a RTP payload format that carries session 
>> identifier and a sufficient large random tag that prevents guessing. 
>> However due to NATs and the setup this can never be ensured against 
>> attackers located on the path between client and server.
>>
>> 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
>
>

-- 

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



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Subject: Re: [MMUSIC] RTSP and NATs
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>,
        jdrosen@dynamicsoft.com
Cc: IETF MMUSIC WG <mmusic@ietf.org>
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Jonathan you wrote:

>I agree that symmetric RTP is clearly the best solution for RTSP as it
>is used today. I do not believe that relying on source IP being the same
> between the RTSP and RTP sessions is a good idea. This won't work in
>enterprises which may be natting into a multiplicity of IP addresses.

YES!
I fully agree, (see item 3 below)


Magnus,

(Sorry for the delay in responding to previous RTSP&NAT)


Having a RTSP server and a STUN server co-located at the same port
is no concern for us neither in terms of security nor in terms of implementation,
or to be more specific it is largely worth the trouble !
i.e. the trade of is rather simple: either you do that or
NATs will prevent you from using RTSP....

And yes, magnus, you are right, the first key issue is to inform the client about the ports
the server will be listening for incomming STUN packets,
which need to be the RTP and RTCP ports....

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

let me summarize the 4 open issues I see for RTSP/STUN/NAT:

(1) server signals its RTP and RTCP ports:

it can be done either:
(a) in the SDP (since its per-media ?) as in draft-ietf-mmusic-sdp4nat-03.txt
or
(b) in the headers of the response to DESCRIBE or SETUP

Note that if SETUP is used it is a failed-in-advance attempt
("hey, try again, because this wont work") causing 1 round trip,
so I dont like this solution much, and also there are cases when
you dont use DESCRIBE ...
so I would prefer solution (a)SDP.


2) there is the possiblity that inside DESCRIBE and/or SETUP the client
gives its IP address, then the server could compare this transported client IP
with the apparent IP address of the client,
Then the server could make a specific reply:
-we would need to define a new status-
for example:

 "463: See you from behind a NAT, use STUN on designated ports"

But this solution would only be a safe-guard,
I mean it is smart to assume that:

"When the server indicates explicit RTP and RTCP ports,
the client SHOULD send STUN queries toward the server
at these port numbers and wait for STUN responses
indicating the possible presence of NAT."

so that if you prefer to wait for a possible 463...
you loose... a round trip !-) too bad!


3) One key missing item  (as noted by Jonathan)
is also the capability at SETUP to indicate not only a specific
port number but also a specific IP address since through
some NATs RTP and RTCP packets will come out not only with
an (apparently) random port number but also
from a different apparent IP address !

So we _must_ add this capability....(but it is no big deal right?)


4) you also need to send packets periodically to keep
the NAT mappings "open". But the issue here is simply
to document that, right ?


regards,


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


                                                                                                                                              
                                                                                                                                              
                                                          To:   Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1                                     
                                                          cc:   IETF MMUSIC WG <mmusic@ietf.org>                                              
                                                          Subject:    Re: [MMUSIC] RTSP and NATs                                              
                                                                                                                                              
               Magnus Westerlund                          Classification:                                                                     
               <magnus.westerlund@era.ericsson                                                                                                
               .se>                                                                                                                           
                                                                                                                                              
               05/02/2003 19:45                                                                                                               
                                                                                                                                              
                                                                                                                                              




Hi Philippe,

philippe.gentric@philips.com wrote:

>Magnus,
>
>I strongly support the recommendation to not use ALGs,
>therefore STUN should be used instead but then you wrote:
>
>
> > To be able to use STUN to traverse symmetric NATs the  STUN  server
> >  needs to be co-located with the streaming server media distribution
> >  ports. As this will create implementations difficulties and  possi-
> > bly security problems this SHOULD NOT be done.
>
>I am surprised, I would have proposed _on the contrary_ a "SHOULD" here
>(since traversing symetric NATs is a key feature) !
>
>
>also what do you mean by  "implementation difficulties" ? (I really cannot see any ?)
>and could you explicit what type of _additional_ security issues this would cause
>(i.e. issues that are not inherent to running either a RTSP or a STUN server in the first place ?)
>
>
The problem of running STUN for a symmetric NAT is that the STUN server
must be located at the servers sending port. So using the same RTSP
mechanism that are used for traversing a cone-nat the client would:
1. first contact a well known port with the server for each his media
stream to get the mapping of the stream. The client can't use any other
then the well known port because it doesn't know what port to send to.
2. In the SETUP ask the server to send from its STUN servers well known
port. It also needs to receive RTCP on that port.
3. For keep alive on the RTP port the client needs to send periodically
STUN messages to the STUN server.

This is a mess for a implementor. It need to have a STUN server
receiving the STUN messages while the RTP/RTCP stack should receive all
other messages. Also all the clients need to reside on the same port
number which creates a multiplexing nightmare.

If you change the setup phase so that the streams are first SETUP to
dummy ports then reconfigured with a later message, the STUN could be
located at each media streams source address. However in this setup it
is actually better to use symmetric RTP in all regards except perhaps
hi-jacking of media streams. However with certain restrictions you can
actually end up with at least the same level of security that RTSP
provides today.

How are you using STUN to traverse a symmetric NAT?

Best Regards

Magnus


--

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 & NAT: Symmetric RTP
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Using SSRC was mentioned a long time ago, although at the time dismissed as 
too protocol-specific.  But the context of the discussion has changed, and 
if that works then I have no problem with it in principle.  That said, 
there was at least one other concern voiced about whether 32 bits is 
sufficient.  It also begs the question of what format to use for an 
endpoint-ID, and whether this format is to be fixed or variable.  One 
approach would be to make the endpoint-ID format specific on a per-protocol 
basis, with a limited number of choices (it seems like string and integer 
would cover most needs).

I'm not sure I followed you exactly, but I read your first paragraph to 
mean that using a windowed protocol approach for endpoint ID is even more 
of a problem in the context of RTCP.  Is that a fair interpretation?

At 10:52 AM 2/10/2003, Magnus Westerlund wrote:
>Hi David,
>
>For RTSP and RTP usage it seems possible to make the whole process work 
>using the RTP SSRC. In RTP flows that is present in the RTP header making 
>it simple to handle. For RTCP it becomes even more important that it can 
>be sent in the context of RTCP. RTCP traffic will flow through out the 
>whole session. Therefore a special demultiplexing between RTCP and binding 
>messages will be difficult.
>
>Using SSRC requires a simple parameter addition to the transport header. 
>The client simple tells the server what to use. However the big question 
>is if a 32 bit SSRC value is unique enough to function well. A RTSP server 
>at least has the advantage that it can use multiple sending ports. If a 
>client SSRC proposal already is in use on some port this can simple be 
>sent to a different sender port.
>
>Using other methods of transporting this ID on top of UDP or other 
>transport protocols create problems. Defining a general format that is 
>simple used creates multiplexing problems for all use cases. For RTP it 
>could be possible to define both a RTP payload format and a RTCP packet 
>type that can be sent containing a general type of ID.
>
>Best Regards
>
>Magnus
>
>David Yon wrote:
>
>>As Jonathan points out in his reply, a similar attack is being discussed 
>>in the context of comedia/comedia-fix.  The proposed solution in 
>>comedia-fix is to window the protocol by always starting the stream with 
>>a unique ID prior to sending packets using the protocol that has been 
>>signalled in the SDP.  My discontent with this mechanism is well 
>>documented in that email thread.
>>
>>If there is to be some sort of session correlation embedded in the media, 
>>I would greatly prefer the approach you are suggesting.  I.e., make the 
>>endpoint ID be something derived from the protocol proper, rather than 
>>inserting the ID out-of-band in relation to the signalled protocol.
>>
>>As evidenced by this RTSP discussion, this need exists outside the scope 
>>of connection-based media, so perhaps comedia isn't the place to document 
>>a generic media/session correlation mechanism.
>>
>>If one accepts the notion that connectionless media has this need for 
>>session correlation, then it becomes clear that the comedia-fix proposal 
>>is not equipped to address the problem in this larger scope.
>>What immediately comes to my mind is how to window the protocol when the 
>>media is carried over unreliable transport (i.e., UDP).  Since there is 
>>no guarantee of ordering or delivery, the sender cannot simply just send 
>>the first packet with the endpoint ID.  The packet may never arrive at 
>>the receiver, or it may arrive after other media packets.  So the 
>>receiver has no reliable way of parcelling out the endpoint ID from the 
>>in-band media, unless of course the endpoint ID is encoded within the 
>>framework of the media itself.
>>
>>This is sounding more and more to me like endpoint ID is a standalone 
>>topic.  Perhaps a short I-D that documents the need and the SDP syntax, 
>>plus globbing together a few of the more obvious protocol extensions 
>>(i.e., the RTP suggestion discussed below).
>>
>>At 07:50 AM 2/10/2003, Magnus Westerlund wrote:
>>
>>>Security risks: Hi-jacking of media streams. The basic problem is that 
>>>any RTP/RTCP traffic arriving to the sender port prior to the clients 
>>>message will result in redirection of the media stream to the sent 
>>>packets address and port. This allows an attacker to spoof the sender 
>>>IP/port number to that of a target of an Distributed DOS attack. This 
>>>can basically be done without being on the path between client and 
>>>server by simple guessing port numbers.
>>>
>>>Good Solution: Create a RTP payload format that carries session 
>>>identifier and a sufficient large random tag that prevents guessing. 
>>>However due to NATs and the setup this can never be ensured against 
>>>attackers located on the path between client and server.
>>>
>>>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
>
>--
>
>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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inline.

Magnus Westerlund wrote:
> Hi Jonathan,
> 
> Comments:
> 
> Jonathan Rosenberg wrote:
> 
>> I agree that symmetric RTP is clearly the best solution for RTSP as it 
>> is used today. I do not believe that relying on source IP being the 
>> same  between the RTSP and RTP sessions is a good idea. This won't 
>> work in enterprises which may be natting into a multiplicity of IP 
>> addresses.
> 
> 
> I agree that requiring IP address to be same may be to restrictive and 
> create a lot of problems. I think that a sufficiently strong solution of 
> identifying the end-point anyway is enough.

I would like to see the ability to specify the client IP/port for rtp 
and rtcp so as to allow RTSP to work with stun, turn and other related 
protocols, even if there is a separate endpoint id mechanism.


-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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From: Tom Marshall <tmarshall@real.com>
To: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP & NAT: Symmetric RTP
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On Mon, Feb 10, 2003 at 11:56:50PM -0500, Jonathan Rosenberg wrote:
> inline.
>=20
> Magnus Westerlund wrote:
> >Hi Jonathan,
> >
> >Comments:
> >
> >Jonathan Rosenberg wrote:
> >
> >>I agree that symmetric RTP is clearly the best solution for RTSP as it=
=20
> >>is used today. I do not believe that relying on source IP being the=20
> >>same  between the RTSP and RTP sessions is a good idea. This won't=20
> >>work in enterprises which may be natting into a multiplicity of IP=20
> >>addresses.
> >
> >
> >I agree that requiring IP address to be same may be to restrictive and=
=20
> >create a lot of problems. I think that a sufficiently strong solution of=
=20
> >identifying the end-point anyway is enough.
>=20
> I would like to see the ability to specify the client IP/port for rtp=20
> and rtcp so as to allow RTSP to work with stun, turn and other related=20
> protocols, even if there is a separate endpoint id mechanism.

It would seem to me that a new parameter is appropriate.  client_port
implies that only a port number is given.  Perhaps something like this could
serve as an endpoint specifier with addresses?

"client_addr" "=3D" addr_list
addr_list -> addr_list "/" addr || addr =20
addr      -> ip4addr || ip6addr =20
ip4addr   -> ip4host || ip4host ":" port =20
ip6addr   -> ip6host || ip6host ":" port  =20
ip4host   -> hostname || ip4literal =20
ip6host   -> hostname || ip6literal =20
port      -> <digits>           =20
hostname  -> <standard dns hostname> =20
ip4literal-> <digits and dots> =20
ip6literal-> "[" <hex quads with colons> "]" =20

=2E. and perhaps a note saying that client_addr is mutually exclusive with
client_port, source, and destination parameters.

Comments?

--=20
Happiness isn't something you experience; it's something you remember.
        -- Oscar Levant

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Subject: Re: [MMUSIC] RTSP and NATs
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Hi Philippe,

Thanks for replying and giving your input. Comments inline.

philippe.gentric@philips.com wrote:

>Magnus,
>
>(Sorry for the delay in responding to previous RTSP&NAT)
>
>
>Having a RTSP server and a STUN server co-located at the same port
>is no concern for us neither in terms of security nor in terms of implementation,
>or to be more specific it is largely worth the trouble !
>i.e. the trade of is rather simple: either you do that or
>NATs will prevent you from using RTSP....
>  
>

Yes, but co-locating STUN and RTP/RTCP sender ports does create 
demultiplexing problems. There is no real multiplexing point between 
these two protocols. Both will be run over IP/UDP. No indication in UDP 
points out if the UDP payload is a STUN message or a RTP/RTCP message. 
Matching based on message structure is not a sure thing. Therefore I 
would much rather use another approach.

 From this point of view symmetric RTP may be a much better solution 
from the multiplexing perspective. Of course this has the security 
problems. The advantage of symmetric RTP using SSRC as identifier is 
that is has no demultiplexing problem and at least gives basic security. 
For all others than the man in the middle guessing correctly a 32 bit 
number is rather hard. And the man in the middle is still impossible to 
protect against.


>And yes, magnus, you are right, the first key issue is to inform the client about the ports
>the server will be listening for incomming STUN packets,
>which need to be the RTP and RTCP ports....
>
>*******************************************
>
>let me summarize the 4 open issues I see for RTSP/STUN/NAT:
>
>(1) server signals its RTP and RTCP ports:
>
>it can be done either:
>(a) in the SDP (since its per-media ?) as in draft-ietf-mmusic-sdp4nat-03.txt
>or
>(b) in the headers of the response to DESCRIBE or SETUP
>
>Note that if SETUP is used it is a failed-in-advance attempt
>("hey, try again, because this wont work") causing 1 round trip,
>so I dont like this solution much, and also there are cases when
>you dont use DESCRIBE ...
>so I would prefer solution (a)SDP.
>
One solution would be to use the well known port of STUN. That allows a 
client to first initiate a STUN request to the server. Upon detecting 
the clients public IP/port it sends a RTSP SETUP with this information 
and actually tells the server by using Server_port of the STUN port that 
it would the server to send media from these ports. This of course has 
the problem that the server will need to demultiplex all incoming 
RTP/RTCP traffic based on source IP/port to know which session it 
belongs to. Also any type of load balancing between ports will be 
impossible.

To achieve load balancing between ports it will be a requirement to have 
signaling indicating which port the client shall connect the different 
flows to when doing STUN. It will still be required for the client to 
tell the server which port it must use.

>
>
>2) there is the possiblity that inside DESCRIBE and/or SETUP the client
>gives its IP address, then the server could compare this transported client IP
>with the apparent IP address of the client,
>Then the server could make a specific reply:
>-we would need to define a new status-
>for example:
>
> "463: See you from behind a NAT, use STUN on designated ports"
>
>But this solution would only be a safe-guard,
>I mean it is smart to assume that:
>
>"When the server indicates explicit RTP and RTCP ports,
>the client SHOULD send STUN queries toward the server
>at these port numbers and wait for STUN responses
>indicating the possible presence of NAT."
>
>so that if you prefer to wait for a possible 463...
>you loose... a round trip !-) too bad!
>  
>

Might be a good addition. A question, are you assuming that any request 
containing destination IP that mismatch the source of the control flow 
will behind a NAT? This will then not work for non-transparent proxies. 
Actually I think we have a large problem in RTSP with proxies and where 
to send the media data. If a proxies is part of the TCP path to a server 
that connection will not have source IP equal to destination address of 
traffic. Then a server shall not send traffic to that source.

>
>3) One key missing item  (as noted by Jonathan)
>is also the capability at SETUP to indicate not only a specific
>port number but also a specific IP address since through
>some NATs RTP and RTCP packets will come out not only with
>an (apparently) random port number but also
>from a different apparent IP address !
>  
>
I think a solution like what Tom has suggested will be required to fix 
this in a general way. However this solution has one very big problem 
with it. It opens up the DOS attack possibilities of RTSP. If your 
allowed to freely specify any IP/port this mechanism will be the certain 
choice of any DOS attacker. So we can't allow this to be used because 
this have worse security properties than symmetric RTP. So either we 
accept that  NATs that gives out different IP address to the same client 
does not work or it will be required to create a security scheme that 
allows us to authorize the destination of the media traffic.

>So we _must_ add this capability....(but it is no big deal right?)
>  
>
It is solveable but it will only work with servers that has been 
updated. But this is of course true with any
server side changes.

>
>4) you also need to send packets periodically to keep
>the NAT mappings "open". But the issue here is simply
>to document that, right ?
>  
>
Yes, that is true independent which NAT traversal technique you use, 
STUN, or symmetric RTP.

Some conclusions:
Actually using STUN might not be possible at all from a security 
perspective if the TCP connection comes from a different IP addresses 
then the media streams. This will only work for single IP address NATs. 
Other solutions needs an authorization mechanism for the destination.

Symmetric RTP seems to have better security despite all. Also it works 
for symmetric NATs without creating multiplexing problems, at least if 
we rely on RTP/RTCP's SSRC.


Best Regards

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
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Subject: Re: [MMUSIC] RTSP & NAT: Symmetric RTP
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Hi David,

David Yon wrote:

> Using SSRC was mentioned a long time ago, although at the time 
> dismissed as too protocol-specific.  But the context of the discussion 
> has changed, and if that works then I have no problem with it in 
> principle.  That said, there was at least one other concern voiced 
> about whether 32 bits is sufficient.  It also begs the question of 
> what format to use for an endpoint-ID, and whether this format is to 
> be fixed or variable.  One approach would be to make the endpoint-ID 
> format specific on a per-protocol basis, with a limited number of 
> choices (it seems like string and integer would cover most needs).


My feeling is that 32 bits are enough for this. Because a brute force 
attack is not possible without the receiving end-point notice it. If 
good random numbers are used then it seem difficult enough to guess 
within a few message the correct one out of the 4.2 billion ones possible.

>
> I'm not sure I followed you exactly, but I read your first paragraph 
> to mean that using a windowed protocol approach for endpoint ID is 
> even more of a problem in the context of RTCP.  Is that a fair 
> interpretation?

Yes, the problem as I see it to safely demultiplex between ordinary 
traffic and the endpoint ID message.

Best Regards 

Magnus Westerlund 

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

I think this is a good idea, however we will still have the DOS attack 
problems when specifying addresses other then the ones the RTSP 
connection uses. I don't have a good way of solving this.

As you used a description format that is not the ABNF of RFC 2234 I 
converted it below.

>It would seem to me that a new parameter is appropriate.  client_port
>implies that only a port number is given.  Perhaps something like this
>could
>serve as an endpoint specifier with addresses?
>  
>
client_addr = "client_addr" "=" addr_list
addr_list = addr  *("/" addr)
addr = ip4addr / ip6addr
ip4addr = ip4host [":" port]
ip6addr = ip6host [":" port]
ip4host = hostname / ip4literal
ip6host = hostname / ip6literal
port = 1*DIGIT
ip4literal = <digits and dots>  
ip6literal = "[" <hex quads with colons> "]"

Some small detail questions:
- Should the port range be limited to 16 bit values?
- What are the rules when it would be optional to specify port numbers?

Cheers

Magnus Westerlund 

Multimedia Technologies, Ericsson Research ERA/TVA/A
----------------------------------------------------------------------
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From: Paul Kyzivat <pkyzivat@cisco.com>
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Gonzalo,

I appreciate the value of this kind of mechanism. But I have a few 
questions about the details.

My main concern is whether it is possible to be certain of ordering of 
media content as usage shifts from one alternative stream to another. 
There are several issues in this regard:

- multiple streams may be established in parallel, but only one should 
be used. In some cases, e.g. UDP, the only way to attempt establishment 
is to send media over the stream. So by definition more than one is 
being used. And if this is done in parallel there is no way to know the 
order in which the media will arrive.

- "During the session, a sender MAY choose to use any of the grouped m 
lines at a particular point in time to send data. This makes it possible 
to use the ALT semantics as a fail-over mechanism for ongoing sessions."

This is described as being useful for failover, but not restricted to 
that. If a sender simply stops using one stream and begins using 
another, there is no guarantee of the order in which the resultant media 
will arrive at the recipient. This may again result in ambiguity about 
the order of media content.

This gets worse if it is permissible to subsequently resume using a 
stream that had previously been used and then abandoned. As you bounce 
between streams it will rapidly become impossible to know the ordering 
of media content.

I think there need to be more constraints over how an alternative is 
selected and how transitions between them are subsequently selected.

	Paul

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From: Tom Marshall <tmarshall@real.com>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: mmusic@ietf.org
Subject: Re: [MMUSIC] RTSP & NAT: Symmetric RTP
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On Tue, Feb 11, 2003 at 12:21:38PM +0100, Magnus Westerlund wrote:
> Hi Tom,
>=20
> I think this is a good idea, however we will still have the DOS attack=20
> problems when specifying addresses other then the ones the RTSP=20
> connection uses. I don't have a good way of solving this.

Unfortunately, I don't either.  Is it mandatory to solve it in the spec?  It
seems to me that the best solution is to have a challenge/response mechanism
for each transport.  But that is beyond the scope of RTSP.  Perhaps it would
be sufficient to say that the possibility for abuse exists and some form of
cryptographically secure challenge/response mechanism should be used if the
server allows sending to a different address than the client RTSP endpoint.

The most troublesome scenario is a NAT that maps the RTSP and transport
flows to different IP addresses.  How common is this?

> As you used a description format that is not the ABNF of RFC 2234 I=20
> converted it below.

[...]

Thank you.

> Some small detail questions:
> - Should the port range be limited to 16 bit values?

For UDP with IPv4 and IPv6, yes.  For others (non-UDP or IPv7+), who knows?

> - What are the rules when it would be optional to specify port numbers?

Perhaps a transport can have an implied or well-known port number, or a port
number is not applicable to it, etc.

--=20
A lot of people are afraid of heights.  Not me.  I'm afraid of widths.
        -- Steven Wright

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  RtspTaskforceTelConMinutes05Jan2003

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

   1. Attendees <#section1>
   2. Minutes <#section2>
         1. Follow up on last meeting <#section2.1>
         2. Administrative <#section2.2>
         3. Issues <#section2.3>
               1. RTSP and NAT: Bug 513888 
               2. Use of destination parameter in Transport header:
bug507347
         4. Next meeting <#section2.3>

  1 Attendees
    * Aravind Narasimhan	
    * Jorg Ott
    * Magnus Westerlund
    * Colin Perkins
    * Rob Lanphier
    * Tom Marshall
    * Thomas Zeng

  2 Minutes

    2.1 Follow up on last meeting
      1. Re-direct, waiting for new text from Tom
	 Tom Marshall:   Not done yet.

      2. Update on Range header text in regards to Live streams.

	 Magnus: We have not published, to MMUSIC WG, Range header change
proposal that we agreed upon last time.
	 Thomas Zeng:  I can try to write something up.

      3. Extension mechanism, have support as SIP. 
	 Jorg:  Seems reasonable to me.
	 
      4. Registration of feature tags, and anything not basic shall be
written as extensions.
         * nobody has comments
   
    2.2 Administrative
      Jorg:   Would like to set a milestone in WG charter: Oct 2003 to
submit a new RTSP RFC to GW last call then IESG. This RFC will unlikely (due
to lack of time for interop tests) to bring RTSP to next maturity level
(Draft Standards) but will still obsolete RFC2326.
      
      Magnus: Does not hurt to have such a goal to work with. 
      Magnus: Regarding document editors, any volunteer?
      Thomas: I thought the plan is divide and conquer?
      Magnus: That's right, but still need an editor-in-chief. Nobody seems
to have time for that.
      Colin:  We can move in a slow pace, as people have time.
      Jorg:   If resolutions to open issues are well-documented in
SourceForge including section numbers (and perhaps page numbers) then the
final editor can pull things together relatively easily.

      Magnus: Should use draft 02 (i.e.,
draft-ietf-mmusic-rfc2326bis-02.txt) as baseline and document changes in
that context.

      Thomas: Perhaps you can send out an example.


  2.3. Issues:
    2.3.1  RTSP and NAT: Bug 513888 <#section2.2.1>
      - What solutions exist to make RTSP traverse NATs without ALG.
          http://rtsp.org/bug513888
      - STUN? See
        <http://www.ietf.org/internet-drafts/draft-ietf-midcom-stun-05.txt>
      - NAT, ALG and Firewall recommendations in relation to RTSP?      

    Magnus:  We should change Transport parameter syntax to use
"client_port" to specify both RTP and RTCP ports
    Tom:     Should make it a generic parameter, not specific to RTP/RTCP.

    Colin:   Old server may not allow this new syntax.
    Tom:     Recommend if client ports are in fact contiguous, please use
old "-" syntax so can still work with old servers.
    Magnus:   Volunteer to change BNF accordingly?
    <silence>
    Magnus:   I may take a shot at it later. 

    Magnus:  As to STUN, I'll propose a change to "client_port" as well. 
             Let's discuss then Symetric RTP for RTSP.

    Magnus:  Sending RTP packets from client to server leaves doors open for
"hijacking", because it's much easier '	  to spoof RTP packets than do TCP
streams. Therefor we need RTP authentication.

    Thomas:  Could require the use of SessinID and/or SSRC in back RTP
packets as a crude authentication.
    Tom:     I prefer using SessionID in this case.

    Magnus:  We should document different NAT mechanisms in relation to
RTSP.

    Magnus:  How can server and client know that Symetric RTP is the
protocol?
  
    Tom:     Use Transport header to signal Symetric RTP.

    Magnus:  Yes, client can use a "symetric" parameter in Transport header
in its Setup request to indicate the
             intention to use symetric RTP for NAT traversal, and server can
use option tag in its Support header to             indicate whether the
server supports symetric RTP.
	    
    CONCLUSION: Add a parameter ("symetric"?) to Transport parameters. Add a
option tag for Symetric RTP feature.

    Magnus:  Moving on to ALG (Application Level Gateway), I think ALGs will
not work with RTSP because:
             1. ALGs have deployment issues 2. ALGs are not up to date   3.
People writing ALGs are generally not familiar with RTSP.

    Magnus:   On the other hand, do we need to write a draft to recommend
how to write ALG for RTSP?

    Rob:     RTSP support has already been added to some application level
firewalls (e.g., ipchain rules in Linux
             Firewalls). NATs and FWs  may need RTSP core spec's support,
but we can early on find those dependencies and address them in core RTSP
spec.

    Magnus:  The section on NATs/FWs could go into a separate draft, which
would also be easier to maintain.

    CONCLUSION: Move NAT/FW section from current rfc2326bis draft 02 to a
separate draft, but make sure the
             core RTSP spec has the right features to support NAT/FW
tranversals schemes, including ALGs, STUN, UPnP or MidComm and Symetric RTP.
Option tags are to be created to extend core RTSP.

    Magnus:  This extension draft should be published at the same time as
the core RTSP spec.

    ACTION ITEM: Magnus to change current text to refer to the separate
draft.



    2.3.2   Use of destination parameter in Transport header: bug507347
     (See http://rtsp.org/bug507347)

     Magnus: I don't think we should remove this parameter, because it seems
to be useful to NAT tranversal.
             We should clearly document the security danger of its use,
though.

     Rob:    Could suggest that the server authenticate receiver using some
out-of-band authentication schemes.

     Mangus: So the wording can be: "One shall not allow destination set to
a different IP unless the receive authorizes it".

     Rob:    The working has to be clear as to which node is sending
credentials... There are three entities: the server, the RTSP client and the
RTP receipient. What the server needs to do is ti make sure RTSP client has
the permission from the RTP receipient to blast packets to it.

     Magnus:   Is it not an issue with multicast RTP?

     Rob:    Should allow "destination" for Multicast ... I can describe a
use case for it...

     Jorg:   It is useful to have the control entity somewhere else than the
conference participants.

     Aravind: I can volunteer to write a proposal. We currently disable
"destination" header by default but
            customers can turn it on using a config parameter.

    2.3.4: Next meeting
	   Magnus:  Next meeting in two weeks?

           Everybody: sounds good.


    

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Subject: Re: [MMUSIC] RTSP & NAT: Symmetric RTP
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
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Magnus,

I count  2 issues left :

1) STUN and RTP/RTCP co-located:

I dont understand your concern here?

the STUN draft says: "A STUN  server can even be embedded within an end system."
I assume it means exactly that, no?

in practice it is easy for the server to discreminate between incomming STUN and incomming RTP or RTCP,
since STUN and RTCP have completely different header structures;
alos the STUN header contain a 128 bits nonce and the RTP/RTCP headers contain
at least one 32 bits nonce ....
So first the risk of collision is extremely small...
secondly a good implementation will always include structural/coherence checks, right ?

so what harm can it do that mis-formed RTCP packets would not cause
in the first place anyway ?


2) Sending RTP at a different IP address than the RTSP client is un-secure.

True.

Unfortunately this is a property of RTSP today ;-)

A) for NATs we simply want to extend it so that the RTP and RTCP
can have both different IP addresses instead of "only" different port numbers
> the additional danger is small compared to the benefit
and:

B) there is a need to fix that security hole _anyway_

One possible fix would be to define a RTCP extension whereby
before sending RTP data the server sends a "confirmation request"
with a Nonce and waits (with as short a time out as possible)
for a RTCP "confirmation responce" containing the same nonce and
originating from the correct target IP.

defeating that is pretty difficult (afaik it requires a man in the middle
in control of a router to trap the request and spoof the reply,
with a guy in that position your concerns are probably elsewhere !-)


regards,



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

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Subject: Re: [MMUSIC] RTSP and NATs
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: IETF MMUSIC WG <mmusic@ietf.org>
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magnus you wrote:


>Also all the clients need to reside on the same port number
> which creates a multiplexing nightmare.

why? if the server _chooses_ its ports and puts them in the SDP ?


(anyway demultiplexing using SSRC was already discussed:
it is maybe not always a very good idea but it is doable...)

regards,


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

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Subject: Re: [MMUSIC] RTSP & NAT: Symmetric RTP
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Hi Philippe,

Comments below:
philippe.gentric@philips.com wrote:

>Magnus,
>
>I count  2 issues left :
>
>1) STUN and RTP/RTCP co-located:
>
>I dont understand your concern here?
>  
>
>the STUN draft says: "A STUN  server can even be embedded within an end system."
>I assume it means exactly that, no?
>  
>

My concern is that it is not a clean, well engineered solution. However 
I can agree that it is possible to make it work. The second question is 
to Jonathan.

>in practice it is easy for the server to discreminate between incomming STUN and incomming RTP or RTCP,
>since STUN and RTCP have completely different header structures;
>alos the STUN header contain a 128 bits nonce and the RTP/RTCP headers contain
>at least one 32 bits nonce ....
>So first the risk of collision is extremely small...
>secondly a good implementation will always include structural/coherence checks, right ?
>
>so what harm can it do that mis-formed RTCP packets would not cause
>in the first place anyway ?
>

I just took to heart the security consideration chapter of STUN that 
recommends that STUN server should avoid having other services that can 
open security holes. So its just a question that the server needs to be 
well built in regards to security and handling of bad messages.

>
>
>2) Sending RTP at a different IP address than the RTSP client is un-secure.
>
>True.
>
>Unfortunately this is a property of RTSP today ;-)
>

Actually, if you look at RFC 2326 you can read the following in regards 
to destination parameter in the transport header:
  "To avoid becoming the unwitting perpetrator of a remote-
  controlled denial-of-service attack, a server SHOULD
  authenticate the client and SHOULD log such attempts before
  allowing the client to direct a media stream to an address not
  chosen by the server. This is particularly important if RTSP
  commands are issued via UDP, but implementations cannot rely
  on TCP as reliable means of client identification by itself. A
  server SHOULD not allow a client to direct media streams to an
  address that differs from the address commands are coming
  from."

So if this should is followed, i.e. do not allow use of destination. 
Then RTSP is not that unsafe.

>
>A) for NATs we simply want to extend it so that the RTP and RTCP
>can have both different IP addresses instead of "only" different port numbers
>  
>
>>the additional danger is small compared to the benefit
>>

This additional danger is way, way to big. You can't allow usage of 
client specified destination on any scale as it is such a potent DOS 
tool. a few messages can generate a multi Mbps attack on a target. So I 
don't see anyway of allowing such a security breach to be used.

>>    
>>
>and:
>
>B) there is a need to fix that security hole _anyway_
>  
>
>One possible fix would be to define a RTCP extension whereby
>before sending RTP data the server sends a "confirmation request"
>with a Nonce and waits (with as short a time out as possible)
>for a RTCP "confirmation responce" containing the same nonce and
>originating from the correct target IP.
>
>defeating that is pretty difficult (afaik it requires a man in the middle
>in control of a router to trap the request and spoof the reply,
>with a guy in that position your concerns are probably elsewhere !-)
>  
>

I think we should look at ways to fix the whole. Because there are also 
other usage's then NAT traversal to such a mechanism.

So from a security perspective I see the useful traversal methods are:
- STUN for single IP NATs, preferable cone NATs.
- Symmetric RTP for symmetric and multi-IP NATs.

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 Philippe,

Comments inline.

philippe.gentric@philips.com wrote:

>magnus you wrote:
>
>
>  
>
>>Also all the clients need to reside on the same port number
>>which creates a multiplexing nightmare.
>>    
>>
>
>why? if the server _chooses_ its ports and puts them in the SDP ?
>  
>
Sorry, was not making clear that I assumed a known port which does not 
require signaling or spending extra RTTs. If you would send out the 
ports within the SDP then it gets simpler. However as discussed in 
another mail, co-locating is not a clean well engineered solution.

>
>(anyway demultiplexing using SSRC was already discussed:
>it is maybe not always a very good idea but it is doable...)
>  
>
We have discussed for symmetric RTP to use SSRC as an random identifier 
ensuring that it not another user that tries to hijack the session. With 
symmetric RTP it is after all the server that gives the client what 
ports to connect to. That allow the server to place a single user on 
each port pair. This way we can avoid doing SSRC multiplexing of 
different RTP sessions.

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] RTSP and NATs
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: IETF MMUSIC WG <mmusic@ietf.org>
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Magnus,

symetric RTP and my other mail about "confirmation"
are the same spirit then no?

it is the principle that "the server sends only if it has received something",
and the something must be a nonce to make hacking harder:
and the SSRC can serve that purpose

 is that what you mean ?

regards,


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


                                                                                                                                              
                                                                                                                                              
                                                          To:   Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1                                     
                                                          cc:   IETF MMUSIC WG <mmusic@ietf.org>                                              
                                                          Subject:    Re: [MMUSIC] RTSP and NATs                                              
                                                                                                                                              
               Magnus Westerlund                          Classification:                                                                     
               <magnus.westerlund@era.ericsson                                                                                                
               .se>                                                                                                                           
                                                                                                                                              
               Sent by:                                                                                                                       
               mmusic-admin@ietf.org                                                                                                          
                                                                                                                                              
               12/02/2003 17:08                                                                                                               
                                                                                                                                              
                                                                                                                                              




Hi Philippe,

Comments inline.

philippe.gentric@philips.com wrote:

>magnus you wrote:
>
>
>
>
>>Also all the clients need to reside on the same port number
>>which creates a multiplexing nightmare.
>>
>>
>
>why? if the server _chooses_ its ports and puts them in the SDP ?
>
>
Sorry, was not making clear that I assumed a known port which does not
require signaling or spending extra RTTs. If you would send out the
ports within the SDP then it gets simpler. However as discussed in
another mail, co-locating is not a clean well engineered solution.

>
>(anyway demultiplexing using SSRC was already discussed:
>it is maybe not always a very good idea but it is doable...)
>
>
We have discussed for symmetric RTP to use SSRC as an random identifier
ensuring that it not another user that tries to hijack the session. With
symmetric RTP it is after all the server that gives the client what
ports to connect to. That allow the server to place a single user on
each port pair. This way we can avoid doing SSRC multiplexing of
different RTP sessions.

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 Philippe,

I get a bit confused by the below but I will try to answer stating my 
understanding of your questions.

philippe.gentric@philips.com wrote:

>Magnus,
>
>symetric RTP and my other mail about "confirmation"
>are the same spirit then no?
>
>it is the principle that "the server sends only if it has received something",
>and the something must be a nonce to make hacking harder:
>and the SSRC can serve that purpose
>
> is that what you mean ?
>  
>
Symmetric RTP and security challenge for the UDP flows are two slightly 
different things.

Symmetric RTP is based on that the client sends packet to the server 
that needs to have some protection against hijacking. That protection 
can be provided with SSRC of the client. This SSRC needs to be 
communicated between the client and server through other means (RTSP). 
This is not safe to a man in the middle attack.

Security challenge is different in some ways. This is my understanding 
of how it could work. In this case the client asks the server to send 
media to any single IP/port number. Before sending a lot of traffic to 
these ports, the server needs to ensure that these ports are prepared to 
receive this traffic. To ensure this we make a security challenge either 
from server to client or the other way around. For STUN and 
"destination" usage the client initiates the challenge. In case of 
symmetric RTP the client could send a response without the challenge to 
id it clearly. The goal is to ensure that the client agrees to receive 
the traffic. By having the receiver sign the message going from client 
to server with keys that are transported through other ways we can 
ensure that no other than the intended receiver can verify that it 
agrees. This makes it possible to protect also against man in the middle 
attacks. It will of course requires secure RTSP signaling.

The security challenge could as describe be taken into use also with 
symmetric RTP to identify that it is the correct receiver that sends 
these packets to use for binding. However this is still possible to 
attack and create an redirect by a man in the middle changing the 
packets source IP/port. By having a signed ACK we can at least ensure 
that the man in the middle can not do more than being a relay.

Conclusion is that the security challenge initiated from server towards 
client can be made safe assuming this mechanism. However symmetric RTP 
can never be safe against man in the middle attacks.

I hope that clarified the difference between the security challenge as a 
general mechanism and symmetric RTP as a special usage.

Best Regards

Magnus

-- 

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: [MMUSIC] RTSP teleconference 19 february - Agenda
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Hi,

Here is the agenda for next weeks RTSP teleconference at wednesday the 
19th of february at 18.00 CET.

Agenda:

1. Follow up on last meeting
- Redirect, waiting for new text from Tom
- Proxy-require, no comments seems okay to edit into spec. See message:
http://www.ietf.org/mail-archive/working-groups/mmusic/current/msg01167.html
- Update on Range header text in regards to Live streams. Need text 
proposal (Thomas Zeng).
-  Restriction text on usage of destination (Aravind Narasimhan)
- Move NAT & FW to separate draft, update intro of core draft to point 
this out (Magnus W).

2. Administrative
- Document Editor(s)!!!

3. Today's new Issues:
 
* RTSP and NATs revisited:
   - Discussion on MMUSIC WG list on how and when you can use STUN and 
Symmetric RTP. STUN for cone NATs which only uses single IP. Symmetric 
RTP requires RTSP and server changes + rules on how to prevent attacks 
using RTP.
   - Security problems. RTSP as Denial of Service tool. Hijacking and 
interception attacks.
   - Destination secure challenge mechanism?

* Use of transport connection interleaved data
   http://rtsp.org/bug507408
   http://rtsp.org/bug556589
   http://rtsp.org/bug662436
   http://rtsp.org/bug672160

* 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

Please be prepared if your participating.
If your not a member of rtsp-logistics and would like to participate 
please send me a mail.

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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From mailnull@www1.ietf.org  Fri Feb 14 08:26:36 2003
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Subject: [MMUSIC] Proposed Resolution on comedia and comedia-fix
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David, Jonathan,

We have reviewed the entire discussion again and have come to few concrete
proposals how we would like to proceed on this issue.  First of all, there
is merit in the current comedia proposal to proceed (quickly), but there is
also value in Jonathan's comments and observations.

What we would like to do is:

1.  Proceed using David's comedia-04 draft as a basis.

2.  Remove the entire discussion of, and attributes for, source IP
     addresses.  As there is no general way to determine when an IP
     addresss can be trusted, we believe its use is worthless, if not
     risky.

3.  Include discussion of the security consideration Jonathan raised
     into the revision to comedia-04.

4.  Include something like the eid: attribute as an optional indicator.
     We need to consider if a unique random indentifier is appropriate, or
     if an application specific identifier (e.g. RTP SSRC) might be better.

5.  Do not follow the comedia-fix idea of sending the eid: as the
     first couple of bytes over any TCP connection.  We believe this
     is going to cause problems with existing protocols and protocol
     implementations which do not natively send this information.

     Where necessary, for example when multiplexing several application
     layer connections into a single transport layer connection, the higher
     layer protocol must obviously already provide some framing structure.
     It is expected that such a structure can be used to tag the individual
     application layer connections and correlate them via the eid:.  In such
     a case, or if the application layer protocol supports flow identifiers,
     this can also be used for single-hop TCP connections. If the transport
     protocol supports additional identifiers they might also be used.

     Obviously, this requires writing up further details. Specific use of,
     or formats for, the eid: per application layer protocol might need to
     be defined.

     It is hoped that this will solve the connection re-use issue for
     certain applications while allowing other applications to continue
     unmodified.

6.  Port re-use on servers should preferably be dealt with on a per
     application basis.  The eid: will be useful for this purpose if
     an application protocol supports referencing it in some fashion.
     There is no need to generally break backwards compatibility with
     arbitrary protocols.

7.  The revised comedia document needs to specify that if eid: is included,
     it MUST be used by the application protocol. If the eid: is not
     present, then identifying incoming connections can only work on a
     receiver port basis (with all the negative implications identified by
     comedia-fix -- which should be included in a note).  New application
     protocol to be used with comedia should incorporate the necessary
     behavior.

Comments?

Cheers,
Colin and Joerg


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I disagree with #2, but so be it.  At this rate, all future IETF protocols 
are going to have use divining rods to determine where to send packets, 
since IP addresses are being banned because of NATs (insert half-smiley).

I agree with your proposal for eid:, but per the recent RTSP discussion I 
propose that eid be made its own draft, as it is arguably not intrinsic to 
connection-oriented media and therefore does not belong in comedia.  I have 
no problem with comedia referencing an externally-documented eid mechanism 
and applying the appropriate MUSTs listed below.


At 08:27 AM 2/14/2003, Joerg Ott wrote:
>David, Jonathan,
>
>We have reviewed the entire discussion again and have come to few concrete
>proposals how we would like to proceed on this issue.  First of all, there
>is merit in the current comedia proposal to proceed (quickly), but there is
>also value in Jonathan's comments and observations.
>
>What we would like to do is:
>
>1.  Proceed using David's comedia-04 draft as a basis.
>
>2.  Remove the entire discussion of, and attributes for, source IP
>     addresses.  As there is no general way to determine when an IP
>     addresss can be trusted, we believe its use is worthless, if not
>     risky.
>
>3.  Include discussion of the security consideration Jonathan raised
>     into the revision to comedia-04.
>
>4.  Include something like the eid: attribute as an optional indicator.
>     We need to consider if a unique random indentifier is appropriate, or
>     if an application specific identifier (e.g. RTP SSRC) might be better.
>
>5.  Do not follow the comedia-fix idea of sending the eid: as the
>     first couple of bytes over any TCP connection.  We believe this
>     is going to cause problems with existing protocols and protocol
>     implementations which do not natively send this information.
>
>     Where necessary, for example when multiplexing several application
>     layer connections into a single transport layer connection, the higher
>     layer protocol must obviously already provide some framing structure.
>     It is expected that such a structure can be used to tag the individual
>     application layer connections and correlate them via the eid:.  In such
>     a case, or if the application layer protocol supports flow identifiers,
>     this can also be used for single-hop TCP connections. If the transport
>     protocol supports additional identifiers they might also be used.
>
>     Obviously, this requires writing up further details. Specific use of,
>     or formats for, the eid: per application layer protocol might need to
>     be defined.
>
>     It is hoped that this will solve the connection re-use issue for
>     certain applications while allowing other applications to continue
>     unmodified.
>
>6.  Port re-use on servers should preferably be dealt with on a per
>     application basis.  The eid: will be useful for this purpose if
>     an application protocol supports referencing it in some fashion.
>     There is no need to generally break backwards compatibility with
>     arbitrary protocols.
>
>7.  The revised comedia document needs to specify that if eid: is included,
>     it MUST be used by the application protocol. If the eid: is not
>     present, then identifying incoming connections can only work on a
>     receiver port basis (with all the negative implications identified by
>     comedia-fix -- which should be included in a note).  New application
>     protocol to be used with comedia should incorporate the necessary
>     behavior.
>
>Comments?
>
>Cheers,
>Colin and Joerg
>
>
>_______________________________________________
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Subject: Re: [MMUSIC] RTSP and NATs
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: IETF MMUSIC WG <mmusic@ietf.org>
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Magnus you write:

>By having the receiver sign the message going from client
>to server with keys that are transported through other ways we can
>ensure that no other than the intended receiver can verify that it
>agrees. This makes it possible to protect also against man in the middle
>attacks. It will of course requires secure RTSP signaling.

I dont see how a "secure RTSP session" solves the case I was thinking about is:

your server thinks that the client is behind a NAT,
because the IP addresses for RTSP and RTP are different,
but in fact that client is a Bad Guy who wants to induce
your server to flood someone else.

if Mr Bad Guy is also a man-in-the-middle you are in real trouble because:

1) your attacker is initiating that RTSP session so

A) your server cannot  trust "a different IP address" indicated in that session, even if the session is TLS-ized  !

B) you cannot use that session to convey crypto stuff for a separate UDP challenge either  because:

2) this guys is also a man-in-the-middle who can trap/spoof UDP traffic
i.e. a challenge/response at that "different IP address".

I dont know how to solve that one (help security experts?)

regards,

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

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Thanks Philippe,

The attack you describe is possible unless we have a way of positively 
identify a entity from two ways. First we must be able to establish the 
identity of the host that are located at the given media destination IP 
address/port. Secondly that host must be able to prove that it is that 
entity.

In slightly more detail: When a client asks the server to send media to 
a given destination, the server must be able to get an global unique 
identifier of the destination from a trusted party. Then the server uses 
this global identifier in the security challenge to the destination. The 
destination answers the challenge and proves that it is really the given 
entity.

The first step of getting a global host entity based on IP will never 
work with an NATed address. It seem to be possible to make it work when 
you use an public IP address.

In basic any NAT traversal mechanism will allow for an attack by a man 
in the middle, see:
http://www.ietf.org/internet-drafts/draft-dupont-transient-pseudonat-01.txt

In fact today's RTSP is definitely not safe against man in the middle 
attacks. The man in the middle must only be able to receive the servers 
response to SETUP request to get the session ID: Then it can control the 
server from any place by simply spoof the sender address to that of the 
target. If you use TCP the attacker must be able to receive the TCP 
flows response. That it is all it takes to use RTSP as a DOS tool.

Best Regards

Magnus Westerlund

philippe.gentric@philips.com wrote:

>Magnus you write:
>
>  
>
>>By having the receiver sign the message going from client
>>to server with keys that are transported through other ways we can
>>ensure that no other than the intended receiver can verify that it
>>agrees. This makes it possible to protect also against man in the middle
>>attacks. It will of course requires secure RTSP signaling.
>>    
>>
>
>I dont see how a "secure RTSP session" solves the case I was thinking about is:
>
>your server thinks that the client is behind a NAT,
>because the IP addresses for RTSP and RTP are different,
>but in fact that client is a Bad Guy who wants to induce
>your server to flood someone else.
>
>if Mr Bad Guy is also a man-in-the-middle you are in real trouble because:
>
>1) your attacker is initiating that RTSP session so
>
>A) your server cannot  trust "a different IP address" indicated in that session, even if the session is TLS-ized  !
>
>B) you cannot use that session to convey crypto stuff for a separate UDP challenge either  because:
>
>2) this guys is also a man-in-the-middle who can trap/spoof UDP traffic
>i.e. a challenge/response at that "different IP address".
>
>I dont know how to solve that one (help security experts?)
>
>regards,
>
>Philippe Gentric
>Software Architect
>Philips MP4Net
>"philippe dot gentric at philips dot com"
>http://www.platform4.philips.com
>
>  
>

-- 

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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I agree with the proposal. Some followup on eid below.

David Yon wrote:
> I disagree with #2, but so be it.  At this rate, all future IETF 
> protocols are going to have use divining rods to determine where to send 
> packets, since IP addresses are being banned because of NATs (insert 
> half-smiley).

Just source addresses.

> 
> I agree with your proposal for eid:, but per the recent RTSP discussion 
> I propose that eid be made its own draft, as it is arguably not 
> intrinsic to connection-oriented media and therefore does not belong in 
> comedia.  I have no problem with comedia referencing an 
> externally-documented eid mechanism and applying the appropriate MUSTs 
> listed below.

At this point, I would actually agree with David here. If the way that 
eid is done is application specific anyway (I concede that sending it as 
the first bytes is not optimal and is better done application specific), 
then its encoding in SDP may as well be application specific too. The 
use of a separate draft would be primarily to outline the concept, I 
suppose. However, I don't see a need for comedia to have a dependency on it.

-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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FYI, I have submitted an updated version of the "SDP source filter" 
Internet draft.  Until it appears on the IETF web site, you an get a copy 
at <http://www.live.com/mmusic-sdp-srcfilter-03.txt>

The major change in this version is that (as a result of feedback from 
Magnus Westerlund), there is now just one attribute defined: 
"a=source-filter:", rather than two: "a=incl:" and "a=excl:".  I.e., the 
syntax is now:
	a=source-filter: <filter-mode> <filter-spec>
where <filter-mode> is either "incl" or "excl".

Also, I've removed the "Multiple Source and Destination Example", because 
it was inconsistent with the rest of the spec.

	Ross.

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From: Kreuter Ruediger <ruediger.kreuter@siemens.com>
To: "'avt@ietf.org'" <avt@ietf.org>, "'mmusic@ietf.org'" <mmusic@ietf.org>,
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Subject: [MMUSIC] updated "Clearmode" (64kbit/s unrestricted) Internet-draft
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FYI, 
I have submitted an update of the CLEARMODE Internet draft. You find it at
http://www.ietf.org/internet-drafts/draft-kreuter-avt-rtp-clearmode-01.txt

The major update, due to comments according the support of Nx64kbit/s
and similar standardization activities in ITU's SG 11, is the introduction of
Nx64kbit/s support. Two remarks to this issue:
1) 1x64kbit/s is used in current VoIP products and supported by most DSP 
    Providers. Therefore I would like to keep Nx64kbit/s with N>1 as an option.
2) We currently do not have a VoIP product supporting Nx64kbit/s, therefore 
    I introduced this merely as a discussion input.
    So if anybody has a working Nx64kbit/s solution I would like to get some 
    comments on the packatization method in that case. If nobody has a
    solution I will delete this feature from the next version of this draft.

Besides, the first draft initiated a controverse discussion, if this should be 
"audio/CLEARMODE" or e.g. "application/CLEARMODE".
IMHO none of the existing IANA content types really fits for CLEARMODE. 
Since 1x64kbit/s is also used for 7kHz voice (despite G.722) I kept "audio"
in this draft. 

Regards
   Ruediger
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Hi,

There is a new version of the Java SDP API available for review.

Thank you for your consideration.

Regards,

Kelvin R. Porter

-----Original Message-----
From: Harold Ogle [mailto:harold.ogle@SUN.COM]
Sent: Thursday, February 13, 2003 12:35 PM
To: JCP-INTEREST@JAVA.SUN.COM
Subject: JSR 141: Public Review 2


The Expert Group for

    JSR-000141 SDP API

has released a second public review draft, which is now available from

    http://jcp.org/aboutJava/communityprocess/review/jsr141/index2.html

This public review closes on 15 March 2003.

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--NextPart

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

	Title		: Session Description Protocol (SDP) Source Filters
	Author(s)	: B. Quinn, R. Finlayson
	Filename	: draft-ietf-mmusic-sdp-srcfilter-03.txt
	Pages		: 10
	Date		: 2003-2-18
	
This document describes how to adapt the Session Description Protocol
(SDP) to express one or more source addresses as a source filter for
one or more destination 'connection' addresses.  It defines the syntax
and semantics for an SDP 'source-filter' attribute that may reference
either IPv4 or IPv6 address(es) as either an inclusive or exclusive
source list for either multicast or unicast destinations.  In particular,
an inclusive source-filter can be used to specify a Source-Specific
Multicast (SSM) session.

Receiver applications are expected use the SDP source-filter
information to identify traffic from legitimate senders and discard
traffic from illegitimate senders.  Applications and hosts may also
share the source-filter information with network elements (e.g., with
routers using IGMPv3) so they can potentially perform the traffic
filtering operation further 'upstream,' closer to the source(s).

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Subject: [MMUSIC] Destination field in Transport Header - suggested text
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Hey Folks,


Here is my attempt to clarify the text for the destination field of the 
transport header.  The action item was from the Feb 05, 03' conference 
call where the idea of having authorization form the recipient of the 
stream was proposed.  The current text in draft-02 is:

>     destination: The address to which a stream will be sent.  The       |
>          client may specify the destination address with the destina-   |
>          tion parameter. To avoid becoming the unwitting perpetrator of |
>          a remote-controlled denial-of-service attack, a server SHOULD  |
>          authenticate the client and SHOULD log such attempts before    |
>          allowing the client to direct a media stream to an address not |
>          chosen by the server. This is particularly important if RTSP   |
>          commands are issued via UDP, but implementations cannot rely   |
>          on TCP as reliable means of client identification by itself.   |
>          IPv6 addresses is RECOMMENDED to be given as fully qualified   |
>          domain to make it backwards compatible with RFC 2326 implemen- |
>          tations.                                                       |
>

The new suggested text to be placed in Section 12.40, p 75:


    destination:  The address of the stream recipient to which a stream  |
          will be sent. The client originating the RTSP request may      |
          specify the destination address of the stream recipient with   |
          the destination parameter.  When the destination field is      |
          specified, the recipient may be a different party than the     |
          originator of the request. To avoid becoming the unwitting     |
          perpetrator of a remote-controlled denial-of-service attack, a |
          server SHOULD authenticate the client originating the request  |
          and SHOULD log such attempts before allowing the client to     |
          direct a media stream to a recipient address not chosen by the |
          server. While, this is particularly important if RTSP commands |
          are issued via UDP, implementations cannot rely on TCP as      |
          reliable means of client identification by itself either.      |
                                                                         |
          The server SHOULD NOT allow the destination field to be set    |
          unless a mechanism exists in the system to authorize the       |
          request originator to direct streams to the recipient.  It is  |
          preferred that this authorization be performed by the          |
          recipient itself and the credentials passed along to the       |
          server.  However, in certain cases, such as when recipient     |
          address is a multicast group, or when the recipient is unable  |
          to communicate with the server in an out-of-band manner, this  |
          may not be possible.  In these cases server may chose another  |
          method such as a server-resident authorization list to ensure  |
          that the request originator has the proper credentials to      |
          request stream delivery to the recipient.                      |
                                                                         |
          IPv6 addresses are RECOMMENDED to be given as fully qualified  |
          domain to make it backwards compatible with RFC 2326 implemen- |
          tations.                                                       |
          

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From: Sasha Vainshtein <Sasha@AXERRA.com>
To: "'Kreuter Ruediger'" <ruediger.kreuter@siemens.com>,
        "'avt@ietf.org'"
	 <avt@ietf.org>,
        "'mmusic@ietf.org'" <mmusic@ietf.org>,
        "'pwe3@ietf.org'"
	 <pwe3@ietf.org>
Cc: "'kschong@att.com'" <kschong@att.com>,
        "'taylor@nortelnetworks.com'"
	 <taylor@nortelnetworks.com>,
        Alik Shimelmits <alik@AXERRA.com>, Akiva Sadovski <akiva@AXERRA.com>,
        Israel Sasson <israel@AXERRA.com>
Date: Wed, 19 Feb 2003 18:50:28 +0200
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Subject: [MMUSIC] RE: [PWE3] updated "Clearmode" (64kbit/s unrestricted) Internet-d
 raft
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Ruediger and all,
Please take a look at
http://www.ietf.org/internet-drafts/draft-vainshtein-cesopsn-05.txt ,
Section 6.6.2. 
It seems that the approach and the resulting format are pretty similar.


With best regards,
                                   Sasha Vainshtein
email:     sasha@axerra.com <mailto:sasha@axerra.com> 
tel:       +972-3-7659993 (office)
           +972-8-9254948 (res.)
           +972-58-674833 (cell.)
 


> -----Original Message-----
> From: Kreuter Ruediger [mailto:ruediger.kreuter@siemens.com]
> Sent: Tuesday, February 18, 2003 12:29 PM
> To: 'avt@ietf.org'; 'mmusic@ietf.org'; 'pwe3@ietf.org'
> Cc: 'kschong@att.com'; 'taylor@nortelnetworks.com'
> Subject: [PWE3] updated "Clearmode" (64kbit/s unrestricted) 
> Internet-draft
> 
> 
> FYI, 
> I have submitted an update of the CLEARMODE Internet draft. 
> You find it at
> http://www.ietf.org/internet-drafts/draft-kreuter-avt-rtp-clea
> rmode-01.txt
> 
> The major update, due to comments according the support of Nx64kbit/s
> and similar standardization activities in ITU's SG 11, is the 
> introduction of
> Nx64kbit/s support. Two remarks to this issue:
> 1) 1x64kbit/s is used in current VoIP products and supported 
> by most DSP 
>     Providers. Therefore I would like to keep Nx64kbit/s with 
> N>1 as an option.
> 2) We currently do not have a VoIP product supporting 
> Nx64kbit/s, therefore 
>     I introduced this merely as a discussion input.
>     So if anybody has a working Nx64kbit/s solution I would 
> like to get some 
>     comments on the packatization method in that case. If nobody has a
>     solution I will delete this feature from the next version 
> of this draft.
> 
> Besides, the first draft initiated a controverse discussion, 
> if this should be 
> "audio/CLEARMODE" or e.g. "application/CLEARMODE".
> IMHO none of the existing IANA content types really fits for 
> CLEARMODE. 
> Since 1x64kbit/s is also used for 7kHz voice (despite G.722) 
> I kept "audio"
> in this draft. 
> 
> Regards
>    Ruediger
> _______________________________________________
> pwe3 mailing list
> pwe3@ietf.org
> https://www1.ietf.org/mailman/listinfo/pwe3
> 
_______________________________________________
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From: Tom Marshall <tmarshall@real.com>
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Here's the new diff that incorporates the action items requested.  Wording
for actions required by the proxy was added and the Location header was made
optional for all 3rr responses (although a 3rr response to a REDIRECT is
probably not such a good idea in the first place...)

--=20
Pardo's First Postulate: Anything good in life is either illegal, immoral,
or fattening.  Arnold's Addendum: Everything else causes cancer in rats.

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--- draft-rfc2326bis-02.txt	2002-11-06 14:34:00.000000000 -0800
+++ draft-rfc2326bis-02-mod.txt	2003-02-19 16:58:20.000000000 -0800
@@ -1202,7 +1202,7 @@
      + 2xx: Success - The action was successfully received, understood,
        and accepted
=20
-     + 3xx: Redirection - Further action must be taken in order to com-
+     + 3rr: Redirection - Further action must be taken in order to com-
        plete the request
=20
      + 4xx: Client Error - The request contains bad syntax or cannot be
@@ -2417,13 +2417,27 @@
=20
 10.10 REDIRECT
=20
-   A redirect request informs the client that it MUST connect to another |
-   server location. REDIRECT SHALL only be sent to the client who cur-   |
-   rently has a session at the server. The REDIRECT request MAY contain  |
-   the header Location, which indicates that the client should issue     |
-   requests for that URL. If the Location URL only contains a host       |
-   address the client shall connect to the given host, while using the   |
-   path from the URL on the current server.                              |
+   A redirect request informs the client that it MUST connect to another
+   server location. The REDIRECT request MAY contain the header Location,
+   which indicates that the client should issue requests for that URL. If
+   the Location URL only contains a host address the client shall connect =
to
+   the given host, while using the path from the URL on the current server.
+
+   If a REDIRECT request contains a Session header, it is end-to-end and
+   applies only to the given session.  If there are proxies in the request
+   chain, they SHOULD NOT disconnect the control channel unless there are
+   no remaining sessions.
+
+   If a REDIRECT request does not contain a Session header, it is next-hop
+   and applies to the control connection.  The Location header SHOULD only
+   contain a host address.  If there are proxies in the request chain, they
+   SHOULD do all of the following: (1) respond to the REDIRECT request, (2)
+   disconnect the control channel from the requestor, (3) reconnect to the
+   given host address, and (4) pass the request to each applicable client
+   (typically those clients with an active session or unanswered request
+   from the requestor).  Note that the proxy is responsible for accepting
+   the REDIRECT response from its clients and these responses MUST NOT be
+   passed on to either the requesting or the destination server.
=20
    The redirect request MAY contain the header Range, which indicates    |
    when the redirection takes effect. If the Range contains a time=3D     =
 |
@@ -2436,11 +2450,6 @@
    time. The range value MUST be a single value in the open ended form,  |
    e.g. npt=3D59-.                                                        =
 |
=20
-   If a Session header is included in the REDIRECT request the client    |
-   MUST redirect the indicated session. If no Session header is included |
-   the client MUST redirect all sessions that it have on the server      |
-   sending the request.                                                  |
-
    If the client wants to continue to send or receive media for this     |
    resource, the client MUST issue a TEARDOWN request for the current    |
    session. A new session must be established with the designated host.  |
@@ -2647,7 +2656,11 @@
    processes on the server may be consuming storage space simultane-
    ously, a client should take this only as an estimate.
=20
-11.3 Redirection 3xx
+11.3 Redirection 3rr
+
+   The notation "3rr" indicates response codes from 300 to 399 inclusive
+   which are meant for redirection.  The response code 304 is excluded
+   from this set, as it is not used for redirection.
=20
    See [H10.3] for definition of status code 300 to 305. However com-    |
    ments are given for some to how they apply to RTSP. Further a couple  |
@@ -2679,6 +2692,24 @@
=20
 11.3.5 304 Not Modified                                                  |
=20
+   If the client has performed a conditional DESCRIBE or SETUP (see 12.23)
+   and the requested resource has not been modified, the server SHOULD send
+   a 304 response.  This response MUST NOT contain a message-body.
+
+   The response MUST include the following header fields:
+
+   - Date
+   - ETag and/or Content-Location, if the header would have been sent in
+     a 200 response to the same request.
+   - Expires, Cache-Control, and/or Vary, if the field-value might
+     differ from that sent in any previous response for the same variant.
+
+   This response is independent for the DESCRIBE and SETUP requests.  That
+   is, a 304 response to DESCRIBE does NOT imply that the resource content
+   is unchanged and a 304 response to SETUP does NOT imply that the resour=
ce
+   description is unchanged.  The ETag and If-Match headers may be used to
+   link the DESCRIBE and SETUP in this manner.
+
 11.3.6 305 Use Proxy                                                     |
=20
    See [H10.3.6].                                                        |
@@ -2948,7 +2979,7 @@
=20
    The From, Location, and RTP-Info header fields contain a URI. If the
    URI contains a comma, or semicolon, the URI MUST be enclosed in dou-
-   ble quotas ("). Any URI parameters are contained within these quotas.
+   ble quotes ("). Any URI parameters are contained within these quotas.
    If the URI is not enclosed in double quotas, any semicolon- delimited
    parameters are header-parameters, not URI parameters.
=20
@@ -3011,7 +3042,7 @@
    If-Match              R      r    -   -    o    -     -   -
    If-Modified-Since     R      r    o   -    o    -     -   -
    Last-Modified         r      r    o   -    -    -     -   -
-   Location             3xx          o   -    o    -     -   -
+   Location             3rr          o   o    o    o     o   o
    Proxy-Authenticate   407    amr   m   m    m    m     m   m
    Proxy-Require         R     ar    o   o    o    o     o   o
    Public                r    admr   -  m*    -    -     -   -
@@ -3028,7 +3059,7 @@
 Internet Draft                    RTSP                 November 01, 2002
=20
=20
-   Retry-After        3xx,503        o   o    o    -     -   -
+   Retry-After        3rr,503        o   o    o    -     -   -
    RTP-Info              r           -   -    o    m     -   -
    Scale                             -   -    -    o     -   -
    Session               R           -   o    o    m     m   m
@@ -3118,6 +3149,7 @@
    Host                              o   o   o   o   o   o
    Last-Modified         R      r    -   -   o   -   -   -
    Last-Modified         r      r    o   -   -   -   -   -
+   Location             3rr          o   o   o   o   o   o
    Location              R           -   -   -   -   m   -
    Proxy-Authenticate   407    amr   m   m   m   m   m   m
    Proxy-Require         R     ar    o   o   o   o   o   o
@@ -3125,7 +3157,7 @@
    Range                 R           -   -   -   -   o   -
    Referer               R           o   o   o   o   o   -
    Require               R           o   o   o   o   o   o
-   Retry-After        3xx,503        o   o   -   -   -   -
+   Retry-After        3rr,503        o   o   -   -   -   -
    Scale                             -   -   -   o   -   -
    Session               R           o   o   m   m   o   m
    Session               r           c   c   m   m   o   m
@@ -5830,8 +5862,9 @@
=20
    type of error occur the appropriate response code MUST be sent. If    |
    the response code is a 4xx the session state is unchanged. A response |
-   code of 3xx will result in that the session is ended and its state is |
-   changed to Init. However there exist restrictions to when a 3xx       |
+   code of 3rr will result in that the session is ended and its state is |
+   changed to Init. A response code of 304 results in no state change.
+   However there exist restrictions to when a 3xx       |
    response may be used. A 5xx response SHALL not result in any change   |
    of the session state, except if the error is not possible to recover  |
    from. A unrecoverable error SHALL result the ending of the session.   |
@@ -5855,7 +5888,7 @@
=20
    Event         Prerequisite      Response
    -----------------------------------------------------------------
-   DESCRIBE      Needs REDIRECT    3xx Redirect
+   DESCRIBE      Needs REDIRECT    3rr Redirect
    DESCRIBE                        200, Session description
    OPTIONS       Session ID        200, Reset session timeout timer
    OPTIONS                         200
@@ -5888,20 +5921,20 @@
             Action  Requisite       New State  Response
             -------------------------------------------------
             SETUP                     Ready    NRM=3D1, RP=3D0.0
-            SETUP   Needs Redirect    Init     3xx Redirect
+            SETUP   Needs Redirect    Init     3rr Redirect
=20
=20
    Table 7: State: Init
=20
    state variables are also set by a correct request. This table also    |
    shows that a correct SETUP can in some cases be redirected to another |
-   URL and/or server by a 3xx response.                                  |
+   URL and/or server by a 3rr response.                                  |
=20
=20
        Action         Requisite       New State  Response
        ----------------------------------------------------------
        SETUP                            Ready    NRM=3D1,RP=3D0.0
-       SETUP          Needs Redirect    Init     3xx
+       SETUP          Needs Redirect    Init     3rr
        TEARDOWN       URL=3D*             Init     No session hdr.
        Timeout                          Init
        S->C:REDIRECT  Range hdr       Ready-nm   Set RedP
@@ -7187,7 +7220,7 @@
 11.1.1     100 Continue  . . . . . . . . . . . . . . . . . . . . . .  47
 11.2       Success 2xx . . . . . . . . . . . . . . . . . . . . . . .  48
 11.2.1     250 Low on Storage Space  . . . . . . . . . . . . . . . .  48
-11.3       Redirection 3xx . . . . . . . . . . . . . . . . . . . . .  48
+11.3       Redirection 3rr . . . . . . . . . . . . . . . . . . . . .  48
 11.3.1     TBW . . . . . . . . . . . . . . . . . . . . . . . . . . .  48
 11.3.2     301 Moved Permanently . . . . . . . . . . . . . . . . . .  48
 11.3.3     302 Found . . . . . . . . . . . . . . . . . . . . . . . .  48

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From: Thomas Zeng <zeng@PacketVideo.COM>
To: "'Tom Marshall'" <tmarshall@real.com>, mmusic@ietf.org
Subject: RE: [MMUSIC] REDIRECT diff, v2
Date: Wed, 19 Feb 2003 17:55:43 -0800
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Tom:

What about the resolution from a couple of RTSP telecon ago to replace
END_OF_STREAM with REDIRECT to the same server?

I don't see it in this diff file. Also it would be nice to see the new
section 10.10 in its final form.

thanks,
thomas

-----Original Message-----
From: Tom Marshall [mailto:tmarshall@real.com]
Sent: Wednesday, February 19, 2003 5:06 PM
To: mmusic@ietf.org
Subject: [MMUSIC] REDIRECT diff, v2


Here's the new diff that incorporates the action items requested.  Wording
for actions required by the proxy was added and the Location header was made
optional for all 3rr responses (although a 3rr response to a REDIRECT is
probably not such a good idea in the first place...)

-- 
Pardo's First Postulate: Anything good in life is either illegal, immoral,
or fattening.  Arnold's Addendum: Everything else causes cancer in rats.
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On Wed, Feb 19, 2003 at 05:55:43PM -0800, Thomas Zeng wrote:
> Tom:
>=20
> What about the resolution from a couple of RTSP telecon ago to replace
> END_OF_STREAM with REDIRECT to the same server?

That discussion started from a proposal by Magnus, so I figured he would
write that part.  I can do it if desired.

> I don't see it in this diff file. Also it would be nice to see the new
> section 10.10 in its final form.

The patch utility will produce the final form.

--=20
Psychiatry is the care of the id by the odd.
        -- Anonymous

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All,

Note the draft below. It is related to Internet Media Guides.

Regards,
 Toni=20

> -----Original Message-----
> From: ext Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
> Sent: 19 February, 2003 17:05
> Subject: I-D ACTION:draft-stallard-mmusic-dvb-thoughts-on-sds-00.txt
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> 	Title		: DVB thoughts on Service Discovery and Selection
> 	Author(s)	: P. Stallard, T. Paila
> 	Filename	: draft-stallard-mmusic-dvb-thoughts-on-sds-00.txt
> 	Pages		: 10
> 	Date		: 2003-2-18
> =09
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> Digital Video Broadcasting (DVB) group.
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Hi Thomas,

Thomas Zeng wrote:

>Tom:
>
>What about the resolution from a couple of RTSP telecon ago to replace
>END_OF_STREAM with REDIRECT to the same server?
>  
>
Just to avoid confusion with naming.

I assume you mean the problem of doing server initiated teardowns.
 From the 22 January notes:

Resolution: use redirect to tell the clients to teardown and reconnect to
the same server. Then tell the clients that
  service is not available via perhaps a info page. That way we have good
backwards compatibility and no 
  need for TEARDOWN S->C nor END_OF_STREAM.


And please Tom I would be thankful if you would include this in your 
text write up.

We have another problem that also has been refered to as END_OF_STREAM. 
This is the problem that a client does not know when the last media 
packets part of the current playout range has been received. Therefore 
one proposal has been to send a END_OF_PLAYOUT request from the server 
to client to tell it that is near and what rtp packets will be the last 
ones.

-- 

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		: SDPng Transition
	Author(s)	: J. Ott, C. Perkins
	Filename	: draft-ietf-mmusic-sdpng-trans-03.txt
	Pages		: 15
	Date		: 2003-2-19
	
The Session Description Protocol (SDP) is today widely used in the
Internet to announce as well as negotiate multimedia sessions and
exchange capabilities.  Having originally been designed for session
announcements only, as opposed to announcements and capabilities
negotiation announcements, native SDP lacks numerous features to be
applicable in many session scenarios.  Numerous extensions have been
developed to circumvent SDP's shortcomings -- but they have also
repeatedly shown its inherent limitations.  A successor protocol --
termed 'SDPng' for the time being -- is developed to address the
aforementioned needs of Internet applications in a more structured
manner.  With the huge installed base of SDP-based applications, a
migration path needs to be developed to move from SDP to SDPng over
time.  This document outlines how this migration can be achieved: in
general as well as for the various IETF control protocols that
potentially make use of SDP and SDPng.

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Date: Thu, 20 Feb 2003 09:54:27 -0800 (PST)
From: Aaron Colwell <acolwell@real.com>
To: mmusic@ietf.org
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Subject: [MMUSIC] Relative URLs in RTP-Info field
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Hi,

In some inter-op testing that I've done, I've discovered an ambiguous
situation related to SDP initiated playback and relative URLs in the PLAY
response.

Say I have the following SDP
v=0
a=control:rtsp://foo.com/blah.3gp
m=audio 0 RTP/AVP 96
a=control:rtsp://foo.com/blah.3gp/trackID=1
a=rtpmap:96 MP4A-LATM/44100
a=range:npt=0-37
m=video 0 RTP/AVP 97
a=control:rtsp://foo.com/blah.3gp/trackID=2
a=rtpmap:97 MP4V-ES/90000

The RTSP chat sequence looks something like
OPTIONS rtsp://foo.com
SETUP rtsp://foo.com/blah.3gp/trackID=1
SETUP response
SETUP rtsp://foo.com/blah.3gp/trackID=2
SETUP response
PLAY rtsp://foo.com/blah.3gp
PLAY response
 RTP-Info: url=trackID=1;seq=0;rtptime=0,url=trackID=2;seq=2234;rtptime=93875

Notice that the URLs in the RTP-Info field are relative. In this case, I
don't have a base URL to apply the relative URLs to. This is because I did
not issue a DESCRIBE and therefore did not get a Content-Base or
Content-Location field from a DESCRIBE response.

What is the proper thing to do in this case? Since I don't have a base URL
to apply the relative URLs to, I can't figure out which SETUP URL the
RTP-Info is referring to. In this particular example I would be able to
figure it out by applying the relative URLs to each SETUP URL and then see
if the resulting URL matches the original URL. That solution won't work
for all cases though. Here is an example of where that won't work.

You have an SDP that looks like..
v=0
a=control:rtsp://foo.com/movie
m=audio 0 RTP/AVP 96
a=control:rtsp://foo.com/audio.3gp/trackID=1
a=rtpmap:96 MP4A-LATM/44100
a=range:npt=0-37
m=video 0 RTP/AVP 97
a=control:rtsp://foo.com/video.3gp/trackID=1
a=rtpmap:97 MP4V-ES/90000

The RTSP chat for this situation would look like
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=trackID=1;seq=0;rtptime=0,url=trackID=1;seq=234;rtptime=23487

In this case I have no way to figure out what info is associated with
which stream.

I think part of the problem here is that the server is assuming that the
client knows what the base URL is. In the case of an SDP initiated
playback, I don't have one. To my knowledge there is no way that the
server can detect whether I have a base URL or not. I suppose it could
check to see if I issued a DESCRIBE or not, but that seems a little messy
to me.

I propose that the RTP-Info field should contain the SETUP URLs exactly
the way that they were given to the server in the SETUP. This allows the
client an easy way to figure out what URLs are associated with which
streams. Relative URLs should not be allowed since it is ambiguous as to
what they are relative to.

I'm open to any questions or comments.

Aaron

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Subject: Re: [MMUSIC] Relative URLs in RTP-Info field
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Hi Aaron,

I will try to provide answers to your questions.

Aaron Colwell wrote:

>Hi,
>
>In some inter-op testing that I've done, I've discovered an ambiguous
>situation related to SDP initiated playback and relative URLs in the PLAY
>response.
>
>Say I have the following SDP
>v=0
>a=control:rtsp://foo.com/blah.3gp
>m=audio 0 RTP/AVP 96
>a=control:rtsp://foo.com/blah.3gp/trackID=1
>a=rtpmap:96 MP4A-LATM/44100
>a=range:npt=0-37
>m=video 0 RTP/AVP 97
>a=control:rtsp://foo.com/blah.3gp/trackID=2
>a=rtpmap:97 MP4V-ES/90000
>
>The RTSP chat sequence looks something like
>OPTIONS rtsp://foo.com
>SETUP rtsp://foo.com/blah.3gp/trackID=1
>SETUP response
>SETUP rtsp://foo.com/blah.3gp/trackID=2
>SETUP response
>PLAY rtsp://foo.com/blah.3gp
>PLAY response
> RTP-Info: url=trackID=1;seq=0;rtptime=0,url=trackID=2;seq=2234;rtptime=93875
>
>Notice that the URLs in the RTP-Info field are relative. In this case, I
>don't have a base URL to apply the relative URLs to. This is because I did
>not issue a DESCRIBE and therefore did not get a Content-Base or
>Content-Location field from a DESCRIBE response.
>

The relative URLs are having the base URL of the Request URL. It might 
not be very clear about how to derive the base URL for the RTP-info 
header. It is something we need to clarify in the updated RTSP spec. 
However the base URL rules that are defined for SDP base URL says that 
the request URL should be used if not Content-Base or Content-Location 
is defined in the response. So in the above case they are actually 
erroneous. The RTP info header should have the following look:

 RTP-Info: url=blah.3gp/trackID=1;seq=0;rtptime=0,url=blah.3gp/trackID=2;seq=2234;rtptime=93875


>
>What is the proper thing to do in this case? Since I don't have a base URL
>to apply the relative URLs to, I can't figure out which SETUP URL the
>RTP-Info is referring to. In this particular example I would be able to
>figure it out by applying the relative URLs to each SETUP URL and then see
>if the resulting URL matches the original URL. That solution won't work
>for all cases though. Here is an example of where that won't work.
>  
>

As said before the relative URLs uses the Request URL as base, i.e. the 
PLAY request for the RTP-Info header. Please also note that normal URL 
handling rules apply, e.g. the last path segment at the end of the URL 
is removed.

>You have an SDP that looks like..
>v=0
>a=control:rtsp://foo.com/movie
>m=audio 0 RTP/AVP 96
>a=control:rtsp://foo.com/audio.3gp/trackID=1
>a=rtpmap:96 MP4A-LATM/44100
>a=range:npt=0-37
>m=video 0 RTP/AVP 97
>a=control:rtsp://foo.com/video.3gp/trackID=1
>a=rtpmap:97 MP4V-ES/90000
>
>The RTSP chat for this situation would look like
>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=trackID=1;seq=0;rtptime=0,url=trackID=1;seq=234;rtptime=23487
>
>In this case I have no way to figure out what info is associated with
>which stream.
>

Yes, you have. By using the PLAY request's URL as base the RTP-Info URLs 
would be:

RTP-Info: url=audio.3gp/trackID=1;seq=0;rtptime=0, url=video.3gp/trackID=1;seq=234;rtptime=23487


>
>I think part of the problem here is that the server is assuming that the
>client knows what the base URL is. In the case of an SDP initiated
>playback, I don't have one. To my knowledge there is no way that the
>server can detect whether I have a base URL or not. I suppose it could
>check to see if I issued a DESCRIBE or not, but that seems a little messy
>to me.
>

There must always be clear what is the base-URL. For RTSP request you 
can't use base-urls that are remembered each request - response pair 
must be self contained in this regard.

>
>I propose that the RTP-Info field should contain the SETUP URLs exactly
>the way that they were given to the server in the SETUP. This allows the
>client an easy way to figure out what URLs are associated with which
>streams. Relative URLs should not be allowed since it is ambiguous as to
>what they are relative to.
>

For RTP-Info you either use relative URLs to the request URL or use 
absolute ones. Unless anyone argues against it I will add the following 
sentence to the RTP-Info URL description to clarify this: "Any relative 
URL SHALL use the request URL as base URL."

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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From: Tom Marshall <tmarshall@real.com>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: Aaron Colwell <acolwell@real.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] Relative URLs in RTP-Info field
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> >You have an SDP that looks like..
> >v=3D0
> >a=3Dcontrol:rtsp://foo.com/movie
> >m=3Daudio 0 RTP/AVP 96
> >a=3Dcontrol:rtsp://foo.com/audio.3gp/trackID=3D1
> >a=3Drtpmap:96 MP4A-LATM/44100
> >a=3Drange:npt=3D0-37
> >m=3Dvideo 0 RTP/AVP 97
> >a=3Dcontrol:rtsp://foo.com/video.3gp/trackID=3D1
> >a=3Drtpmap:97 MP4V-ES/90000
> >
> >The RTSP chat for this situation would look like
> >OPTIONS rtsp://foo.com
> >SETUP rtsp://foo.com/audio.3gp/trackID=3D1
> >SETUP response
> >SETUP rtsp://foo.com/video.3gp/trackID=3D1
> >SETUP response
> >PLAY rtsp://foo.com/movie
> >PLAY response
> >RTP-Info: url=3DtrackID=3D1;seq=3D0;rtptime=3D0,url=3DtrackID=3D1;seq=3D=
234;rtptime=3D23487
> >
> >In this case I have no way to figure out what info is associated with
> >which stream.
> >
>=20
> Yes, you have. By using the PLAY request's URL as base the RTP-Info URLs=
=20
> would be:
>=20
> RTP-Info: url=3Daudio.3gp/trackID=3D1;seq=3D0;rtptime=3D0,=20
> url=3Dvideo.3gp/trackID=3D1;seq=3D234;rtptime=3D23487

You are making an assumption about the ordering here -- that the first url=
=3D
corresponds to the first SETUP.  Is that required?

--=20
Most people are too busy to have time for anything important.

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Date: Fri, 21 Feb 2003 09:43:13 -0800 (PST)
From: Aaron Colwell <acolwell@real.com>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
cc: mmusic@ietf.org
Subject: Re: [MMUSIC] Relative URLs in RTP-Info field
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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. If an absolute URL is
 specified, it MUST match the URL used to SETUP that stream."

I'm sure this could be worded a little better and put into proper
RFC-speak.

Thanks,
Aaron

On Fri, 21 Feb 2003, Magnus Westerlund wrote:

> Hi Aaron,
>
> I will try to provide answers to your questions.
>
> Aaron Colwell wrote:
>
> >Hi,
> >
> >In some inter-op testing that I've done, I've discovered an ambiguous
> >situation related to SDP initiated playback and relative URLs in the PLAY
> >response.
> >
> >Say I have the following SDP
> >v=0
> >a=control:rtsp://foo.com/blah.3gp
> >m=audio 0 RTP/AVP 96
> >a=control:rtsp://foo.com/blah.3gp/trackID=1
> >a=rtpmap:96 MP4A-LATM/44100
> >a=range:npt=0-37
> >m=video 0 RTP/AVP 97
> >a=control:rtsp://foo.com/blah.3gp/trackID=2
> >a=rtpmap:97 MP4V-ES/90000
> >
> >The RTSP chat sequence looks something like
> >OPTIONS rtsp://foo.com
> >SETUP rtsp://foo.com/blah.3gp/trackID=1
> >SETUP response
> >SETUP rtsp://foo.com/blah.3gp/trackID=2
> >SETUP response
> >PLAY rtsp://foo.com/blah.3gp
> >PLAY response
> > RTP-Info: url=trackID=1;seq=0;rtptime=0,url=trackID=2;seq=2234;rtptime=93875
> >
> >Notice that the URLs in the RTP-Info field are relative. In this case, I
> >don't have a base URL to apply the relative URLs to. This is because I did
> >not issue a DESCRIBE and therefore did not get a Content-Base or
> >Content-Location field from a DESCRIBE response.
> >
>
> The relative URLs are having the base URL of the Request URL. It might
> not be very clear about how to derive the base URL for the RTP-info
> header. It is something we need to clarify in the updated RTSP spec.
> However the base URL rules that are defined for SDP base URL says that
> the request URL should be used if not Content-Base or Content-Location
> is defined in the response. So in the above case they are actually
> erroneous. The RTP info header should have the following look:
>
>  RTP-Info: url=blah.3gp/trackID=1;seq=0;rtptime=0,url=blah.3gp/trackID=2;seq=2234;rtptime=93875
>
>
> >
> >What is the proper thing to do in this case? Since I don't have a base URL
> >to apply the relative URLs to, I can't figure out which SETUP URL the
> >RTP-Info is referring to. In this particular example I would be able to
> >figure it out by applying the relative URLs to each SETUP URL and then see
> >if the resulting URL matches the original URL. That solution won't work
> >for all cases though. Here is an example of where that won't work.
> >
> >
>
> As said before the relative URLs uses the Request URL as base, i.e. the
> PLAY request for the RTP-Info header. Please also note that normal URL
> handling rules apply, e.g. the last path segment at the end of the URL
> is removed.
>
> >You have an SDP that looks like..
> >v=0
> >a=control:rtsp://foo.com/movie
> >m=audio 0 RTP/AVP 96
> >a=control:rtsp://foo.com/audio.3gp/trackID=1
> >a=rtpmap:96 MP4A-LATM/44100
> >a=range:npt=0-37
> >m=video 0 RTP/AVP 97
> >a=control:rtsp://foo.com/video.3gp/trackID=1
> >a=rtpmap:97 MP4V-ES/90000
> >
> >The RTSP chat for this situation would look like
> >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=trackID=1;seq=0;rtptime=0,url=trackID=1;seq=234;rtptime=23487
> >
> >In this case I have no way to figure out what info is associated with
> >which stream.
> >
>
> Yes, you have. By using the PLAY request's URL as base the RTP-Info URLs
> would be:
>
> RTP-Info: url=audio.3gp/trackID=1;seq=0;rtptime=0, url=video.3gp/trackID=1;seq=234;rtptime=23487
>
>
> >
> >I think part of the problem here is that the server is assuming that the
> >client knows what the base URL is. In the case of an SDP initiated
> >playback, I don't have one. To my knowledge there is no way that the
> >server can detect whether I have a base URL or not. I suppose it could
> >check to see if I issued a DESCRIBE or not, but that seems a little messy
> >to me.
> >
>
> There must always be clear what is the base-URL. For RTSP request you
> can't use base-urls that are remembered each request - response pair
> must be self contained in this regard.
>
> >
> >I propose that the RTP-Info field should contain the SETUP URLs exactly
> >the way that they were given to the server in the SETUP. This allows the
> >client an easy way to figure out what URLs are associated with which
> >streams. Relative URLs should not be allowed since it is ambiguous as to
> >what they are relative to.
> >
>
> For RTP-Info you either use relative URLs to the request URL or use
> absolute ones. Unless anyone argues against it I will add the following
> sentence to the RTP-Info URL description to clarify this: "Any relative
> URL SHALL use the request URL as base URL."
>
> 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 Aaron,



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. If an absolute URL is
> specified, it MUST match the URL used to SETUP that stream."
>
>I'm sure this could be worded a little better and put into proper
>RFC-speak.
>  
>
Yes, this sounds reasonable. The sever should really reference the media 
resource that it has setup, not something else. However I have not have 
time to look up what RFC 2396 says about matching a URI.

Cheers

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>
To: Magnus Westerlund <magnus.westerlund@era.ericsson.se>
Cc: Aaron Colwell <acolwell@real.com>, mmusic@ietf.org
Subject: Re: [MMUSIC] Relative URLs in RTP-Info field
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--9zSXsLTf0vkW971A
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> >You have an SDP that looks like..
> >v=3D0
> >a=3Dcontrol:rtsp://foo.com/movie
> >m=3Daudio 0 RTP/AVP 96
> >a=3Dcontrol:rtsp://foo.com/audio.3gp/trackID=3D1
> >a=3Drtpmap:96 MP4A-LATM/44100
> >a=3Drange:npt=3D0-37
> >m=3Dvideo 0 RTP/AVP 97
> >a=3Dcontrol:rtsp://foo.com/video.3gp/trackID=3D1
> >a=3Drtpmap:97 MP4V-ES/90000
> >
> >The RTSP chat for this situation would look like
> >OPTIONS rtsp://foo.com
> >SETUP rtsp://foo.com/audio.3gp/trackID=3D1
> >SETUP response
> >SETUP rtsp://foo.com/video.3gp/trackID=3D1
> >SETUP response
> >PLAY rtsp://foo.com/movie
> >PLAY response
> >RTP-Info: url=3DtrackID=3D1;seq=3D0;rtptime=3D0,url=3DtrackID=3D1;seq=3D=
234;rtptime=3D23487
> >
> >In this case I have no way to figure out what info is associated with
> >which stream.
> >
>=20
> Yes, you have. By using the PLAY request's URL as base the RTP-Info URLs=
=20
> would be:
>=20
> RTP-Info: url=3Daudio.3gp/trackID=3D1;seq=3D0;rtptime=3D0,=20
> url=3Dvideo.3gp/trackID=3D1;seq=3D234;rtptime=3D23487

Based on the RTSP spec and a few private emails, I think the correct answer
here is that the server is in error (is this what you meant to say, Magnus?)
When the client combines the relative urls in RTP-Info with the absolute url
in the request using the rules in RFC1808, the result is this for both
streams:

  rtsp://foo.com/movie + trackID=3D1 -> rtsp://foo.com/trackID=3D1

If the server wished to use relative urls in the RTP-Info header, the proper
response would be one of the following:

A completely relative response would be this:

  RTP-Info: url=3Daudio.3gp/trackID=3D1;seq=3D0;rtptime=3D0,
            url=3Dvideo.3gp/trackID=3D1;seq=3D234;rtptime=3D23487

Note that if the stream urls were in a different path from the aggregate
url, the response would contain one or more ../ sequences.

An absolute path without any host would also work:

  RTP-Info: url=3D/audio.3gp/trackID=3D1;seq=3D0;rtptime=3D0,
            url=3D/video.3gp/trackID=3D1;seq=3D234;rtptime=3D23487

--=20
Innovate, v.:
        To annoy people.

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Tom Marshall wrote:

>Based on the RTSP spec and a few private emails, I think the correct
>answer
>here is that the server is in error (is this what you meant to say,
>Magnus?)
>
Yes, you are correct.

Cheers 

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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The following in this draft look unclear to me. Could someone clarify?


Exampe 2.5. The "m=" line in the Answer does not match that in the Offer.

Example 2.8. In the Answer, "c=" line and the second "m=" line appear to be
out of order?

Example 4.1. Same as above.

Example 4.2. Why there is the port change in Bob's Counter-Answer? 

Example 4.3. The "m=" line in the Answer does not match that in the Offer.


In addition, is it possible to add the following to this draft?

1) RFC2833 and audio under the same "m=" line and RFC2833 has a different
packet interval (for example, 50ms)from audio.

2) Show how to negotiate Adaptive Encoding ("on the fly", RFC2198
redundancy, etc.)and asymmetric codecs

3) Answerer's rejection of Offer, etc. 
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 RtspTaskforceTelConMinutesFeb19-2003

= Attendees =
Anders Klements, Joerg Ott, Sean Sheedy, Tom Marshall, Rob Laphier, Aravind
Narasimhan, Magnus Westerlund, Colin Perkins, Andrea Basso, Thomas Zeng
= Agenda = 
1. Follow up on last meeting
- Redirect, waiting for new text from Tom
- Proxy-require, no comments seems okay to edit into spec. See message:
http://www.ietf.org/mail-archive/working-groups/mmusic/current/msg01167.html
- Update on Range header text in regards to Live streams. Need text 
proposal (Thomas Zeng).
-  Restriction text on usage of destination (Aravind Narasimhan)
- Move NAT & FW to separate draft, update intro of core draft to point 
this out (Magnus W).

2. Administrative
- Document Editor(s)!!!

3. Today's new Issues:
 
3.1 RTSP and NATs revisited:
   3.1.1 Discussion on MMUSIC WG list on how and when you can use STUN and 
Symmetric RTP. STUN for cone NATs which only uses single IP. Symmetric 
RTP requires RTSP and server changes + rules on how to prevent attacks 
using RTP.
   3.1.2  Security problems. RTSP as Denial of Service tool. Hijacking and 
interception attacks.
   3.1.3  Destination secure challenge mechanism?

3.2 Use of transport connection interleaved data
   http://rtsp.org/bug507408
   http://rtsp.org/bug556589
   http://rtsp.org/bug662436
   http://rtsp.org/bug672160

3.3 Using option tags for features not related to RTSP

4. Next meeting




== Minutes ==

== 1. Follow up on last meeting ==

 * Aravind: I sent the restriction text on usage of destination 10 minutes
ago. May want to add some best practise guidelines.
 * Magnus:  The restricted use of destination also relates to NAT traversal.
On NAT subject, I am trying to publish a separate draft on NAT and FW as a
new WG work item.
 * Joerg:   That seems to be a good plan.

== 2. Administrative ==
 * Rob: Is next week the due date to submit drafts to San Francisco IETF?  
 * Magnus: Monday after the next for verion 01 and beyond.
 * Joerg:  Magnus, how much time do you need during next meeting to discuss
RTSP revision?
 * Magnus: Between 30 to 60 minutes.


== 3. Today's New Issues ==

=== 3.1  MMUSIC WG list discussion on how and when you can use STUN and
Symmetric RTP ===
  *  Magnus: STUN only useful for Cone NATs. Can use any public STUN
servers. Symetric RTP is not limited to Cone NAT but requires server
modifications.'
  *  Thomas: Even for STUN server must be modified to accomodate new
client_port spec, therefore STUN and Symetric RTP both require server
changes.

  *  Rob: Are we talking about adding more core requirements?
  *  Magnus: Just in the sense of both RTP and RTCP ports in transport.

  *  Andrea (?): How to make it backwards compatible?

  *  Magnus: The client can still choose old syntax, when play.basic option
tag is not present. For symetric RTP will have a new option tag.

  *  Tom: How important is man-in-the-middle vs DOS? Is it necessary to
design RTSP to be robust against man-in-the-middle attacks? TCP itself is
NOT robust to man-in-the-middle. 

  *  Magnus: It's important to describe security implications WRT
man-in-the-middle.
      
  *  Tom: How about simply recommend running RTSP over SSL?

  *  Magnus: SSL is not sufficient.

  *  Sean (?): The only thing secure is to combine SSL and authentication.

  *  Tom:  If the attacker has the ability to redirect the packets to client
then all bets are off. If the attacker is only able to tap the wire, then we
can protect ourselves.

  *  Rob:  SSL can protect security even if the bad guy takes over the
router.

  *  Thomas: What impact does SSL have on RTSP ?

  *  Rob: little things like a different URL starting with "rtsps", similar
to "https".

  *  Anders: A while ago I reserved 322 as RTSP over SSL in IANA.

  *  Sean:  We already have such an implementation.

  *  Magnus: can you write a draft to advice people how to implement RTSP
over SSL?

  *  Sean: I'll see what I can do.

  *  Thomas: If I want to run RTSP over UDP, then can't use SSL.
  *  Jorg:   That's a different can of worms.

  *  Magnus: My guess is RTSP over UDP will no longer be part of core RTSP
spec.

  *  Adlers: If you interleaved RTP over RTSP, there maybe performance
issues by using SSL. If RTP is over UDP, then RTSP over SSL may break
firwalls, since firewalls won't be able to figure out what UDP ports to
open.

  *  Magnus: then we need midcom --- that's also future topic. 
  *  Magnus: Switching to a different topic: how to deal with proxies in
this context?

  *  Tom: Tranparent proxy will not work when RTSP has security protections
such as SSL.

  *  Joerg: With HTTP proxy, if client requests secure page, then proxies
will simply switch to transpartent forward. 

  *  Tom: But if you are using proxy as ALG or cache UDP, then it will not
work.  With secured RTSP, masqurade FWs will not work, but STUN and symetric
RTP will.

 === 3.1.2 Security problems. RTSP as Denial of Service tool. Hijacking and
interception attacks ===
  *  Magnus: NAT traversal opens up security holes. I don't know there's
much more to say about this.

 === 3.1.3 Destination secure challenge mechanism? ===
  * Magnus: defintely for the future, I don't know whether people have time
to do it.
  * Rob: must clarify security implications b/w normal traust relationship
vs trusting client to request different different destination IP.

  * Aravind: if the RTP receipient does not speak RTSP, then we need to make
authentication out-of-band
  * Magnus: or in-band in RTP stream.
 === 3.2  Use of transport connection interleaved data  ===
      * [ http://rtsp.org/bug507408 ]
  *  Magnus:  For this bug, need to claify that it is required to have
either interleave=n or intervleave = n-m. The former means RTCP is not used
at all.
  *  Tom: should allow interleave=n/m to mean n for RTP, m for RTCP
  *  Magnus: play.basic means n-m

  *  Resolution: the client can use "interleave" without channel id, but
server must indicate channel.
  *  Action: Aravind to write text and post to MMUSIC to address all four
bugs.

  * [http://rtsp.org/bug556589 ]

  *  Aravind: To address this bug, shouldn't we change the default to
unicast?
  *  Magnus: No. If you use interleave, you should specifiy UNICAST. If
client requests interleave with MULTICAST, server should ignore it. We must
make sure no bad example in draft wrt interveave and unicast.
     

  * [http://rtsp.org/bug662436 ]

  * Magnus: this one's from John Lazzaro. "RTP/AVP/TCP w/o interleaved". 

  * Joerg: John's draft is not official work item yet.

  * Magnus: it would be a separate draft.

  * Resolution: table it until the dependencies are ready.

  * [http://rtsp.org/bug672160 Link to bug in source forge ]
 
  * Tom: Proxies may need to re-assign  channel IDs.

  * Rob: Channel id assignment is modelled after UDP, server is to allocate
server channels, client is responsible to allocate client channel IDs. We
don't have a lot more lattitude because of proxies.

  * Adlers: I thought client is not required to specify channels. There used
to be a channel parameter. The extra channel key word was removed. Cleint
was to use "interleave=n" to specify channel.
   
   
  * Resolution: clarify that interleave channels are symetric. If the server
uses channel n as down link RTCP SR, client uses channel n for uplink RTCP
RR.

=== 3.3 Using option tags for features not related to RTSP ===

 * Magnus: client uses option tag to signal codec.
 * Joerg:   Not wise, 'cause it would become open ended. A bad example would
be the CONNECT spec 
 * Adlers: get and set parameters seem to be better for negotiate.
 * Magnus: But this is in DESCRIBE.
 * Thomas: how about fixing SDP?

 * Magnus: you'll see a proposal from Nokia to propose this negotiation for
the client to choose a subset of options in SDP. Part of the reason is so
that SDP will not contain  extremely long list of optional tracks.

* Rob: what is the status of Offer/Answer? would it make sense to add to
RTSP?

* Joerg: It would pose backwards compatibility issue. 

* Rob: I think this is the most sensible things to do: propose a option tag
to indicate whether or not Offer/Answer is  supported.

* Jorg: need to careful evaulate offer / answer semantics. 

* Resolution: No to using option tags for Non RTSP features . Should explore
the history of offer/answer for RTSP as a better solution.
* Action Item: Everybody reads on offer/answer and meet in next IETF meeting
to take next step.  
* [ftp://ftp.rfc-editor.org/in-notes/rfc3264.txt This is the link to
RFC3264]


=== 4 Next meeting ===
* Magnus:  Two weeks from now: March 5.  


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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		: How to make Real-Time Streaming Protocol (RTSP) 
                          traverse Network Address Translators (NAT) and 
                          interact with Firewalls
	Author(s)	: M. Westerlund
	Filename	: draft-ietf-mmusic-rtsp-nat-00.txt
	Pages		: 18
	Date		: 2003-2-24
	
This document describes four different types of NAT traversal
techniques that can be used by RTSP. For each technique a description
on how it shall be used, what security implications it has and other
deployment considerations that exist is given. Further a description
on how RTSP relates to firewalls are also given.

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http://www.ietf.org/internet-drafts/draft-ietf-mmusic-rtsp-nat-00.txt

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Magnus,

First bravo for this draft, it is a very useful document.

I have a few comments, mostly about STUN
(you know I am a STUN fan!)

I note that in the STUN section you write:

" To be able to use STUN to traverse symmetric NATs the STUN server
   needs to be co-located with the streaming server media distribution
   ports."

This is true and it is a very interresting feature indeed,
but then when you had the reader say "ahah!" and
start to get really interrested you cold shower it with:

"This creates an unclear demultiplexing point within the
   server. As this will create implementations difficulties and possibly
   security problems this SHOULD NOT be done."

I think that a "should not" is really too strong here if it is
an implementation issue, co-locating is not in itself
a big issue for exemple a lot of RTSP servers
are also HTTP servers right?
(the "possible security problems" would have to be compared
with the alternative solutions!...)

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

BTW in the Symetric RTP section you write:

"It has the advantage of working for all types of NATs. However it requires server modifications."

Therefore I conclude that there is no difference
in terms of implementation between STUN and symetric RTP
since in both case you need to change the RTSP server,
and in a similar fashion!

Also you write in the same section:

"   1. The client knows or has determined by the use of STUN that it is
       located behind a NAT. It may also determined the type of NAT it
       is behind."

So If one wants to use Symetric RTP one will have to implement STUN _anyway_ !

For implementers the advantages of symetric RTP are therefore difficult to see ?

and then the reason why, by comparison, would co-located-STUN get a "SHOULD NOT" gets
really difficult to understand!

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

Anyway I would propose to reword the section  3.1.3. as:


"There are 2 ways to use STUN dependign if the STUN server
is co-located with the streaming server media distribution ports or not.

3.1.3.1. STUN server outside

 Advantages:

   - Does not require RTSP server modifications, totally client
      implemented tool.

   Disadvantages:

   - Requires a STUN server deployed in the public address space.
   - Does only work well behind Cone NATs. Does not work with
      Symmetric NATs.

3.1.3.2. STUN server inside (co-located at the server media distribution
   ports)

 Advantages:

   - Works with all NATs including Symmetric NATs (however see the security section about sending RTP
data at a different public IP address than the one of the RTSP client).

   Disadvantages:

   - Requires a modified RTSP server with co-located STUN server.

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

Last  open issue about STUN:

you describe a number of RTSP modifications for symetric RTP
and none for STUN while there could be one very useful thing:

In the response to DESCRIBE the server should indicate
the UDP ports it is planning to use so that the client can immediately
STUN toward these ports, and then the STUN server reply can
be used for SETUP (saves a lot of round trips).

The question is how can a server indicate its
RTP _AND_ RTCP port numbers ?

For example you can do that with a header in the RTSP response:


   C->S: DESCRIBE rtsp://foo/twister RTSP/1.0
           CSeq: 1

   S->C: RTSP/1.0 200 OK
           CSeq: 1
           Transport: RTP/AVP;server-rtp-port=4056;server-rtcp-port=4057
           Content-Type: application/sdp
           Content-Length: xxx

           v=0
           o=- 2890844256 2890842807 IN IP4 172.16.2.93
           s=RTSP Session
           etc...

Also you can include this informationin SDP,
which may be better since the client may get the SDP
by some other means, right?


Regards,



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

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--NextPart

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


	Title		: Internet Media Guide Metadata Format
	Author(s)	: J. Luoma
	Filename	: draft-luoma-mmusic-img-metadata-00.txt
	Pages		: 27
	Date		: 2003-2-25
	
This document defines the base metadata format for Internet Media
Guides (IMGs), applicable to a wide variety of Internet hosts and
communication links. IMG metadata describes files, resources and
multimedia programs available for streaming or downloading via
multicast or unicast. An IMG metadata model, semantics of the IMG
media metadata entities, and an XML-based framing structure for IMG
media metadata are defined. The aim is to reuse existing metadata
standards for the IMG metadata format where possible.

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From mailnull@www1.ietf.org  Wed Feb 26 11:14:42 2003
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From: Jonathan Rosenberg <jdrosen@dynamicsoft.com>
Organization: dynamicsoft
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CC: magnus.westerlund@era.ericsson.se, mmusic@ietf.org
Subject: Re: [MMUSIC] RE: draft-ietf-mmusic-rtsp-nat-00.txt
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I have yet to read through the doc, but I wanted to let folks know that 
I have released an I-D which is very relevant to this problem:

http://www.jdrosen.net/papers/draft-rosenberg-sipping-ice-00.txt

it should appear in the archives shortly. Although its a sipping draft, 
its applicable to anything that follows an offer/answer style of 
negotiation. One of the things in there is the usage of STUN over media 
ports (called STUN derived transport addresses). It talks about this 
demux issue, and describes how it can be done for RTP and more generally 
for any UDP protocol. As your note below indicates, sending STUN to the 
media port will obviate the need for symmetric RTP, and it works for 
non-RTP transports too. It is my preference to do that in order to have 
a single solution.

The benefit of ICE is that it always ends up selecting the "best" 
connectivity between entities. In the case of the typical RTSP 
deployment, it would end up using connectivity learned by STUNning to 
the media ports on the server. However, ICE would allow for other 
connectivity models, such as cases where a user is in an enterprise that 
wishes to have a media relay on-path for all media (a turn server, for 
example).

Thanks,
Jonathan R.



philippe.gentric@philips.com wrote:
> Magnus,
> 
> First bravo for this draft, it is a very useful document.
> 
> I have a few comments, mostly about STUN
> (you know I am a STUN fan!)
> 
> I note that in the STUN section you write:
> 
> " To be able to use STUN to traverse symmetric NATs the STUN server
>    needs to be co-located with the streaming server media distribution
>    ports."
> 
> This is true and it is a very interresting feature indeed,
> but then when you had the reader say "ahah!" and
> start to get really interrested you cold shower it with:
> 
> "This creates an unclear demultiplexing point within the
>    server. As this will create implementations difficulties and possibly
>    security problems this SHOULD NOT be done."
> 
> I think that a "should not" is really too strong here if it is
> an implementation issue, co-locating is not in itself
> a big issue for exemple a lot of RTSP servers
> are also HTTP servers right?
> (the "possible security problems" would have to be compared
> with the alternative solutions!...)
> 
> **************
> 
> BTW in the Symetric RTP section you write:
> 
> "It has the advantage of working for all types of NATs. However it requires server modifications."
> 
> Therefore I conclude that there is no difference
> in terms of implementation between STUN and symetric RTP
> since in both case you need to change the RTSP server,
> and in a similar fashion!
> 
> Also you write in the same section:
> 
> "   1. The client knows or has determined by the use of STUN that it is
>        located behind a NAT. It may also determined the type of NAT it
>        is behind."
> 
> So If one wants to use Symetric RTP one will have to implement STUN _anyway_ !
> 
> For implementers the advantages of symetric RTP are therefore difficult to see ?
> 
> and then the reason why, by comparison, would co-located-STUN get a "SHOULD NOT" gets
> really difficult to understand!
> 
> ****************
> 
> Anyway I would propose to reword the section  3.1.3. as:
> 
> 
> "There are 2 ways to use STUN dependign if the STUN server
> is co-located with the streaming server media distribution ports or not.
> 
> 3.1.3.1. STUN server outside
> 
>  Advantages:
> 
>    - Does not require RTSP server modifications, totally client
>       implemented tool.
> 
>    Disadvantages:
> 
>    - Requires a STUN server deployed in the public address space.
>    - Does only work well behind Cone NATs. Does not work with
>       Symmetric NATs.
> 
> 3.1.3.2. STUN server inside (co-located at the server media distribution
>    ports)
> 
>  Advantages:
> 
>    - Works with all NATs including Symmetric NATs (however see the security section about sending RTP
> data at a different public IP address than the one of the RTSP client).
> 
>    Disadvantages:
> 
>    - Requires a modified RTSP server with co-located STUN server.
> 
> **************************************
> 
> Last  open issue about STUN:
> 
> you describe a number of RTSP modifications for symetric RTP
> and none for STUN while there could be one very useful thing:
> 
> In the response to DESCRIBE the server should indicate
> the UDP ports it is planning to use so that the client can immediately
> STUN toward these ports, and then the STUN server reply can
> be used for SETUP (saves a lot of round trips).
> 
> The question is how can a server indicate its
> RTP _AND_ RTCP port numbers ?
> 
> For example you can do that with a header in the RTSP response:
> 
> 
>    C->S: DESCRIBE rtsp://foo/twister RTSP/1.0
>            CSeq: 1
> 
>    S->C: RTSP/1.0 200 OK
>            CSeq: 1
>            Transport: RTP/AVP;server-rtp-port=4056;server-rtcp-port=4057
>            Content-Type: application/sdp
>            Content-Length: xxx
> 
>            v=0
>            o=- 2890844256 2890842807 IN IP4 172.16.2.93
>            s=RTSP Session
>            etc...
> 
> Also you can include this informationin SDP,
> which may be better since the client may get the SDP
> by some other means, right?
> 
> 
> Regards,
> 
> 
> 
> Philippe Gentric
> Software Architect
> Philips MP4Net
> philippe.gentric@philips.com
> http://www.platform4.philips.com
> 
> _______________________________________________
> mmusic mailing list
> mmusic@ietf.org
> https://www1.ietf.org/mailman/listinfo/mmusic
> 

-- 
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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Date: Wed, 26 Feb 2003 11:16:08 -0600
From: Alan Johnston <alan.johnston@wcom.com>
Subject: Re: [MMUSIC] draft-johnston-mmusic-offer-answer-examples-00.txt
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Hi Kai,

Thanks for your detailed reading and comments on the draft.  See mine below.

Thanks,
Alan Johnston
WorldCom
sip:alan@digitalari.com

At 08:36 AM 2/24/2003 -0700, Miao, Kai wrote:
>The following in this draft look unclear to me. Could someone clarify?
>
>
>Exampe 2.5. The "m=" line in the Answer does not match that in the Offer.

I don't see a problem with the m= line, but there are two a= lines that 
are: "a=rtpmap:0 PCMU/8000" that should be "a=rtpmap:97 iLBC"


>Example 2.8. In the Answer, "c=" line and the second "m=" line appear to be
>out of order?

Yes - in this example, the second m= line should be swapped with the c= line.

>Example 4.1. Same as above.

Right as well.

>Example 4.2. Why there is the port change in Bob's Counter-Answer?

A port change can happen at any time.  However, in this example, we mixed 
the port numbers of the offer and the answer and will fix this.

>Example 4.3. The "m=" line in the Answer does not match that in the Offer.

This has the same problem as example 2.5.


>In addition, is it possible to add the following to this draft?
>
>1) RFC2833 and audio under the same "m=" line and RFC2833 has a different
>packet interval (for example, 50ms)from audio.
>
>2) Show how to negotiate Adaptive Encoding ("on the fly", RFC2198
>redundancy, etc.)and asymmetric codecs

If you send us a example of each we'll take a look and consider.

>3) Answerer's rejection of Offer, etc.
>_

There are some examples of declines media streams such as example 2.6  Is 
there something particular you are after?

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

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From: Tom Marshall <tmarshall@real.com>
To: philippe.gentric@philips.com
Cc: magnus.westerlund@era.ericsson.se, mmusic@ietf.org
Subject: Re: [MMUSIC] RE: draft-ietf-mmusic-rtsp-nat-00.txt
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On Wed, Feb 26, 2003 at 11:40:04AM +0100, philippe.gentric@philips.com wrot=
e:
>=20
> Magnus,
>=20
> First bravo for this draft, it is a very useful document.
>=20
> I have a few comments, mostly about STUN
> (you know I am a STUN fan!)
>=20
> I note that in the STUN section you write:
>=20
> " To be able to use STUN to traverse symmetric NATs the STUN server
>    needs to be co-located with the streaming server media distribution
>    ports."
>=20
> This is true and it is a very interresting feature indeed,
> but then when you had the reader say "ahah!" and
> start to get really interrested you cold shower it with:
>=20
> "This creates an unclear demultiplexing point within the
>    server. As this will create implementations difficulties and possibly
>    security problems this SHOULD NOT be done."
>=20
> I think that a "should not" is really too strong here if it is
> an implementation issue, co-locating is not in itself
> a big issue for exemple a lot of RTSP servers
> are also HTTP servers right?
> (the "possible security problems" would have to be compared
> with the alternative solutions!...)

It seems to me that an RTSP server which also does STUN is functionally
equivalent to Symmetric RTP.  So why bother to implement STUN in this case?

> BTW in the Symetric RTP section you write:
>=20
> "It has the advantage of working for all types of NATs. However it requir=
es server modifications."
>=20
> Therefore I conclude that there is no difference
> in terms of implementation between STUN and symetric RTP
> since in both case you need to change the RTSP server,
> and in a similar fashion!

I see two big differences:

1. Timing.  In STUN, the client presumedly contacts the STUN server and
receives the answer for its client_port mapping before sending the SETUP
request (most likely introducing an extra round-trip latency).

2. Protocol.  In STUN, a separate non-RTP protocol is used.  The server must
"demultiplex" the packets by using heuristics to determine if the incoming
packet is STUN or RTP.  While this may be possible due to the specific
differences between STUN and RTP, it is not a good idea in general.

> Also you write in the same section:
>=20
> "   1. The client knows or has determined by the use of STUN that it is
>        located behind a NAT. It may also determined the type of NAT it
>        is behind."
>=20
> So If one wants to use Symetric RTP one will have to implement STUN _anyw=
ay_ !

I think the wording is incorrect here.  The client does not need to know
that it is behind a nat to use Symmetric RTP.  It simply uses it and the
server determines whether it needs to adjust its destination.

> you describe a number of RTSP modifications for symetric RTP
> and none for STUN while there could be one very useful thing:
>=20
> In the response to DESCRIBE the server should indicate
> the UDP ports it is planning to use so that the client can immediately
> STUN toward these ports, and then the STUN server reply can
> be used for SETUP (saves a lot of round trips).
>=20
> The question is how can a server indicate its
> RTP _AND_ RTCP port numbers ?
>=20
> For example you can do that with a header in the RTSP response:
>=20
>=20
>    C->S: DESCRIBE rtsp://foo/twister RTSP/1.0
>            CSeq: 1
>=20
>    S->C: RTSP/1.0 200 OK
>            CSeq: 1
>            Transport: RTP/AVP;server-rtp-port=3D4056;server-rtcp-port=3D4=
057
>            Content-Type: application/sdp
>            Content-Length: xxx
>=20
>            v=3D0
>            o=3D- 2890844256 2890842807 IN IP4 172.16.2.93
>            s=3DRTSP Session
>            etc...

Firstly, I assume that the case where a server is behind a NAT is rare.=20
NATs are primarily for clients.  If someone sets up an RTSP server behind a
NAT and forwards port 554 to it, they should also forward UDP traffic to it=
.=20

[My personal opinion: if an administrator is too afraid to handle those
nasty unclean UDP packets, they shouldn't be running a server.  Relying on
the connection oriented nature of TCP for security is surely folly.  But
passing the buck by requiring someone else "take the security hit" is simply
not acceptable and we should not be supporting this scenario at all.]

Secondly, you are correct that there is no way to specify non-contiguous
transport ports in 2326.  The current agreement for 2326bis is to use the
existing transport tokens with the syntax m/n to indicate an arbitrary list
of ports.

--=20
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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To: Tom Marshall <tmarshall@real.com>
Cc: magnus.westerlund@era.ericsson.se, mmusic@ietf.org
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Tom you wrote:

>Firstly, I assume that the case where a server is behind a NAT is rare.
>NATs are primarily for clients.


I dont see how the "localization" of the NAT
makes any difference from the end-to-end point of view
of having RTSP clients and servers ?


[BTWwhat I am seeing (now and not improoving in any future)
is a lot of NATs everywhere !
i.e. several NATs between a server and a client !
actually it was told that IPV6 deployement,
which should solve the addressing issue(!?)
can only happen progressively and that
 IPV6/IPV4 realms will be effectively linked using ...NATs !
arg! getting pessimistic ?-]


> [My personal opinion: if an administrator is too afraid
> to handle those nasty unclean UDP packets,
> they shouldn't be running a server.


yes ... I used to think that too...
but I discovered that often the guy who wants to deploy the RTSP service
only vaguely knows how to possibly get in contact with the guys running the (first) NAT :-!
these are different jobs, typically different corporations and
unfortunately they usually  have so different priorities/agendas
that they have trouble speaking together...


regards,

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


                                                                                                                                       
                                                                                                                                       
                                                   To:   Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1                                     
                                                   cc:   magnus.westerlund@era.ericsson.se                                             
                                                    mmusic@ietf.org                                                                    
                                                   Subject:    Re: [MMUSIC] RE: draft-ietf-mmusic-rtsp-nat-00.txt                      
               Tom Marshall                                                                                                            
               <tmarshall@real.com>                Classification:                                                                     
                                                                                                                                       
               26/02/2003 18:52                                                                                                        
                                                                                                                                       
                                                                                                                                       




On Wed, Feb 26, 2003 at 11:40:04AM +0100, philippe.gentric@philips.com wrote:
>
> Magnus,
>
> First bravo for this draft, it is a very useful document.
>
> I have a few comments, mostly about STUN
> (you know I am a STUN fan!)
>
> I note that in the STUN section you write:
>
> " To be able to use STUN to traverse symmetric NATs the STUN server
>    needs to be co-located with the streaming server media distribution
>    ports."
>
> This is true and it is a very interresting feature indeed,
> but then when you had the reader say "ahah!" and
> start to get really interrested you cold shower it with:
>
> "This creates an unclear demultiplexing point within the
>    server. As this will create implementations difficulties and possibly
>    security problems this SHOULD NOT be done."
>
> I think that a "should not" is really too strong here if it is
> an implementation issue, co-locating is not in itself
> a big issue for exemple a lot of RTSP servers
> are also HTTP servers right?
> (the "possible security problems" would have to be compared
> with the alternative solutions!...)

It seems to me that an RTSP server which also does STUN is functionally
equivalent to Symmetric RTP.  So why bother to implement STUN in this case?

> BTW in the Symetric RTP section you write:
>
> "It has the advantage of working for all types of NATs. However it requires server modifications."
>
> Therefore I conclude that there is no difference
> in terms of implementation between STUN and symetric RTP
> since in both case you need to change the RTSP server,
> and in a similar fashion!

I see two big differences:

1. Timing.  In STUN, the client presumedly contacts the STUN server and
receives the answer for its client_port mapping before sending the SETUP
request (most likely introducing an extra round-trip latency).

2. Protocol.  In STUN, a separate non-RTP protocol is used.  The server must
"demultiplex" the packets by using heuristics to determine if the incoming
packet is STUN or RTP.  While this may be possible due to the specific
differences between STUN and RTP, it is not a good idea in general.

> Also you write in the same section:
>
> "   1. The client knows or has determined by the use of STUN that it is
>        located behind a NAT. It may also determined the type of NAT it
>        is behind."
>
> So If one wants to use Symetric RTP one will have to implement STUN _anyway_ !

I think the wording is incorrect here.  The client does not need to know
that it is behind a nat to use Symmetric RTP.  It simply uses it and the
server determines whether it needs to adjust its destination.

> you describe a number of RTSP modifications for symetric RTP
> and none for STUN while there could be one very useful thing:
>
> In the response to DESCRIBE the server should indicate
> the UDP ports it is planning to use so that the client can immediately
> STUN toward these ports, and then the STUN server reply can
> be used for SETUP (saves a lot of round trips).
>
> The question is how can a server indicate its
> RTP _AND_ RTCP port numbers ?
>
> For example you can do that with a header in the RTSP response:
>
>
>    C->S: DESCRIBE rtsp://foo/twister RTSP/1.0
>            CSeq: 1
>
>    S->C: RTSP/1.0 200 OK
>            CSeq: 1
>            Transport: RTP/AVP;server-rtp-port=4056;server-rtcp-port=4057
>            Content-Type: application/sdp
>            Content-Length: xxx
>
>            v=0
>            o=- 2890844256 2890842807 IN IP4 172.16.2.93
>            s=RTSP Session
>            etc...

Firstly, I assume that the case where a server is behind a NAT is rare.
NATs are primarily for clients.  If someone sets up an RTSP server behind a
NAT and forwards port 554 to it, they should also forward UDP traffic to it.

[My personal opinion: if an administrator is too afraid to handle those
nasty unclean UDP packets, they shouldn't be running a server.  Relying on
the connection oriented nature of TCP for security is surely folly.  But
passing the buck by requiring someone else "take the security hit" is simply
not acceptable and we should not be supporting this scenario at all.]

Secondly, you are correct that there is no way to specify non-contiguous
transport ports in 2326.  The current agreement for 2326bis is to use the
existing transport tokens with the syntax m/n to indicate an arbitrary list
of ports.

--
Coding is easy; All you do is sit staring at a terminal until the drops of
blood form on your forehead.
        -- Anonymous
(See attached file: att5k12c.dat)


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From: Tom Marshall <tmarshall@real.com>
To: philippe.gentric@philips.com
Cc: magnus.westerlund@era.ericsson.se, mmusic@ietf.org
Subject: Re: [MMUSIC] RE: draft-ietf-mmusic-rtsp-nat-00.txt
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On Wed, Feb 26, 2003 at 07:26:15PM +0100, philippe.gentric@philips.com wrot=
e:
> Tom you wrote:
>=20
> >Firstly, I assume that the case where a server is behind a NAT is rare.
> >NATs are primarily for clients.
>=20
>=20
> I dont see how the "localization" of the NAT
> makes any difference from the end-to-end point of view
> of having RTSP clients and servers ?

It makes little difference in the specification, but it a huge difference in
the implementation.  If it's possible for the server to be behind a NAT that
doesn't forward UDP packets, it would need to use NAT traversal to inform
the client of its RTP ports.

We know that STUN does not work for symmetric NATs.  We know that NATs as
implemented in most common Unix routers are symmetric.  Therefore, I am
guessing that most NATted servers will be behind symmetric NATs and
server-to-client STUN will not work anyway.

This is not to say that we should discourage its use, of course.  I am only
saying that NAT should be viewed primarily as a client-side phenomenon.=20
Implementors are free to implement STUN and/or Symmetric RTP on both client
and server and have a product that works when both are behind NAT.  With
careful implementation, a product could even work when both ends are behind
symmetric NATs.  See http://www.peercast.org/ for one possible application
of server side NAT traversal.

> [BTWwhat I am seeing (now and not improoving in any future)
> is a lot of NATs everywhere !
> i.e. several NATs between a server and a client !

Absolutely.  There are two between my home machines and the net.

> actually it was told that IPV6 deployement,
> which should solve the addressing issue(!?)
> can only happen progressively and that
>  IPV6/IPV4 realms will be effectively linked using ...NATs !
> arg! getting pessimistic ?-]

IPv6 is amazingly adept at traversing IPv4 NAT.  I am running an IPv6 tunnel
to my home machines.  You can contact them via IPv6 anywhere in the world
even though they are behind two levels of IPv4 NAT.

> > [My personal opinion: if an administrator is too afraid
> > to handle those nasty unclean UDP packets,
> > they shouldn't be running a server.
>=20
>=20
> yes ... I used to think that too...
> but I discovered that often the guy who wants to deploy the RTSP service
> only vaguely knows how to possibly get in contact with the guys running t=
he (first) NAT :-!
> these are different jobs, typically different corporations and
> unfortunately they usually  have so different priorities/agendas
> that they have trouble speaking together...

The sysadmin vs. netadmin argument holds no water.  We should NOT be
providing a means to deliberately break administrative control.  (It's
redundant anyway, see RFC 3093).

NAT traversal is intended to work around the limitations in common NAT
implementations.  If it happens to work for the server-to-client direction,
well, that's a coincidence.  It might even be seen as a consequence of good
design.  But it is not our intent to deliberately undermine administrative
control.

--=20
A revised schedule is to business what a new season is to an athlete or a
new canvas to an artist.
        -- Norman Augustine

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Tom you write:

> We know that STUN does not work for symmetric NATs.

STUN works for symetric NAT if you run the STUN server co-located with the RTSP server,
and that is the key thing: a solution that works for *all* NATs.

As Jonathan coined it very nicely in draft-rosenberg-sipping-ice-00.txt:

"The need to run STUN on the same transport address as the media
   stream represents the "ugliest" piece of ICE. However, it is an
   essential part of the story. By sending STUN requests to the very
   same place media is sent, any bindings learned through STUN will be
   useful even when communicating through symmetric NATs. This results
   in a substantial increase in the scope of applicability of STUN, in
   terms of cases where it can provide connectivity. In that sense, the
   usage of STUN here is radically different than the usage models
   outlined in [13], where STUN is generally useless for dealing with
   symmetric NAT."

>The sysadmin vs. netadmin argument holds no water.  We should NOT be
>providing a means to deliberately break administrative control.  (It's
>redundant anyway, see RFC 3093).
>
....
>But it is not our intent to deliberately undermine administrative
>control.

ha no! not that!
you cannot 3093'ze me on this because
3093 does not apply a single bit here.
(I feel in the position to debate about 3093 since the London meeting!-)

the plan is not  to undermine "administrative control",
because and *that* is the point: you *cannot* "undermine" it!
 when the adminitrator wants to block RTP traffic  he will,
because he can and it is *very* easy!!!!!!

my original point was that:

(A)in order of priority for operators "NAT comes first" and then
"someone wants to fix the RTP problem" specifically we are told:
"ok, there is this NAT, and we want to deploy
VOD using RTSP, what do you propose ? "

(B)Telling them to start a procedure (internal or external or both!)
that would lead to transforming that NAT is not a good idea,
*especially* since you cannot guarantee it will work  in all cases!

But we will have to influence "them" in at least one respect,
it must be documented somewhere that:

"Symmetric NATs SHOULD be configured so that a given RTP
client is attributed the same public IP address as the corresponding
RTSP client"

(if there are NAT implementors online can they please comment on that?
it does not look like a very difficult thing to implement ?)

Because otherwise whenever the symmetric NAT will allocate different IP addresses
the RTSP server will *have to* deny the service for DDOS prevention.


>NAT traversal is intended to work around the limitations in common NAT
>implementations.

not enough for me,
I want a solution for all NATs, forever.


regards,


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


                                                                                                                                       
                                                                                                                                       
                                                   To:   Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1                                     
                                                   cc:   magnus.westerlund@era.ericsson.se                                             
                                                    mmusic@ietf.org                                                                    
                                                   Subject:    Re: [MMUSIC] RE: draft-ietf-mmusic-rtsp-nat-00.txt                      
               Tom Marshall                                                                                                            
               <tmarshall@real.com>                Classification:                                                                     
                                                                                                                                       
               26/02/2003 21:07                                                                                                        
                                                                                                                                       
                                                                                                                                       




On Wed, Feb 26, 2003 at 07:26:15PM +0100, philippe.gentric@philips.com wrote:
> Tom you wrote:
>
> >Firstly, I assume that the case where a server is behind a NAT is rare.
> >NATs are primarily for clients.
>
>
> I dont see how the "localization" of the NAT
> makes any difference from the end-to-end point of view
> of having RTSP clients and servers ?

It makes little difference in the specification, but it a huge difference in
the implementation.  If it's possible for the server to be behind a NAT that
doesn't forward UDP packets, it would need to use NAT traversal to inform
the client of its RTP ports.

We know that STUN does not work for symmetric NATs.  We know that NATs as
implemented in most common Unix routers are symmetric.  Therefore, I am
guessing that most NATted servers will be behind symmetric NATs and
server-to-client STUN will not work anyway.

This is not to say that we should discourage its use, of course.  I am only
saying that NAT should be viewed primarily as a client-side phenomenon.
Implementors are free to implement STUN and/or Symmetric RTP on both client
and server and have a product that works when both are behind NAT.  With
careful implementation, a product could even work when both ends are behind
symmetric NATs.  See http://www.peercast.org/ for one possible application
of server side NAT traversal.

> [BTWwhat I am seeing (now and not improoving in any future)
> is a lot of NATs everywhere !
> i.e. several NATs between a server and a client !

Absolutely.  There are two between my home machines and the net.

> actually it was told that IPV6 deployement,
> which should solve the addressing issue(!?)
> can only happen progressively and that
>  IPV6/IPV4 realms will be effectively linked using ...NATs !
> arg! getting pessimistic ?-]

IPv6 is amazingly adept at traversing IPv4 NAT.  I am running an IPv6 tunnel
to my home machines.  You can contact them via IPv6 anywhere in the world
even though they are behind two levels of IPv4 NAT.

> > [My personal opinion: if an administrator is too afraid
> > to handle those nasty unclean UDP packets,
> > they shouldn't be running a server.
>
>
> yes ... I used to think that too...
> but I discovered that often the guy who wants to deploy the RTSP service
> only vaguely knows how to possibly get in contact with the guys running the (first) NAT :-!
> these are different jobs, typically different corporations and
> unfortunately they usually  have so different priorities/agendas
> that they have trouble speaking together...

The sysadmin vs. netadmin argument holds no water.  We should NOT be
providing a means to deliberately break administrative control.  (It's
redundant anyway, see RFC 3093).

NAT traversal is intended to work around the limitations in common NAT
implementations.  If it happens to work for the server-to-client direction,
well, that's a coincidence.  It might even be seen as a consequence of good
design.  But it is not our intent to deliberately undermine administrative
control.

--
A revised schedule is to business what a new season is to an athlete or a
new canvas to an artist.
        -- Norman Augustine
(See attached file: att29tr3.dat)


--0__=4EBBE649DFA1FA4B8f9e8a93df938690918c4EBBE649DFA1FA4B
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	name="att29tr3.dat"
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--NextPart

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


	Title		: Protocol Requirements for Internet Media Guides
	Author(s)	: Y. Nomura, H. Schulzrinne
	Filename	: draft-nomura-mmusic-img-requirements-00.txt
	Pages		: 7
	Date		: 2003-2-26
	
This memo specifies requirements for a protocol for accessing and
updating Internet Media Guide (IMG) information for media-on-demand
and multicast applications. The requirements details general protocol
operations and description formats transmitted by an uni- or bi-
directional transport based on the IMG framework.

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

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From: Tom Marshall <tmarshall@real.com>
To: philippe.gentric@philips.com
Cc: magnus.westerlund@era.ericsson.se, mmusic@ietf.org
Subject: Re: [MMUSIC] RE: Symmetric NAT traversal is a key feature. (was: draft-ietf-mmusic-rtsp-nat-00.txt)
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> >The sysadmin vs. netadmin argument holds no water.  We should NOT be
> >providing a means to deliberately break administrative control.  (It's
> >redundant anyway, see RFC 3093).
> >
> ....
> >But it is not our intent to deliberately undermine administrative
> >control.
>=20
> ha no! not that!
> you cannot 3093'ze me on this because
> 3093 does not apply a single bit here.
> (I feel in the position to debate about 3093 since the London meeting!-)
>=20
> the plan is not  to undermine "administrative control",
> because and *that* is the point: you *cannot* "undermine" it!
>  when the adminitrator wants to block RTP traffic  he will,
> because he can and it is *very* easy!!!!!!

Am I the only one here that reads the same argument for 3093?

  the plan is not  to undermine "administrative control",
  because and *that* is the point: you *cannot* "undermine" it!
   when the adminitrator wants to block 3093 traffic  he will, =20
  because he can and it is *very* easy!!!!!! it's a simple matter
  of blocking the "TCP_*", "IP_*", "IP4_*", and "IP6_*" headers!!!

> (A)in order of priority for operators "NAT comes first" and then
> "someone wants to fix the RTP problem" specifically we are told:
> "ok, there is this NAT, and we want to deploy
> VOD using RTSP, what do you propose ? "

> But we will have to influence "them" in at least one respect,
> it must be documented somewhere that:
>=20
> "Symmetric NATs SHOULD be configured so that a given RTP
> client is attributed the same public IP address as the corresponding
> RTSP client"

I have yet to see an example where a client was assigned different public IP
addresses for RTSP and RTP connections to the same server.  Rumors keep
flying that they exist so I'll assume they do.

I like this recommendation -- it should always be the case anyway.  If it's
not, the NAT is being unreasonable.

> (if there are NAT implementors online can they please comment on that?
> it does not look like a very difficult thing to implement ?)

Yes, please.  Does someone run a NAT where a given (client_ip, server_ip)
can be mapped to more than one nat_ip?

> Because otherwise whenever the symmetric NAT will allocate different IP a=
ddresses
> the RTSP server will *have to* deny the service for DDOS prevention.

I am wondering if you are equating "symmetric NAT" with a NAT that maps
clients to multiple public addresses.

My understanding is that symmetric NAT is a more strict case of general NAT=
.=20
The difference is that a general NAT maps a given NAT port based on the pair
(client_ip,client_port) and a symmetric NAT maps a given NAT port based on
the 4-tuple (client_ip,client_port,server_ip,server_port).  Am I mistaken?

> >NAT traversal is intended to work around the limitations in common NAT
> >implementations.
>=20
> not enough for me,
> I want a solution for all NATs, forever.

I want world peace, too.  And no more spam.  And ...

Seriously, there is no all-encompassing solution.  Why not?  Because NAT is
the general name for a family of hacks with no formal specification or
interoperability requirements.  As soon as you design a solution that you
think covers all NATs, someone will build a different NAT that breaks your
solution.

Symmetric RTP and multiplexed STUN/RTP are functionally equivalent for NAT
traversal and they are the best solution you will find in the forseeable
future.

--=20
Any sufficiently advanced technology is indistinguishable from a rigged
demo.
        -- Andy Finkel

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Hi, Philippe,

My comments below:

philippe.gentric@philips.com wrote:

>Magnus,
>
>First bravo for this draft, it is a very useful document.
>
>I have a few comments, mostly about STUN
>(you know I am a STUN fan!)
>
>I note that in the STUN section you write:
>
>" To be able to use STUN to traverse symmetric NATs the STUN server
>   needs to be co-located with the streaming server media distribution
>   ports."
>
>This is true and it is a very interresting feature indeed,
>but then when you had the reader say "ahah!" and
>start to get really interrested you cold shower it with:
>
>"This creates an unclear demultiplexing point within the
>   server. As this will create implementations difficulties and possibly
>   security problems this SHOULD NOT be done."
>
>I think that a "should not" is really too strong here if it is
>an implementation issue, co-locating is not in itself
>a big issue for exemple a lot of RTSP servers
>are also HTTP servers right?
>(the "possible security problems" would have to be compared
>with the alternative solutions!...)
>  
>

I think this is something we need to discuss further. My current view is 
that due to the lack of a well defined demultiplexing structure between 
media protocol and STUN on the servers ports makes STUN solutions for 
symmetric NATs a bad choice. As pointed out you need to compare the 
media protocols bit structure with STUN request. This is not safe 
operation. I do agree that RTP and RTCP seems to be reasonable safe to 
demultiplex, however it is impossible to know that all protocols media 
protocols will work.

I also find your example with HTTP and RTSP a bad example. HTTP and RTSP 
are distinguished using the nice demultiplexing mechanism of ports. 
Normally HTTP request goes to port 80 while RTSP goes to port number 
540. This is a well defined demultiplexing.

>**************
>
>BTW in the Symetric RTP section you write:
>
>"It has the advantage of working for all types of NATs. However it requires server modifications."
>
>Therefore I conclude that there is no difference
>in terms of implementation between STUN and symetric RTP
>since in both case you need to change the RTSP server,
>and in a similar fashion!
>

Yes, unless you can use STUN to find two port numbers that match the RTP 
rule of consecutive even and odd ports. Both solutions requires updates. 
However STUNs are less from a RTSP protocol perspective.

>
>Also you write in the same section:
>
>"   1. The client knows or has determined by the use of STUN that it is
>       located behind a NAT. It may also determined the type of NAT it
>       is behind."
>
>So If one wants to use Symetric RTP one will have to implement STUN _anyway_ !
>
I need to clarify that this is OPTIONAL behavior that MAY improve the 
performance of the solution.

>For implementers the advantages of symetric RTP are therefore difficult to see ?
>and then the reason why, by comparison, would co-located-STUN get a "SHOULD NOT" gets
>really difficult to understand!
>  
>
As explained before this is not a clean solution.

>****************
>
>Anyway I would propose to reword the section  3.1.3. as:
>
>
>"There are 2 ways to use STUN dependign if the STUN server
>is co-located with the streaming server media distribution ports or not.
>
>3.1.3.1. STUN server outside
>
> Advantages:
>
>   - Does not require RTSP server modifications, totally client
>      implemented tool.
>
>   Disadvantages:
>
>   - Requires a STUN server deployed in the public address space.
>   - Does only work well behind Cone NATs. Does not work with
>      Symmetric NATs.
>
>3.1.3.2. STUN server inside (co-located at the server media distribution
>   ports)
>
> Advantages:
>
>   - Works with all NATs including Symmetric NATs (however see the security section about sending RTP
>data at a different public IP address than the one of the RTSP client).
>
>   Disadvantages:
>
>   - Requires a modified RTSP server with co-located STUN server.
>  
>
In the second case you forgot the disadvantage of not being guaranteed 
to be able to correctly distinguish media packets and STUN request.

Another thing that it seems that I forgot being a disadvantage of STUN 
is that it does not work if the media mappings would get different IP 
addresses, either between each UDP stream or with the TCP connection. 
The reason is the restriction that RTSP must use to prevent DOS attacks.

>**************************************
>
>Last  open issue about STUN:
>
>you describe a number of RTSP modifications for symetric RTP
>and none for STUN while there could be one very useful thing:
>
>In the response to DESCRIBE the server should indicate
>the UDP ports it is planning to use so that the client can immediately
>STUN toward these ports, and then the STUN server reply can
>be used for SETUP (saves a lot of round trips).
>
>The question is how can a server indicate its
>RTP _AND_ RTCP port numbers ?
>
>For example you can do that with a header in the RTSP response:
>
>
>   C->S: DESCRIBE rtsp://foo/twister RTSP/1.0
>           CSeq: 1
>
>   S->C: RTSP/1.0 200 OK
>           CSeq: 1
>           Transport: RTP/AVP;server-rtp-port=4056;server-rtcp-port=4057
>           Content-Type: application/sdp
>           Content-Length: xxx
>
>           v=0
>           o=- 2890844256 2890842807 IN IP4 172.16.2.93
>           s=RTSP Session
>           etc...
>
>Also you can include this informationin SDP,
>which may be better since the client may get the SDP
>by some other means, right?
>
>  
>
Actually you can include this information in the SDP without any extra 
extensions is just the matter of defining the usage. The RTP port number 
can be located in its normal place on the m= line. The RTCP port can be 
set by the attribute defined by
draft-ietf-mmusic-sdp4nat-03.txt. To make this in a good fashion the 
server should define a RTSP feature tag that indicates this 
functionality and that the server should provide the information when 
sending the DESCRIBE response.

I think we need to further discuss the RTSP NAT traversal solutions and 
possible optimization. My draft was intended to be a common ground to 
discuss around.

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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Tom,

no more 3093'zation please! my wife is getting worried
she said I am speaking to myself at night and I keep on
saying "non! non! pas la trente quatrevingt treize!
pas la trente quatrevingt treize!"

;-)

>> "Symmetric NATs SHOULD be configured so that a given RTP
>> client is attributed the same public IP address as the corresponding
>> RTSP client"
>
>I have yet to see an example where a client was assigned different public IP
>addresses for RTSP and RTP connections to the same server.  Rumors keep
>flying that they exist so I'll assume they do.
>
>I like this recommendation -- it should always be the case anyway.  If it's
>not, the NAT is being unreasonable.
>
> (if there are NAT implementors online can they please comment on that?
> it does not look like a very difficult thing to implement ?)
>
>Yes, please.  Does someone run a NAT where a given (client_ip, server_ip)
>can be mapped to more than one nat_ip?

I think a lot of NATs are running on a single public IP address:
it gives 60000+ ports to multiplex clients through which is already
quite a bit...so we dont see it happen now.

My concern is to get in trouble for larger deployements where the
operator has many more public IP addresses because I suspect
the "most simple" way to implement such NAT is to pick up the first free pair
(public IP address, port number) and since de-allocation and allocation
are stochastical then there is a fair chance that RTSP and RTP can be mapped
on different public IP addresses..

>I am wondering if you are equating "symmetric NAT" with a NAT that maps
>clients to multiple public addresses.

I am equating "symmetric NAT" with a NAT that *may* maps
clients to multiple public addresses.

>
>My understanding is that symmetric NAT is a more strict case of general NAT.
>The difference is that a general NAT maps a given NAT port based on the pair
>(client_ip,client_port) and a symmetric NAT maps a given NAT port based on
>the 4-tuple (client_ip,client_port,server_ip,server_port).  Am I mistaken?

I dont know what you mean by "general NAT"?
my reference for NAT typologie is the STUN draft?

I think your 4-tuple is accurate for "symetric NAT"
unless it is a 5-tuple including the protocol
i.e. TCP vs UDP?

regards,


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


                                                                                                                                       
                                                                                                                                       
                                                   To:   Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1                                     
                                                   cc:   magnus.westerlund@era.ericsson.se                                             
                                                    mmusic@ietf.org                                                                    
                                                   Subject:    Re: [MMUSIC] RE: Symmetric NAT traversal is a key feature. (was:        
               Tom Marshall                         draft-ietf-mmusic-rtsp-nat-00.txt)                                                 
               <tmarshall@real.com>                                                                                                    
                                                   Classification:                                                                     
               27/02/2003 18:26                                                                                                        
                                                                                                                                       
                                                                                                                                       




> >The sysadmin vs. netadmin argument holds no water.  We should NOT be
> >providing a means to deliberately break administrative control.  (It's
> >redundant anyway, see RFC 3093).
> >
> ....
> >But it is not our intent to deliberately undermine administrative
> >control.
>
> ha no! not that!
> you cannot 3093'ze me on this because
> 3093 does not apply a single bit here.
> (I feel in the position to debate about 3093 since the London meeting!-)
>
> the plan is not  to undermine "administrative control",
> because and *that* is the point: you *cannot* "undermine" it!
>  when the adminitrator wants to block RTP traffic  he will,
> because he can and it is *very* easy!!!!!!

Am I the only one here that reads the same argument for 3093?

  the plan is not  to undermine "administrative control",
  because and *that* is the point: you *cannot* "undermine" it!
   when the adminitrator wants to block 3093 traffic  he will,
  because he can and it is *very* easy!!!!!! it's a simple matter
  of blocking the "TCP_*", "IP_*", "IP4_*", and "IP6_*" headers!!!

> (A)in order of priority for operators "NAT comes first" and then
> "someone wants to fix the RTP problem" specifically we are told:
> "ok, there is this NAT, and we want to deploy
> VOD using RTSP, what do you propose ? "

> But we will have to influence "them" in at least one respect,
> it must be documented somewhere that:
>
> "Symmetric NATs SHOULD be configured so that a given RTP
> client is attributed the same public IP address as the corresponding
> RTSP client"

I have yet to see an example where a client was assigned different public IP
addresses for RTSP and RTP connections to the same server.  Rumors keep
flying that they exist so I'll assume they do.

I like this recommendation -- it should always be the case anyway.  If it's
not, the NAT is being unreasonable.

> (if there are NAT implementors online can they please comment on that?
> it does not look like a very difficult thing to implement ?)

Yes, please.  Does someone run a NAT where a given (client_ip, server_ip)
can be mapped to more than one nat_ip?

> Because otherwise whenever the symmetric NAT will allocate different IP addresses
> the RTSP server will *have to* deny the service for DDOS prevention.

I am wondering if you are equating "symmetric NAT" with a NAT that maps
clients to multiple public addresses.

My understanding is that symmetric NAT is a more strict case of general NAT.
The difference is that a general NAT maps a given NAT port based on the pair
(client_ip,client_port) and a symmetric NAT maps a given NAT port based on
the 4-tuple (client_ip,client_port,server_ip,server_port).  Am I mistaken?

> >NAT traversal is intended to work around the limitations in common NAT
> >implementations.
>
> not enough for me,
> I want a solution for all NATs, forever.

I want world peace, too.  And no more spam.  And ...

Seriously, there is no all-encompassing solution.  Why not?  Because NAT is
the general name for a family of hacks with no formal specification or
interoperability requirements.  As soon as you design a solution that you
think covers all NATs, someone will build a different NAT that breaks your
solution.

Symmetric RTP and multiplexed STUN/RTP are functionally equivalent for NAT
traversal and they are the best solution you will find in the forseeable
future.

--
Any sufficiently advanced technology is indistinguishable from a rigged
demo.
        -- Andy Finkel
(See attached file: attde3bo.dat)


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


	Title		: Media Policy Manipulation in the Conference Policy 
                          Control Protocol
	Author(s)	: R. Mahy, N. Ismail
	Filename	: draft-mahy-sipping-media-policy-control-00.txt
	Pages		: 19
	Date		: 2003-2-27
	
The SIP conferencing framework defines a model for tightly-coupled
conferencing in the Session Initiation Protocol (SIP), in which a
Conference Policy Control Protocol is used to manipulate policies
relevant to a specific conference, such as conference membership
policy, authorization policy, and media policy. This document
describes a logical model to describe media processing in a SIP
conference. It also defines specific protocol semantics and a
specific syntax to manipulate that model.

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This draft is a work item of the Multiparty Multimedia Session Control Working Group of the IETF.

	Title		: A Transport Independent Bandwidth Modifier for the Session Description Protocol (SDP)
	Author(s)	: M. Westerlund
	Filename	: draft-ietf-mmusic-sdp-bwparam-01.txt
	Pages		: 16
	Date		: 2003-2-27
	
The existing Session Description Protocol (SDP) bandwidth modifiers
and their values include the bandwidth needed also for the transport
and IP layers. When using SDP in protocols like Session Announcement
Protocol (SAP), Session Initiation Protocol (SIP) and Real-Time
Streaming Protocol (RTSP) and the involved hosts reside in networks
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Magnus,

I expect indeed that we will discuss this more especially during the SFO meeting and I agree your draft is an excellent basis!

you write:

>My current view is
>that due to the lack of a well defined demultiplexing structure between
>media protocol and STUN on the servers ports makes STUN solutions for
>symmetric NATs a bad choice. As pointed out you need to compare the
>media protocols bit structure with STUN request. This is not safe
>operation. I do agree that RTP and RTCP seems to be reasonable safe to
>demultiplex, however it is impossible to know that all protocols media
>protocols will work.

Well I dont know about other protocols but the use of STUN in the RTSP scenario is *extremely* safe:

1) your client sends and receives STUN on the UDP ports it has choosen for RTP and RTCP
_but_ since this is *before* the SETUP  there is *only* a STUN client behind these UDP ports,
(the RTP/RTCP stack does not have to be initialized yet). And vice versa
i.e. as soon as STUN completed you can stop the STUN client to
start the RTP/RTCP session !

=> so on the client side it is totally safe.

2) the server does NOT listen for incomming data on the RTP ports:
there will be the STUN server listening behind these ports, nothing else

=> no possible collision of STUN with RTP on the server side

3) then the server needs to  distinguish between RTCP RR
and STUN on the RTCP port but the headers are _really_ different.

for starters the first bit is different  (1 for RTCP, 0 for STUN)!

but let us imagine that a STUN packet (of an hypothetical future extended version!)
will by chance match the first 32 bits of a paussible RTCP RR packet,
then what is the probability of having the 32 bits SSRC (random, fixed per transaction)
match the first 32 bits of the 128 STUn tID (random, changing every message) ?
2 to the power 64 ?

etc ...

I dont see how even the worse implementation can collide ?

> As explained before this is not a clean solution.

"Striving to better, oft we mar what's well." (William Shakespeare)

regards,



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


                                                                                                                                              
                                                                                                                                              
                                                          To:   Philippe Gentric/MP4-SUR/CE/PHILIPS@EMEA1                                     
                                                          cc:   mmusic@ietf.org                                                               
                                                          Subject:    [MMUSIC] Re: draft-ietf-mmusic-rtsp-nat-00.txt                          
                                                                                                                                              
               Magnus Westerlund                          Classification:                                                                     
               <magnus.westerlund@era.ericsson                                                                                                
               .se>                                                                                                                           
                                                                                                                                              
               Sent by:                                                                                                                       
               mmusic-admin@ietf.org                                                                                                          
                                                                                                                                              
               28/02/2003 11:28                                                                                                               
                                                                                                                                              
                                                                                                                                              




Hi, Philippe,

My comments below:

philippe.gentric@philips.com wrote:

>Magnus,
>
>First bravo for this draft, it is a very useful document.
>
>I have a few comments, mostly about STUN
>(you know I am a STUN fan!)
>
>I note that in the STUN section you write:
>
>" To be able to use STUN to traverse symmetric NATs the STUN server
>   needs to be co-located with the streaming server media distribution
>   ports."
>
>This is true and it is a very interresting feature indeed,
>but then when you had the reader say "ahah!" and
>start to get really interrested you cold shower it with:
>
>"This creates an unclear demultiplexing point within the
>   server. As this will create implementations difficulties and possibly
>   security problems this SHOULD NOT be done."
>
>I think that a "should not" is really too strong here if it is
>an implementation issue, co-locating is not in itself
>a big issue for exemple a lot of RTSP servers
>are also HTTP servers right?
>(the "possible security problems" would have to be compared
>with the alternative solutions!...)
>
>

I think this is something we need to discuss further. My current view is
that due to the lack of a well defined demultiplexing structure between
media protocol and STUN on the servers ports makes STUN solutions for
symmetric NATs a bad choice. As pointed out you need to compare the
media protocols bit structure with STUN request. This is not safe
operation. I do agree that RTP and RTCP seems to be reasonable safe to
demultiplex, however it is impossible to know that all protocols media
protocols will work.

I also find your example with HTTP and RTSP a bad example. HTTP and RTSP
are distinguished using the nice demultiplexing mechanism of ports.
Normally HTTP request goes to port 80 while RTSP goes to port number
540. This is a well defined demultiplexing.

>**************
>
>BTW in the Symetric RTP section you write:
>
>"It has the advantage of working for all types of NATs. However it requires server modifications."
>
>Therefore I conclude that there is no difference
>in terms of implementation between STUN and symetric RTP
>since in both case you need to change the RTSP server,
>and in a similar fashion!
>

Yes, unless you can use STUN to find two port numbers that match the RTP
rule of consecutive even and odd ports. Both solutions requires updates.
However STUNs are less from a RTSP protocol perspective.

>
>Also you write in the same section:
>
>"   1. The client knows or has determined by the use of STUN that it is
>       located behind a NAT. It may also determined the type of NAT it
>       is behind."
>
>So If one wants to use Symetric RTP one will have to implement STUN _anyway_ !
>
I need to clarify that this is OPTIONAL behavior that MAY improve the
performance of the solution.

>For implementers the advantages of symetric RTP are therefore difficult to see ?
>and then the reason why, by comparison, would co-located-STUN get a "SHOULD NOT" gets
>really difficult to understand!
>
>
As explained before this is not a clean solution.

>****************
>
>Anyway I would propose to reword the section  3.1.3. as:
>
>
>"There are 2 ways to use STUN dependign if the STUN server
>is co-located with the streaming server media distribution ports or not.
>
>3.1.3.1. STUN server outside
>
> Advantages:
>
>   - Does not require RTSP server modifications, totally client
>      implemented tool.
>
>   Disadvantages:
>
>   - Requires a STUN server deployed in the public address space.
>   - Does only work well behind Cone NATs. Does not work with
>      Symmetric NATs.
>
>3.1.3.2. STUN server inside (co-located at the server media distribution
>   ports)
>
> Advantages:
>
>   - Works with all NATs including Symmetric NATs (however see the security section about sending RTP
>data at a different public IP address than the one of the RTSP client).
>
>   Disadvantages:
>
>   - Requires a modified RTSP server with co-located STUN server.
>
>
In the second case you forgot the disadvantage of not being guaranteed
to be able to correctly distinguish media packets and STUN request.

Another thing that it seems that I forgot being a disadvantage of STUN
is that it does not work if the media mappings would get different IP
addresses, either between each UDP stream or with the TCP connection.
The reason is the restriction that RTSP must use to prevent DOS attacks.

>**************************************
>
>Last  open issue about STUN:
>
>you describe a number of RTSP modifications for symetric RTP
>and none for STUN while there could be one very useful thing:
>
>In the response to DESCRIBE the server should indicate
>the UDP ports it is planning to use so that the client can immediately
>STUN toward these ports, and then the STUN server reply can
>be used for SETUP (saves a lot of round trips).
>
>The question is how can a server indicate its
>RTP _AND_ RTCP port numbers ?
>
>For example you can do that with a header in the RTSP response:
>
>
>   C->S: DESCRIBE rtsp://foo/twister RTSP/1.0
>           CSeq: 1
>
>   S->C: RTSP/1.0 200 OK
>           CSeq: 1
>           Transport: RTP/AVP;server-rtp-port=4056;server-rtcp-port=4057
>           Content-Type: application/sdp
>           Content-Length: xxx
>
>           v=0
>           o=- 2890844256 2890842807 IN IP4 172.16.2.93
>           s=RTSP Session
>           etc...
>
>Also you can include this informationin SDP,
>which may be better since the client may get the SDP
>by some other means, right?
>
>
>
Actually you can include this information in the SDP without any extra
extensions is just the matter of defining the usage. The RTP port number
can be located in its normal place on the m= line. The RTCP port can be
set by the attribute defined by
draft-ietf-mmusic-sdp4nat-03.txt. To make this in a good fashion the
server should define a RTSP feature tag that indicates this
functionality and that the server should provide the information when
sending the DESCRIBE response.

I think we need to further discuss the RTSP NAT traversal solutions and
possible optimization. My draft was intended to be a common ground to
discuss around.

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 Phillipe,

Comments below:

philippe.gentric@philips.com wrote:

>you write:
>
>  
>
>>My current view is
>>that due to the lack of a well defined demultiplexing structure between
>>media protocol and STUN on the servers ports makes STUN solutions for
>>symmetric NATs a bad choice. As pointed out you need to compare the
>>media protocols bit structure with STUN request. This is not safe
>>operation. I do agree that RTP and RTCP seems to be reasonable safe to
>>demultiplex, however it is impossible to know that all protocols media
>>protocols will work.
>>    
>>
>
>Well I dont know about other protocols but the use of STUN in the RTSP scenario is *extremely* safe:
>
>1) your client sends and receives STUN on the UDP ports it has choosen for RTP and RTCP
>_but_ since this is *before* the SETUP  there is *only* a STUN client behind these UDP ports,
>(the RTP/RTCP stack does not have to be initialized yet). And vice versa
>i.e. as soon as STUN completed you can stop the STUN client to
>start the RTP/RTCP session !
>
>=> so on the client side it is totally safe.
>  
>
I don't think that client implementation has been my major concern when 
it comes to STUN.

>2) the server does NOT listen for incomming data on the RTP ports:
>there will be the STUN server listening behind these ports, nothing else
>
>=> no possible collision of STUN with RTP on the server side
>  
>
I don't agree with you here. What I believe will happen is that a server 
will use a single port pair for sending all the sessions it needs to be 
STUNed. This will result in that these port will receive traffic from 
other clients that already has established their sessions. Therefore 
there will be a flow of both RTP or RTCP and STUN packets. One must also 
consider any mapping keep alive, especially on the RTP port. This can be 
either RTP or STUN packets. In the draft I recommend the usage of empty 
RTP packets.

>3) then the server needs to  distinguish between RTCP RR
>and STUN on the RTCP port but the headers are _really_ different.
>
>for starters the first bit is different  (1 for RTCP, 0 for STUN)!
>

I agree that there is a pretty good difference here, but RTP and RTCP 
first byte was never designed to be a demultiplexing tool. What happens 
if a server or client would send RAT packets that actually has a version 
number of 0. Then you don't have this nice difference anymore.

>but let us imagine that a STUN packet (of an hypothetical future extended version!)
>will by chance match the first 32 bits of a paussible RTCP RR packet,
>then what is the probability of having the 32 bits SSRC (random, fixed per transaction)
>match the first 32 bits of the 128 STUn tID (random, changing every message) ?
>2 to the power 64 ?
>  
>
Actually having one fixed and the other being random only increases the 
chance of collisions. Of course it will only happened with one STUN 
message not the next. However the whole idea requires both STUN and 
RTP/RTCP to handle packets that are routed wrong. Of course any good 
implementation will anyway have these properties.

>  
>
>>As explained before this is not a clean solution.
>>    
>>
>
>"Striving to better, oft we mar what's well." (William Shakespeare)
>  
>

Yes, the perfect are in many case the enemy of the good. However I don't 
really see how this proverb apply here. First of all, any NAT solution 
is basically a deal with the devil. Secondly I am not trying to better 
anything. Only use what is there and does work, i.e. multiplexing points 
that are designed to be it.

Cheers

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:

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.

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). 


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.

Hence I prefer the inclusion of SessionID.

- thomas




-----Original Message-----
From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]
Sent: Tuesday, February 25, 2003 3:46 AM
Cc: mmusic@ietf.org
Subject: [MMUSIC] I-D ACTION:draft-ietf-mmusic-rtsp-nat-00.txt


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		: How to make Real-Time Streaming Protocol (RTSP) 
                          traverse Network Address Translators (NAT) and 
                          interact with Firewalls
	Author(s)	: M. Westerlund
	Filename	: draft-ietf-mmusic-rtsp-nat-00.txt
	Pages		: 18
	Date		: 2003-2-24
	
This document describes four different types of NAT traversal
techniques that can be used by RTSP. For each technique a description
on how it shall be used, what security implications it has and other
deployment considerations that exist is given. Further a description
on how RTSP relates to firewalls are also given.

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