draft-ietf-rmcat-eval-criteria-14.txt   rfc8868.txt 
RMCAT WG V. Singh Internet Engineering Task Force (IETF) V. Singh
Internet-Draft callstats.io Request for Comments: 8868 callstats.io
Intended status: Informational J. Ott Category: Informational J. Ott
Expires: September 20, 2020 Technical University of Munich ISSN: 2070-1721 Technical University of Munich
S. Holmer S. Holmer
Google Google
March 19, 2020 January 2021
Evaluating Congestion Control for Interactive Real-time Media Evaluating Congestion Control for Interactive Real-Time Media
draft-ietf-rmcat-eval-criteria-14
Abstract Abstract
The Real-time Transport Protocol (RTP) is used to transmit media in The Real-Time Transport Protocol (RTP) is used to transmit media in
telephony and video conferencing applications. This document telephony and video conferencing applications. This document
describes the guidelines to evaluate new congestion control describes the guidelines to evaluate new congestion control
algorithms for interactive point-to-point real-time media. algorithms for interactive point-to-point real-time media.
Status of This Memo Status of This Memo
This Internet-Draft is submitted in full conformance with the This document is not an Internet Standards Track specification; it is
provisions of BCP 78 and BCP 79. published for informational purposes.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months This document is a product of the Internet Engineering Task Force
and may be updated, replaced, or obsoleted by other documents at any (IETF). It represents the consensus of the IETF community. It has
time. It is inappropriate to use Internet-Drafts as reference received public review and has been approved for publication by the
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approved by the IESG are candidates for any level of Internet
Standard; see Section 2 of RFC 7841.
This Internet-Draft will expire on September 20, 2020. Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc8868.
Copyright Notice Copyright Notice
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document authors. All rights reserved. document authors. All rights reserved.
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Table of Contents Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 1. Introduction
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology
3. Metrics . . . . . . . . . . . . . . . . . . . . . . . . . . . 4 3. Metrics
3.1. RTP Log Format . . . . . . . . . . . . . . . . . . . . . 5 3.1. RTP Log Format
4. List of Network Parameters . . . . . . . . . . . . . . . . . 6 4. List of Network Parameters
4.1. One-way Propagation Delay . . . . . . . . . . . . . . . . 6 4.1. One-Way Propagation Delay
4.2. End-to-end Loss . . . . . . . . . . . . . . . . . . . . . 6 4.2. End-to-End Loss
4.3. Drop Tail Router Queue Length . . . . . . . . . . . . . . 7 4.3. Drop-Tail Router Queue Length
4.4. Loss generation model . . . . . . . . . . . . . . . . . . 7 4.4. Loss Generation Model
4.5. Jitter models . . . . . . . . . . . . . . . . . . . . . . 7 4.5. Jitter Models
4.5.1. Random Bounded PDV (RBPDV) . . . . . . . . . . . . . 8 4.5.1. Random Bounded PDV (RBPDV)
4.5.2. Approximately Random Subject to No-Reordering Bounded 4.5.2. Approximately Random Subject to No-Reordering Bounded
PDV (NR-RPVD) . . . . . . . . . . . . . . . . 9 PDV (NR-BPDV)
4.5.3. Recommended distribution . . . . . . . . . . . . . . 10 4.5.3. Recommended Distribution
5. Traffic Models . . . . . . . . . . . . . . . . . . . . . . . 10 5. Traffic Models
5.1. TCP traffic model . . . . . . . . . . . . . . . . . . . . 10 5.1. TCP Traffic Model
5.2. RTP Video model . . . . . . . . . . . . . . . . . . . . . 11 5.2. RTP Video Model
5.3. Background UDP . . . . . . . . . . . . . . . . . . . . . 11 5.3. Background UDP
6. Security Considerations . . . . . . . . . . . . . . . . . . . 12 6. Security Considerations
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 12 7. IANA Considerations
8. Contributors . . . . . . . . . . . . . . . . . . . . . . . . 12 8. References
9. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 12 8.1. Normative References
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 13 8.2. Informative References
10.1. Normative References . . . . . . . . . . . . . . . . . . 13 Contributors
10.2. Informative References . . . . . . . . . . . . . . . . . 14 Acknowledgments
Appendix A. Change Log . . . . . . . . . . . . . . . . . . . . . 15 Authors' Addresses
A.1. Changes in draft-ietf-rmcat-eval-criteria-07 . . . . . . 15
A.2. Changes in draft-ietf-rmcat-eval-criteria-06 . . . . . . 15
A.3. Changes in draft-ietf-rmcat-eval-criteria-05 . . . . . . 15
A.4. Changes in draft-ietf-rmcat-eval-criteria-04 . . . . . . 15
A.5. Changes in draft-ietf-rmcat-eval-criteria-03 . . . . . . 15
A.6. Changes in draft-ietf-rmcat-eval-criteria-02 . . . . . . 15
A.7. Changes in draft-ietf-rmcat-eval-criteria-01 . . . . . . 16
A.8. Changes in draft-ietf-rmcat-eval-criteria-00 . . . . . . 16
A.9. Changes in draft-singh-rmcat-cc-eval-04 . . . . . . . . . 16
A.10. Changes in draft-singh-rmcat-cc-eval-03 . . . . . . . . . 16
A.11. Changes in draft-singh-rmcat-cc-eval-02 . . . . . . . . . 16
A.12. Changes in draft-singh-rmcat-cc-eval-01 . . . . . . . . . 17
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 17
1. Introduction 1. Introduction
This memo describes the guidelines to help with evaluating new This memo describes the guidelines to help with evaluating new
congestion control algorithms for interactive point-to-point real congestion control algorithms for interactive point-to-point real-
time media. The requirements for the congestion control algorithm time media. The requirements for the congestion control algorithm
are outlined in [I-D.ietf-rmcat-cc-requirements]). This document are outlined in [RFC8836]. This document builds upon previous work
builds upon previous work at the IETF: Specifying New Congestion at the IETF: Specifying New Congestion Control Algorithms [RFC5033]
Control Algorithms [RFC5033] and Metrics for the Evaluation of and Metrics for the Evaluation of Congestion Control Algorithms
Congestion Control Algorithms [RFC5166]. [RFC5166].
The guidelines proposed in the document are intended to help prevent The guidelines proposed in the document are intended to help prevent
a congestion collapse, promote fair capacity usage and optimize the a congestion collapse, to promote fair capacity usage, and to
media flow's throughput. Furthermore, the proposed congestion optimize the media flow's throughput. Furthermore, the proposed
control algorithms are expected to operate within the envelope of the congestion control algorithms are expected to operate within the
circuit breakers defined in RFC8083 [RFC8083]. envelope of the circuit breakers defined in RFC 8083 [RFC8083].
This document only provides the broad set of network parameters and This document only provides the broad set of network parameters and
and traffic models for evaluating a new congestion control algorithm. traffic models for evaluating a new congestion control algorithm.
The minimal requirements for congestion control proposals is to The minimal requirement for congestion control proposals is to
produce or present results for the test scenarios described in produce or present results for the test scenarios described in
[I-D.ietf-rmcat-eval-test] (Basic Test Cases), which also defines the [RFC8867] (Basic Test Cases), which also defines the specifics for
specifics for the test cases. Additionally, proponents may produce the test cases. Additionally, proponents may produce evaluation
evaluation results for the wireless test scenarios results for the wireless test scenarios [RFC8869].
[I-D.ietf-rmcat-wireless-tests].
This document does not cover application-specific implications of This document does not cover application-specific implications of
congestion control algorithms and how those could be evaluated. congestion control algorithms and how those could be evaluated.
Therefore, no quality metrics are defined for performance evaluation; Therefore, no quality metrics are defined for performance evaluation;
quality metrics and algorithms to infer those vary between media quality metrics and the algorithms to infer those vary between media
types. Metrics and algorithms to assess, e.g., quality of experience types. Metrics and algorithms to assess, e.g., the quality of
evolve continuously so that determining suitable choices is left for experience, evolve continuously so that determining suitable choices
future work. However, there is consensus that each congestion is left for future work. However, there is consensus that each
control algorithm should be able to show that it is useful for congestion control algorithm should be able to show that it is useful
interactive video by performing analysis using a real codecs and for interactive video by performing analysis using real codecs and
video sequences and state-of-the-art quality metrics. video sequences and state-of-the-art quality metrics.
Beyond optimizing individual metrics, real-time applications may have Beyond optimizing individual metrics, real-time applications may have
further options to trade off performance, e.g., across multiple further options to trade off performance, e.g., across multiple
media; refer to the RMCAT requirements media; refer to the RMCAT requirements [RFC8836] document. Such
[I-D.ietf-rmcat-cc-requirements] document. Such trade-offs may be trade-offs may be defined in the future.
defined in the future.
2. Terminology 2. Terminology
The terminology defined in RTP [RFC3550], RTP Profile for Audio and The terminology defined in RTP [RFC3550], RTP Profile for Audio and
Video Conferences with Minimal Control [RFC3551], RTCP Extended Video Conferences with Minimal Control [RFC3551], RTCP Extended
Report (XR) [RFC3611], Extended RTP Profile for RTCP-based Feedback Report (XR) [RFC3611], Extended RTP Profile for RTCP-Based Feedback
(RTP/AVPF) [RFC4585] and Support for Reduced-Size RTCP [RFC5506] (RTP/AVPF) [RFC4585] and Support for Reduced-Size RTCP [RFC5506]
apply. applies.
3. Metrics 3. Metrics
This document specifies testing criteria for evaluating congestion This document specifies testing criteria for evaluating congestion
control algorithms for RTP media flows. Proposed algorithms are to control algorithms for RTP media flows. Proposed algorithms are to
prove their performance by means of simulation and/or emulation prove their performance by means of simulation and/or emulation
experiments for all the cases described. experiments for all the cases described.
Each experiment is expected to log every incoming and outgoing packet Each experiment is expected to log every incoming and outgoing packet
(the RTP logging format is described in Section 3.1). The logging (the RTP logging format is described in Section 3.1). The logging
can be done inside the application or at the endpoints using PCAP can be done inside the application or at the endpoints using PCAP
(packet capture, e.g., tcpdump [tcpdump], wireshark [wireshark]). (packet capture, e.g., tcpdump [tcpdump], Wireshark [wireshark]).
The following metrics are calculated based on the information in the The following metrics are calculated based on the information in the
packet logs: packet logs:
1. Sending rate, Receiver rate, Goodput (measured at 200ms 1. Sending rate, receiver rate, goodput (measured at 200ms
intervals) intervals)
2. Packets sent, Packets received 2. Packets sent, packets received
3. Bytes sent, bytes received 3. Bytes sent, bytes received
4. Packet delay 4. Packet delay
5. Packets lost, Packets discarded (from the playout or de-jitter 5. Packets lost, packets discarded (from the playout or de-jitter
buffer) buffer)
6. If using, retransmission or FEC: post-repair loss 6. If using retransmission or FEC: post-repair loss
7. Self-Fairness and Fairness with respect to cross traffic: 7. Self-fairness and fairness with respect to cross traffic:
Experiments testing a given congestion control proposal must Experiments testing a given congestion control proposal must
report on relative ratios of the average throughput (measured at report on relative ratios of the average throughput (measured at
coarser time intervals) obtained by each RTP media stream. In coarser time intervals) obtained by each RTP media stream. In
the presence of background cross-traffic such as TCP, the report the presence of background cross-traffic such as TCP, the report
must also include the relative ratio between average throughput must also include the relative ratio between average throughput
of RTP media streams and cross-traffic streams. of RTP media streams and cross-traffic streams.
During static periods of a test (i.e., when bottleneck bandwidth During static periods of a test (i.e., when bottleneck bandwidth
is constant and no arrival/departure of streams), these report is constant and no arrival/departure of streams), these reports
on relative ratios serve as an indicator of how fair the RTP on relative ratios serve as an indicator of how fairly the RTP
streams share bandwidth amongst themselves and against cross- streams share bandwidth amongst themselves and against cross-
traffic streams. The throughput measurement interval should be traffic streams. The throughput measurement interval should be
set at a few values (for example, at 1s, 5s, and 20s) in order set at a few values (for example, at 1 s, 5 s, and 20 s) in
to measure fairness across different time scales. order to measure fairness across different timescales.
As a general guideline, the relative ratio between congestion
As a general guideline, the relative ratio between congestion-
controlled RTP flows with the same priority level and similar controlled RTP flows with the same priority level and similar
path RTT should be bounded between (0.333 and 3.) For example, path RTT should be bounded between 0.333 and 3. For example,
see the test scenarios described in [I-D.ietf-rmcat-eval-test]. see the test scenarios described in [RFC8867].
8. Convergence time: The time taken to reach a stable rate at 8. Convergence time: The time taken to reach a stable rate at
startup, after the available link capacity changes, or when new startup, after the available link capacity changes, or when new
flows get added to the bottleneck link. flows get added to the bottleneck link.
9. Instability or oscillation in the sending rate: The frequency or 9. Instability or oscillation in the sending rate: The frequency or
number of instances when the sending rate oscillates between an number of instances when the sending rate oscillates between an
high watermark level and a low watermark level, or vice-versa in high watermark level and a low watermark level, or vice-versa in
a defined time window. For example, the watermarks can be set a defined time window. For example, the watermarks can be set
at 4x interval: 500 Kbps, 2 Mbps, and a time window of 500ms. at 4x interval: 500 Kbps, 2 Mbps, and a time window of 500 ms.
10. Bandwidth Utilization, defined as ratio of the instantaneous 10. Bandwidth utilization, defined as the ratio of the instantaneous
sending rate to the instantaneous bottleneck capacity. This sending rate to the instantaneous bottleneck capacity: This
metric is useful only when a congestion controlled RTP flow is metric is useful only when a congestion-controlled RTP flow is
by itself or competing with similar cross-traffic. by itself or is competing with similar cross-traffic.
Note that the above metrics are all objective application-independent Note that the above metrics are all objective application-independent
metrics. Refer to Section 3, in [I-D.ietf-netvc-testing] for metrics. Refer to Section 3 of [netvc-testing] for objective metrics
objective metrics for evaluating codecs. for evaluating codecs.
From the logs the statistical measures (min, max, mean, standard From the logs, the statistical measures (min, max, mean, standard
deviation and variance) for the whole duration or any specific part deviation, and variance) for the whole duration or any specific part
of the session can be calculated. Also the metrics (sending rate, of the session can be calculated. Also the metrics (sending rate,
receiver rate, goodput, latency) can be visualized in graphs as receiver rate, goodput, latency) can be visualized in graphs as
variation over time, the measurements in the plot are at 1 second variation over time; the measurements in the plot are at one-second
intervals. Additionally, from the logs it is possible to plot the intervals. Additionally, from the logs, it is possible to plot the
histogram or CDF of packet delay. histogram or cumulative distribution function (CDF) of packet delay.
3.1. RTP Log Format 3.1. RTP Log Format
Having a common log format simplifies running analyses across and Having a common log format simplifies running analyses across
comparing different measurements. The log file should be tab or different measurement setups and comparing their results.
comma separated containing the following details:
Send or receive timestamp (Unix): <int>.<int> -- sec.usec decimal Send or receive timestamp (Unix): <int>.<int> -- sec.usec decimal
RTP payload type <int> -- decimal RTP payload type <int> -- decimal
SSRC <int> -- hexadecimal SSRC <int> -- hexadecimal
RTP sequence no <int> -- decimal RTP sequence no <int> -- decimal
RTP timestamp <int> -- decimal RTP timestamp <int> -- decimal
marker bit 0|1 -- character marker bit 0|1 -- character
Payload size <int> -- # bytes, decimal Payload size <int> -- # bytes, decimal
Each line of the log file should be terminated with CRLF, CR, or LF Each line of the log file should be terminated with CRLF, CR, or LF
characters. Empty lines are disregarded. characters. Empty lines are disregarded.
If the congestion control implements retransmissions or FEC, the If the congestion control implements retransmissions or Forward Error
evaluation should report both packet loss (before applying error- Correction (FEC), the evaluation should report both packet loss
resilience) and residual packet loss (after applying error- (before applying error resilience) and residual packet loss (after
resilience). applying error resilience).
These data should suffice to compute the media-encoding independent These data should suffice to compute the media-encoding independent
metrics described above. Use of a common log will allow simplified metrics described above. Use of a common log will allow simplified
post-processing and analysis across different implementations. post-processing and analysis across different implementations.
4. List of Network Parameters 4. List of Network Parameters
The implementors initially are encouraged to choose evaluation The implementors are encouraged to choose evaluation settings from
settings from the following values: the following values initially:
4.1. One-way Propagation Delay 4.1. One-Way Propagation Delay
Experiments are expected to verify that the congestion control is Experiments are expected to verify that the congestion control is
able to work across a broad range of path characteristics, also able to work across a broad range of path characteristics, including
including challenging situations, for example over trans-continental challenging situations, for example, over transcontinental and/or
and/or satellite links. Tests thus account for the following satellite links. Tests thus account for the following different
different latencies: latencies:
1. Very low latency: 0-1ms 1. Very low latency: 0-1 ms
2. Low latency: 50ms 2. Low latency: 50 ms
3. High latency: 150ms 3. High latency: 150 ms
4. Extreme latency: 300ms 4. Extreme latency: 300 ms
4.2. End-to-end Loss 4.2. End-to-End Loss
Many paths in the Internet today are largely lossless but, with Many paths in the Internet today are largely lossless; however, in
wireless networks and interference, towards remote regions, or in scenarios featuring interference in wireless networks, sending to and
scenarios featuring high/fast mobility, media flows may exhibit receiving from remote regions, or high/fast mobility, media flows may
substantial packet loss. This variety needs to be reflected exhibit substantial packet loss. This variety needs to be reflected
appropriately by the tests. appropriately by the tests.
To model a wide range of lossy links, the experiments can choose one To model a wide range of lossy links, the experiments can choose one
of the following loss rates, the fractional loss is the ratio of of the following loss rates; the fractional loss is the ratio of
packets lost and packets sent. packets lost and packets sent:
1. no loss: 0% 1. no loss: 0%
2. 1% 2. 1%
3. 5% 3. 5%
4. 10% 4. 10%
5. 20% 5. 20%
4.3. Drop Tail Router Queue Length 4.3. Drop-Tail Router Queue Length
Routers should be configured to use Drop Trail queues in the Routers should be configured to use drop-tail queues in the
experiments due to their (still) prevalent nature. Experimentation experiments due to their (still) prevalent nature. Experimentation
with AQM schemes is encouraged but not mandatory. with Active Queue Management (AQM) schemes is encouraged but not
mandatory.
The router queue length is measured as the time taken to drain the The router queue length is measured as the time taken to drain the
FIFO queue. It has been noted in various discussions that the queue FIFO queue. It has been noted in various discussions that the queue
length in the current deployed Internet varies significantly. While length in the currently deployed Internet varies significantly.
the core backbone network has very short queue length, the home While the core backbone network has very short queue length, the home
gateways usually have larger queue length. Those various queue gateways usually have larger queue length. Those various queue
lengths can be categorized in the following way: lengths can be categorized in the following way:
1. QoS-aware (or short): 70ms 1. QoS-aware (or short): 70 ms
2. Nominal: 300-500ms 2. Nominal: 300-500 ms
3. Buffer-bloated: 1000-2000ms 3. Buffer-bloated: 1000-2000 ms
Here the size of the queue is measured in bytes or packets and to Here the size of the queue is measured in bytes or packets. To
convert the queue length measured in seconds to queue length in convert the queue length measured in seconds to queue length in
bytes: bytes:
QueueSize (in bytes) = QueueSize (in sec) x Throughput (in bps)/8 QueueSize (in bytes) = QueueSize (in sec) x Throughput (in bps)/8
4.4. Loss generation model 4.4. Loss Generation Model
Many models for generating packet loss are available, some yield Many models for generating packet loss are available: some generate
correlated, others independent losses; losses can also be extracted correlated packet losses, others generate independent packet losses.
from packet traces. As a (simple) minimum loss model with minimal In addition, packet losses can also be extracted from packet traces.
parameterization (i.e., the loss rate), independent random losses As a (simple) minimum loss model with minimal parameterization (i.e.,
must be used in the evaluation. the loss rate), independent random losses must be used in the
evaluation.
It is known that independent loss models may reflect reality poorly It is known that independent loss models may reflect reality poorly,
and hence more sophisticated loss models could be considered. and hence more sophisticated loss models could be considered.
Suitable models for correlated losses includes the Gilbert-Elliot Suitable models for correlated losses include the Gilbert-Elliot
model [gilbert-elliott] and losses generated by modeling a queue model [gilbert-elliott] and models that generate losses by modeling a
including its (different) drop behaviors. queue with its (different) drop behaviors.
4.5. Jitter models 4.5. Jitter Models
This section defines jitter models for the purposes of this document. This section defines jitter models for the purposes of this document.
When jitter is to be applied to both the congestion controlled RTP When jitter is to be applied to both the congestion-controlled RTP
flow and any competing flow (such as a TCP competing flow), the flow and any competing flow (such as a TCP competing flow), the
competing flow will use the jitter definition below that does not competing flow will use the jitter definition below that does not
allow for re-ordering of packets on the competing flow (see NR-RBPDV allow for reordering of packets on the competing flow (see NR-BPDV
definition below). definition below).
Jitter is an overloaded term in communications. It is typically used Jitter is an overloaded term in communications. It is typically used
to refer to the variation of a metric (e.g., delay) with respect to to refer to the variation of a metric (e.g., delay) with respect to
some reference metric (e.g., average delay or minimum delay). For some reference metric (e.g., average delay or minimum delay). For
example, RFC 3550 jitter is computed as the smoothed difference in example in RFC 3550, jitter is computed as the smoothed difference in
packet arrival times relative to their respective expected arrival packet arrival times relative to their respective expected arrival
times, which is particularly meaningful if the underlying packet times, which is particularly meaningful if the underlying packet
delay variation was caused by a Gaussian random process. delay variation was caused by a Gaussian random process.
Because jitter is an overloaded term, we use the term Packet Delay Because jitter is an overloaded term, we use the term Packet Delay
Variation (PDV) instead to describe the variation of delay of Variation (PDV) instead to describe the variation of delay of
individual packets in the same sense as the IETF IPPM WG has defined individual packets in the same sense as the IETF IP Performance
PDV in their documents (e.g., RFC 3393) and as the ITU-T SG16 has Metrics (IPPM) working group has defined PDV in their documents
defined IP Packet Delay Variation (IPDV) in their documents (e.g., (e.g., RFC 3393) and as the ITU-T SG16 has defined IP Packet Delay
Y.1540). Variation (IPDV) in their documents (e.g., Y.1540).
Most PDV distributions in packet network systems are one-sided Most PDV distributions in packet network systems are one-sided
distributions, the measurement of which with a finite number of distributions, the measurement of which with a finite number of
measurement samples results in one-sided histograms. In the usual measurement samples results in one-sided histograms. In the usual
packet network transport case, there is typically one packet that packet network transport case, there is typically one packet that
transited the network with the minimum delay; a (large) number of transited the network with the minimum delay; a (large) number of
packets transit the network within some (smaller) positive variation packets transit the network within some (smaller) positive variation
from this minimum delay, and a (small) number of the packets transit from this minimum delay, and a (small) number of the packets transit
the network with delays higher than the median or average transit the network with delays higher than the median or average transit
time (these are outliers). Although infrequent, outliers can cause time (these are outliers). Although infrequent, outliers can cause
significant deleterious operation in adaptive systems and should be significant deleterious operation in adaptive systems and should be
considered in rate adaptation designs for RTP congestion control. considered in rate adaptation designs for RTP congestion control.
In this section we define two different bounded PDV characteristics, In this section we define two different bounded PDV characteristics,
1) Random Bounded PDV and 2) Approximately Random Subject to No- 1) Random Bounded PDV and 2) Approximately Random Subject to No-
Reordering Bounded PDV. Reordering Bounded PDV.
The former, 1) Random Bounded PDV is presented for information only, The former, 1) Random Bounded PDV, is presented for information only,
while the latter, 2) Approximately Random Subject to No-Reordering while the latter, 2) Approximately Random Subject to No-Reordering
Bounded PDV, must be used in the evaluation. Bounded PDV, must be used in the evaluation.
4.5.1. Random Bounded PDV (RBPDV) 4.5.1. Random Bounded PDV (RBPDV)
The RBPDV probability distribution function (PDF) is specified to be The RBPDV probability distribution function (PDF) is specified to be
of some mathematically describable function which includes some of some mathematically describable function that includes some
practical minimum and maximum discrete values suitable for testing. practical minimum and maximum discrete values suitable for testing.
For example, the minimum value, x_min, might be specified as the For example, the minimum value, x_min, might be specified as the
minimum transit time packet and the maximum value, x_max, might be minimum transit time packet, and the maximum value, x_max, might be
defined to be two standard deviations higher than the mean. defined to be two standard deviations higher than the mean.
Since we are typically interested in the distribution relative to the Since we are typically interested in the distribution relative to the
mean delay packet, we define the zero mean PDV sample, z(n), to be mean delay packet, we define the zero mean PDV sample, z(n), to be
z(n) = x(n) - x_mean, where x(n) is a sample of the RBPDV random z(n) = x(n) - x_mean, where x(n) is a sample of the RBPDV random
variable x and x_mean is the mean of x. variable x and x_mean is the mean of x.
We assume here that s(n) is the original source time of packet n and We assume here that s(n) is the original source time of packet n and
the post-jitter induced emission time, j(n), for packet n is: the post-jitter induced emission time, j(n), for packet n is:
j(n) = {[z(n) + x_mean] + s(n)}. j(n) = {[z(n) + x_mean] + s(n)}.
It follows that the separation in the post-jitter time of packets n It follows that the separation in the post-jitter time of packets n
and n+1 is {[s(n+1)-s(n)] - [z(n)-z(n+1)]}. Since the first term is and n+1 is {[s(n+1)-s(n)] - [z(n)-z(n+1)]}. Since the first term is
always a positive quantity, we note that packet reordering at the always a positive quantity, we note that packet reordering at the
receiver is possible whenever the second term is greater than the receiver is possible whenever the second term is greater than the
first. Said another way, whenever the difference in possible zero first. Said another way, whenever the difference in possible zero
mean PDV sample delays (i.e., [x_max-x_min]) exceeds the inter- mean PDV sample delays (i.e., [x_max-x_min]) exceeds the inter-
departure time of any two sent packets, we have the possibility of departure time of any two sent packets, we have the possibility of
packet re-ordering. packet reordering.
There are important use cases in real networks where packets can There are important use cases in real networks where packets can
become re-ordered such as in load balancing topologies and during become reordered, such as in load-balancing topologies and during
route changes. However, for the vast majority of cases there is no route changes. However, for the vast majority of cases, there is no
packet re-ordering because most of the time packets follow the same packet reordering because most of the time packets follow the same
path. Due to this, if a packet becomes overly delayed, the packets path. Due to this, if a packet becomes overly delayed, the packets
after it on that flow are also delayed. This is especially true for after it on that flow are also delayed. This is especially true for
mobile wireless links where there are per-flow queues prior to base mobile wireless links where there are per-flow queues prior to base
station scheduling. Owing to this important use case, we define station scheduling. Owing to this important use case, we define
another PDV profile similar to the above, but one that does not allow another PDV profile similar to the above, but one that does not allow
for re-ordering within a flow. for reordering within a flow.
4.5.2. Approximately Random Subject to No-Reordering Bounded PDV (NR- 4.5.2. Approximately Random Subject to No-Reordering Bounded PDV (NR-
RPVD) BPDV)
No Reordering RPDV, NR-RPVD, is defined similarly to the above with No Reordering BPDV, NR-BPDV, is defined similarly to the above with
one important exception. Let serial(n) be defined as the one important exception. Let serial(n) be defined as the
serialization delay of packet n at the lowest bottleneck link rate serialization delay of packet n at the lowest bottleneck link rate
(or other appropriate rate) in a given test. Then we produce all the (or other appropriate rate) in a given test. Then we produce all the
post-jitter values for j(n) for n = 1, 2, ... N, where N is the post-jitter values for j(n) for n = 1, 2, ... N, where N is the
length of the source sequence s to be offset-ed. The exception can length of the source sequence s to be offset. The exception can be
be stated as follows: We revisit all j(n) beginning from index n=2, stated as follows: We revisit all j(n) beginning from index n=2, and
and if j(n) is determined to be less than [j(n-1)+serial(n-1)], we if j(n) is determined to be less than [j(n-1)+serial(n-1)], we
redefine j(n) to be equal to [j(n-1)+serial(n-1)] and continue for redefine j(n) to be equal to [j(n-1)+serial(n-1)] and continue for
all remaining n (i.e., n = 3, 4, .. N). This models the case where all remaining n (i.e., n = 3, 4, .. N). This models the case where
the packet n is sent immediately after packet (n-1) at the bottleneck the packet n is sent immediately after packet (n-1) at the bottleneck
link rate. Although this is generally the theoretical minimum in link rate. Although this is generally the theoretical minimum in
that it assumes that no other packets from other flows are in-between that it assumes that no other packets from other flows are in between
packet n and n+1 at the bottleneck link, it is a reasonable packet n and n+1 at the bottleneck link, it is a reasonable
assumption for per flow queuing. assumption for per-flow queuing.
We note that this assumption holds for some important exception We note that this assumption holds for some important exception
cases, such as packets immediately following outliers. There are a cases, such as packets immediately following outliers. There are a
multitude of software controlled elements common on end-to-end multitude of software-controlled elements common on end-to-end
Internet paths (such as firewalls, ALGs and other middleboxes) which Internet paths (such as firewalls, application-layer gateways, and
stop processing packets while servicing other functions (e.g., other middleboxes) that stop processing packets while servicing other
garbage collection). Often these devices do not drop packets, but functions (e.g., garbage collection). Often these devices do not
rather queue them for later processing and cause many of the drop packets, but rather queue them for later processing and cause
outliers. Thus NR-RPVD models this particular use case (assuming many of the outliers. Thus NR-BPDV models this particular use case
serial(n+1) is defined appropriately for the device causing the (assuming serial(n+1) is defined appropriately for the device causing
outlier) and thus is believed to be important for adaptation the outlier) and is believed to be important for adaptation
development for congestion controlled RTP streams. development for congestion-controlled RTP streams.
4.5.3. Recommended distribution 4.5.3. Recommended Distribution
Whether Random Bounded PDV or Approximately Random Subject to No- Whether Random Bounded PDV or Approximately Random Subject to No-
Reordering Bounded PDV, it is recommended that z(n) is distributed Reordering Bounded PDV, it is recommended that z(n) is distributed
according to a truncated Gaussian for the above jitter models: according to a truncated Gaussian for the above jitter models:
z(n) ~ |max(min(N(0, std^2), N_STD * std), -N_STD * std)| z(n) ~ |max(min(N(0, std^(2)), N_STD * std), -N_STD * std)|
where N(0, std^2) is the Gaussian distribution with zero mean and where N(0, std^(2)) is the Gaussian distribution with zero mean and
standard deviation std. Recommended values: std is standard deviation. Recommended values:
o std = 5 ms std = 5 ms
o N_STD = 3 N_STD = 3
5. Traffic Models 5. Traffic Models
5.1. TCP traffic model 5.1. TCP Traffic Model
Long-lived TCP flows will download data throughout the session and Long-lived TCP flows will download data throughout the session and
are expected to have infinite amount of data to send or receive. are expected to have infinite amount of data to send or receive.
This roughly applies, for example, when downloading software This roughly applies, for example, when downloading software
distributions. distributions.
Each short TCP flow is modeled as a sequence of file downloads Each short TCP flow is modeled as a sequence of file downloads
interleaved with idle periods. Not all short TCP flows start at the interleaved with idle periods. Not all short TCP flows start at the
same time, i.e., some start in the ON state while others start in the same time, i.e., some start in the ON state while others start in the
OFF state. OFF state.
The short TCP flows can be modeled as follows: 30 connections start The short TCP flows can be modeled as follows: 30 connections start
simultaneously fetching small (30-50 KB) amounts of data, evenly simultaneously fetching small (30-50 KB) amounts of data, evenly
distributed. This covers the case where the short TCP flows are distributed. This covers the case where the short TCP flows are
fetching web page resources rather than video files. fetching web page resources rather than video files.
The idle period between bursts of starting a group of TCP flows is The idle period between bursts of starting a group of TCP flows is
typically derived from an exponential distribution with the mean typically derived from an exponential distribution with the mean
value of 10 seconds. value of 10 seconds.
[These values were picked based on the data available at | These values were picked based on the data available at
http://httparchive.org/interesting.php as of October 2015]. | <https://httparchive.org/reports/state-of-the-
| web?start=2015_10_01&end=2015_11_01&view=list> as of October
| 2015.
Many different TCP congestion control schemes are deployed today. Many different TCP congestion control schemes are deployed today.
Therefore, experimentation with a range of different schemes, Therefore, experimentation with a range of different schemes,
especially including CUBIC, is encouraged. Experiments must document especially including CUBIC [RFC8312], is encouraged. Experiments
in detail which congestion control schemes they tested against and must document in detail which congestion control schemes they tested
which parameters were used. against and which parameters were used.
5.2. RTP Video model 5.2. RTP Video Model
[RFC8593] describes two types of video traffic models for evaluating [RFC8593] describes two types of video traffic models for evaluating
candidate algorithms for RTP congestion control. The first model candidate algorithms for RTP congestion control. The first model
statistically characterizes the behavior of a video encoder, whereas statistically characterizes the behavior of a video encoder, whereas
the second model uses video traces. the second model uses video traces.
Sample video test sequences are available at [xiph-seq]. The Sample video test sequences are available at [xiph-seq]. The
following two video streams are the recommended minimum for testing: following two video streams are the recommended minimum for testing:
Foreman (CIF sequence) and FourPeople (720p); both come as raw video Foreman (CIF sequence) and FourPeople (720p); both come as raw video
data to be encoded dynamically. As these video sequences are short data to be encoded dynamically. As these video sequences are short
(300 and 600 frames, respectively, they shall be stitched together (300 and 600 frames, respectively), they shall be stitched together
repeatedly until the desired length is reached. repeatedly until the desired length is reached.
5.3. Background UDP 5.3. Background UDP
Background UDP flow is modeled as a constant bit rate (CBR) flow. It Background UDP flow is modeled as a constant bit rate (CBR) flow. It
will download data at a particular CBR rate for the complete session, will download data at a particular CBR for the complete session, or
or will change to particular CBR rate at predefined intervals. The will change to particular CBR at predefined intervals. The inter-
inter packet interval is calculated based on the CBR and the packet packet interval is calculated based on the CBR and the packet size
size (is typically set to the path MTU size, the default value can be (typically set to the path MTU size, the default value can be 1500
1500 bytes). bytes).
Note that new transport protocols such as QUIC may use UDP but, due Note that new transport protocols such as QUIC may use UDP; however,
to their congestion control algorithms, will exhibit behavior due to their congestion control algorithms, they will exhibit
conceptually similar in nature to TCP flows above and can thus be behavior conceptually similar in nature to TCP flows above and can
subsumed by the above, including the division into short- and long- thus be subsumed by the above, including the division into short-
lived flows. As QUIC evolves independently of TCP congestion control lived and long-lived flows. As QUIC evolves independently of TCP
algorithms, its future congestion control should be considered as congestion control algorithms, its future congestion control should
competing traffic as appropriate. be considered as competing traffic as appropriate.
6. Security Considerations 6. Security Considerations
This document specifies evaluation criteria and parameters for This document specifies evaluation criteria and parameters for
assessing and comparing the performance of congestion control assessing and comparing the performance of congestion control
protocols and algorithms for real-time communication. This memo protocols and algorithms for real-time communication. This memo
itself is thus not subject to security considerations but the itself is thus not subject to security considerations but the
protocols and algorithms evaluated may be. In particular, successful protocols and algorithms evaluated may be. In particular, successful
operation under all tests defined in this document may suffice for a operation under all tests defined in this document may suffice for a
comparative evaluation but must not be interpreted that the protocol comparative evaluation but must not be interpreted that the protocol
skipping to change at page 12, line 27 skipping to change at line 528
Such evaluations are expected to be carried out in controlled Such evaluations are expected to be carried out in controlled
environments for limited numbers of parallel flows. As such, these environments for limited numbers of parallel flows. As such, these
evaluations are by definition limited and will not be able to evaluations are by definition limited and will not be able to
systematically consider possible interactions or very large groups of systematically consider possible interactions or very large groups of
communicating nodes under all possible circumstances, so that careful communicating nodes under all possible circumstances, so that careful
protocol design is advised to avoid incidentally contributing traffic protocol design is advised to avoid incidentally contributing traffic
that could lead to unstable networks, e.g., (local) congestion that could lead to unstable networks, e.g., (local) congestion
collapse. collapse.
This specification focuses on assessing the regular operation of the This specification focuses on assessing the regular operation of the
protocols and algorithms under considerations. It does not suggest protocols and algorithms under consideration. It does not suggest
checks against malicious use of the protocols -- by the sender, the checks against malicious use of the protocols -- by the sender, the
receiver, or intermediate parties, e.g., through faked, dropped, receiver, or intermediate parties, e.g., through faked, dropped,
replicated, or modified congestion signals. It is up to the protocol replicated, or modified congestion signals. It is up to the protocol
specifications themselves to ensure that authenticity, integrity, specifications themselves to ensure that authenticity, integrity,
and/or plausibility of received signals are checked and the and/or plausibility of received signals are checked, and the
appropriate actions (or non-actions) are taken. appropriate actions (or non-actions) are taken.
7. IANA Considerations 7. IANA Considerations
There are no IANA impacts in this memo. This document has no IANA actions.
8. Contributors
The content and concepts within this document are a product of the
discussion carried out in the Design Team.
Michael Ramalho provided the text for the Jitter model.
9. Acknowledgments
Much of this document is derived from previous work on congestion
control at the IETF.
The authors would like to thank Harald Alvestrand, Anna Brunstrom,
Luca De Cicco, Wesley Eddy, Lars Eggert, Kevin Gross, Vinayak Hegde,
Randell Jesup, Mirja Kuehlewind, Karen Nielsen, Piers O'Hanlon, Colin
Perkins, Michael Ramalho, Zaheduzzaman Sarker, Timothy B.
Terriberry, Michael Welzl, Mo Zanaty, and Xiaoqing Zhu for providing
valuable feedback on earlier versions of this draft. Additionally,
also thank the participants of the design team for their comments and
discussion related to the evaluation criteria.
10. References
10.1. Normative References 8. References
[I-D.ietf-rmcat-cc-requirements] 8.1. Normative References
Jesup, R. and Z. Sarker, "Congestion Control Requirements
for Interactive Real-Time Media", draft-ietf-rmcat-cc-
requirements-09 (work in progress), December 2014.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <https://www.rfc-editor.org/info/rfc3550>. July 2003, <https://www.rfc-editor.org/info/rfc3550>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551, Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003, DOI 10.17487/RFC3551, July 2003,
<https://www.rfc-editor.org/info/rfc3551>. <https://www.rfc-editor.org/info/rfc3551>.
skipping to change at page 14, line 10 skipping to change at line 580
[RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control: [RFC8083] Perkins, C. and V. Singh, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", RFC 8083, Circuit Breakers for Unicast RTP Sessions", RFC 8083,
DOI 10.17487/RFC8083, March 2017, DOI 10.17487/RFC8083, March 2017,
<https://www.rfc-editor.org/info/rfc8083>. <https://www.rfc-editor.org/info/rfc8083>.
[RFC8593] Zhu, X., Mena, S., and Z. Sarker, "Video Traffic Models [RFC8593] Zhu, X., Mena, S., and Z. Sarker, "Video Traffic Models
for RTP Congestion Control Evaluations", RFC 8593, for RTP Congestion Control Evaluations", RFC 8593,
DOI 10.17487/RFC8593, May 2019, DOI 10.17487/RFC8593, May 2019,
<https://www.rfc-editor.org/info/rfc8593>. <https://www.rfc-editor.org/info/rfc8593>.
10.2. Informative References [RFC8836] Jesup, R. and Z. Sarker, Ed., "Congestion Control
Requirements for Interactive Real-Time Media", RFC 8836,
DOI 10.17487/RFC8836, January 2021,
<https://www.rfc-editor.org/info/rfc8836>.
8.2. Informative References
[gilbert-elliott] [gilbert-elliott]
Hasslinger, G. and O. Hohlfeld, "The Gilbert-Elliott Model Hasslinger, G. and O. Hohlfeld, "The Gilbert-Elliott Model
for Packet Loss in Real Time Services on the Internet", for Packet Loss in Real Time Services on the Internet",
14th GI/ITG Conference - Measurement, Modelling and 14th GI/ITG Conference - Measurement, Modelling and
Evalutation of Computer and Communication Systems , 3 Evalutation [sic] of Computer and Communication Systems,
2008. March 2008,
<https://ieeexplore.ieee.org/document/5755057>.
[I-D.ietf-netvc-testing] [netvc-testing]
Daede, T., Norkin, A., and I. Brailovskiy, "Video Codec Daede, T., Norkin, A., and I. Brailovskiy, "Video Codec
Testing and Quality Measurement", draft-ietf-netvc- Testing and Quality Measurement", Work in Progress,
testing-09 (work in progress), January 2020. Internet-Draft, draft-ietf-netvc-testing-09, 31 January
2020,
[I-D.ietf-rmcat-eval-test] <https://tools.ietf.org/html/draft-ietf-netvc-testing-09>.
Sarker, Z., Singh, V., Zhu, X., and M. Ramalho, "Test
Cases for Evaluating RMCAT Proposals", draft-ietf-rmcat-
eval-test-10 (work in progress), May 2019.
[I-D.ietf-rmcat-wireless-tests]
Sarker, Z., Zhu, X., and J. Fu, "Evaluation Test Cases for
Interactive Real-Time Media over Wireless Networks",
draft-ietf-rmcat-wireless-tests-11 (work in progress),
March 2020.
[RFC5033] Floyd, S. and M. Allman, "Specifying New Congestion [RFC5033] Floyd, S. and M. Allman, "Specifying New Congestion
Control Algorithms", BCP 133, RFC 5033, Control Algorithms", BCP 133, RFC 5033,
DOI 10.17487/RFC5033, August 2007, DOI 10.17487/RFC5033, August 2007,
<https://www.rfc-editor.org/info/rfc5033>. <https://www.rfc-editor.org/info/rfc5033>.
[RFC5166] Floyd, S., Ed., "Metrics for the Evaluation of Congestion [RFC5166] Floyd, S., Ed., "Metrics for the Evaluation of Congestion
Control Mechanisms", RFC 5166, DOI 10.17487/RFC5166, March Control Mechanisms", RFC 5166, DOI 10.17487/RFC5166, March
2008, <https://www.rfc-editor.org/info/rfc5166>. 2008, <https://www.rfc-editor.org/info/rfc5166>.
[tcpdump] "Homepage of tcpdump and libpcap", [RFC8312] Rhee, I., Xu, L., Ha, S., Zimmermann, A., Eggert, L., and
https://www.tcpdump.org/index.html . R. Scheffenegger, "CUBIC for Fast Long-Distance Networks",
RFC 8312, DOI 10.17487/RFC8312, February 2018,
[wireshark] <https://www.rfc-editor.org/info/rfc8312>.
"Homepage of Wireshark", https://www.wireshark.org .
[xiph-seq]
Daede, T., "Video Test Media Set",
https://media.xiph.org/video/derf/ .
Appendix A. Change Log
Note to the RFC-Editor: please remove this section prior to
publication as an RFC.
A.1. Changes in draft-ietf-rmcat-eval-criteria-07
Updated the draft according to the discussion at IETF-101.
o Updated the discussion on fairness. Thanks to Xiaoqing Zhu for
providing text.
o Fixed a simple loss model and provided pointers to more
sophisticated ones.
o Fixed the choice of the jitter model.
A.2. Changes in draft-ietf-rmcat-eval-criteria-06
o Updated Jitter.
A.3. Changes in draft-ietf-rmcat-eval-criteria-05
o Improved text surrounding wireless tests, video sequences, and
short-TCP model.
A.4. Changes in draft-ietf-rmcat-eval-criteria-04
o Removed the guidelines section, as most of the sections are now
covered: wireless tests, video model, etc.
o Improved Short TCP model based on the suggestion to use
httparchive.org.
A.5. Changes in draft-ietf-rmcat-eval-criteria-03
o Keep-alive version.
o Moved link parameters and traffic models from eval-test
A.6. Changes in draft-ietf-rmcat-eval-criteria-02
o Incorporated fairness test as a working test.
o Updated text on mimimum evaluation requirements.
A.7. Changes in draft-ietf-rmcat-eval-criteria-01
o Removed Appendix B.
o Removed Section on Evaluation Parameters.
A.8. Changes in draft-ietf-rmcat-eval-criteria-00
o Updated references.
o Resubmitted as WG draft.
A.9. Changes in draft-singh-rmcat-cc-eval-04
o Incorporate feedback from IETF 87, Berlin.
o Clarified metrics: convergence time, bandwidth utilization.
o Changed fairness criteria to fairness test.
o Added measuring pre- and post-repair loss.
o Added open issue of measuring video quality to appendix.
o clarified use of DropTail and AQM.
o Updated text in "Minimum Requirements for Evaluation"
A.10. Changes in draft-singh-rmcat-cc-eval-03
o Incorporate the discussion within the design team.
o Added a section on evaluation parameters, it describes the flow
and network characteristics.
o Added Appendix with self-fairness experiment. [RFC8867] Sarker, Z., Singh, V., Zhu, X., and M. Ramalho, "Test
Cases for Evaluating Congestion Control for Interactive
Real-Time Media", RFC 8867, DOI 10.17487/RFC8867, January
2021, <https://www.rfc-editor.org/info/rfc8867>.
o Changed bottleneck parameters from a proposal to an example set. [RFC8869] Sarker, Z., Zhu, X., and J. Fu, "Evaluation Test Cases for
Interactive Real-Time Media over Wireless Networks",
RFC 8869, DOI 10.17487/RFC8869, January 2021,
<https://www.rfc-editor.org/info/rfc8869>.
o [tcpdump] "Homepage of tcpdump and libpcap",
<https://www.tcpdump.org/index.html>.
A.11. Changes in draft-singh-rmcat-cc-eval-02 [wireshark]
"Homepage of Wireshark", <https://www.wireshark.org>.
o Added scenario descriptions. [xiph-seq] Daede, T., "Video Test Media Set",
<https://media.xiph.org/video/derf/>.
A.12. Changes in draft-singh-rmcat-cc-eval-01 Contributors
o Removed QoE metrics. The content and concepts within this document are a product of the
discussion carried out in the Design Team.
o Changed stability to steady-state. Michael Ramalho provided the text for the jitter models
(Section 4.5).
o Added measuring impact against few and many flows. Acknowledgments
o Added guideline for idle and data-limited periods. Much of this document is derived from previous work on congestion
control at the IETF.
o Added reference to TCP evaluation suite in example evaluation The authors would like to thank Harald Alvestrand, Anna Brunstrom,
scenarios. Luca De Cicco, Wesley Eddy, Lars Eggert, Kevin Gross, Vinayak Hegde,
Randell Jesup, Mirja Kühlewind, Karen Nielsen, Piers O'Hanlon, Colin
Perkins, Michael Ramalho, Zaheduzzaman Sarker, Timothy B. Terriberry,
Michael Welzl, Mo Zanaty, and Xiaoqing Zhu for providing valuable
feedback on draft versions of this document. Additionally, thanks to
the participants of the Design Team for their comments and discussion
related to the evaluation criteria.
Authors' Addresses Authors' Addresses
Varun Singh Varun Singh
CALLSTATS I/O Oy CALLSTATS I/O Oy
Runeberginkatu 4c A 4 Rauhankatu 11 C
Helsinki 00100 FI-00100 Helsinki
Finland Finland
Email: varun.singh@iki.fi Email: varun.singh@iki.fi
URI: https://www.callstats.io/about URI: https://www.callstats.io/
Joerg Ott Jörg Ott
Technical University of Munich Technical University of Munich
Faculty of Informatics Department of Informatics
Chair of Connected Mobility
Boltzmannstrasse 3 Boltzmannstrasse 3
Garching bei Muenchen, DE 85748 85748 Garching
Germany Germany
Email: ott@in.tum.de Email: ott@in.tum.de
Stefan Holmer Stefan Holmer
Google Google
Kungsbron 2 Kungsbron 2
Stockholm 11122 SE-11122 Stockholm
Sweden Sweden
Email: holmer@google.com Email: holmer@google.com
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