This document describes two methods of congestion control when using real-time communications on the World Wide Web (RTCWEB); one delay- based and one loss-based. It is published as an input document to the RMCAT working group on congestion control for media streams. The mailing list of that working group is rmcat@ietf.org.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [RFC2119].
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1. Introduction·····························································3 1.1. Mathematical notation conventions····························3 2. System model·························································4 3. Feedback and extensions··········································4 4. Sending Engine·······················································5 5. Delay-based control··················································5 5.1. Arrival-time model··················································6 5.2. Pre-filtering···························································7 5.3. Arrival-time filter····················································7 5.4. Over-use detector·················································8 5.5. Rate control·······················································10 5.6. Parameters settings·············································12 6. Loss-based control·················································13 7. Interoperability Considerations··································14 8. Implementation Experience······································14 9. Further Work·························································15 10. IANA Considerations·············································15 11. Security Considerations·········································15 12. Acknowledgements···············································15 13. References··························································16 13.1. Normative References·········································16 13.2. Informative References········································16 Appendix A. Change log··············································16 A.1. Version -00 to -01················································16 A.2. Version -01 to -02················································17 A.3. Version -02 to -03················································17 A.4. rtcweb-03 to rmcat-00···········································17 A.5. rmcat -00 to -01···················································17 A.6. rmcat -01 to -02···················································17 A.7. rmcat -02 to -03···················································18 A.8. ietf-rmcat -00 to ietf-rmcat -01·································18 A.9. ietf-rmcat -01 to ietf-rmcat -02·································18 Authors' Addresses·····················································18
1. Introduction 简介
Congestion control is a requirement for all applications sharing the Internet resources [RFC2914].
Congestion control for real-time media is challenging for a number of reasons:
拥塞控制是针对所有共享Internet网络资源的应用程序的一种要求[RFC2914]。
针对实时流媒体的拥塞控制面临一些挑战,原因如下:
o The media is usually encoded in forms that cannot be quickly changed to accommodate varying bandwidth, and bandwidth requirements can often be changed only in discrete, rather large steps
o The participants may have certain specific wishes on how to respond - which may not be reducing the bandwidth required by the flow on which congestion is discovered
o The encodings are usually sensitive to packet loss, while the real-time requirement precludes the repair of packet loss by retransmission
o 流媒体通常形式的编码无法针对不同带宽做出快速改变,流媒体需要的带宽通常是离散的、甚至是大跨度的变化
o 当网络拥塞发生时,共享网络的其他程序有可能不会减小带宽。
o 编码(输出的数据)通常对数据包丢失很敏感,然而实时性的要求有时会阻止通过重传恢复丢失的数据包。
This memo describes two congestion control algorithms that together are able to provide good performance and reasonable bandwidth sharing with other video flows using the same congestion control and with TCP flows that share the same links.
The signaling used consists of experimental RTP header extensions and RTCP messages RFC 3550 [RFC3550] as defined in [abs-send-time], [I-D.alvestrand-rmcat-remb] and [I-D.holmer-rmcat-transport-wide-cc-extensions].
The mathematics of this document have been transcribed from a more formula-friendly format.
The following notational conventions are used:
X_hat An estimate of the true value of variable X - conventionally marked by a circumflex accent on top of the variable name.
X(i) The "i"th value of vector X - conventionally marked by a subscript i.
E{X} The expected value of the stochastic variable X
本文的数学内容是从一种更为公式友好的格式转录而来的。
使用以下符号约定:
X_hat 变量X的真实值的估计值 - 通常在变量名的顶部用符号标记。
X(i) 数组X的第i个值-通常用下标i标记。
E{X} 随机变量X的期望值
2. System model 系统模型
The following elements are in the system: 本系统中包含下列元素:
o RTP packet - an RTP packet containing media data.
o Group of packets - a set of RTP packets transmitted from the sender uniquely identified by the group departure and group arrival time (absolute send time) [abs-send-time]. These could be video packets, audio packets, or a mix of audio and video packets.
o Incoming media stream - a stream of frames consisting of RTP packets.
o RTP sender - sends the RTP stream over the network to the RTP receiver. It generates the RTP timestamp and the abs-send-time header extension
o RTP receiver - receives the RTP stream, marks the time of arrival.
o RTP数据包 - 包含流媒体数据的RTP数据包。
o 数据包组 - (连续)RTP数据包构成的集合,通过数据包组的发送时间和到达时间(绝对发送时间)[abs-send-time]可以唯一标识这个由发送方发送的RTP数据包组。这里的RTP包可以是视频包、音频包或音频和视频包的混合。
o 传入媒体流 - 由RTP数据包组成的帧流。
o RTP发送方 - 通过网络向RTP接收方发送RTP流。RTP发送方负责生成RTP时间戳以及abs-send-time头扩展
o RTP接收器 - 接收RTP流,并记录RTP报文的到达时间。
o RTCP sender at RTP receiver - sends receiver reports, REMB messages and transport-wide RTCP feedback messages.
o RTCP receiver at RTP sender - receives receiver reports and REMB messages and transport-wide RTCP feedback messages, reports these to the sender side controller.
o RTCP receiver at RTP receiver, receives sender reports from the sender.
o Loss-based controller - takes loss rate measurement, round trip time measurement and REMB messages, and computes a target sending bitrate.
o Delay-based controller - takes the packet arrival info, either at the RTP receiver, or from the feedback received by the RTP sender, and computes a maximum bitrate which it passes to the loss-based controller.
o RTP接收方同时作为RTCP的发送方 - 负责发送RR报文、REMB报文和传输层RTCP反馈报文(transport wide feedback packet)。
o RTP发送方同时作为RTCP接收方 - 负责接收RR报文、REMB报文、传输层RTCP反馈报文(transport wide feedback packet),并将接收信息通知到本方控制器。
o RTP接收方同时作为RTCP的接收方,负责接收发送方生成的SR报文。
o 基于丢包的控制器 - 进行丢失率测量、RTT时间测量和(接收)REMB消息,并计算发送端的目标发送比特率。
o 基于延迟的控制器 - 可以部署在RTP接收端记录RTP报文的到达时间或部署在RTP发送端从接收端的反馈报文中获取数据包的到达时间,计算最大比特率并传递给基于丢包的控制器。
Together, loss-based controller and delay-based controller implement the congestion control algorithm.
基于丢包的控制器和基于延迟的控制器共同实现了拥塞控制算法。
3. Feedback and extensions 反馈和扩展
There are two ways to implement the proposed algorithm. One where both the controllers are running at the send-side, and one where the delay-based controller runs on the receive-side and the loss-based controller runs on the send-side.
The first version can be realized by using a per-packet feedback protocol as described in [I-D.holmer-rmcat-transport-wide-cc-extensions]. Here, the RTP receiver will record the arrival time and the transport-wide sequence number of each received packet, which will be sent back to the sender periodically using the transport-wide feedback message. The RECOMMENDED feedback interval is once per received video frame or at least once every 30 ms if audio-only or multi-stream. If the feedback overhead needs to be limited this interval can be increased to 100 ms.
The sender will map the received {sequence number, arrival time} pairs to the send-time of each packet covered by the feedback report, and feed those timestamps to the delay-based controller. It will also compute a loss ratio based on the sequence numbers in the feedback message.
The second version can be realized by having a delay-based controller at the receive-side, monitoring and processing the arrival time and size of incoming packets. The sender SHOULD use the abs-send-time RTP header extension [abs-send-time] to enable the receiver to compute the inter-group delay variation. The output from the delay- based controller will be a bitrate, which will be sent back to the sender using the REMB feedback message [I-D.alvestrand-rmcat-remb]. The packet loss ratio is sent back via RTCP receiver reports. At the sender the bitrate in the REMB message and the fraction of packets lost are fed into the loss-based controller, which outputs a final target bitrate. It is RECOMMENDED to send the REMB message as soon as congestion is detected, and otherwise at least once every second.
Pacing is used to actuate the target bitrate computed by the controllers.
When media encoder produces data, this is fed into a Pacer queue. The Pacer sends a group of packets to the network every burst_time interval. RECOMMENDED value for burst_time is 5 ms. The size of a group of packets is computed as the product between the target bitrate and the burst_time.
The delay-based control algorithm can be further decomposed into four parts: a pre-filtering, an arrival-time filter, an over-use detector, and a rate controller.
基于延迟的控制算法可以进一步分解为四个部分:预滤波、到达时间滤波器、过载检测器和速率控制器。
5.1. Arrival-time model 到达时间模型
This section describes an adaptive filter that continuously updates estimates of network parameters based on the timing of the received groups of packets.
We define the inter-arrival time, t(i) - t(i-1), as the difference in arrival time of two groups of packets. Correspondingly, the inter- departure time, T(i) - T(i-1), is defined as the difference in departure-time of two groups of packets. Finally, the inter-group delay variation, d(i), is defined as the difference between the inter-arrival time and the inter-departure time. Or interpreted differently, as the difference between the delay of group i and group i-1.
d
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)
d(i) = t(i) - t(i-1) - (T(i) - T(i-1))
d(i)=t(i)−t(i−1)−(T(i)−T(i−1))
An inter-departure time is computed between consecutive groups as T(i) - T(i-1), where T(i) is the departure timestamp of the last packet in the current packet group being processed. Any packets received out of order are ignored by the arrival-time model.
Each group is assigned a receive time t(i), which corresponds to the time at which the last packet of the group was received. A group is delayed relative to its predecessor if t(i) - t(i-1) > T(i) - T(i-1), i.e., if the inter-arrival time is larger than the inter-departure time.
We can model the inter-group delay variation as: d(i) = w(i)
Here, w(i) is a sample from a stochastic process W, which is a function of the link capacity, the current cross traffic, and the current sent bitrate. We model W as a white Gaussian process. If we are over-using the channel we expect the mean of w(i) to increase, and if a queue on the network path is being emptied, the mean of w(i) will decrease; otherwise the mean of w(i) will be zero.
Breaking out the mean, m(i), from w(i) to make the process zero mean, we get
Equation 1 : d(i) = m(i) + v(i)
The noise term v(i) represents network jitter and other delay effects not captured by the model.
从w(i)中取平均值m(i),使过程的平均值为零,我们得到
方程式1: d(i) = m(i) + v(i)
噪声项v(i)表示网络抖动和模型未捕获的其它影响延迟的因素。
5.2. Pre-filtering 预滤波
The pre-filtering aims at handling delay transients caused by channel outages. During an outage, packets being queued in network buffers, for reasons unrelated to congestion, are delivered in a burst when the outage ends.
The pre-filtering merges together groups of packets that arrive in a burst. Packets are merged in the same group if one of these two conditions holds:
o A sequence of packets which are sent within a burst_time interval constitute a group.
o A Packet which has an inter-arrival time less than burst_time and an inter-group delay variation d(i) less than 0 is considered being part of the current group of packets.
The parameter d(i) is readily available for each group of packets, i > 1. We want to estimate m(i) and use this estimate to detect whether or not the bottleneck link is over-used. The parameter can be estimated by any adaptive filter - we are using the Kalman filter.
Let m(i) be the estimate at time i
We model the state evolution from time i to time i+1 as
m(i+1) = m(i) + u(i)
where u(i) is the state noise that we model as a stationary process with Gaussian statistic with zero mean and variance
where f_max = max {1/(T(j) - T(j-1))} for j in i-K+1,...,i is the highest rate at which the last K packet groups have been received and chi is a filter coefficient typically chosen as a number in the interval [0.1, 0.001]. Since our assumption that v(i) should be zero mean WGN is less accurate in some cases, we have introduced an additional outlier filter around the updates of var_v_hat. If z(i) > 3*sqrt(var_v_hat) the filter is updated with 3*sqrt(var_v_hat) rather than z(i). For instance v(i) will not be white in situations where packets are sent at a higher rate than the channel capacity, in which case they will be queued behind each other.
The inter-group delay variation estimate m(i), obtained as the output of the arrival-time filter, is compared with a threshold del_var_th(i). An estimate above the threshold is considered as an indication of over-use. Such an indication is not enough for the detector to signal over-use to the rate control subsystem. A definitive over-use will be signaled only if over-use has been detected for at least overuse_time_th milliseconds. However, if m(i) < m(i-1), over-use will not be signaled even if all the above conditions are met. Similarly, the opposite state, under-use, is detected when m(i) < -del_var_th(i). If neither over-use nor under- use is detected, the detector will be in the normal state.
The threshold del_var_th has a remarkable impact on the overall dynamics and performance of the algorithm. In particular, it has been shown that using a static threshold del_var_th, a flow controlled by the proposed algorithm can be starved by a concurrent TCP flow [Pv13]. This starvation can be avoided by increasing the threshold del_var_th to a sufficiently large value.
The reason is that, by using a larger value of del_var_th, a larger queuing delay can be tolerated, whereas with a small del_var_th, the over-use detector quickly reacts to a small increase in the offset estimate m(i) by generating an over-use signal that reduces the delay-based estimate of the available bandwidth A_hat (see Section 4.4). Thus, it is necessary to dynamically tune the threshold del_var_th to get good performance in the most common scenarios, such as when competing with loss-based flows.
For this reason, we propose to vary the threshold del_var_th(i) according to the following dynamic equation:
with K(i)=K_d if |m(i)| < del_var_th(i-1) or K(i)=K_u otherwise. The rationale is to increase del_var_th(i) when m(i) is outside of the range [-del_var_th(i-1),del_var_th(i-1)], whereas, when the offset estimate m(i) falls back into the range, del_var_th is decreased. In this way when m(i) increases, for instance due to a TCP flow entering the same bottleneck, del_var_th(i) increases and avoids the uncontrolled generation of over-use signals which may lead to starvation of the flow controlled by the proposed algorithm [Pv13]. Moreover, del_var_th(i) SHOULD NOT be updated if this condition holds:
|m(i)| - del_var_th(i) > 15
It is also RECOMMENDED to clamp del_var_th(i) to the range [6, 600], since a too small del_var_th(i) can cause the detector to become overly sensitive.
On the other hand, when m(i) falls back into the range [-del_var_th(i-1),del_var_th(i-1)] the threshold del_var_th(i) is decreased so that a lower queuing delay can be achieved.
It is RECOMMENDED to choose K_u > K_d so that the rate at which del_var_th is increased is higher than the rate at which it is decreased. With this setting it is possible to increase the threshold in the case of a concurrent TCP flow and prevent starvation as well as enforcing intra-protocol fairness. RECOMMENDED values for del_var_th(0), overuse_time_th, K_u and K_d are respectively 12.5 ms, 10 ms, 0.01 and 0.00018.
The rate control is split in two parts, one controlling the bandwidth estimate based on delay, and one controlling the bandwidth estimate based on loss. Both are designed to increase the estimate of the available bandwidth A_hat as long as there is no detected congestion and to ensure that we will eventually match the available bandwidth of the channel and detect an over-use.
As soon as over-use has been detected, the available bandwidth estimated by the delay-based controller is decreased. In this way we get a recursive and adaptive estimate of the available bandwidth.
In this document we make the assumption that the rate control subsystem is executed periodically and that this period is constant.
The rate control subsystem has 3 states: Increase, Decrease and Hold. "Increase" is the state when no congestion is detected; "Decrease" is the state where congestion is detected, and "Hold" is a state that waits until built-up queues have drained before going to "increase" state.
The state transitions (with blank fields meaning "remain in state") are: 【信号触发】状态迁移规则表(空白表示“停留在【之前的】状态”):
Signal \ State
Hold
Increase
Decrease
Over-use
Decrease
Decrease
Normal
Increase
Hold
Under-use
Hold
Hold
信号 \ 状态
保持
增加
减少
过载
减少
减少
正常
增加
保持
轻载
保持
保持
The subsystem starts in the increase state, where it will stay until over-use or under-use has been detected by the detector subsystem. On every update the delay-based estimate of the available bandwidth is increased, either multiplicatively or additively, depending on its current state.
The system does a multiplicative increase if the current bandwidth estimate appears to be far from convergence, while it does an additive increase if it appears to be closer to convergence. We assume that we are close to convergence if the currently incoming bitrate, R_hat(i), is close to an average of the incoming bitrates at the time when we previously have been in the Decrease state. "Close" is defined as three standard deviations around this average. It is RECOMMENDED to measure this average and standard deviation with an exponential moving average with the smoothing factor 0.95, as it is expected that this average covers multiple occasions at which we are in the Decrease state. Whenever valid estimates of these statistics are not available, we assume that we have not yet come close to convergence and therefore remain in the multiplicative increase state.
If R_hat(i) increases above three standard deviations of the average max bitrate, we assume that the current congestion level has changed, at which point we reset the average max bitrate and go back to the multiplicative increase state.
R_hat(i) is the incoming bitrate measured by the delay-based controller over a T seconds window:
R_hat(i) = 1/T * sum(L(j)) for j from 1 to N(i)
N(i) is the number of packets received the past T seconds and L(j) is the payload size of packet j. A window between 0.5 and 1 second is RECOMMENDED.
During multiplicative increase, the estimate is increased by at most 8% per second.
eta = 1.08^min(time_since_last_update_ms / 1000, 1.0) A_hat(i) = eta * A_hat(i-1)
During the additive increase the estimate is increased with at most half a packet per response_time interval. The response_time interval is estimated as the round-trip time plus 100 ms as an estimate of over-use estimator and detector reaction time.
在乘法增加期间,估计值每秒最多被增加8%。
eta = 1.08^min(time_since_last_update_ms / 1000, 1.0) A_hat(i) = eta * A_hat(i-1)
expected_packet_size_bits is used to get a slightly slower slope for the additive increase at lower bitrates. It can for instance be computed from the current bitrate by assuming a frame rate of 30 frames per second:
Since the system depends on over-using the channel to verify the current available bandwidth estimate, we must make sure that our estimate does not diverge from the rate at which the sender is actually sending. Thus, if the sender is unable to produce a bit stream with the bitrate the congestion controller is asking for, the available bandwidth estimate should stay within a given bound. Therefore we introduce a threshold
When an over-use is detected the system transitions to the decrease state, where the delay-based available bandwidth estimate is decreased to a factor times the currently incoming bitrate.
A_hat(i) = beta * R_hat(i)
beta is typically chosen to be in the interval [0.8, 0.95], 0.85 is the RECOMMENDED value.
When the detector signals under-use to the rate control subsystem, we know that queues in the network path are being emptied, indicating that our available bandwidth estimate A_hat is lower than the actual available bandwidth. Upon that signal the rate control subsystem will enter the hold state, where the receive-side available bandwidth estimate will be held constant while waiting for the queues to stabilize at a lower level - a way of keeping the delay as low as possible. This decrease of delay is wanted, and expected, immediately after the estimate has been reduced due to over-use, but can also happen if the cross traffic over some links is reduced.
It is RECOMMENDED that the routine to update A_hat(i) is run at least once every response_time interval.
Time limit in milliseconds between packet bursts which identifies a group
5 ms
q
State noise covariance matrix
q = 10^-3
e(0)
Initial value of the system error covariance
e(0) = 0.1
chi
Coefficient used for the measured noise variance
[0.1, 0.001]
del_var_th(0)
Initial value for the adaptive threshold
12.5 ms
overuse_time_th
Time required to trigger an overuse signal
10 ms
K_u
Coefficient for the adaptive threshold
0.01
K_d
Coefficient for the adaptive threshold
0.00018
T
Time window for measuring the received bitrate
[0.5, 1] s
beta
Decrease rate factor
0.85
Table 1: RECOMMENDED values for delay based controller
参数
描述
推荐值
突发时间
标识组的数据包突发之间的时间限制(以毫秒为单位)
5 ms
q
状态噪声协方差矩阵
q = 10^-3
e(0)
系统误差协方差的初始值
e(0) = 0.1
chi
用于测量噪声方差的系数
[0.1, 0.001]
del_var_th(0)
自适应阈值的初始值
12.5 ms
overuse_time_th
触发过载信号所需的时间
10 ms
K_u
自适应阈值系数
0.01
K_d
自适应阈值系数
0.00018
T
用于测量接收比特率的时间窗口
[0.5, 1] s
beta
递减率系数
0.85
表1: 基于延时的控制器参数的推荐者
6. Loss-based control 基于丢包的(码率)控制
A second part of the congestion controller bases its decisions on the round-trip time, packet loss and available bandwidth estimates A_hat received from the delay-based controller. The available bandwidth estimates computed by the loss-based controller are denoted with As_hat.
The available bandwidth estimates A_hat produced by the delay-based controller are only reliable when the size of the queues along the path sufficiently large. If the queues are very short, over-use will only be visible through packet losses, which are not used by the delay-based controller.
The loss-based controller SHOULD run every time feedback from the receiver is received.
o If 2-10% of the packets have been lost since the previous report from the receiver, the sender available bandwidth estimate As_hat(i) will be kept unchanged.
o If more than 10% of the packets have been lost a new estimate is calculated as As_hat(i) = As_hat(i-1)(1-0.5p), where p is the loss ratio.
o As long as less than 2% of the packets have been lost As_hat(i) will be increased as As_hat(i) = 1.05(As_hat(i-1))
o 如果自上一次来自接收方的报告以来有2-10%的数据包丢失,则发送方可用带宽估计值As_hat(i)将保持不变。
o 如果超过10%的数据包丢失,则新的估计值计算为As_hat(i) =As_hat(i-1)(1-0.5p),其中p是丢失率。
o 只要丢失的数据包不到2%,那么将增加As_hat(i) =1.05(As_hat(i-1))
The loss-based estimate As_hat is compared with the delay-based estimate A_hat. The actual sending rate is set as the minimum between As_hat and A_hat.
We motivate the packet loss thresholds by noting that if the transmission channel has a small amount of packet loss due to over- use, that amount will soon increase if the sender does not adjust his bitrate. Therefore we will soon enough reach above the 10% threshold and adjust As_hat(i). However, if the packet loss ratio does not increase, the losses are probably not related to self-inflicted congestion and therefore we should not react on them.
In case a sender implementing these algorithms talks to a receiver which do not implement any of the proposed RTCP messages and RTP header extensions, it is suggested that the sender monitors RTCP receiver reports and uses the fraction of lost packets and the round- trip time as input to the loss-based controller. The delay-based controller should be left disabled.
This algorithm has been implemented in the open-source WebRTC project, has been in use in Chrome since M23, and is being used by Google Hangouts.
Deployment of the algorithm have revealed problems related to, e.g, congested or otherwise problematic WiFi networks, which have led to algorithm improvements. The algorithm has also been tested in a multi-party conference scenario with a conference server which terminates the congestion control between endpoints. This ensures that no assumptions are being made by the congestion control about maximum send and receive bitrates, etc., which typically is out of control for a conference server.
This draft is offered as input to the congestion control discussion.
Work that can be done on this basis includes:
o Considerations of integrated loss control: How loss and delay control can be better integrated, and the loss control improved. o Considerations of locus of control: evaluate the performance of having all congestion control logic at the sender, compared to splitting logic between sender and receiver. o Considerations of utilizing ECN as a signal for congestion estimation and link over-use detection.
提交本草案作为拥塞控制讨论的输入。 在此基础上可以完成的工作包括: o 综合损失控制的考虑因素:如何更好地整合损失和延迟控制,并改进损失控制。 o 对控制点的考虑:对于拥塞控制逻辑集中部署在发送方和塞控制逻辑分置部署在发送方和接收方,需要比较并评估这两种部署方式的性能差异。 o 考虑将ECN用作拥塞估计和链路过载的检测信号。
10. IANA Considerations IANA 考虑
This document makes no request of IANA. Note to RFC Editor: this section may be removed on publication as an RFC.
本文件对于IANA无任何要求。 注意RFC编辑器:本文在RFC发布时本节可能会被删除。
11. Security Considerations 安全考虑
An attacker with the ability to insert or remove messages on the connection would have the ability to disrupt rate control. This could make the algorithm to produce either a sending rate under- utilizing the bottleneck link capacity, or a too high sending rate causing network congestion.
In this case, the control information is carried inside RTP, and can be protected against modification or message insertion using SRTP, just as for the media. Given that timestamps are carried in the RTP header, which is not encrypted, this is not protected against disclosure, but it seems hard to mount an attack based on timing information only.
Thanks to Randell Jesup, Magnus Westerlund, Varun Singh, Tim Panton, Soo-Hyun Choo, Jim Gettys, Ingemar Johansson, Michael Welzl and others for providing valuable feedback on earlier versions of this draft.
[I-D.alvestrand-rmcat-remb] Alvestrand, H., "RTCP message for Receiver Estimated Maximum Bitrate", draft-alvestrand-rmcat-remb-03 (work in progress), October 2013.
[I-D.holmer-rmcat-transport-wide-cc-extensions] Holmer, S., Flodman, M., and E. Sprang, "RTP Extensions for Transport-wide Congestion Control", draft-holmer- rmcat-transport-wide-cc-extensions-00 (work in progress), March 2015.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003.
[abs-send-time] "RTP Header Extension for Absolute Sender Time",
13.2. Informative References 资料性引用
[Pv13] De Cicco, L., Carlucci, G., and S. Mascolo, "Understanding the Dynamic Behaviour of the Google Congestion Control", Packet Video Workshop , December 2013.
[RFC2914] Floyd, S., "Congestion Control Principles", BCP 41, RFC 2914, September 2000.
Appendix A. Change log 附录A. 修订记录
A.1. Version -00 to -01
o Added change log o Added appendix outlining new extensions o Added a section on when to send feedback to the end of section 3.3 "Rate control", and defined min/max FB intervals. o Added size of over-bandwidth estimate usage to "further work" section. o Added startup considerations to "further work" section. o Added sender-delay considerations to "further work" section. o Filled in acknowledgments section from mailing list discussion.
o 添加了更改日志 o 增加了概述新扩展的附录 o 在第3.3节“速率控制”末尾增加了一节,说明何时发送反馈,并定义了最小/最大FB间隔。 o 在“进一步工作”部分增加了带宽估计使用量的大小。 o 在“进一步工作”部分增加了启动注意事项。 o 在“进一步工作”部分增加了发送方延迟注意事项。 o 填写邮件列表讨论中的确认部分。
A.2. Version -01 to -02
o Defined the term "frame", incorporating the transmission time offset into its definition, and removed references to "video frame". o Referred to "m(i)" from the text to make the derivation clearer. o Made it clearer that we modify our estimates of available bandwidth, and not the true available bandwidth. o Removed the appendixes outlining new extensions, added pointers to REMB draft and RFC 5450.
o 定义术语“帧”,将传输时间偏移纳入其定义,并删除对“视频帧”的引用。 o 从案文中提到“m(i)”,以使推导更清楚。 o 更清楚地表明,我们修改了对可用带宽的估计,而不是真正的可用带宽。 o 删除概述新扩展的附录,添加指向REMB草案和RFC 5450的指针。
A.3. Version -02 to -03
o Added a section on how to process multiple streams in a single estimator using RTP timestamps to NTP time conversion. o Stated in introduction that the draft is aimed at the RMCAT working group.
o 增加了一节,介绍如何使用RTP时间戳到NTP时间转换在单个估计器中处理多个流。 o 在导言中指出,草案的目标是RMCAT工作组。
A.4. rtcweb-03 to rmcat-00
Renamed draft to link the draft name to the RMCAT WG.
重命名草稿以将草稿名称链接到RMCAT WG。
A.5. rmcat -00 to -01
Spellcheck. Otherwise no changes, this is a "keepalive" release.
拼写检查。否则没有更改,这是一个“keepalive”版本。
A.6. rmcat -01 to -02
o Added Luca De Cicco and Saverio Mascolo as authors. o Extended the "Over-use detector" section with new technical details on how to dynamically tune the offset del_var_th for improved fairness properties. o Added reference to a paper analyzing the behavior of the proposed algorithm.
o 增加了卢卡·德·西科和萨维里奥·马斯科洛作为作者。 o 扩展了“过度使用检测器”部分,提供了关于如何动态调整偏移量del_var_th以改进公平性属性的新技术细节。 o 在一篇分析所提出算法行为的论文中增加了参考。
A.7. rmcat -02 to -03
o Swapped receiver-side/sender-side controller with delay-based/ loss-based controller as there is no longer a requirement to run the delay-based controller on the receiver-side. o Removed the discussion about multiple streams and transmission time offsets. o Introduced a new section about "Feedback and extensions". o Improvements to the threshold adaptation in the "Over-use detector" section. o Swapped the previous MIMD rate control algorithm for a new AIMD rate control algorithm.
o 由于不再需要在接收方运行基于延迟的控制器,因此将接收方/发送方控制器与基于延迟/丢失的控制器交换。 o 删除了关于多个流和传输时间偏移的讨论。 o 引入了关于“反馈和扩展”的新章节。 o 改进“过度使用检测器”部分中的阈值自适应。 o 将以前的MIMD速率控制算法替换为新的AIMD速率控制算法。
A.8. ietf-rmcat -00 to ietf-rmcat -01
o Arrival-time filter converted from a two dimensional Kalman filter to a scalar Kalman filter. o The use of the TFRC equation was removed from the loss-based controller, as it turned out to have little to no effect in practice.
o 到达时间滤波器从二维卡尔曼滤波器转换为标量卡尔曼滤波器。 o 从基于损耗的控制器中删除了TFRC方程的使用,因为它在实践中几乎没有影响。
A.9. ietf-rmcat -01 to ietf-rmcat -02
o Added a section which better describes the pre-filtering algorithm.
o 增加了一节,更好地描述了预过滤算法。
Authors’ Addresses 作者联系方式
Stefan Holmer Google Kungsbron 2 Stockholm 11122 Sweden Email: holmer@google.com
Henrik Lundin Google Kungsbron 2 Stockholm 11122 Sweden Email: hlundin@google.com
Gaetano Carlucci Politecnico di Bari Via Orabona, 4 Bari 70125 Italy Email: gaetano.carlucci@poliba.it
Luca De Cicco Politecnico di Bari Via Orabona, 4 Bari 70125 Italy Email: l.decicco@poliba.it
Saverio Mascolo Politecnico di Bari Via Orabona, 4 Bari 70125 Italy Email: mascolo@poliba.it