One document matched: draft-wu-xrblock-rtcp-xr-quality-monitoring-01.xml
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<rfc category="std" docName="draft-wu-xrblock-rtcp-xr-quality-monitoring-01"
ipr="pre5378Trust200902">
<front>
<title abbrev="RTCP XR Quality Report Blocks">RTP Control Protocol
Extended Reports (RTCP XR) Report Blocks for Real-time Application Quality
Monitoring</title>
<author fullname="Qin Wu" initials="Q." surname="Wu">
<organization>Huawei</organization>
<address>
<postal>
<street>101 Software Avenue, Yuhua District</street>
<city>Nanjing</city>
<region>Jiangsu</region>
<code>210012</code>
<country>China</country>
</postal>
<email>sunseawq@huawei.com</email>
</address>
</author>
<author fullname="Glen Zorn" initials="G." surname="Zorn">
<organization>Network Zen</organization>
<address>
<postal>
<street>77/440 Soi Phoomjit, Rama IV Road</street>
<street>Phra Khanong, Khlong Toie</street>
<city>Bangkok</city>
<code>10110</code>
<country>Thailand</country>
</postal>
<phone>+66 (0) 87 502 4274</phone>
<email>gwz@net-zen.net</email>
</address>
</author>
<author fullname="Roland Schott" initials="R." surname="Schott">
<organization>Deutsche Telekom Laboratories</organization>
<address>
<postal>
<street>Deutsche-Telekom-Allee 7</street>
<street></street>
<city>Darmstadt</city>
<code>64295</code>
<country>Germany</country>
</postal>
<email>Roland.Schott@telekom.de</email>
</address>
</author>
<date year="2011" />
<abstract>
<t>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.</t>
</abstract>
</front>
<middle>
<section title="Introduction">
<t>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. <vspace blankLines="1" /> 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. <vspace
blankLines="1" /> 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. <vspace blankLines="1" /> 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<xref target="RFC3611"></xref> and may have the
following three benefits: <list>
<t><list style="symbols">
<t>Tuning the content encoder algorithm to satisfy real time
data quality requirements</t>
<t>Determining which system techniques to use in a given
situation and when to switch from one technique to another as
system parameters change</t>
<t>Verifying the continued correct operation of an existing
system</t>
</list></t>
</list> <vspace blankLines="1" /> <xref target="RFC3611">RFC
3611</xref> 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.<vspace
blankLines="1" /> 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.<vspace blankLines="1" /> Seven report block formats
are defined by this document. Of these, three are transport layer
metrics: <list>
<t><list style="symbols">
<t>RTP Flows Initial Synchronization Delay Report Block</t>
<t>RTP Flows General Synchronization Offset Metrics Block</t>
<t>Layered Streams Statistics Metrics Block</t>
</list></t>
</list> <vspace blankLines="1" /> The other four are application layer
metrics: <list>
<t><list style="symbols">
<t>Transport Stream Statistics Summary Report Block</t>
<t>Transport Stream Loss and Discard Metrics Block</t>
<t>Transport Stream Burst Metrics Block</t>
<t>Synthetical Multimedia Quality Metrics Block</t>
</list></t>
</list></t>
</section>
<section title="Terminology">
<section title="Standards Language">
<t>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 <xref
target="RFC2119">RFC 2119</xref>.</t>
<t>In addition, the following terms are defined:</t>
<t><list style="hanging">
<t hangText="Layered Component Packet"><vspace blankLines="1" />a
RTP packet using layered codecs containing the specified layered
component, e.g., encoded stream at the base layer or at the
enhancement layer.<vspace blankLines="1" /></t>
<t hangText="Picture Type"><vspace blankLines="1" />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. <vspace blankLines="1" /></t>
</list></t>
</section>
<section title="Acronyms">
<t><list style="hanging">
<t hangText="SSRC"><vspace blankLines="0" /> <xref
target="RFC3550">Synchronization Source</xref> <vspace
blankLines="1" /></t>
<t hangText="TS"><vspace blankLines="0" /> <xref
target="ISO-IEC.13818-1.2007">Transport Stream</xref></t>
</list></t>
</section>
</section>
<section title="Applicability">
<t>All the report blocks defined in this document could be used by
dedicated network monitoring applications. As specified in <xref
target="RFC3611">RFC 3611</xref>, 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. <vspace blankLines="1" /> RTP Flows General
Synchronization Offset Metrics Block in <xref target="RFGSO"></xref> 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. <vspace
blankLines="1" /> 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. <vspace blankLines="1" /> 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 AV application.</t>
</section>
<section title="Transport Layer Metrics">
<section anchor="RFISD"
title="RTP Flows Initial Synchronization Delay Report Block">
<t>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
<xref target="RFC6051"></xref>. 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 <xref target="RFC3550"></xref>. 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. <vspace
blankLines="1" /> The RTP Flows Initial Synchronization Delay Report
Block has the following format: <figure>
<artwork>
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
</artwork>
</figure> <list style="hanging">
<t hangText="Block type (BT): 8 bits"><vspace blankLines="0" />
The Statistics Summary Report Block is identified by the constant
<RFISD>. <vspace blankLines="1" /></t>
<t hangText="Reserved: 8 bits"><vspace blankLines="0" /> 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. <vspace blankLines="1" /></t>
<t hangText="Block length: 16 bits"><vspace blankLines="0" /> The
constant 3, in accordance with the definition of this field in
Section 3 of <xref target="RFC3611">RFC 3611</xref>. <vspace
blankLines="1" /></t>
<t hangText="SSRC of Sender: 32 bits"><vspace blankLines="0" />The
SSRC of the RTP data packet source being reported upon by this
report block. (<xref target="RFC3611">Section 4.1 of</xref>).
<vspace blankLines="1" /></t>
<t hangText="Initial Synchronization Delay: 32 bits"><vspace
blankLines="0" /> 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 <xref
target="RFC6051"></xref>. The value is calculated as follows:
<vspace blankLines="1" /> <list>
<t>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 <xref
target="RFC6051"></xref></t>
</list></t>
</list></t>
</section>
<section anchor="RFGSO"
title="RTP Flows General Synchronization Offset Metrics Block">
<t>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: <figure>
<artwork>
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
</artwork>
</figure> <list style="hanging">
<t hangText="Block type (BT): 8 bits"><vspace blankLines="0" />
The Statistics Summary Report Block is identified by the constant
<RFGSO>. <vspace blankLines="1" /></t>
<t hangText="Interval Metric flag (I): 1 bit"><vspace
blankLines="1" /> 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).<vspace blankLines="1" /></t>
<t hangText="Reserved: 8 bits"><vspace blankLines="0" /> 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. <vspace blankLines="1" /></t>
<t hangText="Block length: 16 bits"><vspace blankLines="0" /> The
constant 2, in accordance with the definition of this field in
Section 3 of <xref target="RFC3611">RFC 3611</xref>. <vspace
blankLines="1" /></t>
<t hangText="SSRC of source: 32 bits"><vspace blankLines="0" /> As
defined in Section 4.1 of <xref target="RFC3611">RFC 3611</xref>.
<vspace blankLines="1" /></t>
<t hangText="General synchronization offset: 32 bits"><vspace
blankLines="0" /> 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 <xref target="RFC6051"></xref> 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.</t>
</list></t>
</section>
<section anchor="LSSM" title="Layered Streams Statistics Metrics Block">
<t>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 <xref target="RFC3611">RFC
3611</xref>. Information is recorded about lost layered component
packets, duplicated layered component packets. Such information can be
useful for network management and video quality monitoring. <vspace
blankLines="1" /> 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. <vspace
blankLines="1" /> The Layered Stream Statistics metrics Block has the
following format: <figure>
<artwork>
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
</artwork>
</figure> <list style="hanging">
<t hangText="Block type (BT): 8 bits"><vspace blankLines="0" />
The Layered stream Statistics Metrics Block is identified by the
constant <LSSM>. <vspace blankLines="1" /></t>
<t hangText="Layer Type flag (T): 1 bits"><vspace
blankLines="0" /> 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 <xref
target="I-D.ietf-avt-rtp-svc"></xref>, 1 if the loss_component
packet field and dup_component packet contain enhancement layer
packet in layered codec. <vspace blankLines="1" /></t>
<t hangText="Rsd.: 3 bits"><vspace blankLines="0" /> 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.<vspace blankLines="1" /></t>
<t hangText="Block length: 16 bits"><vspace blankLines="0" /> The
constant 3, in accordance with the definition of this field in
Section 3 of <xref target="RFC3611">RFC 3611</xref>. <vspace
blankLines="1" /></t>
<t hangText="SSRC of source: 32 bits "><vspace blankLines="0" />
As defined in Section 4.1 of <xref target="RFC3611">RFC
3611</xref>. <vspace blankLines="1" /></t>
<t hangText="begin_seq: 16 bits "><vspace blankLines="0" /> As
defined in Section 4.1 of <xref target="RFC3611">RFC 3611</xref>.
<vspace blankLines="1" /></t>
<t hangText="end_seq: 16 bits "><vspace blankLines="0" /> As
defined in Section 4.1 of <xref target="RFC3611">RFC 3611</xref>.
<vspace blankLines="1" /></t>
<t hangText="Lost_Layered Component Packets: 32 bits"><vspace
blankLines="0" />Number of lost_component packets in the above
sequence number interval.<vspace blankLines="1" /></t>
<t hangText="Dup_Layered Component Packets: 32 bits"><vspace
blankLines="0" />Number of dup_component packets in the above
sequence number interval.<vspace blankLines="1" /></t>
</list></t>
</section>
</section>
<section title="Application Layer Metrics">
<section anchor="TSSS"
title="Transport Streams Statistics Summary Report Block">
<t>This block reports statistics beyond the information carried in the
Statistics Summary Report Block RTCP packet specified in the section
4.6 of <xref target="RFC3611">RFC 3611</xref>. 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. <vspace blankLines="1" /> 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.
<vspace blankLines="1" /> The Transport Streams Statistics Summary
Report Block has the following format: <figure>
<artwork>
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
</artwork>
</figure> <list style="hanging">
<t hangText="Block type (BT): 8 bits"><vspace blankLines="0" />
The Transport Statistics Summary Report Block is identified by the
constant <TSSS>. <vspace blankLines="1" /></t>
<t hangText="Picture type indicator (T): 1 bits"><vspace
blankLines="0" /> 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. <vspace
blankLines="1" /></t>
<t hangText="P: 1 bit"><vspace blankLines="0" /> Bit set to 1 if
the partial_lost_frames field or the partial_dup_ frames field
contains a report, 0 otherwise. <vspace blankLines="1" /></t>
<t hangText="Rsd.: 3 bits"><vspace blankLines="0" /> 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.<vspace blankLines="1" /></t>
<t hangText="Block length: 16 bits"><vspace blankLines="0" /> The
constant 5, in accordance with the definition of this field in
Section 3 of <xref target="RFC3611">RFC 3611</xref>. <vspace
blankLines="1" /></t>
<t hangText="SSRC of source: 32 bits "><vspace blankLines="0" />
As defined in Section 4.1 of <xref target="RFC3611">RFC
3611</xref>. <vspace blankLines="1" /></t>
<t hangText="begin_seq: 16 bits "><vspace blankLines="0" /> As
defined in Section 4.1 of <xref target="RFC3611">RFC 3611</xref>.
<vspace blankLines="1" /></t>
<t hangText="end_seq: 16 bits "><vspace blankLines="0" /> As
defined in Section 4.1 of <xref target="RFC3611">RFC 3611</xref>.
<vspace blankLines="1" /></t>
<t hangText="lost_frames: 32 bits"><vspace blankLines="0" />
Number of lost_frames in the above sequence number interval.
<vspace blankLines="1" /></t>
<t hangText="dup_frames: 32 bits"><vspace blankLines="0" /> Number
of dup_frames in the above sequence number interval. <vspace
blankLines="1" /></t>
<t hangText="partial lost_frames: 32 bits"><vspace
blankLines="0" />Number of partial lost_frames in the above
sequence number interval. <vspace blankLines="1" /></t>
<t hangText="partial dup_frames: 32 bits"><vspace
blankLines="0" />Number of partial_dup_frames in the above
sequence number interval. <vspace blankLines="1" /></t>
<t hangText="key frames impairment proportion:32bits"><vspace
blankLines="0" />The proportion of key frame impaired by packet
loss,discard and duplication.<vspace blankLines="1" /></t>
</list></t>
</section>
<section anchor="TSLDM"
title="Transport Stream Loss and Discard Metrics Block">
<t>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.
<vspace blankLines="1" />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. <vspace blankLines="1" /> 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.<vspace blankLines="1" />The
block is encoded as six 32-bit words: <figure>
<artwork>
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
</artwork>
</figure><list style="hanging">
<t hangText="block type (BT): 8 bits"><vspace blankLines="0" /> A
Transport Stream Metrics Report Block is identified by the
constant <TSLDM>. <vspace blankLines="1" /></t>
<t hangText="Picture type indicator (T): 1 bits"><vspace
blankLines="0" /> 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. <vspace
blankLines="1" /></t>
<t hangText="reserved: 6 bits"><vspace blankLines="0" /> 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. <vspace blankLines="1" /></t>
<t hangText="block length: 16 bits"><vspace blankLines="0" /> The
constant 1, in accordance with the definition of this field in
Section 3 of <xref target="RFC3611">RFC 3611</xref>. <vspace
blankLines="1" /></t>
<t hangText="SSRC of source: 32 bits"><vspace blankLines="0" />
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 <xref target="RFC3611">RFC 3611</xref>.</t>
<t hangText="Loss rate: 8 bits"><vspace blankLines="0" /> 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
that triggers a loss SHOULD be significantly greater than that
which triggers a discard.<vspace blankLines="1" /></t>
<t hangText="Discard rate: 8 bits"><vspace blankLines="0" /> 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.<vspace
blankLines="1" /></t>
</list></t>
</section>
<section anchor="TSBM" title="Transport Stream Burst Metrics Block">
<t>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. <vspace blankLines="1" /> In order to properly assess the
quality of a video stream, it is desirable to consider the degree of
burstiness of packet loss <xref target="RFC3357">RFC 3357</xref>.
Following the one-way loss pattern sample metrics discussed in <xref
target="RFC3357"></xref>, 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. <vspace blankLines="1" />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. <vspace blankLines="1" />The block is
encoded as six 32-bit words: <figure>
<artwork>
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. |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
</artwork>
</figure><list style="hanging">
<t hangText="block type (BT): 8 bits"><vspace blankLines="0" /> A
Transport Stream Metrics Report Block is identified by the
constant <TSBM>. <vspace blankLines="1" /></t>
<t hangText="reserved: 8 bits"><vspace blankLines="0" /> 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. <vspace blankLines="1" /></t>
<t hangText="block length: 16 bits"><vspace blankLines="0" /> The
constant 2, in accordance with the definition of this field in
Section 3 of <xref target="RFC3611">RFC 3611</xref>. <vspace
blankLines="1" /></t>
<t hangText="SSRC of source: 32 bits"><vspace blankLines="0" />
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 <xref target="RFC3611">RFC 3611</xref>.<vspace
blankLines="1" /></t>
<t hangText="Loss Distance: 16 bits"><vspace blankLines="0" /> The
mean duration, expressed in milliseconds, of the loss intervals
that have occurred since the beginning of reception <xref
target="DSLF"></xref>. 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. <vspace blankLines="1" /></t>
<t hangText="Loss Period: 16 bits"><vspace blankLines="0" /> The
mean duration, expressed in milliseconds, of the burst loss
periods that have occurred since the beginning of reception <xref
target="DSLF"></xref>. 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 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. <vspace
blankLines="1" /></t>
<t hangText="Max Loss Duration of a single error: 16 bits"><vspace
blankLines="0" /> 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 <xref target="DSLF"></xref>, 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.<vspace blankLines="1" /></t>
<t hangText="Reserved: 16 bits"><vspace blankLines="0" />All bits
SHALL be set to 0 by the sender and SHALL be ignored on
reception.<vspace blankLines="1" /></t>
<t hangText="block length: 16 bits"><vspace blankLines="0" /> The
constant 2, in accordance with the definition of this field in
Section 3 of <xref target="RFC3611">RFC 3611</xref>. <vspace
blankLines="1" /></t>
</list></t>
</section>
<section anchor="SMQM"
title="Synthetical Multimedia Quality Metrics Block">
<t>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 <xref
target="G.1082"></xref><xref target="P.NAMS"></xref><xref
target="P.NBAMS"></xref> 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. <vspace blankLines="1" /> 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. <vspace
blankLines="1" /> The Synthetical Multimedia Quality Metrics Block has
the following format: <figure>
<artwork>
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 |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
</artwork>
</figure> <list style="hanging">
<t hangText="Block type (BT): 8 bits"><vspace blankLines="0" />
The Perceptual Quality Metrics Block is identified by the constant
<SMQM>. <vspace blankLines="1" /></t>
<t hangText="Interval Metric flag (I): 1 bit "><vspace
blankLines="0" /> 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). <vspace
blankLines="1" /></t>
<t hangText="MoS Class (MC): 4 bits"><vspace blankLines="0" />This
field is used to indicate the MOS type to be reported. The MOS
type is defined as follows: <list>
<t>0000 MOS-A - Audio Quality MOS <xref
target="G.107"></xref><xref target="P.564"></xref>.</t>
<t>0001 MOS-V - Video Quality MOS <xref
target="P.NAMS"></xref><xref target="P.NBAMS"></xref>.</t>
<t>0010 MOS-AV - Audio-Video Quality MOS<xref
target="P.NAMS"></xref><xref target="P.NBAMS"></xref>.</t>
<t>0100~1111 - Reserved</t>
</list><vspace blankLines="1" /></t>
<t hangText="Rsd.: 7 bits "><vspace blankLines="0" /> 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. <vspace blankLines="1" /></t>
<t hangText="SSRC of source: 32 bits"><vspace blankLines="0" /> As
defined in Section 4.1 of <xref target="RFC3611"></xref>. <vspace
blankLines="1" /></t>
<t hangText="MOS Value: Variable Length"><vspace
blankLines="1" />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 <xref target="G.1082"></xref><xref
target="P.NAMS"></xref><xref target="P.NBAMS"></xref>. 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. <vspace blankLines="1" /></t>
</list></t>
</section>
</section>
<section title="SDP Signaling">
<t>Six new parameters are defined for the six report blocks defined in
this document to be used with Session Description Protocol (SDP) <xref
target="RFC4566"></xref> using the Augmented Backus-Naur Form (ABNF)
<xref target="RFC5234"></xref>. They have the following syntax within
the "rtcp-xr" attribute <xref target="RFC3611"></xref>: <figure
align="left">
<artwork>
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"
</artwork>
</figure> Refer to Section 5.1 of <xref target="RFC3611">RFC
3611</xref> for a detailed description and the full syntax of the
"rtcp-xr" attribute.</t>
</section>
<section title="IANA Considerations">
<t>New report block types for RTCP XR are subject to IANA registration.
For general guidelines on IANA allocations for RTCP XR, refer to <xref
target="RFC3611">Section 6.2 of</xref>. <vspace blankLines="1" /> This
document assigns six new block type values in the RTCP XR Block Type
Registry: <list>
<t><list hangIndent="12" style="hanging">
<t hangText="Name:">RFISD</t>
<t hangText="Long Name:">RTP Flows Initial Synchronization
Delay</t>
<t hangText="Value"><RFISD></t>
<t hangText="Reference:"><xref target="RFISD"></xref> <vspace
blankLines="1" /></t>
<t hangText="Name:">RFGSO</t>
<t hangText="Long Name:">RTP Flows General Synchronization
Offset Metrics Block</t>
<t hangText="Value"><RFGSO></t>
<t hangText="Reference:"><xref target="RFGSO"></xref> <vspace
blankLines="1" /></t>
<t hangText="Name:">TSSS</t>
<t hangText="Long Name:">Transport Stream Statistics Summary</t>
<t hangText="Value"><TSSS></t>
<t hangText="Reference:"><xref target="TSSS"></xref> <vspace
blankLines="1" /></t>
<t hangText="Name:">LSSM</t>
<t hangText="Value"><LSSM></t>
<t hangText="Long Name:">Layered Stream Statistics Metrics</t>
<t hangText="Reference:"><xref target="LSSM"></xref> <vspace
blankLines="1" /></t>
<t hangText="Name:">TSLDM</t>
<t hangText="Long Name:">Transport Stream Loss and Discard
Metrics</t>
<t hangText="Value"><TSLDM></t>
<t hangText="Reference:"><xref target="TSLDM"></xref> <vspace
blankLines="1" /></t>
<t hangText="Name:">TSBM</t>
<t hangText="Long Name:">Transport Stream Burst Metrics</t>
<t hangText="Value"><TSBM></t>
<t hangText="Reference:"><xref target="TSBM"></xref> <vspace
blankLines="1" /></t>
<t hangText="Name:">SMQM</t>
<t hangText="Long Name:">Synthetical Multimedia Quality
Metric</t>
<t hangText="Value"><SMQM></t>
<t hangText="Reference:"><xref target="SMQM"></xref> <vspace
blankLines="1" /></t>
</list></t>
</list> This document also registers seven SDP <xref
target="RFC4566"></xref> parameters for the "rtcp-xr" attribute in the
RTCP XR SDP Parameters Registry: <list>
<t><list style="symbols">
<t>"RTP-flows-init-syn"</t>
<t>“RTP-flows-general-syn”</t>
<t>“multimedia-quality-metrics”</t>
<t>“transport-stream-loss-metrics”</t>
<t>“transport-stream-burst-metrics”</t>
<t>“transport-stat-summary”</t>
<t>“layered-stream-stat-metrics”</t>
</list></t>
</list> <vspace blankLines="1" /> The contact information for the
registrations is: <figure align="center">
<artwork>
Qin Wu
sunseawq@huawei.com
101 Software Avenue, Yuhua District
Nanjing, JiangSu 210012 China
</artwork>
</figure></t>
</section>
<section title="Security Considerations">
<t>The new RTCP XR report blocks proposed in this document introduces no
new security considerations beyond those described in <xref
target="RFC3611"></xref>.</t>
</section>
<section title="Acknowledgements">
<t>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.</t>
</section>
</middle>
<back>
<references title="Normative References">
<reference anchor="RFC6051">
<front>
<title>Rapid Synchronisation of RTP Flows</title>
<author fullname="Colin Perkins" initials="C." surname="Perkins">
<organization></organization>
</author>
<author fullname="Thomas Schierl" initials="T" surname="Schierl">
<organization></organization>
</author>
<date day="10" month="November" year="2010" />
<abstract>
<t>This memo outlines how RTP sessions are synchronised, and
discusses how rapidly such synchronisation can occur. We show that
most RTP sessions can be synchronised immediately, but that the
use of video switching multipoint conference units (MCUs) or large
source specific multicast (SSM) groups can greatly increase the
synchronisation delay. This increase in delay can be unacceptable
to some applications that use layered and/or multi-description
codecs. This memo introduces three mechanisms to reduce the
synchronisation delay for such sessions. First, it updates the RTP
Control Protocol (RTCP) timing rules to reduce the initial
synchronisation delay for SSM sessions. Second, a new feedback
packet is defined for use with the Extended RTP Profile for
RTCP-based Feedback (RTP/AVPF), allowing video switching MCUs to
rapidly request resynchronisation. Finally, new RTP header
extensions are defined to allow rapid synchronisation of late
joiners, and guarantee correct timestamp based decoding order
recovery for layered codecs in the presence of clock skew.</t>
</abstract>
</front>
<seriesInfo name="RFC" value="6051" />
<format target="http://tools.ietf.org/html/rfc6051" type="TXT" />
</reference>
&I-D.ietf-avt-rtp-svc;
&rfc2250;
&rfc3611;
&rfc2119;
&rfc4566;
&rfc5234;
&rfc3550;
<reference anchor="ISO-IEC.13818-1.2007">
<front>
<title>Information technology - Generic coding of moving pictures
and associated audio information: Systems</title>
<author>
<organization>International Organization for
Standardization</organization>
</author>
<date month="October" year="2007" />
</front>
<seriesInfo name="ISO" value="International Standard 13818-1" />
</reference>
</references>
<references title="Informative References">
&rfc3357;
<reference anchor="DSLF">
<front>
<title>Triple-play Services Quality of Experience (QoE)
Requirements</title>
<author fullname="Tim Rahrer" initials="T.R." role="editor"
surname="Rahrer">
<organization>Nortel</organization>
<address>
<postal>
<street>3500 Carling Ave.</street>
<city>Ottawa</city>
<region>Ontario</region>
<country>Canada</country>
</postal>
<phone>+1.613.763.4479</phone>
<email>tim.rahrer@nortel.com</email>
</address>
</author>
<author fullname="Riccardo Fiandra" role="editor" surname="Fiandra">
<organization>FastWeb, SpA</organization>
<address>
<email>riccardo.fiandra@fastweb.it</email>
</address>
</author>
<author fullname="Steven Wright" role="editor" surname="Wright">
<organization>BellSouth Telecommunications</organization>
<address>
<email>steven.wright@bellsouth.com</email>
</address>
</author>
<date day="13" month="December" year="2006" />
</front>
<seriesInfo name="DSL Forum Technical Report" value="TR-126" />
</reference>
<reference anchor="G.1082">
<front>
<title>Measurement-based methods for improving the robustness of
IPTV performance</title>
<author>
<organization>ITU-T</organization>
</author>
<date month="April" year="2009" />
</front>
<seriesInfo name="ITU-T Recommendation" value="G.1082" />
</reference>
<reference anchor="P.564">
<front>
<title>Conformance testing for narrowband Voice over IP transmission
quality assessment models</title>
<author>
<organization>ITU-T</organization>
</author>
<date month="July" year="2006" />
</front>
<seriesInfo name="ITU-T Recommendation" value="P.564" />
</reference>
<reference anchor="G.107">
<front>
<title>The E Model, a computational model for use in transmission
planning</title>
<author>
<organization>ITU-T</organization>
</author>
<date month="April" year="2009" />
</front>
<seriesInfo name="ITU-T Recommendation" value="G.107" />
</reference>
<reference anchor="P.NAMS">
<front>
<title>Non-intrusive parametric model for the Assessment of
performance of Multimedia Streaming</title>
<author>
<organization>ITU-T</organization>
</author>
<date month="November" year="2009" />
</front>
<seriesInfo name="ITU-T Recommendation" value="P.NAMS" />
</reference>
<reference anchor="P.NBAMS">
<front>
<title>non-intrusive bit-stream model for assessment of performance
of multimedia streaming</title>
<author>
<organization>ITU-T</organization>
</author>
<date month="November" year="2009" />
</front>
<seriesInfo name="ITU-T Recommendation" value="P.NBAMS" />
</reference>
<reference anchor="I-D.ietf-pmol-metrics-framework-02">
<front>
<title>Framework for Performance Metric Development</title>
<author fullname="Alan Clark " initials="A." surname="Clark">
<organization>Telchemy Incorporated</organization>
</author>
<date />
</front>
</reference>
<reference anchor="IEEE">
<front>
<title>Human Perception of Jitter and Media Synchronization</title>
<author>
<organization>IEEE</organization>
</author>
<date month="January" year="1996" />
</front>
<seriesInfo name="IEEE Journal on Selected Areas in Communications"
value="Vol. 14, No. 1" />
</reference>
</references>
</back>
</rfc>
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