Network Working Group                                           R. Jesup
Internet-Draft                                                   Mozilla
Intended status: Informational                              July 4, 2014
Expires: January 5, 2015

               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.  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 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

Requirements Language

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   document are to be interpreted as described in RFC 2119 [RFC2119].
   The terms are presented in many cases using lowercase for

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
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   Drafts is at

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any

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   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 January 5, 2015.

Copyright Notice

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   document authors.  All rights reserved.

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   described in the Simplified BSD License.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   2
   2.  Requirements  . . . . . . . . . . . . . . . . . . . . . . . .   3
   3.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   8
   4.  Security Considerations . . . . . . . . . . . . . . . . . . .   8
   5.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .   8
   6.  References  . . . . . . . . . . . . . . . . . . . . . . . . .   8
     6.1.  Normative 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.

   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

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   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 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

   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 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

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.  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 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

        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.

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        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
            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.

        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.

        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 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 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.

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        A.  An algorithm may be more successful at avoiding starvation
            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
            (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
            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 (Voice Activity Detection/Discontinuous

        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
        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,

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        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 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.

   7.   The algorithm should not require any special support from
        network elements (Explicit Congestion Notification (ECN)
        [RFC3168], 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
            to sampling times), but should not be required.

        B.  When additional input signals such as ECN are available,
            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.

        B.  Note that in some cases, backchannel messages may be delayed
            until the RTCP channel can be allocated enough bandwidth,

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            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 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.
            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.

   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
        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
        problem and react accordingly to avoid burst events causing a
        congestion collapse.

   11.  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

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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.

6.  References

6.1.  Normative References

              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

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   [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

              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
              Channels", draft-ietf-rtcweb-data-channel-10 (work in
              progress), June 2014.

              Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion
              control for RTP media", draft-welzl-rmcat-coupled-cc-03
              (work in progress), May 2014.

              "Dynamic adaptive streaming over HTTP (DASH) -- Part 1:
              Media presentation description and segment formats", April

   [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
              of Explicit Congestion Notification (ECN) to IP", RFC
              3168, September 2001.

   [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


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