DSCP and other packet markings for RTCWeb QoS
draft-ietf-tsvwg-rtcweb-qos-02

Abstract

Many networks, such as service provider and enterprise networks, can provide per packet treatments based on Differentiated Services Code Points (DSCP) 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 December 25, 2014.

Copyright Notice

Copyright (c) 2014 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. Document History
• 11. References
• 11.1. Normative References
• 11.2. 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 draft proposes how WebRTC applications can mark packets. This draft does not contradict or redefine any advice from previous IETF RFCs but simply provides a simple set of recommendations for implementers based on the previous RFCs.

There are some environments where priority markings frequently help. These include:

1. Private networks (Wide Area).

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 WiFi network, marking may help.

Traditionally DSCP values have been thought of as being site specific, with each site selecting its own code points for each QoS level. However in the RTCWeb use cases, the browsers need to set them to something when there is no site specific information. Browsers, in this document is used synonmously with "interactive User Agent" as defined in teh HTML specification, [W3C.WD-html-20110525]. This document describes a reasonable default set of DSCP code point values drawn from existing RFCs and common usage. These code points are solely defaults. Future drafts may define mechanisms for site specific mappings to override the values provided in this draft.

This draft defines some inputs that the browser in an WebRTC application can look at to determine 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 specification does not change or override the advice in any other standards about setting packet markings. It simply provides a summary of them and provides the context of how they relate in the RTCWeb context. In some cases, such as DSCP where the normative RFC leaves open multiple options from which to choose, this clarifies which choice should be used in the RTCWeb 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 the network edge through policy to any other DSCP value, including best effort. The mitigation for such action is through an authorization mechanism. Such authorization mechanism is outside the scope of this document.

3. Terminology

The key words "MUST", "MUST NOT", "SHOULD", "SHOULD NOT", and "MAY" in this document are to be interpreted as described in [RFC2119].

4. Inputs

The below uses the concept of a media flow, however these are commonly not equivalent to a transport flow, i.e. as defined by a 5-tuple (source address, destination address, source port, destination port, and protocol). Instead each media flow 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 priorities. 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.
• Priority: Another input is the relative treatment of the flow within that data type. Many applications have multiple media flows of the same data type and often some are more important than others. Likewise, in a video conference where the flows in the conference is of the same data type but contains different media types, the flow for audio 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.

When it comes to data transmission, a media (data) flow is the SCTP stream under a common congestion control (currently within the same SCTP association).

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

5. DSCP Mappings

Below is a table of DSCP markings for each data type of interest to RTCWeb. These DSCP values for each data type listed are a reasonable default set 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].