Internet Engineering Task Force (IETF) P. Jones
Request for Comments: 8837 Cisco Systems
Category: Standards Track S. Dhesikan
ISSN: 2070-1721 Individual
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
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Table of Contents
3. Relation to Other Specifications
5. DSCP Mappings
6. Security Considerations
7. IANA Considerations
8. Downward References
9.1. Normative References
9.2. Informative References
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.
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
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 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
The following are the inputs provided by the WebRTC application to
* 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
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
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
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
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
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.
9.1. 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, <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,
[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, <https://www.rfc-editor.org/info/rfc8834>.
[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, <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>.
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
Paul E. Jones