TOC 
Network Working GroupQ. Wu
Internet-DraftHuawei
Intended status: Standards TrackG. Zorn
Expires: January 10, 2011Network Zen
 July 9, 2010


RTP Control Protocol Extended Reports (RTCP XR) Report Blocks for Real-time Video Quality Monitoring
draft-wu-avt-rtcp-xr-quality-monitoring-01

Abstract

This document defines a set of RTP Control Protocol Extended Reports (RTCP XR) Report Blocks and associated SDP parameters allowing the report of 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 January 10, 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 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
2.  Terminology
    2.1.  Standards Language
    2.2.  Acronyms
3.  Applicability
4.  RTP Flows Synchronization Delay Report Block
5.  Audio-Video Synchronization Report Block
6.  Video Statistics Summary Report Block
7.  TR 101 290 Decodability Metrics Report Block
8.  Video Stream Metrics Report Block
    8.1.  Packet Loss and Discard Metrics
    8.2.  Burst Metrics
    8.3.  Delay Metrics
    8.4.  Configuration Parameter Metrics
    8.5.  Jitter Buffer Parameters
9.  Perceptual Quality Metrics Report Block
10.  SDP Signaling
11.  IANA Considerations
12.  Security Considerations
13.  Acknowledgements
14.  References
    14.1.  Normative References
    14.2.  Informative References
§  Authors' Addresses




 TOC 

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 network-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 its 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 network-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, conveying video quality metrics in RTCP XR packets [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) has the following three benefits:



RFC 3611 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611] defines seven report block formats for network 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 synchronization information on received RTP packets. The report blocks in the second category convey metrics relating to packet receipts defined in RFC 3611 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611] that are summary in nature but include data that is more detailed, or of a different type, than that conveyed in existing RTCP packets.

Six report block formats are defined by this document. Of these, two are synchronization information blocks:



The other four are summary metrics blocks:



 TOC 

2.  Terminology



 TOC 

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 (Bradner, S., “Key words for use in RFCs to Indicate Requirement Levels,” March 1997.) [RFC2119].



 TOC 

2.2.  Acronyms

SSRC
Synchronization Source (Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” July 2003.) [RFC3550]

TS
Transport Stream (International Organization for Standardization, “Information technology - Generic coding of moving pictures and associated audio information: Systems,” October 2007.) [ISO‑IEC.13818‑1.2007]



 TOC 

3.  Applicability

All the report blocks defined in this document could be used by dedicated network monitoring applications. As specified in RFC 3611 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [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 5 (Audio-Video Synchronization Report Block) 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 Stream Metrics Report Block and Statistics Summary Report Block can be applied to any real time video application, while the TR 101 290 Decodability Metrics Report Block and Perceptual Quality Metrics Report Block can be used in any real-time AV application [ETSI] (ETSI, “Digital Video Broadcasting (DVB); Measurement guidelines for DVB systems,” 2001.).



 TOC 

4.  RTP Flows Synchronization Delay Report Block

This block reports synchronization delay between RTP sessions of the same video stream sent using Multi-Session Transmission [I‑D.ietf‑avt‑rtp‑svc] (Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, “RTP Payload Format for SVC Video,” April 2010.) beyond the information carried in the standard RTCP packet format. Information is recorded about session bandwidth and synchronization delay.

The RTP Flows Synchronization Delay Report Block has the following format:

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|     BT=TBD    |   Reserved    |          Block length         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                      SSRC of Sender                           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                     Session Bandwidth                         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|               Initial Synchronization Delay                   |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Block type (BT): 8 bits
The Statistics Summary Report Block is identified by the constant <RFSD>.

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 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [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 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611]).

Session Bandwidth: 32 bits
This field indicates the maximum bandwidth allocated to each RTP Session. The value in this field represents the bandwidth allocation per RTP Session in kilobits per second, with values in the range 0 <= BW < 65536.

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] (Perkins, C. and T. Schierl, “Rapid Synchronisation of RTP Flows,” May 2010.). The value is calculated as follows:

Initial Synchronization Delay = max((the time interval between receiving the first RTP packet with synchronization metadata and the start of a session), (the time interval between receiving the first RTCP packet in the RTP session with the longest RTCP reporting interval and the start of a session)).



 TOC 

5.  Audio-Video Synchronization Report Block

This block reports the audio-video synchronization requirements between audio and video components beyond the information carried in the standard RTCP packet format. Information is recorded about tolerant audio lead video time and tolerant audio lag video time. The Audio-Video Synchronization 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|  Reserved   |         Block length          |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                        SSRC of source                         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|        Audio Lead Video       |       Audio Lag Video         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Block type (BT): 8 bits
The Statistics Summary Report Block is identified by the constant <AVS>.

Audio Lead/Lag Video flag (L): 1 bit
Bit set to 1 if Audio Lead Video field contains a report, 0 if Audio Lag Video field contains a report.

Reserved: 7 bits
This field is reserved for future definition. In the absence of such a definition, the bits in this field MUST be set to zero and MUST be ignored by the receiver.

Block length: 16 bits
The constant 2, in accordance with the definition of this field in Section 3 of RFC 3611 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611].

SSRC of source: 32 bits
As defined in Section 4.1 of RFC 3611 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611].

Audio Lead Video time: 16 bits
This field indicates the tolerant audio lead video time in milliseconds [DSLF] (Rahrer, T., Ed., Fiandra, Ed., and Wright, Ed., “Triple-play Services Quality of Experience (QoE) Requirements,” December 2006.). This value is calculated based on the interarrival time between previous video RTP packet and the next first audio RTP packet and timestamps of both previous video RTP packet and the next first audio packet.

Audio Lag Video time: 16 bits
This field indicates the tolerant audio lag video time in milliseconds [DSLF] (Rahrer, T., Ed., Fiandra, Ed., and Wright, Ed., “Triple-play Services Quality of Experience (QoE) Requirements,” December 2006.). This value is calculated based on the interarrival time between previous video RTP packet and the next first audio RTP packet and timestamps of both previous video RTP packet and the next first audio packet.



 TOC 

6.  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 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611]. Information is recorded about lost frame packets, duplicated frame packets, lost layered component packets, duplicated layered component packets. Such information can be useful for network management and video quality monitoring.

The report block contents are dependent upon a series of flag bits carried in the first part of the header. Not all parameters need to be reported in each block. Flags indicate which parameters are reported and which are not. The fields corresponding to unreported parameters MUST be present, but are set to zero. The receiver MUST ignore any Video Statistics Summary Report Block with a non-zero value in any field flagged as unreported.

The 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 |  LT |P|        block length           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                        SSRC of source                         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|          begin_seq            |             end_seq           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                         lost_frames                           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                          dup frames                           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                 lost_layered component packets                |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                  dup layered component_packets                |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                    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.

Layer Type indicator (LT): 3 bits
This field is used to indicate the layer type of layered video to be reported. LT is set to 001 if the loss_component_packet field and dup_component packet contain the base layer packet in SVC [I‑D.ietf‑avt‑rtp‑svc] (Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, “RTP Payload Format for SVC Video,” April 2010.), 010 if the loss_component packet field and dup_component packet contain enhancement layer 1 packet in SVC, 011 if the loss_component packet field and dup_component packet contain the enhancement layer 2 packet, 000 otherwise.

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.

Block length: 16 bits
The constant 8, in accordance with the definition of this field in Section 3 of RFC 3611 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611].

SSRC of source: 32 bits
As defined in Section 4.1 of RFC 3611 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611].

begin_seq: 16 bits
As defined in Section 4.1 of RFC 3611 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611].

end_seq: 16 bits
As defined in Section 4.1 of RFC 3611 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [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_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.

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.



 TOC 

7.  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] (ETSI, “Digital Video Broadcasting (DVB); Measurement guidelines for DVB systems,” 2001.). Such information can be useful for network management and video quality monitoring.

Note that this is only applicable to MPEG-2 RTP streams [RFC2250] (Hoffman, D., Fernando, G., Goyal, V., and M. Civanlar, “RTP Payload Format for MPEG1/MPEG2 Video,” January 1998.). 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>.

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 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611].

SSRC of source: 32 bits
As defined in Section 4.1 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).

begin_seq: 16 bits
As defined in Section 4.1 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).

end_seq: 16 bits
As defined in Section 4.1 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).

Number of Packets: 16 bits
Number of RTP packets in the above sequence number interval.

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.



 TOC 

8.  Video Stream Metrics Report Block

The Video Stream Metrics Report Block provides metrics for monitoring the quality of video stream. This metrics includes Loss and discard metrics, Burst metrics, Delay metrics for I-Frame packets, B-Frame packets and P-Frame packets, Configuration parameter metrics. 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 also 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 (Koodli, R. and R. Ravikanth, “One-way Loss Pattern Sample Metrics,” August 2002.) [RFC3357]. Following the one-way loss pattern sample metrics discussed in [RFC3357] (Koodli, R. and R. Ravikanth, “One-way Loss Pattern Sample Metrics,” August 2002.), 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:

 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 |          Loss Period          |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|         Loss Distance         |        Max Loss Duration      |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|     Round trip delay          |       End system delay        |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|     Gmin      |   RX config   |          JB nominal           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|          JB maximum           |          JB abs max           |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

block type (BT): 8 bits
A Video Stream Metrics Report Block is identified by the constant <VSM>.

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 8, in accordance with the definition of this field in Section 3 of RFC 3611 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [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 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611].

The remaining fields are described in the following four sections: Packet Loss and Discard Metrics, Delay Metrics, Configuration Metrics, and Jitter Buffer Parameters.



 TOC 

8.1.  Packet Loss and Discard Metrics

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.

Loss rate for I-Frame packets: 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 I frame packets lost (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.

Discard rate for I-Frame packets: 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 I-Frame packets discarded (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.

Loss rate for P-Frame packets: 8 bits
The loss rate for P-Frame packets is similar to the loss rate for I-Frame packets. The value is calculated by dividing the total number of P frame packets lost (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.

Discard rate for P-Frame packets: 8 bits
The discard rate for P-Frame packets is similar to the discard rate for I-Frame packets. The value is calculated by dividing the total number of P-Frame packets discarded (excluding duplicate packet discards) by the total number of packets expected, multiplying the result of the division by 256.

Loss rate for B-Frame packets: 8 bits
The loss rate for B-Frame packets is similar to the loss rate for P-Frame packets. The value is calculated by dividing the total number of B frame packets lost (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.

Discard rate for P-Frame packets: 8 bits
The discard rate for B-Frame packets is similar to the discard rate for P-Frame packets. The value is calculated by dividing the total number of B-Frame packet discarded (excluding duplicate packet discards) by the total number of packets expected, multiplying the result of the division by 256.



 TOC 

8.2.  Burst Metrics

Loss Distance: 16 bits
The mean duration, expressed in milliseconds, of the loss intervals that have occurred since the beginning of reception [DSLF] (Rahrer, T., Ed., Fiandra, Ed., and Wright, Ed., “Triple-play Services Quality of Experience (QoE) Requirements,” December 2006.). 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] (Rahrer, T., Ed., Fiandra, Ed., and Wright, Ed., “Triple-play Services Quality of Experience (QoE) Requirements,” December 2006.). 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.

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] (Rahrer, T., Ed., Fiandra, Ed., and Wright, Ed., “Triple-play Services Quality of Experience (QoE) Requirements,” December 2006.), 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.



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8.3.  Delay Metrics

Similar to the delay metrics for audio stream defined in the section 4.7.3 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.), the delay metrics for video stream fall into two categories: end-system delay due to RTP payload multiplexing and round trip delay due to multiplexing RTP frames within a UDP frame.

Round trip delay: 16 bits
As defined in Section 4.7.3 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).

End system delay: 16 bits
As defined in Section 4.7.3 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).



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8.4.  Configuration Parameter Metrics

Similar to the configuration metrics defined in the section 4.7.6 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.), the configuration metrics for video stream include Gmin, receiver configuration byte, packet loss concealment, jitter buffer adaptive, and jitter buffer rate.

Gmin: 8 bits
As defined in Section 4.7.6 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).

Receiver configuration byte (RX config): 8 bits
As defined in Section 4.7.6 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).

Jitter buffer adaptive (JBA): 2 bits
As defined in Section 4.7.6 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).

Jitter buffer rate (JB rate): 4 bits
As defined in Section 4.7.6 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).



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8.5.  Jitter Buffer Parameters

Similar to the jitter buffer parameters defined in the section 4.7.7 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.), the Jitter Buffer Parameters for video streams include jitter buffer nominal delay, jitter buffer maximum delay and jitter buffer absolute maximum delay.

Jitter buffer nominal delay (JB nominal): 16 bits
As defined in Section 4.7.7 of[RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).

Jitter buffer maximum delay (JB maximum): 16 bits
As defined in Section 4.7.7 of[RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).

Jitter buffer absolute maximum delay (JB abs max): 16 bits
As defined in Section 4.7.7 of[RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).



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9.  Perceptual Quality Metrics Report Block

This block reports perceptual quality metrics beyond the information carried in the standard RTCP packet format. Information is recorded about Video MOS, Audio Video MOS, Video Service Transmission Quality, Video Service Audio Quality, Video Service Multimedia Quality and Video Service Picture Quality.

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 Perceptual Quality Metrics Block has the following format:

 0                   1                   2                   3
 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|     BT=TBD    |I|S|T|A|M|P|Rsd|         block length          |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|                        SSRC of source                         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|              MOS-V            |           MOS-AV              |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|              VSTQ             |           VSAQ                |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|              VSMQ             |           VSPQ                |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|        Loss Period VSPQ       |    Loss Distance VSPQ         |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

Block type (BT): 8 bits
The Perceptual Quality Metrics Block is identified by the constant <PQM>.

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 (S): 1 bit
Bit set to 1 if the MOS-V field and 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.

Video Service Audio Quality flag (A): 1 bit
Bit set to 1 if the VSAQ field contains a report, 0 otherwise.

Video Service Multimedia Quality flag (M): 1 bit
Bit set to 1 if the VSMQ field contains a report, 0 otherwise.

Video Service Picture Quality flag (P): 1 bit
Bit set to 1 if the VSPQ field, Loss Period VSPQ, Loss Distance VSPQ contains a report, 0 otherwise.

Rsd.: 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.

SSRC of source: 32 bits
As defined in Section 4.1 of [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.).

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. 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-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. 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)

VSAQ: 16 bits
Video Service Audio Quality (TBC)

VSMQ: 16 bits
Video Service Multimedia Quality (TBC)

VSPQ: 16 bits
Video Service Picture Quality (TBC)

Loss Period VSPQ: 16 bits
Video Service Picture Quality during Loss Period (TBC)

Loss Distance VSPQ: 16 bits
Video Service Picture Quality during Loss Distance (TBC)



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10.  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] (Handley, M., Jacobson, V., and C. Perkins, “SDP: Session Description Protocol,” July 2006.) using the Augmented Backus-Naur Form (ABNF) [RFC5234] (Crocker, D. and P. Overell, “Augmented BNF for Syntax Specifications: ABNF,” January 2008.). They have the following syntax within the "rtcp-xr" attribute [RFC3611] (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.):

rtcp-xr-attrib =  "a=rtcp-xr:"
                  [xr-format *(SP xr-format)] CRLF

   xr-format = RTP-flows-syn
               / audio-video-syn
               / percept-quality-metrics
               / video-stream-metric
               / video-stat-summary
               / decodability-metric

      RTP-flows-syn = "RTP-flows-syn"
                      ["=" max-size]
         max-size = 1*DIGIT ; maximum block size in octets

      audio-video-syn = "audio-video-syn"
                        ["=" max-size]
         max-size = 1*DIGIT ; maximum block size in octets

      percept-quality-metrics = "percept-quality-metrics"
                                ["=" max-size]
         max-size = 1*DIGIT ; maximum block size in octets

      video-stream-metric = "video-stream-metric"
                            ["=" 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"

      decodability-metric = "decodability-metric"
                            ["=" stat-flag *("," stat-flag)]
         stat-flag = "Interval Metric"
                     / "MOS-V and MOS-AV"
                     / "VSTQ"
                     / "VSAQ"
                     / "VSMQ"
                     / "VSPQ"

Refer to Section 5.1 of RFC 3611 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611] for a detailed description and the full syntax of the "rtcp-xr" attribute.



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11.  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 (Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” November 2003.) [RFC3611].

This document assigns six new block type values in the RTCP XR Block Type Registry:

Name:
RFSD
Long Name:
RTP Flows Synchronization Delay
Value
<RFSD>
Reference:
Section 4 (RTP Flows Synchronization Delay Report Block)
Name:
AVS
Long Name:
Audio-Video Synchronization
Value
<AVS>
Reference:
Section 5 (Audio-Video Synchronization Report Block)
Name:
VSS
Long Name:
Video Statistics Summary
Value
<VSS>
Reference:
Section 6 (Video Statistics Summary Report Block)
Name:
TDM
Long Name:
TR 101 290 Decodability Metrics
Value
<TDM>
Reference:
Section 7 (TR 101 290 Decodability Metrics Report Block)
Name:
VSM
Long Name:
Video Stream Metrics
Value
<VSM>
Reference:
Section 8 (Video Stream Metrics Report Block)
Name:
PQM
Long Name:
Perceptual Quality Metric
Value
<PQM>
Reference:
Section 9 (Perceptual Quality Metrics Report Block)

This document also registers six SDP [RFC4566] (Handley, M., Jacobson, V., and C. Perkins, “SDP: Session Description Protocol,” July 2006.) parameters for the "rtcp-xr" attribute in the RTCP XR SDP Parameters Registry:



The contact information for the registrations is:

Qin Wu
sunseawq@huawei.com
101 Software Avenue, Yuhua District
Nanjing, JiangSu 210012 China



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12.  Security Considerations

TBC



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13.  Acknowledgements

The authors would like to thank Youqing Yang, Wenxiao Yu and Yinliang Hu for their valuable comments and suggestions on this document.



 TOC 

14.  References



 TOC 

14.1. 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-11 (work in progress), May 2010 (TXT).
[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-21 (work in progress), April 2010 (TXT).
[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 (TXT, HTML, XML).
[RFC2250] Hoffman, D., Fernando, G., Goyal, V., and M. Civanlar, “RTP Payload Format for MPEG1/MPEG2 Video,” RFC 2250, January 1998 (TXT, HTML, XML).
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, “RTP: A Transport Protocol for Real-Time Applications,” STD 64, RFC 3550, July 2003 (TXT, PS, PDF).
[RFC3611] Friedman, T., Caceres, R., and A. Clark, “RTP Control Protocol Extended Reports (RTCP XR),” RFC 3611, November 2003 (TXT).
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, “SDP: Session Description Protocol,” RFC 4566, July 2006 (TXT).
[RFC5234] Crocker, D. and P. Overell, “Augmented BNF for Syntax Specifications: ABNF,” STD 68, RFC 5234, January 2008 (TXT).


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14.2. 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.
[RFC3357] Koodli, R. and R. Ravikanth, “One-way Loss Pattern Sample Metrics,” RFC 3357, August 2002 (TXT).


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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