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rfc:rfc8837



Internet Engineering Task Force (IETF) P. Jones Request for Comments: 8837 Cisco Systems Category: Standards Track S. Dhesikan ISSN: 2070-1721 Individual

                                                           C. Jennings
                                                         Cisco Systems
                                                              D. Druta
                                                                  AT&T
                                                          January 2021

Differentiated Services Code Point (DSCP) Packet Markings for WebRTC QoS

Abstract

 Networks can provide different forwarding treatments for individual
 packets based on Differentiated Services Code Point (DSCP) values on
 a per-hop basis.  This document provides the recommended DSCP values
 for web browsers to use for various classes of Web Real-Time
 Communication (WebRTC) traffic.

Status of This Memo

 This is an Internet Standards Track document.
 This document is a product of the Internet Engineering Task Force
 (IETF).  It represents the consensus of the IETF community.  It has
 received public review and has been approved for publication by the
 Internet Engineering Steering Group (IESG).  Further information on
 Internet Standards is available in Section 2 of RFC 7841.
 Information about the current status of this document, any errata,
 and how to provide feedback on it may be obtained at
 https://www.rfc-editor.org/info/rfc8837.

Copyright Notice

 Copyright (c) 2021 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
 (https://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.  Terminology
 3.  Relation to Other Specifications
 4.  Inputs
 5.  DSCP Mappings
 6.  Security Considerations
 7.  IANA Considerations
 8.  Downward References
 9.  References
   9.1.  Normative References
   9.2.  Informative References
 Acknowledgements
 Dedication
 Authors' Addresses

1. Introduction

 Differentiated Services Code Point (DSCP) [RFC2474] packet marking
 can help provide QoS in some environments.  This specification
 provides default packet marking for browsers that support WebRTC
 applications, but does not change any advice or requirements in other
 RFCs.  The contents of this specification are intended to be a simple
 set of implementation recommendations based on previous RFCs.
 Networks in which these DSCP markings are beneficial (likely to
 improve QoS for WebRTC traffic) include:
 1.  Private, wide-area networks.  Network administrators have control
     over remarking packets and treatment of packets.
 2.  Residential Networks.  If the congested link is the broadband
     uplink in a cable or DSL scenario, residential routers/NAT often
     support preferential treatment based on DSCP.
 3.  Wireless Networks.  If the congested link is a local wireless
     network, marking may help.
 There are cases where these DSCP markings do not help but, aside from
 possible priority inversion for "Less-than-Best-Effort traffic" (see
 Section 5), they seldom make things worse if packets are marked
 appropriately.
 DSCP values are, in principle, site specific with each site selecting
 its own code points for controlling per-hop behavior to influence the
 QoS for transport-layer flows.  However, in the WebRTC use cases, the
 browsers need to set them to something when there is no site-specific
 information.  This document describes a subset of DSCP code point
 values drawn from existing RFCs and common usage for use with WebRTC
 applications.  These code points are intended to be the default
 values used by a WebRTC application.  While other values could be
 used, using a non-default value may result in unexpected per-hop
 behavior.  It is RECOMMENDED that WebRTC applications use non-default
 values only in private networks that are configured to use different
 values.
 This specification defines inputs that are provided by the WebRTC
 application hosted in the browser that aid the browser in determining
 how to set the various packet markings.  The specification also
 defines the mapping from abstract QoS policies (flow type, priority
 level) to those packet markings.

2. Terminology

 The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
 "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
 "OPTIONAL" in this document are to be interpreted as described in BCP
 14 [RFC2119] [RFC8174] when, and only when, they appear in all
 capitals, as shown here.
 The terms "browser" and "non-browser" are defined in [RFC7742] and
 carry the same meaning in this document.

3. Relation to Other Specifications

 This document is a complement to [RFC7657], which describes the
 interaction between DSCP and real-time communications.  That RFC
 covers the implications of using various DSCP values, particularly
 focusing on the Real-time Transport Protocol (RTP) [RFC3550] streams
 that are multiplexed onto a single transport-layer flow.
 There are a number of guidelines specified in [RFC7657] that apply to
 marking traffic sent by WebRTC applications, as it is common for
 multiple RTP streams to be multiplexed on the same transport-layer
 flow.  Generally, the RTP streams would be marked with a value as
 appropriate from Table 1.  A WebRTC application might also multiplex
 data channel [RFC8831] traffic over the same 5-tuple as RTP streams,
 which would also be marked per that table.  The guidance in [RFC7657]
 says that all data channel traffic would be marked with a single
 value that is typically different from the value(s) used for RTP
 streams multiplexed with the data channel traffic over the same
 5-tuple, assuming RTP streams are marked with a value other than
 Default Forwarding (DF).  This is expanded upon further in the next
 section.
 This specification does not change or override the advice in any
 other RFCs about setting packet markings.  Rather, it simply selects
 a subset of DSCP values that is relevant in the WebRTC context.
 The DSCP value set by the endpoint is not trusted by the network.  In
 addition, the DSCP value may be remarked at any place in the network
 for a variety of reasons to any other DSCP value, including the DF
 value to provide basic best-effort service.  Even so, there is a
 benefit to marking traffic even if it only benefits the first few
 hops.  The implications are discussed in Section 3.2 of [RFC7657].
 Further, a mitigation for such action is through an authorization
 mechanism.  Such an authorization mechanism is outside the scope of
 this document.

4. Inputs

 This document recommends DSCP values for two classes of WebRTC flows:
  • media flows that are RTP streams [RFC8834]
  • data flows that are data channels [RFC8831]
 Each of the RTP streams and distinct data channels consist of all of
 the packets associated with an independent media entity, so an RTP
 stream or distinct data channel is not always equivalent to a
 transport-layer flow defined by a 5-tuple (source address,
 destination address, source port, destination port, and protocol).
 There may be multiple RTP streams and data channels multiplexed over
 the same 5-tuple, with each having a different level of importance to
 the application and, therefore, potentially marked using different
 DSCP values than another RTP stream or data channel within the same
 transport-layer flow.  (Note that there are restrictions with respect
 to marking different data channels carried within the same Stream
 Control Transmission Protocol (SCTP) association as outlined in
 Section 5.)
 The following are the inputs provided by the WebRTC application to
 the browser:
  • Flow Type: The application provides this input because it knows if

the flow is audio, interactive video ([RFC4594] [G.1010]) with or

    without audio, or data.
  • Application Priority: Another input is the relative importance of

an RTP stream or data channel. Many applications have multiple

    flows of the same flow type and some flows are often more
    important than others.  For example, in a video conference where
    there are usually audio and video flows, the audio flow may be
    more important than the video flow.  JavaScript applications can
    tell the browser whether a particular flow is of High, Medium,
    Low, or Very Low importance to the application.
 [RFC8835] defines in more detail what an individual flow is within
 the WebRTC context and priorities for media and data flows.
 Currently in WebRTC, media sent over RTP is assumed to be interactive
 [RFC8835] and browser APIs do not exist to allow an application to
 differentiate between interactive and non-interactive video.

5. DSCP Mappings

 The DSCP values for each flow type of interest to WebRTC based on
 application priority are shown in Table 1.  These values are based on
 the framework and recommended values in [RFC4594].  A web browser
 SHOULD use these values to mark the appropriate media packets.  More
 information on Expedited Forwarding (EF) and Assured Forwarding (AF)
 can be found in [RFC3246] and [RFC2597], respectively.  DF is Default
 Forwarding, which provides the basic best-effort service [RFC2474].
 WebRTC's use of multiple DSCP values may result in packets with
 certain DSCP values being blocked by a network.  See Section 4.2 of
 [RFC8835] for further discussion, including how WebRTC
 implementations establish and maintain connectivity when such
 blocking is encountered.
 +=======================+==========+=====+============+============+
 |       Flow Type       | Very Low | Low |   Medium   |    High    |
 +=======================+==========+=====+============+============+
 |         Audio         |  LE (1)  |  DF |  EF (46)   |  EF (46)   |
 |                       |          | (0) |            |            |
 +-----------------------+----------+-----+------------+------------+
 +-----------------------+----------+-----+------------+------------+
 |   Interactive Video   |  LE (1)  |  DF | AF42, AF43 | AF41, AF42 |
 | with or without Audio |          | (0) |  (36, 38)  |  (34, 36)  |
 +-----------------------+----------+-----+------------+------------+
 +-----------------------+----------+-----+------------+------------+
 | Non-Interactive Video |  LE (1)  |  DF | AF32, AF33 | AF31, AF32 |
 | with or without Audio |          | (0) |  (28, 30)  |  (26, 28)  |
 +-----------------------+----------+-----+------------+------------+
 +-----------------------+----------+-----+------------+------------+
 |          Data         |  LE (1)  |  DF |    AF11    |    AF21    |
 |                       |          | (0) |            |            |
 +-----------------------+----------+-----+------------+------------+
       Table 1: Recommended DSCP Values for WebRTC Applications
 The application priority, indicated by the columns "Very Low", "Low",
 "Medium", and "High", signifies the relative importance of the flow
 within the application.  It is an input that the browser receives to
 assist in selecting the DSCP value and adjusting the network
 transport behavior.
 The above table assumes that packets marked with LE are treated as
 lower effort (i.e., "less than best effort"), such as the LE behavior
 described in [RFC8622].  However, the treatment of LE is
 implementation dependent.  If an implementation treats LE as other
 than "less than best effort", then the actual priority (or, more
 precisely, the per-hop behavior) of the packets may be changed from
 what is intended.  It is common for LE to be treated the same as DF,
 so applications and browsers using LE cannot assume that LE will be
 treated differently than DF [RFC7657].  During development of this
 document, the CS1 DSCP was recommended for "very low" application
 priority traffic; implementations that followed that recommendation
 SHOULD be updated to use the LE DSCP instead of the CS1 DSCP.
 Implementers should also note that excess EF traffic is dropped.
 This could mean that a packet marked as EF may not get through,
 although the same packet marked with a different DSCP value would
 have gotten through.  This is not a flaw, but how excess EF traffic
 is intended to be treated.
 The browser SHOULD first select the flow type of the flow.  Within
 the flow type, the relative importance of the flow SHOULD be used to
 select the appropriate DSCP value.
 Currently, all WebRTC video is assumed to be interactive [RFC8835],
 for which the interactive video DSCP values in Table 1 SHOULD be
 used.  Browsers MUST NOT use the AF3x DSCP values (for non-
 interactive video in Table 1) for WebRTC applications.  Non-browser
 implementations of WebRTC MAY use the AF3x DSCP values for video that
 is known not to be interactive, e.g., all video in a WebRTC video
 playback application that is not implemented in a browser.
 The combination of flow type and application priority provides
 specificity and helps in selecting the right DSCP value for the flow.
 All packets within a flow SHOULD have the same application priority.
 In some cases, the selected application priority cell may have
 multiple DSCP values, such as AF41 and AF42.  These offer different
 drop precedences.  The different drop precedence values provide
 additional granularity in classifying packets within a flow.  For
 example, in a video conference, the video flow may have medium
 application priority, thus either AF42 or AF43 may be selected.  More
 important video packets (e.g., a video picture or frame encoded
 without any dependency on any prior pictures or frames) might be
 marked with AF42 and less important packets (e.g., a video picture or
 frame encoded based on the content of one or more prior pictures or
 frames) might be marked with AF43 (e.g., receipt of the more
 important packets enables a video renderer to continue after one or
 more packets are lost).
 It is worth noting that the application priority is utilized by the
 coupled congestion control mechanism for media flows per [RFC8699]
 and the SCTP scheduler for data channel traffic per [RFC8831].
 For reasons discussed in Section 6 of [RFC7657], if multiple flows
 are multiplexed using a reliable transport (e.g., TCP), then all of
 the packets for all flows multiplexed over that transport-layer flow
 MUST be marked using the same DSCP value.  Likewise, all WebRTC data
 channel packets transmitted over an SCTP association MUST be marked
 using the same DSCP value, regardless of how many data channels
 (streams) exist or what kind of traffic is carried over the various
 SCTP streams.  In the event that the browser wishes to change the
 DSCP value in use for an SCTP association, it MUST reset the SCTP
 congestion controller after changing values.  However, frequent
 changes in the DSCP value used for an SCTP association are
 discouraged, as this would defeat any attempts at effectively
 managing congestion.  It should also be noted that any change in DSCP
 value that results in a reset of the congestion controller puts the
 SCTP association back into slow start, which may have undesirable
 effects on application performance.
 For the data channel traffic multiplexed over an SCTP association, it
 is RECOMMENDED that the DSCP value selected be the one associated
 with the highest priority requested for all data channels multiplexed
 over the SCTP association.  Likewise, when multiplexing multiple
 flows over a TCP connection, the DSCP value selected SHOULD be the
 one associated with the highest priority requested for all
 multiplexed flows.
 If a packet enters a network that has no support for a flow-type-
 application priority combination specified in Table 1, then the
 network node at the edge will remark the DSCP value based on
 policies.  This could result in the flow not getting the network
 treatment it expects based on the original DSCP value in the packet.
 Subsequently, if the packet enters a network that supports a larger
 number of these combinations, there may not be sufficient information
 in the packet to restore the original markings.  Mechanisms for
 restoring such original DSCP is outside the scope of this document.
 In summary, DSCP marking provides neither guarantees nor promised
 levels of service.  However, DSCP marking is expected to provide a
 statistical improvement in real-time service as a whole.  The service
 provided to a packet is dependent upon the network design along the
 path, as well as the network conditions at every hop.

6. Security Considerations

 Since the JavaScript application specifies the flow type and
 application priority that determine the media flow DSCP values used
 by the browser, the browser could consider application use of a large
 number of higher priority flows to be suspicious.  If the server
 hosting the JavaScript application is compromised, many browsers
 within the network might simultaneously transmit flows with the same
 DSCP marking.  The Diffserv architecture requires ingress traffic
 conditioning for reasons that include protecting the network from
 this sort of attack.
 Otherwise, this specification does not add any additional security
 implications beyond those addressed in the following DSCP-related
 specifications.  For security implications on use of DSCP, please
 refer to Section 7 of [RFC7657] and Section 6 of [RFC4594].  Please
 also see [RFC8826] as an additional reference.

7. IANA Considerations

 This document has no IANA actions.

8. Downward References

 This specification contains downwards references to [RFC4594] and
 [RFC7657].  However, the parts of the former RFCs used by this
 specification are sufficiently stable for these downward references.
 The guidance in the latter RFC is necessary to understand the
 Diffserv technology used in this document and the motivation for the
 recommended DSCP values and procedures.

9. References

9.1. Normative References

 [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
            Requirement Levels", BCP 14, RFC 2119,
            DOI 10.17487/RFC2119, March 1997,
            <https://www.rfc-editor.org/info/rfc2119>.
 [RFC4594]  Babiarz, J., Chan, K., and F. Baker, "Configuration
            Guidelines for DiffServ Service Classes", RFC 4594,
            DOI 10.17487/RFC4594, August 2006,
            <https://www.rfc-editor.org/info/rfc4594>.
 [RFC7657]  Black, D., Ed. and P. Jones, "Differentiated Services
            (Diffserv) and Real-Time Communication", RFC 7657,
            DOI 10.17487/RFC7657, November 2015,
            <https://www.rfc-editor.org/info/rfc7657>.
 [RFC7742]  Roach, A.B., "WebRTC Video Processing and Codec
            Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,
            <https://www.rfc-editor.org/info/rfc7742>.
 [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
            2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
            May 2017, <https://www.rfc-editor.org/info/rfc8174>.
 [RFC8622]  Bless, R., "A Lower-Effort Per-Hop Behavior (LE PHB) for
            Differentiated Services", RFC 8622, DOI 10.17487/RFC8622,
            June 2019, <https://www.rfc-editor.org/info/rfc8622>.
 [RFC8826]  Rescorla, E., "Security Considerations for WebRTC",
            RFC 8826, DOI 10.17487/RFC8826, January 2021,
            <https://www.rfc-editor.org/info/rfc8826>.
 [RFC8831]  Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data
            Channels", RFC 8831, DOI 10.17487/RFC8831, January 2021,
            <https://www.rfc-editor.org/info/rfc8831>.
 [RFC8834]  Perkins, C., Westerlund, M., and J. Ott, "Media Transport
            and Use of RTP in WebRTC", RFC 8834, DOI 10.17487/RFC8834,
            January 2021, <https://www.rfc-editor.org/info/rfc8834>.
 [RFC8835]  Alvestrand, H., "Transports for WebRTC", RFC 8835,
            DOI 10.17487/RFC8835, January 2021,
            <https://www.rfc-editor.org/info/rfc8835>.

9.2. Informative References

 [G.1010]   ITU-T, "End-user multimedia QoS categories", ITU-T
            Recommendation G.1010, November 2001,
            <https://www.itu.int/rec/T-REC-G.1010-200111-I/en>.
 [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,
            DOI 10.17487/RFC2474, December 1998,
            <https://www.rfc-editor.org/info/rfc2474>.
 [RFC2597]  Heinanen, J., Baker, F., Weiss, W., and J. Wroclawski,
            "Assured Forwarding PHB Group", RFC 2597,
            DOI 10.17487/RFC2597, June 1999,
            <https://www.rfc-editor.org/info/rfc2597>.
 [RFC3246]  Davie, B., Charny, A., Bennet, J.C.R., Benson, K., Le
            Boudec, J.Y., Courtney, W., Davari, S., Firoiu, V., and D.
            Stiliadis, "An Expedited Forwarding PHB (Per-Hop
            Behavior)", RFC 3246, DOI 10.17487/RFC3246, March 2002,
            <https://www.rfc-editor.org/info/rfc3246>.
 [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
            Jacobson, "RTP: A Transport Protocol for Real-Time
            Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
            July 2003, <https://www.rfc-editor.org/info/rfc3550>.
 [RFC8699]  Islam, S., Welzl, M., and S. Gjessing, "Coupled Congestion
            Control for RTP Media", RFC 8699, DOI 10.17487/RFC8699,
            January 2020, <https://www.rfc-editor.org/info/rfc8699>.

Acknowledgements

 Thanks to David Black, Magnus Westerlund, Paolo Severini, Jim
 Hasselbrook, Joe Marcus, Erik Nordmark, Michael Tüxen, and Brian
 Carpenter for their invaluable input.

Dedication

 This document is dedicated to the memory of James Polk, a long-time
 friend and colleague.  James made important contributions to this
 specification, including serving initially as one of the primary
 authors.  The IETF global community mourns his loss and he will be
 missed dearly.

Authors' Addresses

 Paul E. Jones
 Cisco Systems
 Email: paulej@packetizer.com
 Subha Dhesikan
 Individual
 Email: sdhesikan@gmail.com
 Cullen Jennings
 Cisco Systems
 Email: fluffy@cisco.com
 Dan Druta
 AT&T
 Email: dd5826@att.com
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