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Versions: (draft-dhesikan-tsvwg-rtcweb-qos) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 Draft is active
In: MissingRef
Network Working Group                                        S. Dhesikan
Internet-Draft                                               C. Jennings
Intended status: Standards Track                           Cisco Systems
Expires: June 20, 2016                                     D. Druta, Ed.
                                                                    AT&T
                                                                P. Jones
                                                           Cisco Systems
                                                       December 18, 2015


             DSCP and other packet markings for WebRTC QoS
                     draft-ietf-tsvwg-rtcweb-qos-06

Abstract

   Many networks, such as service provider and enterprise networks, can
   provide treatment for individual packets based on Differentiated
   Services Code Point (DSCP) values on a per-hop basis.  This document
   provides the recommended DSCP values for browsers to use for various
   classes of 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 June 20, 2016.

Copyright Notice

   Copyright (c) 2015 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



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   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
   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  . . . . . . . . . . . . . . . . . . . . . . . . .   8
   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 Points (DSCP) [RFC2474] style packet
   marking can help provide QoS in some environments.  There are many
   use cases where such marking does not help, but it seldom makes
   things worse if packets are marked appropriately.  In other words, if
   too many packets, say all audio or all audio and video, are marked
   for a given network condition then it can prevent desirable results.
   Either too much other traffic will be starved, or there is not enough
   capacity for the preferentially marked packets (i.e., audio and/or
   video).

   This specification proposes how WebRTC applications can mark packets.
   This specification does not contradict or redefine any advice from
   previous IETF RFCs, but merely provides a simple set of
   recommendations for implementers based on the previous RFCs

   There are some environments where DSCP markings frequently help.
   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.



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   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
   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 some inputs that the browser in a WebRTC
   application can consider to aid in determining how to set the various
   packet markings and defines the mapping from abstract QoS policies
   (data type, priority level) to those packet markings.

2.  Relation to Other Standards

   This document exists as a complement to [I-D.ietf-dart-dscp-rtp],
   which describes the interaction between DSCP and real-time
   communications.  It 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
   [I-D.ietf-dart-dscp-rtp] that should be followed when 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 [I-D.ietf-dart-dscp-rtp] 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.  It simply selects a
   subset of DSCP values that is relevant in the WebRTC context.  This
   document also specifies the inputs that are needed by the browser to
   provide to the media engine.



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   The DSCP value set by the endpoint is not always trusted by the
   network.  Therefore, 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.  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

   The below uses the concept of a media flow, however this is usually
   not equivalent to a transport-layer flow defined by a 5-tuple (source
   address, destination address, source port, destination port, and
   protocol).  Instead each media flow, such as an RTP stream
   [I-D.ietf-rtcweb-rtp-usage] or SCTP association carrying data channel
   packets [I-D.ietf-rtcweb-data-channel], contains all the packets
   associated with an independent media entity within one 5-tuple.
   Specifically, a media flow is the transmitted packets for an RTP
   session or an SCTP association.  There may be multiple media flows
   within the same 5-tuple.  These media flows might consist of
   different media types and have different levels of importance to the
   application and, therefore, each potentially marked using different
   DSCP values than for another media flow multiplexed over the same
   transport-layer flow.  The following are the inputs that the browser
   provides to the media engine:

   o  Data 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
      the flow within that data type.  Many applications have multiple
      media flows of the same data 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 media flow is high, medium,
      low or very low importance to the application.

   [I-D.ietf-rtcweb-transports] defines in more detail what an
   individual media flow is within the WebRTC context.




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   As an example of different media flows that might be multiplexed over
   the same transport-layer flow, packets related to one RTP stream
   (e.g., an audio flow) carried over UDP might be one media flow,
   packets related to a second RTP stream (e.g., presentation video)
   carried over UDP might be a second media flow, and finally data
   channel packets carried via SCTP over DTLS might be third media flow.

5.  DSCP Mappings

   Below is a table of DSCP markings for each data type of interest to
   WebRTC.  These DSCP values for each data type listed are a reasonable
   subset of code point values taken from [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.

   +------------------------+-------+------+-------------+-------------+
   |       Data 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 columns "very low", "low", "Medium" and "high" signify the
   relative importance of the media flow within the application and is
   an input that the browser receives to assist it in selecting the DSCP
   value.  These are referred to as application priority in this
   document.  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



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   what is intended.  It is common for CS1 to be treated the same as DF
   so anyone using CS1 cannot assume that CS1 will be treated
   differently than DF.  Implementers should also note that the excess
   EF traffic is dropped.  This could mean that a packet marked as EF
   may not get through as opposed to a packet marked with a different
   DSCP value.

   The browser SHOULD first select the data type of the media flow.
   Within the data type, the relative importance of the media flow
   SHOULD be used to select the appropriate DSCP value.

   The combination of data type and application priority provides
   specificity and helps in selecting the right DSCP value for the media
   flow.  In some cases, the different drop precedence values provides
   additional granularity in classifying packets within a media flow.
   For example, in a video conference, the video media 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.

   All packets within a media flow SHOULD have the same application
   priority.  In some cases, the selected cell may have multiple DSCP
   values, such as AF41 and AF42.  These offer different drop
   precedences.  With the exception of data channel traffic, one may
   select different drop precedences for the different packets in the
   same media flow.  Therefore, all packets in the media flow SHOULD be
   marked with the same application priority, but can have different
   drop precedences.

   For reasons discussed in Section 6 of [I-D.ietf-dart-dscp-rtp], if
   multiple media flows are multiplexed using a reliable transport
   (e.g., TCP) then all of the packets for all media 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.





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   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
   media 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 QoS domain that has no support for the above
   defined data types/application priority (service class), then the
   network node at the edge will remark the DSCP value based on
   policies.  This could result in the media flow not getting the
   network treatment it expects based on the original DSCP value in the
   packet.  Subsequently, if the packet enters a QoS domain that
   supports a larger number of service classes, 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, there are no guarantees or promised level of service with
   the use of DSCP.  The service provided to a packet is dependent upon
   the network design along the path, as well as the congestion levels
   at every hop.

6.  Security Considerations

   This specification does not add any additional security implication
   other than the normal application use of DSCP.  For security
   implications on use of DSCP, please refer to Section 6 of RFC 4594.
   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
   help.




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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.

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.  This document is now being
   submitted as individual submission to the TSVWG with the hope that WG
   will select it as a WG draft and move it forward to an RFC.

12.  References

12.1.  Normative References

   [I-D.ietf-dart-dscp-rtp]
              Black, D. and P. Jones, "Differentiated Services
              (DiffServ) and Real-time Communication", draft-ietf-dart-
              dscp-rtp-10 (work in progress), November 2014.

   [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, March 1997.



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

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, December
              1998.

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

   [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, 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, <http://www.rfc-editor.org/info/rfc3550>.

   [W3C.REC-html5-20141028]
              Hickson, I., Berjon, R., Faulkner, S., Leithead, T.,
              Navara, E., O&#039;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

   Subha Dhesikan
   Cisco Systems

   Email: sdhesika@cisco.com


   Cullen Jennings
   Cisco Systems

   Email: fluffy@cisco.com








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   Dan Druta (editor)
   AT&T

   Email: dd5826@att.com


   Paul E. Jones
   Cisco Systems

   Email: paulej@packetizer.com









































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