Skip to main content

Differentiated Services Code Point (DSCP) Packet Markings for WebRTC QoS
RFC 8837

Document Type RFC - Proposed Standard (January 2021)
Authors Paul Jones , Subha Dhesikan , Cullen Fluffy Jennings , Dan Druta
Last updated 2021-01-18
RFC stream Internet Engineering Task Force (IETF)
Additional resources Mailing list discussion
IESG Responsible AD Magnus Westerlund
Send notices to (None)
RFC 8837

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
                                                            January 2021

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


   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

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
   ( 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
   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

   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

   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",
   "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

   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,

   [RFC4594]  Babiarz, J., Chan, K., and F. Baker, "Configuration
              Guidelines for DiffServ Service Classes", RFC 4594,
              DOI 10.17487/RFC4594, August 2006,

   [RFC7657]  Black, D., Ed. and P. Jones, "Differentiated Services
              (Diffserv) and Real-Time Communication", RFC 7657,
              DOI 10.17487/RFC7657, November 2015,

   [RFC7742]  Roach, A.B., "WebRTC Video Processing and Codec
              Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,

   [RFC8174]  Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
              2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
              May 2017, <>.

   [RFC8622]  Bless, R., "A Lower-Effort Per-Hop Behavior (LE PHB) for
              Differentiated Services", RFC 8622, DOI 10.17487/RFC8622,
              June 2019, <>.

   [RFC8826]  Rescorla, E., "Security Considerations for WebRTC",
              RFC 8826, DOI 10.17487/RFC8826, January 2021,

   [RFC8831]  Jesup, R., Loreto, S., and M. Tüxen, "WebRTC Data
              Channels", RFC 8831, DOI 10.17487/RFC8831, January 2021,

   [RFC8834]  Perkins, C., Westerlund, M., and J. Ott, "Media Transport
              and Use of RTP in WebRTC", RFC 8834, DOI 10.17487/RFC8834,
              January 2021, <>.

   [RFC8835]  Alvestrand, H., "Transports for WebRTC", RFC 8835,
              DOI 10.17487/RFC8835, January 2021,

9.2.  Informative References

   [G.1010]   ITU-T, "End-user multimedia QoS categories", ITU-T
              Recommendation G.1010, November 2001,

   [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,

   [RFC2597]  Heinanen, J., Baker, F., Weiss, W., and J. Wroclawski,
              "Assured Forwarding PHB Group", RFC 2597,
              DOI 10.17487/RFC2597, June 1999,

   [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,

   [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, <>.

   [RFC8699]  Islam, S., Welzl, M., and S. Gjessing, "Coupled Congestion
              Control for RTP Media", RFC 8699, DOI 10.17487/RFC8699,
              January 2020, <>.


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


   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


   Subha Dhesikan


   Cullen Jennings
   Cisco Systems


   Dan Druta