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Network Working Group                                      M. Westerlund
Internet-Draft                                                  Ericsson
Intended status: Standards Track                              C. Perkins
Expires: April 25, 2013                            University of Glasgow
                                                        October 22, 2012


        Multiple RTP Sessions on a Single Lower-Layer Transport
           draft-westerlund-avtcore-transport-multiplexing-04

Abstract

   This document specifies how multiple RTP sessions are to be
   multiplexed on the same lower-layer transport, e.g. a UDP flow.  It
   discusses various requirements that have been raised and their
   feasibility, which results in a solution with a certain
   applicability.  A solution is recommended and that solution is
   provided in more detail, including signalling and examples.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
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   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
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   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."

   This Internet-Draft will expire on April 25, 2013.

Copyright Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
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   publication of this document.  Please review these documents
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   include Simplified BSD License text as described in Section 4.e of



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   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.


Table of Contents

   1.  Introduction . . . . . . . . . . . . . . . . . . . . . . . . .  4
   2.  Conventions  . . . . . . . . . . . . . . . . . . . . . . . . .  4
     2.1.  Terminology  . . . . . . . . . . . . . . . . . . . . . . .  4
     2.2.  Requirements Language  . . . . . . . . . . . . . . . . . .  5
   3.  Motivations  . . . . . . . . . . . . . . . . . . . . . . . . .  5
     3.1.  NAT and Firewalls  . . . . . . . . . . . . . . . . . . . .  5
     3.2.  No Transport Level QoS . . . . . . . . . . . . . . . . . .  5
     3.3.  Multiple RTP sessions  . . . . . . . . . . . . . . . . . .  6
     3.4.  Usage of RTP Extensions  . . . . . . . . . . . . . . . . .  6
     3.5.  Incremental Deployment . . . . . . . . . . . . . . . . . .  7
     3.6.  Summary  . . . . . . . . . . . . . . . . . . . . . . . . .  7
   4.  Requirements . . . . . . . . . . . . . . . . . . . . . . . . .  7
     4.1.  Support Use of Multiple RTP Sessions . . . . . . . . . . .  7
     4.2.  Same SSRC Value in Multiple RTP Sessions . . . . . . . . .  8
     4.3.  SRTP . . . . . . . . . . . . . . . . . . . . . . . . . . .  8
     4.4.  Don't Redefine Used Bits . . . . . . . . . . . . . . . . .  9
     4.5.  Firewall Friendly  . . . . . . . . . . . . . . . . . . . .  9
     4.6.  Monitoring and Reporting . . . . . . . . . . . . . . . . .  9
     4.7.  Usable Also Over Multicast . . . . . . . . . . . . . . . . 10
     4.8.  Incremental Deployment . . . . . . . . . . . . . . . . . . 10
   5.  Design Considerations  . . . . . . . . . . . . . . . . . . . . 10
     5.1.  Location of SHIM . . . . . . . . . . . . . . . . . . . . . 10
     5.2.  ICE and DTLS-SRTP Integration  . . . . . . . . . . . . . . 12
     5.3.  Signalling Fallback  . . . . . . . . . . . . . . . . . . . 12
   6.  Specification  . . . . . . . . . . . . . . . . . . . . . . . . 13
     6.1.  Shim Layer . . . . . . . . . . . . . . . . . . . . . . . . 14
     6.2.  Signalling . . . . . . . . . . . . . . . . . . . . . . . . 17
     6.3.  SRTP Key Management  . . . . . . . . . . . . . . . . . . . 18
       6.3.1.  Security Description . . . . . . . . . . . . . . . . . 19
       6.3.2.  DTLS-SRTP  . . . . . . . . . . . . . . . . . . . . . . 19
       6.3.3.  MIKEY  . . . . . . . . . . . . . . . . . . . . . . . . 19
     6.4.  Examples . . . . . . . . . . . . . . . . . . . . . . . . . 20
       6.4.1.  RTP Packet with Transport Header . . . . . . . . . . . 20
       6.4.2.  SDP Offer/Answer example . . . . . . . . . . . . . . . 21
   7.  Open Issues  . . . . . . . . . . . . . . . . . . . . . . . . . 25
   8.  IANA Considerations  . . . . . . . . . . . . . . . . . . . . . 26
   9.  Security Considerations  . . . . . . . . . . . . . . . . . . . 26
   10. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 26
   11. References . . . . . . . . . . . . . . . . . . . . . . . . . . 27
     11.1. Normative References . . . . . . . . . . . . . . . . . . . 27
     11.2. Informational References . . . . . . . . . . . . . . . . . 27
   Appendix A.  Possible Solutions  . . . . . . . . . . . . . . . . . 29



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     A.1.  Header Extension . . . . . . . . . . . . . . . . . . . . . 29
     A.2.  Multiplexing Shim  . . . . . . . . . . . . . . . . . . . . 30
     A.3.  Single Session . . . . . . . . . . . . . . . . . . . . . . 31
     A.4.  Use the SRTP MKI field . . . . . . . . . . . . . . . . . . 32
     A.5.  Use an Octet in the Padding  . . . . . . . . . . . . . . . 33
     A.6.  Redefine the SSRC field  . . . . . . . . . . . . . . . . . 33
   Appendix B.  Comparison  . . . . . . . . . . . . . . . . . . . . . 34
     B.1.  Support of Multiple RTP Sessions Over Single Transport . . 34
     B.2.  Enable Same SSRC Value in Multiple RTP Sessions  . . . . . 34
       B.2.1.  Avoid SSRC Translation in Gateways/Translation . . . . 34
       B.2.2.  Support Existing Extensions  . . . . . . . . . . . . . 35
     B.3.  Ensure SRTP Functions  . . . . . . . . . . . . . . . . . . 35
     B.4.  Don't Redefine Used Bits . . . . . . . . . . . . . . . . . 36
     B.5.  Firewall Friendly  . . . . . . . . . . . . . . . . . . . . 37
     B.6.  Monitoring and Reporting . . . . . . . . . . . . . . . . . 38
     B.7.  Usable over Multicast  . . . . . . . . . . . . . . . . . . 39
     B.8.  Incremental Deployment . . . . . . . . . . . . . . . . . . 39
     B.9.  Summary and Conclusion . . . . . . . . . . . . . . . . . . 40
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 41
































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1.  Introduction

   There has been renewed interest for having a solution that allows
   multiple RTP sessions [RFC3550] to use a single lower layer
   transport, such as a bi-directional UDP flow.  The main reason is the
   cost of doing NAT/FW traversal for each individual flow.  ICE and
   other NAT/FW traversal solutions are clearly capable of attempting to
   open multiple flows.  However, there is both increased risk for
   failure and an increased cost in the creation of multiple flows.  The
   increased cost comes as slightly higher delay in establishing the
   traversal, and the amount of consumed NAT/FW resources.  The latter
   might be an increasing problem in the IPv4 to IPv6 transition period.

   There is ongoing work on specifying how and when one RTP session may
   contain multiple media types
   [I-D.ietf-avtcore-multi-media-rtp-session].  That addresses certain
   use cases, while this proposal addresses a different set of use cases
   and motivations.  This is further discussed in the section on
   Motivations (Section 3).  The classical method of having one RTP
   session over a specific transport flow is still motivated for a
   number of use cases, especially when flow based QoS is to be used for
   some media streams.

   This document draws up some requirements for consideration on how to
   transport multiple RTP sessions over a single lower-layer transport.
   These requirements had to be weighted as the combined set of
   requirements result in that no known solution exist that can fulfill
   them completely.

   A number of possible solutions where considered and discussed with
   respect to their properties.  Based on that, the authors recommended
   a shim layer variant as single solution, which specified in detail
   including signalling solution and examples.  The other considered
   proposals and the comparison is available as appendices.


2.  Conventions

2.1.  Terminology

   Some terminology used in this document.

   Multiplexing:  Unless specifically noted, all mentioning of
      multiplexing in this document refer to the multiplexing of
      multiple RTP Sessions on the same lower layer transport.  It is
      important to make this distinction as RTP does contain a number of
      multiplexing points for various purposes, such as media formats
      (Payload Type), media sources (SSRC), and RTP sessions.



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2.2.  Requirements Language

   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].


3.  Motivations

   This section looks at the motivations why an additional solution is
   needed assuming that you can do both the classical method of having
   one RTP session per transport flow as defined by the RTP
   specification [RFC3550] and when you have multiple media types within
   one RTP session [I-D.ietf-avtcore-multi-media-rtp-session].

3.1.  NAT and Firewalls

   The existence of NATs and Firewalls at almost all Internet access has
   had implications on protocols like RTP that were designed to use
   multiple transport flows.  First of all, the NAT/FW traversal
   solution one uses needs to ensure that all these transport flows are
   established.  This has three different impacts:

   1.  Increased delay to perform the transport flow establishment

   2.  The more transport flows, the more state and the more resource
       consumption in the NAT and Firewalls.  When the resource
       consumption in NAT/FWs reaches their limits, unexpected behaviors
       usually occur.  Commonly resulting in service disruptions.

   3.  More transport flows means a higher risk that some transport flow
       fails to be established, thus preventing the application to
       communicate.

   Using fewer transport flows reduces the risk of communication
   failure, improved establishment behavior and less load on NAT and
   Firewalls.

3.2.  No Transport Level QoS

   Many RTP-using applications don't utilize any network level Quality
   of Service functions.  Nor do they expect or desire any separation in
   network treatment of its media packets, independent of whether they
   are audio, video or text.  When an application has no such desire, it
   doesn't need to provide a transport flow structure that simplifies
   flow based QoS.





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3.3.  Multiple RTP sessions

   The usage of multiple RTP sessions allow separation of media streams
   that have different usages or purposes in an RTP based application,
   for example to separate the video of a presenter or most important
   current talker, from those of the listeners that not all end-points
   receive.  Also separation for different processing based on media
   types such as audio and video in end-points and central nodes.  Thus
   providing the node with the knowledge that any SSRC within the
   session is supposed to be processed in a similar or same way.

   For simpler cases, where the streams within each media type need the
   same processing, it is clearly possible to find other multiplex
   solutions, for example based on the Payload Type and the differences
   in encoding that the payload type allows to describe.  This may
   anyhow be insufficient when you get into more advanced usages where
   you have multiple sources of the same media type, but for different
   usages or as alternatives.  For example when you have one set of
   video sources that shows session participants and another set of
   video sources that shares an application or presentation slides, you
   likely want to separate those streams for various reasons such as
   control, prioritization, QoS, methods for robustification, etc.  In
   those cases, using the RTP session for separation of properties is a
   powerful tool.  A tool with properties that need to be preserved when
   providing a solution for how to use only a single lower-layer
   transport.

   For more discussion of the usage of RTP sessions verses other
   multiplexing we recommend RTP Multiplexing Architecture
   [I-D.westerlund-avtcore-multiplex-architecture].

3.4.  Usage of RTP Extensions

   Applications uses different sets of RTP extensions.  The solution for
   multiple media types in one RTP session
   [I-D.ietf-avtcore-multi-media-rtp-session] is known to have
   limitations that prevent the usage of the following RTP mechanisms
   and extensions:

   o  XOR FEC (RFC5109)

   o  RTP Retransmission in session mode (RFC4588)

   o  Certain Layered Coding

   A developed solution should minimize the number of RTP/RTCP extension
   and mechanism that can't be used.




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3.5.  Incremental Deployment

   In various multi-party communication scenarios deployment can become
   an issue if all session participants are required to have the
   functionality before enabling its usage.  This is especially
   difficult in communication scenarios where not all possible
   participants and their capabilities are know ahead of establishing
   the communication session with some sub-set of the participants.  At
   least for centralized communication sessions it is desirable to have
   a solution that enables allows the solution to be used on a single
   leg without affecting any other leg, nor require advanced translation
   functionality in any central node.

3.6.  Summary

   The center of the motivation is to ensure that the RTP session is a
   available and usable tool also for applications that has no need for
   network level separation of its media streams and wants to reduce its
   exposure to any NAT or Firewall inconsistencies and minimize the
   resource consumption.  As a benefit a well designed solution will
   enable incremental deployment and minimal limitations in what
   existing RTP mechanisms or extensions that can be used by the RTP
   using application.


4.  Requirements

   This section lists and discusses a number of potential requirements.
   However, it is not difficult to realize that it is in fact possible
   to put requirements that makes the set of feasible solutions an empty
   set.  It is thus necessary to consider which requirements that are
   essential to fulfill and which can be compromised on to arrive at a
   solution.

4.1.  Support Use of Multiple RTP Sessions

   Section 3.3 discusses a number of reasons why an application may like
   to have multiple RTP sessions.  Considering the motivations for this
   work this must be an absolute requirement.  We also are of the
   opinion that the session provided by the solution must fulfill the
   definition in the RTP [RFC3550] specification:

      "The distinguishing feature of an RTP session is that each
      maintains a full, separate space of SSRC identifiers (defined
      next).  The set of participants included in one RTP session
      consists of those that can receive an SSRC identifier transmitted
      by any one of the participants either in RTP as the SSRC or a CSRC
      (also defined below) or in RTCP."



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4.2.  Same SSRC Value in Multiple RTP Sessions

   Two different RTP sessions being multiplexed on the same lower layer
   transport need to be able to use the same SSRC value.  This is a
   absolute requirement, for two reasons:

   1.  To avoid mandating SSRC assignment rules that are coordinated
       between the sessions.  If the RTP sessions multiplexed together
       must have unique SSRC values, then additional code that works
       between RTP Sessions is needed in the implementations.  Thus
       raising the bar for implementing this solution.  In addition, if
       one gateways between parts of a system using this multiplexing
       and parts that aren't multiplexing, the part that isn't
       multiplexing must also fulfill the requirements on how SSRC is
       assigned or force the gateway to translate SSRCs.  Translating
       SSRC is actually hard as it requires one to understand the
       semantics of all current and future RTP and RTCP extensions.
       Otherwise a barrier for deploying new extensions is created.

   2.  There are some few RTP extensions that currently rely on being
       able to use the same SSRC in different RTP sessions:

       *  XOR FEC (RFC5109)

       *  RTP Retransmission in session mode (RFC4588)

       *  Certain Layered Coding

4.3.  SRTP

   SRTP [RFC3711] is one of the most commonly used security solutions
   for RTP.  In addition, it is the only one defined by IETF that is
   integrated into RTP.  This integration has several aspects that needs
   to be considered when designing a solution for multiplexing RTP
   sessions on the same lower layer transport.

   Determining Crypto Context:  SRTP first of all needs to know which
      session context a received or to-be-sent packet relates to.  It
      also normally relies on the lower layer transport to identify the
      session.  It uses the Master Key Indicatior (MKI), if present, to
      determine which key set is to be used.  Then the SSRC and sequence
      number are used by most crypto suites, including the most common
      use of AES Counter Mode, to actually generate the correct cipher
      stream.







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   Unencrypted Headers:  SRTP has chosen to leave the RTP headers and
      the first two 32-bit words of the first RTCP header unencrypted,
      to allow for both header compression and monitoring to work also
      in the presence of encryption.  As these fields are in clear text
      they are used in most crypto suites for SRTP to determine how to
      protect or recover the plain text.

   It is here important to contrast SRTP against a set of other possible
   protection mechanisms.  DTLS, TLS, and IPsec are all protecting and
   encapsulating the entire RTP and RTCP packets.  They don't perform
   any partial operations on the RTP and RTCP packets.  Any change that
   is considered to be part of the RTP and RTCP packet is transparent to
   them, but possibly not to SRTP.  Thus the impact on SRTP operations
   must be considered when defining a mechanism.

4.4.  Don't Redefine Used Bits

   As the core of RTP is in use in many systems and has a really large
   deployment story and numerous implementations, changing any of the
   field definitions is highly problematic.  First of all, the
   implementations need to change to support this new semantics.
   Secondly, you get a large transition issue when you have some session
   participants that support the new semantics and some that don't.
   Combing the two behaviors in the same session can force the
   deployment of costly and less than perfect translation devices.

4.5.  Firewall Friendly

   It is desirable that current Firewalls will accept the solutions as
   normal RTP packets.  However, in the authors' opinion we can't let
   the firewall stifle invention and evolution of the protocol.  It is
   also necessary to be aware that a change that will make most deep
   inspecting firewall consider the packet as not valid RTP/RTCP will
   have a more difficult deployment story.

4.6.  Monitoring and Reporting

   It is desirable that a third party monitor can still operate on the
   multiplexed RTP Sessions.  It is however likely that they will
   require an update to correctly monitor and report on multiplexed RTP
   Sessions.

   Another type of function to consider is packet sniffers and their
   selector filters.  These may be impacted by a change of the fields.
   An observation is that many such systems are usually quite rapidly
   updated to consider new types of standardized or simply common packet
   formats.




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4.7.  Usable Also Over Multicast

   It is desirable that a solution should be possible to use also when
   RTP and RTCP packets are sent over multicast, both Any Source
   Multicast (ASM) and Single Source Multicast (SSM).  The reason for
   this requirement is to allow a system using RTP to use the same
   configuration regardless of the transport being done over unicast or
   multicast.  In addition, multicast can't be claimed to have an issue
   with using multiple ports, as each multicast group has a complete
   port space scoped by address.

4.8.  Incremental Deployment

   A good solution has the property that in topologies that contains RTP
   mixers or Translators, a single session participant can enable
   multiplexing without having any impact on any other session
   participants.  Thus a node should be able to take a multiplexed
   packet and then easily send it out with minimal or no modification on
   another leg of the session, where each RTP session is transported
   over its own lower-layer transport.  It should also be as easy to do
   the reverse forwarding operation.


5.  Design Considerations

   When defining a SHIM solution for identifying RTP sessions over a
   single transport layer there has been some special considerations
   that is discussed in this section.

5.1.  Location of SHIM

   A major question affecting the SHIM is the location of the SHIM
   header providing the Identifier of the session the packet relate to.
   This section will discuss in detail about the impact of making the
   different choices.

   Identified aspects to consider are:

   Possibility to Process:  A prefixed shim header, i.e. between the
      transport protocol and the RTP/RTCP packet header has the
      advantage that any node on the network that likes to include the
      header in any per-packet processing can reach it.  Reasons for
      per-packet processing are:

      A.  Quality of Service classification

      B.  SHIM ingress or egress




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      C.  Monitoring

      Many routers or similar devices can only read and process the
      first N bytes of the whole packet, where N is commonly on the
      order of 64-128 bytes.  Any other type of processing means putting
      the packet on the slow path.  Thus a prefixed solution enables
      this processing while a post fixed solution will most likely
      forever prevent this type of devices to process it.

   Legacy Processing:  Packets or at least flows of the type IP/UDP/RTP
      can in many cases be identified in Deep Packet Inspection,
      Firewalls or other network entities that concern themselves with
      trying determine what traffic that flows in a particular packet.
      These nodes can clearly be updated but until they have they may
      create a hinder against deployment.  Thus a post fix gives likely
      the least resistance for initial deployment.  However, also for
      postfix location the deployment can be hindered in cases multiple
      RTP sessions using the same SSRC values due to irregular behavior
      of the fields for what the third party believes is one media
      stream rather than multiple ones.  The prefixed will however
      maintain the long-term capabilities of such devices assuming they
      can be updated to include the SHIM header as part of the
      classification.

   Header Compression:  The different header compression techniques that
      has been developed compresses IP/UDP/RTP as complete combination.
      If one instead have a IP/UDP/SHIM/RTP then the compression for the
      full set may not work or poorly.  Instead only IP/UDP header
      compression is likely to be applied.  Thus a prefix will loose
      some compression efficiency until compression profiles for IP/UDP/
      SHIM/RTP has been developed, implemented and deployed.  Postfix
      don't have that issue, but nor can it ever gain anything from
      header compression which an prefixed solution could once an
      updated profile is deployed.  Postfix also will have reduced
      efficiency compressing sessions when the same SSRC is used in two
      different RTP sessions as the RTP header fields like sequence
      number etc will not behave as expected and need frequent explicit
      updates.

   The question of a prefixed or a postfixed header comes down to a
   trade-off between long term usability and deployment issues:

   Prefixed:  Long term good possibility to adapt any network function
      that needs to take the SHIM header into account.  At the same time
      any function that tries to analyze packets and because of that may
      block the packets will be a hinder to deployment.





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   Postfixed:  This solution will likely short term have the best
      possibilities to deploy successfully.  However, long term this
      choice will likely prevent many network nodes that like to be
      capable of separating the RTP sessions being multiplexed together
      from successfully doing that.

   After discussion in the WG it has been determined that prefixed is
   the prefered solution.

5.2.  ICE and DTLS-SRTP Integration

   When using ICE [RFC5245] or DTLS-SRTP [RFC5764] or both with RTP
   there exist the issue that RTP, STUN [RFC5389] and DTLS-SRTP are
   simultanously in use over the same lower layer transport flow, like
   UDP.  This multiplexing is based on the value of the first byte of
   the lower layer transport payload as discussed in Section 5.1.2 of
   DTLS-SRTP [RFC5764].

   The replacement of a single RTP session with the multiple RTP
   sessions idenfied by a SHIM must not be missidentified to be either
   STUN or DTLS-SRTP or any other protocol intending to take the
   available free code-points in the range 193-255 (Decimal).  Thus a
   prefixed SHIM must have its first byte have the two first bits set to
   10 (Binary).  Having the SHIM share the identity of RTP is not an
   issue as one must have mutual agreement that the SHIM is used instead
   of RTP.

      Note: This limits a single byte SHIM to only allow a maximum of 64
      RTP sessions over a single transport flow.

5.3.  Signalling Fallback

   There exist an important aspect in how the SDP signalling functions,
   especially Offer/Answer [RFC3264].  The initial idea for the
   signalling was to build on top of bundle
   [I-D.ietf-mmusic-sdp-bundle-negotiation] which in its default
   function negotiate multiple media types over one RTP session
   [I-D.ietf-avtcore-multi-media-rtp-session].  If the signalling for
   the solution that main purpose is to enable multiple RTP sessions
   results in those cases the peer doesn't support this specification
   the communicating peer can end up in single RTP session if the peer
   supports that.

   We consider it important that in the signalling design that the
   application developer can decide what type of fallback that will
   occur.  It is also important to consider that one have to signal SHIM
   based multiplexing of RTP sessions that are in fact of the type with
   multiple media types.  Thus the signalling for SHIM must be able to



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   describe multiple different scenarios:

   1.  Multiple RTP sessions multiplexed together using SHIM over one
       transport

   2.  Like 1 but where at least one RTP session is containing multiple
       media types

   3.  Like 1, but where the peer doesn't support SHIM and the initiator
       wants to fallback to independent transports

   4.  Like 2, but where the peer doesn't support SHIM and wants to
       fallback to multiple BUNDLED sessions over independent
       transports.

   In addition it must be possible to have multiple different transports
   where each is a SHIM multiplex.  This is to support decomposed end-
   points or cases where certain media traffic is required to go to a
   central processing node while others goes directly to a peer.

   To enable all of these scenarios we propose a solution where each
   indicates SHIM multiplex is indicated as its own grouping attribute
   across all media blocks that are included in some form in the
   multiplex.  This resulting in that these media blocks fall under a
   form of BUNDLE super set.  This super set will also have some of
   bundles restrictions on the transport layer, but not on higher layer.
   Which Session ID pair a particular media block is associated is
   signalled using a SDP attribute (a=session-mux-id) in each media
   block.  When multiple media block are assigned the same session ID
   pair, they form a RTP session with multiple media types and have the
   full restriction of bundle between them.

   The method of fallback is indicated by providing explicit BUNDLE
   grouping in addition to the SHIM when the fallback from SHIM is to
   BUNDLE.

   Note: Signalling solution is awaiting resolution of design path for
   bundle and will then consider that solution and issues raised.


6.  Specification

   This section contains the specification of the RTP session
   multiplexing SHIM, using an explicit session identifier of the
   encapsulated payload.






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6.1.  Shim Layer

   This solution is based on a shim layer that is inserted in the stack
   between the regular RTP and RTCP packets and the transport layer
   being used by the RTP sessions.  Thus the layering looks like the
   following:

   +---------------------+
   |  RTP / RTCP Packet  |
   +---------------------+
   |  Session ID Layer   |
   +---------------------+
   |  Transport layer    |
   +---------------------+

                      Stack View with Session ID SHIM

   The above stack is in fact a layered one as it does allow multiple
   RTP Sessions to be multiplexed on top of the Session ID shim layer.
   This enables the example presented in Figure 1 where four sessions,
   S1-S4 is sent over the same Transport layer and where the Session ID
   layer will combine and encapsulate them with the session ID on
   transmission and separate and decapsulate them on reception.

   +-------------------+
   | S1 | S2 | S3 | S4 |
   +-------------------+
   |  Session ID Layer |
   +-------------------+
   |  Transport layer  |
   +-------------------+

         Figure 1: Multiple RTP Session On Top of Session ID Layer

   The Session ID layer encapsulates one RTP or RTCP packet from a given
   RTP session and prefixes the 2-byte Session ID layer to the packet.
   The Session ID layer is depicted below (Figure 2) and consists of
   first 2 fixed bit values (10b) followed by a 14 bits unsigned integer
   field with the Session ID (SID) value.
    0                   1
    0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |1 0|  Session ID (SID)         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

                        Figure 2: Session ID layer

   Each RTP session being multiplexed on top of a given transport layer



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   is assigned either a single or a pair of unique SID in the range
   0-16383.  The reason for assigning a pair of SIDs to a given RTP
   session are for RTP Sessions that doesn't support "Multiplexing RTP
   Data and Control Packets on a Single Port" [RFC5761] to still be able
   to use a single 5-tuple.  The reasons for supporting this extra
   functionality is that RTP and RTCP multiplexing based on the payload
   type/packet type fields enforces certain restrictions on the RTP
   sessions.  These restrictions may not be acceptable.  As this
   solution does not have these restrictions, performing RTP and RTCP
   multiplexing in this way has benefits.

   Each Session ID value space is scoped by the underlying transport
   protocol.  Common transport protocols like UDP [RFC0768], DCCP
   [RFC4340], TCP [RFC0793], and SCTP [RFC4960] can all be scoped by one
   or more 5-tuple (Transport protocol, source address and port,
   destination address and port).  The case of multiple 5-tuples occur
   in the case of multi-unicast topologies, also called meshed
   multiparty RTP sessions or in case any application would need more
   than 8192 RTP sessions.

      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
                                     +-------------------------------+
                                     |1 0|       Session ID          |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
     |V=2|P|X|  CC   |M|     PT      |       sequence number         | |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
     |                           timestamp                           | |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
     |           synchronization source (SSRC) identifier            | |
     +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
     |            contributing source (CSRC) identifiers             | |
     |                               ....                            | |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
     |                   RTP extension (OPTIONAL)                    | |
   +>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | |                          payload  ...                         | |
   | |                               +-------------------------------+ |
   | |                               | RTP padding   | RTP pad count | |
   +>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
   | ~                     SRTP MKI (OPTIONAL)                       ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | :                 authentication tag (RECOMMENDED)              : |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   +- Encrypted Portion*                      Authenticated Portion ---+

          Figure 3: SRTP Packet encapsulated by Session ID Layer




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      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
                                     +-------------------------------+
                                     |1 0|       Session ID          |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
     |V=2|P|    RC   |   PT=SR or RR   |             length          | |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
     |                         SSRC of sender                        | |
   +>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
   | ~                          sender info                          ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | ~                         report block 1                        ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | ~                         report block 2                        ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | ~                              ...                              ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | |V=2|P|    SC   |  PT=SDES=202  |             length            | |
   | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
   | |                          SSRC/CSRC_1                          | |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | ~                           SDES items                          ~ |
   | +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
   | ~                              ...                              ~ |
   +>+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
   | |E|                         SRTCP index                         | |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
   | ~                     SRTCP MKI (OPTIONAL)                      ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | :                     authentication tag                        : |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   +-- Encrypted Portion                    Authenticated Portion -----+


          Figure 4: SRTCP packet encapsulated by Session ID layer

   The processing in a receiver when the Session ID layer is present
   will be to

   1.  Pick up the packet from the lower layer transport

   2.  Inspect the SID field value

   3.  Strip the SID field from the packet

   4.  Forward it to the (S)RTP Session context identified by the SID
       value




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6.2.  Signalling

   Note: This section may need updating as the direction of the solution
   for Bundle has settled and the impact of the raised issues has been
   analyzed.

   The use of the Session ID layer needs to be explicitly agreed on
   between the communicating parties.  Each RTP Session the application
   uses must in addition to the regular configuration such as payload
   types, RTCP extension etc, have both the underlying 5-tuple (source
   address and port, destination address and port, and transport
   protocol) and the Session ID used for the particular RTP session.
   The signalling requirement is to assign unique Session ID values to
   all RTP Sessions being sent over the same 5-tuple.  The same Session
   ID shall be used for an RTP session independently of the traffic
   direction.  Note that nothing prevents a multi-media application from
   using multiple 5-tuples if desired for some reason, in which case
   each 5-tuple has its own session ID value space.

   This section defines how to negotiate the use of the Session ID
   layer, using the Session Description Protocol (SDP) Offer/Answer
   mechanism [RFC3264].  A new SDP grouping semantics is defined "SHIM"
   and a new media-level SDP attribute, 'session-mux-id.  The attribute
   allows each media description ("m=" line) associated with a 'SHIM'
   group to be identified in which RTP session it belongs.

   The 'session-mux-id' attribute is included for a media description,
   in order to indicate the Session ID for that particular media
   description.  Every media description that shares a common attribute
   value is assumed to be part of a single RTP session.  An SDP Offerer
   MUST include the 'session-mux-id' attribute for every media
   description associated with a 'SHIM' group.  If the SDP Answer does
   not contain the SHIM group, the SDP Offerer MUST NOT use SHIM based
   layering.  However, if that is separate RTP sessions or BUNDLE is
   determined on what was present in the offer and answer.  This will
   depend on what the offering party likes to happen.  If they want a
   failure to negotiate a SHIM, instead may be one or more bundle groups
   then also the BUNDLE grouping is included in the offer.  If the SDP
   Answer still describes a 'BUNDLE' group, the procedures in
   [I-D.ietf-mmusic-sdp-bundle-negotiation] apply.  If not independent
   transports and sessions are used.

   An SDP Answerer MUST NOT include the 'SHIM' group and
   'session-mux-id' attribute in an SDP Answer, unless they where
   included in the SDP Offer.

   The attribute has the following ABNF [RFC5234] definition.




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   Session-mux-id-attr = "a=session-mux-id:" SID *SID-prop
   SID                 = SID-value / SID-pairs
   SID-value           = 1*3DIGIT / "NoN"
   SID-pairs           = SID-value "/" SID-value ; RTP/RTCP SIDs
   SID-prop            = SP assignment-policy / prop-ext
   prop-ext            = token "=" value
   assignment-policy   = "policy=" ("tentative" / "fixed")

   The SHIM group SHALL contain all media descriptions that are intended
   to be sent over the same transport flow, independent of Session ID.
   For all media descriptions part of the same SHIM group the transport
   parameters, i.e. ports, ICE-candidates etc MUST be the same and
   handled as described by BUNDLE.  Note, the parameters related to the
   RTP session does not need to be same.

   For media descriptions that have the same value of the Session ID
   SHALL be treated the same way as if they where part of a BUNDLE
   group, independently if that is indicated or not in the SDP.

   The SID property "policy" is used in negotiation by an end-point to
   indicate if the session ID values are merely a tentative suggestion
   or if they must have these values.  This is used when negotiating SID
   for multi-party RTP sessions to support shared transports such as
   multicast or RTP translators that are unable to produce renumbered
   SIDs on a per end-point basis.  The normal behavior is that the offer
   suggest a tentative set of values, indicated by "policy=tentative".
   These SHOULD be accepted by the peer unless that peer negotiate
   session IDs on behalf of a centralized policy, in which case it MAY
   change the value(s) in the answer.  If the offer represents a policy
   that does not allow changing the session ID values, it can indicate
   that to the answerer by setting the policy to "fixed".  This enables
   the answering peer to either accept the value or indicate that there
   is a conflict in who is performing the assignment by setting the SID
   value to NoN (Not a Number).  Offerer and answerer SHOULD always
   include the policy they are operating under.  Thus, in case of no
   centralized behaviors, both offerer and answerer will indicate the
   tentative policy.

6.3.  SRTP Key Management

   Key management for SRTP do needs discussion as we do cause multiple
   SRTP sessions to exist on the same underlying transport flow.  Thus
   we need to ensure that the key management mechanism still are
   properly associated with the SRTP session context it intends to key.
   To ensure that we do look at the three SRTP key management mechanism
   that IETF has specified, one after another.





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6.3.1.  Security Description

   Session Description Protocol (SDP) Security Descriptions for Media
   Streams [RFC4568] as being based on SDP has no issue with the RTP
   session multiplexing on lower layer specified here.  The reason is
   that the actual keying is done using a media level SDP attribute.
   Thus the attribute is already associated with a particular media
   description.  A media description that also will have an instance of
   the "a=session-mux-id" attribute carrying the SID value/pair used
   with this particular crypto parameters.

6.3.2.  DTLS-SRTP

   Datagram Transport Layer Security (DTLS) Extension to Establish Keys
   for the Secure Real-time Transport Protocol (SRTP) [RFC5764] is a
   keying mechanism that works on the media plane on the same lower
   layer transport that SRTP/SRTCP will be transported over.

   The most direct solution would be to use the SHIM and the SID context
   identifier to be applied also on DTLS packets.  Thus using the same
   SID that is used with RTP and/or RTCP also for the DTLS message
   intended to key that particular SRTP and/or SRTCP flow(s).  This of
   course requires independent usage of DTLS-SRTP for each RTP session.
   In addition it requires changing the layering for DTLS-SRTP as well
   as RTP.  Thus this behavior doesn't gain you anything in regards to
   key-management when using SHIM and have some costs.

   Instead we propose that an DTLS-SRTP key-derivation change is
   introduced.  By including the Session ID value in the derivation of
   the keying material a single DTLS-SRTP key-management operation could
   apply keys and parameters for all the RTP sessions in the same
   transport flow.  Thus the keying cost is significantly reduced,
   especially in regards to network communication and delay impact and
   vunerability to packet loss.

   Details to be written up.

6.3.3.  MIKEY

   MIKEY: Multimedia Internet KEYing [RFC3830] is a key management
   protocol that has several transports.  In some cases it is used
   directly on a transport protocol such as UDP, but there is also a
   specification for how MIKEY is used with SDP "Key Management
   Extensions for Session Description Protocol (SDP) and Real Time
   Streaming Protocol (RTSP)" [RFC4567].

   Lets start with the later, i.e. the SDP transport, which shares the
   properties with Security Description in that is can be associated



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   with a particular media description in a SDP.  As long as one avoids
   using the session level attribute one can be certain to correctly
   associate the key exchange with a given SRTP/SRTCP context.

   It does appear that MIKEY directly over a lower layer transport
   protocol will have similar issues as DTLS.

6.4.  Examples

6.4.1.  RTP Packet with Transport Header

   The below figure contains an RTP packet with SID field encapsulated
   by a UDP packet (added UDP header).

      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     | Source Port                   | Destination Port              |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     | Length                        | Checksum                      |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |1 0|     Session ID            |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
     |V=2|P|X|  CC   |M|     PT      |       sequence number         | |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
     |                           timestamp                           | |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
     |           synchronization source (SSRC) identifier            | |
     +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+ |
     |            contributing source (CSRC) identifiers             | |
     |                               ....                            | |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
     |                   RTP extension (OPTIONAL)                    | |
   +>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | |                          payload  ...                         | |
   | |                               +-------------------------------+ |
   | |                               | RTP padding   | RTP pad count | |
   +>+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+<+
   | ~                     SRTP MKI (OPTIONAL)                       ~ |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   | :                 authentication tag (RECOMMENDED)              : |
   | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
   +- Encrypted Portion*                      Authenticated Portion ---+

               SRTP Packet Encapsulated by Session ID Layer






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6.4.2.  SDP Offer/Answer example

6.4.2.1.  Basic Example

   This section contains SDP offer/answer examples.  First one example
   of successful SHIMing, and then two where fallback occurs.  The
   fallback option here is to fallback to individual transports, thus no
   BUNDLE group.

   In the below SDP offer, one audio and one video is being offered.
   The audio is using SID 0, and the video is using SID 1 to indicate
   that they are different RTP sessions despite being offered over the
   same 5-tuple.
   v=0
   o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
   s=
   c=IN IP4 atlanta.example.com
   t=0 0
   a=group:SHIM foo bar
   m=audio 10000 RTP/AVP 0 8 97
   b=AS:200
   a=mid:foo
   a=session-mux-id:0 policy=tentative
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 iLBC/8000
   m=video 10000 RTP/AVP 31 32
   b=AS:1000
   a=mid:bar
   a=session-mux-id:1 policy=tentative
   a=rtpmap:31 H261/90000
   a=rtpmap:32 MPV/90000

   The SDP answer from an end-point that supports this BUNDLEing:

















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   v=0
   o=bob 2808844564 2808844564 IN IP4 biloxi.example.com
   s=
   c=IN IP4 biloxi.example.com
   t=0 0
   a=group:SHIM foo bar
   m=audio 20000 RTP/AVP 0
   b=AS:200
   a=mid:foo
   a=session-mux-id:0 policy=tentative
   a=rtpmap:0 PCMU/8000
   m=video 20000 RTP/AVP 32
   b=AS:1000
   a=mid:bar
   a=session-mux-id:1 policy=tentative
   a=rtpmap:32 MPV/90000

   The SDP answer from an end-point that does not support this SHIMing.
   v=0
   o=bob 2808844564 2808844564 IN IP4 biloxi.example.com
   s=
   c=IN IP4 biloxi.example.com
   t=0 0
   m=audio 20000 RTP/AVP 0
   b=AS:200
   a=rtpmap:0 PCMU/8000
   m=video 30000 RTP/AVP 32
   b=AS:1000
   a=rtpmap:32 MPV/90000

6.4.2.2.  Advanced Example

   In this example we have two BUNDLED sessions, one with audio and
   video and one with XOR based FEC [RFC5109] for the audio and the
   video.  These two RTP session are then SHIMed into a single transport
   flow.















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   v=0
   o=alice 2890844526 2890844526 IN IP4 atlanta.example.com
   s=
   c=IN IP4 atlanta.example.com
   t=0 0
   a=group:SHIM foo bar 1 2
   a=group:BUNDLE 1 2
   a=group:BUNDLE foo bar
   a=group:FEC foo 1
   a=group:FEC bar 2
   m=audio 10000 RTP/AVP 0 8 97
   b=AS:200
   a=mid:foo
   a=session-mux-id:0 policy=tentative
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 iLBC/8000
   m=video 10000 RTP/AVP 31 32
   b=AS:1000
   a=mid:bar
   a=session-mux-id:0 policy=tentative
   a=rtpmap:31 H261/90000
   a=rtpmap:32 MPV/90000
   m=audio 10000 RTP/AVP 100
   b=AS:100
   a=rtpmap:100 ulpfec/8000
   a=mid:1
   a=session-mux-id:1 policy=tentative
   m=video 10000 RTP/AVP 101
   b=AS:500
   a=mid:2
   a=session-mux-id:1 policy=tentative
   a=rtpmap:101 ulpfec/90000

   The SDP answer of a client supporting
   [I-D.ietf-mmusic-sdp-bundle-negotiation] but not this SHIMing would
   look like this:














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   v=0
   o=bob 2808844564 2808844564 IN IP4 biloxi.example.com
   s=
   c=IN IP4 biloxi.example.com
   t=0 0
   a=group:BUNDLE 1 2
   a=group:BUNDLE foo bar
   a=group:FEC foo 1
   a=group:FEC bar 2
   m=audio 20000 RTP/AVP 0 8 97
   b=AS:200
   a=mid:foo
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 iLBC/8000
   m=video 20000 RTP/AVP 31 32
   b=AS:1000
   a=mid:bar
   a=rtpmap:31 H261/90000
   a=rtpmap:32 MPV/90000
   m=audio 20002 RTP/AVP 100
   b=AS:100
   a=rtpmap:100 ulpfec/8000
   a=mid:1
   m=video 20002 RTP/AVP 101
   b=AS:500
   a=mid:2
   a=rtpmap:101 ulpfec/90000

   In the above case two different RTP sessions, both being of a BUNDLE
   type with multiple media types in each.  The two established flows
   will be Alice:10000<->Bob:20000, and Alice:10000<->Bob:20002.

   If the peer did support neither of the SHIM or BUNDLE extension the
   answer would look like this:
















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   v=0
   o=bob 2808844564 2808844564 IN IP4 biloxi.example.com
   s=
   c=IN IP4 biloxi.example.com
   t=0 0
   a=group:FEC foo 1
   a=group:FEC bar 2
   m=audio 20000 RTP/AVP 0 8 97
   b=AS:200
   a=mid:foo
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:97 iLBC/8000
   m=video 20002 RTP/AVP 31 32
   b=AS:1000
   a=mid:bar
   a=rtpmap:31 H261/90000
   a=rtpmap:32 MPV/90000
   m=audio 20004 RTP/AVP 100
   b=AS:100
   a=rtpmap:100 ulpfec/8000
   a=mid:1
   m=video 20006 RTP/AVP 101
   b=AS:500
   a=mid:2
   a=rtpmap:101 ulpfec/90000

   In this case four different transport flows would be established for
   RTP, each with a different RTP session over them.  The answer also
   knows the binding between the sessions with FEC and their source data
   thanks to the FEC specification.


7.  Open Issues

   This work is still in the early phase of specification.  This section
   contains a list of open issues where the author desires some input.

   1.  In Section 6.2 there is a discussion of which parameters that
       must be configured.  The scope of these rules and if they do make
       sense needs additional discussion.

   2.  Can we provide better control so that applications that doesn't
       desire fallback to single RTP session when Multiplexing shim
       fails to be supported but Bundle is supported ends up with a
       better alternative?





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   3.  The details for how to do key-derivation, preferably in such a
       way that it can be reused by multiple key-management solutions
       like MIKEY and DTLS-SRTP

   4.  The signalling solution will be revisited when the BUNDLE
       solution discussion has yeild some result.


8.  IANA Considerations

   This document request the registration of one SDP attribute.  Details
   of the registration to be filled in.


9.  Security Considerations

   The security properties of the Session ID layer is depending on what
   mechanism is used to protect the RTP and RTCP packets of a given RTP
   session.  If IPsec or transport layer security solutions such as DTLS
   or TLS are being used then both the encapsulated RTP/RTCP packets and
   the session ID layer will be protected by that security mechanism.
   Thus potentially providing both confidentiality, integrity and source
   authentication.  If SRTP is used, the session ID layer will not be
   directly protected by SRTP.  However, it will be implicitly integrity
   protected (assuming the RTP/RTCP packet is integrity protected) as
   the only function of the field is to identify the session context.
   Thus any modification of the SID field will attempt to retrieve the
   wrong SRTP crypto context.  If that retrieval fails, the packet will
   be anyway be discarded.  If it is successful, the context will not
   lead to successful verification of the packet.


10.  Acknowledgements

   This document is based on the input from various people, especially
   in the context of the RTCWEB discussion of how to use only a single
   lower layer transport.  The RTP and RTCP packet figures are borrowed
   from RFC3711.  The SDP example is extended from the one present in
   [I-D.ietf-mmusic-sdp-bundle-negotiation].  Eric Rescorla contributed
   the basic idea of optimizing the DTLS-SRTP key-management by
   modifying the key derivation process.

   The proposal in Appendix A.5 is original suggested by Colin Perkins.
   The idea in Appendix A.6 is from an Internet Draft
   [I-D.rosenberg-rtcweb-rtpmux] written by Jonathan Rosenberg et. al.
   The proposal in Appendix A.3 is a result of discussion by a group of
   people at IETF meeting #81 in Quebec.




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11.  References

11.1.  Normative References

   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C. and H. Alvestrand, "Multiplexing Negotiation
              Using Session Description Protocol (SDP) Port Numbers",
              draft-ietf-mmusic-sdp-bundle-negotiation-01 (work in
              progress), August 2012.

   [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.

   [RFC3711]  Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
              Norrman, "The Secure Real-time Transport Protocol (SRTP)",
              RFC 3711, March 2004.

   [RFC5234]  Crocker, D. and P. Overell, "Augmented BNF for Syntax
              Specifications: ABNF", STD 68, RFC 5234, January 2008.

11.2.  Informational References

   [I-D.ietf-avtcore-multi-media-rtp-session]
              Westerlund, M., Perkins, C., and J. Lennox, "Multiple
              Media Types in an RTP Session",
              draft-ietf-avtcore-multi-media-rtp-session-00 (work in
              progress), October 2012.

   [I-D.lennox-rtcweb-rtp-media-type-mux]
              Rosenberg, J. and J. Lennox, "Multiplexing Multiple Media
              Types In a Single Real-Time Transport Protocol (RTP)
              Session", draft-lennox-rtcweb-rtp-media-type-mux-00 (work
              in progress), October 2011.

   [I-D.rosenberg-rtcweb-rtpmux]
              Rosenberg, J., Jennings, C., Peterson, J., Kaufman, M.,
              Rescorla, E., and T. Terriberry, "Multiplexing of Real-
              Time Transport Protocol (RTP) Traffic for Browser based
              Real-Time Communications (RTC)",
              draft-rosenberg-rtcweb-rtpmux-00 (work in progress),
              July 2011.

   [I-D.westerlund-avtcore-multiplex-architecture]
              Westerlund, M., Burman, B., Perkins, C., and H.



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              Alvestrand, "Guidelines for using the Multiplexing
              Features of RTP",
              draft-westerlund-avtcore-multiplex-architecture-02 (work
              in progress), July 2012.

   [RFC0768]  Postel, J., "User Datagram Protocol", STD 6, RFC 768,
              August 1980.

   [RFC0793]  Postel, J., "Transmission Control Protocol", STD 7,
              RFC 793, September 1981.

   [RFC3264]  Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model
              with Session Description Protocol (SDP)", RFC 3264,
              June 2002.

   [RFC3830]  Arkko, J., Carrara, E., Lindholm, F., Naslund, M., and K.
              Norrman, "MIKEY: Multimedia Internet KEYing", RFC 3830,
              August 2004.

   [RFC4340]  Kohler, E., Handley, M., and S. Floyd, "Datagram
              Congestion Control Protocol (DCCP)", RFC 4340, March 2006.

   [RFC4567]  Arkko, J., Lindholm, F., Naslund, M., Norrman, K., and E.
              Carrara, "Key Management Extensions for Session
              Description Protocol (SDP) and Real Time Streaming
              Protocol (RTSP)", RFC 4567, July 2006.

   [RFC4568]  Andreasen, F., Baugher, M., and D. Wing, "Session
              Description Protocol (SDP) Security Descriptions for Media
              Streams", RFC 4568, July 2006.

   [RFC4960]  Stewart, R., "Stream Control Transmission Protocol",
              RFC 4960, September 2007.

   [RFC5109]  Li, A., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, December 2007.

   [RFC5245]  Rosenberg, J., "Interactive Connectivity Establishment
              (ICE): A Protocol for Network Address Translator (NAT)
              Traversal for Offer/Answer Protocols", RFC 5245,
              April 2010.

   [RFC5285]  Singer, D. and H. Desineni, "A General Mechanism for RTP
              Header Extensions", RFC 5285, July 2008.

   [RFC5389]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing,
              "Session Traversal Utilities for NAT (STUN)", RFC 5389,
              October 2008.



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   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, April 2009.

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, April 2010.

   [RFC5764]  McGrew, D. and E. Rescorla, "Datagram Transport Layer
              Security (DTLS) Extension to Establish Keys for the Secure
              Real-time Transport Protocol (SRTP)", RFC 5764, May 2010.


Appendix A.  Possible Solutions

   This section documents the solutions explored when selecting a SHIM
   based one and discusses their feasibility.

A.1.  Header Extension

   One proposal is to define an RTP header extension [RFC5285] that
   explicitly enumerates the session identifier in each packet.  This
   proposal has some merits regarding RTP, since it uses an existing
   extension mechanism; it explicitly enumerates the session allowing
   for third parties to associate the packet to a given RTP session; and
   it works with SRTP as currently defined since a header extension is
   by default not encrypted, and is thus readable by the receiving stack
   without needing to guess which session it belongs to and attempt to
   decrypt it.  This approach does, however, conflict with the
   requirement from [RFC5285] that "header extensions using this
   specification MUST only be used for data that can be safely ignored
   by the recipient", since correct processing of the received packet
   depends on using the header extension to demultiplex it to the
   correct RTP session.

   Using a header extension also result in the session ID is in the
   integrity protected part of the packet.  Thus a translator between
   multiplexed and non-multiplexed has the options:

   1.  to be part of the security context to verify the field

   2.  to be part of the security context to verify the field and remove
       it before forwarding the packet

   3.  to be outside of the security context and leave the header
       extension in the packet.  However, that requires successful
       negotiation of the header extension, but not of the
       functionality, with the receiving end-points.




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   The biggest existing hurdle for this solution is that there exist no
   header extension field in the RTCP packets.  This requires defining a
   solution for RTCP that allows carrying the explicit indicator,
   preferably in a position that isn't encrypted by SRTCP.  However, the
   current SRTCP definition does not offer such a position in the
   packet.

   Modifying the RR or SR packets is possible using profile specific
   extensions.  However, that has issues when it comes to deployability
   and in addition any information placed there would end up in the
   encrypted part.

   Another alternative could be to define another RTCP packet type that
   only contains the common header, using the 5 bits in the first byte
   of the common header to carry a session id.  That would allow SRTCP
   to work correctly as long it accepts this new packet type being the
   first in the packet.  Allowing a non-SR/RR packet as the first packet
   in a compound RTCP packet is also needed if an implementation is to
   support Reduced Size RTCP packets [RFC5506].  The remaining downside
   with this is that all stack implementations supporting multiplexing
   would need to modify its RTCP compound packet rules to include this
   packet type first.  Thus a translator box between supporting nodes
   and non-supporting nodes needs to be in the crypto context.

   This solution's per packet overhead is expected to be 64-bits for
   RTCP.  For RTP it is 64-bits if no header extension was otherwise
   used, and an additional 16 bits (short header), or 24 bits plus (if
   needed) padding to next 32-bits boundary if other header extensions
   are used.

A.2.  Multiplexing Shim

   This proposal is to prefix or postfix all RTP and RTCP packets with a
   session ID field.  This field would be outside of the normal RTP and
   RTCP packets, thus having no impact on the RTP and RTCP packets and
   their processing.  An additional step of demultiplexing processing
   would be added prior to RTP stack processing to determine in which
   RTP session context the packet shall be included.  This has also no
   impact on SRTP/SRTCP as the shim layer would be outside of its
   protection context.  The shim layer's session ID is however
   implicitly integrity protected as any error in the field will result
   in the packet being placed in the wrong or non-existing context, thus
   resulting in a integrity failure if processed by SRTP/SRTCP.

   This proposal is quite simple to implement in any gateway or
   translating device that goes from a multiplexed to a non-multiplexed
   domain or vice versa, as only an additional field needs to be added
   to or removed from the packet.



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   The main downside of this proposal is that it is very likely to
   trigger a firewall response from any deep packet inspection device.
   If the field is prefixed, the RTP fields are not matching the
   heuristics field (unless the shim is designed to look like an RTP
   header, in which case the payload length is unlikely to match the
   expected value) and thus are likely preventing classification of the
   packet as an RTP packet.  If it is postfixed, it is likely classified
   as an RTP packet but may not correctly validate if the content
   validation is such that the payload length is expected to match
   certain values.  It is expected that a postfixed shim will be less
   problematic than a prefixed shim in this regard, but we are lacking
   hard data on this.

   This solution's per packet overhead is 1 byte.

A.3.  Single Session

   Given the difficulty of multiplexing several RTP sessions onto a
   single lower-layer transport, it's tempting to send multiple media
   streams in a single RTP session.  Doing this avoids the need to de-
   multiplex several sessions on a single transport, but at the cost of
   losing the RTP session as a separator for different type of streams.
   Lacking different RTP sessions to demultiplex incoming packets, a
   receiver will have to dig deeper into the packet before determining
   what to do with it.  Care must be taken in that inspection.  For
   example, you must be careful to ensure that each real media source
   uses its own SSRC in the session and that this SSRC doesn't change
   media type.

   The loss of the RTP session as a separator for different usages or
   purpose would be an minor issue if the only difference between the
   RTP sessions is the media type.  In this case, the application could
   use the Payload Type field to identify the media type.  The loss of
   the RTP Session functionality is however severe, if the application
   uses the RTP Session for separating different treatments, contexts
   etc.  Then you would need additional signalling to bind the different
   sources to groups which can help make the necessary distinctions.

   However, the loss of the RTP session as separator is not the only
   issue with this approach.  The RTP Multiplexing Architecture
   [I-D.westerlund-avtcore-multiplex-architecture] discusses a number of
   issues in Section 6.7.  These include RTCP bandwidth differences,
   limitations in the number of payload types, media aware RTP mixers
   and interactions with Legacy end-points.

   Additional attention should be place on this important aspect.  In
   multi-party situations using central nodes there exist some
   difficulties in having a legacy implementation using multiple RTP



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   sessions interworking with an end-point having only a single RTP
   session across the central node.  The main reason is the fact that
   the one using single session with multiple media types has only one
   SSRC space, while the other end-points have multiple spaces.  Thus
   translation may have to occur because there is several RTP sessions
   using the same SSRC value.  This has both limitations, processing
   overhead and the possibility of becoming an deployment obstacle for
   new RTP/RTCP extensions.

   This approach has been proposed in the RTCWeb context in
   [I-D.lennox-rtcweb-rtp-media-type-mux] and
   [I-D.ietf-mmusic-sdp-bundle-negotiation].  These drafts describe how
   to signal multiple media streams multiplexed into a single RTP
   session, and address some of the issues raised here and in Section
   6.7 of the RTP Multiplexing Architecture
   [I-D.westerlund-avtcore-multiplex-architecture] draft.

   This method has several limitations that limits its usage as solution
   in providing multiple RTP sessions on the same lower layer transport.
   However, we acknowledge that there are some uses for which this
   method may be sufficient and which can accept the methods limitations
   and downsides.  The RTCWEB WG has a working assumption to support
   this method.  For more details of this method, see the relevant
   drafts under development.  We do include this method in the
   comparison to provide a more complete picture of the pro and cons of
   this method.

   This solution has no per packet overhead.  The signalling overhead
   will be a different question.

A.4.  Use the SRTP MKI field

   This proposal is to overload the MKI SRTP/SRTCP identifier to not
   only identify a particular crypto context, but also identify the
   actual RTP Session.  This clearly is a miss use of the MKI field,
   however it appears to be with little negative implications.  SRTP
   already supports handling of multiple crypto contexts.

   The two major downsides with this proposal is first the fact that it
   requires using SRTP/SRTCP to multiplex multiple sessions on a single
   lower layer transport.  The second issue is that the session ID
   parameter needs to be put into the various key-management schemes and
   to make them understand that the reason to establish multiple crypto
   contexts is because they are connected to various RTP Sessions.
   Considering that SRTP have at least 3 used keying mechanisms, DTLS-
   SRTP [RFC5764], Security Descriptions [RFC4568], and MIKEY [RFC3830],
   this is not an insignificant amount of work.




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   This solution has 32-bit per packet overhead, but only if the MKI was
   not already used.

A.5.  Use an Octet in the Padding

   The basics of this proposal is to have the RTP packet and the last
   (required by RFC3550) RTCP packet in a compound to include padding,
   at least 2 bytes.  One byte for the padding count (last byte) and one
   byte just before the padding count containing the session ID.

   This proposal uses bytes to carry the session ID that have no defined
   value and is intended to be ignored by the receiver.  From that
   perspective it only causes packet expansion that is supported and
   handled by all existing equipment.  If an implementation fails to
   understand that it is required to interpret this padding byte to
   learn the session ID, it will see a mostly coherent RTP session
   except where SSRCs overlap or where the payload types overlap.
   However, reporting on the individual sources or forwarding the RTCP
   RR are not completely without merit.

   There is one downside of this proposal and that has to do with SRTP.
   To be able to determine the crypto context, it is necessary to access
   to the encrypted payload of the packet.  Thus, the only mechanism
   available for a receiver to solve this issue is to try the existing
   crypto contexts for any session on the same lower layer transport and
   then use the one where the packet decrypts and verifies correctly.
   Thus for transport flows with many crypto contexts, an attacker could
   simply generate packets that don't validate to force the receiver to
   try all crypto contexts they have rather than immediately discard it
   as not matching a context.  A receiver can mitigate this somewhat by
   using heuristics based on the RTP header fields to determine which
   context applies for a received packet, but this is not a complete
   solution.

   This solution has a 16-bit per packet overhead.

A.6.  Redefine the SSRC field

   The Rosenberg et. al.  Internet draft "Multiplexing of Real-Time
   Transport Protocol (RTP) Traffic for Browser based Real-Time
   Communications (RTC)" [I-D.rosenberg-rtcweb-rtpmux] proposed to
   redefine the SSRC field.  This has the advantage of no packet
   expansion.  It also looks like regular RTP.  However, it has a number
   of implications.  First of all it prevents any RTP functionality that
   require the same SSRC in multiple RTP sessions.

   Secondly its interoperability with end-point using multiple RTP
   sessions are problematic.  Such interoperability will requires an



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   SSRC translator function in the gatewaying node to ensure that the
   SSRCs fulfill the semantic rules of the different domains.  That
   translator is actually far from easy as it needs to understand the
   semantics of all RTP and RTCP extensions that include SSRC/CSRC.
   This as it is necessary to know when a particular matching 32-bit
   pattern is an SSRC field and when the field is just a combination of
   other fields that create the same matching 32-bit pattern.  Thus
   there is a possibility that such a translator becomes a obstacle in
   deploying future RTP/RTCP extensions.  In addition the translator
   actually have significant overhead when SRTP are in use.  This as a
   verification that the packet is authentic, decryption, SSRC
   translation, encryption and finally generation of authentication tags
   are required.  In addition the translator must be part of the
   security context.

   This solution has no per packet overhead.


Appendix B.  Comparison

   This section compares the above potential solutions with the
   requirements.  Motivations are provided in addition to a high level
   metric of successfully, partially and failing to meet requirement.
   In the end a summary table (Figure 5) of the high level value are
   provided.

B.1.  Support of Multiple RTP Sessions Over Single Transport

   This one is easy to determine.  Only the single session proposal
   fails this requirement as it is not at all designed to meet it.  The
   rest fully support this requirement.  The main question around this
   requirement is how important it is to have as discussed in
   Section 4.1.

B.2.  Enable Same SSRC Value in Multiple RTP Sessions

   Based on the discussion in Section 4.2 two sub-requirements have been
   derived.

B.2.1.  Avoid SSRC Translation in Gateways/Translation

   This sub-requirement is derived based on the desire to avoid having
   gateways or translators perform full SSRC translation to minimize
   complexity, avoid the requirement to have gateways in security
   context, and as a hinder to long-term evolution.  Two of the
   proposals have issues with this, due to their lack of support for
   multiple 32-bit SSRC spaces and lacking possibility to have the same
   SSRC value in multiple RTP sessions.  The proposals that have these



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   properties and thus are marked as failing are the Single Session and
   Redefine the SSRC field.  The other proposals are all successful in
   meeting this requirement.

B.2.2.  Support Existing Extensions

   The second sub-requirement is how well the proposals support using
   the existing RTP mechanisms.  Here both Single Session and Redefine
   the SSRC field will have clear issues as they cannot support the same
   full 32-bit SSRC value in two different RTP sessions.  This is
   clearly an issue for the XOR based FEC.  RTP retransmission and
   scalable encoding are minor issues as there exist alternatives to
   those mechanisms that works with the structure of these two
   proposals.  Thus we give them a fail.  The Header Extension gets a
   partial due to unclear interaction between putting in an header
   extension and these mechanisms.

B.3.  Ensure SRTP Functions

   This requirement is about ensuring both secure and efficient usage of
   SRTP.  The Octet in Padding field proposal gets a fail as the
   receiving end-point cannot determine the intended RTP session prior
   to de-encryption of the padding field.  Thus a catch-22 arises which
   can only be resolved by trying all session contexts and see what
   decrypts.  This causes a security vulnerability as an attacker can
   inject a packet which does not meet any of the session contexts.  The
   receiver will then attempt decryption and authentication of it using
   all its session contexts, increasing the amount of wasted resources
   by a factor equal to the number of multiplexed sessions.  Thus this
   proposal gets a fail.

   The proposal of Overloading the SRTP MKI field as session identifier
   gets a partial due to the fact that it cannot use SRTP's key-
   management mechanism out of the box.  It forces the key-management
   mechanism and the SRTP implementations to maintain the MKI-to-RTP
   session bindings to maintain secure and correct function.

   The Redefine the SSRC field gets a partial due to its need to modify
   the key-management mechanisms to correctly identify the partial SSRC
   space the parameters applies to.  Similarly, the SRTP implementation
   also needs to be updated to correctly support this security context
   differentiation.

   The header extension based solution gets a less severe partial than
   Redefine the SSRC and the MKI.  It will however have an issue when
   being gatewayed to a domain that does not multiplex multiple RTP
   sessions over the same transport.  Then the gateway will require to
   be in the security context to be able to add or remove the header



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   extension as it is in the part of the packet that is integrity
   protected by SRTP.

   The remaining two proposals do not affect SRTP mechanisms and thus
   successfully meet this requirement.

B.4.  Don't Redefine Used Bits

   This requirement is all about RTP and RTCP header fields having a
   given definition should not be changed as it can cause
   interoperability problems between modified and non-modified
   implementations.  This becomes especially problematic in RTP sessions
   used for multi-party sessions.

   Redefine the SSRC field gets a big fail on this as it redefines the
   SSRC field, a core field in RTP.  It has been identified that such a
   change will have issues since if it gets connected to a non-modified
   end-point that randomly assigns the SSRC, as supposed by RFC 3550,
   those SSRCs will be distributed over different RTP sessions at the
   modified end-point.  Also other functions using the SSRC field, not
   understanding the additional semantics of the SSRC field, is likely
   to have issues.

   Using the SRTP MKI field to identify a session is overloading that
   field with double semantics.  This likely has minimal negative impact
   in RTP since it should be possible to have the SRTP stack use the MKI
   field to both look up the security context and which output RTP
   session the processed packet belongs to.  However, this redefinition
   clearly creates issues with the key-management scheme.  That will
   have to be modified to handle both this change and deal with the
   interoperability issues when negotiating its usage.  This gets a full
   fail due to that it makes the problem someone else's, namely the RTP
   implementors.

   Defining an Octet in the Padding field redefines a field, whose
   definition is to have zero value and is expected to be ignored by the
   receiver according to the original semantics.  Thus this is one of
   the more benign modifications one can do, however this can still
   cause issues in implementations that unnecessarily check the field
   values, or in Firewalls.  This is judged to be partially meeting the
   requirement.

   The Header Extension proposal does in fact not redefine any currently
   used bits in RTP.  The header extension would be a correctly
   identified extension with its own definition.  However, it does
   redefine a rule on what header extensions are for.  The RTCP solution
   however would have more severe impact as it would need to redefine
   the standard meaning of an RTCP packet header in addition to the



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   default compound packet rules.  Due to these issues the proposal
   fails to meet this requirement.

   The multiplexing shim and the single session both successfully meet
   this requirement.

B.5.  Firewall Friendly

   This requirement is clearly difficult to judge as firewall
   implementations are highly different in both implementation, scope of
   what it investigates in packets, and set policies.  A reasonable goal
   is to minimize the likeliness that rules and policies intended to let
   RTP media streams pass, will also let these streams through when
   multiplexing RTP sessions over a single transport.  The below
   analysis shows that no solution is truly firewall friendly and all
   are judged as being partially meeting this goal.  However, the reason
   why it is believed that a firewall might react to the streams are
   quite different.

   The Single Session and Redefine the SSRC field are likely the least
   suspect solutions from a firewall perspective.  However, as their
   transport flows contain multiple SSRCs with payloads that indicate
   likely multiple different media types they are still likely to make a
   picky firewall block the transport.  This is especially true for
   Firewalls that take signalling messages into account where it will
   expect a particular media type in a given context.  A non upgraded
   firewall might in fact produce two different contexts with
   overlapping transport parameters where both rules will receive media
   streams of the other media type that are outside of the allowed rule.
   However, to be clear if these proposals doesn't get through, none of
   the other will either as they all will have this behavior.

   The header extension proposal is potentially problematic for two
   reasons.  The first reason, which also other proposals has, is
   related to that the same SSRC value can exist in two RTP sessions
   over the same underlying flow.  Anyone tracking the sequence number
   and timestamp will react badly as the second media stream with the
   same SSRC causes constant jumps back and forth in these fields
   compared to the first stream, if packets are transmitted
   simultaneously for both SSRCs.  This issue can likely only be solved
   by having the Firewalls that like to track flows to also use the
   session identifier to create context.  This is possible as the header
   extension will be in the clear and in the front.  The second issue is
   that the header extension itself may get the firewall to react.
   Especially very picky ones that expect packets with certain media
   types to have certain packet lengths.  They are not compatible with a
   header extension.




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   The Multiplexing Shim shares the issue with multiple flows for the
   same SSRC.  Firewalls and deep packet inspection cause the shim
   placement to be in question.  If it is a pre-fixed shim, it prevents
   the packet from looking like regular IP/UDP/RTP packets and be
   correctly classified in Firewalls and DPI engines.  However, if one
   puts it last, it is unlikely that any firewall or DPI ever will be
   able to take the session context into account as it is at the end of
   the packet.  This as many line rate processing devices only take a
   certain amount of the headers into account.

   The SRTP MKI field is likely the solution that has least firewall and
   DPI issues, after the single RTP session.  There is no additional
   suspect field.  The only difference from a single RTP session in the
   transport flow is the fact that multiple MKI are guaranteed to be
   used.  However, that may occur also in a single RTP session usage.
   Thus the only issues are the one shared with single session and the
   one that several RTP media streams may use the same SSRC.

   The octet in the padding field has, in addition to the issues the
   SRTP MKI field has, the single issue that it redefines something that
   is supposed to be zero into a value.  Thus potentially causing a
   deeply inspecting firewall to clamp the flow in fear of covert
   channel or non-compliance.

B.6.  Monitoring and Reporting

   The monitoring and reporting requirement considers several aspects.
   How useful monitoring can one get from an existing legacy monitor,
   and secondary any issues in upgrading them to handle the selected
   solution.  Thirdly, packet selector filters and packet sniffers
   concerns are considered.

   In general one can expect the proposals that have only a single SSRC
   space to work better with legacy.  Thus both Single Session and
   Redefine SSRC space can gather and report data on media flows most
   likely.  The only potential issue is that due to the different media
   types and clock rates, some failure may occur.  In particular a third
   party monitor may be targeted to a specific media type, like
   monitoring VoIP.  That monitor will have problems processing any
   video packets correctly and generate the VoIP specific metrics for
   any video sending SSRC.  In general, no legacy solution for
   monitoring will be able to correctly create the sub-contexts that
   each RTP session has in the solutions, without update to handle the
   new semantics.  Also when it comes to the packet filtering and
   selector filters, fine grained control can only be accomplished
   implementing the new semantics.  Therefore only the Single Session
   meets this requirement fully.




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   Redefine the SSRC field is close to fully meeting the requirement,
   however due to that there exist a session structure that is hidden to
   anyone that is not upgraded to understand the semantics, this only
   gets a partial.

   The other proposals all can have multiple RTP sessions using the same
   SSRC.  This will create significant issues for any legacy third party
   monitor.  Only an updated monitor, or for that matter packet
   selector, can pick out the individual media streams and their
   associated RTCP traffic.  Thus all these proposals gets a failure to
   meet the requirement.

B.7.  Usable over Multicast

   As discussed earlier the goal with having the option usable also over
   multicast is to remove the need to produce different media streams
   for transport over unicast and multicast.  All of the proposals
   successfully meet the requirement.

B.8.  Incremental Deployment

   The possibility to deploy the usage of the multiplexing of multiple
   RTP sessions over a single transport, especially in the context of
   multi-party sessions, is a great benefit for any of the proposals.
   Thus not all end-point implementations needs to be upgraded before
   one start enabling it in the central node and any signalling.

   Considering a centralized multi-party application where some
   participants are using multiple transport flows and you want to
   enable one particular participant to use the single transport to the
   central node, one criteria stands out.  The possibility to have one
   RTP session per transport in one leg, and in the next multiplex them
   together with minimal complexity and packet changes.  Here there are
   significant differences.

   The Multiplexing Shim has the least overhead for this.  As the
   central node or gateway between deployments only needs to either add
   or remove the shim identifier and then forward the packet over the
   corresponding transport, either a joint one on the single transport
   side, or over the individual one on the multiple transport side.

   The SRTP MKI field proposal is almost as good, as the only main
   difference is the need to coordinate the used MKIs on the non-
   multiplexed legs so that there is no overlap between the RTP
   sessions.  And if there is, the MKI can be translated in gateway as
   SRTP has no integrity protection over the MKI.  Thus both
   multiplexing shim and SRTP MKI field does successfully meet this
   requirement.



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   The Header Extension supports multiple full 32-bit SSRC spaces and
   can thus handle all the RTP sessions without need for any SSRC
   translation, however this proposal does run into the problem that the
   gateway needs to be in the security context to be able to add or
   remove the header extension when SRTP is used.  In addition to the
   security implications of that, there is a complexity overhead due to
   the need to redo the authentication tags on all RTP/RTCP packets.
   Thus it gets a partial.

   The Octet in the Padding field share issues with the header extension
   but have even higher complexities for this.  The reason is that the
   padding field is also encrypted.  Thus to add or remove it (although
   removing it may be unnecessary) forces the end-point to encrypt at
   least that byte also, and for ciphers that are not stream-ciphers,
   the whole packet needs to be re-encrypted.  Thus this proposal gets a
   very weak partially meeting the requirement.

   The Single Session and Redefine the SSRC field do not allow several
   vanilla RTP sessions to be connected to these proposals.  The reason
   is the single 32-bit SSRC space they have.  Single Session only has
   one session and the Redefine the SSRC fields uses some of the bits as
   session identifier.  This forces the gateway to translate the SSRC
   whenever it does not fulfill the rules or semantics of the
   multiplexed side.  For Redefine SSRC field this becomes almost
   constant as the session identifier part of the SSRC must be the same
   over all SSRCs from the same session.  For Single Session it may only
   be needed when there otherwise would be an SSRC collision between the
   sessions.  This further assumes that the non-multiplexed side would
   never use any of the RTP mechanisms that require the same SSRC in
   multiple RTP sessions, as they cannot be gatewayed at all.  When
   translating an SSRC there is first of all an overhead, with SRTP that
   includes a complete authenticate, decrypt, encrypt and create a new
   authentication tag cycle.  In addition, the SSRC translation could
   potentially be a deployment obstacle for new RTP/RTCP extensions
   required to be understood by the translator to be correctly
   translated.  Therefore these two proposals gets a fail to meet the
   requirements.

B.9.  Summary and Conclusion

   This section contains a summary table of the high level outcome
   against the different requirements.

   A table mapping the requirements against the ID numbers used in the
   table is the following:






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   1: Support multiple RTP sessions over one transport flow

   2: Enable same SSRC value in multiple RTP sessions

      2.1:  Avoid SSRC translation in gateways/translators

      2.2:  Support existing extensions

   3: Ensure SRTP functions

   4: Don't Redefine used bits

   5: Firewall Friendly

   6: Monitoring and Reporting should still function

   7: Usable over Multicast

   8: Incremental deployment

   OH:  Overhead in Bytes. + means variable


          ---------------+---+---+---+---+---+---+---+---+---+----
          Solution       | 1 |2.1|2.2| 3 | 4 | 5 | 6 | 7 | 8 | OH
          ---------------+---+---+---+---+---+---+---+---+---+----
          Header Ext.    | S | S | P | P | F | P | F | S | P | 8+
          Multiplex Shim | S | S | S | S | S | P | F | S | S | 1
          Single Session | F | F | F | S | S | P | S | S | F | 0
          SRTP MKI Field | S | S | S | P | F | P | F | S | S | 4
          Padding Field  | S | S | S | F | P | P | F | S | P | 2
          Redefine SSRC  | S | F | F | P | F | P | P | S | S | 0
          ---------------+---+---+---+---+---+---+---+---+---+----

    Figure 5: Summary Table of Evaluation (Successfully (S), Partially
                   (P) or Fails (F) to meet requirement)

   Considering these options, the authors would recommend that AVTCORE
   standardize a solution based on a post or prefixed multiplexing
   field, i.e. a shim approach combined with the appropriate signalling
   as described in Appendix A.2.










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Authors' Addresses

   Magnus Westerlund
   Ericsson
   Farogatan 6
   SE-164 80 Kista
   Sweden

   Phone: +46 10 714 82 87
   Email: magnus.westerlund@ericsson.com


   Colin Perkins
   University of Glasgow
   School of Computing Science
   Glasgow  G12 8QQ
   United Kingdom

   Email: csp@csperkins.org
































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