Congeston Control Evaluation

Abstract

Congeston Control Evaluation

Category

Learning note

Authors

Walter Fan

Status

WIP

Updated

2024-08-21

概述

Evaluating Congestion Control for Interactive Real-Time Media

度量

从数据包日志中可以计算出下列度量指标

  1. Sending rate, receiver rate, goodput (measured at 200ms intervals) 发送速率，接收速率，和实际吞吐量（以 200ms 间隔来度量）

  2. Packets sent, packets received

  3. Bytes sent, bytes received

  4. Packet delay

  5. Packets lost, packets discarded (from the playout or de-jitter buffer)

  6. If using retransmission or FEC: post-repair loss

  7. Self-fairness and fairness with respect to cross traffic:

Experiments testing a given congestion control proposal must report on relative ratios of the average throughput (measured at coarser time intervals) obtained by each RTP media stream. In the presence of background cross-traffic such as TCP, the report must also include the relative ratio between average throughput of RTP media streams and cross-traffic streams.

During static periods of a test (i.e., when bottleneck bandwidth is constant and no arrival/departure of streams), these reports on relative ratios serve as an indicator of how fairly the RTP streams share bandwidth amongst themselves and against cross- traffic streams. The throughput measurement interval should be set at a few values (for example, at 1 s, 5 s, and 20 s) in order to measure fairness across different timescales.

As a general guideline, the relative ratio between congestion- controlled RTP flows with the same priority level and similar path RTT should be bounded between 0.333 and 3. For example, see the test scenarios described in [RFC8867].

  1. Convergence time: The time taken to reach a stable rate at startup, after the available link capacity changes, or when new flows get added to the bottleneck link.

  2. Instability or oscillation in the sending rate:

    The frequency or number of instances when the sending rate oscillates between an high watermark level and a low watermark level, or vice-versa in a defined time window. For example, the watermarks can be set at 4x interval: 500 Kbps, 2 Mbps, and a time window of 500 ms.

  3. Bandwidth utilization, defined as the ratio of the instantaneous sending rate to the instantaneous bottleneck capacity:

    This metric is useful only when a congestion-controlled RTP flow is by itself or is competing with similar cross-traffic.

From the logs, the statistical measures (min, max, mean, standard deviation, and variance) for the whole duration or any specific part of the session can be calculated. Also the metrics (sending rate, receiver rate, goodput, latency) can be visualized in graphs as variation over time; the measurements in the plot are at one-second intervals. Additionally, from the logs, it is possible to plot the histogram or cumulative distribution function (CDF) of packet delay.

List of Network Parameters

One-Way Propagation Delay

  1. Very low latency: 0-1 ms

  2. Low latency: 50 ms

  3. High latency: 150 ms

  4. Extreme latency: 300 ms

End-to-End Loss

  1. no loss: 0%

  2. 1%

  3. 5%

  4. 10%

  5. 20%

Drop-Tail Router Queue Length

Loss Generation Model

Jitter Models

Random Bounded PDV (RBPDV)

Approximately Random Subject to No-Reordering Bounded PDV (NR-BPDV)

Traffic Models

TCP Traffic Model

RTP Video Model

Background UDP

Evaluation by ns-3

  • edit /etc/profile

export WEBRTC_LIB=/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib
export LD_LIBRARY_PATH=$WEBRTC_LIB/webrtc/system_wrappers:$WEBRTC_LIB/webrtc/rtc_base:$WEBRTC_LIB/webrtc/api:$WEBRTC_LIB/webrtc/logging:$WEBRTC_LIB/webrtc/modules/utility:$WEBRTC_LIB/webrtc/modules/pacing:$WEBRTC_LIB/webrtc/modules/congestion_controller:$WEBRTC_LIB/webrtc/modules/bitrate_controller:$WEBRTC_LIB/webrtc/modules/remote_bitrate_estimator:$WEBRTC_LIB/webrtc/modules/rtp_rtcp:$LD_LIBRARY_PATH
export CPLUS_INCLUDE_PATH=CPLUS_INCLUDE_PATH:$WEBRTC_LIB/webrtc/:$WEBRTC_LIB/webrtc/system_wrappers:$WEBRTC_LIB/webrtc/rtc_base:$WEBRTC_LIB/webrtc/api:$WEBRTC_LIB/webrtc/logging:$WEBRTC_LIB/webrtc/modules/utility:$WEBRTC_LIB/webrtc/modules/pacing:$WEBRTC_LIB/webrtc/modules/congestion_controller:$WEBRTC_LIB/webrtc/modules/bitrate_controller:$WEBRTC_LIB/webrtc/modules/remote_bitrate_estimator:$WEBRTC_LIB/webrtc/modules/rtp_rtcp

  1. edit webrtc-ns3/wscript

The path about the headers and so libs in wscript(under webrtc-ns3) should also be changed accordingly:

conf.env.append_value('INCLUDES', ['/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib/webrtc/'])
conf.env.append_value("LINKFLAGS", ['-L/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib/webrtc/system_wrappers','-L/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib/webrtc/rtc_base','-L/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib/webrtc/api','-L/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib/webrtc/logging','-L/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib/webrtc/modules/utility','-L/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib/webrtc/modules/pacing','-L/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib/webrtc/modules/congestion_controller','-L/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib/webrtc/modules/bitrate_controller','-L/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib/webrtc/modules/remote_bitrate_estimator','-L/home/walter/workspace/webrtc/rmcat-ns3/webrtc-lib/webrtc/modules/rtp_rtcp'])

  1. put these modules to /home/walter/workspace/webrtc/ns-allinone-3.35/ns-3.35/src

  • mystrace

  • webrtc-ns3

  • multipathvid

./waf –run webrtc-scratch