Network Working Group                                           R. Jesup
Internet-Draft                                                   Mozilla
Intended status: Informational                         December 21, 2013
Expires: June 24, 2014

               Congestion Control Requirements For RMCAT


   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

Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   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
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   This Internet-Draft will expire on June 24, 2014.

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Table of Contents

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

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.

   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

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   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.

   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

   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

   The terminology defined in [I-D.ietf-rtcweb-overview] is used in this

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-
        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.

        B.  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.

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        C.  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.

   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

   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

        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

        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.

   4.   The algorithm should not require any special support from
        network elements (ECN, etc).  As much as possible, it should

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        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
            to sampling times), but should not be required.

        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.

        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)

        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.
            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.

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   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.

        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
        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
        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) algorithms.  Also note that these
        algorithms may apply across multiple queues in the bottleneck,
        or to a single queue

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

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

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            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).

   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

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
   mailing list.  Many people contributed their thoughts to this.

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6.  References

6.1.  Normative References

              Alvestrand, H., "Overview: Real Time Protocols for Brower-
              based Applications", draft-ietf-rtcweb-overview-08 (work
              in progress), September 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

6.2.  Informative References

              Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind,
              "Low Extra Delay Background Transport (LEDBAT)", draft-
              ietf-ledbat-congestion-10 (work in progress), September

              Jesup, R. and H. Alvestrand, "Congestion Control
              Requirements For Real Time Media", draft-jesup-rtp-
              congestion-reqs-00 (work in progress), March 2012.

              Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion
              control for RTP media", draft-welzl-rmcat-coupled-cc-02
              (work in progress), October 2013.

   [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

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   Randell Jesup


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