--- 1/draft-ietf-rmcat-cc-requirements-04.txt 2014-07-04 12:14:29.771740530 -0700 +++ 2/draft-ietf-rmcat-cc-requirements-05.txt 2014-07-04 12:14:29.795741103 -0700 @@ -1,84 +1,87 @@ Network Working Group R. Jesup Internet-Draft Mozilla -Intended status: Informational April 19, 2014 -Expires: October 21, 2014 +Intended status: Informational July 4, 2014 +Expires: January 5, 2015 Congestion Control Requirements For RMCAT - draft-ietf-rmcat-cc-requirements-04 + draft-ietf-rmcat-cc-requirements-05 Abstract Congestion control is needed for all data transported across the Internet, in order to promote fair usage and prevent congestion collapse. The requirements for interactive, point-to-point real time multimedia, which needs low-delay, semi-reliable data delivery, are different from the requirements for bulk transfer like FTP or bursty transfers like Web pages. Due to an increasing amount of RTP-based real-time media traffic on the Internet (e.g. with the introduction of WebRTC[I-D.ietf-rtcweb-overview]), it is especially important to ensure that this kind of traffic is congestion controlled. - This document attempts to describe a set of requirements that can be - used to evaluate other congestion control mechanisms in order to - figure out their fitness for this purpose, and in particular to - provide a set of possible requirements for proposals coming out of - the RMCAT Working Group. + This document describes a set of requirements that can be used to + evaluate other congestion control mechanisms in order to figure out + their fitness for this purpose, and in particular to provide a set of + possible requirements for realtime media congestion avoidance + technique. Requirements Language 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 RFC 2119 [RFC2119]. + The terms are presented in many cases using lowercase for + readability. 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 October 21, 2014. + + This Internet-Draft will expire on January 5, 2015. Copyright Notice Copyright (c) 2014 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 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. Requirements . . . . . . . . . . . . . . . . . . . . . . . . 3 - 3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 7 + 3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 8 4. Security Considerations . . . . . . . . . . . . . . . . . . . 8 5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 8 6. References . . . . . . . . . . . . . . . . . . . . . . . . . 8 6.1. Normative References . . . . . . . . . . . . . . . . . . 8 - 6.2. Informative References . . . . . . . . . . . . . . . . . 8 + 6.2. Informative References . . . . . . . . . . . . . . . . . 9 Author's Address . . . . . . . . . . . . . . . . . . . . . . . . 9 1. Introduction Most of today's TCP congestion control schemes were developed with a focus on an use of the Internet for reliable bulk transfer of non- time-critical data, such as transfer of large files. They have also been used successfully to govern the reliable transfer of smaller chunks of data in as short a time as possible, such as when fetching Web pages. @@ -130,21 +133,21 @@ A. It should provide this as-low-as-possible-delay transit even when faced with intermediate bottlenecks and competing flows. Competing flows may limit what's possible to achieve. B. It should handle routing changes which may alter or remove bottlenecks or change the bandwidth available, and react quickly, especially if there is a reduction in available bandwidth or increase in bottleneck delay. - C. It should handle interface changes (WiFi to 3G data, etc) + C. It should handle interface changes (WLAN to 3G data, etc) which may radically change the bandwidth available or bottlenecks, and react quickly, especially if there is a reduction in available bandwidth or increase in bottleneck delay. It is assumed that an interface change can generate a notification to the algorithm. D. The offered load may be less than the available bandwidth at any given moment, and may vary dramatically over time, including dropping to no load and then resuming a high load, such as in a mute operation. The reaction time between a @@ -157,43 +160,44 @@ bottleneck router or link, but also clear quickly, and should recover quickly when the burst ends. This is inherently at odds with the need to react quickly-enough to avoid queue buildup. F. Similarly periodic bursty flows such as MPEG DASH [MPEG_DASH] or proprietary media streaming algorithms may compete in bursts with the algorithm, and may not be adaptive within a burst. They are often layered on top of TCP. The algorithm must avoid too much delay buildup during - those bursts, and quickly recover. Note that this traffic - may on an access link, or may cause a shift in the location - of the bottleneck for the duration of the burst. + those bursts, and quickly recover. Note that this competing + traffic may on a shared access link, or the traffic burst + may cause a shift in the location of the bottleneck for the + duration of the burst. 2. The algorithm must be fair to other flows, both realtime flows (such as other instances of itself), and TCP flows, both long- lived and bursts such as the traffic generated by a typical web browsing session. Note that 'fair' is a rather hard-to-define - term. It should be self-fair with itself, giving roughly equal + term. It should be fair with itself, giving roughly equal bandwidth to multiple flows with similar RTTs, and if possible to multiple flows with different RTTs. A. Existing flows at a bottleneck must also be fair to new flows to that bottleneck, and must allow new flows to ramp up to a useful share of the bottleneck bandwidth quickly. Note that relative RTTs may affect the rate new flows can ramp up to a reasonable share. 3. The algorithm should not starve competing TCP flows, and should as best as possible avoid starvation by TCP flows. A. An algorithm may be more successful at avoiding starvation - from short-lived TCP long-lived/saturating TCP flows. + from short-lived TCP than long-lived/saturating TCP flows. B. In order to avoid starvation other goals may need to be compromised (such as delay). 4. The algorithm should quickly adapt to initial network conditions at the start of a flow. This should occur both if the initial bandwidth is above or below the bottleneck bandwidth. A. The startup adaptation may be faster than adaptation later in a flow. It should allow for both slow-start operation @@ -209,21 +213,22 @@ B. Alternative methods to help startup like probing during setup with dummy data may be useful in some applications; in some cases there will be a considerable gap in time between flow creation and the initial flow of data. C. A flow may need to change adaptation rates due to network conditions or changes in the provided flows (such as un- muting or sending data after a gap). 5. It should be stable if the RTP streams are halted or - discontinuous (VAD/DTX). + discontinuous (Voice Activity Detection/Discontinuous + Transmission). A. After a resumption of RTP data it may adapt more quickly (similar to the start of a flow), and previous bandwidth estimates may need to be aged or thrown away. 6. The algorithm should where possible merge information across multiple RTP streams between the same endpoints, whether or not they're multiplexed on the same ports, in order to allow congestion control of the set of streams together instead of as multiple independent streams. This allows better overall @@ -268,30 +273,31 @@ they should be utilized if possible. 8. Since the assumption here is a set of RTP streams, the backchannel typically should be done via RTCP; one alternative would be to include it instead in a reverse RTP channel using header extensions. A. In order to react sufficiently quickly when using RTCP for a backchannel, an RTP profile such as RTP/AVPF [RFC4585] or RTP/SAVPF [RFC5124] that allows sufficiently frequent - feedback MUST be used. + feedback must be used. B. Note that in some cases, backchannel messages may be delayed until the RTCP channel can be allocated enough bandwidth, even under AVPF rules. This may also imply negotiating a higher maximum percentage for RTCP data or allowing RMCAT solutions to violate or modify the rules specified for AVPF. C. Bandwidth for the feedback messages should be minimized - (such as via RFC 5506 [RFC5506]to allow RTCP without SR/RR) + (such as via RFC 5506 [RFC5506]to allow RTCP without Sender + Reports/Receiver Reports) D. Header extensions would avoid the RTCP timing rules issues, and allow the application to allocate bandwidth as needed for the congestion algorithm. E. Backchannel data should be minimized to avoid taking too much reverse-channel bandwidth (since this will often be used in a bidirectional set of flows). In areas of stability, backchannel data may be sent more infrequently so long as algorithm stability and fairness are maintained. @@ -299,21 +305,21 @@ equilibrium after a change, backchannel feedback may be more frequent and use more reverse-channel bandwidth. This is an area with considerable flexibility of design, and different approaches to backchannel messages and frequency are expected to be evaluated. 9. Flows managed by this algorithm and flows competing against at a bottleneck may have different DSCP[RFC5865] markings depending on the type of traffic, or may be subject to flow-based QoS. A particular bottleneck or section of the network path may or may - not honor DSCP markings. The algorithm SHOULD attempt to + not honor DSCP markings. The algorithm should attempt to leverage DSCP markings when they're available. A. In WebRTC, a division of packets into 4 classes is envisioned in order of priority: faster-than-audio, audio, video, best-effort, and bulk-transfer. Typically the flows managed by this algorithm would be audio or video in that heirarchy, and feedback flows would be faster-than-audio. 10. The algorithm should sense the unexpected lack of backchannel information as a possible indication of a channel overuse @@ -338,61 +344,61 @@ the flow can fake congestion signals, unless they are passed on a tamper-proof path. Since some possible algorithms depend on the timing of packet arrival, even a traditional protected channel does not fully mitigate such attacks. An attack that reduces bandwidth is not necessarily significant, since an on-path attacker could break the connection by discarding all packets. Attacks that increase the percieved available bandwidth are concievable, and need to be evaluated. - Algorithm designers SHOULD consider the possibility of malicious on- + Algorithm designers should consider the possibility of malicious on- path attackers. 5. Acknowledgements This document is the result of discussions in various fora of the WebRTC effort, in particular on the rtp-congestion@alvestrand.no mailing list. Many people contributed their thoughts to this. 6. References 6.1. Normative References [I-D.ietf-rtcweb-overview] - Alvestrand, H., "Overview: Real Time Protocols for Brower- - based Applications", draft-ietf-rtcweb-overview-09 (work - in progress), February 2014. + Alvestrand, H., "Overview: Real Time Protocols for + Browser-based Applications", draft-ietf-rtcweb-overview-10 + (work in progress), June 2014. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. [RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey, "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 2006. [RFC5124] Ott, J. and E. Carrara, "Extended Secure RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/SAVPF)", RFC 5124, February 2008. 6.2. Informative References [I-D.ietf-rtcweb-data-channel] Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data - Channels", draft-ietf-rtcweb-data-channel-08 (work in - progress), April 2014. + Channels", draft-ietf-rtcweb-data-channel-10 (work in + progress), June 2014. [I-D.welzl-rmcat-coupled-cc] Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion - control for RTP media", draft-welzl-rmcat-coupled-cc-02 - (work in progress), October 2013. + control for RTP media", draft-welzl-rmcat-coupled-cc-03 + (work in progress), May 2014. [MPEG_DASH] "Dynamic adaptive streaming over HTTP (DASH) -- Part 1: Media presentation description and segment formats", April 2012. [RFC3168] Ramakrishnan, K., Floyd, S., and D. Black, "The Addition of Explicit Congestion Notification (ECN) to IP", RFC 3168, September 2001.