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Versions: 00 01 02

DiffServ Applied to Real-time Transports                         D. York
Internet-Draft                                          Internet Society
Intended status: Standards Track                           D. Black, Ed.
Expires: January 4, 2015                                             EMC
                                                             C. Jennings
                                                                P. Jones
                                                                   Cisco
                                                            July 3, 2014


     Differentiated Services (DiffServ) and Real-time Communication
                      draft-york-dart-dscp-rtp-01

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
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   Drafts is at http://datatracker.ietf.org/drafts/current/.

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

   This Internet-Draft will expire on January 4, 2015.







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Copyright Notice

   Copyright (c) 2014 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
     1.1.  Requirements Language . . . . . . . . . . . . . . . . . .   3
   2.  Background  . . . . . . . . . . . . . . . . . . . . . . . . .   3
     2.1.  RTP Background  . . . . . . . . . . . . . . . . . . . . .   3
     2.2.  Differentiated Services (DiffServ) Background . . . . . .   5
     2.3.  Diffserv PHBs (Per-Hop Behaviors) . . . . . . . . . . . .   7
     2.4.  DiffServ, Reordering  and Transport Protocols . . . . . .   8
     2.5.  DiffServ, Reordering and Real-Time Communication  . . . .   9
     2.6.  Traffic Classifiers and DSCP Remarking  . . . . . . . . .  10
   3.  RTP Multiplexing Background . . . . . . . . . . . . . . . . .  11
   4.  Recommendations . . . . . . . . . . . . . . . . . . . . . . .  12
   5.  Examples  . . . . . . . . . . . . . . . . . . . . . . . . . .  13
   6.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .  14
   7.  Security Considerations . . . . . . . . . . . . . . . . . . .  14
   8.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  15
   9.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  15
     9.1.  Normative References  . . . . . . . . . . . . . . . . . .  15
     9.2.  Informative References  . . . . . . . . . . . . . . . . .  16
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  19

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



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   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].

1.1.  Requirements Language

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

2.  Background

   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 in order
   to provide a richer communication experience.

   A simple example of real-time communications is a voice call placed
   over the Internet wherein 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 Packet Stream:  A stream of RTP packets containing media data,
      which may be source data or redundant data.  The RTP Packet Stream



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      is 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 packet stream plus zero or more redundancy RTP packet
   streams that provide resilience against loss of packets from the
   source RTP packet stream [I-D.ietf-avtext-rtp-grouping-taxonomy].
   Redundancy information may also be carried in the same RTP packet
   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 packet streams.  This is
   referred to as RTP multiplexing.

   The number of source streams and RTP packet 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 packet streams that provide protection to a
   source RTP packet stream, using techniques such as Forward Error
   Correction.  Another example is simulcast transmission, where a video
   source stream can be transmitted at high resolution and low
   resolution RTP packet streams at the same time.  In this case, a
   media processing function might choose to send one or both RTP packet
   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 packet streams using different encodings (e.g., 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.

   Other transport protocols may also be used to transmit real-time data
   or near-real-time data.  For example, SCTP 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 [I-D.ietf-rtcweb-transports] employs two
   layers of multiplexing:

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





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

   For RTCWEB,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 packet stream plus zero or more
   redundancy RTP packet streams, so a MediaStream that consists of one
   MediaStreamTrack is transmitted as a single source RTP packet stream
   plus zero or more redundancy RTP packet streams.

   For more information on use of RTP in RTCWEB, see
   [I-D.ietf-rtcweb-rtp-usage].

   [I-D.westerlund-avtcore-transport-multiplexing] proposes to allow
   multiple RTP sessions to be multiplexed over a single UDP 5-tuple;
   the future of that expired proposal is uncertain.

   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.

   [Editor's Note: Multiple RTP sessions cannot be multiplexed on the
   same UDP 5-tuple, but what about multiple DTLS sessions for RTP?  RFC
   5764 appears to allow multiple DTLS sessions.]

   [Editor's Note: Should RTCP multiplexing w/RTP be mentioned here, as
   described in RFC 5761?]

2.2.  Differentiated Services (DiffServ) Background

   The DiffServ architecture 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:



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   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 (e.g., meter, mark, shape, police) at
      network boundaries in accordance with the requirements or rules of
      each service.

   A network node that supports DiffServ includes a classifier that
   selects packets based on the value of the DS field in IP headers,
   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 DS field value (DS codepoint, or 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.

   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



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   DiffServ network node to allocate resources to traffic subsets, with
   network-scope differentiated services constructed on top of this
   basic hop-by-hop (per-node) 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 were originally specified 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.

   o  CS1 ('001000') was subsequently recommended for "less than best
      effort" service when such a service is offered by a network
      [RFC3662].  Not all networks offer such a service.

   Applications and traffic sources cannot rely upon different class
   selector codepoints providing differentiated services or upon the
   presence of a "less than best effort" service that is selected by the
   CS1 DSCP, and there is no effective way for a network endpoint to
   determine whether the CS1 DSCP selects such a service on a specific
   network, let alone end-to-end.  Packets marked with the CS1 DSCP may
   be carried as best effort or even "better than best effort", see
   [RFC2474].

2.3.  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.



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   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 expected 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.

2.4.  DiffServ, Reordering and Transport Protocols

   [Editor's note: This section and the recommendations in Section 4 are
   centered on TCP, UDP, and SCTP.  They could use generalization to
   include other transport protocols - DCCP is a likely one to include,
   although it is not necessary to include every known transport
   protocol.]

   Transport protocols provide data communication behaviors beyond those
   possible at the IP layer.  An important example is that TCP provides
   reliable in-order delivery of data with congestion control.  SCTP
   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 is a basic unreliable datagram
   protocol that provides port-based multiplexing and demultiplexing on
   top of IP.

   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 for an ordered connection or stream, it 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





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

   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).

   UDP is the primary transport protocol that is not sensitive to
   reordering in the network, because it does not provide reliable
   delivery or congestion control.  On the other hand, when UDP is used
   to encapsulate other protocols (e.g., as is the case for RTCWEB, see
   Section 2.1), the reordering considerations for the encapsulated
   protocols apply.  For RTCWEB example in particular, every
   encapsulated protocol (i.e., RTP, SCTP and TCP) is sensitive to
   reordering as further discussed in this document.

2.5.  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 Section 2.4).  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 packet
      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 choose to discard data that is sufficiently
      late to prevent it from being rendered in source stream order,
      making retransmission counterproductive.  For this reason,
      implementations of interactive real-time communication often do
      not use retransmission.



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   o  In contrast, replay of recorded media can typically uses
      significantly larger jitter buffers than can be tolerated for
      interactive conversations, with the result that replay is more
      tolerant to reordering than 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
      will 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.

2.6.  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
   RFC 4594 [RFC4594] recommends their end-to-end usage, there is no
   requirement that every network support any PHBs or use any 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



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   presented at the destination endpoint.  On the contrary, 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 (MF) classifier for whatever
   5-tuple it carries.  DiffServ classifiers generally ignore embedded
   protocol headers (e.g., for SCTP or RTP embedded in UDP,
   classification will be only on 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
   packet 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
   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].

3.  RTP Multiplexing Background

   Section 2 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
   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.  A good example is a NAT/FW device
   sitting at the network edge.  As the number of simultaneous protocol
   sessions increases, so does the burden placed on these devices in
   order to provide port mapping.



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   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 and requiring an endpoint to perform these
   tasks for multiple protocol sessions can increase the time required.
   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 to users can begin.  Reducing
   the number of STUN/ICE/TURN steps reduces the probability of losing a
   packet and introducing delay in setting up a communication session.
   Further, reducing the number of STUN/ICE/TURN tasks means that there
   is a lower burden placed 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.

4.  Recommendations

   The only standardized use of multiple PHBs and DSCPs that avoids
   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 may cause reordering
      within a single RTP packet stream.  If this is not done,
      significant network reordering may overwhelm implementation
      assumptions about limits on reordering, e.g., jitter buffer size,
      causing poor user experiences, see Section Section 2.5 above.

   o  SHOULD NOT use different PHBs and DSCPs that may cause reordering
      within an ordered session for a reliable transport protocol (e.g.,
      TCP, SCTP).  Receivers for such protocols interpret reordering as
      indicating loss of out-of-order packets causing undesired
      retransmission requests, and will infer congestion from
      significant reordering, causing throughput reduction.  This
      requirement applies to both unencapsulated and encapsulated (e.g.,
      via UDP) uses of reliable transport protocols.



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   o  MAY use different PHBs and DSCPs that cause reordering within a
      single UDP 5-tuple, subject to the above constraints.  The service
      differentiation provided by such usage is unreliable, as it may be
      removed at network boundaries for the reasons described in
      Section 2.6 above.

   o  MUST NOT rely on end-to-end preservation of DSCPs as network node
      remarking can change DSCPs and remove drop precedence distinctions
      see Section 2.6 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 use the CS1 codepoint only for traffic that is acceptable
      to forward as best effort traffic, 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).

5.  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,



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   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 packet 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 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.

   Additional RTP packet 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 when redundancy
   encoding is used (as required by RFC 4103).  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.

6.  IANA Considerations

   This document includes no request to IANA.

7.  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 is
   based on monitoring or inspection of individual protocol flows, e.g.,
   firewalls and traffic monitoring systems.  When implementations of



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   such functionality lack visibility into encapsulated traffic (likely
   for many current implementations), it may be difficult or impossible
   to apply network security policy and controls at a finer grain than
   the overall UDP 5-tuple.

   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 packet 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 packet stream.

8.  Acknowledgements

   This document is the result of many conversations that have occurred
   within multiple working groups in the RAI and Transport areas.  Many
   thanks to Harald Alvestrand, Erin Bournival, Brian Carpenter,
   Ruediger Geib and James Polk for their reviews and input.

9.  References

9.1.  Normative References

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

   [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.

   [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.





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   [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.

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

9.2.  Informative References

   [I-D.geib-tsvwg-diffserv-intercon]
              Geib, R., "DiffServ interconnection classes and practice",
              draft-geib-tsvwg-diffserv-intercon-05 (work in progress),
              February 2014.

   [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-01 (work in progress), February
              2014.

   [I-D.ietf-rtcweb-overview]
              Alvestrand, H., "Overview: Real Time Protocols for Brower-
              based Applications", draft-ietf-rtcweb-overview-09 (work
              in progress), February 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-15 (work in progress), May
              2014.

   [I-D.ietf-rtcweb-transports]
              Alvestrand, H., "Transports for RTCWEB", draft-ietf-
              rtcweb-transports-04 (work in progress), April 2014.

   [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.




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

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

   [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.

   [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.





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

   [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.

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

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





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Authors' Addresses

   Dan York
   Internet Society
   Keene, N.H.
   USA

   Phone: +1-802-735-1624
   Email: dyork@lodestar2.com


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

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


   Cullen Jennings
   Cisco
   170 West Tasman Drive
   MS: SJC-21/2
   San Jose, CA  95134
   USA

   Phone: +1 408 421-9990
   Email: fluffy@cisco.com


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

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











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