One document matched: draft-wu-xrblock-rtcp-xr-quality-monitoring-01.txt
Differences from draft-wu-xrblock-rtcp-xr-quality-monitoring-00.txt
Network Working Group Q. Wu
Internet-Draft Huawei
Intended status: Standards Track G. Zorn
Expires: September 3, 2011 Network Zen
R. Schott
Deutsche Telekom Laboratories
March 2, 2011
RTP Control Protocol Extended Reports (RTCP XR) Report Blocks for Real-
time Application Quality Monitoring
draft-wu-xrblock-rtcp-xr-quality-monitoring-01
Abstract
This document defines a set of RTP Control Protocol Extended Reports
(RTCP XR) Report Blocks and associated SDP parameters allowing the
report of real time application quality metrics, primarily for video
applications of RTP.
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
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Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
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material or to cite them other than as "work in progress."
This Internet-Draft will expire on September 3, 2011.
Copyright Notice
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document authors. All rights reserved.
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described in the Simplified BSD License.
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Standards Language . . . . . . . . . . . . . . . . . . . . 4
2.2. Acronyms . . . . . . . . . . . . . . . . . . . . . . . . . 5
3. Applicability . . . . . . . . . . . . . . . . . . . . . . . . 5
4. Transport Layer Metrics . . . . . . . . . . . . . . . . . . . 6
4.1. RTP Flows Initial Synchronization Delay Report Block . . . 6
4.2. RTP Flows General Synchronization Offset Metrics Block . . 7
4.3. Layered Streams Statistics Metrics Block . . . . . . . . . 8
5. Application Layer Metrics . . . . . . . . . . . . . . . . . . 10
5.1. Transport Streams Statistics Summary Report Block . . . . 10
5.2. Transport Stream Loss and Discard Metrics Block . . . . . 12
5.3. Transport Stream Burst Metrics Block . . . . . . . . . . . 14
5.4. Synthetical Multimedia Quality Metrics Block . . . . . . . 16
6. SDP Signaling . . . . . . . . . . . . . . . . . . . . . . . . 18
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 20
8. Security Considerations . . . . . . . . . . . . . . . . . . . 21
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 21
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 21
10.1. Normative References . . . . . . . . . . . . . . . . . . . 21
10.2. Informative References . . . . . . . . . . . . . . . . . . 22
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 23
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1. Introduction
Along with the wide deployment of broadband access and the
development of new IPTV services (e.g., broadcast video, video on
demand), there is increasing interest in monitoring and managing
networks and applications that deliver real-time applications over
RTP or IP, to ensure that all end users obtain acceptable video/audio
quality. The main drives come from operators and enterprises, since
offering performance monitoring capability can help diagnose network
impairments, facilitate in root cause analysis and aid in verifying
compliance with service level agreements (SLAs) between Internet
Service Providers (ISPs) and content providers.
The factors that affect real-time application quality can be split
into two categories. The first category consists of transport-
dependent factors such as packet loss, delay and jitter (which also
translates into losses in the playback buffer). The factors in the
second category are application-specific factors that affect real
time application (e.g., video) quality and are sensitivity to network
errors. These factors can be but not limited to video codec and loss
recovery technique, coding bit rate, packetization scheme, and
content characteristics.
Compared with application-specific factors, the transport-dependent
factors sometimes are not sufficient to measure real time data
quality, since the ability to analyze the real time data in the
application layer provides quantifiable measurements for subscriber
Quality of Experience (QoE) that may not be captured in the
transmission layers or from the RTP layer down. In a typical
scenario, monitoring of the transmission layers can produce
statistics suggesting that quality is not an issue, such as the fact
that network jitter is not excessive. However, problems may occur in
the service layers leading to poor subscriber QoE. Therefore
monitoring using only network-level measurements may be insufficient
when application layer video quality is required.
In order to provide accurate measures of real time application
quality for operators when transporting real time contents across a
network, the synthentical multimedia quality Metrics is highly
required which can be conveyed in the RTCP XR packets[RFC3611] and
may have the following three benefits:
* Tuning the content encoder algorithm to satisfy real time data
quality requirements
* Determining which system techniques to use in a given situation
and when to switch from one technique to another as system
parameters change
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* Verifying the continued correct operation of an existing system
RFC 3611 [RFC3611] defines seven report block formats for network
management and quality monitoring. However, some of these metrics
are mostly for multicast inference of network characteristics (MINC)
or voice over IP (VoIP) monitoring and not widely applicable to other
applications, e.g., video quality monitoring. This document focuses
on specifying new additional report block types used to convey QoE
related parameters that is genericly designed for use in voice, audio
and video services.
The report block types defined in this document fall into two
categories. The first category consists of general information
regarding transmission quality, to be generated and processed by the
RTP transport. The report blocks in the second category convey
metrics above transport that affect real time application quality and
are sensitivity to network errors.
Seven report block formats are defined by this document. Of these,
three are transport layer metrics:
* RTP Flows Initial Synchronization Delay Report Block
* RTP Flows General Synchronization Offset Metrics Block
* Layered Streams Statistics Metrics Block
The other four are application layer metrics:
* Transport Stream Statistics Summary Report Block
* Transport Stream Loss and Discard Metrics Block
* Transport Stream Burst Metrics Block
* Synthetical Multimedia Quality Metrics Block
2. Terminology
2.1. Standards Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
In addition, the following terms are defined:
Layered Component Packet
a RTP packet using layered codecs containing the specified layered
component, e.g., encoded stream at the base layer or at the
enhancement layer.
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Picture Type
Picture types used in the different video algorithms compose of
the key-frame and the Derivation frame. Key-frame is also called
a reference frame and used as a reference for predicting other
pictures. It is coded without prediction from other pictures.
The Derivation frame is derived from Key-frame using prediction
from the reference frame.
2.2. Acronyms
SSRC
Synchronization Source [RFC3550]
TS
Transport Stream [ISO-IEC.13818-1.2007]
3. Applicability
All the report blocks defined in this document could be used by
dedicated network monitoring applications. As specified in RFC 3611
[RFC3611], for such an application it might be appropriate to allow
more than 5% of RTP data bandwidth to be used for RTCP packets, thus
allowing proportionately larger and more detailed report blocks.
RTP Flows General Synchronization Offset Metrics Block in Section 4.2
has been defined for various multimedia applications. Such
applications can use this report block to monitor offset between two
RTP streams synchronization to ensure satisfactory QoE. Tighter
tolerances than typically used have been recommended for such
applications.
The RTP Flows Initial Synchronization Delay Report Block has been
defined primarily for layered or multi-description video coding
applications. When joining a layered video session in such an
application, a receiver may not synchronize playout across the
multimedia session until RTCP SR packets have been received on all of
the component RTP sessions. This report block can be used to measure
synchronization between different media layers for the same
multimedia session.
The Transport Stream Loss and Discard Metrics Report Block, Transport
Stream Burst Metrics report Block, Transport Statistics Summary
Report Block and Layered Streams Statistics Metrics Block can be
applied to any real time video application, while Synthetical
Multimedia Quality Metrics Report Block can be used in any real-time
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AV application.
4. Transport Layer Metrics
4.1. RTP Flows Initial Synchronization Delay Report Block
This block reports Initial synchronization delay beyond the
information carried in the standard RTCP packet format. Information
is recorded about the the difference between the start of RTP
sessions and the time the RTP receiver acquires all components of RTP
sessions [RFC6051]. The components of RTP session are referred to as
one RTP session for each media type or the media content in each
layer contained in RTP Control Protocol (RTCP) sender report (SR)
packets [RFC3550]. For unicast session, the delay due to negotiation
of NAT pinholes, firewall holes, quality-of-service, and media
security keys is contributed to such initial synchronization delay.
For multicast session, the initial synchronization delay varies with
the session bandwidth and the number of members, the number of
senders in the session. In the absence of packet loss, the initial
synchronisation delay equals to the average time taken to receive the
first RTCP packet in the RTP session with the longest RTCP reporting
interval.In the presence of packet loss, the media synchronization
needs to wait until the reporting interval has passed, and the next
RTCP SR packet is sent.
The RTP Flows Initial Synchronization Delay Report Block has the
following format:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=TBD | Reserved | Block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of Sender |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Initial Synchronization Delay |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Block type (BT): 8 bits
The Statistics Summary Report Block is identified by the constant
<RFISD>.
Reserved: 8 bits
This field is reserved for future definition. In the absence of
such a definition, the bits in this field MUST be set to zero and
MUST be ignored by the receiver.
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Block length: 16 bits
The constant 3, in accordance with the definition of this field in
Section 3 of RFC 3611 [RFC3611].
SSRC of Sender: 32 bits
The SSRC of the RTP data packet source being reported upon by this
report block. (Section 4.1 of [RFC3611]).
Initial Synchronization Delay: 32 bits
The average delay, expressed in units of 1/65536 seconds, between
the RTCP packets received on all of the components RTP sessions
and the beginning of session [RFC6051]. The value is calculated
as follows:
The average time, expressed in units of 1/65536 seconds, taken
to receive the first RTCP packet in the RTP session with the
longest RTCP reporting interval [RFC6051]
4.2. RTP Flows General Synchronization Offset Metrics Block
In an RTP multimedia session, there can be an arbitrary number of
streams, with the same RTCP CNAME. This block reports the general
Synchronization offset status of these RTP streams beyond the
information carried in the standard RTCP packet format. Information
is recorded about the synchronization offset time of each RTP stream
relative to the reference RTP stream with the same CNAME and General
Synchronisation Offset of zero. For layered session or multimedia
session,the first RTP packet can be chosen as the basic packet of
reference RTP stream. The RTP Flow General Synchronization Offset
Report Block has the following format:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=TBD |I| Reserved | Block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| General Synchronization Offset |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Block type (BT): 8 bits
The Statistics Summary Report Block is identified by the constant
<RFGSO>.
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Interval Metric flag (I): 1 bit
This field is used to indicate whether the Audio-Video
synchronization metrics are Interval or Cumulative metrics, that
is, whether the reported values applies to the most recent
measurement interval duration between successive metrics reports
(I=1) (the Interval Duration) or to the accumulation period
characteristic of cumulative measurements (I=0) (the Cumulative
Duration).
Reserved: 8 bits
This field is reserved for future definition. In the absence of
such a definition, the bits in this field MUST be set to zero and
MUST be ignored by the receiver.
Block length: 16 bits
The constant 2, in accordance with the definition of this field in
Section 3 of RFC 3611 [RFC3611].
SSRC of source: 32 bits
As defined in Section 4.1 of RFC 3611 [RFC3611].
General synchronization offset: 32 bits
This field represents the synchronization offset time of one RTP
stream in milliseconds relative to the reference RTP stream with
the same CNAME and General Synchronisation Offset of zero
[RFC6051] This value is calculated based on the interarrival time
between arbitray RTP packet and the reference RTP packet with the
same CNAME , and timestamps of this arbitray RTP packet and the
reference RTP packet with the same CNAME.
4.3. Layered Streams Statistics Metrics Block
This block reports layered streams statistics beyond the information
carried in the Statistics Summary Report Block RTCP packet specified
in the section 4.6 of RFC 3611 [RFC3611]. Information is recorded
about lost layered component packets, duplicated layered component
packets. Such information can be useful for network management and
video quality monitoring.
The report block contents are dependent upon a series of flag bits
carried in the first part of the header. Not all parameters need to
be reported in each block. Flags indicate which parameters are
reported and which are not. The fields corresponding to unreported
parameters MUST be present, but are set to zero. The receiver MUST
ignore any Layered Streams Statistics Metrics Block with a non-zero
value in any field flagged as unreported.
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The Layered Stream Statistics metrics Block has the following format:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=TBD |T| rsd. | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Lost_Layered Component Packets |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Dup Layered Component_Packets |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Block type (BT): 8 bits
The Layered stream Statistics Metrics Block is identified by the
constant <LSSM>.
Layer Type flag (T): 1 bits
This field is used to indicate the Layer Type of layered video to
be reported. LT is set to 0 if the loss_component_packet field
and dup_component packet contain the base layer packet in layered
codecs,e.g, SVC in [I-D.ietf-avt-rtp-svc], 1 if the loss_component
packet field and dup_component packet contain enhancement layer
packet in layered codec.
Rsd.: 3 bits
This field is reserved for future definition. In the absence of
such a definition, the bits in this field MUST be set to zero and
MUST be ignored by the receiver.
Block length: 16 bits
The constant 3, in accordance with the definition of this field in
Section 3 of RFC 3611 [RFC3611].
SSRC of source: 32 bits
As defined in Section 4.1 of RFC 3611 [RFC3611].
begin_seq: 16 bits
As defined in Section 4.1 of RFC 3611 [RFC3611].
end_seq: 16 bits
As defined in Section 4.1 of RFC 3611 [RFC3611].
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Lost_Layered Component Packets: 32 bits
Number of lost_component packets in the above sequence number
interval.
Dup_Layered Component Packets: 32 bits
Number of dup_component packets in the above sequence number
interval.
5. Application Layer Metrics
5.1. Transport Streams Statistics Summary Report Block
This block reports statistics beyond the information carried in the
Statistics Summary Report Block RTCP packet specified in the section
4.6 of RFC 3611 [RFC3611]. Information is recorded about lost frame
packets, duplicated frame packets, lost layered component packets,
duplicated layered component packets. Such information can be useful
for network management and video quality monitoring.
The report block contents are dependent upon a series of flag bits
carried in the first part of the header. Not all parameters need to
be reported in each block. Flags indicate which parameters are
reported and which are not. The fields corresponding to unreported
parameters MUST be present, but are set to zero. The receiver MUST
ignore any Video Statistics Summary Report Block with a non-zero
value in any field flagged as unreported.
The Transport Streams Statistics Summary Report Block has the
following format:
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=TBD |T|P| rsd. | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| lost_frames |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| dup frames |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| partial_lost_frames |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| partial_dup_frames |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| key frames impairement proportion |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Block type (BT): 8 bits
The Transport Statistics Summary Report Block is identified by the
constant <TSSS>.
Picture type indicator (T): 1 bits
Picture types used in the different video algorithms compose of
key-frame and derivation frame. This field is used to indicate
the frame type to be reported. Bits set to 0 if the lost_frames
field or dup_frames field contain a key_frame report or reference
frame report, 1 if the lost_frames field and dup_frames field
contain other derivation frame report.
P: 1 bit
Bit set to 1 if the partial_lost_frames field or the partial_dup_
frames field contains a report, 0 otherwise.
Rsd.: 3 bits
This field is reserved for future definition. In the absence of
such a definition, the bits in this field MUST be set to zero and
MUST be ignored by the receiver.
Block length: 16 bits
The constant 5, in accordance with the definition of this field in
Section 3 of RFC 3611 [RFC3611].
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SSRC of source: 32 bits
As defined in Section 4.1 of RFC 3611 [RFC3611].
begin_seq: 16 bits
As defined in Section 4.1 of RFC 3611 [RFC3611].
end_seq: 16 bits
As defined in Section 4.1 of RFC 3611 [RFC3611].
lost_frames: 32 bits
Number of lost_frames in the above sequence number interval.
dup_frames: 32 bits
Number of dup_frames in the above sequence number interval.
partial lost_frames: 32 bits
Number of partial lost_frames in the above sequence number
interval.
partial dup_frames: 32 bits
Number of partial_dup_frames in the above sequence number
interval.
key frames impairment proportion:32bits
The proportion of key frame impaired by packet loss,discard and
duplication.
5.2. Transport Stream Loss and Discard Metrics Block
This block reports Loss and Discard metrics statistics beyond the
information carried in the standard RTCP packet format. The block
reports separately on packets lost on the IP channel, and those that
have been received but then discarded by the receiving jitter buffer.
It is very useful to distinguish between packets lost by the network
and those discarded due to jitter. Both have equal effect on the
quality of the video stream, however, having separate counts helps
identify the source of quality degradation. These fields MUST be
populated, and MUST be set to zero if no packets have been received.
Implementations MUST provide values for all the fields defined here.
For certain metrics, if the value is undefined or unknown, then the
specified default or unknown field value MUST be provided.
The block is encoded as six 32-bit words:
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=TBD |T | reserved | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Loss rate | Discard rate |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
A Transport Stream Metrics Report Block is identified by the
constant <TSLDM>.
Picture type indicator (T): 1 bits
Picture types used in the different video algorithms compose of
key-frame and derivation frame. This field is used to indicate
the picture type to be reported. Bits set to 0 if the Loss rate
field and discard rate field contain a Key_frame report or
reference frame report, 1 if the Loss rate field and discard rate
field contain other derivation frame reports.
reserved: 6 bits
This field is reserved for future definition. In the absence of
such a definition, the bits in this field MUST be set to zero and
MUST be ignored by the receiver.
block length: 16 bits
The constant 1, in accordance with the definition of this field in
Section 3 of RFC 3611 [RFC3611].
SSRC of source: 32 bits
The SSRC of the RTP data packet source being reported upon by this
report block. in accordance with the definition of this field in
Section 3 of RFC 3611 [RFC3611].
Loss rate: 8 bits
The fraction of RTP data packets from the source lost since the
beginning of reception, expressed as a fixed point number with the
binary point at the left edge of the field. This value is
calculated by dividing the total number of lost packets containing
specified frame (e.g., Key frame) (after the effects of applying
any error protection such as FEC) by the total number of packets
expected, multiplying the result of the division by 256, limiting
the maximum value to 255 (to avoid overflow), and taking the
integer part. The numbers of duplicated packets and discarded
packets do not enter into this calculation. Since receivers
cannot be required to maintain unlimited buffers, a receiver MAY
categorize late-arriving packets as lost. The degree of lateness
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that triggers a loss SHOULD be significantly greater than that
which triggers a discard.
Discard rate: 8 bits
The fraction of RTP data packets from the source that have been
discarded since the beginning of reception, due to late or early
arrival, under-run or overflow at the receiving jitter buffer.
This value is expressed as a fixed point number with the binary
point at the left edge of the field. It is calculated by dividing
the total number of discarded packets containing specified frame
(e.g., Key Frame) (excluding duplicate packet discards) by the
total number of packets expected, multiplying the result of the
division by 256, limiting the maximum value to 255 (to avoid
overflow), and taking the integer part.
5.3. Transport Stream Burst Metrics Block
This block reports Burst metrics statistics beyond the information
carried in the standard RTCP packet format. It reports on the
combined effect of losses and discards, as both have equal effect on
video quality.
In order to properly assess the quality of a video stream, it is
desirable to consider the degree of burstiness of packet loss RFC
3357 [RFC3357]. Following the one-way loss pattern sample metrics
discussed in [RFC3357], a measure of the spacing between consecutive
network packet loss or error events, is a "loss distance". The loss
distance metric captures the spacing between the loss periods. The
duration of a loss or error event (e.g. and how many packets are lost
in that duration) is a "loss period", the loss period metric captures
the frequency and length (burstiness) of loss once it starts. Delay
reports include the transit delay between RTP end points and the end
system processing delays, both of which contribute to the user
perceived delay.
Implementations MUST provide values for all the fields defined here.
For certain metrics, if the value is undefined or unknown, then the
specified default or unknown field value MUST be provided.
The block is encoded as six 32-bit words:
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0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=TBD | Reserved | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Loss Distance | Loss Period |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Max Loss Duration | Reserved. |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
A Transport Stream Metrics Report Block is identified by the
constant <TSBM>.
reserved: 8 bits
This field is reserved for future definition. In the absence of
such a definition, the bits in this field MUST be set to zero and
MUST be ignored by the receiver.
block length: 16 bits
The constant 2, in accordance with the definition of this field in
Section 3 of RFC 3611 [RFC3611].
SSRC of source: 32 bits
The SSRC of the RTP data packet source being reported upon by this
report block. in accordance with the definition of this field in
Section 3 of RFC 3611 [RFC3611].
Loss Distance: 16 bits
The mean duration, expressed in milliseconds, of the loss
intervals that have occurred since the beginning of reception
[DSLF]. The duration of each loss distance is calculated based
upon the frames that mark the beginning and end of that period.
It is equal to the timestamp of the end frame, plus the duration
of the end frame, minus the timestamp of the beginning frame. If
the actual values are not available, estimated values MUST be
used. If there have been no burst periods, the burst duration
value MUST be zero.
Loss Period: 16 bits
The mean duration, expressed in milliseconds, of the burst loss
periods that have occurred since the beginning of reception
[DSLF]. The duration of each period is calculated based upon the
frame that marks the end of the prior burst loss and the frame
that marks the beginning of the subsequent burst loss. It is
equal to the timestamp of the subsequent burst frame, minus the
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timestamp of the prior burst packet, plus the duration of the
prior burst packet. If the actual values are not available,
estimated values MUST be used. In the case of a gap that occurs
at the beginning of reception, the sum of the timestamp of the
prior burst packet and the duration of the prior burst packet are
replaced by the reception start time. In the case of a gap that
occurs at the end of reception, the timestamp of the subsequent
burst packet is replaced by the reception end time. If there have
been no gap periods, the gap duration value MUST be zero.
Max Loss Duration of a single error: 16 bits
The maximum loss duration, expressed in milliseconds, of the loss
periods that have occurred since the beginning of reception. The
recommended max loss duration is specified as less than 16 ms in
[DSLF], which provides a balance between interleaver depth
protection from xDSL errors induced by impulse noise, delay added
to other applications and video service QoE requirements to reduce
visible impairments.
Reserved: 16 bits
All bits SHALL be set to 0 by the sender and SHALL be ignored on
reception.
block length: 16 bits
The constant 2, in accordance with the definition of this field in
Section 3 of RFC 3611 [RFC3611].
5.4. Synthetical Multimedia Quality Metrics Block
This block reports the multimedia application performance or quality
metrics beyond the information carried in the standard RTCP packet
format. Information is recorded about multimedia application QoE
metric which is expressed as a MOS ("Mean Opinion Score"), MOS is on
a scale from 1 to 5, in which 5 represents excellent and 1 represents
unacceptable. MOS scores are usually obtained using subjective
testing or using objective algorithm to estimate the multimedia
quality. However Subjective testing is not suitable for measuring
the multimedia quality since the results may vary from test to test.
Therefore using objective algorithm to calculate MOS scores is
recommended. ITU-T recommendation [G.1082][P.NAMS][P.NBAMS] defines
a methodology for verifying the performance of QoE estimation
algorithms for video and audio. Hence this document recommends
vendors and implementers to use International Telecommunication Union
(ITU)-specified methodologies to measure parameters when possible.
The report block contents are dependent upon a series of flag bits
carried in the first part of the header. Not all parameters need to
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be reported in each block. Flags indicate which are and which are
not reported. The fields corresponding to unreported parameters MUST
be present, but are set to zero. The receiver MUST ignore any
Perceptual Quality Metrics Block with a non-zero value in any field
flagged as unreported.
The Synthetical Multimedia Quality Metrics Block has the following
format:
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| BT=TBD |I| MC | Rsd.| block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MOS Value |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Block type (BT): 8 bits
The Perceptual Quality Metrics Block is identified by the constant
<SMQM>.
Interval Metric flag (I): 1 bit
This field is used to indicate whether the Basic Loss/Discard
metrics are Interval or Cumulative metrics, that is, whether the
reported values applies to the most recent measurement interval
duration between successive metrics reports (I=1) (the Interval
Duration) or to the accumulation period characteristic of
cumulative measurements (I=0) (the Cumulative Duration).
MoS Class (MC): 4 bits
This field is used to indicate the MOS type to be reported. The
MOS type is defined as follows:
0000 MOS-A - Audio Quality MOS [G.107][P.564].
0001 MOS-V - Video Quality MOS [P.NAMS][P.NBAMS].
0010 MOS-AV - Audio-Video Quality MOS[P.NAMS][P.NBAMS].
0100~1111 - Reserved
Rsd.: 7 bits
This field is reserved for future definition. In the absence of
such a definition, the bits in this field MUST be set to zero and
MUST be ignored by the receiver.
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SSRC of source: 32 bits
As defined in Section 4.1 of [RFC3611].
MOS Value: Variable Length
The estimated mean opinion score for Audio Qulity, Video Quality
or Audio-Video quality is defined as including the effects of
delay and other effects that would affect Audio-Video quality
[G.1082][P.NAMS][P.NBAMS]. It is expressed as an integer in the
range 10 to 50, corresponding to MOS x 10, as for MOS. A value of
127 indicates that this parameter is unavailable. Values other
than 127 and the valid range defined above MUST NOT be sent and
MUST be ignored by the receiving system.
6. SDP Signaling
Six new parameters are defined for the six report blocks defined in
this document to be used with Session Description Protocol (SDP)
[RFC4566] using the Augmented Backus-Naur Form (ABNF) [RFC5234].
They have the following syntax within the "rtcp-xr" attribute
[RFC3611]:
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rtcp-xr-attrib = "a=rtcp-xr:"
[xr-format *(SP xr-format)] CRLF
xr-format = RTP-flows-init-syn
/ RTP-flows-general-syn
/ multimedia-quality-metrics
/ transport-stream-loss-metrics
/ transport-stream-burst-metrics
/ transport-stat-summary
/ layered-stream-stat-metrics
RTP-flows-init-syn = "RTP-flows-init-syn"
["=" max-size]
max-size = 1*DIGIT ; maximum block size in octets
RTP-flow-general-syn = "RTP-flows-general-syn"
["=" max-size]
max-size = 1*DIGIT ; maximum block size in octets
transport-stream-burst-metrics = "transport-stream-burst-metrics"
["=" max-size]
max-size = 1*DIGIT ; maximum block size in octets
transport-stream-loss-metrics = "transport-stream-loss-metrics"
["=" stat-flag *("," stat-flag)]
stat-flag = "key Frame loss and duplication"
/ "derivation Frame loss and duplication"
transport-stream-stat-summary = "transport-stream-stat-summary"
["=" stat-flag *("," stat-flag)]
stat-flag = "key Frame loss and duplication"
/ "derivation Frame loss and duplication"
layered-stream-stat-metrics = "layered-stream-stat-metrics"
["=" stat-flag *("," stat-flag)]
stat-flag = "base layer packet"
/ "enhancment layer packet"
multimedia-quality-metrics = "multimedia-quality-metrics"
["=" stat-flag *("," stat-flag)]
stat-flag = "Interval Metrics"
/"Cumulative metrics"
Refer to Section 5.1 of RFC 3611 [RFC3611] for a detailed description
and the full syntax of the "rtcp-xr" attribute.
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7. IANA Considerations
New report block types for RTCP XR are subject to IANA registration.
For general guidelines on IANA allocations for RTCP XR, refer to
Section 6.2 of [RFC3611].
This document assigns six new block type values in the RTCP XR Block
Type Registry:
Name: RFISD
Long Name: RTP Flows Initial Synchronization Delay
Value <RFISD>
Reference: Section 4.1
Name: RFGSO
Long Name: RTP Flows General Synchronization Offset Metrics Block
Value <RFGSO>
Reference: Section 4.2
Name: TSSS
Long Name: Transport Stream Statistics Summary
Value <TSSS>
Reference: Section 5.1
Name: LSSM
Value <LSSM>
Long Name: Layered Stream Statistics Metrics
Reference: Section 4.3
Name: TSLDM
Long Name: Transport Stream Loss and Discard Metrics
Value <TSLDM>
Reference: Section 5.2
Name: TSBM
Long Name: Transport Stream Burst Metrics
Value <TSBM>
Reference: Section 5.3
Name: SMQM
Long Name: Synthetical Multimedia Quality Metric
Value <SMQM>
Reference: Section 5.4
This document also registers seven SDP [RFC4566] parameters for the
"rtcp-xr" attribute in the RTCP XR SDP Parameters Registry:
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* "RTP-flows-init-syn"
* "RTP-flows-general-syn"
* "multimedia-quality-metrics"
* "transport-stream-loss-metrics"
* "transport-stream-burst-metrics"
* "transport-stat-summary"
* "layered-stream-stat-metrics"
The contact information for the registrations is:
Qin Wu
sunseawq@huawei.com
101 Software Avenue, Yuhua District
Nanjing, JiangSu 210012 China
8. Security Considerations
The new RTCP XR report blocks proposed in this document introduces no
new security considerations beyond those described in [RFC3611].
9. Acknowledgements
The authors would like to thank Bill Ver Steeg, David R Oran, Ali
Begen,Colin Perkins, Roni Even,Youqing Yang, Wenxiao Yu and Yinliang
Hu for their valuable comments and suggestions on this document.
10. References
10.1. Normative References
[I-D.ietf-avt-rtp-svc]
Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
"RTP Payload Format for Scalable Video Coding",
draft-ietf-avt-rtp-svc-27 (work in progress),
February 2011.
[ISO-IEC.13818-1.2007]
International Organization for Standardization,
"Information technology - Generic coding of moving
pictures and associated audio information: Systems",
ISO International Standard 13818-1, October 2007.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
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[RFC2250] Hoffman, D., Fernando, G., Goyal, V., and M. Civanlar,
"RTP Payload Format for MPEG1/MPEG2 Video", RFC 2250,
January 1998.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, July 2003.
[RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control
Protocol Extended Reports (RTCP XR)", RFC 3611,
November 2003.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, July 2006.
[RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax
Specifications: ABNF", STD 68, RFC 5234, January 2008.
[RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", RFC 6051, November 2010.
10.2. Informative References
[DSLF] Rahrer, T., Ed., Fiandra, Ed., and Wright, Ed., "Triple-
play Services Quality of Experience (QoE) Requirements",
DSL Forum Technical Report TR-126, December 2006.
[G.107] ITU-T, "The E Model, a computational model for use in
transmission planning", ITU-T Recommendation G.107,
April 2009.
[G.1082] ITU-T, "Measurement-based methods for improving the
robustness of IPTV performance", ITU-T
Recommendation G.1082, April 2009.
[I-D.ietf-pmol-metrics-framework-02]
Clark, A., "Framework for Performance Metric Development".
[IEEE] IEEE, "Human Perception of Jitter and Media
Synchronization", IEEE Journal on Selected Areas in
Communications Vol. 14, No. 1, January 1996.
[P.564] ITU-T, "Conformance testing for narrowband Voice over IP
transmission quality assessment models", ITU-T
Recommendation P.564, July 2006.
[P.NAMS] ITU-T, "Non-intrusive parametric model for the Assessment
of performance of Multimedia Streaming", ITU-T
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Recommendation P.NAMS, November 2009.
[P.NBAMS] ITU-T, "non-intrusive bit-stream model for assessment of
performance of multimedia streaming", ITU-T
Recommendation P.NBAMS, November 2009.
[RFC3357] Koodli, R. and R. Ravikanth, "One-way Loss Pattern Sample
Metrics", RFC 3357, August 2002.
Authors' Addresses
Qin Wu
Huawei
101 Software Avenue, Yuhua District
Nanjing, Jiangsu 210012
China
Email: sunseawq@huawei.com
Glen Zorn
Network Zen
77/440 Soi Phoomjit, Rama IV Road
Phra Khanong, Khlong Toie
Bangkok 10110
Thailand
Phone: +66 (0) 87 502 4274
Email: gwz@net-zen.net
Roland Schott
Deutsche Telekom Laboratories
Deutsche-Telekom-Allee 7
Darmstadt 64295
Germany
Email: Roland.Schott@telekom.de
Wu, et al. Expires September 3, 2011 [Page 23]
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