Internet DRAFT - draft-ietf-xrblock-rtcweb-rtcp-xr-metrics
draft-ietf-xrblock-rtcweb-rtcp-xr-metrics
XR Block Working Group V. Singh
Internet-Draft callstats.io
Intended status: Informational R. Huang
Expires: November 26, 2018 R. Even
Huawei
D. Romascanu
Individual
L. Deng
China Mobile
May 25, 2018
Considerations for Selecting RTCP Extended Report (XR) Metrics for the
WebRTC Statistics API
draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-10
Abstract
This document describes monitoring features related to media streams
in Web real-time communication (WebRTC). It provides a list of RTCP
Sender Report, Receiver Report and Extended Report metrics, which may
need to be supported by RTP implementations in some diverse
environments. It lists a set of identifiers for the WebRTC's
statistics API. These identifiers are a set of RTCP SR, RR, and XR
metrics related to the transport of multimedia flows.
Status of This Memo
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This Internet-Draft will expire on January 21, 2018.
Copyright Notice
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Copyright (c) 2018 IETF Trust and the persons identified as the
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. RTP Statistics in WebRTC Implementations . . . . . . . . . . 3
4. Considerations for Impact of Measurement Interval . . . . . . 4
5. Candidate Metrics . . . . . . . . . . . . . . . . . . . . . . 5
5.1. Network Impact Metrics . . . . . . . . . . . . . . . . . 5
5.1.1. Loss and Discard Packet Count Metric . . . . . . . . 5
5.1.2. Burst/Gap Pattern Metrics for Loss and Discard . . . 6
5.1.3. Run Length Encoded Metrics for Loss, Discard . . . . 7
5.2. Application Impact Metrics . . . . . . . . . . . . . . . 7
5.2.1. Discard Octets Metric . . . . . . . . . . . . . . . . 7
5.2.2. Frame Impairment Summary Metrics . . . . . . . . . . 8
5.2.3. Jitter Buffer Metrics . . . . . . . . . . . . . . . . 8
5.3. Recovery metrics . . . . . . . . . . . . . . . . . . . . 9
5.3.1. Post-repair Packet Count Metrics . . . . . . . . . . 9
5.3.2. Run Length Encoded Metric for Post-repair . . . . . . 9
6. Identifiers from Sender, Receiver, and Extended Report Blocks 10
6.1. Cumulative Number of Packets and Octets Sent . . . . . . 10
6.2. Cumulative Number of Packets and Octets Received . . . . 10
6.3. Cumulative Number of Packets Lost . . . . . . . . . . . . 11
6.4. Interval Packet Loss and Jitter . . . . . . . . . . . . . 11
6.5. Cumulative Number of Packets and Octets Discarded . . . . 11
6.6. Cumulative Number of Packets Repaired . . . . . . . . . . 11
6.7. Burst Packet Loss and Burst Discards . . . . . . . . . . 11
6.8. Burst/Gap Rates . . . . . . . . . . . . . . . . . . . . . 12
6.9. Frame Impairment Metrics . . . . . . . . . . . . . . . . 12
7. Adding new metrics to WebRTC Statistics API . . . . . . . . . 13
8. Security Considerations . . . . . . . . . . . . . . . . . . . 13
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 13
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 13
10.1. Normative References . . . . . . . . . . . . . . . . . . 13
10.2. Informative References . . . . . . . . . . . . . . . . . 15
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Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16
1. Introduction
Web real-time communication (WebRTC) [I-D.ietf-rtcweb-overview]
deployments are emerging and applications need to be able to estimate
the service quality. If sufficient information (metrics or
statistics) is provided to the application, it can attempt to improve
the media quality. [RFC7478] specifies a requirement for statistics:
F38 The browser must be able to collect statistics, related to the
transport of audio and video between peers, needed to estimate
quality of experience.
The WebRTC Stats API [W3C.WD-webrtc-stats] currently lists metrics
reported in the RTCP Sender and Receiver Report (SR/RR) [RFC3550] to
fulfill this requirement. However, the basic metrics from RTCP SR/RR
are not sufficient for precise quality monitoring, or diagnosing
potential issues.
Standards such as "RTP Control Protocol Extended Reports (RTCP XR)"
[RFC3611] as well as other extensions standardized in the XRBLOCK
working group, e.g., burst/gap loss metric reporting [RFC6958],
burst/gap discard metric reporting [RFC7003], and etc., have been
produced for the purpose of collecting and reporting performance
metrics from RTP endpoint devices that can be used to have a end-to-
end service visibility and measure the delivering quality in various
RTP services. These metrics are able to complement those in
[RFC3550].
In this document, we provide rationale for choosing additional RTP
metrics for the WebRTC getStats() API [W3C.WD-webrtc]. All
identifiers proposed in this document are recommended to be
implemented by an WebRTC endpoint. An endpoint may choose not to
expose an identifier if it does not implement the corresponding RTCP
Report. This document only considers RTP layer metrics. Other
metrics, e.g., IP layer metrics, are out of scope.
2. Terminology
ReportGroup: It is a set of metrics identified by a common
Synchronization source (SSRC).
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3. RTP Statistics in WebRTC Implementations
The RTCP Sender Reports (SRs) and Receiver Reports (RRs) [RFC3550]
expose the basic metrics for the local and remote media streams.
However, these metrics provide only partial or limited information,
which may not be sufficient for diagnosing problems or quality
monitoring. For example, it may be useful to distinguish between
packets lost and packets discarded due to late arrival. Even though
they have the same impact on the multimedia quality, it helps in
identifying and diagnosing problems. RTP Control Protocol Extended
Reports (XRs) [RFC3611] and other extensions discussed in the XRBLOCK
working group provide more detailed statistics, which complement the
basic metrics reported in the RTCP SR and RRs.
The WebRTC application extracts the statistic from the browser by
querying the getStats() API [W3C.WD-webrtc]. The browser can easily
report the local variables i.e., the statistics related to the
outgoing RTP media streams and the incoming RTP media streams.
However, without the support of RTCP XRs or some other signaling
mechanism, the WebRTC application cannot expose the remote endpoints'
statistics. [I-D.ietf-rtcweb-rtp-usage] does not mandate the use of
any RTCP XRs and their usage is optional. If the use of RTCP XRs is
successfully negotiated between endpoints (via SDP), thereafter the
application has access to both local and remote statistics.
Alternatively, once the WebRTC application gets the local
information, they can report it to an application server or a third-
party monitoring system, which provides quality estimations or
diagnosis services for application developers. The exchange of
statistics between endpoints or between a monitoring server and an
endpoint is outside the scope of this document.
4. Considerations for Impact of Measurement Interval
RTCP extensions like RTCP XR usually share the same timing interval
with the RTCP SR/RR, i.e., they are sent as compound packets,
together with the RTCP SR/RR. Alternatively, if the RTCP XR uses a
different measurement interval, all XRs using the same measurement
interval are compounded together and the measurement interval is
indicated in a specific measurement information block defined in
[RFC6776].
When using WebRTC getStats() APIs (see section 7 of [W3C.WD-webrtc]),
the applications can query this information at arbitrary intervals.
For the statistics reported by the remote endpoint, e.g., those
conveyed in an RTCP SR/RR/XR, these will not change until the next
RTCP report is received. However, statistics generated by the local
endpoint have no such restrictions as long as the endpoint is sending
and receiving media. For example, an application may choose to poll
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the stack for statistics every 1 second. In this case the underlying
stack local will return the current snapshot of the local statistics
(for incoming and outgoing media streams). However, it may return
the same remote statistics as before for the remote statistics, as no
new RTCP reports may have been received in the past 1 second. This
can occur when the polling interval is shorter than the average RTCP
reporting interval.
5. Candidate Metrics
Since the following metrics are all defined in RTCP XR, which is not
mandated in WebRTC, all of them are local. However, if RTCP XR is
supported by negotiation between two browsers, the following metrics
can also be generated remotely and be sent to local by RTCP XR
packets.
The following metrics are classified into 3 categories: network
impact metrics, application impact metrics and recovery metrics.
Network impact metrics are the statistics recording the information
only for network transmission. They are useful for network problem
diagnosis. Application impact metrics mainly collect the information
from the viewpoint of application, e.g., bit rate, frame rate or
jitter buffers. Recovery metrics reflect how well the repair
mechanisms perform, e.g. loss concealment, retransmission or Forward
Error Correction (FEC). All of the 3 types of metrics are useful for
quality estimations of services in WebRTC implementations. WebRTC
applications can use these metrics to calculate the estimated Mean
Opinion Score (MOS) [ITU-T P.800.1] values or Media Delivery Index
(MDI) [RFC4445] for their services.
5.1. Network Impact Metrics
5.1.1. Loss and Discard Packet Count Metric
In multimedia transport, packets which are received abnormally are
classified into 3 types: lost, discarded and duplicate packets.
Packet loss may be caused by network device breakdown, bit-error
corruption or network congestion (packets dropped by an intermediate
router queue). Duplicate packets may be a result of network delays
that causes the sender to retransmit the original packets. Discarded
packets are packets that have been delayed long enough (perhaps they
missed the playout time) and are considered useless by the receiver.
Lost and discarded packets cause problems for multimedia services, as
missing data and long delays can cause degradation in service
quality, e.g., missing large blocks of contiguous packets (lost or
discarded) may cause choppy audio, and long network transmission
delay time may cause audio or video buffering. The RTCP SR/RR
defines a metric for counting the total number of RTP data packets
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that have been lost since the beginning of reception. But this
statistic does not distinguish lost packets from discarded and
duplicate packets. Packets that arrive late will be discarded and
are not reported as lost, and duplicate packets will be regarded as a
normally received packet. Hence, the loss metric can be misleading
if many duplicate packets are received or packets are discarded,
which causes the quality of the media transport to appear okay from
the statistic point of view, but meanwhile the users may actually be
experiencing bad service quality. So in such cases, it is better to
use more accurate metrics in addition to those defined in RTCP SR/RR.
The lost packets and duplicated packets metrics defined in Statistics
Summary Report Block of [RFC3611] extend the information of loss
carried in standard RTCP SR/RR. They explicitly give an account of
lost and duplicated packets. Lost packet counts are useful for
network problem diagnosis. It is better to use the loss packets
metrics of [RFC3611] to indicate the packet lost count instead of the
cumulative number of packets lost metric of [RFC3550]. Duplicated
packets are usually rare and have little effect on QoS evaluation. So
it may not be suitable for use in WebRTC.
Using loss metrics without considering discard metrics may result in
inaccurate quality evaluation, as packet discard due to jitter is
often more prevalent than packet loss in modern IP networks. The
discarded metric specified in [RFC7002] counts the number of packets
discarded due to the jitter. It augments the loss statistics metrics
specified in standard RTCP SR/RR. For those RTCWEB services with
jitter buffers requiring precise quality evaluation and accurate
troubleshooting, this metric is useful as a complement to the metrics
of RTCP SR/RR.
5.1.2. Burst/Gap Pattern Metrics for Loss and Discard
RTCP SR/RR defines coarse metrics regarding loss statistics: the
metrics are all about per call statistics and are not detailed enough
to capture the transitory nature of some impairments like bursty
packet loss. Even if the average packet loss rate is low, the lost
packets may occur during short dense periods, resulting in short
periods of degraded quality. Bursts cause lower quality experience
than the non-bursts for low packet loss rates, whereas for high
packet loss rates the converse is true. So capturing burst gap
information is very helpful for quality evaluation and locating
impairments. If the WebRTC application needs to evaluate the
services quality, burst gap metrics provides more accurate
information than RTCP SR/RR.
[RFC3611] introduces burst gap metrics in VoIP report block. These
metrics record the density and duration of burst and gap periods,
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which are helpful in isolating network problems since bursts
correspond to periods of time during which the packet loss/discard
rate is high enough to produce noticeable degradation in audio or
video quality. Burst gap related metrics are also introduced in
[RFC7003] and [RFC6958] which define two new report blocks for usage
in a range of RTP applications beyond those described in [RFC3611].
These metrics distinguish discarded packets from loss packets that
occur in the bursts period and provides more information for
diagnosing network problems. Additionally, the block reports the
frequency of burst events which is useful information for evaluating
the quality of experience. Hence, if WebRTC applications need to do
quality evaluation and observe when and why quality degrades, these
metrics should be considered.
5.1.3. Run Length Encoded Metrics for Loss, Discard
Run-length encoding uses a bit vector to encode information about the
packet. Each bit in the vector represents a packet and depending on
the signaled metric it defines if the packet was lost, duplicated,
discarded, or repaired. An endpoint typically uses the run length
encoding to accurately communicate the status of each packet in the
interval to the other endpoint. [RFC3611], [RFC7097] define run-
length encoding for lost and duplicate packets, and discarded
packets, respectively.
The WebRTC application could benefit from the additional information.
If losses occur after discards, an endpoint may be able to correlate
the two run length vectors to identify congestion-related losses,
e.g., a router queue became overloaded causing delays and then
overflowed. If the losses are independent, it may indicate bit-error
corruption. For the WebRTC Stats API [W3C.WD-webrtc-stats], these
types of metrics are not recommended for use due to the large amount
of data and the computation involved.
5.2. Application Impact Metrics
5.2.1. Discarded Octets Metric
The metric reports the cumulative size of the packets discarded in
the interval. It is complementary to number of discarded packets. An
application measures sent octets and received octets to calculate
sending rate and receiving rate, respectively. The application can
calculate the actual bit rate in a particular interval by subtracting
the discarded octets from the received octets.
For WebRTC, discarded octets supplements the sent and received octets
and provides an accurate method for calculating the actual bit rate
which is an important parameter to reflect the quality of the media.
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The discarded bytes metric is defined in [RFC7243].
5.2.2. Frame Impairment Summary Metrics
RTP has different framing mechanisms for different payload types. For
audio streams, a single RTP packet may contain one or multiple audio
frames. On the other hand, in video streams, a single video frame
may be transmitted in multiple RTP packets. The size of each packet
is limited by the Maximum Transmission Unit (MTU) of the underlying
network. However, statistics from standard SR/RR only collect
information from transport layer, which may not fully reflect the
quality observed by the application. Video is typically encoded
using two frame types i.e., key frames and derived frames. Key
frames are normally just spatially compressed, i.e., without
prediction from other pictures. The derived frames are temporally
compressed, i.e., depend on the key frame for decoding. Hence, key
frames are much larger in size than derived frames. The loss of
these key frames results in a substantial reduction in video quality.
Thus it is reasonable to consider this application layer information
in WebRTC implementations, which influence sender strategies to
mitigate the problem or require the accurate assessment of users'
quality of experience.
The metrics in this category include: number of discarded key frames,
number of lost key frames, number of discarded derived frames, number
of lost derived frames. These metrics can be used to calculate Media
Loss Rate (MLR) of MDI [RFC4445]. Details of the definition of these
metrics are described in [RFC7003]. Additionally, the metric
provides the rendered frame rate, an important parameter for quality
estimation.
5.2.3. Jitter Buffer Metrics
The size of the jitter buffer affects the end-to-end delay on the
network and also the packet discard rate. When the buffer size is
too small, slower packets are not played out and dropped, while when
the buffer size is too large, packets are held longer than necessary
and consequently reduce conversational quality. Measurement of
jitter buffer should not be ignored in the evaluation of end user
perception of conversational quality. Jitter buffer related metrics,
such as maximum and nominal jitter buffer, could be used to show how
the jitter buffer behaves at the receiving endpoint. They are useful
for providing better end-user quality of experience (QoE) when jitter
buffer factors are used as inputs to calculate estimated MOS values.
Thus for those cases, jitter buffer metrics should be considered.
The definition of these metrics is provided in [RFC7005].
5.3. Recovery metrics
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This document does not consider concealment metrics [RFC7294] as part
of recovery metrics.
5.3.1. Post-repair Packet Count Metrics
Web applications can support certain RTP error-resilience mechanisms
following the recommendations specified in [draft-ietf-rtcweb-rtp-
usage]. For these web applications using repair mechanisms, providing
some statistic information for the performance of their repair
mechanisms could help to have a more accurate quality evaluation.
The unrepaired packet count and repaired loss count defined in
[RFC7509] provide the recovery information of the error-resilience
mechanisms to the monitoring application or the sending endpoint. The
endpoint can use these metrics to ascertain the ratio of repaired
packets to lost packets. Including post-repair packet count metrics
helps the application evaluate the effectiveness of the applied
repair mechanisms.
5.3.2. Run Length Encoded Metric for Post-repair
[RFC5725] defines run-length encoding for post-repair packets. When
using error-resilience mechanisms, the endpoint can correlate the
loss run length with this metric to ascertain where the losses and
repairs occurred in the interval. This provides more accurate
information for recovery mechanisms evaluation than those in Section
5.3.1. However, it is not suggested to use due to their enormous
amount of data when RTCP XR are supported.
For WebRTC, the application may benefit from the additional
information. If losses occur after discards, an endpoint may be able
to correlate the two run length vectors to identify congestion-
related losses, e.g., a router queue became overloaded causing delays
and then overflowed. If the losses are independent, it may indicate
bit-error corruption. Lastly, when using error-resilience
mechanisms, the endpoint can correlate the loss and post-repair run
lengths to ascertain where the losses and repairs occurred in the
interval. For example, consecutive losses are likely not to be
repaired by a simple FEC scheme.
6. Identifiers from Sender, Receiver, and Extended Report Blocks
This document describes a list of metrics and corresponding
identifiers relevant to RTP media in WebRTC. This group of
identifiers are defined on a ReportGroup corresponding to a
synchronization source (SSRC). In practice the application needs to
be able to query the statistic identifiers on both an incoming
(remote) and outgoing (local) media stream. Since sending and
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receiving SR and RR are mandatory, the metrics defined in the SR and
RR report blocks are always available. For XR metrics, it depends on
two factors: 1) if it is measured at the endpoint, 2) if it is
reported by the endpoint in an XR report. If a metric is only
measured by the endpoint and not reported, the metrics will only be
available for the incoming (remote) media stream. Alternatively, if
the corresponding metric is also reported in an XR report, it will be
available for both the incoming (remote) and outgoing (local) media
stream.
For a remote statistic, the timestamp represents the timestamp from
an incoming SR/RR/XR packet. Conversely, for a local statistic, it
refers to the current timestamp generated by the local clock
(typically the POSIX timestamp, i.e., milliseconds since Jan 1,
1970).
As per [RFC3550], the octets metrics represent the payload size
(i.e., not including header or padding).
6.1. Cumulative Number of Packets and Octets Sent
Name: packetsSent
Definition: section 6.4.1 in [RFC3550].
Name: bytesSent
Definition: section 6.4.1 in [RFC3550].
6.2. Cumulative Number of Packets and Octets Received
Name: packetsReceived
Definition: section 6.4.1 in [RFC3550].
Name: bytesReceived
Definition: section 6.4.1 in [RFC3550].
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6.3. Cumulative Number of Packets Lost
Name: packetsLost
Definition: section 6.4.1 in [RFC3550].
6.4. Interval Packet Loss and Jitter
Name: jitter
Definition: section 6.4.1 in [RFC3550].
Name: fractionLost
Definition: section 6.4.1 in [RFC3550].
6.5. Cumulative Number of Packets and Octets Discarded
Name: packetsDiscarded
Definition: The cumulative number of RTP packets discarded due to
late or early-arrival, Appendix A (a) of [RFC7002].
Name: bytesDiscarded
Definition: The cumulative number of octets discarded due to late or
early-arrival, Appendix A of [RFC7243].
6.6. Cumulative Number of Packets Repaired
Name: packetsRepaired
Definition: The cumulative number of lost RTP packets repaired after
applying a error-resilience mechanism, Appendix A (b) of [RFC7509].
To clarify, the value is upper bound to the cumulative number of lost
packets.
6.7. Burst Packet Loss and Burst Discards
Name: burstPacketsLost
Definition: The cumulative number of RTP packets lost during loss
bursts, Appendix A (c) of [RFC6958].
Name: burstLossCount
Definition: The cumulative number of bursts of lost RTP packets,
Appendix A (e) of [RFC6958].
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Name: burstPacketsDiscarded
Definition: The cumulative number of RTP packets discarded during
discard bursts, Appendix A (b) of [RFC7003].
Name: burstDiscardCount
Definition: The cumulative number of bursts of discarded RTP packets,
Appendix A (e) of [RFC8015].
[RFC3611] recommends a Gmin (threshold) value of 16 for classifying
packet loss or discard burst.
6.8. Burst/Gap Rates
Name: burstLossRate
Definition: The fraction of RTP packets lost during bursts,
Appendix A (a) of [RFC7004].
Name: gapLossRate
Definition: The fraction of RTP packets lost during gaps, Appendix A
(b) of [RFC7004].
Name: burstDiscardRate
Definition: The fraction of RTP packets discarded during bursts,
Appendix A (e) of [RFC7004].
Name: gapDiscardRate
Definition: The fraction of RTP packets discarded during gaps,
Appendix A (f) of [RFC7004].
6.9. Frame Impairment Metrics
Name: framesLost
Definition: The cumulative number of full frames lost, Appendix A (i)
of [RFC7004].
Name: framesCorrupted
Definition: The cumulative number of frames partially lost,
Appendix A (j) of [RFC7004].
Name: framesDropped
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Definition: The cumulative number of full frames discarded,
Appendix A (g) of [RFC7004].
Name: framesSent
Definition: The cumulative number of frames sent.
Name: framesReceived
Definition: The cumulative number of partial or full frames received.
7. Adding new metrics to WebRTC Statistics API
During the progress of this work, the metrics defined in this draft
have already been added to the W3C WebRTC specification. The working
process to add new metrics for future is to create an issue or pull
request on the repository of the W3C WebRTC specification
(https://github.com/w3c/webrtc-stats).
8. Security Considerations
This document focuses on listing the RTCP XR metrics defined in the
corresponding RTCP reporting extensions and do not give rise to any
new security vulnerabilities beyond those described in [RFC3611] and
[RFC6792].
The overall security considerations for RTP used in WebRTC
applications is described in [I-D.ietf-rtcweb-rtp-usage] and
[I-D.ietf-rtcweb-security], which are also apply to this memo.
9. IANA Consideration
This document requests no action by IANA.
10. Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand, Al
Morton, Colin Perkins, and Shida Schubert for their valuable comments
and suggestions on earlier version of this document.
11. References
11.1. Normative References
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
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[RFC3611] Friedman, T., Ed., Caceres, R., Ed., and A. Clark, Ed.,
"RTP Control Protocol Extended Reports (RTCP XR)",
RFC 3611, DOI 10.17487/RFC3611, November 2003,
<http://www.rfc-editor.org/info/rfc3611>.
[RFC5725] Begen, A., Hsu, D., and M. Lague, "Post-Repair Loss RLE
Report Block Type for RTP Control Protocol (RTCP) Extended
Reports (XRs)", RFC 5725, DOI 10.17487/RFC5725, February
2010, <http://www.rfc-editor.org/info/rfc5725>.
[RFC6776] Clark, A. and Q. Wu, "Measurement Identity and Information
Reporting Using a Source Description (SDES) Item and an
RTCP Extended Report (XR) Block", RFC 6776,
DOI 10.17487/RFC6776, October 2012,
<http://www.rfc-editor.org/info/rfc6776>.
[RFC6792] Qu, Q. and P. Arden, "Guidelines for Use of the RTP
Monitoring Framework", RFC 6792, November 2012,
<http://www.rfc-editor.org/info/rfc6792>
[RFC6958] Clark, A., Zhang, S., Zhao, J., and Q. Wu, Ed., "RTP
Control Protocol (RTCP) Extended Report (XR) Block for
Burst/Gap Loss Metric Reporting", RFC 6958,
DOI 10.17487/RFC6958, May 2013,
<http://www.rfc-editor.org/info/rfc6958>.
[RFC7002] Clark, A., Zorn, G., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for Discard Count Metric
Reporting", RFC 7002, DOI 10.17487/RFC7002, September
2013, <http://www.rfc-editor.org/info/rfc7002>.
[RFC7003] Clark, A., Huang, R., and Q. Wu, Ed., "RTP Control
Protocol (RTCP) Extended Report (XR) Block for Burst/Gap
Discard Metric Reporting", RFC 7003, DOI 10.17487/RFC7003,
September 2013, <http://www.rfc-editor.org/info/rfc7003>.
[RFC7004] Zorn, G., Schott, R., Wu, Q., Ed., and R. Huang, "RTP
Control Protocol (RTCP) Extended Report (XR) Blocks for
Summary Statistics Metrics Reporting", RFC 7004,
DOI 10.17487/RFC7004, September 2013,
<http://www.rfc-editor.org/info/rfc7004>.
[RFC7005] Clark, A., Singh, V., and Q. Wu, "RTP Control Protocol
(RTCP) Extended Report (XR) Block for De-Jitter Buffer
Metric Reporting", RFC 7005, DOI 10.17487/RFC7005,
September 2013, <http://www.rfc-editor.org/info/rfc7005>.
[RFC7097] Ott, J., Singh, V., Ed., and I. Curcio, "RTP Control
Protocol (RTCP) Extended Report (XR) for RLE of Discarded
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Packets", RFC 7097, DOI 10.17487/RFC7097, January 2014,
<http://www.rfc-editor.org/info/rfc7097>.
[RFC7243] Singh, V., Ed., Ott, J., and I. Curcio, "RTP Control
Protocol (RTCP) Extended Report (XR) Block for the Bytes
Discarded Metric", RFC 7243, DOI 10.17487/RFC7243, May
2014, <http://www.rfc-editor.org/info/rfc7243>.
[RFC7509] Huang, R. and V. Singh, "RTP Control Protocol (RTCP)
Extended Report (XR) for Post-Repair Loss Count Metrics",
RFC 7509, DOI 10.17487/RFC7509, May 2015,
<http://www.rfc-editor.org/info/rfc7509>.
[RFC8015] Singh, V., Perkins, C., Clark, A., and R. Huang, "RTP
Control Protocol (RTCP) Extended Report (XR) Block for
Independent Reporting of Burst/Gap Discard Metrics",
RFC 8015, DOI 10.17487/RFC8015, November 2016,
<http://www.rfc-editor.org/info/rfc8015>.
11.2. Informative References
[I-D.ietf-rtcweb-overview] H. Alverstrand, "Overview: Real Time
Protocols for Browser-based Applications", draft-ietf-
rtcweb-overview-19 (work in progress), November 2017.
[ITU-T P.800.1] "Mean Opinion Score (MOS) terminology", ITU-T
P.800.1, July 2016.
[RFC4445] Welch, J. and J. Clark, "A Proposed Media Delivery Index
(MDI)", RFC4445, April 2006.
[I-D.ietf-rtcweb-rtp-usage] Perkins, C., Westerlund, M., and J. Ott,
"Web Real-Time Communication (WebRTC): Media Transport and
Use of RTP", draft-ietf-rtcweb-rtp-usage-26 (work in
progress), March 2016.
[I-D.ietf-rtcweb-security] Rescorla, E., "Security Considerations for
WebRTC", draft-ietf-rtcweb-security-10 (work in progress),
January 2018.
[RFC7294] Clark, A., Zorn, G., Bi, C. and Q. Wu, "RTP Control
Protocol (RTCP) Extended Report (XR) Blocks for
Concealment Metrics Reporting on Audio Applications", RFC
7294, July 2014, <http://www.rfc-editor.org/info/rfc7294>
[RFC7478] Holmberg, C., Hakansson, S., and G. Eriksson, "Web Real-
Time Communication Use Cases and Requirements", RFC 7478,
Singh, et al. Expires November 26, 2018 [Page 15]
Internet-Draft RTCP XR Metrics for RTCWEB May 25, 2018
DOI 10.17487/RFC7478, March 2015, <http://www.rfc-
editor.org/info/rfc7478>.
[W3C.WD-webrtc] Bergkvist, A., Burnett, C., Jennings, C., Narayanan,
A., Aboba B. and T. Brandstetter, "WebRTC 1.0: Real-time
Communication Between Browsers", World Wide Web Consortium
WD WD-webrtc-20171102, November 2017,
<https://www.w3.org/TR/2017/CR-webrtc-20171102/>.
[W3C.WD-webrtc-stats] Alvestrand, H. and V. Singh, "Identifiers for
WebRTC's Statistics API", World Wide Web Consortium WD WD-
webrtc-stats-20180519, May 2018,
<https://www.w3.org/TR/2018/WD-webrtc-stats-20180519/>.
Authors' Addresses
Varun Singh
CALLSTATS I/O Oy
Annankatu 31-33 C 42
Helsinki 00100
Finland
Email: varun@callstats.io
URI: https://www.callstats.io/about
Rachel Huang
Huawei
101 Software Avenue, Yuhua District
Nanjing, CN 210012
China
Email: rachel.huang@huawei.com
Roni Even
Huawei
14 David Hamelech
Tel Aviv 64953
Israel
Email: roni.even@huawei.com
Dan Romascanu
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Email: dromasca@gmail.com
Lingli Deng
China Mobile
Email: denglingli@chinamobile.com
Singh, et al. Expires November 26, 2018 [Page 17]
