AVTCORE WG                                                 M. Westerlund
Internet-Draft                                                  Ericsson
Updates: 3550, 3551 (if approved)                             C. Perkins
Intended status: Standards Track                   University of Glasgow
Expires: January 08, 21, 2016                                      J. Lennox
                                                           July 07, 20, 2015

        Sending Multiple Types of Media in a Single RTP Session


   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, 21, 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
   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   6
     5.1.  Allowing Multiple Media Types in an RTP Session . . . . .   7   6
     5.2.  Demultiplexing Media Streams  . . . . . . . . . . media types within an RTP session  . . . .   8   7
     5.3.  Per-SSRC Media Type Restrictions  . . . . . . . . . . . .   8
     5.4.  RTCP Considerations . . . . . . . . . . . . . . . . . . .   9   8
   6.  Extension Considerations  . . . . . . . . . . . . . . . . . .   9   8
     6.1.  RTP Retransmission Payload Format . . . . . . . . . . . .   9
     6.2.  RTP Payload Format for Generic FEC  . . . . . . . . . . .  11  10
     6.3.  RTP Payload Format for Redundant Audio  . . . . . . . . .  11
   7.  Signalling  . . . . . . . . . . . . . . . . . . . . . . . . .  12
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  12
   9.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  13  12
   10. Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  13
   11. References  . . . . . . . . . . . . . . . . . . . . . . . . .  13
     11.1.  Normative References . . . . . . . . . . . . . . . . . .  13
     11.2.  Informative References . . . . . . . . . . . . . . . . .  14  13
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  15  16

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

   [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)  Section 5 discusses issues arising from the base RTP and
   RTCP specification when using multiple types of media in a single RTP
   session, while Section 6 considers the impact of RTP extensions.  We
   discuss signalling in Section 7.  Finally, security considerations
   are discussed in Section 8.

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",
   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, 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

   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, the above issues, it might seem desirable appropriate for RTP-based
   applications to send all their media streams bundled into one RTP
   session, that runs
   on running over a single transport layer flow.  Unfortunately,  However, this
   is prohibited by the RTP specification, since because the design of 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  Various RTP extensions also make
   assumptions about SSRC use and RTCP reporting 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, type in certain circumstances, and gives
   guidance on when it is safe to perform such multiplexing. send multiple media types in a single
   RTP session.

4.  Applicability

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

   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

   Compatible Media Requirements: RTCP Behaviour:  The RTCP timing rules enforce a single
      RTCP reporting interval for all participants in an RTP session.
      Flows with very different media requirements, for
      example a low-rate audio flow with no sending rate or RTCP feedback needs and a high-
      quality video flow with different repair mechanisms,
      requirements cannot be multiplexed together together, since this results in leads to
      either excessive or insufficient RTCP for some flows, depending
      how the RTCP session bandwidth, and hence reporting interval, is
      configured.  For example, it is likely not feasible to find a
      single RTCP configuration that simultaneously suits both a low-
      rate audio flow with no feedback and a high-quality video flow
      with sophisticated RTCP-based feedback needs, making it difficult
      to combine these into a single RTP session.

   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 multiparty RTP sessions:  If it is desired to
      send multiple media types of media in a single RTP session:
      In multiparty communication scenarios it is important to separate
      two different cases.  One case is where the RTP session, then all
      participants in that session contains need to support sending multiple participants type
      of media in a common single 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 is not possible, in
      some implementations the
      general case, to implement a gateway that can interconnect an
      endpoint using multiple types of media sent using separate RTP mixers
      sessions, with one or more endpoints 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.

      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 send multiple media types of
      media 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 RTP session.

      One reason for this is that in case of multiple RTP sessions, the same SSRC value can safely be use used
      for different streams in both multiple RTP sessions without any
      issues, sessions, but when collapsed
      to a single RTP session there is an SSRC collision.  In addition some collisions can't  This would
      not be represented in
      the multiple separate RTP sessions.  For example, in a session
      with audio and video, an issue, since SSRC value used for video collision detection will not show
      up in the Audio RTP session at resolve 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
      conflict, except that participant
      and the rest of the RTP session.

      For the second case of separate some RTP sessions for each multiparty
      participant payload formats and a central node it is possible extensions use
      matching SSRCs to have identify related flows, and break when a mix of single
      RTP session users and is used.

      A middlebox that remaps SSRC values when combining multiple RTP session users as long as
      sessions into one is willing also needs to be aware of all possible RTCP
      packet types that might be used, so that it can remap the SSRCs used by a participant with
      multiple RTP sessions into non-used SSRC
      values in those packets.  This is impossible to do without
      restricting the single RTP
      session SSRC space for each set of the participants using a single RTP
      session with multiple media types.  It RTCP packet types that 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 those
      that are known by the RTP sessions.  It middlebox.  Such a middlebox might also suffer
      issues have
      difficulty due to differencies differences in configured RTCP bandwidth and
      other parameters between the RTP sessions.  It

      Finally, the use of a middlebox that translates SSRC values 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.

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

   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.

   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 [RFCXXXX] is used,
      under the restriction in Multiple Media Types in an RTP Session
      [RFCXXXX]. application conforms to the applicability constraints.
      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 media types within an RTP session

   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
   distinguish media streams.  Media  The RTP packets are demultiplexed into
   media streams are distinguished by based on their SSRC, and then the RTP payload type is then used
   to route data for a particular SSRC to select the right correct media decoder. decoding pathway for each media stream.

5.3.  Per-SSRC Media Type Restrictions

   An SSRC in an RTP session can change between media formats of the
   same type, subject to certain restrictions [RFC7160], but MUST NOT
   change media type during its lifetime.  For example, an SSRC can
   change between different audio formats, but cannot start sending audio,
   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
   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


   The RTP retransmission Retransmission Payload Format [RFC4588] is actually very
   straightforward.  Each retransmission can operate in either
   SSRC-multiplexed mode or session-multiplex mode.

   In SSRC-multiplexed mode, retransmitted RTP payload type is explicitly
   connected to an associated payload type. packets are sent in the
   same RTP session as the original packets, but use a different SSRC
   with the same RTCP SDES CNAME.  If retransmission is each endpoint sends only
   to be used with a subset of all payload types, single
   original RTP stream and a single retransmission RTP stream in the
   session, this is not a problem, sufficient.  If an endpoint sends multiple original
   and retransmission RTP streams, as it will be evident from would occur when sending multiple
   media types in a single RTP session, then each original RTP stream
   and the retransmission payload types which
   payload types RTP stream have to be associated using
   heuristics.  By having retransmission enabled requests outstanding for them.

   Session-multiplexed RTP only
   one SSRC not yet mapped, a receiver can determine the binding between
   original and retransmission RTP stream.  Another alternative is also possible to use where
   an retransmission session contains the retransmissions
   use of different RTP payload types, allowing the
   associated signalled "apt"
   (associated payload types in type) parameter of the source RTP session.  The only
   difference retransmission payload
   format to the previous case is if the source RTP session is one
   which contains multiple media types.  This results in be used to associate retransmitted and original packets.

   Session-multiplexed mode sends the retransmission streams RTP stream in the a
   separate RTP session to the original RTP stream, but using the same
   SSRC for each, with association being done by matching SSRCs between
   the two sessions.  This is unaffected by the retransmission
   having use of multiple associated media types.

   When using SDP signalling for
   types in a multiple single RTP session, since each media type will be sent
   using a different SSRC in the original RTP session, i.e.
   BUNDLE [I-D.ietf-mmusic-sdp-bundle-negotiation], and the session
   multiplexed case do require some recommendations on how to signal
   this.  To avoid breaking same
   SSRCs can be used in the semantics of retransmission session, allowing the streams
   to be associated.  This can be signalled using SDP with the BUNDLE
   [I-D.ietf-mmusic-sdp-bundle-negotiation] and FID grouping as defined
   by [RFC5888]
   extensions.  These SDP extensions require each media "m=" line can to only be
   included in one FID group. a single FID is used by group, but the RTP retransmission payload
   format uses FID groups to indicate the SDP media m= lines that is a source form an original
   and retransmission pair.  Thus, for SDP  Accordingly, when using
   extension to allow multiple media types to be sent in a single RTP
   session, each original media source (m= line) that is retransmitted
   needs a corresponding media m= 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

   An example (Figure 1) which shows SDP fragment showing the grouping
   structures. structures is provided
   in Figure 1.  This example is not legal SDP and only the most
   important attributes has have 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=rtpmap:0 PCMU/8000
       m=video 10000 RTP/AVP 31
       a=rtpmap:31 H261/90000
       m=video 10000 RTP/AVP 31
       a=rtpmap:31 H261/90000
       m=audio 40000 RTP/AVPF 99
       a=rtpmap:99 rtx/90000
       a=fmtp:99 apt=0;rtx-time=3000
       m=video 40000 RTP/AVPF 100
       a=rtpmap:100 rtx/90000
       a=fmtp:199 apt=31;rtx-time=3000
       m=video 40000 RTP/AVPF 100
       a=rtpmap:100 rtx/90000
       a=fmtp:199 apt=31;rtx-time=3000

      Figure 1: SDP example of Session Multiplexed RTP Retransmission

6.2.  RTP Payload Format for Generic FEC

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

   Independent [RFC2733]) can either send the FEC
   stream as a separate RTP Sessions, i.e.  where source and repair data are sent
   in different stream, or it can send the FEC combined with
   the original RTP sessions.  As this mode of configuration requires
   different stream as a redundant encoding [RFC2198].

   When sending FEC as a separate stream, the RTP session, there has Payload Format for
   generic FEC requires that FEC stream to be at least one sent in a separate RTP
   session for
   source data, this to the original stream, using the same SSRC, with the FEC
   stream being associated by matching the SSRC between sessions.  The
   RTP session used for the original streams can be one using include multiple RTP
   streams, and those RTP stream can use multiple media types.  The
   repair session only needs one RTP Payload type indicating repair to indicate FEC data, i.e.  x/ulpfec or x/parityfec depending if RFC 5109 or RFC 2733
   irrespective of the number of FEC streams sent, since the SSRC is used.  The
   used to associate the FEC streams with the original streams.  Hence,
   it is RECOMMENDED that FEC stream use the "application/ulpfec" media
   type in this session is not relevant for [RFC5109], and can in
   theory be any of the defined ones. "application/parityfec" media type for
   [RFC2733].  It is RECOMMENDED that one uses

   If one legal, but NOT RECOMMENDED, to send FEC streams
   using media specific payload format names (e.g., if an original RTP
   session contains audio and video flows, for the associated FEC RTP
   session where to use the "audio/ulpfec" and "video/ulpfec" payload
   formats), since this unnecessarily uses up RTP payload type values,
   and adds no value for demultiplexing since there might be multiple
   streams of the same media type).

   The combination of an original RTP session using multiple media types
   with a associated generic FEC session can be signalled using SDP signalling with
   the BUNDLE
   [I-D.ietf-mmusic-sdp-bundle-negotiation], then extension [I-D.ietf-mmusic-sdp-bundle-negotiation].  In
   this case, the RTP session carrying the FEC streams will be its own
   BUNDLE group.  The media m= line with the source stream for the FEC each original stream and the FEC stream's media m= line will be
   for the corresponding FEC stream are grouped using media line the SDP grouping using
   framework with either the FEC [RFC4756] or FEC-
   FR the FEC-FR [RFC5956]
   grouping.  This is very similar to the situation that
   arise arises for RTP
   retransmission with session multiplexing discussed
   above inSection in Section 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

   Note that the Source-Specific Media Attributes [RFC5576] specification defines
   an SDP syntax extension (the "FEC" semantic of the "ssrc-
   group" "ssrc-group" attribute)
   to signal FEC relationships between multiple RTP streams within a
   single RTP session.  However, this can't  This cannot be used as with generic FEC, since 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].

   When the FEC is sent as a redundant encoding, the considerations in
   Section 6.3 apply.

6.3.  RTP Payload Format for Redundant Audio

   In stream, using

   The RTP Payload Format for Redundant Audio Data [RFC2198]
   combining repair can be used to
   protect audio streams.  It can also be used along with the generic
   FEC payload format to send original and source repair data in the same RTP
   packets.  Both are compatible with RTP sessions containing multiple
   media types.

   This is
   possible to payload format requires each different redundant encoding use within a single
   different 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 number.  When used with generic FEC in
   sessions that contain multiple media types, this requires each media
   type use a different payload type for the FEC repair data payload format, i.e.  one for audio/ulpfec and one
   for text/ulpfec stream.  For example,
   if audio and text are combined sent in an a single RTP session.
   Secondly each combination of source payload and its session with generic ULP
   FEC repair data
   has sent as a redundant encoding for each, then payload types need to
   be an explicit configured assigned for FEC using the audio/ulpfec and text/ulpfec payload
   formats.  If multiple original payload types of used in the session,
   different redundant payload type. types need to be allocated for each one.
   This has potential
   for making to rapidly exhaust the limitation of available RTP payload types available into a real
   issue. type

7.  Signalling

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

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

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

   3.  Configure the  ensure RTP session level parameters, such as for example the RTCP RR and
       RS bandwidth, AVPF trr-int, underlying transport, bandwidth modifiers, the RTP/AVPF trr-int parameter, transport
       protocol, RTCP extensions in use, and any security parameters, commonly for
       are consistent across the RTP
       session. session; and

   4.  ensure that 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"

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. type are reused where possible, rather than
       configuring multiple code-points for the same thing.

   When combining using SDP signalling, the BUNDLE extension
   [I-D.ietf-mmusic-sdp-bundle-negotiation] is used to signal RTP
   sessions containing multiple media types in one
   session, their types.

8.  Security Considerations

   RTP provides a range of strong security requirements also have to mechanisms that can be combined by
   selecting used
   to secure sessions [RFC7201], [RFC7202].  The majority of these are
   independent of the most demanding for each property.  Thus having multiple type of media types can result sent 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
   for the need to be able session, however it
   is important to ensure check that the participants are aware
   that it security mechanism chosen is a multiple
   compatible with all types of media type sent within the session.  If not that is ensured it
   can cause issues

   Sending multiple media types in the a single RTP session for both the unaware and will generally
   require that all use the
   aware one.  Similar issues same security mechanism, whereas media sent
   using different RTP sessions can also be produced secured in an normal different ways.  When
   different media types have different security requirements, it might
   be necessary to send them using separate RTP
   session by creating configurations for sessions to meet those
   different end-points that
   doesn't match each other. requirements.  This can have significant costs in terms of
   resource usage, session set-up time, etc.

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

              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session",
              draft-ietf-avtcore-rtp-multi-stream-08 (work in progress),
              July 2015.

              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, DOI 10.17487/
              RFC2119, March 1997. 1997,

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
              July 2003. 2003, <http://www.rfc-editor.org/info/rfc3550>.

   [RFC3551]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
              Video Conferences with Minimal Control", STD 65, RFC 3551,
              DOI 10.17487/RFC3551, July 2003. 2003,

11.2.  Informative References

              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.

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

              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

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

              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.

              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., J.C., Vega-Garcia, A., and S. Fosse-
              Parisis, "RTP Payload for Redundant Audio Data", RFC 2198,
              DOI 10.17487/RFC2198, September 1997. 1997,

   [RFC2733]  Rosenberg, J. and H. Schulzrinne, "An RTP Payload Format
              for Generic Forward Error Correction", RFC 2733, DOI
              10.17487/RFC2733, December
              1999. 1999,

   [RFC4566]  Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
              Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
              July 2006. 2006, <http://www.rfc-editor.org/info/rfc4566>.

   [RFC4588]  Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R.
              Hakenberg, "RTP Retransmission Payload Format", RFC 4588,
              DOI 10.17487/RFC4588, July 2006. 2006,

   [RFC4756]  Li, A., "Forward Error Correction Grouping Semantics in
              Session Description Protocol", RFC 4756, DOI 10.17487/
              RFC4756, November 2006,

   [RFC5109]  Li, A., Ed., "RTP Payload Format for Generic Forward Error
              Correction", RFC 5109, DOI 10.17487/RFC5109, December 2007.
              2007, <http://www.rfc-editor.org/info/rfc5109>.

   [RFC5576]  Lennox, J., Ott, J., and T. Schierl, "Source-Specific
              Media Attributes in the Session Description Protocol
              (SDP)", RFC 5576, DOI 10.17487/RFC5576, June 2009. 2009,

   [RFC5761]  Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
              Control Packets on a Single Port", RFC 5761, DOI 10.17487/
              RFC5761, April 2010. 2010,

   [RFC5888]  Camarillo, G. and H. Schulzrinne, "The Session Description
              Protocol (SDP) Grouping Framework", RFC 5888, DOI 10.17487
              /RFC5888, June 2010. 2010,

   [RFC5956]  Begen, A., "Forward Error Correction Grouping Semantics in
              the Session Description Protocol", RFC 5956, DOI 10.17487/
              RFC5956, September
              2010. 2010,

   [RFC7160]  Petit-Huguenin, M. and G. Zorn, Ed., "Support for Multiple
              Clock Rates in an RTP Session", RFC 7160, DOI 10.17487/
              RFC7160, April 2014,

   [RFC7201]  Westerlund, M. and C. Perkins, "Options for Securing RTP
              Sessions", RFC 7201, DOI 10.17487/RFC7201, April 2014,

   [RFC7202]  Perkins, C. and M. Westerlund, "Securing the RTP
              Framework: Why RTP Does Not Mandate a Single Media
              Security Solution", RFC 7202, DOI 10.17487/RFC7202, April
              2014, <http://www.rfc-editor.org/info/rfc7202>.

Authors' Addresses

   Magnus Westerlund
   Farogatan 6
   SE-164 80 Kista

   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

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

   Email: jonathan@vidyo.com