One document matched: draft-wu-avt-rtcp-xr-quality-monitoring-02.txt
Differences from draft-wu-avt-rtcp-xr-quality-monitoring-01.txt
Network Working Group Q. Wu
Internet-Draft Huawei
Intended status: Standards Track G. Zorn
Expires: March 14, 2011 Network Zen
R. Schott
Deutsche Telekom Laboratories
September 10, 2010
RTP Control Protocol Extended Reports (RTCP XR) Report Blocks for Real-
time Video Quality Monitoring
draft-wu-avt-rtcp-xr-quality-monitoring-02
Abstract
This document defines a set of RTP Control Protocol Extended Reports
(RTCP XR) Report Blocks and associated SDP parameters allowing the
report of video 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
working documents as Internet-Drafts. The list of current Internet-
Drafts is at http://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on March 14, 2011.
Copyright Notice
Copyright (c) 2010 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(http://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
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include Simplified BSD License text as described in Section 4.e of
the Trust Legal Provisions and are provided without warranty as
described in the Simplified BSD License.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1. Standards Language . . . . . . . . . . . . . . . . . . . . 4
2.2. Acronyms . . . . . . . . . . . . . . . . . . . . . . . . . 4
3. Applicability . . . . . . . . . . . . . . . . . . . . . . . . 5
4. Transport Layer Metrics . . . . . . . . . . . . . . . . . . . 5
4.1. RTP Flows Initial Synchronization Delay Report Block . . . 5
4.2. Audio-Video Playout Offset Report Block . . . . . . . . . 6
4.3. Layered Streams Statistics Metrics Block . . . . . . . . . 7
5. Application Layer Metrics . . . . . . . . . . . . . . . . . . 9
5.1. Video Statistics Summary Report Block . . . . . . . . . . 9
5.2. TR 101 290 Decodability Metrics Report Block . . . . . . . 11
5.3. Video Stream Loss and Discard Metrics Block . . . . . . . 14
5.4. Video Stream Burst Metrics Block . . . . . . . . . . . . . 16
5.5. Synthetical Multimedia Quality Metrics Block . . . . . . . 18
5.6. RTP Terminal Configuration Metrics Block . . . . . . . . . 20
6. SDP Signaling . . . . . . . . . . . . . . . . . . . . . . . . 22
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 23
8. Security Considerations . . . . . . . . . . . . . . . . . . . 25
9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 25
10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 25
10.1. Normative References . . . . . . . . . . . . . . . . . . . 25
10.2. Informative References . . . . . . . . . . . . . . . . . . 26
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 27
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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
IP, to ensure that all end users obtain acceptable video/audio
quality. The main drives come from operators, 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 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 video quality, since
the ability to analyze the video 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 video quality for operators
when transporting video across a network, the video quality Metrics
is highly required which can be conveyed in the RTCP XR
packets[RFC3611] and may have the following three benefits:
* Tuning the video encoder algorithm to satisfy video 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
* Verifying the continued correct operation of an existing system
RFC 3611 [RFC3611] defines seven report block formats for network
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management and quality monitoring. However, there are no block types
specifically designed for conveying video quality metrics. This
document focuses on specifying new report block types used to convey
video-specific quality metrics.
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 video quality and are sensitivity
to network errors.
Nine report block formats are defined by this document. Of these,
three are transport layer metrics:
* RTP Flows Initial Synchronization Delay Report Block
* Audio-Video Playout Offset Report Block
* Layered Streams Statistics Metrics Block
The other six are application layer metrics:
* Video Statistics Summary Report Block
* TR 101 290 Decodability Metrics Block [ETSI]
* Video Stream Loss and Discard Metrics Block
* Video Stream Burst Metrics Block
* Synthetical Multimedia Quality Metrics Block
* RTP Terminal Configuration 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].
2.2. Acronyms
SSRC
Synchronization Source [RFC3550]
TS
Transport Stream [ISO-IEC.13818-1.2007]
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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.
The Audio-Video Synchronization Report Block Section 4.2 has been
defined for video conferencing applications. Such applications can
use this report block to monitor A/V component synchronization to
ensure satisfactory QoE. Tighter tolerances than typically used have
been recommended for such applications.
The Flows 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 ensure synchronization
between different media layers for the same multimedia session.
The Video Stream Loss and Discard Metrics Report Block, Video Stream
Burst Metrics report Block, Video Statistics Summary Report Block and
Layered Video Statistics Metrics Block can be applied to any real
time video application, while the TR 101 290 Decodability Metrics
Report Block and Synthetical Multimedia Quality Metrics Report Block
can be used in any real-time AV application [ETSI].
4. Transport Layer Metrics
4.1. RTP Flows Initial Synchronization Delay Report Block
This block reports the initial synchronization delay between RTP
sessions of the same media stream sent using Multi-Session
Transmission [I-D.ietf-avt-rtp-svc] or the initial synchronization
delay betwen RTP session of the different media types
[I-D.ietf-avt-rapid-rtp-sync], which is beyond the information
carried in the standard RTCP packet format. Information is recorded
about session bandwidth and synchronization delay.
The RTP Flows Intial Synchronization Delay 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 | 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.
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 [I-D.ietf-avt-rapid-rtp-sync]. 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 [I-D.ietf-avt-rapid-rtp-sync]
4.2. Audio-Video Playout Offset Report Block
This block reports the audio-video playout offset requirements
between audio and video components beyond the information carried in
the standard RTCP packet format. Information is recorded about
average tolerant audio lag video time and typically used for video
conferencing applications. The Audio-Video Playout Offset 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 |I| Reserved | Block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Audio Lag Video |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Block type (BT): 8 bits
The Statistics Summary Report Block is identified by the constant
<AVPO>.
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].
Audio Lag Video time: 32 bits
This field indicates the tolerant audio lag video time in
milliseconds [DSLF][I-D.ietf-avt-rapid-rtp-sync]. This value is
calculated based on the interarrival time between previous video/
audio RTP packet and the next first audio/video RTP packet and
timestamps of both previous video/audio RTP packet and the next
first audio/video packet. The recommended acceptable audio lag
video is specified as less than tolerance 80 ms [IEEE]
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
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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 Video Statistics Summary Report Block with a non-zero
value in any field flagged as unreported.
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 |L|D|LT | 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>.
loss report flag (L): 1 bit
Bit set to 1 if the lost_frame packets field, the
lost_layered_component field or the lost partial frame packets
field contains a report, 0 otherwise.
duplicate report flag (D): 1 bit
Bit set to 1 of the dup_frame packets field, the
dup_layered_component packets field or the dup partial frame
packets field contains a report, 0 otherwise.
Layer Type flag (LT): 3 bits
This field is used to indicate the Layer Type of layered video to
be reported. LT is set to 1 if the loss_component_packet field
and dup_component packet contain the base layer packet in SVC
[I-D.ietf-avt-rtp-svc], 2 if the loss_component packet field and
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dup_component packet contain enhancement layer 1 packet in SVC, 3
if the loss_component packet field and dup_component packet
contain the enhancement layer 2 packet, 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 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].
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. Video 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
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ignore any Video Statistics Summary Report Block with a non-zero
value in any field flagged as unreported.
The Video Statistics Summary 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 |L|D|FT |P| rsd.| block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| lost_frames |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| dup frames |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| lost_partial frame packets |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| dup partial frame_packets |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
Block type (BT): 8 bits
The Video Statistics Summary Report Block is identified by the
constant <VSS>.
loss report flag (L): 1 bit
Bit set to 1 if the lost_frame packets field, the
lost_layered_component field or the lost partial frame packets
field contains a report, 0 otherwise.
duplicate report flag (D): 1 bit
Bit set to 1 of the dup_frame packets field, the
dup_layered_component packets field or the dup partial frame
packets field contains a report, 0 otherwise.
Frame type indicator (FT): 2 bits
This field is used to indicate the frame type to be reported.
Bits set to 01 if the lost_frames field or dup_frames field
contain a I_frame report, 10 if the lost_frames field and
dup_frames field contain a P frame report, 11 if the lost_frames
field and dup_frames field contain a B frame report, 00 otherwise.
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P: 1 bit
Bit set to 1 if the lost_partial frame packets field or the
dup_partial_frame packets 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].
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.
lost_partial frame packets: 32 bits
Number of lost_partial frame packets in the above sequence number
interval.
dup_partial frame packets: 32 bits
Number of dup_partial frame packets in the above sequence number
interval.
5.2. TR 101 290 Decodability Metrics Report Block
This block reports decodability metrics statistics beyond the
information carried in the standard RTCP packet format. Information
is recorded about the number of Transport Stream Synchronization
Losses, Sync byte errors, Continuity count errors, Transport errors,
Program Clock Reference (PCR) errors, PCR repetition errors, PCR
discontinuity indicator errors, and Presentation Time Stamp (PTS)
errors [ETSI]. Such information can be useful for network management
and video quality monitoring.
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Note that this is only applicable to MPEG-2 RTP streams [RFC2250].
and not to any other video codec.
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 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
Decodability Metrics Block with a non-zero value in any field flagged
as unreported.
The Decodability 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 |L|B|C|T|P|S|rvd| block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| begin_seq | end_seq |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Number of packets | Number of TSs |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Transport Stream Synchronization Losses |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Sync byte errors |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Continuity count errors |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Transport errors |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| PCR errors |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| PCR repetition errors |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| PCR discontinuity indicator errors |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| PTS errors |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
A TR 101 290 decodability metrics report block is identified by
the constant <TDM>.
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Transport Stream Synchronization Loss flag (L): 1 bit
Bit set to 1 if the Transport Stream Synchronization Loss field
contains a report, 0 otherwise.
Sync byte error flag (B): 1 bit
Bit set to 1 if the Sync byte error field contains a report, 0
otherwise.
Continuity count error flag (C): 1 bit
Bit set to 1 if the Continuity count error field contains a
report, 0 otherwise.
Transport error flag (T): 1 bit
Bit set to 1 if the Transport error field contains a report, 0
otherwise.
PCR related error flag (P): 1 bit
Bit set to 1 if the PCR error field, PCR repetition error field
and PCR discontinuity indicator error fields contain a report, 0
otherwise.
PTS error flag (S): 1 bit
Bit set to 1 if the PTS error field contains a report, 0
otherwise.
rvd: 2 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 10, 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 [RFC3611].
begin_seq: 16 bits
As defined in Section 4.1 of [RFC3611].
end_seq: 16 bits
As defined in Section 4.1 of [RFC3611].
Number of Packets: 16 bits
Number of RTP packets in the above sequence number interval.
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Number of TSs: 16 bits
Number of Transport Streams in the above sequence number interval.
Transport Stream Synchronization Losses: 32 bits
Number of Transport Stream Synchronization Losses in the above
sequence number interval.
Sync byte errors: 32 bits
Number of Transport sync byte errors in the above sequence number
interval.
Continuity count error: 32 bits
Number of Transport Continuity count errors in the above sequence
number interval.
Transport errors: 32 bits
Number of Transport errors in the above sequence number interval.
PCR errors: 32 bits
Number of PCR errors in the above sequence number interval.
PCR repetition errors: 32 bits
Number of Transport PCR repetition errors in the above sequence
number interval.
PCR discontinuity indicator errors: 32 bits
Number of PCR discontinuity indicator errors in the above sequence
number interval.
PTS errors: 32 bits
Number of PTS errors in the above sequence number interval.
5.3. Video 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.
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The block is encoded as six 32-bit words:
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 |FT | reserved | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| Loss rate | Discard rate |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
A Video Stream Metrics Report Block is identified by the constant
<VSLDM>.
Frame type indicator (FT): 2 bits
This field is used to indicate the frame type to be reported.
Bits set to 01 if the Loss rate field and discard rate field
contain a I_frame report, 10 if the Loss rate field and discard
rate field contain a P frame report, 11 if the Loss rate field and
discard rate field contain a B frame report, 00 otherwise.
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., I 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
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categorize late-arriving packets as lost. The degree of lateness
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., I-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.
5.4. Video 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 Video Stream Metrics Report Block is identified by the constant
<VSBM>.
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
timestamp of the prior burst packet, plus the duration of the
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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].
5.5. Synthetical Multimedia Quality Metrics Block
This block reports the multimedia quality metrics beyond the
information carried in the standard RTCP packet format. Information
is recorded about Video MOS, Audio MOS, Audio Video MOS, Video
Service Transmission Quality [G.1082][P.NAMS].
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 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:
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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 |I|V|A|M|T|Rsd. | block length |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| SSRC of source |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MOS-V | MOS-A |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| MOS-AV | VSTQ |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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-V flag (V): 1 bit
Bit set to 1 if the MOS-V field and MOS-AV field contain a report,
0 otherwise.
MOS-A flag (A): 1 bit
Bit set to 1 if the MOS-A field contain a report, 0 otherwise.
MOS-AV flag (M): 1 bit
Bit set to 1 if the MOS-AV field contain a report, 0 otherwise.
Video Service Transmission Quality flag (T): 1 bit
Bit set to 1 if the VSTQ 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.
SSRC of source: 32 bits
As defined in Section 4.1 of [RFC3611].
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MOS-V: 16 bits
The estimated mean opinion score for video quality (MOS-V) is a
video quality metric on a scale from 1 to 5, in which 5 represents
excellent and 1 represents unacceptable [G.1082][P.NAMS]. This
metric is defined as not including the effects of audio
impairments and can be compared to MOS scores obtained from video
quality tests. It is expressed as an integer in the range 10 to
50, corresponding to MOS x 10. For example, a value of 35 would
correspond to an estimated MOS score of 3.5.
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.
MOS-A: 16 bits
The estimated mean opinion score for Audio quality (MOS-A) is
defined as including the effects of delay and other effects that
would affect Audio-Video quality [RFC3611]. It is expressed as an
integer in the range 10 to 50, corresponding to MOS x 10, as for
MOS-A.
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.
MOS-AV: 16 bits
The estimated mean opinion score for Audio-Video quality (MOS-AV)
is defined as including the effects of delay and other effects
that would affect Audio-Video quality [G.1082][P.NAMS]. It is
expressed as an integer in the range 10 to 50, corresponding to
MOS x 10, as for MOS-AV. 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.
VSTQ: 16 bits
Video Service Transmission Quality (TBC) .
5.6. RTP Terminal Configuration Metrics Block
This block reports the RTP Terminal Configuration metrics beyond the
information carried in the standard RTCP packet format. Information
is recorded about packet loss concealment,jitter buffer
adaptive,jitter buffer rate described in section 4.7.6 of [RFC3611]
and FEC type described in [I-D.ietf-fecframe-raptor],
[I-D.ietf-fecframe-interleaved-fec-scheme].
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The block is encoded as six 32-bit words:
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|PLC|JBA|JB Rate|FEC| Rsd. | Reserved |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
block type (BT): 8 bits
A Video Stream Metrics Report Block is identified by the constant
<RTCM>.
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.
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].
Rsd.: 6 bits
All bits SHALL be set to 0 by the sender and SHALL be ignored on
reception.
Reserved: 16 bits
All bits SHALL be set to 0 by the sender and SHALL be ignored on
reception.
Packet loss concealment (PLC): 2 bits
Standard (11) / enhanced (10) / disabled (01) / unspecified (00)
.As described in [RFC3611].When PLC = 11, then a simple replay or
interpolation algorithm is being used to fill-in the missing
packet; this approach is typically able to conceal isolated lost
packets at low packet loss rates. When PLC = 10, then an enhanced
interpolation algorithm is being used; algorithms of this type are
able to conceal high packet loss rates effectively. When PLC =
01, then silence is being inserted in place of lost packets. When
PLC = 00, then no information is available concerning the use of
PLC; however, for some codecs this may be inferred.
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Jitter buffer adaptive (JBA): 2 bits
Adaptive (11) / non-adaptive (10) / reserved (01)/ unknown (00).
As described in [RFC3611], when the jitter buffer is adaptive,
then its size is being dynamically adjusted to deal with varying
levels of jitter. When non-adaptive, the jitter buffer size is
maintained at a fixed level. When either adaptive or non-
adaptive modes are specified, then the jitter buffer size
parameters below MUST be specified.
Jitter buffer rate (JB rate): 4 bits
As described in [RFC3611]J = adjustment rate (0-15), This
represents the implementation specific adjustment rate of a jitter
buffer in adaptive mode. This parameter is defined in terms of
the approximate time taken to fully adjust to a step change in
peak to peak jitter from 30 ms to 100 ms such that: adjustment
time = 2 * J * frame size (ms)
Forward Error Correction (FEC):2 bits
Raptor FEC SchemeA (00),Raptor FEC Scheme B(01),Raptor FEC Scheme
C (10), 1D Interleaved parity codec(11)
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]:
rtcp-xr-attrib = "a=rtcp-xr:"
[xr-format *(SP xr-format)] CRLF
xr-format = RTP-flows-syn
/ audio-video-ofset
/ multimedia-quality-metrics
/ video-stream-loss-metrics
/ video-stream-burst-metrics
/ video-stat-summary
/ layed-video-stat-metrics
/ decodability-metrics
/terminal-metrics
RTP-flows-syn = "RTP-flows-syn"
["=" max-size]
max-size = 1*DIGIT ; maximum block size in octets
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audio-video-ofset = "audio-video-ofset"
["=" max-size]
max-size = 1*DIGIT ; maximum block size in octets
multimedia-quality-metrics = "multimedia-quality-metrics"
["=" max-size]
max-size = 1*DIGIT ; maximum block size in octets
video-stream-loss-metrics = "video-stream-loss-metrics"
["=" max-size]
max-size = 1*DIGIT ; maximum block size in octets
video-stream-burst-metrics = "video-stream-burst-metrics"
["=" max-size]
max-size = 1*DIGIT ; maximum block size in octets
video-stat-summary = "video-stat-summary"
["=" stat-flag *("," stat-flag)]
stat-flag = "I Frame loss and duplication"
/ "P Frame loss and duplication"
/ "B Frame loss and duplication"
layered-stream-stat-metrics = "layered-stream-stat-metrics"
["=" stat-flag *("," stat-flag)]
stat-flag = "I Frame loss and duplication"
/ "P Frame loss and duplication"
/ "B Frame loss and duplication"
decodability-metrics = "decodability-metrics"
["=" stat-flag *("," stat-flag)]
stat-flag = "Interval Metric"
/ "MOS-V"
/ "MOS-A"
/ "MOS-AV"
/ "VSTQ"
Terminal-metrics = "Terminal-metrics"
["=" max-size]
max-size = 1*DIGIT ; maximum block size in octets
Refer to Section 5.1 of RFC 3611 [RFC3611] for a detailed description
and the full syntax of the "rtcp-xr" attribute.
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].
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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: AVPO
Long Name: Audio-Video Playout Offset
Value <AVPO>
Reference: Section 4.2
Name: VSS
Long Name: Video Statistics Summary
Value <VSS>
Reference: Section 5.1
Name: LSSM
Value <LSSM>
Long Name: Layered Stream Statistics Metrics
Reference: Section 4.3
Name: TDM
Long Name: TR 101 290 Decodability Metrics
Value <TDM>
Reference: Section 5.2
Name: VSLDM
Long Name: Video Stream Loss and Discard Metrics
Value <VSLDM>
Reference: Section 5.3
Name: VSBM
Long Name: Video Stream Burst Metrics
Value <VSBM>
Reference: Section 5.4
Name: SMQM
Long Name: Synthetical Multimedia Quality Metric
Value <SMQM>
Reference: Section 5.5
Name: RTCM
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Long Name: RTP Terminal Configuration Metrics
Value <RTCM>
Reference: Section 5.6
This document also registers six SDP [RFC4566] parameters for the
"rtcp-xr" attribute in the RTCP XR SDP Parameters Registry:
* "RTP-flows-syn"
* "audio-video-ofset"
* "multimedia-quality-metrics"
* "video-stream-loss-metrics"
* "video-stream-burst-metrics"
* "video-stat-summary"
* "layered-stream-stat-metrics"
* "decodability-metrics"
* "terminal-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
TBC
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
[ETSI] ETSI, "Digital Video Broadcasting (DVB); Measurement
guidelines for DVB systems", Technical Report TR 101 290,
2001.
[I-D.ietf-avt-rapid-rtp-sync]
Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP
Flows", draft-ietf-avt-rapid-rtp-sync-12 (work in
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progress), July 2010.
[I-D.ietf-avt-rtp-svc]
Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis,
"RTP Payload Format for SVC Video",
draft-ietf-avt-rtp-svc-22 (work in progress), August 2010.
[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.
[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.
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.1082] ITU-T, "Measurement-based methods for improving the
robustness of IPTV performance", ITU-T
Recommendation G.1082, April 2009.
[I-D.ietf-fecframe-interleaved-fec-scheme]
Begen, A., "RTP Payload Format for 1-D Interleaved Parity
FEC", draft-ietf-fecframe-interleaved-fec-scheme-09 (work
in progress), January 2010.
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[I-D.ietf-fecframe-raptor]
Waston, M., "Raptor FEC Schemes for FECFRAME",
draft-ietf-fecframe-raptor-02 (work in progress),
March 2010.
[IEEE] IEEE, "Human Perception of Jitter and Media
Synchronization", IEEE Journal on Selected Areas in
Communications Vol. 14, No. 1, January 1996.
[P.NAMS] ITU-T, "Non-intrusive parametric model for the Assessment
of performance of Multimedia Streaming", ITU-T
Recommendation P.NAMS, 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
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