DSCP and other packet markings for WebRTC QoS
draft-ietf-tsvwg-rtcweb-qos-07

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.

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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.

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Table of Contents

• 1. Introduction
• 2. Relation to Other Standards
• 3. Terminology
• 4. Inputs
• 5. DSCP Mappings
• 6. Security Considerations
• 7. IANA Considerations
• 8. Downward References
• 9. Acknowledgements
• 10. Dedication
• 11. Document History
• 12. References
• 12.1. Normative References
• 12.2. Informative References
• Authors' Addresses

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.

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.

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. 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:

• 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.
• 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.

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.