path: root/doc/rfc/rfc7295.txt

Internet Architecture Board (IAB)                          H. Tschofenig
Request for Comments: 7295                                     L. Eggert
Category: Informational                                        Z. Sarker
ISSN: 2070-1721                                                July 2014


        Report from the IAB/IRTF Workshop on Congestion Control
                for Interactive Real-Time Communication

Abstract

   This document provides a summary of the IAB/IRTF Workshop on
   'Congestion Control for Interactive Real-Time Communication', which
   took place in Vancouver, Canada, on July 28, 2012.  The main goal of
   the workshop was to foster a discussion on congestion control
   mechanisms for interactive real-time communication.  This report
   summarizes the discussions and lists recommendations to the Internet
   Engineering Task Force (IETF) community.

   The views and positions in this report are those of the workshop
   participants and do not necessarily reflect the views and positions
   of the authors, the Internet Architecture Board (IAB), or the
   Internet Research Task Force (IRTF).

Status of This Memo

   This document is not an Internet Standards Track specification; it is
   published for informational purposes.

   This document is a product of the Internet Architecture Board (IAB)
   and represents information that the IAB has deemed valuable to
   provide for permanent record.  It represents the consensus of the
   Internet Architecture Board (IAB).  Documents approved for
   publication by the IAB are not a candidate for any level of Internet
   Standard; see Section 2 of RFC 5741.

   Information about the current status of this document, any errata,
   and how to provide feedback on it may be obtained at
   http://www.rfc-editor.org/info/rfc7295.












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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.

Table of Contents

   1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .   3
   2.  Workshop Structure  . . . . . . . . . . . . . . . . . . . . .   5
     2.1.  History and Current Challenges  . . . . . . . . . . . . .   5
     2.2.  Simulations and Measurements  . . . . . . . . . . . . . .   8
     2.3.  Design Aspects of Problems and Solutions  . . . . . . . .   9
   3.  Recommendations . . . . . . . . . . . . . . . . . . . . . . .  13
     3.1.  Changes to Network Infrastructure . . . . . . . . . . . .  14
     3.2.  Avoiding Self-Inflicted Queuing . . . . . . . . . . . . .  15
   4.  Security Considerations . . . . . . . . . . . . . . . . . . .  17
   5.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .  17
   6.  Informative References  . . . . . . . . . . . . . . . . . . .  17
   Appendix A.  Program Committee  . . . . . . . . . . . . . . . . .  22
   Appendix B.  Workshop Material  . . . . . . . . . . . . . . . . .  22
   Appendix C.  Accepted Position Papers . . . . . . . . . . . . . .  22
   Appendix D.  Workshop Participants  . . . . . . . . . . . . . . .  24






















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1.  Introduction

   The Internet Architecture Board (IAB) holds occasional workshops
   designed to consider long-term issues and strategies for the
   Internet, and to suggest future directions for the Internet
   architecture.  This long-term planning function of the IAB is
   complementary to the ongoing engineering efforts performed by working
   groups of the Internet Engineering Task Force (IETF), under the
   leadership of the Internet Engineering Steering Group (IESG) and area
   directorates.

   Any application that sends significant amounts of data over the
   Internet is expected to implement reasonable congestion control
   behavior.  The goals for congestion control are well understood and
   documented in RFC 2914 [2] and RFC 5405 [1]:

   1.  Preventing congestion collapse.

   2.  Allowing multiple flows to share the network fairly.

   The Internet has been used for interactive real-time communication
   for decades, most of which is being transmitted using the Real-Time
   Transport Protocol (RTP) over UDP, often over provisioned capacity
   and/or using only rudimentary congestion control mechanisms.  In
   2004, the IAB raised concerns regarding possibilities of a congestion
   collapse due to a rapid growth in real-time voice traffic that does
   not practice end-to-end congestion control [17].  That congestion
   collapse did not happen, but concerns raised about both congestion
   collapse and fairness are still valid and have gained more relevance
   when applied to more bandwidth-hungry video applications.  The
   development and upcoming widespread deployment of web-based real-time
   media communication -- where RTP is used to and from web browsers to
   transmit audio, video, and data -- will likely result in substantial
   new Internet traffic.  Due to the projected volume of this traffic,
   as well as the fact that it is more likely to use unprovisioned
   capacity, it is essential that it is transmitted with robust and
   effective congestion control mechanisms.

   Designing congestion control mechanisms that perform well under a
   wide variety of traffic mixes and over network paths with widely
   varying characteristics is not easy.  Prevention of congestion
   collapse can be achieved through a "circuit breaker" mechanism (see,
   for example, [3]), but for media flows that are supposed to coexist
   with a user's other ongoing communication sessions, a congestion
   control mechanism that shares capacity fairly in the presence of a
   mix of TCP, UDP, and other protocol flows is needed.





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   Many additional complications arise.  Here are some examples:

   1.  Real-time interactive media sessions require low latencies,
       whereas streaming media can use large play-out buffers.

   2.  In an RTP session, feedback exchanged via the RTP Control
       Protocol (RTCP) typically arrives much less frequently than, for
       example, TCP ACKs for a given TCP connection.  Theoretically, the
       RTP/RTCP control loop can lead to a longer reaction time.

   3.  Media codecs can usually only adjust their output rates in a much
       more coarse-grained fashion than, for example, TCP, and user
       experience suffers if encoding rates are switched too frequently.
       Codecs typically have a minimum sending rate as well.

   4.  Some bits of an encoded media stream are more important than
       others.  For example, losing or dropping an I-frame of a video
       stream is more problematic than dropping a P-frame [40].

   5.  Ramping up the transmission rate can be problematic.  Simply
       increasing the output rate of the codec without knowing whether
       the network path can sustain transmission at the increased rate
       runs the danger of incurring a significant amount of packet loss
       that can cause playback artifacts.

   6.  A congestion control scheme for interactive media needs to handle
       bundles of interrelated flows (audio, video, and data) in a way
       that accommodates the preferences of the application in the event
       of congestion.

   7.  The desire to provide a congestion control mechanism that can be
       efficiently implemented inside an application imposes additional
       restrictions.  For example, a web browser is not able to take the
       protocol interactions of a software download happening in another
       application into account.

   8.  There are explicit congestion signals (such as Explicit
       Congestion Notification (ECN) [19]), and there are implicit
       indications of congestion (e.g., packet delay and loss).  Care
       must be taken to account for each of these signals, particularly
       if various applications react on the same set of signals.

   9.  Large buffers are often used in network elements and end device
       operating systems to better support TCP-based applications.
       These buffers introduce additional communication delay, which
       harms the small delay budget available for interactive real-time
       applications.




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2.  Workshop Structure

   The IETF has a long history of work on congestion control mechanisms.
   With ongoing standardization work on real-time interactive media
   communication on the web, new challenges have emerged that have
   refocused engineering attention on congestion control issues.  To
   take a deeper look at congestion control in light of the growth of
   real-time traffic, workshop participants were invited to submit
   position papers that were then used to organize the workshop agenda
   into three principal components: a keynote talk given by Mark Handley
   describing the history of the work on congestion control for real-
   time media followed and his views of current problems; a presentation
   of simulations and data demonstrating current problems and solutions;
   and a discussion of desirable solution properties and challenges in
   deploying solutions.

2.1.  History and Current Challenges

   Mark Handley argued that since 1988, the Internet has remained
   functional despite exponential growth, routers that are sometimes
   buggy or misconfigured, rapidly changing applications and usage
   patterns, and flash crowds.  This is largely because most
   applications use TCP, and TCP implements end-to-end congestion
   control.

   TCP's congestion control adapts the window to fit the capacity
   available in the network and accomplishes approximate fairness
   between two competing flows over a period of time.  Mark indicated
   that the provided level of fairness is not necessarily what we want:
   The 1/round-trip-time relationship in TCP is not ideal since it means
   that network operators can decide to lower packet loss by adding
   bigger buffers (which unfortunately leads to bufferbloat problems;
   see [31] and [39]).  The 1/sqrt(packet drop rate) relationship is
   also not necessarily desirable since TCP initially did not work
   particularly well for high-speed flows (which had been the subject of
   much TCP research).

   TCP controls the congestion window in bytes.  For bulk transfer,
   usually this results in controlling the number of 1500-byte packets
   sent per second.  Real-time media is different since it has its own
   time constraints.  For audio, one wants to send one packet per 20 ms
   and for video, the ideal value would be 25 to 30 frames per second.
   One, therefore, wants to avoid additional sending delay.

   As an example, in case of video, to relieve congestion one has to
   reduce the number of packets-per-second transmission rate rather than
   transmit smaller packets, since at higher bitrates on WiFi the time
   it takes to send a packet is almost negligible compared to the time



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   that is spent with Media Access Control (MAC) layer operations.
   Reducing the packet size makes little difference to the available
   capacity.  For a serial line, it does not matter how big the packets
   are.

   From a network point of view, the goals of congestion control
   therefore are:

   1.  Avoid congestion collapse

   2.  Avoid starvation of TCP flows

   3.  Avoid starvation of real-time flows, specifically in the case
       where TCP and real-time flows share the same FIFO queue.

   From an application point of view, the goals of congestion control
   are different, namely:

   1.  Robust behavior.  One wants to have a good throughput when the
       network is working well and passable performance when the network
       is working poorly.

   2.  Predictable behavior.  This matters from a usability point of
       view since variable media creates a bad user experience.

   3.  Low latency.  With large buffers along the end-to-end path,
       latency will increase when interactive real-time flows compete
       with TCP flows.  This results in TCP filling up the buffers;
       increased buffering will lead to additional delays for the
       delivery of the interactive real-time media.

   Attempts to provide congestion control for interactive real-time
   media have previously been made in the IETF, for example, with the
   work on TCP Friendly Rate Control (TFRC) [12].  TFRC illustrates the
   challenges quite well.  TFRC tries to accomplish the same throughput
   as TCP, but with a smoother transmission rate.  It measures the loss
   and the round-trip time but follows a similar model as TCP to
   determine the sending rate.

   In a link with low statistical multiplexing, TCP can lead to bad
   oscillations.  The sending rate hits the maximum rate of a bottleneck
   link, a lot of loss occurs, and then the sending rate peaks again.
   For very small buffers the result is acceptable, but bigger buffers
   lead to oscillations.  The result is bad for networks and for
   applications.  To deal with large buffers on these links, a short-
   term rate adaptation based on round-trip time (RTT) information is
   utilized in TRFC, but this requires good short-term RTT measurements.




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   TRFC works pretty well in theory.  TFRC assumes the network is in
   charge of the codec and that the codec can produce data at the
   demanded rate.  Modern video codecs inherently produce variable-
   bitrate video streams based on the content being encoded, and it is
   hard to produce data at exactly the desired bitrate without excessive
   buffering or ugly quality changes.

   What if the codec is put in charge instead of the network?  The
   network tells the codec the mean rate, but it does not worry about
   what happens in short time scales, and the codec matches the mean
   rate and does not worry whether it is over or under the rate for a
   relatively short time scale.  This again leads to the low statistical
   multiplexing problem and leads to oscillations.

   Known congestion control mechanisms work well if they can respond
   quickly enough to changes and if they do not bump into the low
   statistical multiplexing problem.

   To avoid the low statistical multiplexing problem, techniques for
   inferring link speed are needed.  The work from Van Jacobson's
   pathchar [37] (and successors) serve as valuable input.  The idea is
   to send short packet trains, to measure timing accurately, and to
   infer the link speed from the relative delay.  If we know the link
   speed, we can avoid exceeding it.  Congestion control can give us an
   approximate rate, but we must not exceed link speed.  This is a
   hybrid between codec being in charge (most of the time) and the
   network being in charge.  These work well for some links, but not for
   others.  Wireless links where speed can change in less than a single
   RTT because of fading, bitrate adaption, etc., cause problems.  We
   would like to have the codec and the network be in charge.  However,
   they both cannot be in charge at the same time.

   Mark indicated that he is not entirely sure whether RTCP is suitable
   for congestion control.  RTCP gives feedback, but it cannot send it
   often enough to avoid bumping into link speed.  Circuit breakers [3],
   on the other hand, do not help to give good performance on an
   uncongested path.  With circuit breakers, the sender measures the
   loss rate and RTT, and runs with a loose "cap."

   In conclusion, Mark Handley claimed that we know how to do good
   congestion control, but only if congestion control is in charge, and
   that's not acceptable for real-time applications.  We only know how
   to do good congestion control if we change the packet/sec rate and
   not the packet size.







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2.2.  Simulations and Measurements

   This second part of the workshop was focused on the presentation and
   the discussion of data gathered from simulations and real-world
   measurements.

   Keith Winstein started the discussion with his presentation of
   measurements performed in cellular operator networks in the US [22].
   The measurements indicate that the analyzed cellular networks showed
   varying RTT with transient latency spikes to hundreds of
   milliseconds, link speed that varies by a factor of 10 in a short
   time scale, and buffers that do not drop packets until they contain
   5-10 seconds of data at bottleneck link speed.

   Zaheduzzaman Sarker [21] presented results from real-time video
   communication in a Long Term Evolution (LTE) simulator utilizing ECN-
   based packet marking and adaptation using implicit methods like
   packet loss and delay.  ECN marking provides ways for the network to
   explicitly signal congestion and hence distributes the cost of
   congestion well and helps achieve lower latency.  However, although
   RFC 3168 [19] was finalized in 2001, the deployment of ECN is still
   lacking as investigated by Bauer, et al. [25].  A few participants
   noted that they believe that the deployment of LTE networks will also
   increase the deployment of ECN with the recent work on ECN for RTP
   over UDP [11].

   Mo Zahaty [20] discussed TFRC [12] and TFRC with weighted fairness
   (MulTFRC) [4], which tunes TFRC to consider multiple flows, and
   showed the impact of RTT and loss rates on the type of video quality
   that can be achieved under those conditions.  TFRC requires frequent
   feedback, which RTCP does not provide even when considering the
   extended RTP profile for RTCP-based feedback (RFC 4585 [5]).  Mo
   argued that application-specified weighted fairness is important but
   while MulTFRC provides better performance than TFRC, it is not clear
   whether the added complexity over an n-times-TFRC approach is indeed
   worth the effort.

   Markku Kojo shared analysis results of how real-time audio is
   affected by competing TCP flows.  In the experiments shown in
   Figure 2 of [27], a real-time interactive audio stream had to compete
   against one TCP flow and, as a comparison, against six TCP flows.
   With one concurrent TCP flow, voice is impacted on startup and six
   TCP flows destroy the quality of the call.  Two types of losses were
   analyzed, namely losses that result from a packet being dropped in
   the network (e.g., due to congestion or link errors) and losses that
   result from the delayed arrival of the packet (due to buffering) when
   the audio packet misses the deadline for the codec to decode and play
   the transmitted content.  Consequently, even a moderate number of TCP



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   flows typically used by browsers to retrieve content on web pages in
   parallel causes irreparable harm for audio transfers.  The size of
   the initial window (IW) also impacts interactive real-time
   communication since a larger TCP IW size (e.g., IW10 with ten
   segments, as proposed in [18], instead of three) leads to a bigger
   burst of packets because of the initial window transmission.  Note
   that the study in [24] does not necessarily lead to the same
   conclusion.  It claims that the increased initial window size leads
   to no impact or only modest impact for buffering in the majority of
   cases.

   Cullen Jennings [28] presented measurement results showing
   interactions between RTP and TCP flows for several widely deployed
   video communication products: Apple FaceTime, Google Hangout, Cisco
   Movi, and Microsoft Skype.  While all tested products implemented
   some form of congestion control, none of the applications did
   additive increase and multiplicative decrease (AIMD).  In general, it
   was observable that video adapts more slowly than AIMD to changes in
   available bandwidth because most codecs cannot make small increases
   in sending rates when available bandwidth increases, and do not make
   large decreases in sending rates when available bandwidth decreases,
   in order to improve the user's experience.

   Stefan Holmer [43] investigated the difference between loss-based and
   delay-based congestion control algorithms.  The suitability of loss-
   based congestion control schemes for interactive real-time
   communication systems heavily depends on buffer sizes and the
   deployment of active queue management mechanisms.  If most routers
   are using tail-drop queuing, then loss-based congestion control
   cannot fulfill the requirements of interactive real-time applications
   since those flows will effectively increase the bitrate until a loss
   event is identified, which only happens when the bottleneck queue is
   full.

2.3.  Design Aspects of Problems and Solutions

   During the remaining part of the workshop, the participants discussed
   design aspects of both the problem and solution spaces.  The
   discussions started with a presentation by Jim Gettys about problems
   related to bufferbloat [31][36].  Bufferbloat is "a phenomenon in
   packet-switched networks, in which excess buffering of packets causes
   high latency and packet delay variation (also known as jitter), as
   well as reducing the overall network throughput" [39].  A certain
   amount of buffering is helpful to improve the efficiency.  Not
   dropping packets in the event of congestion leads to increasing
   delays for interactive real-time communication.





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   Packets may get buffered at various places along the end-to-end path
   including in the operating system/device drivers, customer premise
   equipment (such as cable modem and DSL routers), base stations, and
   routers.  While the understanding of too large buffers has improved
   over the last few years, workshop participants were still concerned
   that many equipment manufacturers and network operators do not yet
   acknowledge the existence of the problem.  This lack of understanding
   is caused by the strong focus on throughput network performance
   measurements that do not take latency into account.  For example,
   only recently the Federal Communications Commission (FCC) has added
   latency tests to their test suites [41].

   Active queue management (AQM) aims to prevent queues from growing too
   large.  This is accomplished by monitoring queue length and informing
   the sender by dropping or marking packets to lower their transmission
   rate.  Random Early Detection (RED) [9] is one such AQM algorithm,
   but it has not been widely deployed in routers largely because of
   challenges to configure it correctly [32].  According to [23], RED
   does not work with the default settings as it is "too "gentle" to
   handle fast changes due to TCP slow start, when the aggregate traffic
   is limited."  There may also be a lack of incentives to deploy AQM
   algorithms.  Participants speculated about the time it takes to
   update network equipment (to support AQM algorithms), considering the
   different replacement cycles of these devices.

   One outcome of that discussion on AQM at the workshop was a Birds of
   a Feather ("BoF") meeting on "Active Queue Management and Packet
   Scheduling" at IETF 87 (July 28 - August 5, 2013, Berlin, Germany).
   The AQM WG [35] was chartered a few weeks later and is now designing
   AQM and network infrastructure improvements to deal with bufferbloat
   and related issues.

   Measurement tools that allow an end user to determine the performance
   of his or her network, including latency, is seen as a promising
   approach to motivate network operators to upgrade their equipment and
   to make use of AQM algorithms.  Measurement tools would allow users
   to determine how bad their networks perform and to complain to their
   ISP, thereby creating a market force.  As to what the right
   performance measurement metrics are, it was noted that the intent of
   the IETF IP Performance Metrics (IPPM) working group [33] was to
   develop such metrics to qualify networks.  That work may have begun
   before its time, but there have been recent attempts to revisit the
   measurement work and an effort by the FCC has gotten a lot of
   attention recently (see [7] and [42]).

   Matt Mathis and others argued that the traffic of throughput-
   maximizing and delay-minimizing applications need to be in separate
   queues (segregated queuing).  Requiring segregated queues assumes you



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   are sharing the network with other greedy traffic.
   Quality-of-Service (QoS) signaling is a way to deploy segregated
   queuing, but there are several simpler alternatives, such as
   Stochastic Fair Queuing [38].  The Controlled Delay (CoDel) AQM
   algorithm [6] can also be used in combination with stochastic fair
   queuing.  Note that queue segregation is not necessary for every
   router to implement; using it at the edge of a network where
   bottleneck links are located is already sufficient.

   It was noted that current interactive voice usage over the Internet
   works most of the time satisfactorily.  In typical networks, the
   reason voice works is because networks are underloaded.  As long as
   there is idle capacity and the queue is empty when packets arrive,
   traffic does not need to be separated into distinct queues.  Further
   explanations were offered as to why many networks work surprisingly
   well: Low Extra Delay Background Transport (LEDBAT) [8] is used for
   the download of software updates, voice traffic contributes only a
   small percentage of the overall Internet traffic, and users employ
   "human protocols" (e.g., parents asking their kids to get off the
   network during the time of a conference call).

   Cullen Jennings raised a concern that although interactive voice may
   be functional without a congestion control mechanism, the potentially
   large uptake of interactive video spurred on by Real-Time
   Communications on the Web (RTCWEB) could create substantially more
   significant problems.  In the class of space where voice is currently
   working, video may fail.  Ted Hardie countered by saying that RTCWEB
   is trying to replace existing proprietary technologies.  It may ramp
   up the amount of use we are expecting, but it is not doing much that
   was not being done by Adobe Flash or Skype.  RTCWEB is not a totally
   novel context of Internet usage.  Magnus Westerlund added that RTCWEB
   might be the driver for the moment, but web browsers are not the only
   consumers of such congestion control algorithm.

   Furthermore, Ted Hardie noted that applications will not produce
   media streams that grow to 10 Mbps because their sending rate is auto
   rate limited by the production of the video.  He suggested to ask
   ourselves if we are trying to get TCP to be friendly to media streams
   that are already rate limited or if we are asking media streams that
   are already rate limited to be TCP friendly.  To quote Andrew
   McGregor: "It's really not good to be TCP friendly because it's not
   going to return the favor."  If the desired properties we want are no
   starvation, fairness, and effective goodput for the offered loads,
   are we only willing to consider changes in RTP control, or are we
   willing to consider changes in TCP congestion control?






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   This led to a discussion about whether the development of a
   congestion control algorithm for interactive real-time applications
   provides any value if network equipment suffers from bufferbloat.  Is
   there something that can be done today to help interactive real-time
   media or do we have to wait to get the network updated first?
   Replacing home routers and updating routers with modern AQM
   algorithms was seen as a longer-term effort.  Also, the time scale
   for changing TCP's congestion control is on the same time scale as
   deploying ECN [19].  Colin Perkins noted that we cannot change TCP
   quickly; the way TCP is being used is changing quickly, and we can
   impact the way TCP is used.  When TCP is used for file transfer, it
   will send data as fast as it can, but when TCP is used for
   WebSockets, the dynamics are different.  WebSockets and SPDY are
   clearly changing the behavior of TCP.  Also, Netflix-style video-
   streaming applications are huge users of TCP and those applications
   can change rather quickly.  Matt Mathis added that real-time
   videoconferencing almost always produces video streams at a lower
   bitrate than downloading equivalent-sized stored video using best-
   effort file-sharing.

   Bill Ver Steeg suggested to consider three different deployment
   environments, namely:

   1.  Flows competing with flows from the host ("self-inflicted queuing
       delay")

   2.  Flows competing with flows in the same subnetwork (e.g., home
       network)

   3.  Flows competing with flows from other networks (e.g., traffic
       from different households that utilize the same DSL provider)

   The narrowest problem domain that makes sense is to avoid self-
   inflicted queuing delay.  Michael Welzl indicated that this requires
   an information exchange (called flow-state exchange) inside a browser
   (at the level of the same host or even beyond, as described in [29])
   to synchronize congestion control of different audio, video, and data
   flows.  Although it would provide great benefits if one could share
   information about a bottleneck with all the flows sharing that
   bottleneck, this is considered challenging even within a single host.
   John Leslie [30] also noted: "We're acting as if we believe
   congestion will magically be solved by a new transport algorithm.  It
   won't."  Instead, an interaction between the network layer, transport
   layer, and the application layer is needed whereby the application
   layer is the only practical place to balance what piece(s) to
   constrain to lower bandwidths.  All flows relating to a user session





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   should have a common congestion controller.  For many applications,
   audio is much more critical than video.  In those cases, the video
   may back off, but the audio transmission remains unchanged.

   Mo Zanaty pointed to the importance of the media start-up behavior,
   which is an area where the exchange of real-time interactive media is
   different from a TCP-based file transfer.  The instantaneous
   experience in the first part of a video call is highly determinative
   of people's perception of the call quality.  Vendors are using vague
   heuristics, for example, data from the last call to figure out what
   to do on the next call.  Lars Eggert highlighted that the start-up
   behavior of an application affects ongoing performance of other flows
   if, for example, an application blasts at line rate at the beginning
   of a video stream.  You need to start slow enough to not cause
   congestion to others.  Randell Jesup argued that for an interactive
   real-time video application, you really need to have most of your
   bandwidth right away.  Colin Perkins agreed and added that on startup
   you need good quality video quickly, but perhaps not as quickly as
   voice.  The requirements are likely going to be different from audio
   to video and maybe even vary between different applications.  Various
   protocol exchanges take place before media is exchanged between
   endpoints (such as Session Traversal Utilities for NAT (STUN) packets
   [13] as part of the Interactive Connectivity Establishment (ICE) [15]
   or a Datagram Transport Layer Security (DTLS) handshake [14]) and may
   be used to obtain simple start-up measurements.

   The group agreed that it is feasible to design a congestion control
   algorithm that works on mostly idle networks.  In the view of the
   participants, upgrades of the network infrastructure can happen in
   parallel.  This view was later confirmed at the RTP Media Congestion
   Avoidance Techniques (RMCAT) BoF meeting at IETF 84 (July 29 - August
   3, 2012, Vancouver, BC, Canada) that led to the formation of the
   RMCAT working group [34].

3.  Recommendations

   The participants suggested to explore two primary solution tracks:
   changes to network infrastructure and the development of algorithms
   to avoid self-inflicted queuing.  These are discussed below.  A third
   approach recommended by some participants was to change the way TCP
   is used in browsers and other HTTP-based applications.  For example,
   by not opening too many concurrent TCP connections, and by improving
   the interaction with other non-real-time applications (such as video
   streaming and file sharing), additional improvements can be made.
   The work on HTTP 2.0 with SPDY [16] is already a step in the right
   direction since SPDY makes use of a more aggressive form of
   multiplexing instead of opening a larger number of TCP connections.




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3.1.  Changes to Network Infrastructure

   As for all other traffic on the network, better data plane
   infrastructure improves the perceived quality of the best-effort
   service that the Internet provides for RTCWEB flows.  The IETF has
   already developed several technologies that would be of immediate
   usefulness if they were to be deployed.  The workshop participants
   expressed the hope that due to the volume and importance of RTCWEB
   traffic, some of these technologies might finally see widespread use.

   The first and by far most important improvement is traffic
   segregation: the ability to use different queues for different
   traffic types.  Specifically, jitter- and delay-sensitive protocols
   would benefit from being in different queues from throughput-
   maximizing protocols.  It is not possible for a single queue/AQM to
   be optimal for both.

   Furthermore, ECN allows routers along the end-to-end path to signal
   the onset of congestion and allows applications to respond early,
   avoiding losses and keeping queue sizes short and, therefore,
   end-to-end delay low.  ECN is implemented on some end system stacks
   and routers, but is frequently not enabled.  The participants
   expressed the importance of increasing the deployment of ECN, even if
   used initially only in closed environments, such as data centers (as
   with Data Center TCP (DCTCP) [26]).

   Different mechanisms have been developed to facilitate traffic
   segregation.  Differentiated Services [10] is one possibility in this
   space.  If applications start to mark outgoing traffic appropriately
   and routers segregate traffic accordingly, browsers could more
   directly control the relative importance of their various flows and
   avoid self-competition.  Compared to ECN, however, DiffServ is far
   more difficult to deploy meaningfully end to end, especially given
   that Differentiated Services Code Points (DSCPs) have no defined end-
   to-end meaning and packets can be re-marked.

   QoS signaling together with resource reservation facilities would
   enable a fine-grained and flexible way to indicate resource needs to
   network elements, but it is also by far the most heavyweight
   proposal, and unlikely to be viable in the global Internet.  However,
   as mentioned in Section 2.3, QoS signaling is not the only way to
   accomplish traffic segregation.  Further investigations regarding
   stochastic fair queuing and new AQM algorithms are seen as desirable.

   In any case, network infrastructure updates will take time,
   particularly if the interest of the involved stakeholders is not
   aligned (as is often the case for network operators when dealing with




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   over-the-top real-time traffic).  It is, therefore, imperative that
   RTCWEB congestion control provides adequate improvement in the
   absence of any of the aforementioned schemes.

3.2.  Avoiding Self-Inflicted Queuing

   This approach tries to ensure that the network does not suffer from
   congestion collapse and that one data flow from a single host does
   not harm another data flow from the same host.  A single congestion
   manager within the end host or the browser could help to coordinate
   various congestion control activities and to ensure a more
   coordinated approach between different applications and different
   flows.

   The following design and testing aspects were considered relevant to
   this approach:

   Reacting to All Congestion Signals:

      To initiate the congestion control process, it is important to
      detect congestion in the communication path.  Congestion can be
      detected using either an explicit mechanism or an implicit
      mechanism.  An explicit mechanism involves direct congestion
      signaling usually from the congested network node, such as ECN.
      In case of an implicit mechanism, packet-loss events or observed
      delay increases are used as an indication for congestion.  These
      measurements can also be made available in a variety of different
      protocols, such as RTCP reports or transport protocols.  It is
      recommended for applications to take all available congestion
      signals into account and to couple the congestion control
      algorithm, the codec, and the application so that better
      information exchange between these components is possible since
      there are constraints on how quickly a codec can adapt to a
      specific sending rate.

   Delay- and Loss-Based Algorithms:

      The main goal of designing a congestion control algorithm for
      real-time conversational media is to achieve low latency.
      Explicit congestion signals provide the most reliable way for
      applications to react, but due to the lack of ECN deployment,
      delay-based algorithms are needed.  Since there is large delay
      variation in wireless networks (even in a non-congested network),
      the workshop participants recommended that more research should be
      done to better understand non-congestion-related delay variation
      in the network.  General consensus among the workshop participants
      was that latency-based congestion control algorithms are needed




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      due to the lack of loss indications caused by large buffers, even
      though loss-based techniques dominate latency-based techniques
      when the two are competing for bandwidth.

   Algorithm Evaluation:

      The Internet consists of heterogeneous networks, which include
      misconfigured and unmanaged network nodes.  Bandwidth and latency
      vary a lot.  Different services deployed using RTP/UDP have
      different requirements in terms of media quality.  A congestion
      control algorithm needs to perform well not only in simulators but
      also in the deployed Internet.  To achieve this, it is recommended
      to test the algorithms with real-world loss and delay figures to
      ensure that the desired audio/video rates are attainable using the
      proposed algorithms for the desired services.

   Media Characteristics:

      Interactive real-time voice and video data are inherently
      variable.  Usually the content of the media and service
      requirements dictate the media coding.  The codec may be bursty
      and not all frames are equally important (e.g., I-frames are more
      important than P-frames).  Thus, codecs have limited room for
      adaptation.  Congestion control for audio and video codecs is,
      therefore, different from congestion control applied to bulk file
      transfers where buffering is not a problem and the transmission
      rate can be changed to any rate suitable for the congestion
      control algorithm.  In the workshop, these limitations were
      brought up and the workshop participants recommended that a
      congestion controller needs to be aware of these constraints.
      However, further investigation is needed to decide what
      information needs to be exchanged between a codec and the
      congestion manager.

   Start-up Behavior:

      The start-up media quality is very important for real-time
      interactive applications and for user-perceived application
      performance.  The start-up behavior of these is also different
      from other traffic.  By nature, real-time interactive
      communication applications want to provide a smooth user
      experience and maintain the best media quality possible to ease
      the interaction.  While it may be desirable from a user-experience
      point of view to immediately start streaming video with high-
      definition quality and audio of a wideband codec, this will have
      impacts on the bandwidth of the already ongoing flows.  As such,





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      it would be ideal to start slow enough to avoid causing excessive
      congestion to other flows but fast enough to offer a good user
      experience.  The sweet spot, however, yet has to be found.

4.  Security Considerations

   Two position papers focused on security, but these papers were not
   discussed during the workshop.  As such, nothing beyond the material
   contained in those position papers can be reported.

5.  Acknowledgments

   We would like to thank the participants and the paper authors of the
   position papers for their input.

   Additionally, we would like to thank the following persons for their
   review comments: Michael Welzl, John Leslie, Mirja Kuehlewind, Matt
   Mathis, Mary Barnes, Spencer Dawkins, Dave Thaler, and Alissa Cooper.

6.  Informative References

   [1]   Eggert, L. and G. Fairhurst, "Unicast UDP Usage Guidelines for
         Application Designers", BCP 145, RFC 5405, November 2008.

   [2]   Floyd, S., "Congestion Control Principles", BCP 41, RFC 2914,
         September 2000.

   [3]   Perkins, C. and V. Singh, "Multimedia Congestion Control:
         Circuit Breakers for Unicast RTP Sessions", Work in Progress,
         February 2014.

   [4]   Welzl, M., Damjanovic, D., and S. Gjessing, "MulTFRC: TFRC with
         weighted fairness", Work in Progress, July 2010.

   [5]   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]   Nichols, K. and V. Jacobson, "Controlled Delay Active Queue
         Management", Work in Progress, March 2014.

   [7]   Schulzrinne, H., Johnston, W., and J. Miller, "Large-Scale
         Measurement of Broadband Performance: Use Cases, Architecture
         and Protocol Requirements", Work in Progress, September 2012.

   [8]   Shalunov, S., Hazel, G., Iyengar, J., and M. Kuehlewind, "Low
         Extra Delay Background Transport (LEDBAT)", RFC 6817, December
         2012.



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   [9]   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.

   [10]  Blake, S., Black, D., Carlson, M., Davies, E., Wang, Z., and W.
         Weiss, "An Architecture for Differentiated Services", RFC 2475,
         December 1998.

   [11]  Westerlund, M., Johansson, I., Perkins, C., O'Hanlon, P., and
         K. Carlberg, "Explicit Congestion Notification (ECN) for RTP
         over UDP", RFC 6679, August 2012.

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

   [13]  Rosenberg, J., Mahy, R., Matthews, P., and D. Wing, "Session
         Traversal Utilities for NAT (STUN)", RFC 5389, October 2008.

   [14]  Rescorla, E. and N. Modadugu, "Datagram Transport Layer
         Security Version 1.2", RFC 6347, January 2012.

   [15]  Rosenberg, J., "Interactive Connectivity Establishment (ICE): A
         Protocol for Network Address Translator (NAT) Traversal for
         Offer/Answer Protocols", RFC 5245, April 2010.

   [16]  Belshe, M., Peon, R., and M. Thomson, "Hypertext Transfer
         Protocol version 2", Work in Progress, June 2014.

   [17]  Floyd, S. and J. Kempf, "IAB Concerns Regarding Congestion
         Control for Voice Traffic in the Internet", RFC 3714, March
         2004.

   [18]  Chu, J., Dukkipati, N., Cheng, Y., and M. Mathis, "Increasing
         TCP's Initial Window", RFC 6928, April 2013.

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

   [20]  Zanaty, M., "Fairness Considerations for Congestion Control for
         Interactive Real-Time Communication (IRTC)", IAB/ RTF Workshop
         on Congestion Control for Interactive Real-Time Communication,
         July 2012.





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   [21]  Sarker, Z. and I. Johansson, "Improving the Interactive
         Real-Time Video Communication with Network Provided Congestion
         Notification", IAB/IRTF Workshop on Congestion Control for
         Interactive Real-Time Communication, July 2012.

   [22]  Winstein, K., Sivaraman, A., and H. Balakrishnan, "Congestion
         Control for Interactive Real-Time Flows on Today's Internet",
         IAB/IRTF Workshop on Congestion Control for Interactive
         Real-Time Communication, July 2012.

   [23]  Jarvinen, I., Ding, A., Nyrhinen, A., and M. Kojo, "Harsh RED:
         Improving RED for Limited Aggregate Traffic", In Proceedings of
         the 26th IEEE International Conference on Advanced Information
         Networking and Applications (AINA 2012), March 2012.

   [24]  Allman, M., "Comments on Bufferbloat", In ACM SIGCOMM Computer
         Communication Review, Volume 43, Issue 1, pp.  30-37, January
         2013, <http://dl.acm.org/citation.cfm?doid=2427036.2427041>.

   [25]  Bauer, S., Beverly, R., and A. Berger, "Measuring the state of
         ECN readiness in servers, clients,and routers", In Proceedings
         of the 2011 ACM SIGCOMM conference on Internet measurement
         conference (IMC '11), New York, NY, USA, pp. 171-180, February
         2011, <http://dl.acm.org/citation.cfm?doid=2068816.2068833>.

   [26]  Bauer, S., Greenberg, A., Maltz, D., Padhye, J., Patel, P.,
         Prabhakar, B., Sengupta, S., and M. Sridharan, "Data center TCP
         (DCTCP)", In Proceedings of the ACM SIGCOMM 2010 conference
         (SIGCOMM '10), New York, NY, USA, pp.  63-74, August 2010,
         <http://dl.acm.org/citation.cfm?doid=1851182.1851192>.

   [27]  Jarvinen, I., Chemmagate, B., Daniel, L., Ding, A., Kojo, M.,
         and M. Isomaki, "Impact of TCP on Interactive Real- Time
         Communication", IAB/IRTF Workshop on Congestion Control for
         Interactive Real-Time Communication, July 2012.

   [28]  Jennings, C., Nandakumar, S., and H. Phan, "Vendors Considered
         Harmfull", IAB/IRTF Workshop on Congestion Control for
         Interactive Real-Time Communication, July 2012.

   [29]  Welzl, M., "One control to rule them all", IAB/IRTF Workshop on
         Congestion Control for Interactive Real-Time Communication,
         July 2012.

   [30]  Leslie, J., "There is No Magic Transport Wand", IAB/IRTF
         Workshop on Congestion Control for Interactive Real-Time
         Communication, July 2012.




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   [31]  Gettys, J. and J. Gettys, "Bufferbloat: Dark Buffers in the
         Internet", IEEE Internet Computing, Volume 15, Issue 3, pp.
         95-96, May/June 2011.

   [32]  Feng, W., Shin, K., Kandlur, D., and D. Saha, "The BLUE active
         queue management algorithms", In IEEE/ACM Transactions on
         Networking, Volume 10, Issue 4, pp.  513-528, August 2002.

   [33]  IETF, "IP Performance Metrics (ippm) Working Group", January
         2012, <http://datatracker.ietf.org/wg/ippm/charter/>.

   [34]  IETF, "RTP Media Congestion Avoidance Techniques (rmcat)
         Working Group", January 2012,
         <http://datatracker.ietf.org/wg/rmcat/charter/>.

   [35]  IETF, "Active Queue Management and Packet Scheduling (aqm)
         Working Group", September 2013,
         <http://datatracker.ietf.org/wg/aqm/charter/>.

   [36]  Gettys, J. and K. Nichols, "Bufferbloat: Dark Buffers in the
         Internet", Communications of the ACM, Vol. 55, No. 1, pp.
         57-65, January 2012,
         <http://cacm.acm.org/magazines/2012/1/144810-bufferbloat/>.

   [37]  Jacobson, V., "pathchar - a tool to infer characteristics of
         Internet paths", Presented at the Mathematical Sciences
         Research Institute, April 1997,
         <ftp://ftp.ee.lbl.gov/pathchar/msri-talk.pdf>.

   [38]  McKenney, P., "Stochastic Fairness Queuing", In IEEE INFOCOM'90
         Proceedings, Volume 2, pp. 733-740, June 1990.

   [39]  Wikipedia, "Bufferbloat", May 2014, <http://en.wikipedia.org/w/
         index.php?title=Bufferbloat&oldid=608805474>.

   [40]  Wikipedia, "Video compression picture types", March 2014,
         <http://en.wikipedia.org/w/index.php?
         title=Video_compression_picture_types&oldid=602183340>.

   [41]  FCC, "Methodology - Measuring Broadband America July Report
         2012", July 2012, <http://www.fcc.gov/
         measuring-broadband-america/2012/methodology-july-report-2012>.

   [42]  IETF, "lmap -- Large Scale Measurement of Access network
         Performance Mailing List", 2012,
         <https://www.ietf.org/mailman/listinfo/lmap>.





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   [43]  Holmer, S., "On Fairness, Delay and Signaling of Different
         Approaches to Real-time Congestion Control", IAB/IRTF Workshop
         on Congestion Control for Interactive Real-Time Communication,
         July 2012.















































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Appendix A.  Program Committee

   This workshop was organized by Harald Alvestrand, Bernard Aboba, Mary
   Barnes, Gonzalo Camarillo, Spencer Dawkins, Lars Eggert, Matthew
   Ford, Randell Jesup, Cullen Jennings, Jon Peterson, Robert Sparks,
   and Hannes Tschofenig.

Appendix B.  Workshop Material

   o  Main Workshop Page:
      http://www.iab.org/activities/workshops/cc-workshop/

   o  Position Papers:
      http://www.iab.org/activities/workshops/cc-workshop/papers/

   o  Slides:
      http://www.iab.org/activities/workshops/cc-workshop/slides/

Appendix C.  Accepted Position Papers

   1.   "One control to rule them all" by Michael Welzl

   2.   "Congestion Avoidance Through Deterministic" by Pier Luca
        Montessoro, Riccardo Bernardini, Franco Blanchini, Daniele
        Casagrande, Mirko Loghi, and Stefan Wieser

   3.   "Congestion Control in Real Time Media - Context" by Harald
        Alvestrand

   4.   "Improving the Interactive Real-Time Video Communication with
        Network Provided Congestion Notification" by ANM Zaheduzzaman
        Sarker and Ingemar Johansson

   5.   "Multiparty Requirements in Congestion Control for Real-Time
        Interactive Media" by Magnus Westerlund

   6.   "On Fairness, Delay and Signaling of Different Approaches to
        Real-time Congestion Control" by Stefan Holmer

   7.   "RTP Congestion Control and RTCWEB Application Feedback" by Ted
        Hardie

   8.   "Issues with Using Packet Delays and Inter-arrival Times for
        Inference of Internet Congestion" by Wesley M.  Eddy

   9.   "Impact of TCP on Interactive Real-Time Communication" by Ilpo
        Jarvinen, Binoy Chemmagate, Laila Daniel, Aaron Yi Ding, Markku
        Kojo, and Markus Isomaki



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   10.  "Security Concerns For RTCWEB Congestion Control" by Dan York

   11.  "Vendors Considered Harmfull" by Cullen Jennings, Suhas
        Nandakumar, and Hein Phan

   12.  "Network-Assisted Dynamic Adaptation" by Xiaoqing Zhu and Rong
        Pan

   13.  "Congestion Control for Interactive Real-Time Applications" by
        Sanjeev Mehrotra and Jin Li

   14.  "There is No Magic Transport Wand" by John Leslie

   15.  "Towards Adaptive Congestion Management for Interactive Real-
        Time Communications" by Dirk Kutscher and Miriam Kuehlewind

   16.  "Enlarge the pre-congestion spectrum usage?" by Xavier Marjou
        and Emile Stephan

   17.  "Congestion control for users who don't have first-class
        internet access" by Maire Reavy

   18.  "Realtime Congestion Challenges" by Randell Jesup

   19.  "Congestion Control for Interactive Media: Control Loops & APIs"
        by Varun Singh, Joerg Ott, and Colin Perkins

   20.  "Some Notes on Threat Modelling Congestion Management" by Eric
        Rescorla

   21.  "Timely Detection of Lost Packets" by Ali C.  Begen

   22.  "Congestion Control Considerations for Data Channels" by Michael
        Tuexen

   23.  "Position paper on CC for Interactive RT" by Matt Mathis

   24.  "Overall Considerations for Congestion Control" by Mo Zanaty,
        Bill VerSteeg, Bent Christensen, David Benham, and Allyn Romanow

   25.  "Fairness Considerations for Congestion Control" by Mo Zanaty

   26.  "Media is not Data: The Meaning of Fairness for Competing
        Multimedia Flows" by Timothy B.  Terriberry

   27.  "Thoughts on Real-Time Congestion Control" by Murari Sridharan





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   28.  "Congestion Control for Interactive Real-Time Flows on Today's
        Internet" by Keith Winstein, Anirudh Sivaraman, and Hari
        Balakrishnan

   29.  "Congestion Control Principles for CoAP" by Carsten Bormann and
        Klaus Hartke

   30.  "Erasure Coding and Congestion Control for Interactive Real-Time
        Communication" by Pierre-Ugo Tournoux, Tuan Tran Thai, Emmanuel
        Lochin, Jerome Lacan, and Vincent Roca

   31.  "Video Conferencing Specific Considerations for RTP Congestion
        Control" by Stephen Botzko and Mary Barnes

   32.  "The Internet is Broken, and How to Fix It" by Jim Gettys

   33.  "Deployment Considerations for Congestion Control in Real-Time
        Interactive Media Systems" by Jari Arkko

Appendix D.  Workshop Participants

   We would like to thank the following workshop participants for
   attending the workshop:

   o  Mat Ford
   o  Bernard Aboba
   o  Alissa Cooper
   o  Mary Barnes
   o  Lars Eggert
   o  Harald Alvestrand
   o  Gonzalo Camarillo
   o  Robert Sparks
   o  Cullen Jennings
   o  Dirk Kutscher
   o  Carsten Bormann
   o  Michael Welzl
   o  Magnus Westerlund
   o  Colin Perkins
   o  Murari Sridharan
   o  Klaus Hartke
   o  Pier Luca Montessoro
   o  Xavier Marjou
   o  Vincent Roca
   o  Wes Eddy
   o  Ali C.  Begen
   o  Mo Zanaty
   o  Jin Li
   o  Dave Thaler



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   o  Bob Briscoe
   o  Barry Leiba
   o  Jari Arkko
   o  Stewart Bryant
   o  Martin Stiemerling
   o  Russ Housley
   o  Marc Blanchet
   o  Zaheduzzaman Sarker
   o  Xiaoqing Zhu
   o  Randell Jesup
   o  Eric Rescorla
   o  Suhas Nandakumar
   o  Hannes Tschofenig
   o  Bill VerSteeg
   o  Sean Turner
   o  Keith Winstein
   o  Jon Peterson
   o  Maire Reavy
   o  Michael Tuexen
   o  Stefan Holmer
   o  Joerg Ott
   o  Timothy Terriberry
   o  Benoit Claise
   o  Ted Hardie
   o  Stephen Botzko
   o  Matt Mathis
   o  David Benham
   o  Jim Gettys
   o  Spencer Dawkins
   o  Sanjeev Mehrotra
   o  Adrian Farrel
   o  Greg White
   o  Markku Kojo

   We also had remote participants, namely:

   o  Emmanuel Lochin
   o  Mark Handley
   o  Anirudh Sivaraman
   o  John Leslie
   o  Varun Singh










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Authors' Addresses

   Hannes Tschofenig
   Hall, Tirol  6060
   Austria

   EMail: Hannes.Tschofenig@gmx.net
   URI:   http://www.tschofenig.priv.at


   Lars Eggert
   Sonnenallee 1
   Kirchheim  85551
   Germany

   Phone: +49 151 12055791
   EMail: lars@netapp.com
   URI:   http://eggert.org/


   Zaheduzzaman Sarker
   Lulea  SE-971 28
   Sweden

   Phone: +46 10 717 37 43
   EMail: zaheduzzaman.sarker@ericsson.com

























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