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Versions: 00 01 draft-ietf-rmcat-cc-requirements

Network Working Group                                           R. Jesup
Internet-Draft                                                   Mozilla
Intended status: Informational                              Feb 25, 2013
Expires: August 29, 2013


               Congestion Control Requirements For RMCAT
                       draft-jesup-rmcat-reqs-01

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 by low-delay, semi-reliable data delivery,
   are different from the requirements for bulk transfer like FTP or
   bursty transfers like Web pages, and the TCP algorithms are not
   suitable for this traffic.

   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."



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   This Internet-Draft will expire on August 29, 2013.

Copyright Notice

   Copyright (c) 2013 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
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   described in the Simplified BSD License.



































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Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . . . 4
   2.  Requirements  . . . . . . . . . . . . . . . . . . . . . . . . . 4
   3.  IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 7
   4.  Security Considerations . . . . . . . . . . . . . . . . . . . . 8
   5.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . . . 8
   6.  References  . . . . . . . . . . . . . . . . . . . . . . . . . . 8
     6.1.  Normative References  . . . . . . . . . . . . . . . . . . . 8
     6.2.  Informative References  . . . . . . . . . . . . . . . . . . 8
   Author's Address  . . . . . . . . . . . . . . . . . . . . . . . . . 9








































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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 fast as possible" mode, such as when
   fetching Web pages.

   These algorithms have also been used for transfer of media streams
   that are viewed in a non-interactive manner, such as "streaming"
   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.

   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.

   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-



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        latency video+audio, but enough for audio-only.)

        A.  It should also deal well with routing changes and interface
            changes (WiFi to 3G data, etc) which may radically change
            the bandwidth available.

   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.

        A.  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.

        B.  We will need make some evaluation of fairness, but deciding
            what is "fair" is a tough question and likely to be
            partially subjective, but we should specify some of the
            inputs needed in order to select among algorithms and
            tunings presented as options.

   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.

        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

   4.   The algorithm should not require any special support from
        network elements (ECN, etc).  As much as possible, it should
        leverage existing information about the incoming flows to
        provide feedback to the sender.  Examples of this information
        are the packet arrival times, acknowledgments and feedback,
        packet timestamps, packet sizes, packet losses.  Extra
        information could be added to the packets to provide more
        detailed information on actual send times (as opposed to
        sampling times), but should not be required.

        A.  When additional input signals such as ECN are available,
            they should be utilized if possible.




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   5.   Since the assumption here is a set of RTP streams, the
        backchannel typically should be done via RTCP; the alternative
        would be to include it in a reverse RTP channel using header
        extensions.

        A.  In order to react sufficiently quickly, the AVPF/SAVPF RTP
            profile[RFC4585] 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.  Note that RTCP is of course unreliable

        D.  Bandwidth for the feedback messages should be minimized
            (such as via RFC 5506 [RFC5506]to allow RTCP without SR/RR)

        E.  Header extensions would avoid the RTCP timing rules issues,
            and allow the application to allocate bandwidth as needed
            for the congestion algorithm.

        F.  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.
            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 fequency are expected
            to be evaluated.

   6.   Where possible and helpful, the algorithm should leverage and
        piggyback on other RTP/RTCP communications, such as SR/RR,
        rctp-fb PLI, RPSI, SLI or application-specific NACK messages
        (such as for loss information), and also reverse-direction RTP.

   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
        congestion collapse.

   8.   It should attempt to avoid bandwidth 'collapse' when facing a
        long-lived saturating TCP flow or flows.  (I.e. a classic delay-
        sensitive algorithm will reduce bandwidth to keep delay down



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        until the TCP flow has all the bandwidth).  See the Cx-TCP
        algorithm discussed in a recent Transactions On Networking
        [cx-tcp] for an example of a delay-sensitive congestion-control
        algorithm that transitions to a loss-based mode when competing
        with TCP flows - at the cost of increased delay.

   9.   The algorithm should be stable and low-delay when faced with
        active queue management (AQM) such as RED [RFC2309] or CoDel
        [I-D.nichols-tsvwg-codel] in the channel.

   10.  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
            (adapt up) and history-based startup (start at a point
            expected to be at or below channel bandwidth from historical
            information, which may need to adapt down quickly if the
            initial guess is wrong).  Starting too low and/or adapting
            up too slowly can cause a critical point in a personal
            communication to be poor ("Hello!").

        B.  Starting over-bandwidth causes other problems for user
            experience, so there's a tension here.

        C.  Alternative methods to help startup like probing during
            setup with dummy data may be useful in some applications.

   11.  It should be evaluated in how it works both with backbone-router
        bottlenecks, (asymmetric) local-loop bottlenecks, and local-lan
        (WiFi/etc) bottlenecks, and in competition with varying numbers
        and types of streams (TCP, TCP variants in use, LEDBAT
        [I-D.ietf-ledbat-congestion], inflexible VoIP UDP flows).

   12.  It should be stable if the RTP streams are halted or
        discontinuous (VAD/DTX).

        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.


3.  IANA Considerations

   This document makes no request of IANA.

   Note to RFC Editor: this section may be removed on publication as an



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   RFC.


4.  Security Considerations

   An attacker with the ability to delete, delay or insert messages in
   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-
   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-06 (work
              in progress), February 2013.

   [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.

6.2.  Informative References

   [I-D.ietf-ledbat-congestion]
              Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,



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              "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.nichols-tsvwg-codel]
              Nichols, K., "Controlled Delay Active Queue Management",
              draft-nichols-tsvwg-codel-00 (work in progress),
              July 2012.

   [RFC2309]  Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering,
              S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,
              Partridge, C., Peterson, L., Ramakrishnan, K., Shenker,
              S., Wroclawski, J., and L. Zhang, "Recommendations on
              Queue Management and Congestion Avoidance in the
              Internet", RFC 2309, April 1998.

   [RFC5506]  Johansson, I. and M. Westerlund, "Support for Reduced-Size
              Real-Time Transport Control Protocol (RTCP): Opportunities
              and Consequences", RFC 5506, April 2009.

   [cx-tcp]   Budzisz, L., Stanojevic, R., Schlote, A., Baker, F., and
              R. Shorten, "On the Fair Coexistence of Loss- and Delay-
              Based TCP", December 2011.


Author's Address

   Randell Jesup
   Mozilla
   USA

   Email: randell-ietf@jesup.org













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