< draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-06.txt   draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-07.txt >
XR Block Working Group V. Singh XR Block Working Group V. Singh
Internet-Draft callstats.io Internet-Draft callstats.io
Intended status: Standards Track R. Huang Intended status: Informational R. Huang
Expires: January 21, 2018 R. Even Expires: June 14, 2018 R. Even
Huawei Huawei
D. Romascanu D. Romascanu
Individual
L. Deng L. Deng
China Mobile China Mobile
July 20, 2017 December 11, 2017
Considerations for Selecting RTCP Extended Report (XR) Metrics for the Considerations for Selecting RTCP Extended Report (XR) Metrics for the
WebRTC Statistics API WebRTC Statistics API
draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-06 draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-07
Abstract Abstract
This document describes monitoring features related to media streams This document describes monitoring features related to media streams
in Web real-time communication (WebRTC). It provides a list of RTCP in Web real-time communication (WebRTC). It provides a list of RTCP
Sender Report, Receiver Report and Extended Report metrics, which may Sender Report, Receiver Report and Extended Report metrics, which may
need to be supported by RTP implementations in some diverse need to be supported by RTP implementations in some diverse
environments. It lists a set of identifiers for the WebRTC's environments. It lists a set of identifiers for the WebRTC's
statistics API. These identifiers are a set of RTCP SR, RR, and XR statistics API. These identifiers are a set of RTCP SR, RR, and XR
metrics related to the transport of multimedia flows. metrics related to the transport of multimedia flows.
skipping to change at page 2, line 49 skipping to change at page 3, line 4
6.6. Cumulative Number of Packets Repaired . . . . . . . . . . 11 6.6. Cumulative Number of Packets Repaired . . . . . . . . . . 11
6.7. Burst Packet Loss and Burst Discards . . . . . . . . . . 11 6.7. Burst Packet Loss and Burst Discards . . . . . . . . . . 11
6.8. Burst/Gap Rates . . . . . . . . . . . . . . . . . . . . . 12 6.8. Burst/Gap Rates . . . . . . . . . . . . . . . . . . . . . 12
6.9. Frame Impairment Metrics . . . . . . . . . . . . . . . . 12 6.9. Frame Impairment Metrics . . . . . . . . . . . . . . . . 12
7. Adding new metrics to WebRTC Statistics API . . . . . . . . . 13 7. Adding new metrics to WebRTC Statistics API . . . . . . . . . 13
8. Security Considerations . . . . . . . . . . . . . . . . . . . 13 8. Security Considerations . . . . . . . . . . . . . . . . . . . 13
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 13 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 13
10. References . . . . . . . . . . . . . . . . . . . . . . . . . 13 10. References . . . . . . . . . . . . . . . . . . . . . . . . . 13
10.1. Normative References . . . . . . . . . . . . . . . . . . 13 10.1. Normative References . . . . . . . . . . . . . . . . . . 13
10.2. Informative References . . . . . . . . . . . . . . . . . 15 10.2. Informative References . . . . . . . . . . . . . . . . . 15
Appendix A. Change Log . . . . . . . . . . . . . . . . . . . . . 16 Appendix A. Change Log . . . . . . . . . . . . . . . . . . . . . 16
A.1. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-05 A.1. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-05
and -06 . . . . . . . . . . . . . . . . . . . . . . . . . 16 and -06 . . . . . . . . . . . . . . . . . . . . . . . . . 16
A.2. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-04 . 16 A.2. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-04 . 16
A.3. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-02, A.3. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-02,
-03 . . . . . . . . . . . . . . . . . . . . . . . . . . . 16 -03 . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
A.4. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-01 . 16 A.4. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-01 . 16
A.5. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00 . 16 A.5. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00 . 16
A.6. changes in draft-huang-xrblock-rtcweb-rtcp-xr-metrics-04 16 A.6. changes in draft-huang-xrblock-rtcweb-rtcp-xr-metrics-04 16
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16
1. Introduction 1. Introduction
Web real-time communication (WebRTC) deployments are emerging and Web real-time communication (WebRTC) deployments are emerging and
applications need to be able to estimate the service quality. If applications need to be able to estimate the service quality. If
sufficient information (metrics or statistics) are provided to the sufficient information (metrics or statistics) is provided to the
applications, it can attempt to improve the media quality. [RFC7478] application, it can attempt to improve the media quality. [RFC7478]
specifies a requirement for statistics: specifies a requirement for statistics:
F38 The browser must be able to collect statistics, related to the F38 The browser must be able to collect statistics, related to the
transport of audio and video between peers, needed to estimate transport of audio and video between peers, needed to estimate
quality of experience. quality of experience.
The WebRTC Stats API [W3C.WD-webrtc-stats-20161214] currently lists The WebRTC Stats API [W3C.WD-webrtc-stats-20161214] currently lists
metrics reported in the RTCP Sender and Receiver Report (SR/RR) metrics reported in the RTCP Sender and Receiver Report (SR/RR)
[RFC3550] to fulfill this requirement. However, the basic metrics [RFC3550] to fulfill this requirement. However, the basic metrics
from RTCP SR/RR are not sufficient for precise quality monitoring, or from RTCP SR/RR are not sufficient for precise quality monitoring, or
diagnosing potential issues. diagnosing potential issues.
In this document, we provide rationale for choosing additional RTP In this document, we provide rationale for choosing additional RTP
metrics for the WebRTC getStats() API [W3C.WD-webrtc-20161124]. The metrics for the WebRTC getStats() API [W3C.WD-webrtc-20161124]. All
document also creates a registry containing identifiers from the identifiers proposed in this document are recommended to be
metrics reported in the RTCP Sender, Receiver, and Extended Reports. implemented by an WebRTC endpoint. An endpoint may choose not to
All identifiers proposed in this document are RECOMMENDED to be expose an identifier if it does not implement the corresponding RTCP
implemented by an endpoint. An endpoint MAY choose not to expose an Report.
identifier if it does not implement the corresponding RTCP Report.
2. Terminology 2. Terminology
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in [RFC2119]. document are to be interpreted as described in [RFC2119].
ReportGroup: It is a set of metrics identified by a common ReportGroup: It is a set of metrics identified by a common
Synchronization source (SSRC). Synchronization source (SSRC).
3. RTP Statistics in WebRTC Implementations 3. RTP Statistics in WebRTC Implementations
The RTCP Sender Reports (SRs) and Receiver Reports (RRs) [RFC3550] The RTCP Sender Reports (SRs) and Receiver Reports (RRs) [RFC3550]
exposes the basic metrics for the local and remote media streams. exposes the basic metrics for the local and remote media streams.
However, these metrics provides only partial or limited information, However, these metrics provides only partial or limited information,
which may not be sufficient for diagnosing problems or quality which may not be sufficient for diagnosing problems or quality
monitoring. For example, it may be useful to distinguish between monitoring. For example, it may be useful to distinguish between
packets lost and packets discarded due to late arrival, even though packets lost and packets discarded due to late arrival, even though
they have the same impact on the multimedia quality, it helps in they have the same impact on the multimedia quality, it helps in
identifying and diagnosing issues. identifying and diagnosing issues. RTP Control Protocol Extended
Reports (XRs) [RFC3611] and other extensions discussed in the XRBLOCK
RTP Control Protocol Extended Reports (XRs) [RFC3611] and other working group provide more detailed statistics, which complement the
extensions discussed in the XRBLOCK working group provide more basic metrics reported in the RTCP SR and RRs.
detailed statistics, which complement the basic metrics reported in
the RTCP SR and RRs. Section 5 discusses the use of XR metrics that
may be useful for monitoring the performance of WebRTC applications.
Section 6 proposes a set of candidate metrics.
The WebRTC application extracts the statistic from the browser by The WebRTC application extracts the statistic from the browser by
querying the getStats() API [W3C.WD-webrtc-20161124], but the browser querying the getStats() API [W3C.WD-webrtc-20161124], but the browser
currently only reports the local variables i.e., the statistics currently only reports the local variables i.e., the statistics
related to the outgoing RTP media streams and the incoming RTP media related to the outgoing RTP media streams and the incoming RTP media
streams. Without the support of RTCP XRs or some other signaling streams. Without the support of RTCP XRs or some other signaling
mechanism, the WebRTC application cannot expose the remote endpoints' mechanism, the WebRTC application cannot expose the remote endpoints'
statistics. At the moment [I-D.ietf-rtcweb-rtp-usage] does not statistics. [I-D.ietf-rtcweb-rtp-usage] does not mandate the use of
mandate the use of any RTCP XRs and since their usage is optional. any RTCP XRs and since their usage is optional. If the use of RTCP
If the use of RTCP XRs is successfully negotiated between endpoints XRs is successfully negotiated between endpoints (via SDP),
(via SDP), thereafter the application has access to both local and thereafter the application has access to both local and remote
remote statistics. Alternatively, once the WebRTC application gets statistics. Alternatively, once the WebRTC application gets the
the local information, they can report it to an application server or local information, they can report it to an application server or a
a third-party monitoring system, which provides quality estimations third-party monitoring system, which provides quality estimations or
or diagnosis services for application developers. The exchange of diagnosis services for application developers. The exchange of
statistics between endpoints or between a monitoring server and an statistics between endpoints or between a monitoring server and an
endpoint is outside the scope of this document. endpoint is outside the scope of this document.
4. Considerations for Impact of Measurement Interval 4. Considerations for Impact of Measurement Interval
RTCP extensions like RTCP XR usually share the same timing 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, 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 together with the RTCP SR/RR. Alternatively, if the RTCP XR uses a
different measurement interval, all XRs using the same measurement different measurement interval, all XRs using the same measurement
interval are compounded together and the measurement interval is interval are compounded together and the measurement interval is
indicated in a specific measurement information block defined in indicated in a specific measurement information block defined in
[RFC6776]. [RFC6776].
When using WebRTC getStats() APIs (see section 7 of When using WebRTC getStats() APIs (see section 7 of [W3C.WD-webrtc-
[W3C.WD-webrtc-20161124]), the applications can query this 20161124]), the applications can query this information at arbitrary
information at arbitrary intervals. For the statistics reported by intervals. For the statistics reported by the remote endpoint, e.g.,
the remote endpoint, e.g., those conveyed in an RTCP SR/RR/XR, these those conveyed in an RTCP SR/RR/XR, these will not change until the
will not change until the next RTCP report is received. However, next RTCP report is received. However, statistics generated by the
statistics generated by the local endpoint have no such restrictions local endpoint have no such restrictions as long as the endpoint is
as long as the endpoint is sending and receiving media. For example, sending and receiving media. For example, an application may choose
an application may choose to poll the stack for statistics every 1 to poll the stack for statistics every 1 second, in this case the
second, in this case the underlying stack local will return the underlying stack local will return the current snapshot of the local
current snapshot of the local statistics (for incoming and outgoing statistics (for incoming and outgoing media streams). However it may
media streams). However it may return the same remote statistics as return the same remote statistics as before for the remote
before for the remote statistics, as no new RTCP reports may have statistics, as no new RTCP reports may have been received in the past
been received in the past 1 second. This can occur when the polling 1 second. This can occur when the polling interval is shorter than
interval is shorter than the average RTCP reporting interval. the average RTCP reporting interval.
5. Candidate Metrics 5. Candidate Metrics
Since following metrics are all defined in RTCP XR which is not Since following metrics are all defined in RTCP XR which is not
mandated in WebRTC, all of them are local. However, if RTCP XR is mandated in WebRTC, all of them are local. However, if RTCP XR is
supported by negotiation between two browsers, following metrics can supported by negotiation between two browsers, following metrics can
also be generated remotely and be sent to local by RTCP XR packets. also be generated remotely and be sent to local by RTCP XR packets.
Following metrics are classified into 3 categories: network impact Following metrics are classified into 3 categories: network impact
metrics, application impact metrics and recovery metrics. Network metrics, application impact metrics and recovery metrics. Network
impact metrics are the statistics recording the information only for impact metrics are the statistics recording the information only for
network transmission. They are useful for network problem diagnosis. network transmission. They are useful for network problem diagnosis.
Application impact metrics mainly collect the information in the Application impact metrics mainly collect the information from the
viewpoint of application, e.g., bit rate, frames rate or jitter viewpoint of application, e.g., bit rate, frames rate or jitter
buffers. Recovery metrics reflect how well the repair mechanisms buffers. Recovery metrics reflect how well the repair mechanisms
perform, e.g. loss concealment, retransmission or FEC. All of the 3 perform, e.g. loss concealment, retransmission or Forward Error
types of metrics are useful for quality estimations of services in Correction (FEC). All of the 3 types of metrics are useful for
WebRTC implementations. WebRTC application can use these metrics to quality estimations of services in WebRTC implementations. WebRTC
calculate the Mean Opinion Score (MoS) values or Media Delivery Index application can use these metrics to calculate the Mean Opinion Score
(MDI) for their services. (MoS) values or Media Delivery Index (MDI) for their services.
5.1. Network Impact Metrics 5.1. Network Impact Metrics
5.1.1. Loss and Discard Packet Count Metric 5.1.1. Loss and Discard Packet Count Metric
In multimedia transport, packets which are received abnormally are In multimedia transport, packets which are received abnormally are
classified into 3 types: lost, discarded and duplicate packets. classified into 3 types: lost, discarded and duplicate packets.
Packet loss may be caused by network device breakdown, bit-error Packet loss may be caused by network device breakdown, bit-error
corruption or network congestion (packets dropped by an intermediate corruption or network congestion (packets dropped by an intermediate
router queue). Duplicate packets may be a result of network delays, router queue). Duplicate packets may be a result of network delays
which causes the sender to retransmit the original packets. that causes the sender to retransmit the original packets. Discarded
Discarded packets are packets that have been delayed long enough packets are packets that have been delayed long enough (perhaps they
(perhaps they missed the playout time) and are considered useless by missed the playout time) and are considered useless by the receiver.
the receiver. Lost and discarded packets cause problems for Lost and discarded packets cause problems for multimedia services, as
multimedia services, as missing data and long delays can cause missing data and long delays can cause degradation in service
degradation in service quality, e.g., missing large blocks of quality, e.g., missing large blocks of contiguous packets (lost or
contiguous packets (lost or discarded) may cause choppy audio, and discarded) may cause choppy audio, and long network transmission
long network transmission delay time may cause audio or video delay time may cause audio or video buffering. The RTCP SR/RR
buffering. The RTCP SR/RR defines a metric for counting the total defines a metric for counting the total number of RTP data packets
number of RTP data packets that have been lost since the beginning of that have been lost since the beginning of reception. But this
reception. But this statistic does not distinguish lost packets from statistic does not distinguish lost packets from discarded and
discarded and duplicate packets. Packets that arrive late will be duplicate packets. Packets that arrive late will be discarded and
discarded and are not reported as lost, and duplicate packets will be are not reported as lost, and duplicate packets will be regarded as a
regarded as a normally received packet. Hence, the loss metric can normally received packet. Hence, the loss metric can be misleading
be misleading if many duplicate packets are received or packets are if many duplicate packets are received or packets are discarded,
discarded, which causes the quality of the media transport to appear which causes the quality of the media transport to appear okay from
okay from the statistic point of view, but meanwhile the users may the statistic point of view, but meanwhile the users may actually be
actually be experiencing bad service quality. So in such cases, it experiencing bad service quality. So in such cases, it is better to
is better to use more accurate metrics in addition to those defined use more accurate metrics in addition to those defined in RTCP SR/RR.
in RTCP SR/RR.
The lost packets and duplicated packets metrics defined in Statistics The lost packets and duplicated packets metrics defined in Statistics
Summary Report Block of [RFC3611] extend the information of loss Summary Report Block of [RFC3611] extend the information of loss
carried in standard RTCP SR/RR. They explicitly give an account of carried in standard RTCP SR/RR. They explicitly give an account of
lost and duplicated packets. Lost packets counts are useful for lost and duplicated packets. Lost packets counts are useful for
network problem diagnosis. It is better to use the loss packets network problem diagnosis. It is better to use the loss packets
metrics of [RFC3611] to indicate the packet lost count instead of the metrics of [RFC3611] to indicate the packet lost count instead of the
cumulative number of packets lost metric of [RFC3550]. Duplicated cumulative number of packets lost metric of [RFC3550]. Duplicated
packets are usually rare and have little effect on QoS evaluation. packets are usually rare and have little effect on QoS evaluation. So
So it may not be suitable for use in WebRTC. it may not be suitable for use in WebRTC.
Using loss metrics without considering discard metrics may result in Using loss metrics without considering discard metrics may result in
inaccurate quality evaluation, as packet discard due to jitter is inaccurate quality evaluation, as packet discard due to jitter is
often more prevalent than packet loss in modern IP networks. The often more prevalent than packet loss in modern IP networks. The
discarded metric specified in [RFC7002] counts the number of packets discarded metric specified in [RFC7002] counts the number of packets
discarded due to the jitter. It augments the loss statistics metrics discarded due to the jitter. It augments the loss statistics metrics
specified in standard RTCP SR/RR. For those RTCWEB services with specified in standard RTCP SR/RR. For those RTCWEB services with
jitter buffer requiring precise quality evaluation and accurate jitter buffers requiring precise quality evaluation and accurate
troubleshooting, this metric is useful as a complement to the metrics troubleshooting, this metric is useful as a complement to the metrics
of RTCP SR/RR. of RTCP SR/RR.
5.1.2. Burst/Gap Pattern Metrics for Loss and Discard 5.1.2. Burst/Gap Pattern Metrics for Loss and Discard
RTCP SR/RR defines coarse metrics regarding loss statistics, the RTCP SR/RR defines coarse metrics regarding loss statistics: the
metrics are all about per call statistics and are not detailed enough metrics are all about per call statistics and are not detailed enough
to capture some transitory nature of the impairments like bursty to capture the transitory nature of some impairments like bursty
packet loss. Even if the average packet loss rate is low, the lost packet loss. Even if the average packet loss rate is low, the lost
packets may occur during short dense periods, resulting in short packets may occur during short dense periods, resulting in short
periods of degraded quality. Distributed burst provides a higher periods of degraded quality. Bursts cause lower quality experience
subjective quality than a non-burst distribution for low packet loss than the non-bursts for low packet loss rates, whereas for high
rates whereas for high packet loss rates the converse is true. So packet loss rates the converse is true. So capturing burst gap
capturing burst gap information is very helpful for quality information is very helpful for quality evaluation and locating
evaluation and locating impairments. If the WebRTC application needs impairments. If the WebRTC application needs to evaluate the
to evaluate the services quality, burst gap metrics provides more services quality, burst gap metrics provides more accurate
accurate information than RTCP SR/RR. information than RTCP SR/RR.
[RFC3611] introduces burst gap metrics in VoIP report block. These [RFC3611] introduces burst gap metrics in VoIP report block. These
metrics record the density and duration of burst and gap periods, metrics record the density and duration of burst and gap periods,
which are helpful in isolating network problems since bursts which are helpful in isolating network problems since bursts
correspond to periods of time during which the packet loss/discard correspond to periods of time during which the packet loss/discard
rate is high enough to produce noticeable degradation in audio or rate is high enough to produce noticeable degradation in audio or
video quality. Burst gap related metrics are also introduced in video quality. Burst gap related metrics are also introduced in
[RFC7003] and [RFC6958] which define two new report blocks for usage [RFC7003] and [RFC6958] which define two new report blocks for usage
in a range of RTP applications beyond those described in [RFC3611]. in a range of RTP applications beyond those described in [RFC3611].
These metrics distinguish discarded packets from loss packets that These metrics distinguish discarded packets from loss packets that
occur in the bursts period and provides more information for occur in the bursts period and provides more information for
diagnosing network problems. Additionally, the block reports the diagnosing network problems. Additionally, the block reports the
frequency of burst events which is useful information for evaluating frequency of burst events which is useful information for evaluating
the quality of experience. Hence, if WebRTC application need to do the quality of experience. Hence, if WebRTC applications need to do
quality evaluation and observe when and why quality degrades, these quality evaluation and observe when and why quality degrades, these
metrics should be considered. metrics should be considered.
5.1.3. Run Length Encoded Metrics for Loss, Discard 5.1.3. Run Length Encoded Metrics for Loss, Discard
Run-length encoding uses a bit vector to encode information about the Run-length encoding uses a bit vector to encode information about the
packet. Each bit in the vector represents a packet and depending on packet. Each bit in the vector represents a packet and depending on
the signaled metric it defines if the packet was lost, duplicated, the signaled metric it defines if the packet was lost, duplicated,
discarded, or repaired. An endpoint typically uses the run length discarded, or repaired. An endpoint typically uses the run length
encoding to accurately communicate the status of each packet in the encoding to accurately communicate the status of each packet in the
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If losses occur after discards, an endpoint may be able to correlate If losses occur after discards, an endpoint may be able to correlate
the two run length vectors to identify congestion-related losses, the two run length vectors to identify congestion-related losses,
i.e., a router queue became overloaded causing delays and then i.e., a router queue became overloaded causing delays and then
overflowed. If the losses are independent, it may indicate bit-error overflowed. If the losses are independent, it may indicate bit-error
corruption. For the WebRTC Stats API [W3C.WD-webrtc-stats-20161214], corruption. For the WebRTC Stats API [W3C.WD-webrtc-stats-20161214],
these types of metrics are not recommended for use due to the large these types of metrics are not recommended for use due to the large
amount of data and the computation involved. amount of data and the computation involved.
5.2. Application Impact Metrics 5.2. Application Impact Metrics
5.2.1. Discard Octets Metric 5.2.1. Discarded Octets Metric
The metric reports the cumulative size of the packets discarded in The metric reports the cumulative size of the packets discarded in
the interval, it is complementary to number of discarded packets. An the interval, it is complementary to number of discarded packets. An
application measures sent octets and received octets to calculate application measures sent octets and received octets to calculate
sending rate and receiving rate, respectively. The application can sending rate and receiving rate, respectively. The application can
calculate the actual bit rate in a particular interval by subtracting calculate the actual bit rate in a particular interval by subtracting
the discarded octets from the received octets. the discarded octets from the received octets.
For WebRTC, discarded octets supplements the sent and received octets For WebRTC, discarded octets supplements the sent and received octets
and provides an accurate method for calculating the actual bit rate and provides an accurate method for calculating the actual bit rate
which is an important parameter to reflect the quality of the media. which is an important parameter to reflect the quality of the media.
The discarded bytes metric is defined in [RFC7243]. The discarded bytes metric is defined in [RFC7243].
5.2.2. Frame Impairment Summary Metrics 5.2.2. Frame Impairment Summary Metrics
RTP has different framing mechanisms for different payload types. RTP has different framing mechanisms for different payload types. For
For audio streams, a single RTP packet may contain one or multiple audio streams, a single RTP packet may contain one or multiple audio
audio frames, each of which has a fixed length. On the other hand, frames, each of which has a fixed length. On the other hand, in
in video streams, a single video frame may be transmitted in multiple video streams, a single video frame may be transmitted in multiple
RTP packets. The size of each packet is limited by the Maximum RTP packets. The size of each packet is limited by the Maximum
Transmission Unit (MTU) of the underlying network. However, Transmission Unit (MTU) of the underlying network. However,
statistics from standard SR/RR only collect information from statistics from standard SR/RR only collect information from
transport layer, which may not fully reflect the quality observed by transport layer, which may not fully reflect the quality observed by
the application. Video is typically encoded using two frame types the application. Video is typically encoded using two frame types
i.e., key frames and derived frames. Key frames are normally just i.e., key frames and derived frames. Key frames are normally just
spatially compressed, i.e., without prediction from other pictures. spatially compressed, i.e., without prediction from other pictures.
The derived frames are temporally compressed, i.e., depend on the key The derived frames are temporally compressed, i.e., depend on the key
frame for decoding. Hence, key frames are much larger in size than frame for decoding. Hence, key frames are much larger in size than
derived frames. The loss of these key frames results in a derived frames. The loss of these key frames results in a
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the original streams. But they do help to greatly reduce the impact the original streams. But they do help to greatly reduce the impact
of packet loss and enhance the quality of transmission. Web of packet loss and enhance the quality of transmission. Web
applications could support certain repair mechanism after negotiation applications could support certain repair mechanism after negotiation
between both sides of browsers when needed. For these web between both sides of browsers when needed. For these web
applications using repair mechanisms, providing some statistic applications using repair mechanisms, providing some statistic
information for the performance of their repair mechanisms could help information for the performance of their repair mechanisms could help
to have a more accurate quality evaluation. to have a more accurate quality evaluation.
The un-repaired packets count and repaired loss count defined in The un-repaired packets count and repaired loss count defined in
[RFC7509] provide the recovery information of the error-resilience [RFC7509] provide the recovery information of the error-resilience
mechanisms to the monitoring application or the sending endpoint. mechanisms to the monitoring application or the sending endpoint. The
The endpoint can use these metrics to ascertain the ratio of repaired endpoint can use these metrics to ascertain the ratio of repaired
packets to lost packets. Including this kind of metrics helps the packets to lost packets. Including this kind of metrics helps the
application evaluate the effectiveness of the applied repair application evaluate the effectiveness of the applied repair
mechanisms. mechanisms.
5.3.2. Run Length Encoded Metric for Post-repair 5.3.2. Run Length Encoded Metric for Post-repair
[RFC5725] defines run-length encoding for post-repair packets. When [RFC5725] defines run-length encoding for post-repair packets. When
using error-resilience mechanisms, the endpoint can correlate the using error-resilience mechanisms, the endpoint can correlate the
loss run length with this metric to ascertain where the losses and loss run length with this metric to ascertain where the losses and
repairs occurred in the interval. This provides more accurate repairs occurred in the interval. This provides more accurate
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mechanisms, the endpoint can correlate the loss and post-repair run mechanisms, the endpoint can correlate the loss and post-repair run
lengths to ascertain where the losses and repairs occurred in the lengths to ascertain where the losses and repairs occurred in the
interval. For example, consecutive losses are likely not to be interval. For example, consecutive losses are likely not to be
repaired by a simple FEC scheme. repaired by a simple FEC scheme.
6. Identifiers from Sender, Receiver, and Extended Report Blocks 6. Identifiers from Sender, Receiver, and Extended Report Blocks
This document describes a list of metrics and corresponding This document describes a list of metrics and corresponding
identifiers relevant to RTP media in WebRTC. These group of identifiers relevant to RTP media in WebRTC. These group of
identifiers are defined on a ReportGroup corresponding to an identifiers are defined on a ReportGroup corresponding to an
Synchronization source (SSRC). In practice the application MUST be Synchronization source (SSRC). In practice the application need to
able to query the statistic identifiers on both an incoming (remote) be able to query the statistic identifiers on both an incoming
and outgoing (local) media stream. Since sending and receiving SR (remote) and outgoing (local) media stream. Since sending and
and RR are mandatory, the metrics defined in the SR and RR report receiving SR and RR are mandatory, the metrics defined in the SR and
blocks are always available. For XR metrics, it depends on two RR report blocks are always available. For XR metrics, it depends on
factors: 1) if it measured at the endpoint, 2) if it reported by the two factors: 1) if it measured at the endpoint, 2) if it reported by
endpoint in an XR report. If a metric is only measured by the 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 endpoint and not reported, the metrics will only be available for the
incoming (remote) media stream. Alternatively, if the corresponding incoming (remote) media stream. Alternatively, if the corresponding
metric is also reported in an XR report, it will be available for metric is also reported in an XR report, it will be available for
both the incoming (remote) and outgoing (local) media stream. both the incoming (remote) and outgoing (local) media stream.
For a remote statistic, the timestamp represents the timestamp from For a remote statistic, the timestamp represents the timestamp from
an incoming SR/RR/XR packet. Conversely, for a local statistic, it an incoming SR/RR/XR packet. Conversely, for a local statistic, it
refers to the current timestamp generated by the local clock refers to the current timestamp generated by the local clock
(typically the POSIX timestamp, i.e., milliseconds since Jan 1, (typically the POSIX timestamp, i.e., milliseconds since Jan 1,
1970). 1970).
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Name: framesSent Name: framesSent
Definition: The cumulative number of frames sent. Definition: The cumulative number of frames sent.
Name: framesReceived Name: framesReceived
Definition: The cumulative number of partial or full frames received. Definition: The cumulative number of partial or full frames received.
7. Adding new metrics to WebRTC Statistics API 7. Adding new metrics to WebRTC Statistics API
The metrics defined in this draft have already been added to the W3C During the progress of this work, the metrics defined in this draft
WebRTC specification. The current working process to add new metrics have already been added to the W3C WebRTC specification. The working
is, create an issue or pull request on the repository of the W3C process to add new metrics for future is to create an issue or pull
WebRTC specification (https://github.com/w3c/webrtc-stats). request on the repository of the W3C WebRTC specification
(https://github.com/w3c/webrtc-stats).
8. Security Considerations 8. Security Considerations
The monitoring activities are implemented between two browsers or This document focuses on listing the RTCP XR metrics defined in the
between a browser and a server. Therefore encryption procedures, corresponding RTCP reporting extensions and do not give rise to any
such as the ones suggested for a Secure RTCP (SRTCP), need to be new security vulnerabilities beyond those described in [RFC3611] and
used. Currently, the monitoring in RTCWEB introduces no new security [RFC3792].
considerations beyond those described in [I-D.ietf-rtcweb-rtp-usage],
[I-D.ietf-rtcweb-security]. The overall security considerations for RTP used in WebRTC
applicaitons is described in [I-D.ietf-rtcweb-rtp-usage] and [I-
D.ietf-rtcweb-security], which are also apply to this memo.
9. Acknowledgements 9. Acknowledgements
The authors would like to thank Bernard Aboba, Harald Alvestrand, Al The authors would like to thank Bernard Aboba, Harald Alvestrand, Al
Morton, Colin Perkins, and Shida Schubert for their valuable comments Morton, Colin Perkins, and Shida Schubert for their valuable comments
and suggestions on earlier version of this document. and suggestions on earlier version of this document.
10. References 10. References
10.1. Normative References 10.1. Normative References
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A.5. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00 A.5. changes in draft-ietf-xrblock-rtcweb-rtcp-xr-metrics-00
o Submitted as WG Draft. o Submitted as WG Draft.
A.6. changes in draft-huang-xrblock-rtcweb-rtcp-xr-metrics-04 A.6. changes in draft-huang-xrblock-rtcweb-rtcp-xr-metrics-04
o Addressed comments from the London IETF meeting: o Addressed comments from the London IETF meeting:
o Removed ECN metrics. o Removed ECN metrics.
o Merged draft-singh-xrblock-webrtc-additional-stats-01
Authors' Addresses Authors' Addresses
Varun Singh Varun Singh
CALLSTATS I/O Oy CALLSTATS I/O Oy
Annankatu 31-33 C 42 Annankatu 31-33 C 42
Helsinki 00100 Helsinki 00100
Finland Finland
Email: varun@callstats.io Email: varun@callstats.io
URI: https://www.callstats.io/about URI: https://www.callstats.io/about
skipping to change at page 17, line 27 skipping to change at page 17, line 27
China China
Email: rachel.huang@huawei.com Email: rachel.huang@huawei.com
Roni Even Roni Even
Huawei Huawei
14 David Hamelech 14 David Hamelech
Tel Aviv 64953 Tel Aviv 64953
Israel Israel
Email: roni.even@mail01.huawei.com Email: roni.even@huawei.com
Dan Romascanu Dan Romascanu
Email: dromasca@gmail.com Email: dromasca@gmail.com
Lingli Deng Lingli Deng
China Mobile China Mobile
Email: denglingli@chinamobile.com Email: denglingli@chinamobile.com
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