--- 1/draft-ietf-rmcat-cc-requirements-02.txt 2014-03-31 14:14:35.569967024 -0700 +++ 2/draft-ietf-rmcat-cc-requirements-03.txt 2014-03-31 14:14:35.589967509 -0700 @@ -1,84 +1,82 @@ Network Working Group R. Jesup Internet-Draft Mozilla -Intended status: Informational February 14, 2014 -Expires: August 18, 2014 +Intended status: Informational March 31, 2014 +Expires: October 2, 2014 Congestion Control Requirements For RMCAT - draft-ietf-rmcat-cc-requirements-02 + draft-ietf-rmcat-cc-requirements-03 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. 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 is derived from draft-jesup-rtp-congestion-reqs - [I-D.jesup-rtp-congestion-reqs]. - 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]. 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 August 18, 2014. + + This Internet-Draft will expire on October 2, 2014. 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 - 4. Security 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 The traditional TCP congestion control requirements were developed in order to promote efficient 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. @@ -88,117 +86,137 @@ video, where having the data ready when the viewer wants it is important, but the exact timing of the delivery is not. When doing real time interactive media, the requirements are different; one needs to provide the data continuously, within a very limited time window (no more than 100s of milliseconds end-to-end delay), the sources of data may be able to adapt the amount of data that needs sending within fairly wide margins, and may tolerate some amount of packet loss, but since the data is generated in real time, sending "future" data is impossible, and since it's consumed in real - time, data delivered late is useless. + time, data delivered late is commonly useless. While the requirements for RMCAT differ from the requirements for the other flow types, these other flow types will be present in the network. The RMCAT congestion control algorithm must work properly when these other flow types are present as cross traffic on the network. One particular protocol portofolio being developed for this use case is WebRTC [I-D.ietf-rtcweb-overview], where one envisions sending multiple RTP-based flows between two peers, in conjunction with data flows, all at the same time, without having special arrangements with the intervening service providers. Given that this use case is the focus of this document, use cases - involving noninteractive media such as YouTube-like video streaming, - and use cases using multicast/broadcast-type technologies, are out of - scope. + involving noninteractive media such as video streaming, and use cases + using multicast/broadcast-type technologies, are out of scope. The terminology defined in [I-D.ietf-rtcweb-overview] is used in this memo. 2. Requirements 1. The congestion control algorithm must attempt to provide as-low- as-possible-delay transit for real-time traffic while still - providing a useful amount of bandwidth, even when faced with - intermediate bottlenecks and competing flows. There may be - lower limits on the amount of bandwidth that is useful, but this - is largely application-specific and the application may be able - to modify or remove flows in order allow some useful flows to - get enough bandwidth. (Example: not enough bandwidth for low- + providing a useful amount of bandwidth. There may be lower + limits on the amount of bandwidth that is useful, but this is + largely application-specific and the application may be able to + modify or remove flows in order allow some useful flows to get + enough bandwidth. (Example: not enough bandwidth for low- latency video+audio, but enough for audio-only.) - A. It should also handle routing changes and interface changes - (WiFi to 3G data, etc) which may radically change the - bandwidth available, and react quickly, especially if there - is a reduction in available bandwidth. + 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. The offered load may be less than the available bandwidth at + 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) + 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 change in the bandwidth available from the algorithm and a change in the offered load is variable, and may be different when increasing versus decreasing. - C. The algorithm must not overreact to short-term bursts (such + E. The algorithm must not overreact to short-term bursts (such as web-browsing) which can quickly saturate a local- 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. - D. Similarly periodic bursty flows such as DASH or proprietary - media streaming algorithms may compete in bursts with the - algorithm, and may not be adaptive within a burst. They are - often are 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 fir the - duration of the burst. + 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 are 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. 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. + term. It should be self-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 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 bandwidth management, faster response to changing conditions, and fairer sharing of bandwidth with other network users. + Alternatively, it should work with an external bandwidth control framework to coordinate bandwidth usage across a bottleneck, such as draft-welzl-rmcat-coupled-cc [I-D.welzl-rmcat-coupled-cc]. A. If possible, it should also share information and adaptation with other non-RTP flows between the same endpoints, such as - a WebRTC data channel + a WebRTC DataChannel[I-D.ietf-rtcweb-data-channel] B. The most correlated bandwidth usage would be with other flows on the same 5-tuple, but there may be use in coordinating measurement and control of the local link(s). C. Use of information about previous flows, especially on the same 5-tuple, may be useful input to the algorithm, especially to startup performance of a new flow. + D. When there are multiple streams across the same 5-tuple + coordinating their bandwidth use and congestion control, it + should be possible for the application to control the + relative split of available bandwidth. + 4. The algorithm should not require any special support from network elements (ECN, etc). As much as possible, it should leverage available information about the incoming flow to provide feedback to the sender. Examples of this information are the ECN, packet arrival times, acknowledgments and feedback, packet timestamps, and packet losses; all of these can provide information about the state of the path and any bottlenecks. A. Extra information could be added to the packets to provide more detailed information on actual send times (as opposed @@ -206,22 +224,23 @@ B. When additional input signals such as ECN are available, they should be utilized if possible. 5. 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 AVPF/SAVPF that allows - sufficiently frequent feedback [RFC4585] MUST be used. + backchannel, an RTP profile such as RTP/AVPF [RFC4585] or + RTP/SAVPF [RFC5124] that allows sufficiently frequent + 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) @@ -234,24 +253,25 @@ 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. When the channel is unstable or has not yet reached 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. - 6. Flows managed by this algorithm and flows competed against at a - bottleneck may have different DSCP markings depending on the - type of traffic. A particular bottleneck or section of the - network path may or may not honor these markings. + 6. 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. 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. 7. The algorithm should sense the unexpected lack of backchannel information as a possible indication of a channel overuse problem and react accordingly to avoid burst events causing a @@ -336,50 +356,63 @@ 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-08 (work - in progress), September 2013. + based Applications", draft-ietf-rtcweb-overview-09 (work + in progress), February 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-ledbat-congestion] Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind, "Low Extra Delay Background Transport (LEDBAT)", draft- ietf-ledbat-congestion-10 (work in progress), September 2012. - [I-D.jesup-rtp-congestion-reqs] - Jesup, R. and H. Alvestrand, "Congestion Control - Requirements For Real Time Media", draft-jesup-rtp- - congestion-reqs-00 (work in progress), March 2012. + [I-D.ietf-rtcweb-data-channel] + Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data + Channels", draft-ietf-rtcweb-data-channel-07 (work in + progress), February 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. + [MPEG_DASH] + "Dynamic adaptive streaming over HTTP (DASH) -- Part 1: + Media presentation description and segment formats", April + 2012. + [RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size Real-Time Transport Control Protocol (RTCP): Opportunities and Consequences", RFC 5506, April 2009. + [RFC5865] Baker, F., Polk, J., and M. Dolly, "A Differentiated + Services Code Point (DSCP) for Capacity-Admitted Traffic", + RFC 5865, May 2010. + Author's Address Randell Jesup Mozilla USA Email: randell-ietf@jesup.org