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In: MissingRef
Network Working Group                                           R. Jesup
Internet-Draft                                                   Mozilla
Intended status: Informational                            Z. Sarker, Ed.
Expires: June 15, 2015                                          Ericsson
                                                       December 12, 2014

    Congestion Control Requirements for Interactive Real-Time Media


   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 the Web Real-Time Communication (WebRTC)), it is especially
   important to ensure that this kind of traffic is congestion

   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 real-time 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
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

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   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 June 15, 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
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   publication of this document.  Please review these documents
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   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.  Deficiencies of existing mechanisms . . . . . . . . . . . . .   8
   4.  IANA Considerations . . . . . . . . . . . . . . . . . . . . .   9
   5.  Security Considerations . . . . . . . . . . . . . . . . . . .   9
   6.  Acknowledgements  . . . . . . . . . . . . . . . . . . . . . .  10
   7.  References  . . . . . . . . . . . . . . . . . . . . . . . . .  10
     7.1.  Normative References  . . . . . . . . . . . . . . . . . .  10
     7.2.  Informative References  . . . . . . . . . . . . . . . . .  10
   Authors' Addresses  . . . . . . . . . . . . . . . . . . . . . . .  11

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.

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   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 but can be rate limited
   by the application- even not always have data to send, 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 real-time interactive media differ from
   the requirements for the other flow types, these other flow types
   will be present in the network.  The congestion control algorithm for
   real-time interactive media must work properly when these other flow
   types are present as cross traffic on the network.

   One particular protocol portfolio being developed for this use case
   is WebRTC [I-D.ietf-rtcweb-overview], where one envisions sending
   multiple flows using the Real-time Transport Protocol (RTP) [RFC3550]
   between two peers, in conjunction with data flows, all at the same
   time, without having special arrangements with the intervening
   service providers.  As RTP does not provide any congestion control
   mechanism; a set of circuit breakers, such as
   [I-D.ietf-avtcore-rtp-circuit-breakers], are required to protect the
   network from excessive congestion caused by the non-congestion
   controlled flows.  When the real-time interactive media is congestion
   controlled, it is recommended that the congestion control mechanism
   operates within the constraints defined by these circuit breakers
   when circuit breaker is present and that it should not cause
   congestion collapse when circuit breaker is not implemented.

   Given that this use case is the focus of this document, use cases
   involving non-interactive 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 interactive 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.)

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        A.  Jitter (variation in the bitrate over short time scales)
            also is relevant, though moderate amounts of jitter will be
            absorbed by jitter buffers.  Transit delay should be
            considered to track the short-term maximums of delay
            including jitter.

        B.  It should provide this as-low-as-possible-delay transit and
            minimize self-induced latency even when faced with
            intermediate bottlenecks and competing flows.  Competing
            flows may limit what's possible to achieve.

        C.  It should be resilience to the effects of the events, such
            as routing changes, which may alter or remove bottlenecks or
            change the bandwidth available especially if there is a
            reduction in available bandwidth or increase in observed
            delay.  It is expected that the mechanism reacts quickly to
            the such events to avoid delay buildup.  In the context of
            this memo, a 'quick' reaction is on the order of a few RTTs,
            subject to the constraints of the media codec, but is likely
            within a second.  Reaction on the next RTT is explicitly not
            required, since many codecs cannot adapt their sending rate
            that quickly, but equally response cannot be arbitrarily

        D.  It should react quickly to handle both local and remote
            interface changes (WLAN to 3G data, etc) which may radically
            change the bandwidth available or bottlenecks, 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.

        E.  The real-time interactive media applications can be rate
            limited.  This means the offered loads can 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/unmute
            operation.  Hence, the algorithm must be designed to handle
            such behavior from media source or application.  Note that
            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.

        F.  The algorithm requires to avoid building up queues when
            competing with short-term bursts of traffic (for example,
            traffic generated by web-browsing) which can quickly
            saturate a local-bottleneck router or link, but also clear
            quickly.  The algorithm should also react quickly to regain

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            its previous share of the bandwidth when the local-
            bottleneck or link is cleared.

        G.  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 but use TCP in a bursty manner that can interact poorly
            with competing flows during the bursts.  The algorithm must
            not increase the already existing delay buildup during those
            bursts.  Note that this competing traffic may be 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 real-time 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 fair share of the
        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 as quickly
            as possible.  A useful share will depend on the media types
            involved, total bandwidth available and the user experience
            requirements of a particular service.  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.  The congestion control should prioritise achieving a useful
            share of the bandwidth depending on the media types and
            total available bandwidth over achieving as low as possible
            transit delay, when these two requirements are in conflict.

   4.   The algorithm should as quickly as possible 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

        A.  The algorithm should allow different modes of adaptation for
            example,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

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            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.  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.  Again, 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

   5.   The algorithm should be stable if the RTP streams are halted or
        discontinuous (for example - Voice Activity Detection).

        A.  After stream resumption, the algorithm should attempt to
            rapidly regain its previous share of the bandwidth; the
            aggressiveness with which this is done will decay with the
            length of the pause.

   6.   The algorithm should where possible merge information across
        multiple RTP streams sent between two endpoints, when those RTP
        streams share a common bottleneck, whether or not those streams
        are multiplexed onto 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.  The algorithm 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], when

        B.  When there are multiple streams across the same 5-tuple
            coordinating their bandwidth use and congestion control, the
            algorithm should allow the application to control the
            relative split of available bandwidth.  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).  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.

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   7.   The algorithm should not require any special support from
        network elements to convey congestion related information to be
        functional.  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 packet arrival
        times, acknowledgements and feedback, packet timestamps, and
        packet losses, Explicit Congestion Notification (ECN) [RFC3168];
        all of these can provide information about the state of the path
        and any bottlenecks.  However, the use of available information
        is algorithm dependent.

        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.

   8.   Since the assumption here is a set of RTP streams, the
        backchannel typically should be done via RTCP[RFC3550]; 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.  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 solutions to violate or modify the rules
            specified for AVPF.

        B.  Bandwidth for the feedback messages should be minimized
            (such as via RFC 5506 [RFC5506]to allow RTCP without Sender
            Reports/Receiver Reports)

        C.  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

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        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
            hierarchy, 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 maintain 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

3.  Deficiencies of existing mechanisms

   Among the existing congestion control mechanisms TCP Friendly Rate
   Control (TFRC) [RFC5348] is the one which claims to be suitable for
   real-time interactive media.  TFRC is, an equation based, congestion
   control mechanism which provides reasonably fair share of the
   bandwidth when competing with TCP flows and offers much lower
   throughput variations than TCP.  This is achieved by a slower
   response to the available bandwidth change than TCP.  TFRC is
   designed to perform best with applications which has fixed packet
   size and does not have fixed period between sending packets.

   TFRC operates on detecting loss events and reacts to loss caused by
   congestion by reducing its sending rate.  It allows applications to
   increase the sending rate until loss is observed in the flows.  As it
   is noted in IAB/IRTF report [RFC7295] large buffers are available in
   the network elements which introduces additional delay in the
   communication, it becomes important to take all possible congestion
   indications into considerations.  Looking at the current Internet
   deployment, TFRC's only consideration of loss events as congestion
   indication can be considered as biggest lacking.

   A typical real-time interactive communication includes live encoded
   audio and video flow(s).  In such a communication scenario an audio
   source typically needs fixed interval between packets, needs to vary
   their segment size instead of their packet rate in response to
   congestion and sends smaller packets, a variant of TFRC , Small-

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   Packet TFRC (TFRC-SP) [RFC4828] addresses the issues related to such
   kind of sources ; a video source generally varies video frame sizes,
   can produce large frames which need to be further fragmented to fit
   into path Maximum Transmission Unit (MTU) size, and have almost fixed
   interval between producing frames under a certain frame rate, TFRC is
   known to be less optimal when using with such video sources.

   There are also some mismatches between TFRC's design assumptions and
   how the media sources in a typical real-time interactive application
   works.  TFRC is design to maintain smooth sending rate however media
   sources can change rates in steps for both rate increase and rate
   decrease.  TFRC can operate in two modes - i) Bytes per second and
   ii) packets per second, where typical real-time interactive media
   sources operates on bit per second.  There are also limitations on
   how quickly the media sources can adapt to specific sending rates.
   The modern video encoders can operate on a mode where they can vary
   the output bitrate a lot depending on the way there are configured,
   the current scene it is encoding and more.  Therefore, it is possible
   that the video source does not always output at a bitrate they are
   allowed to.  TFRC tries to raise its sending rate when transmitting
   at maximum allowed rate and increases only twice the current
   transmission rate hence it may create issues when the video source
   vary their bitrates.

   Moreover, there are number of studies on TFRC which shows it's
   limitations which includes TFRC's unfairness on low statistically
   multiplexed links, oscillatory behavior, performance issue in highly
   dynamic loss rates conditions and more [CH09].

   Looking at all these deficiencies it can be concluded that the
   requirements of congestion control mechanism for real-time
   interactive media cannot be met by TFRC as defined in the standard.

4.  IANA Considerations

   This document makes no request of IANA.

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

5.  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.

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   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 perceived available bandwidth
   are conceivable, and need to be evaluated.  Such attacks could result
   in starvation of competing flows and permit amplification attacks.

   Algorithm designers should consider the possibility of malicious on-
   path attackers.

6.  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.

7.  References

7.1.  Normative References

              Alvestrand, H., "Overview: Real Time Protocols for
              Browser-based Applications", draft-ietf-rtcweb-overview-13
              (work in progress), November 2014.

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

   [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

   [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.

7.2.  Informative References

   [CH09]     Choi, S. and M. Handley, "Designing TCP-Friendly Window-
              based Congestion Control for Real-time Multimedia
              Applications", PFLDNeT 2009 Workshop , May 2009.

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              Perkins, C. and V. Singh, "Multimedia Congestion Control:
              Circuit Breakers for Unicast RTP Sessions", draft-ietf-
              avtcore-rtp-circuit-breakers-08 (work in progress),
              December 2014.

              Jesup, R., Loreto, S., and M. Tuexen, "WebRTC Data
              Channels", draft-ietf-rtcweb-data-channel-12 (work in
              progress), September 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.

   [RFC4828]  Floyd, S. and E. Kohler, "TCP Friendly Rate Control
              (TFRC): The Small-Packet (SP) Variant", RFC 4828, April

   [RFC5348]  Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
              Friendly Rate Control (TFRC): Protocol Specification", RFC
              5348, September 2008.

   [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.

   [RFC7295]  Tschofenig, H., Eggert, L., and Z. Sarker, "Report from
              the IAB/IRTF Workshop on Congestion Control for
              Interactive Real-Time Communication", RFC 7295, July 2014.

Authors' Addresses

   Randell Jesup

   Email: randell-ietf@jesup.org

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   Zaheduzzaman Sarker (editor)

   Email: zaheduzzaman.sarker@ericsson.com

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