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authorThomas Voss <mail@thomasvoss.com> 2024-11-27 20:54:24 +0100
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+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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+Tschofenig, et al. Informational [Page 1]
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+RFC 7295 Congestion Control Workshop Report July 2014
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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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+RFC 7295 Congestion Control Workshop Report July 2014
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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
+
+
+
+
+
+Tschofenig, 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 2014
+
+
+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
+
+
+
+
+Tschofenig, 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 needed
+
+
+
+
+Tschofenig, 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", 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.
+
+
+
+
+
+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 2014
+
+
+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
+
+
+
+Tschofenig, 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 Sridharan
+
+
+
+
+
+Tschofenig, 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 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
+
+
+
+Tschofenig, 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 Singh
+
+
+
+
+
+
+
+
+
+
+Tschofenig, et al. Informational [Page 25]
+
+RFC 7295 Congestion Control Workshop Report July 2014
+
+
+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
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+Tschofenig, et al. Informational [Page 26]
+