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Versions: (draft-york-dart-dscp-rtp) 00 01 02 03 04 05 06 07 08 09 10 RFC 7657

DiffServ Applied to Real-time Transports                   D. Black, Ed.
Internet-Draft                                                       EMC
Intended status: Informational                                  P. Jones
Expires: February 9, 2015                                          Cisco
                                                          August 8, 2014


     Differentiated Services (DiffServ) and Real-time Communication
                      draft-ietf-dart-dscp-rtp-02

Abstract

   This document describes the interaction between Differentiated
   Services (DiffServ) network quality of service (QoS) functionality
   and real-time network communication, including communication based on
   the Real-time Transport Protocol (RTP).  DiffServ is based on network
   nodes applying different forwarding treatments to packets whose IP
   headers are marked with different DiffServ Code Points (DSCPs).  As a
   result, use of different DSCPs within a single traffic stream may
   cause transport protocol interactions (e.g., reordering).  In
   addition, DSCP markings may be changed or removed between the traffic
   source and destination.  This document covers the implications of
   these DiffServ aspects for real-time network communication, including
   RTCWEB.

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 February 9, 2015.

Copyright Notice

   Copyright (c) 2014 IETF Trust and the persons identified as the
   document authors.  All rights reserved.





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   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.  Real Time Communications  . . . . . . . . . . . . . . . . . .   3
     2.1.  RTP Background  . . . . . . . . . . . . . . . . . . . . .   3
     2.2.  RTP Multiplexing  . . . . . . . . . . . . . . . . . . . .   5
   3.   Differentiated Services (DiffServ) . . . . . . . . . . . . .   7
     3.1.  Diffserv PHBs (Per-Hop Behaviors) . . . . . . . . . . . .   9
     3.2.  Traffic Classifiers and DSCP Remarking  . . . . . . . . .   9
   4.  Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .  11
   5.  DiffServ Interactions . . . . . . . . . . . . . . . . . . . .  12
     5.1.  DiffServ, Reordering and Transport Protocols  . . . . . .  12
     5.2.  DiffServ, Reordering and Real-Time Communication  . . . .  14
     5.3.  Drop Precedence and Transport Protocols . . . . . . . . .  14
   6.  Guidelines  . . . . . . . . . . . . . . . . . . . . . . . . .  16
   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  17
   8.  Security Considerations . . . . . . . . . . . . . . . . . . .  17
   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  18
   10. References  . . . . . . . . . . . . . . . . . . . . . . . . .  18
     10.1.  Normative References . . . . . . . . . . . . . . . . . .  18
     10.2.  Informative References . . . . . . . . . . . . . . . . .  20
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  24

1.  Introduction

   This document describes the interactions between Differentiated
   Services (DiffServ) network quality of service (QoS) functionality
   [RFC2475] and real-time network communication, including
   communication based on the Real-time Transport Protocol (RTP)
   [RFC3550].  DiffServ is based on network nodes applying different
   forwarding treatments to packets whose IP headers are marked with
   different DiffServ Code Points (DSCPs)[RFC2474].  As a result use of
   different DSCPs within a single traffic stream may cause transport
   protocol interactions (e.g., reordering).  In addition, DSCP markings
   may be changed or removed between the traffic's source and
   destination.  This document covers the implications of these DiffServ
   aspects for real-time network communication, including RTCWEB traffic
   [I-D.ietf-rtcweb-overview].



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   The document is organized as follows.  Background is provided in
   Section 2 on real time communications and Section 3 on Differentiated
   Services.  Section 4 describes some examples of DiffServ usage with
   real time communications.  Section 5 explains how use of DiffServ
   features interacts with both transport and real time communications
   protocols and Section 6 provides guidance on DiffServ feature usage
   to control undesired interactions.  Security considerations are
   discussed in Section 8 (Section 7 is an empty IANA Considerations
   section).

2.  Real Time Communications

   Real-time communications enables communication in real-time over an
   IP network using voice, video, text, content sharing, etc.  It is
   possible to use one or more of these modalities in parallel to
   provide a richer communication experience.

   A simple example of real-time communications is a voice call placed
   over the Internet where an audio stream is transmitted in each
   direction between two users.  A more complex example is an immersive
   videoconferencing system that has multiple video screens, multiple
   cameras, multiple microphones, and some means of sharing content.
   For such complex systems, there may be multiple media streams that
   may be transmitted via a single IP address and port or via multiple
   IP addresses and ports.

2.1.  RTP Background

   The most common protocol used for real time media is the Real-Time
   Transport Protocol (RTP) [RFC3550].  RTP defines a common
   encapsulation format and handling rules for real-time data
   transmitted over the Internet.  Unfortunately, RTP terminology usage
   has been inconsistent.  For example, this document on RTP grouping
   terminology [I-D.ietf-avtext-rtp-grouping-taxonomy] observes that:

      RFC 3550 [RFC3550] uses the terms media stream, audio stream,
      video stream and streams of (RTP) packets interchangeably.

   Terminology in this document is based on that RTP grouping
   terminology document with the following terms being of particular
   importance (see that terminology document for full definitions):

   Source Stream:  A reference clock synchronized, time progressing,
      digital media stream.

   RTP Stream:  A stream of RTP packets containing media data, which may
      be source data or redundant data.  The RTP Packet Stream is




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      identified by an RTP synchronization source (SSRC) belonging to a
      particular RTP session.

   Media encoding and packetization of a source stream results in a
   source RTP stream plus zero or more redundancy RTP streams that
   provide resilience against loss of packets from the source RTP stream
   [I-D.ietf-avtext-rtp-grouping-taxonomy].  Redundancy information may
   also be carried in the same RTP stream as the encoded source stream,
   e.g., see Section 7.2 of [RFC5109].  With most applications, a single
   media type (e.g., audio) is transmitted within a single RTP session.
   However, it is possible to transmit multiple, distinct source streams
   over the same RTP session as one or more individual RTP streams.
   This is referred to as RTP multiplexing.  In addition, an RTP stream
   may contain multiple source streams that use the same reference clock
   (SSRC), e.g., components or programs in an MPEG Transport Stream
   [H.222.0].

   The number of source streams and RTP streams in an overall real-time
   interaction can be surprisingly large.  In addition to a voice source
   stream and a video source stream, there could be separate source
   streams for each of the cameras or microphones on a videoconferencing
   system.  As noted above, there might also be separate redundancy RTP
   streams that provide protection to a source RTP stream, using
   techniques such as Forward Error Correction.  Another example is
   simulcast transmission, where a video source stream can be
   transmitted as high resolution and low resolution RTP streams at the
   same time.  In this case, a media processing function might choose to
   send one or both RTP streams onward to a receiver based on bandwidth
   availability or who the active speaker is in a multipoint conference.
   Lastly, a transmitter might send a the same media content
   concurrently as two RTP streams using different encodings (e.g.,
   video/audio encoded as VP8 in parallel with H.264) to allow a media
   processing function to select a media encoding that best matches the
   capabilities of the receiver.

   For the RTCWEB protocol suite [I-D.ietf-rtcweb-transports], an
   individual source stream is a MediaStreamTrack, and a MediaStream
   contains one or more MediaStreamTracks
   [W3C.WD-mediacapture-streams-20130903].  A MediaStreamTrack is
   transmitted as a source RTP stream plus zero or more redundancy RTP
   streams, so a MediaStream that consists of one MediaStreamTrack is
   transmitted as a single source RTP stream plus zero or more
   redundancy RTP streams.  For more information on use of RTP in
   RTCWEB, see [I-D.ietf-rtcweb-rtp-usage].

   RTP is usually carried over a datagram protocol, such as
   UDP[RFC0768], UDP-Lite [RFC3828] or DCCP [RFC4340]; UDP is most
   commonly used.  Other transport protocols may also be used to



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   transmit real-time data or near-real-time data.  For example, SCTP
   [RFC4960] can be utilized to carry application sharing or
   whiteboarding information as part of an overall interaction that
   includes real time media.  These additional transport protocols can
   be multiplexed with an RTP session via UDP encapsulation, thereby
   using a single pair of UDP ports.

   The RTCWEB protocol suite encompasses a number of forms of
   multiplexing:

   1.  Individual source streams are carried in one or more individual
       RTP streams that can be multiplexed into a single RTP session as
       described in [RFC3550];

   2.  RTCP (see [RFC3550]) may be multiplexed with the RTP session as
       described in [RFC5761];

   3.  An RTP session could be multiplexed with other protocols via UDP
       encapsulation over a common pair of UDP ports as described in
       [RFC5764] as updated by
       [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]; and

   4.  The data may be further encapsulated via STUN [RFC5389] and TURN
       [RFC5766] for NAT (Network Address Translator) traversal.

   The resulting unidirectional UDP packet flow is identified by a
   5-tuple, i.e., a combination of two IP addresses (source and
   destination), two UDP ports (source and destination), and the use of
   the UDP protocol.  SDP bundle negotiation restrictions
   [I-D.ietf-mmusic-sdp-bundle-negotiation] limit RTCWEB to using at
   most a single DTLS session per UDP 5-tuple.  In contrast, multiple
   SCTP associations can be multiplexed over a single UDP 5-tuple
   [RFC6951].

   For IPv6, addition of the flow label [RFC6437] to 5-tuples results in
   6-tuples, but in practice, use of a flow label is unlikely to result
   in a finer-grain traffic subset than the corresponding 5-tuple (e.g.,
   the flow label is likely to represent the combination of two ports
   with use of the UDP protocol).  For that reason, discussion in this
   draft focuses on UDP 5-tuples.

2.2.  RTP Multiplexing

   Section 2.1 explains how source streams can be multiplexed over RTP
   sessions, which can in turn be multiplexed over UDP with packets
   generated by other transport protocols.  This section provides
   background on why this level of multiplexing is desirable.  The
   rationale in this section applies both to multiplexing of source



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   streams in RTP sessions and multiplexing of an RTP session with
   traffic from other transport protocols via UDP encapsulation.

   Multiplexing reduces the number of ports utilized for real-time and
   related communication in an overall interaction.  While a single
   endpoint might have plenty of ports available for communication, this
   traffic often traverses points in the network that are constrained on
   the number of available ports or whose performance degrades as the
   number of ports in use increases.  A good example is a Network
   Address Translator and Firewall (NAT/FW) device sitting at the
   network edge.  As the number of simultaneous protocol sessions
   increases, so does the burden placed on these devices to provide port
   mapping.

   The STUN [RFC5389] / ICE [RFC5245] / TURN [RFC5766] protocol family
   provides NAT/FW traversal and port mapping for protocols (e.g., those
   in the RTCWEB protocol suite) via communication with a relay server.
   These protocols were originally designed for use of UDP, however,
   they have been extended to use TCP as a transport for situations in
   which UDP does not work [RFC6062].

   Another reason for multiplexing is to help reduce the time required
   to establish bi-directional communication.  Since any two
   communicating users might be situated behind different NAT/FW
   devices, it is necessary to employ techniques like STUN/ICE/TURN in
   order to get traffic to flow between the two devices
   [I-D.ietf-rtcweb-transports].  Performing the tasks required of
   STUN/ICE/TURN take time, especially when multiple protocol sessions
   are involved.  While tasks for different sessions can be performed in
   parallel, it is nonetheless necessary for applications to wait for
   all sessions to be opened before communication between two users can
   begin.  Reducing the number of STUN/ICE/TURN steps reduces the
   likelihood of loss of a packet for one of these protocols; any such
   loss adds delay to setting up a communication session.  Further,
   reducing the number of STUN/ICE/TURN tasks places a lower burden on
   the STUN and TURN servers.

   Multiplexing may reduce the complexity and resulting load on an
   endpoint.  A single instance of STUN/ICE/TURN is simpler to execute
   and manage than multiple instances STUN/ICE/TURN operations happening
   in parallel, as the latter require synchronization and create more
   complex failure situations that have to be cleaned up by additional
   code.








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3.  Differentiated Services (DiffServ)

   The DiffServ architecture [RFC2475][RFC4594] is intended to enable
   scalable service discrimination in the Internet without requiring
   each network node to store per-flow state and participate in per-flow
   signaling.  The services may be end-to-end or within a network; they
   include both those that can satisfy quantitative performance
   requirements (e.g., peak bandwidth) and those based on relative
   performance (e.g., "class" differentiation).  Services can be
   constructed by a combination of well-defined building blocks deployed
   in network nodes that:

   o  classify traffic and set bits in an IP header field at network
      boundaries or hosts,

   o  use those bits to determine how packets are forwarded by the nodes
      inside the network, and

   o  condition the marked packets at network boundaries in accordance
      with the requirements or rules of each service.  Traffic
      conditioning may change the DSCP in a packet (remark it), delay
      the packet (as a consequence of traffic shaping) or drop the
      packet (as a consequence of traffic policing).

   A network node that supports DiffServ includes a classifier that
   selects packets based on the value of the DS field in IP headers (the
   DiffServ codepoint or DSCP), along with buffer management and packet
   scheduling mechanisms capable of delivering the specific packet
   forwarding treatment indicated by the DS field value.  Setting of the
   DS field and fine-grain conditioning of marked packets need only be
   performed at network boundaries; internal network nodes operate on
   traffic aggregates that share a DS field value, or in some cases, a
   small set of related values.

   The DiffServ architecture[RFC2475] maintains distinctions among:

   o  the QoS service provided to a traffic aggregate,

   o  the conditioning functions and per-hop behaviors (PHBs) used to
      realize services,

   o  the DSCP in the IP header used to mark packets to select a per-hop
      behavior, and

   o  the particular implementation mechanisms that realize a per-hop
      behavior.





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   This document focuses on PHBs and the usage of DSCPs to obtain those
   behaviors.  In a network node's forwarding path, the DSCP is used to
   map a packet to a particular forwarding treatment, or per-hop
   behavior (PHB) that specifies the forwarding treatment.

   A per-hop behavior (PHB) is a description of the externally
   observable forwarding behavior of a network node for network traffic
   marked with a DSCP that selects that PHB.  In this context,
   "forwarding behavior" is a general concept - for example, if only one
   DSCP is used for all traffic on a link, the observable forwarding
   behavior (e.g., loss, delay, jitter) will often depend only on the
   relative loading of the link.  To obtain useful behavioral
   differentiation, multiple traffic subsets are marked with different
   DSCPs for different PHBs for which node resources such as buffer
   space and bandwidth are allocated.  PHBs provide the framework for a
   DiffServ network node to allocate resources to traffic subsets, with
   network-scope differentiated services constructed on top of this
   basic hop-by-hop resource allocation mechanism.

   The codepoints (DSCPs) may be chosen from a small set of fixed values
   (the class selector codepoints), or from a set of recommended values
   defined in PHB specifications, or from values that have purely local
   meanings to a specific network that supports DiffServ; in general,
   packets may be forwarded across multiple such networks between source
   and destination.

   The mandatory DSCPs are the class selector code points as specified
   in [RFC2474].  The class selector codepoints (CS0-CS7) extend the
   deprecated concept of IP Precedence in the IPv4 header; three bits
   are added, so that the class selector DSCPs are of the form 'xxx000'.
   The all-zero DSCP ('000000' or CS0) designates a Default PHB that
   provides best-effort forwarding behavior and the remaining class
   selector code points are intended to provide relatively better per-
   hop-forwarding behavior in increasing numerical order, but:

   o  There is no requirement that any two adjacent class selector
      codepoints select different PHBs; adjacent class selector
      codepoints may use the same pool of resources on each network node
      in some networks.  This generalizes to ranges of class selector
      codepoints, but with limits - for example CS6 and CS7 are often
      used for network control (e.g., routing) traffic [RFC4594] and
      hence are likely to provide better forwarding behavior under
      network load in order to prioritize network recovery from
      disruptions.

   o  CS1 ('001000') was subsequently designated as the recommended
      codepoint for the Lower Effort (LE) PHB [RFC3662].  An LE service
      forwards traffic with "lower" priority than best effort and can be



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      "starved" by best effort and other "higher" priority traffic.  Not
      all networks offer an LE service, hence traffic marked with the
      CS1 DSCP may not receive lower effort forwarding; such traffic may
      be forwarded with a different PHB (the Default PHB is likely),
      remarked to another DSCP (CS0 is likely) and forwarded
      accordingly, or dropped.  See [RFC3662] for further discussion of
      the LE PHB and service.

   One cannot rely upon different class selector codepoints providing
   differentiated services or upon the presence of an LE service that is
   selected by the CS1 DSCP.  There is no effective way for a network
   endpoint to determine which PHBs are selected by the class selector
   codepoints or whether the CS1 DSCP selects an LE service on a
   specific network, let alone end-to-end.  Packets marked with the CS1
   DSCP may be forwarded with best effort service or another "higher"
   priority service, see [RFC2474].

3.1.  Diffserv PHBs (Per-Hop Behaviors)

   Although Differentiated Services is a general architecture that may
   be used to implement a variety of services, three fundamental
   forwarding behaviors (PHBs) have been defined and characterized for
   general use.  These are:

   1.  Default Forwarding (DF) for elastic traffic [RFC2474].  The
       Default PHB is always selected by the all-zero DSCP and provides
       best-effort forwarding.

   2.  Assured Forwarding (AF) [RFC2597] to provide differentiated
       service to elastic traffic.  Each instance of the AF behavior
       consists of three PHBs that differ only in drop precedence, e.g.,
       AF11, AF12 and AF13; such a set of three AF PHBs is referred to
       as an AF class, e.g., AF1x.  There are four defined AF classes,
       AF1x through AF4x, with higher numbered classes intended to
       receive better forwarding treatment than lower numbered classes.

   3.  Expedited Forwarding (EF) [RFC3246] intended for inelastic
       traffic.  Beyond the basic EF PHB, the VOICE-ADMIT PHB [RFC5865]
       is an admission controlled variant of the EF PHB.  Both of these
       PHBs are based on pre-configured limited forwarding capacity;
       traffic that exceeds that capacity may be shaped, remarked to a
       different DSCP, or dropped.

3.2.  Traffic Classifiers and DSCP Remarking

   DSCP markings are not end-to-end in general.  Each network can make
   its own decisions about what PHBs to use and which DSCP maps to each
   PHB.  While every PHB specification includes a recommended DSCP, and



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   RFC 4594 [RFC4594] recommends their end-to-end usage, there is no
   requirement that every network support any PHBs or use any specific
   DSCPs, with the exception of the class selector codepoint
   requirements in RFC 2474 [RFC2474].  When DiffServ is used, the edge
   or boundary nodes of a network are responsible for ensuring that all
   traffic entering that network conforms to that network's policies for
   DSCP and PHB usage, and such nodes remark traffic (change the DSCP
   marking as part of traffic conditioning) accordingly.  As a result,
   DSCP remarking is possible at any network boundary, including the
   first network node that traffic sent by a host encounters.  Remarking
   is also possible within a network, e.g., for traffic shaping.

   DSCP remarking is part of traffic conditioning; the traffic
   conditioning functionality applied to packets at a network node is
   determined by a traffic classifier [RFC2475].  Edge nodes of a
   DiffServ network classify traffic based on selected packet header
   fields; typical implementations do not look beyond the traffic's
   5-tuple in the IP and transport protocol headers.  As a result, when
   multiple DSCPs are used for traffic that shares a 5-tuple, remarking
   at a network boundary may result in all of the traffic being
   forwarded with a single DSCP, thereby removing any differentiation
   within the 5-tuple downstream of the remarking location.  Network
   nodes within a DiffServ network generally classify traffic based
   solely on DSCPs, but may perform finer grain traffic conditioning
   similar to that performed by edge nodes.

   So, for two arbitrary network endpoints, there can be no assurance
   that the DSCP set at the source endpoint will be preserved and
   presented at the destination endpoint.  Rather, it is quite likely
   that the DSCP will be set to zero (e.g., at the boundary of a network
   operator that distrusts or does not use the DSCP field) or to a value
   deemed suitable by an ingress classifier for whatever 5-tuple it
   carries.  DiffServ classifiers generally ignore embedded protocol
   headers (e.g., for SCTP or RTP embedded in UDP, header-based
   classification is unlikely to look beyond the outer UDP header).

   In addition, remarking may remove application-level distinctions in
   forwarding behavior - e.g., if multiple PHBs within an AF class are
   used to distinguish different types of frames within a video RTP
   stream, token-bucket-based remarkers operating in Color-Blind mode
   (see [RFC2697] and [RFC2698] for examples) may remark solely based on
   flow rate and burst behavior, removing the drop precedence
   distinctions specified by the source.

   Backbone and other carrier networks may employ a small number of
   DSCPs (e.g., less than half a dozen) in order to manage a small
   number of traffic aggregates; hosts that use a larger number of DSCPs
   can expect to find that much of their intended differentiation is



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   removed by such networks.  Better results may be achieved when DSCPs
   are used to spread traffic among a smaller number of DiffServ-based
   traffic subsets or aggregates, see [I-D.geib-tsvwg-diffserv-intercon]
   for one proposal.  This is of particular importance for MPLS-based
   networks due to the limited size of the Traffic Class (TC) field in
   an MPLS label [RFC5462] that is used to carry DiffServ information
   and the use of that TC field for other purposes, e.g., ECN [RFC5129].
   For further discussion on use of DiffServ with MPLS, see [RFC3270]
   and [RFC5127].

4.  Examples

   For real-time communications, one might want to mark the audio
   packets using EF and the video packets as AF41.  However, in a video
   conference receiving the audio packets ahead of the video is not
   useful because lip sync is necessary between audio and video.  It may
   still be desirable to send audio with a PHB that provides better
   service, because early arrival of audio helps assure smooth audio
   rendering, which is often more important than fully faithful video
   rendering.  There are also limits, as some devices have difficulties
   in synchronizing voice and video when packets that need to be
   rendered together arrive at significantly different times.  It makes
   more sense to use different PHBs when the audio and video source
   streams do not share a strict timing relationship.  For example,
   video content may be shared within a video conference via playback,
   perhaps of an unedited video clip that is intended to become part of
   a television advertisement.  Such content sharing video does not need
   precise synchronization with video conference audio, and could use a
   different PHB, as content sharing video is more tolerant to jitter,
   loss, and delay.

   Within a layered video RTP stream, ordering of frame communication is
   preferred, but importance of frame types varies, making use of PHBs
   with different drop precedences appropriate.  For example, I-frames
   that contain an entire image are usually more important than P-frames
   that contain only changes from the previous image because loss of a
   P-frame (or part thereof) can be recovered (at the latest) via the
   next I-frame, whereas loss of an I-frame (or part thereof) may cause
   rendering problems for all of the P-frames that depend on the missing
   I-frame.  For this reason, it is appropriate to mark I-frame packets
   with a PHB that has lower drop precedence than the PHB used for
   P-frames, as long as the PHBs preserve ordering among frames (e.g.,
   are in an AF class) - AF41 for I-frames and AF43 for P-frames is one
   possibility.  Additional spatial and temporal layers beyond the base
   video layer could also be marked with higher drop precedence than the
   base video layer, as their loss reduces video quality, but does not
   disrupt video rendering.




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   Additional RTP streams in a real-time communication interaction could
   be marked with CS0 and carried as best effort traffic.  One example
   is real-time text transmitted as specified in RFC 4103 [RFC4103].
   Best effort forwarding suffices because such real-time text has loose
   timing requirements; RFC 4103 recommends sending text in chunks every
   300ms.  Such text is technically real-time, but does not need a PHB
   promising better service than best effort, in contrast to audio or
   video.

5.  DiffServ Interactions

5.1.  DiffServ, Reordering and Transport Protocols

   Transport protocols provide data communication behaviors beyond those
   possible at the IP layer.  An important example is that TCP [RFC0793]
   provides reliable in-order delivery of data with congestion control.
   SCTP [RFC4960] provides additional properties such as preservation of
   message boundaries, and the ability to avoid head-of-line blocking
   that may occur with TCP.

   In contrast, UDP [RFC0768] is a basic unreliable datagram protocol
   that provides port-based multiplexing and demultiplexing on top of
   IP.  Two other unreliable datagram protocols are UDP-Lite [RFC3828],
   a variant of UDP that may deliver partially corrupt payloads when
   errors occur, and DCCP [RFC4340], which provides a range of
   congestion control modes for its unreliable datagram service.

   Transport protocols that provide reliable delivery (e.g., TCP, SCTP)
   are sensitive to network reordering of traffic.  When a protocol that
   provides reliable delivery receives a packet other than the next
   expected packet, the protocol usually assumes that the expected
   packet has been lost and respond with a retransmission request for
   that packet.  In addition, congestion control functionality in
   transport protocols usually infers congestion when packets are lost,
   creating an additional sensitivity to significant reordering - such
   reordering may be (mis-)interpreted as indicating congestion-caused
   packet loss, causing a reduction in transmission rate.  This remains
   true even when ECN [RFC3168] is in use, as ECN receivers are required
   to treat missing packets as potential indications of congestion.
   This requirement is based on two factors:

   o  Severe congestion may cause ECN-capable network nodes to drop
      packets, and

   o  ECN traffic may be forwarded by network nodes that do not support
      ECN and hence use packet drops to indicate congestion.





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   Congestion control is an important aspect of the Internet
   architecture, see [RFC2914] for further discussion.

   In general, marking packets with different DSCPs results in different
   PHBs being applied at network nodes, making reordering possible due
   to use of different pools of forwarding resources for each PHB.  The
   primary exception is that reordering is prohibited within each AF
   class (e.g., AF1x), as the three PHBs in an AF class differ solely in
   drop precedence.  Reordering within a PHB or AF class may occur for
   other transient reasons (e.g., route flap or ECMP rebalancing).

   Reordering also affects other forms of congestion control, such as
   techniques for RTP congestion control that were under development
   when this document was published, see
   [I-D.ietf-rmcat-cc-requirements] for requirements.  These techniques
   prefer use of a common (coupled) congestion controller for RTP
   streams between the same endpoints in order to reduce packet loss and
   delay by reducing competition for resources at any shared bottleneck.

   Shared bottlenecks can be detected via correlations of measured
   metrics such as one-way delay.  An alternative approach assumes that
   the set of packets on a single 5-tuple marked with DSCPs that do not
   allow reordering will utilize a common network path and common
   forwarding resources at each network node.  Under that assumption,
   any bottleneck encountered by such packets is shared among all of
   them, making it safe to use a common (coupled) congestion controller,
   see [I-D.welzl-rmcat-coupled-cc].  This is not a safe assumption when
   the packets involved are marked with DSCP values that allow
   reordering because a bottleneck may not be shared among all such
   packets (e.g., if the DSCPs result in use of different queues at a
   network node, only one of which is a bottleneck).

   Unreliable datagram protocols (e.g., UDP, UDP-Lite, DCCP) are not
   sensitive to reordering in the network, because they do not provide
   reliable delivery or congestion control.  On the other hand, when
   used to encapsulate other protocols (e.g., as UDP is used by RTCWEB,
   see Section 2.1), the reordering considerations for the encapsulated
   protocols apply.  For the specific usage of UDP by RTCWEB, every
   encapsulated protocol (i.e., RTP, SCTP and TCP) is sensitive to
   reordering as further discussed in this document.  In addition,
   [RFC5405] provides general guidelines for use of UDP (and UDP-Lite);
   the congestion control guidelines in that document apply to protocols
   encapsulated in UDP (or UDP-Lite).








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5.2.  DiffServ, Reordering and Real-Time Communication

   Real-time communications are also sensitive to network reordering of
   packets.  Such reordering may lead to spurious NACK generation and
   unneeded retransmission, as is the case for reliable delivery
   protocols (see Section 5.1).  The degree of sensitivity depends on
   protocol or stream timers, in contrast to reliable delivery protocols
   that usually react to all reordering.

   Receiver jitter buffers have important roles in the effect of
   reordering on real time communications:

   o  Minor packet reordering that is contained within a jitter buffer
      usually has no effect on rendering of the received RTP stream.

   o  Packet reordering that exceeds the capacity of a jitter buffer can
      cause user-perceptible quality problems (e.g., glitches, noise)
      for delay sensitive communication, such as interactive
      conversations.  Interactive real-time communication
      implementations often discard data that is sufficiently late that
      it cannot be rendered in source stream order, making
      retransmission counterproductive.  For this reason,
      implementations of interactive real-time communication often do
      not use retransmission.

   o  In contrast, replay of recorded media can tolerate significantly
      longer delays than interactive conversations, so replay is likely
      to use larger jitter buffers than interactive conversations.
      These larger jitter buffers increase the tolerance of replay to
      reordering by comparison to interactive conversations.  The size
      of the jitter buffer imposes an upper bound on replay tolerance to
      reordering, but does enable retransmission to be used when the
      jitter buffer is significantly larger than the amount of data that
      can be expected to arrive during the round-trip latency for
      retransmission.

   Network packet reordering caused by use of different DSCPs has no
   effective upper bound, and can exceed the size of any reasonable
   jitter buffer - in practice, the size of jitter buffers for replay is
   limited by external factors such as the amount of time that a human
   is willing to wait for replay to start.

5.3.  Drop Precedence and Transport Protocols

   Each DiffServ AF class consists of three PHBs that differ solely in
   drop precedence (e.g., AF3x consists of AF31, AF32 and AF33).
   Reordering is prohibited among packets on the same 5-tuple that use
   PHBs within a single AF class; further, these packets can be expected



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   to draw upon the same forwarding resources on network nodes (e.g.,
   use the same router queue) and hence use of multiple drop precedences
   within an AF class is not expected to impact latency.

   When PHBs within a single AF class are mixed for a protocol session,
   the resulting drop likelihood is a mix of the drop likelihoods of the
   PHBs involved.  The primary effect of multiple drop precedences is to
   influence decisions on what to drop with the goal that less important
   packets are dropped in preference to more important packets.

   There are situations in which drop precedences should not be mixed.
   A simple example is that there is little value in mixing drop
   precedences within a TCP connection, because TCP's ordered delivery
   behavior results in any drop requiring the receiver to wait for the
   dropped packet to be retransmitted.  Any resulting delay depends on
   the RTT and not the packet that was dropped.  Hence a single PHB and
   DSCP should be used for all packets in a TCP connection.

   As a consequence, when TCP is selected for NAT/FW traversal, a single
   PHB and DSCP should be used for all traffic on that TCP connection.
   An additional reason for this recommendation is that packetization
   for STUN/ICE/TURN occurs before passing the resulting packets to TCP;
   TCP resegmentation may result in a different packetization on the
   wire, breaking any association between DSCPs and specific data to
   which they are intended to apply.

   SCTP [RFC4960] differs from TCP in a number of ways, including the
   ability to deliver messages in an order that differs from the order
   in which they were sent and support for unreliable streams.  However,
   SCTP performs congestion control and retransmission across the entire
   association, and not on a per-stream basis.  Although there may be
   advantages to using multiple drop precedence across SCTP streams or
   within an SCTP stream that does not use reliable ordered delivery,
   there is no practical operational experience in doing so (e.g., the
   SCTP sockets API [RFC6458] does not support use of more than one DSCP
   for an SCTP association).  As a consequence, the impacts on SCTP
   protocol and implementation behavior are unknown and difficult to
   predict.  Hence a single PHB and DSCP should be used for all packets
   in an SCTP association, independent of the number or nature of
   streams in that association.  Similar reasoning applies to a DCCP
   connection; a single PHB and DSCP should be used because the scope of
   congestion control is the connection and there is no operational
   experience with using more than one PHB or DSCP.

   RTCP multi-stream reporting optimizations for an RTP session
   [I-D.ietf-avtcore-rtp-multi-stream-optimisation] assume that the RTP
   streams involved experience the same packet loss behavior.  This




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   mechanism is highly inappropriate when the RTP streams involved use
   different PHBs, even if those PHBs differ solely in drop precedence.

6.  Guidelines

   The only use of multiple standardized PHBs and DSCPs that prevents
   network reordering among packets marked with different DSCPs is use
   of PHBs within a single AF class.  All other uses of multiple PHBs
   and/or the class selector DSCPs allow network reordering of packets
   that are marked with different DSCPs.  Based on this and the
   foregoing discussion, the following requirements apply to use of
   DiffServ with real-time communications - applications and other
   traffic sources:

   o  Should not use different PHBs and DSCPs that allow reordering
      within a single RTP stream.  If this is not done, significant
      network reordering may overwhelm implementation assumptions about
      reordering limits, e.g., jitter buffer size, causing poor user
      experiences, see Section 5.2 above.  When a common (coupled)
      congestion controller is used across multiple RTP streams, this
      recommendation against use of PHBs and DSCPs that allow reordering
      applies across all of the RTP streams that are within the scope of
      a single common (coupled) congestion controller.

   o  Should use a single PHB and DSCP for an RTCP session, primarily to
      avoid RTCP reordering (and because there is no compelling reason
      for use of different drop precedences).  One of the PHBs and
      associated DSCP used for the associated RTP traffic would be an
      appropriate choice.

   o  Should use a single PHB and DSCP for all packets within a reliable
      transport protocol session (e.g., TCP connection, SCTP
      association) or DCCP connection.  Receivers for such protocols
      interpret reordering as indicating loss of some of the out-of-
      order packets; see Section 5.1 and there is no operational
      experience with multiple PHBs and DSCPs for SCTP or DCCP, see
      Section 5.3.  For SCTP, this requirement applies across the entire
      SCTP association, and not just to individual streams within an
      association because SCTP's reliable transmission functionality
      operates on the overall association.

   o  May use different PHBs and DSCPs that cause reordering within a
      single UDP (or UDP-Lite) 5-tuple, subject to the above
      constraints.  The service differentiation provided by such usage
      is unreliable, as it may be removed or changed by DSCP remarking
      at network boundaries as described in Section 3.2 above.





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   o  Cannot rely on end-to-end preservation of DSCPs as network node
      remarking can change DSCPs and remove drop precedence distinctions
      see Section 3.2 above.  For example, if a source uses drop
      precedence distinctions within an AF class to identify different
      types of video frames, using those DSCP values at the receiver to
      identify frame type is inherently unreliable.

   o  Should limit use of the CS1 codepoint to traffic for which best
      effort forwarding is acceptable, as network support for use of CS1
      to select a "less than best effort" PHB is inconsistent.  Further,
      some networks may treat CS1 as providing "better than best effort"
      forwarding behavior.

   There is no requirement in this document for network operators to
   differentiate traffic in any fashion.  Networks may support all of
   the PHBs discussed herein, classify EF and AFxx traffic identically,
   or even remark all traffic to best effort at some ingress points.
   Nonetheless, it is useful for network endpoints to provide finer
   granularity DSCP marking on packets for the benefit of networks that
   offer QoS service differentiation.  A specific example is that
   traffic originating from a browser may benefit from QoS service
   differentiation in within-building and residential access networks,
   even if the DSCP marking is subsequently removed or simplified.  This
   is because such networks and the boundaries between them are likely
   traffic bottleneck locations (e.g., due to customer aggregation onto
   common links and/or speed differences among links used by the same
   traffic).

   [Editor's note: rtcweb-transports draft is not aligned with the
   above.  The rtcweb WG and the draft author will bring it into line.]

7.  IANA Considerations

   This document includes no request to IANA.

8.  Security Considerations

   The security considerations for all of the technologies discussed in
   this document apply; in particular see the security considerations
   for RTP in [RFC3550] and DiffServ in [RFC2474] and[RFC2475].

   Multiplexing of multiple protocols onto a single UDP 5-tuple via
   encapsulation has implications for network functionality that
   monitors or inspects individual protocol flows, e.g., firewalls and
   traffic monitoring systems.  When implementations of such
   functionality lack visibility into encapsulated traffic (likely for
   many current implementations), it may be difficult or impossible to




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   apply network security policy and associated controls at a finer
   granularity than the overall UDP 5-tuple.

   Use of multiple PHBs and DSCPs that allow reordering within an
   overall real-time communication interaction enlarges the set of
   network forwarding resources used by that interaction, thereby
   increasing exposure to resource depletion or failure, independent of
   whether the underlying cause is benign or malicious.  This represents
   an increase in the effective attack surface of the interaction, and
   is a consideration in selecting an appropriate degree of QoS
   differentiation among the components of the real-time communication
   interaction.

   Use of multiple DSCPs to provide differentiated QoS service may
   reveal information about the encrypted traffic to which different
   service levels are provided.  For example, DSCP-based identification
   of RTP streams combined with packet frequency and packet size could
   reveal the type or nature of the encrypted source streams.  The IP
   header used for forwarding has to be unencrypted for obvious reasons,
   and the DSCP likewise has to be unencrypted in order to enable
   different IP forwarding behaviors to be applied to different packets.
   The nature of encrypted traffic components can be disguised via
   encrypted dummy data padding and encrypted dummy packets, e.g., see
   the discussion of traffic flow confidentiality in [RFC4303].
   Encrypted dummy packets could even be added in a fashion that an
   observer of the overall encrypted traffic might mistake for another
   encrypted RTP stream.

9.  Acknowledgements

   This document is the result of many conversations that have occurred
   within the dart working group and multiple other working groups in
   the RAI and Transport areas.  Many thanks to Harald Alvestrand, Erin
   Bournival, Brian Carpenter, Keith Drage, Gorry Fairhurst, Ruediger
   Geib, Cullen Jennings, Jonathan Lennox, Karen Nielsen, Colin Perkins,
   James Polk, Michael Welzl, Dan York and the dart WG participants for
   their reviews and comments.

10.  References

10.1.  Normative References

   [I-D.ietf-avtext-rtp-grouping-taxonomy]
              Lennox, J., Gross, K., Nandakumar, S., and G. Salgueiro,
              "A Taxonomy of Grouping Semantics and Mechanisms for Real-
              Time Transport Protocol (RTP) Sources", draft-ietf-avtext-
              rtp-grouping-taxonomy-02 (work in progress), June 2014.




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   [I-D.petithuguenin-avtcore-rfc5764-mux-fixes]
              Petit-Huguenin, M. and G. Salgueiro, "Multiplexing Scheme
              Updates for Secure Real-time Transport Protocol (SRTP)
              Extension for Datagram Transport Layer Security (DTLS)",
              draft-petithuguenin-avtcore-rfc5764-mux-fixes-00 (work in
              progress), July 2014.

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

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

   [RFC2474]  Nichols, K., Blake, S., Baker, F., and D. Black,
              "Definition of the Differentiated Services Field (DS
              Field) in the IPv4 and IPv6 Headers", RFC 2474, December
              1998.

   [RFC2475]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z.,
              and W. Weiss, "An Architecture for Differentiated
              Services", RFC 2475, December 1998.

   [RFC2597]  Heinanen, J., Baker, F., Weiss, W., and J. Wroclawski,
              "Assured Forwarding PHB Group", RFC 2597, June 1999.

   [RFC3246]  Davie, B., Charny, A., Bennet, J., Benson, K., Le Boudec,
              J., Courtney, W., Davari, S., Firoiu, V., and D.
              Stiliadis, "An Expedited Forwarding PHB (Per-Hop
              Behavior)", RFC 3246, March 2002.

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

   [RFC3662]  Bless, R., Nichols, K., and K. Wehrle, "A Lower Effort
              Per-Domain Behavior (PDB) for Differentiated Services",
              RFC 3662, December 2003.

   [RFC3828]  Larzon, L-A., Degermark, M., Pink, S., Jonsson, L-E., and
              G. Fairhurst, "The Lightweight User Datagram Protocol
              (UDP-Lite)", RFC 3828, July 2004.

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

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




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   [RFC5405]  Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines
              for Application Designers", BCP 145, RFC 5405, November
              2008.

   [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.

   [RFC5865]  Baker, F., Polk, J., and M. Dolly, "A Differentiated
              Services Code Point (DSCP) for Capacity-Admitted Traffic",
              RFC 5865, May 2010.

   [RFC6951]  Tuexen, M. and R. Stewart, "UDP Encapsulation of Stream
              Control Transmission Protocol (SCTP) Packets for End-Host
              to End-Host Communication", RFC 6951, May 2013.

10.2.  Informative References

   [H.222.0]  ITU-T, "H.222.0 : Information technology - Generic coding
              of moving pictures and associated audio information", June
              2012.

   [I-D.geib-tsvwg-diffserv-intercon]
              Geib, R., "DiffServ interconnection classes and practice",
              draft-geib-tsvwg-diffserv-intercon-06 (work in progress),
              July 2014.

   [I-D.ietf-avtcore-rtp-multi-stream-optimisation]
              Lennox, J., Westerlund, M., Wu, W., and C. Perkins,
              "Sending Multiple Media Streams in a Single RTP Session:
              Grouping RTCP Reception Statistics and Other Feedback",
              draft-ietf-avtcore-rtp-multi-stream-optimisation-03 (work
              in progress), July 2014.

   [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-07 (work in progress), April 2014.

   [I-D.ietf-rmcat-cc-requirements]
              Jesup, R., "Congestion Control Requirements For RMCAT",
              draft-ietf-rmcat-cc-requirements-05 (work in progress),
              July 2014.







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   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-10
              (work in progress), June 2014.

   [I-D.ietf-rtcweb-rtp-usage]
              Perkins, C., Westerlund, M., and J. Ott, "Web Real-Time
              Communication (WebRTC): Media Transport and Use of RTP",
              draft-ietf-rtcweb-rtp-usage-16 (work in progress), July
              2014.

   [I-D.ietf-rtcweb-transports]
              Alvestrand, H., "Transports for RTCWEB", draft-ietf-
              rtcweb-transports-05 (work in progress), June 2014.

   [I-D.welzl-rmcat-coupled-cc]
              Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion
              control for RTP media", draft-welzl-rmcat-coupled-cc-03
              (work in progress), May 2014.

   [RFC2697]  Heinanen, J. and R. Guerin, "A Single Rate Three Color
              Marker", RFC 2697, September 1999.

   [RFC2698]  Heinanen, J. and R. Guerin, "A Two Rate Three Color
              Marker", RFC 2698, September 1999.

   [RFC2914]  Floyd, S., "Congestion Control Principles", BCP 41, RFC
              2914, September 2000.

   [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
              of Explicit Congestion Notification (ECN) to IP", RFC
              3168, September 2001.

   [RFC3270]  Le Faucheur, F., Wu, L., Davie, B., Davari, S., Vaananen,
              P., Krishnan, R., Cheval, P., and J. Heinanen, "Multi-
              Protocol Label Switching (MPLS) Support of Differentiated
              Services", RFC 3270, May 2002.

   [RFC4103]  Hellstrom, G. and P. Jones, "RTP Payload for Text
              Conversation", RFC 4103, June 2005.

   [RFC4303]  Kent, S., "IP Encapsulating Security Payload (ESP)", RFC
              4303, December 2005.

   [RFC4594]  Babiarz, J., Chan, K., and F. Baker, "Configuration
              Guidelines for DiffServ Service Classes", RFC 4594, August
              2006.




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

   [RFC5127]  Chan, K., Babiarz, J., and F. Baker, "Aggregation of
              Diffserv Service Classes", RFC 5127, February 2008.

   [RFC5129]  Davie, B., Briscoe, B., and J. Tay, "Explicit Congestion
              Marking in MPLS", RFC 5129, January 2008.

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

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

   [RFC5462]  Andersson, L. and R. Asati, "Multiprotocol Label Switching
              (MPLS) Label Stack Entry: "EXP" Field Renamed to "Traffic
              Class" Field", RFC 5462, February 2009.

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

   [RFC5766]  Mahy, R., Matthews, P., and J. Rosenberg, "Traversal Using
              Relays around NAT (TURN): Relay Extensions to Session
              Traversal Utilities for NAT (STUN)", RFC 5766, April 2010.

   [RFC6062]  Perreault, S. and J. Rosenberg, "Traversal Using Relays
              around NAT (TURN) Extensions for TCP Allocations", RFC
              6062, November 2010.

   [RFC6437]  Amante, S., Carpenter, B., Jiang, S., and J. Rajahalme,
              "IPv6 Flow Label Specification", RFC 6437, November 2011.

   [RFC6458]  Stewart, R., Tuexen, M., Poon, K., Lei, P., and V.
              Yasevich, "Sockets API Extensions for the Stream Control
              Transmission Protocol (SCTP)", RFC 6458, December 2011.

   [W3C.WD-mediacapture-streams-20130903]
              Burnett, D., Bergkvist, A., Jennings, C., and A.
              Narayanan, "Media Capture and Streams", World Wide Web
              Consortium WD WD-mediacapture-streams-20130903, September
              2013, <http://www.w3.org/TR/2013/
              WD-mediacapture-streams-20130903>.





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Appendix A.  Change History

   [To be removed before RFC publication.]

   Changes from draft-york-dart-dscp-rtp-00 to -01

   o  Added examples (Section 5)

   o  Reworked text on RTP session multiplexing, at most one RTP session
      can be used per UDP 5-tuple.

   o  Initial terminology alignment with RTP grouping taxonomy draft.

   o  Added Section 2.5 on real-time communication interaction w/
      reordering based on text from Harald Alvestrand.

   o  Strengthened warnings on loss of differentiation, but indicate
      that differentiation may still be useful from source to point of
      loss.

   o  Added a few sentences on DiffServ and MPLS.

   o  Added discussion of UDP-encapsulated protocols that are reordering
      sensitive.

   o  Added initial security considerations.

   o  Many editorial changes

   Changes from draft-york-dart-dscp-rtp-01 to -02

   o  More terminology alignment with RTP grouping taxonomy draft: "RTP
      packet stream" -> "RTP stream"

   o  Aligned terminology for less-than-best-effort with RFC 3662 - LE
      (Lower Effort) PHB and service

   o  Minor reference updates

   Changes from draft-york-dart-dscp-rtp-02 to draft-ietf-dart-dscp-
   rtp-00

   o  Reduce author list and convert to Informational - remove RFC 2119
      reference and keywords

   o  Strengthen TCP and SCTP text.

   o  Add section 2.6 on drop precedence.



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   o  Remove discussion of multiplexing multiple RTP sessions on a
      single UDP 5-tuple

   o  Add discussions of RTCP,STUN/ICE/TURN and coupled congestion
      control

   o  Many editorial changes.

   o  Lots of additional references

   Changes from draft-ietf-dart-dscp-rtp-00 to draft-ietf-dart-dscp-
   rtp-01

   o  Merge the two TCP/SCTP guideline bullets.

   o  Add DCCP and UDP-Lite material, plus reference to RFC 5405 for UDP
      (and UDP-Lite) usage guidelines.

   o  Add "attack surface" security consideration.

   o  Many editorial changes.

   o  More references, and moved some references to normative.

   Changes from draft-ietf-dart-dscp-rtp-01 to draft-ietf-dart-dscp-
   rtp-02

   o  Reorganize text for better topic flow and make related edits.

Authors' Addresses

   David Black (editor)
   EMC
   176 South Street
   Hopkinton, MA  01748
   USA

   Phone: +1 508 293-7953
   Email: david.black@emc.com












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Internet-Draft        DiffServ and RT Communication          August 2014


   Paul Jones
   Cisco
   7025 Kit Creek Road
   Research Triangle Park, MA  27502
   USA

   Phone: +1 919 476 2048
   Email: paulej@packetizer.com











































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