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INFORMATIONAL
Internet Architecture Board (IAB)                          H. TschofenigRequest for Comments: 7295                                     L. EggertCategory: Informational                                        Z. SarkerISSN: 2070-1721                                                July 2014Report from the IAB/IRTF Workshop on Congestion Controlfor Interactive Real-Time CommunicationAbstract   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; seeSection 2 of RFC 5741.   Information about the current status of this document, any errata,   and how to provide feedback on it may be obtained athttp://www.rfc-editor.org/info/rfc7295.Tschofenig, et al.            Informational                     [Page 1]

RFC 7295           Congestion Control Workshop Report          July 2014Copyright Notice   Copyright (c) 2014 IETF Trust and the persons identified as the   document authors.  All rights reserved.   This document is subject toBCP 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 Contents1.  Introduction  . . . . . . . . . . . . . . . . . . . . . . . .32.  Workshop Structure  . . . . . . . . . . . . . . . . . . . . .52.1.  History and Current Challenges  . . . . . . . . . . . . .52.2.  Simulations and Measurements  . . . . . . . . . . . . . .82.3.  Design Aspects of Problems and Solutions  . . . . . . . .93.  Recommendations . . . . . . . . . . . . . . . . . . . . . . .133.1.  Changes to Network Infrastructure . . . . . . . . . . . .143.2.  Avoiding Self-Inflicted Queuing . . . . . . . . . . . . .154.  Security Considerations . . . . . . . . . . . . . . . . . . .175.  Acknowledgments . . . . . . . . . . . . . . . . . . . . . . .176.  Informative References  . . . . . . . . . . . . . . . . . . .17Appendix A.  Program Committee  . . . . . . . . . . . . . . . . .22Appendix B.  Workshop Material  . . . . . . . . . . . . . . . . .22Appendix C.  Accepted Position Papers . . . . . . . . . . . . . .22Appendix D.  Workshop Participants  . . . . . . . . . . . . . . .24Tschofenig, et al.            Informational                     [Page 2]

RFC 7295           Congestion Control Workshop Report          July 20141.  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 inRFC 2914 [2] andRFC 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.Tschofenig, et al.            Informational                     [Page 3]

RFC 7295           Congestion Control Workshop Report          July 2014   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.Tschofenig, et al.            Informational                     [Page 4]

RFC 7295           Congestion Control Workshop Report          July 20142.  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 timeTschofenig, et al.            Informational                     [Page 5]

RFC 7295           Congestion Control Workshop Report          July 2014   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.Tschofenig, et al.            Informational                     [Page 6]

RFC 7295           Congestion Control Workshop Report          July 2014   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.Tschofenig, et al.            Informational                     [Page 7]

RFC 7295           Congestion Control Workshop Report          July 20142.2.  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, althoughRFC 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 TCPTschofenig, et al.            Informational                     [Page 8]

RFC 7295           Congestion Control Workshop Report          July 2014   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.Tschofenig, et al.            Informational                     [Page 9]

RFC 7295           Congestion Control Workshop Report          July 2014   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 youTschofenig, et al.            Informational                    [Page 10]

RFC 7295           Congestion Control Workshop Report          July 2014   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?Tschofenig, et al.            Informational                    [Page 11]

RFC 7295           Congestion Control Workshop Report          July 2014   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 sessionTschofenig, et al.            Informational                    [Page 12]

RFC 7295           Congestion Control Workshop Report          July 2014   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.Tschofenig, et al.            Informational                    [Page 13]

RFC 7295           Congestion Control Workshop Report          July 20143.1.  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 inSection 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 withTschofenig, et al.            Informational                    [Page 14]

RFC 7295           Congestion Control Workshop Report          July 2014   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 neededTschofenig, et al.            Informational                    [Page 15]

RFC 7295           Congestion Control Workshop Report          July 2014      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,Tschofenig, et al.            Informational                    [Page 16]

RFC 7295           Congestion Control Workshop Report          July 2014      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.Tschofenig, et al.            Informational                    [Page 17]

RFC 7295           Congestion Control Workshop Report          July 2014   [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",RFC5348, 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.Tschofenig, et al.            Informational                    [Page 18]

RFC 7295           Congestion Control Workshop Report          July 2014   [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.Tschofenig, et al.            Informational                    [Page 19]

RFC 7295           Congestion Control Workshop Report          July 2014   [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>.Tschofenig, et al.            Informational                    [Page 20]

RFC 7295           Congestion Control Workshop Report          July 2014   [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.Tschofenig, et al.            Informational                    [Page 21]

RFC 7295           Congestion Control Workshop Report          July 2014Appendix 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 IsomakiTschofenig, et al.            Informational                    [Page 22]

RFC 7295           Congestion Control Workshop Report          July 2014   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 SridharanTschofenig, et al.            Informational                    [Page 23]

RFC 7295           Congestion Control Workshop Report          July 2014   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 ArkkoAppendix 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 ThalerTschofenig, et al.            Informational                    [Page 24]

RFC 7295           Congestion Control Workshop Report          July 2014   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 SinghTschofenig, et al.            Informational                    [Page 25]

RFC 7295           Congestion Control Workshop Report          July 2014Authors' 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.comTschofenig, et al.            Informational                    [Page 26]

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