Network Working Group P. Jones
Internet-Draft S. Dhesikan
Intended status: Standards Track C. Jennings
Expires: July 30, 2016 Cisco Systems
D. Druta
AT&T
January 27, 2016
DSCP and other packet markings for WebRTC QoS
draft-ietf-tsvwg-rtcweb-qos-12
Abstract
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.
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
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
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 July 30, 2016.
Copyright Notice
Copyright (c) 2016 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
Jones, et al. Expires July 30, 2016 [Page 1]
Internet-Draft WebRTC QoS January 2016
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
2. Relation to Other Standards . . . . . . . . . . . . . . . . . 3
3. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
4. Inputs . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
5. DSCP Mappings . . . . . . . . . . . . . . . . . . . . . . . . 5
6. Security Considerations . . . . . . . . . . . . . . . . . . . 7
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7
8. Downward References . . . . . . . . . . . . . . . . . . . . . 7
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 7
10. Dedication . . . . . . . . . . . . . . . . . . . . . . . . . 7
11. Document History . . . . . . . . . . . . . . . . . . . . . . 8
12. References . . . . . . . . . . . . . . . . . . . . . . . . . 8
12.1. Normative References . . . . . . . . . . . . . . . . . . 8
12.2. Informative References . . . . . . . . . . . . . . . . . 9
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 9
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
existing IETF RFCs. The contents of this specification are intended
to be a simple set of implementation recommendations based on the
previous RFCs.
There are many use cases where such marking does not help, but it
seldom makes things worse if packets are marked appropriately. There
are some environments where DSCP markings frequently help, though.
These include:
1. Private, wide-area networks.
2. 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.
3. Wireless Networks. If the congested link is a local wireless
network, marking may help.
Traditionally, DSCP values have been thought of as being site
specific, with each site selecting its own code points for
Jones, et al. Expires July 30, 2016 [Page 2]
Internet-Draft WebRTC QoS January 2016
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,
[W3C.REC-html5-20141028]. 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 solely defaults.
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. Relation to Other Standards
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 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 [I-D.ietf-rtcweb-data-channel] traffic over the same
5-tuple as RTP streams, which would also be marked as per that table.
The guidance in [RFC7657] 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 standards 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. The
mitigation for such action is through an authorization mechanism.
Such authorization mechanism is outside the scope of this document.
Jones, et al. Expires July 30, 2016 [Page 3]
Internet-Draft WebRTC QoS January 2016
There is benefit in marking traffic even if it only benefits the
first few hops.
3. Terminology
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 [RFC2119].
4. Inputs
WebRTC applications send and receive two types of flows of
significance to this document:
o media flows which are RTP streams [I-D.ietf-rtcweb-rtp-usage]
o data flows which are data channels [I-D.ietf-rtcweb-data-channel]
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 Section 5.)
The following are the inputs provided by the WebRTC application to
the browser:
o Flow Type: The browser provides this input as it knows if the flow
is audio, interactive video with or without audio, non-interactive
video with or without audio, or data.
o 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.
Jones, et al. Expires July 30, 2016 [Page 4]
Internet-Draft WebRTC QoS January 2016
[I-D.ietf-rtcweb-transports] defines in more detail what an
individual flow is within the WebRTC context and priorities for media
and data flows.
5. DSCP Mappings
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 [RFC4594]. A
web browser SHOULD use these values to mark the appropriate media
packets. More information on EF can be found in [RFC3246]. More
information on AF can be found in [RFC2597]. DF is default
forwarding which provides the basic best effort service [RFC2474].
+------------------------+-------+------+-------------+-------------+
| Flow Type | Very | Low | Medium | High |
| | Low | | | |
+------------------------+-------+------+-------------+-------------+
| Audio | CS1 | DF | EF (46) | EF (46) |
| | (8) | (0) | | |
| | | | | |
| Interactive Video with | CS1 | DF | AF42, AF43 | AF41, AF42 |
| or without audio | (8) | (0) | (36, 38) | (34, 36) |
| | | | | |
| Non-Interactive Video | CS1 | DF | AF32, AF33 | AF31, AF32 |
| with or without audio | (8) | (0) | (28, 30) | (26, 28) |
| | | | | |
| Data | CS1 | DF | AF11 | AF21 |
| | (8) | (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 it in selecting the DSCP value. Application priority does not
refer to priority in the network transport.
The above table assumes that packets marked with CS1 are treated as
"less than best effort". 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 [RFC7657]. Implementers should also note
that excess EF traffic is dropped. This could mean that a packet
Jones, et al. Expires July 30, 2016 [Page 5]
Internet-Draft WebRTC QoS January 2016
marked as EF may not get through as opposed to a packet marked with a
different DSCP value.
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.
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. If so, either AF42 or AF43 may be selected.
If the I-frames in the stream are more important than the P-frames,
then the I-frames can be marked with AF42 and the P-frames marked
with AF43.
It is worth noting that the application priority is utilized by the
SCTP scheduler for data channel traffic per
[I-D.ietf-rtcweb-data-channel].
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. 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.
Jones, et al. Expires July 30, 2016 [Page 6]
Internet-Draft WebRTC QoS January 2016
If a packet enters a network that has no support for a flow type-
application priority combination specified in Table 1 (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. 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
This specification does not add any additional security implication
other than the normal application use of DSCP not already addressed
by the following specifications. For security implications on use of
DSCP, please refer to Section 7 of [RFC7657] and Section 6 of
[RFC4594]. Please also see [I-D.ietf-rtcweb-security] as an
additional reference.
7. IANA Considerations
This specification does not require any actions from IANA.
8. Downward References
This specification contains a downwards reference to [RFC4594].
However, the parts of that RFC used by this specification are
sufficiently stable for this downward reference.
9. Acknowledgements
Thanks to David Black, Magnus Westerland, Paolo Severini, Jim
Hasselbrook, Joe Marcus, Erik Nordmark, and Michael Tuexen for their
invaluable input.
10. 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 being one of its primary authors. The IETF
global community mourns his loss and he will be missed dearly.
Jones, et al. Expires July 30, 2016 [Page 7]
Internet-Draft WebRTC QoS January 2016
11. Document History
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.
12. References
12.1. Normative References
[I-D.ietf-rtcweb-data-channel]
Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
Channels", draft-ietf-rtcweb-data-channel-13 (work in
progress), January 2015.
[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-25 (work in progress), June
2015.
[I-D.ietf-rtcweb-security]
Rescorla, E., "Security Considerations for WebRTC", draft-
ietf-rtcweb-security-08 (work in progress), February 2015.
[I-D.ietf-rtcweb-transports]
Alvestrand, H., "Transports for WebRTC", draft-ietf-
rtcweb-transports-10 (work in progress), October 2015.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/
RFC2119, March 1997,
<http://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,
<http://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,
<http://www.rfc-editor.org/info/rfc7657>.
Jones, et al. Expires July 30, 2016 [Page 8]
Internet-Draft WebRTC QoS January 2016
12.2. Informative References
[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,
<http://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,
<http://www.rfc-editor.org/info/rfc2597>.
[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, DOI 10.17487/RFC3246, March 2002,
<http://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, <http://www.rfc-editor.org/info/rfc3550>.
[W3C.REC-html5-20141028]
Hickson, I., Berjon, R., Faulkner, S., Leithead, T.,
Navara, E., O'Connor, E., and S. Pfeiffer, "HTML5",
World Wide Web Consortium Recommendation REC-
html5-20141028, October 2014,
<http://www.w3.org/TR/2014/REC-html5-20141028>.
Authors' Addresses
Paul E. Jones
Cisco Systems
Email: paulej@packetizer.com
Subha Dhesikan
Cisco Systems
Email: sdhesika@cisco.com
Jones, et al. Expires July 30, 2016 [Page 9]
Internet-Draft WebRTC QoS January 2016
Cullen Jennings
Cisco Systems
Email: fluffy@cisco.com
Dan Druta
AT&T
Email: dd5826@att.com
Jones, et al. Expires July 30, 2016 [Page 10]