Network Working Group                                           P. JonesInternet-Draft                                               S. DhesikanIntended status: Standards Track                             C. JenningsExpires: February 20, 2017                                 Cisco Systems                                                                D. Druta                                                                    AT&T                                                         August 19, 2016                  DSCP Packet Markings for WebRTC QoS                     draft-ietf-tsvwg-rtcweb-qos-18Abstract   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 February 20, 2017.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 mustJones, et al.           Expires February 20, 2017               [Page 1]Internet-Draft                 WebRTC QoS                    August 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.  Terminology . . . . . . . . . . . . . . . . . . . . . . . . .   3   3.  Relation to Other Specifications  . . . . . . . . . . . . . .   3   4.  Inputs  . . . . . . . . . . . . . . . . . . . . . . . . . . .   4   5.  DSCP Mappings . . . . . . . . . . . . . . . . . . . . . . . .   5   6.  Security Considerations . . . . . . . . . . . . . . . . . . .   8   7.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   8   8.  Downward References . . . . . . . . . . . . . . . . . . . . .   9   9.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   9   10. Dedication  . . . . . . . . . . . . . . . . . . . . . . . . .   9   11. Document History  . . . . . . . . . . . . . . . . . . . . . .   9   12. References  . . . . . . . . . . . . . . . . . . . . . . . . .   9     12.1.  Normative References . . . . . . . . . . . . . . . . . .   9     12.2.  Informative References . . . . . . . . . . . . . . . . .  10   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  111.  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   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:   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, 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.   There are cases where these DSCP markings do not help, but, aside   from possible priority inversion for "less than best effort traffic"Jones, et al.           Expires February 20, 2017               [Page 2]Internet-Draft                 WebRTC QoS                    August 2016   (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.  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.2.  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].   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 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 theJones, et al.           Expires February 20, 2017               [Page 3]Internet-Draft                 WebRTC QoS                    August 2016   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   [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   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 application provides this input because it knows if      the flow is audio, interactive video [RFC4594] [G.1010] with or      without audio, or data.Jones, et al.           Expires February 20, 2017               [Page 4]Internet-Draft                 WebRTC QoS                    August 2016   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.   [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.   Currently in WebRTC, media sent over RTP is assumed to be interactive   [I-D.ietf-rtcweb-transports] and browser APIs do not exist to allow   an application to 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 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].   WebRTC use of multiple DSCP values may encounter network blocking of   packets with certain DSCP values.  See section 4.2 of   [I-D.ietf-rtcweb-transports] for further discussion, including how   WebRTC implementations establish and maintain connectivity when such   blocking is encountered.Jones, et al.           Expires February 20, 2017               [Page 5]Internet-Draft                 WebRTC QoS                    August 2016   +------------------------+-------+------+-------------+-------------+   |       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 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   [RFC3662].  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].  However, it is also possible per   [RFC2474] 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.Jones, et al.           Expires February 20, 2017               [Page 6]Internet-Draft                 WebRTC QoS                    August 2016   Currently, all WebRTC video is assumed to be interactive   [I-D.ietf-rtcweb-transports], 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   [I-D.ietf-rmcat-coupled-cc] and 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.Jones, et al.           Expires February 20, 2017               [Page 7]Internet-Draft                 WebRTC QoS                    August 2016   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 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 [I-D.ietf-rtcweb-security] as an additional reference.7.  IANA Considerations   This specification does not require any actions from IANA.Jones, et al.           Expires February 20, 2017               [Page 8]Internet-Draft                 WebRTC QoS                    August 20168.  Downward References   This specification contains a downwards reference to [RFC4594] and   [RFC7657].  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.9.  Acknowledgements   Thanks to David Black, Magnus Westerlund, Paolo Severini, Jim   Hasselbrook, Joe Marcus, Erik Nordmark, Michael Tuexen, and Brian   Carpenter 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 serving initially as one of the 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.12.  References12.1.  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, D., Westerlund, M., and J. Ott, "Web Real-Time              Communication (WebRTC): Media Transport and Use of RTP",              draft-ietf-rtcweb-rtp-usage-26 (work in progress), March              2016.Jones, et al.           Expires February 20, 2017               [Page 9]Internet-Draft                 WebRTC QoS                    August 2016   [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-15 (work in progress), August 2016.   [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>.   [RFC7742]  Roach, A., "WebRTC Video Processing and Codec              Requirements", RFC 7742, DOI 10.17487/RFC7742, March 2016,              <http://www.rfc-editor.org/info/rfc7742>.12.2.  Informative References   [G.1010]   International Telecommunications Union, "End-user              multimedia QoS categories", Recommendation ITU-T G.1010,              November 2001.   [I-D.ietf-rmcat-coupled-cc]              Islam, S., Welzl, M., and S. Gjessing, "Coupled congestion              control for RTP media", draft-ietf-rmcat-coupled-cc-03              (work in progress), July 2016.   [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>.Jones, et al.           Expires February 20, 2017              [Page 10]Internet-Draft                 WebRTC QoS                    August 2016   [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>.   [RFC3662]  Bless, R., Nichols, K., and K. Wehrle, "A Lower Effort              Per-Domain Behavior (PDB) for Differentiated Services",              RFC 3662, DOI 10.17487/RFC3662, December 2003,              <http://www.rfc-editor.org/info/rfc3662>.Authors' Addresses   Paul E. Jones   Cisco Systems   Email: paulej@packetizer.com   Subha Dhesikan   Cisco Systems   Email: sdhesika@cisco.com   Cullen Jennings   Cisco Systems   Email: fluffy@cisco.com   Dan Druta   AT&T   Email: dd5826@att.comJones, et al.           Expires February 20, 2017              [Page 11]