DSCP Packet Markings for WebRTC QoS
Cisco Systems
paulej@packetizer.com
Cisco Systems
sdhesika@cisco.com
Cisco Systems
fluffy@cisco.com
AT&T
dd5826@att.com
Many networks, such as service provider and enterprise 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 WebRTC traffic.
Differentiated Services Code Point (DSCP)
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 IETF RFCs. The contents of this
specification are intended to be a simple set of implementation
recommendations based on the previous RFCs.
Networks where these DSCP markings are beneficial (likely to
improve QoS for WebRTC traffic) include:
Private, wide-area networks. Network administrators have
control over remarking packets and treatment of packets.
Residential Networks. If the congested link is the
broadband uplink in a cable or DSL scenario, often
residential routers/NAT support preferential treatment based
on DSCP.
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. In this document,
"browsers" is used synonymously with "Interactive User Agent" as
defined in the HTML specification, . 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.
This document is a complement to , which
describes the interaction between DSCP and real-time
communications. That RFC covers the implications of using
various DSCP values, particularly focusing on Real-time
Transport Protocol (RTP) streams that
are multiplexed onto a single transport-layer flow.
There are a number of guidelines specified in
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
. A WebRTC application might also
multiplex data channel
traffic over the
same 5-tuple as RTP streams, which would also be marked as per
that table. The guidance in says that
all data channel traffic would be marked with a single value
that is typically different than 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 IETF 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 default forwarding (DF) value to provide basic
best effort service. Even so, there is benefit in marking
traffic even if it only benefits the first few hops. The
implications are discussed in Secton 3.2 of
. Further, a mitigation for such action
is through an authorization mechanism. Such an authorization
mechanism is outside the scope of this document.
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 .
WebRTC applications send and receive two types of flows of
significance to this document:
media flows which are RTP streams
data flows which are data channels
Each of the RTP streams and distinct data channels consists 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 SCTP association
as outlined in .)
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
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
often some flows are 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 high, medium, low or
very low importance to the application.
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 and
browser APIs do not exist to allow an application to to
differentiate between interactive and non-interactive video.
The DSCP values for each flow type of interest to WebRTC based
on application priority are shown in the following table. These
values are based on the framework and recommended values in
. A web browser SHOULD use these values
to mark the appropriate media packets. More information on EF
can be found in . More information on
AF can be found in . DF is default
forwarding which provides the basic best effort service
.
Flow Type
Very Low
Low
Medium
High
Audio
CS1 (8)
DF (0)
EF (46)
EF (46)
Interactive Video with or without Audio
CS1 (8)
DF (0)
AF42, AF43 (36, 38)
AF41, AF42 (34, 36)
Non-Interactive Video with or without Audio
CS1 (8)
DF (0)
AF32, AF33 (28, 30)
AF31, AF32 (26, 28)
Data
CS1 (8)
DF (0)
AF11
AF21
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 CS1 are treated
as "less than best effort", such as the LE behavior described in
. However, the treatment of CS1 is
implementation dependent. If an implementation treats CS1 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 CS1 to be
treated the same as DF, so applications and browsers using CS1
cannot assume that CS1 will be treated differently than DF
. However, it is also possible per
for CS1 traffic to be given better
treatment than DF, thus caution should be exercised when
electing to use CS1. This is one of the cases where marking
packets using these recommendations can make things worse.
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
, 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 provides 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
and the SCTP
scheduler for data channel traffic per
.
For reasons discussed in Section 6 of
, 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.
Frequent changes in the DSCP value used for an SCTP association
are discouraged, though, 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 DCSP 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
(above), 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.
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
and Section 6 of . Please also see
as an additional reference.
This specification does not require any actions from IANA.
This specification contains a downwards reference to
and . However,
the parts of the former RFC used by this specification are
sufficiently stable for this downward reference. 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.
Thanks to David Black, Magnus Westerlund, Paolo Severini, Jim
Hasselbrook, Joe Marcus, Erik Nordmark, Michael Tuexen, 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.
Note to RFC Editor: Please remove this section.
This document was originally an individual submission in RTCWeb WG.
The RTCWeb working group selected it to be become a WG document.
Later the transport ADs requested that this be moved to the TSVWG WG
as that seemed to be a better match.
End-user multimedia QoS categories
International Telecommunications Union