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Versions: (draft-wenger-avt-topologies) 00 01 02 03 04 05 06 07 RFC 5117

Network Working Group                                Magnus Westerlund
INTERNET-DRAFT                                                Ericsson
Expires: March 2007                                     Stephan Wenger
                                                                 Nokia

                                                    September 17, 2006

                             RTP Topologies
                   draft-ietf-avt-topologies-01.txt>


Status of this Memo

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Copyright Notice

   Copyright (C) The Internet Society (2006).

Abstract

   This document disucsses multi-endpoint topologies commonly used in
   RTP based environments.  In particular, centralized topologies
   commonly employed in the video conferencing industry are mapped to
   the RTP terminology.







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TABLE OF CONTENTS

Status of this Memo................................................1
Copyright Notice...................................................1
Abstract...........................................................1
TABLE OF CONTENTS..................................................2
1. Introduction....................................................3
2. Definitions.....................................................3
  2.1. Glossary...................................................3
  2.2. Terminology................................................3
  2.3. Topologies.................................................4
     2.3.1. TOPO10: Point to Point................................4
     2.3.2. TOPO20: Point to Multi-point using Multicast..........4
     2.3.3. TOPO30: Point to Multipoint using the RFC 3550
     translator...................................................5
     2.3.4. TOPO40: Point to Multipoint using the RFC 3550 mixer
     model........................................................8
     2.3.5. TOPO50: Point to Multipoint using video switching MCU 10
     2.3.6. TOPO60: Point to Multipoint using RTCP-terminating MCU11
     2.3.7. Combining Topologies.................................12
3. Security Considerations........................................13
4. IANA Considerations............................................13
5. Acknowledgements...............................................13
6. References.....................................................14
  6.1. Normative references......................................14
  6.2. Informative references....................................14
7. Authors' Addresses.............................................14
8. List of Changes relative to previous drafts....................15
RFC Editor Considerations.........................................16























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1.  Introduction

   When working on the Codec Control Messages [CCM], we noticed a
   considerable confusion in the community with respect to terms such
   as MCU, mixer, and translator.  In the process of writing, we
   became increasingly unsure of our own understanding, and therefore
   added what became the core of this draft to the CCM draft.  Later,
   it was found that this information has its own value, and was
   "outsourced" from the CCM draft into the present memo.

   It could be argued that this document clarifies and explains
   sections of the RTP spec [RFC3550], and is therefore of
   informational nature.  In this case, the present memo may end up
   as an informational RFC.

   When the Audio-Visual Profile with Feedback (AVPF) [AVPF] was
   developed, the main emphasis lied in the efficient support of
   point-to-point and small multipoint scenarios without centralized
   multipoint control.  However, in practice, many small multipoint
   conferences operate utilizing devices known as Multipoint Control
   Units (MCUs).  MCUs comprise mixers and translators (in RTP
   [RFC3550] terminology), but also signalling support



2.  Definitions


2.1.    Glossary

   ASM    - Asynchronous Multicast
   AVPF   - The Extended RTP Profile for RTCP-based Feedback
   MCU    - Multipoint Control Unit
   PtM    - Point to Multipoint
   PtP    - Point to Point


2.2.    Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL
   NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED",  "MAY", and
   "OPTIONAL" in this document are to be interpreted as described in
   RFC 2119 [RFC2119].







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2.3.    Topologies

   This subsection defines several basic topologies that are relevant
   for codec control. The first four relate to the RTP system model
   utilizing multicast and/or unicast, as envisioned in RFC 3550.
   The last two topologies, in contrast, describe the widely deployed
   system model as used in most H.323 video conferences, where both
   the media streams and the RTCP control traffic terminate at the
   MCU.  More topologies can be constructed by combining any of the
   models, see Section 2.3.7.
   The topologies may be referenced by a shortcut name, indicated by
   the prefix "Topo-".

2.3.1.      Point to Point

   Shortcut name: Topo-Point-to-Point

   The Point to Point (PtP) topology (Figure 1) consists of two end-
   points with unicast capabilities between them.  Both RTP and RTCP
   traffic are conveyed endpoint to endpoint using unicast traffic
   only (even if this unicast traffic happens to be conveyed over an
   IP-multicast address).

      +---+         +---+
      | A |<------->| B |
      +---+         +---+

   Figure 1 - Point to Point

   The main property of this topology is that A sends to B and only
   B, while B sends to A and only A. This avoids all complexities of
   handling multiple endpoints and combining the requirements from
   them.  Do note that an endpoint may still use multiple RTP
   Synchronization Sources (SSRCs) in an RTP session.


2.3.2.      Point to Multi-point using Multicast

   Shortcut name: Topo- Multicast














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                 +-----+
      +---+     /       \    +---+
      | A |----/         \---| B |
      +---+   /   Multi-  \  +---+
             +    Cast     +
      +---+   \  Network  /  +---+
      | C |----\         /---| D |
      +---+     \       /    +---+
                 +-----+

   Figure 2 - Point to Multipoint using Multicast

   We define Point to Multipoint (PtM) using multicast topology as a
   transmission model in which traffic from any participant reaches
   all the other participants, except for cases such as
     o packet loss occurs,
     o a participant does not wish to receive the traffic
       for a specific media stream, and therefore has not
       subscribed to the IP multicast group in question.

   In this sense, "traffic" encompasses both RTP and RTCP traffic.
   The number of participants can be between one and many -- as RTP
   and RTCP scales to very large multicast groups (the theoretical
   limit of RTP is approximately two billion participants).

   This draft is primarily interested in the subset of multicast
   session where the number of participants in the multicast group
   allows the participants to use early or immediate feedback as
   defined in AVPF.  This document refers to those groups as as
   "small multicast groups".


2.3.3.      Point to Multipoint using the RFC 3550 translator

   Shortcut name: Topo-Translator

   Two main categories of Translators can be distinguished.

   Transport Translators do not modify the media stream itself, but
   are concerned with transport parameters.  Transport parameters, in
   the sense of this section, comprise the transport addresses to
   bridge different domains, and the media packetization to allow
   other transport protocols to be interconnected to a session
   (gateways).

   Media Translators, in contrast, modify the media stream itself.
   This process is commonly known as transcoding.  The modification
   of the media stream can be as small as removing parts of the




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   stream, and can go all the way to a full transcoding utilizing a
   different media codec.   Media translators are commonly used to
   connect entities without a common interoperability point.

   Stand-alone Media Translators are rare.  Most commonly, a
   combination of Transport and Media Translators are used to
   translate both the media stream and the transport aspects of a
   stream between two transport domains (or clouds).

   Both Translator types share common attributes that separates them
   from mixers.  For each media stream that the translator receives,
   it generates an individual stream in the other domain.  However, a
   translator maintains a complete view of all existing participants
   between both domains. Therefore, the SSRC space is shared across
   the two domains.

   The RTCP translation process can be trivial, for example when
   Transport translators just need to adjust IP addresses, and can be
   quite complex in the case of media translators.  See section 7.2
   of [RFC 3550].


                 +-----+
      +---+     /       \     +------------+      +---+
      | A |<---/         \    |            |<---->| B |
      +---+   /   Multi-  \   |            |      +---+
             +    Cast     +->| Translator |
      +---+   \  Network  /   |            |      +---+
      | C |<---\         /    |            |<---->| D |
      +---+     \       /     +------------+      +---+
                 +-----+

   Figure 3 - Point to Multipoint using a Translator

   Figure 3 depicts an example of a Transport Translator performing
   at least IP address translation.  It allows the (non multicast
   capable) participants B and D to take part in a multicasted
   session by having the translator forward their unicast traffic to
   the multicast addresses in use, and vice versa.  It must also
   forward B's traffic to D and vice versa, to provide each of B and
   D with a complete view of the session.

   If B were behind a limited link, the translator may perform media
   transcoding to allow the traffic received from the other
   participants to reach B without overloading the link.

   When in the example depicted in Figure 3 the translator acts only
   as a Transport Translator, then the RTCP traffic can simply be
   forwarded, similar to the media traffic.  However, when media




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   translation occurs, the translator's task becomes substantially
   more complex even with respect to the RTCP traffic.  In this case,
   the translator needs to rewrite B's RTCP receiver report, before
   forwarding them to D and the multicast network.  The rewriting is
   needed as the stream received by B is not the same stream as the
   other participants receive. For example, the number of packets
   transmitted to B may be lower than what D receives, due to the
   different media format. Therefore, if the receiver reports were
   forwarded without changes, the extended highest sequence number
   would indicate that B were substantially behind in reception --
   while it most likely it would not be. Therefore, the translator
   must translate that number to a corresponding sequence number for
   the stream the translator received.  Similar arguments can be made
   for most other fields in the RTCP receiver reports.

   As specified in Section 7.1 of [RFC3550] the SSRC space is common
   for all participants in the session, independent of which side
   they are of the translator. Thus it is the responsibility of the
   participants to run SSRC collision detection, and the SSRC a field
   the translator should not change.

      +---+      +------------+      +---+
      | A |<---->| Multipoint |<---->| B |
      +---+      |  Control   |      +---+
                 |   Unit     |
      +---+      |   (MCU)    |      +---+
      | C |<---->|            |<---->| D |
      +---+      +------------+      +---+

   Figure 4 - MCU with RTP Translator (relay) with only unicast links

   A common MCU scenario is the one depicted in Figure 4.  Herein,
   the MCU connects multiple users of a conference through unicast.
   This can be implemented using a very simple transport translator,
   which could be called a relay. The relay forwards all traffic it
   receives, both RTP and RTCP, to all other participants. In doing
   so, a multicast network is emulated without relying on a multicast
   capable network structure.

   A translator normally does not use an SSRC of its own, and is not
   visible as an active participant in the session. However, it may
   act as a media receiver, thus have an SSRC, and use RTCP to report
   reception statistics.  However this behavior should only be used
   when it is really desirable to have this feedback, i.e. having it
   act as special type of quality monitor.

   It also needs to be noted that the translator, in some cases, may
   act on behalf of the "real" source and respond to codec control
   messages. in his capacity as media translator. This for example




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   may occur if a receiver requests a bandwidth reduction, and the
   media translator has not detected any congestion or other reasons
   for bandwidth reduction between the media source and itself. In
   that case, a translator should be able to react to codec control
   messages, as it is capable of fulfilling the request on behalf of
   the media sender. If it wouldn't react to codec control, and
   therefore couldn't fullfil the request, the media quality in the
   media senders domain would suffer.


2.3.4.      Point to Multipoint using the RFC 3550 mixer model

   Shortcut name: Topo-Mixer

   A mixer is a middlebox that aggregates multiple RTP streams that
   are part of a session, by mixing the media data and generating a
   new RTP stream.  One common application for a mixer is to allow a
   participant to receive a session with a reduced amount of
   resources.

                 +-----+
      +---+     /       \     +-----------+      +---+
      | A |<---/         \    |           |<---->| B |
      +---+   /   Multi-  \   |           |      +---+
             +    Cast     +->|   Mixer   |
      +---+   \  Network  /   |           |      +---+
      | C |<---\         /    |           |<---->| D |
      +---+     \       /     +-----------+      +---+
                 +-----+

   Figure 5 - Point to Multipoint using RFC 3550 mixer model

   A mixer can be viewed as a device terminating the media streams
   received from other session participants.  Using the media data
   from the received media streams, a mixer generates a media stream
   that is sent to the session participant.

   The content that the mixer provides is the mixed aggregate of what
   the mixer receives from the PtP or PtM links, which are part of
   the same conference session.

   The mixer is the content source, as it mixes the content (often in
   the uncompressed domain) and then encodes it for transmission to a
   participant. The CC and CSRC fields in the RTP header are used to
   indicate the contributors of to the newly generated stream.  The
   SSRCs of the to-be-mixed streams on the mixer input appear as the
   CSRCs at the mixer output.  That output stream uses a new SSRC
   that identifies the Mixer.  The CSRC are forwarded between the two
   domains to allow for loop detection and identification of sources




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   that are part of the global session. Note that Section 7.1 of RFC
   3550 requires the SSRC space to be shared between domains for
   these reasons.

   The mixer is responsible for generating RTCP packets in accordance
   with its role. It is a receiver and should therefore send
   reception reports for the media streams it receives. As a media
   sender itself it should also generate sender report for those
   media streams sent.  The content of the SRs created by the mixer
   may or may not take into account the situation on its receiving
   side.  Similarly, the content of RRs created by the mixer may or
   may not be based on the situation on the mixer's sending side.
   This is left open to the implementation.  As specified in Section
   7.3 of RFC 3550, a mixer must not forward RTCP unaltered between
   the two domains.

   The mixer depicted in Figure 5 has three domains that needs to be
   separated; the multicast network, participant B and participant D.
   The Mixer produces different mixed streams to B and D, as the one
   to B may contain D and vice versa. However the mixer does only
   need one SSRC in each domain that is the receiving entity and
   transmitter of mixed content.

   In the multicast domain, the mixer does not need to provide a
   mixed view of the other domains and will commonly only forward the
   media from B and D into the multicast network using B's and D's
   SSRC.

   The mixer is responsible for receiving the codec control messages
   and handles them appropriately.  The definition of "appropriate"
   depends on the message itself and the context. In some cases, the
   reception of a codec control message may result in the generation
   and transmission of codec control messages by the mixer to the
   participants in the other domain. In other cases, a message is
   handled by the mixer itself and therefore not forwarded to any
   other domains.

   It should be noted that this form of mixing technology is not
   widely deployed.  Most multipoint video conferences used today
   employ one of the models discussed in the next sections.

   When replacing the multicast network in Figure 5 (to the left of
   the mixer) with individual unicast links as depicted in Figure 6,
   the mixer model is very similar to the one discussed in section
   2.3.6 below.








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      +---+      +------------+      +---+
      | A |<---->| Multipoint |<---->| B |
      +---+      |  Control   |      +---+
                 |   Unit     |
      +---+      |   (MCU)    |      +---+
      | C |<---->|            |<---->| D |
      +---+      +------------+      +---+

   Figure 6 - RTP Mixer with only unicast links




2.3.5.      Point to Multipoint using video switching MCU

   Shortcut name: Topo- Video-switch-MCU

      +---+      +------------+      +---+
      | A |------| Multipoint |------| B |
      +---+      |  Control   |      +---+
                 |   Unit     |
      +---+      |   (MCU)    |      +---+
      | C |------|            |------| D |
      +---+      +------------+      +---+

   Figure 7 - Point to Multipoint using relaying MCU

   This PtM topology is, today, still deployed, although the RTCP-
   terminating MCUs, as discussed in the next section, are perhaps
   more common..  this topology, as well as the following one,
   reflect today's lack of wide availability of IP multicast
   technologies , as well as the simplicity of content switching when
   compared to content mixing.  The technology is commonly
   implemented in what is known as "Video Switching MCUs".

   A video switching MCU forwards to a participant a single media
   stream, selected from the available streams.  The criteria for
   selection are often based on voice activity in the audio-visual
   conference, but other conference management mechanisms (like
   presentation mode or explicit floor control) are known to exist as
   well.

   The video switching MCU may also perform media translation to
   modify the content in bit-rate, encoding, resolution; however it
   still may indicate the original sender of the content through the
   SSRC.  In this case the values of the CC and CSRC fields are
   retained.






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   If not terminating RTP, the RTCP Sender Reports are forwarded for
   the currently selected sender. All RTCP receiver reports are
   freely forward between the participants. In addition, the MCU may
   also originate RTCP control traffic in order to control the
   session and/or report on status from its viewpoint.

   The video switching MCU has mostly the attributes of a translator.
   However its stream selection is a mixing behaviour. This behaviour
   has some RTP and RTCP issues associated with it. The suppression
   of all but one media stream results in that most participants see
   only a subset of the sent media streams at any given time; often a
   single stream per conference. Therefore, RTCP receiver reports
   only report on these streams.  In consequence, the media senders
   that are not currently forwarded receive a view of the session
   that indicates their media streams disappearing somewhere en
   route. This makes the use of RTCP for congestion control very
   problematic. To avoid these issues the MCU needs to modify the
   RTCP RRs.


2.3.6.      Point to Multipoint using RTCP-terminating MCU

   Shortcut name: Topo-RTCP-terminating-MCU

      +---+      +------------+      +---+
      | A |<---->| Multipoint |<---->| B |
      +---+      |  Control   |      +---+
                 |   Unit     |
      +---+      |   (MCU)    |      +---+
      | C |<---->|            |<---->| D |
      +---+      +------------+      +---+

   Figure 8 - Point to Multipoint using content modifying MCU

   In this PtM scenario, each participant runs an RTP point-to-point
   session between itself and the MCU, this is the mostly deployed
   topology. The content that the MCU provides to each participant is
   either:

     a) A selection of the content received from the other
        participants, or

     b) The mixed aggregate of what the MCU receives from the other
        PtP links, which are part of the same conference session.

   In case a) the MCU may modify the content in bit-rate, encoding,
   resolution. No explicit RTP mechanism is used to establish the
   relationship between the original media sender and the version the
   MCU sends.  In other words, the outgoing session typically uses a




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   different SSRC, and may well use a different PT, even if this
   different PT happens to be mapped to the same media type.  (This
   is the definition of this topology and distinguishes it from the
   topologies previously discussed).

   In case b) the MCU is the content source as it mixes the content
   and then encodes it for transmission to a participant. The
   participant's content that is included in the aggregated content
   is not indicated through any explicit RTP mechanism.  For example,
   regardless of the number of streams that are aggregated, in the
   MCU generated streams CC is zero and therefore no CSRC fields are
   present (this is true for most shipping MCUS). The participants
   contributing to the mix are reported using signalling mechanism
   like conference event package in SIP.

   The MCU is responsible for receiving the codec control messages
   and handle them appropriately. In some cases, the reception of a
   codec control message may result in the generation and
   transmission of codec control messages by the MCU to some or all
   of the other participants.

   An MCU may transparently relay some codec control messages and
   intercept, modify, and (when appropriate) generate codec control
   messages of its own and transmit them to the media senders.

   The main feature that sets this topology apart from what RFC 3550
   describes, is the lack of an explicit RTP level indication of all
   participants. If one were using the mechanisms available in RTP
   and RTCP to signal this explicitly, the topology would follow the
   approach of an RTP mixer. The lack of explicit indication has at
   least the following potential problems:

    1) Loop detection cannot be performed on the RTP level.  When
        carelessly connecting two misconfigured MCUs, a loop could be
        generated.
    2) There is no information about active media senders available
        in the RTP packet.  As this information is missing, receivers
        cannot use it.  It also deprive the participant's clients
        information about who are actively sending in a machine
        usable way. Thus preventing clients from doing indication of
        currently active speakers in user interfaces, etc. It is
        known in the signaling layer.



2.3.7.      Combining Topologies

   Topologies can be combined and linked to each other using mixers
   or translators. Care must however be taken to how the SSRC space
   is handled, mixers separate the SSRC space into two parts, while



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   translators maintain the space across themselves. Any hybrid, like
   the video switching MCU, 2.3.5, requires considerable afterthought
   on how RTCP is dealt with. But do note that the SSRC uniquenss
   always needs to global across the different domains.


3.  Security Considerations

   The usage of mixers and translators do have impact on security and
   the security functions used. The primary issue is that both mixers
   and translators do modify packets, thus preventing the usage of
   integrity and source authentication unless they are a trusted
   device which takes part of the security context. If encryption is
   employed the media translator and mixers will need to be able to
   decrypt the media to perform its function. A transport translator
   may be used without access to the security association in cases
   they touches parts that are not included in the integrity
   protection, for example IP address and UDP port numbers in a media
   stream using SRTP [RFC3711]. However in general the translator or
   mixer needs to be part of the signalling context and get the
   necessary security associations established with its RTP session
   participants.
   Including the mixer and translator in the security context allows
   the entity if subverted or misbehaving to perform a number of very
   serious attacks as it has full access. It can perform all the
   attacks possible, see RFC 3550 and any applicable profiles, as if
   the media session was not protected at all, while giving the
   impression to the session participants that they are protected
   against them.


4.  IANA Considerations

   This document specifies no actions for IANA.


5.  Acknowledgements

   The authors would like to thank N.N.














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6.  References


6.1.    Normative references

   [AVPF]   Ott, J., Wenger, S., Sato, N., Burmeister, C., and J.
            Rey, "Extended RTP Profile for Real-time Transport
            Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC
            4585, July 2006.
   [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
            Requirement Levels", BCP 14, RFC 2119, March 1997.
   [RFC3550] Schulzrinne, H.,  Casner, S., Frederick, R., and V.
            Jacobson, "RTP: A Transport Protocol for Real-Time
            Applications", STD 64, RFC 3550, July 2003.
   [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
            Video Conferences with Minimal Control", STD 65, RFC
            3551, July 2003.
   [RFC3711] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and
            K. Norrman, "The Secure Real-time Transport Protocol
            (SRTP)", RFC 3711, March 2004.



6.2.    Informative references


   Any 3GPP document can be downloaded from the 3GPP web server,
   "http://www.3gpp.org/", see specifications.


7.  Authors' Addresses

   Magnus Westerlund
   Ericsson Research
   Ericsson AB
   SE-164 80 Stockholm, SWEDEN

   Phone: +46 8 7190000
   EMail: magnus.westerlund@ericsson.com


   Stephan Wenger
   Nokia Corporation
   P.O. Box 100
   FIN-33721 Tampere
   FINLAND

   Phone: +358-50-486-0637
   EMail: stewe@stewe.org




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8.  List of Changes relative to previous drafts



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INTERNET-DRAFT               RTP Topologies        September 17, 2006


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