AVTCORE WG                                                 M. Westerlund
Internet-Draft                                                  Ericsson
Updates: 3550, 3551 (if approved)                             C. Perkins
Intended status: Standards Track                   University of Glasgow
Expires: January 08, 2016                                      J. Lennox
                                                                   Vidyo
                                                           July 07, 2015


        Sending Multiple Types of Media in a Single RTP Session
             draft-ietf-avtcore-multi-media-rtp-session-08

Abstract

   This document specifies how an RTP session can contain RTP Streams
   with media from multiple media types such as audio, video, and text.
   This has been restricted by the RTP Specification, and thus this
   document updates RFC 3550 and RFC 3551 to enable this behaviour for
   applications that satisfy the applicability for using multiple media
   types in a single RTP session.

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
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   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 January 08, 2016.

Copyright Notice

   Copyright (c) 2015 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
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect



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   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  Background and Motivation . . . . . . . . . . . . . . . . . .   3
   4.  Applicability . . . . . . . . . . . . . . . . . . . . . . . .   4
   5.  Using Multiple Media Types in a Single RTP Session  . . . . .   7
     5.1.  Allowing Multiple Media Types in an RTP Session . . . . .   7
     5.2.  Demultiplexing Media Streams  . . . . . . . . . . . . . .   8
     5.3.  Per-SSRC Media Type Restrictions  . . . . . . . . . . . .   8
     5.4.  RTCP Considerations . . . . . . . . . . . . . . . . . . .   9
   6.  Extension Considerations  . . . . . . . . . . . . . . . . . .   9
     6.1.  RTP Retransmission Payload Format . . . . . . . . . . . .   9
     6.2.  RTP Payload Format for Generic FEC  . . . . . . . . . . .  11
     6.3.  RTP Payload Format for Redundant Audio  . . . . . . . . .  11
   7.  Signalling  . . . . . . . . . . . . . . . . . . . . . . . . .  12
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  12
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  13
   10. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  13
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .  13
     11.1.  Normative References . . . . . . . . . . . . . . . . . .  13
     11.2.  Informative References . . . . . . . . . . . . . . . . .  14
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  15

1.  Introduction

   The Real-time Transport Protocol [RFC3550] was designed to use
   separate RTP sessions to transport different types of media.  This
   implies that different transport layer flows are used for different
   media streams.  For example, a video conferencing application might
   send audio and video traffic RTP flows on separate UDP ports.  With
   increased use of network address/port translation, firewalls, and
   other middleboxes it is, however, becoming difficult to establish
   multiple transport layer flows between endpoints.  Hence, there is
   pressure to reduce the number of concurrent transport flows used by
   RTP applications.

   This memo updates [RFC3550] and [RFC3551] to allow multiple media
   types to be sent in a single RTP session in certain cases, thereby
   reducing the number of transport layer flows that are needed.  It
   makes no changes to RTP behaviour when using multiple RTP streams
   containing media of the same type (e.g., multiple audio streams or
   multiple video streams) in a single RTP session, however



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   [I-D.ietf-avtcore-rtp-multi-stream] provides important clarifications
   to RTP behaviour in that case.

   This memo is structured as follows.  Section 2 defines terminology.
   Section 3 further describes the background to, and motivation for,
   this memo and Section 4 describes the scenarios where this memo is
   applicable.  (tbd: fixme)

2.  Terminology

   The terms Encoded Stream, Endpoint, Media Source, RTP Session, and
   RTP Stream are used as defined in
   [I-D.ietf-avtext-rtp-grouping-taxonomy].  We also define the
   following terms:

   Media Type:  The general type of media data used by a real-time
      application.  The media type corresponds to the value used in the
      <media> field of an SDP m= line.  The media types defined at the
      time of this writing are "audio", "video", "text", "application",
      and "message".

   Quality of Service (QoS):  Network mechanisms that are intended to
      ensure that the packets within a flow or with a specific marking
      are transported with certain properties.

   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 [RFC2119].

3.  Background and Motivation

   RTP was designed to support multimedia sessions, containing multiple
   types of media sent simultaneously, by using multiple transport layer
   flows.  The existence of network address translators, firewalls, and
   other middleboxes complicates this, however, since a mechanism is
   needed to ensure that all the transport layer flows needed by the
   application can be established.  This has three consequences:

   1.  increased delay to establish a complete session, since each of
       the transport layer flows needs to be negotiated and established;

   2.  increased state and resource consumption in the middleboxes, that
       can lead to unexpected behaviour when middlebox resource limits
       are reached; and

   3.  increased risk that a subset of the transport layer flows will
       fail to be established, thus preventing the application from
       communicating.



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   Using fewer transport layer flows can hence be seen to reduce the
   risk of communication failure, and can lead to improved reliability
   and performance.

   One of the benefits of using multiple transport layer flows is that
   it makes it easy to use network layer quality of service (QoS)
   mechanisms to give differentiated performance for different flows.
   However, we note that many RTP-using application don't use network
   QoS features, and don't expect or desire any separation in network
   treatment of their 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.

   Given this, it might seem desirable for RTP-based applications to
   send all their media streams bundled into one RTP session, that runs
   on a single transport layer flow.  Unfortunately, this is prohibited
   by the RTP specification, since RTP makes certain assumptions that
   can be incompatible with sending multiple media types in a single RTP
   session.  Specifically, the RTP control protocol (RTCP) timing rules
   assume that all RTP media flows in a single RTP session have broadly
   similar RTCP reporting and feedback requirements, which can be
   problematic when different types of media are multiplexed together.
   Certain RTP extensions also make assumptions that are incompatible
   with sending different media types in a single RTP session.

   This memo updates [RFC3550] and [RFC3551] to allow RTP sessions to
   contain more than just one media type, and gives guidance on when it
   is safe to perform such multiplexing.

4.  Applicability

   This specification has limited applicability, and anyone intending to
   use it MUST ensure that their application and use meets the following
   criteria:

   Equal treatment of media:  The use of a single RTP session enforces
      similar treatment on all types of media used within the session.
      Applications that require significantly different network QoS or
      RTCP configuration for different media streams are better suited
      by sending those media streams on separate RTP session, using
      separate transport layer flows for each, since that gives greater
      flexibility.  Further guidance is given in
      [I-D.ietf-avtcore-multiplex-guidelines] and
      [I-D.ietf-dart-dscp-rtp].

   Compatible Media Requirements:  The RTCP timing rules enforce a
      single RTCP reporting interval for all participants in an RTP



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      session.  Flows with very different media requirements, for
      example a low-rate audio flow with no feedback needs and a high-
      quality video flow with different repair mechanisms, cannot be
      multiplexed together since this results in either excessive or
      insufficient RTCP for some flows, depending how the RTCP session
      bandwidth, and hence reporting interval, is configured.

   Signalled Support:  The extensions defined in this memo are not
      compatible with unmodified [RFC3550]-compatible endpoints.  Their
      use requires signalling and mutual agreement by all participants
      within an RTP session.  This requirement can be a problem for
      signalling solutions that can't negotiate with all participants.
      For declarative signalling solutions, mandating that the session
      is using multiple media types in one RTP session can be a way of
      attempting to ensure that all participants in the RTP session
      follow the requirement.  However, for signalling solutions that
      lack methods for enforcing that a receiver supports a specific
      feature, this can still cause issues.

   Consistent support for multiple media types in a single RTP session:
      In multiparty communication scenarios it is important to separate
      two different cases.  One case is where the RTP session contains
      multiple participants in a common RTP session.  This occurs for
      example in Any Source Multicast (ASM) and Relay (Transport
      Translator) topologies as defined in RTP Topologies
      [I-D.ietf-avtcore-rtp-topologies-update].  It can also occur in
      some implementations of RTP mixers that share the same SSRC/CSRC
      space across all participants.  The second case is when the RTP
      session is terminated in a middlebox and the other participants
      sources are projected or switched into each RTP session and
      rewritten on RTP header level including SSRC mappings.




















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      For the first case, with a common RTP session or at least shared
      SSRC/CSRC values, all participants in multiparty communication are
      REQUIRED to support multiple media types in an RTP session.  An
      participant using two or more RTP sessions towards a multiparty
      session can't be collapsed into a single session with multiple
      media types.  The reason is that in case of multiple RTP sessions,
      the same SSRC value can be use in both RTP sessions without any
      issues, but when collapsed to a single session there is an SSRC
      collision.  In addition some collisions can't be represented in
      the multiple separate RTP sessions.  For example, in a session
      with audio and video, an SSRC value used for video will not show
      up in the Audio RTP session at the participant using multiple RTP
      sessions, and thus not trigger any collision handling.  Thus any
      application using this type of RTP session structure MUST have a
      homogeneous support for multiple media types in one RTP session,
      or be forced to insert a translator node between that participant
      and the rest of the RTP session.

      For the second case of separate RTP sessions for each multiparty
      participant and a central node it is possible to have a mix of
      single RTP session users and multiple RTP session users as long as
      one is willing to remap the SSRCs used by a participant with
      multiple RTP sessions into non-used values in the single RTP
      session SSRC space for each of the participants using a single RTP
      session with multiple media types.  It can be noted that this type
      of implementation has to understand all types of RTP/RTCP
      extension being used in the RTP sessions to correctly be able to
      translate them between the RTP sessions.  It might also suffer
      issues due to differencies in configured RTCP bandwidth and other
      parameters between the RTP sessions.  It can also negatively
      impact the possibility for loop detection, as SSRC/CSRC can't be
      used to detect the loops, instead some other RTP stream or media
      source identity name space that is common across all interconnect
      parts are needed.

   Ability to operate with limited payload type space:  An RTP session
      has only a single 7-bit payload type space for all its payload
      type numbers.  Some applications might find this space limiting
      when media different media types and RTP payload formats are using
      within a single RTP session.

   Avoids incompatible Extensions:  Some RTP and RTCP extensions rely on
      the existence of multiple RTP sessions and relate media streams
      between sessions.  Others report on particular media types, and
      cannot be used with other media types.  Applications that send
      multiple types of media into a single RTP session need to avoid
      such extensions.




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5.  Using Multiple Media Types in a Single RTP Session

   This section defines what needs to be done or avoided to make an RTP
   session with multiple media types function without issues.

5.1.  Allowing Multiple Media Types in an RTP Session

   Section 5.2 of "RTP: A Transport Protocol for Real-Time Applications"
   [RFC3550] states:

      For example, in a teleconference composed of audio and video media
      encoded separately, each medium SHOULD be carried in a separate
      RTP session with its own destination transport address.

      Separate audio and video streams SHOULD NOT be carried in a single
      RTP session and demultiplexed based on the payload type or SSRC
      fields.

   This specification changes both of these sentences.  The first
   sentence is changed to:

      For example, in a teleconference composed of audio and video media
      encoded separately, each medium SHOULD be carried in a separate
      RTP session with its own destination transport address, unless
      specification [RFCXXXX] is followed and the application meets the
      applicability constraints.

   The second sentence is changed to:

      Separate audio and video media sources SHOULD NOT be carried in a
      single RTP session, unless the guidelines specified in [RFCXXXX]
      are followed.

   Second paragraph of Section 6 in RTP Profile for Audio and Video
   Conferences with Minimal Control [RFC3551] says:

      The payload types currently defined in this profile are assigned
      to exactly one of three categories or media types: audio only,
      video only and those combining audio and video.  The media types
      are marked in Tables 4 and 5 as "A", "V" and "AV", respectively.
      Payload types of different media types SHALL NOT be interleaved or
      multiplexed within a single RTP session, but multiple RTP sessions
      MAY be used in parallel to send multiple media types.  An RTP
      source MAY change payload types within the same media type during
      a session.  See the section "Multiplexing RTP Sessions" of RFC
      3550 for additional explanation.





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   This specifications purpose is to violate that existing SHALL NOT
   under certain conditions.  Thus this sentence also has to be changed
   to allow for multiple media type's payload types in the same session.
   The above sentence is changed to:

      Payload types of different media types SHALL NOT be interleaved or
      multiplexed within a single RTP session unless as specified and
      under the restriction in Multiple Media Types in an RTP Session
      [RFCXXXX].  Multiple RTP sessions MAY be used in parallel to send
      multiple media types.

   RFC-Editor Note: Please replace RFCXXXX with the RFC number of this
   specification when assigned.

5.2.  Demultiplexing Media Streams

   When receiving packets from a transport layer flow, an endpoint will
   first separate the RTP and RTCP packets from the non-RTP packets, and
   pass them to the RTP/RTCP protocol handler.  The RTP and RTCP packets
   are then demultiplexed based on their SSRC into the different media
   streams.  For each media stream, incoming RTCP packets are processed,
   and the RTP payload type is used to select the appropriate media
   decoder.

   This process remains the same irrespective of whether multiple media
   types are sent in a single RTP session or not.  It is important to
   note that the RTP payload type is never used to demultiplex media
   streams.  Media streams are distinguished by SSRC, and the payload
   type is then used to route data for a particular SSRC to the right
   media decoder.

5.3.  Per-SSRC Media Type Restrictions

   An SSRC in an RTP session MUST NOT change media type during its
   lifetime.  For example, an SSRC cannot start sending audio, then
   change to sending video.  The lifetime of an SSRC ends when an RTCP
   BYE packet for that SSRC is sent, or when it ceases transmission for
   long enough that it times out for the other participants in the
   session.

   The main motivation is that a given SSRC has its own RTP timestamp
   and sequence number spaces.  The same way that you can't send two
   encoded streams of audio on the same SSRC, you can't send one encoded
   audio and one encoded video stream on the same SSRC.  Each encoded
   stream when made into an RTP stream needs to have the sole control
   over the sequence number and timestamp space.  If not, one would not
   be able to detect packet loss for that particular encoded stream.
   Nor can one easily determine which clock rate a particular SSRCs



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   timestamp will increase with.  For additional arguments why RTP
   payload type based multiplexing of multiple media sources doesn't
   work see [I-D.ietf-avtcore-multiplex-guidelines].

   Within an RTP session where multiple media types have been configured
   for use, an SSRC can only send one type of media during its lifetime
   (i.e., it can switch between different audio codecs, since those are
   both the same type of media, but cannot switch between audio and
   video).  Different SSRCs MUST be used for the different media
   sources, the same way multiple media sources of the same media type
   already have to do.  The payload type will inform a receiver which
   media type the SSRC is being used for.  Thus the payload type MUST be
   unique across all of the payload configurations independent of media
   type that is used in the RTP session.

5.4.  RTCP Considerations

   When sending multiple types of media that have different rates in a
   single RTP session, endpoints MUST follow the guidelines for handling
   RTCP described in Section 7 of [I-D.ietf-avtcore-rtp-multi-stream].

6.  Extension Considerations

   This section outlines known issues and incompatibilities with RTP and
   RTCP extensions when multiple media types are used in a single RTP
   sessions.  Future extensions to RTP and RTCP need to consider, and
   document, any potential incompatibility.

6.1.  RTP Retransmission Payload Format

   SSRC-multiplexed RTP retransmission [RFC4588] is actually very
   straightforward.  Each retransmission RTP payload type is explicitly
   connected to an associated payload type.  If retransmission is only
   to be used with a subset of all payload types, this is not a problem,
   as it will be evident from the retransmission payload types which
   payload types have retransmission enabled for them.

   Session-multiplexed RTP retransmission is also possible to use where
   an retransmission session contains the retransmissions of the
   associated payload types in the source RTP session.  The only
   difference to the previous case is if the source RTP session is one
   which contains multiple media types.  This results in the
   retransmission streams in the RTP session for the retransmission
   having multiple associated media types.

   When using SDP signalling for a multiple media type RTP session, i.e.
   BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation], the session
   multiplexed case do require some recommendations on how to signal



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   this.  To avoid breaking the semantics of the FID grouping as defined
   by [RFC5888] each media line can only be included in one FID group.
   FID is used by RTP retransmission to indicate the SDP media lines
   that is a source and retransmission pair.  Thus, for SDP using
   BUNDLE, each original media source (m= line) that is retransmitted
   needs a corresponding media line in the retransmission RTP session.
   In case there are multiple media lines for retransmission, these
   media lines will form a independent BUNDLE group from the BUNDLE
   group with the source streams.

   Below is an SDP example (Figure 1) which shows the grouping
   structures.  This example is not legal SDP and only the most
   important attributes has been left in place.  Note that this SDP is
   not an initial BUNDLE offer.  As can be seen there are two bundle
   groups, one for the source RTP session and one for the
   retransmissions.  Then each of the media sources are grouped with its
   retransmission flow using FID, resulting in three more groupings.

       a=group:BUNDLE foo bar fiz
       a=group:BUNDLE zoo kelp glo
       a=group:FID foo zoo
       a=group:FID bar kelp
       a=group:FID fiz glo
       m=audio 10000 RTP/AVP 0
       a=mid:foo
       a=rtpmap:0 PCMU/8000
       m=video 10000 RTP/AVP 31
       a=mid:bar
       a=rtpmap:31 H261/90000
       m=video 10000 RTP/AVP 31
       a=mid:fiz
       a=rtpmap:31 H261/90000
       m=audio 40000 RTP/AVPF 99
       a=rtpmap:99 rtx/90000
       a=fmtp:99 apt=0;rtx-time=3000
       a=mid:zoo
       m=video 40000 RTP/AVPF 100
       a=rtpmap:100 rtx/90000
       a=fmtp:199 apt=31;rtx-time=3000
       a=mid:kelp
       m=video 40000 RTP/AVPF 100
       a=rtpmap:100 rtx/90000
       a=fmtp:199 apt=31;rtx-time=3000
       a=mid:glo

      Figure 1: SDP example of Session Multiplexed RTP Retransmission





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6.2.  RTP Payload Format for Generic FEC

   The RTP Payload Format for Generic Forward Error Correction
   [RFC5109], and also its predecessor [RFC2733], requires some
   considerations, and they are different depending on what type of
   configuration of usage one has.

   Independent RTP Sessions, i.e.  where source and repair data are sent
   in different RTP sessions.  As this mode of configuration requires
   different RTP session, there has to be at least one RTP session for
   source data, this session can be one using multiple media types.  The
   repair session only needs one RTP Payload type indicating repair
   data, i.e.  x/ulpfec or x/parityfec depending if RFC 5109 or RFC 2733
   is used.  The media type in this session is not relevant and can in
   theory be any of the defined ones.  It is RECOMMENDED that one uses
   "Application".

   If one uses SDP signalling with BUNDLE
   [I-D.ietf-mmusic-sdp-bundle-negotiation], then the RTP session
   carrying the FEC streams will be its own BUNDLE group.  The media
   line with the source stream for the FEC and the FEC stream's media
   line will be grouped using media line grouping using the FEC or FEC-
   FR [RFC5956] grouping.  This is very similar to the situation that
   arise for RTP retransmission with session multiplexing discussed
   above inSection 6.1.

   The RTP Payload Format for Generic Forward Error Correction [RFC5109]
   and its predecessor [RFC2733] requires a separate RTP session unless
   the FEC data is carried in RTP Payload for Redundant Audio Data
   [RFC2198].

   Note that the Source-Specific Media Attributes [RFC5576]
   specification defines an SDP syntax (the "FEC" semantic of the "ssrc-
   group" attribute) to signal FEC relationships between multiple RTP
   streams within a single RTP session.  However, this can't be used as
   the FEC repair packets need to have the same SSRC value as the source
   packets being protected.  [RFC5576] does not normatively update and
   resolve that restriction.  There is ongoing work on an ULP extension
   to allow it be use FEC RTP streams within the same RTP Session as the
   source stream [I-D.lennox-payload-ulp-ssrc-mux].

6.3.  RTP Payload Format for Redundant Audio

   In stream, using RTP Payload for Redundant Audio Data [RFC2198]
   combining repair and source data in the same packets.  This is
   possible to use within a single RTP session.  However, the usage and
   configuration of the payload types can create an issue.  First of all
   it might be necessary to have one payload type per media type for the



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   FEC repair data payload format, i.e.  one for audio/ulpfec and one
   for text/ulpfec if audio and text are combined in an RTP session.
   Secondly each combination of source payload and its FEC repair data
   has to be an explicit configured payload type.  This has potential
   for making the limitation of RTP payload types available into a real
   issue.

7.  Signalling

   Establishing an RTP session with multiple media types requires
   signalling.  This signalling needs to fulfil the following
   requirements:

   1.  Ensure that any participant in the RTP session is aware that this
       is an RTP session with multiple media types.

   2.  Ensure that the payload types in use in the RTP session are using
       unique values, with no overlap between the media types.

   3.  Configure the RTP session level parameters, such as RTCP RR and
       RS bandwidth, AVPF trr-int, underlying transport, the RTCP
       extensions in use, and security parameters, commonly for the RTP
       session.

   4.  RTP and RTCP functions that can be bound to a particular media
       type SHOULD be reused when possible also for other media types,
       instead of having to be configured for multiple code-points.
       Note: In some cases one will not have a choice but to use
       multiple configurations.

   The signalling of multiple media types in one RTP session in SDP is
   specified in "Multiplexing Negotiation Using Session Description
   Protocol (SDP) Port Numbers"
   [I-D.ietf-mmusic-sdp-bundle-negotiation].

8.  Security Considerations

   Having an RTP session with multiple media types doesn't change the
   methods for securing a particular RTP session.  One possible
   difference is that the different media have often had different
   security requirements.  When combining multiple media types in one
   session, their security requirements also have to be combined by
   selecting the most demanding for each property.  Thus having multiple
   media types can result in increased overhead for security for some
   media types to ensure that all requirements are meet.

   Otherwise, the recommendations for how to configure and RTP session
   do not add any additional requirements compared to normal RTP, except



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   for the need to be able to ensure that the participants are aware
   that it is a multiple media type session.  If not that is ensured it
   can cause issues in the RTP session for both the unaware and the
   aware one.  Similar issues can also be produced in an normal RTP
   session by creating configurations for different end-points that
   doesn't match each other.

9.  IANA Considerations

   This memo makes no request of IANA.

10.  Acknowledgements

   The authors would like to thank Christer Holmberg, Gunnar Hellstroem,
   and Charles Eckel for the feedback on the document.

11.  References

11.1.  Normative References

   [I-D.ietf-avtcore-rtp-multi-stream]
              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-07 (work in progress),
              March 2015.

   [I-D.ietf-mmusic-sdp-bundle-negotiation]
              Holmberg, C., Alvestrand, H., and C. Jennings,
              "Negotiating Media Multiplexing Using the Session
              Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
              negotiation-22 (work in progress), June 2015.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              July 2003.









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11.2.  Informative References

   [I-D.ietf-avtcore-multiplex-guidelines]
              Westerlund, M., Perkins, C., and H. Alvestrand,
              "Guidelines for using the Multiplexing Features of RTP to
              Support Multiple Media Streams", draft-ietf-avtcore-
              multiplex-guidelines-03 (work in progress), October 2014.

   [I-D.ietf-avtcore-rtp-topologies-update]
              Westerlund, M. and S. Wenger, "RTP Topologies", draft-
              ietf-avtcore-rtp-topologies-update-10 (work in progress),
              July 2015.

   [I-D.ietf-avtext-rtp-grouping-taxonomy]
              Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and
              B. Burman, "A Taxonomy of Semantics and Mechanisms for
              Real-Time Transport Protocol (RTP) Sources", draft-ietf-
              avtext-rtp-grouping-taxonomy-07 (work in progress), June
              2015.

   [I-D.ietf-dart-dscp-rtp]
              Black, D. and P. Jones, "Differentiated Services
              (DiffServ) and Real-time Communication", draft-ietf-dart-
              dscp-rtp-10 (work in progress), November 2014.

   [I-D.lennox-payload-ulp-ssrc-mux]
              Lennox, J., "Supporting Source-Multiplexing of the Real-
              Time Transport Protocol (RTP) Payload for Generic Forward
              Error Correction", draft-lennox-payload-ulp-ssrc-mux-00
              (work in progress), February 2013.

   [I-D.westerlund-avtcore-transport-multiplexing]
              Westerlund, M. and C. Perkins, "Multiplexing Multiple RTP
              Sessions onto a Single Lower-Layer Transport", draft-
              westerlund-avtcore-transport-multiplexing-07 (work in
              progress), October 2013.

   [RFC2198]  Perkins, C., Kouvelas, I., Hodson, O., Hardman, V.,
              Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse-
              Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
              September 1997.

   [RFC2733]  Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format
              for Generic Forward Error Correction", RFC 2733, December
              1999.

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, July 2006.



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   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              July 2006.

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

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, June 2009.

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

   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", RFC 5888, June 2010.

   [RFC5956]  Begen, A., "Forward Error Correction Grouping Semantics in
              the Session Description Protocol", RFC 5956, September
              2010.

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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   Jonathan Lennox
   Vidyo, Inc.
   433 Hackensack Avenue
   Seventh Floor
   Hackensack, NJ  07601
   US

   Email: jonathan@vidyo.com










































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