Network Working Group Q. Wu Internet-Draft Huawei Intended status: Standards Track G. Zorn Expires: September 3, 2011 Network Zen R. Schott Deutsche Telekom Laboratories March 2, 2011 RTP Control Protocol Extended Reports (RTCP XR) Report Blocks for Real- time Application Quality Monitoring draft-wu-xrblock-rtcp-xr-quality-monitoring-01 Abstract This document defines a set of RTP Control Protocol Extended Reports (RTCP XR) Report Blocks and associated SDP parameters allowing the report of real time application quality metrics, primarily for video applications of RTP. Status of this Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute 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 September 3, 2011. Copyright Notice Copyright (c) 2011 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 Wu, et al. Expires September 3, 2011 [Page 1] Internet-Draft RTCP XR Quality Report Blocks March 2011 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. This document may contain material from IETF Documents or IETF Contributions published or made publicly available before November 10, 2008. The person(s) controlling the copyright in some of this material may not have granted the IETF Trust the right to allow modifications of such material outside the IETF Standards Process. Without obtaining an adequate license from the person(s) controlling the copyright in such materials, this document may not be modified outside the IETF Standards Process, and derivative works of it may not be created outside the IETF Standards Process, except to format it for publication as an RFC or to translate it into languages other than English. Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . 4 2.1. Standards Language . . . . . . . . . . . . . . . . . . . . 4 2.2. Acronyms . . . . . . . . . . . . . . . . . . . . . . . . . 5 3. Applicability . . . . . . . . . . . . . . . . . . . . . . . . 5 4. Transport Layer Metrics . . . . . . . . . . . . . . . . . . . 6 4.1. RTP Flows Initial Synchronization Delay Report Block . . . 6 4.2. RTP Flows General Synchronization Offset Metrics Block . . 7 4.3. Layered Streams Statistics Metrics Block . . . . . . . . . 8 5. Application Layer Metrics . . . . . . . . . . . . . . . . . . 10 5.1. Transport Streams Statistics Summary Report Block . . . . 10 5.2. Transport Stream Loss and Discard Metrics Block . . . . . 12 5.3. Transport Stream Burst Metrics Block . . . . . . . . . . . 14 5.4. Synthetical Multimedia Quality Metrics Block . . . . . . . 16 6. SDP Signaling . . . . . . . . . . . . . . . . . . . . . . . . 18 7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 20 8. Security Considerations . . . . . . . . . . . . . . . . . . . 21 9. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 21 10. References . . . . . . . . . . . . . . . . . . . . . . . . . . 21 10.1. Normative References . . . . . . . . . . . . . . . . . . . 21 10.2. Informative References . . . . . . . . . . . . . . . . . . 22 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 23 Wu, et al. Expires September 3, 2011 [Page 2] Internet-Draft RTCP XR Quality Report Blocks March 2011 1. Introduction Along with the wide deployment of broadband access and the development of new IPTV services (e.g., broadcast video, video on demand), there is increasing interest in monitoring and managing networks and applications that deliver real-time applications over RTP or IP, to ensure that all end users obtain acceptable video/audio quality. The main drives come from operators and enterprises, since offering performance monitoring capability can help diagnose network impairments, facilitate in root cause analysis and aid in verifying compliance with service level agreements (SLAs) between Internet Service Providers (ISPs) and content providers. The factors that affect real-time application quality can be split into two categories. The first category consists of transport- dependent factors such as packet loss, delay and jitter (which also translates into losses in the playback buffer). The factors in the second category are application-specific factors that affect real time application (e.g., video) quality and are sensitivity to network errors. These factors can be but not limited to video codec and loss recovery technique, coding bit rate, packetization scheme, and content characteristics. Compared with application-specific factors, the transport-dependent factors sometimes are not sufficient to measure real time data quality, since the ability to analyze the real time data in the application layer provides quantifiable measurements for subscriber Quality of Experience (QoE) that may not be captured in the transmission layers or from the RTP layer down. In a typical scenario, monitoring of the transmission layers can produce statistics suggesting that quality is not an issue, such as the fact that network jitter is not excessive. However, problems may occur in the service layers leading to poor subscriber QoE. Therefore monitoring using only network-level measurements may be insufficient when application layer video quality is required. In order to provide accurate measures of real time application quality for operators when transporting real time contents across a network, the synthentical multimedia quality Metrics is highly required which can be conveyed in the RTCP XR packets[RFC3611] and may have the following three benefits: * Tuning the content encoder algorithm to satisfy real time data quality requirements * Determining which system techniques to use in a given situation and when to switch from one technique to another as system parameters change Wu, et al. Expires September 3, 2011 [Page 3] Internet-Draft RTCP XR Quality Report Blocks March 2011 * Verifying the continued correct operation of an existing system RFC 3611 [RFC3611] defines seven report block formats for network management and quality monitoring. However, some of these metrics are mostly for multicast inference of network characteristics (MINC) or voice over IP (VoIP) monitoring and not widely applicable to other applications, e.g., video quality monitoring. This document focuses on specifying new additional report block types used to convey QoE related parameters that is genericly designed for use in voice, audio and video services. The report block types defined in this document fall into two categories. The first category consists of general information regarding transmission quality, to be generated and processed by the RTP transport. The report blocks in the second category convey metrics above transport that affect real time application quality and are sensitivity to network errors. Seven report block formats are defined by this document. Of these, three are transport layer metrics: * RTP Flows Initial Synchronization Delay Report Block * RTP Flows General Synchronization Offset Metrics Block * Layered Streams Statistics Metrics Block The other four are application layer metrics: * Transport Stream Statistics Summary Report Block * Transport Stream Loss and Discard Metrics Block * Transport Stream Burst Metrics Block * Synthetical Multimedia Quality Metrics Block 2. Terminology 2.1. Standards Language The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [RFC2119]. In addition, the following terms are defined: Layered Component Packet a RTP packet using layered codecs containing the specified layered component, e.g., encoded stream at the base layer or at the enhancement layer. Wu, et al. Expires September 3, 2011 [Page 4] Internet-Draft RTCP XR Quality Report Blocks March 2011 Picture Type Picture types used in the different video algorithms compose of the key-frame and the Derivation frame. Key-frame is also called a reference frame and used as a reference for predicting other pictures. It is coded without prediction from other pictures. The Derivation frame is derived from Key-frame using prediction from the reference frame. 2.2. Acronyms SSRC Synchronization Source [RFC3550] TS Transport Stream [ISO-IEC.13818-1.2007] 3. Applicability All the report blocks defined in this document could be used by dedicated network monitoring applications. As specified in RFC 3611 [RFC3611], for such an application it might be appropriate to allow more than 5% of RTP data bandwidth to be used for RTCP packets, thus allowing proportionately larger and more detailed report blocks. RTP Flows General Synchronization Offset Metrics Block in Section 4.2 has been defined for various multimedia applications. Such applications can use this report block to monitor offset between two RTP streams synchronization to ensure satisfactory QoE. Tighter tolerances than typically used have been recommended for such applications. The RTP Flows Initial Synchronization Delay Report Block has been defined primarily for layered or multi-description video coding applications. When joining a layered video session in such an application, a receiver may not synchronize playout across the multimedia session until RTCP SR packets have been received on all of the component RTP sessions. This report block can be used to measure synchronization between different media layers for the same multimedia session. The Transport Stream Loss and Discard Metrics Report Block, Transport Stream Burst Metrics report Block, Transport Statistics Summary Report Block and Layered Streams Statistics Metrics Block can be applied to any real time video application, while Synthetical Multimedia Quality Metrics Report Block can be used in any real-time Wu, et al. Expires September 3, 2011 [Page 5] Internet-Draft RTCP XR Quality Report Blocks March 2011 AV application. 4. Transport Layer Metrics 4.1. RTP Flows Initial Synchronization Delay Report Block This block reports Initial synchronization delay beyond the information carried in the standard RTCP packet format. Information is recorded about the the difference between the start of RTP sessions and the time the RTP receiver acquires all components of RTP sessions [RFC6051]. The components of RTP session are referred to as one RTP session for each media type or the media content in each layer contained in RTP Control Protocol (RTCP) sender report (SR) packets [RFC3550]. For unicast session, the delay due to negotiation of NAT pinholes, firewall holes, quality-of-service, and media security keys is contributed to such initial synchronization delay. For multicast session, the initial synchronization delay varies with the session bandwidth and the number of members, the number of senders in the session. In the absence of packet loss, the initial synchronisation delay equals to the average time taken to receive the first RTCP packet in the RTP session with the longest RTCP reporting interval.In the presence of packet loss, the media synchronization needs to wait until the reporting interval has passed, and the next RTCP SR packet is sent. The RTP Flows Initial Synchronization Delay Report Block has the following format: 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | BT=TBD | Reserved | Block length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of Sender | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Initial Synchronization Delay | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Block type (BT): 8 bits The Statistics Summary Report Block is identified by the constant . 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. Wu, et al. Expires September 3, 2011 [Page 6] Internet-Draft RTCP XR Quality Report Blocks March 2011 Block length: 16 bits The constant 3, in accordance with the definition of this field in Section 3 of RFC 3611 [RFC3611]. SSRC of Sender: 32 bits The SSRC of the RTP data packet source being reported upon by this report block. (Section 4.1 of [RFC3611]). Initial Synchronization Delay: 32 bits The average delay, expressed in units of 1/65536 seconds, between the RTCP packets received on all of the components RTP sessions and the beginning of session [RFC6051]. The value is calculated as follows: The average time, expressed in units of 1/65536 seconds, taken to receive the first RTCP packet in the RTP session with the longest RTCP reporting interval [RFC6051] 4.2. RTP Flows General Synchronization Offset Metrics Block In an RTP multimedia session, there can be an arbitrary number of streams, with the same RTCP CNAME. This block reports the general Synchronization offset status of these RTP streams beyond the information carried in the standard RTCP packet format. Information is recorded about the synchronization offset time of each RTP stream relative to the reference RTP stream with the same CNAME and General Synchronisation Offset of zero. For layered session or multimedia session,the first RTP packet can be chosen as the basic packet of reference RTP stream. The RTP Flow General Synchronization Offset Report Block has the following format: 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | BT=TBD |I| Reserved | Block length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | General Synchronization Offset | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Block type (BT): 8 bits The Statistics Summary Report Block is identified by the constant . Wu, et al. Expires September 3, 2011 [Page 7] Internet-Draft RTCP XR Quality Report Blocks March 2011 Interval Metric flag (I): 1 bit This field is used to indicate whether the Audio-Video synchronization metrics are Interval or Cumulative metrics, that is, whether the reported values applies to the most recent measurement interval duration between successive metrics reports (I=1) (the Interval Duration) or to the accumulation period characteristic of cumulative measurements (I=0) (the Cumulative Duration). Reserved: 8 bits This field is reserved for future definition. In the absence of such a definition, the bits in this field MUST be set to zero and MUST be ignored by the receiver. Block length: 16 bits The constant 2, in accordance with the definition of this field in Section 3 of RFC 3611 [RFC3611]. SSRC of source: 32 bits As defined in Section 4.1 of RFC 3611 [RFC3611]. General synchronization offset: 32 bits This field represents the synchronization offset time of one RTP stream in milliseconds relative to the reference RTP stream with the same CNAME and General Synchronisation Offset of zero [RFC6051] This value is calculated based on the interarrival time between arbitray RTP packet and the reference RTP packet with the same CNAME , and timestamps of this arbitray RTP packet and the reference RTP packet with the same CNAME. 4.3. Layered Streams Statistics Metrics Block This block reports layered streams statistics beyond the information carried in the Statistics Summary Report Block RTCP packet specified in the section 4.6 of RFC 3611 [RFC3611]. Information is recorded about lost layered component packets, duplicated layered component packets. Such information can be useful for network management and video quality monitoring. The report block contents are dependent upon a series of flag bits carried in the first part of the header. Not all parameters need to be reported in each block. Flags indicate which parameters are reported and which are not. The fields corresponding to unreported parameters MUST be present, but are set to zero. The receiver MUST ignore any Layered Streams Statistics Metrics Block with a non-zero value in any field flagged as unreported. Wu, et al. Expires September 3, 2011 [Page 8] Internet-Draft RTCP XR Quality Report Blocks March 2011 The Layered Stream Statistics metrics Block has the following format: 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | BT=TBD |T| rsd. | block length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | begin_seq | end_seq | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Lost_Layered Component Packets | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Dup Layered Component_Packets | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Block type (BT): 8 bits The Layered stream Statistics Metrics Block is identified by the constant . Layer Type flag (T): 1 bits This field is used to indicate the Layer Type of layered video to be reported. LT is set to 0 if the loss_component_packet field and dup_component packet contain the base layer packet in layered codecs,e.g, SVC in [I-D.ietf-avt-rtp-svc], 1 if the loss_component packet field and dup_component packet contain enhancement layer packet in layered codec. Rsd.: 3 bits This field is reserved for future definition. In the absence of such a definition, the bits in this field MUST be set to zero and MUST be ignored by the receiver. Block length: 16 bits The constant 3, in accordance with the definition of this field in Section 3 of RFC 3611 [RFC3611]. SSRC of source: 32 bits As defined in Section 4.1 of RFC 3611 [RFC3611]. begin_seq: 16 bits As defined in Section 4.1 of RFC 3611 [RFC3611]. end_seq: 16 bits As defined in Section 4.1 of RFC 3611 [RFC3611]. Wu, et al. Expires September 3, 2011 [Page 9] Internet-Draft RTCP XR Quality Report Blocks March 2011 Lost_Layered Component Packets: 32 bits Number of lost_component packets in the above sequence number interval. Dup_Layered Component Packets: 32 bits Number of dup_component packets in the above sequence number interval. 5. Application Layer Metrics 5.1. Transport Streams Statistics Summary Report Block This block reports statistics beyond the information carried in the Statistics Summary Report Block RTCP packet specified in the section 4.6 of RFC 3611 [RFC3611]. Information is recorded about lost frame packets, duplicated frame packets, lost layered component packets, duplicated layered component packets. Such information can be useful for network management and video quality monitoring. The report block contents are dependent upon a series of flag bits carried in the first part of the header. Not all parameters need to be reported in each block. Flags indicate which parameters are reported and which are not. The fields corresponding to unreported parameters MUST be present, but are set to zero. The receiver MUST ignore any Video Statistics Summary Report Block with a non-zero value in any field flagged as unreported. The Transport Streams Statistics Summary Report Block has the following format: Wu, et al. Expires September 3, 2011 [Page 10] Internet-Draft RTCP XR Quality Report Blocks March 2011 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | BT=TBD |T|P| rsd. | block length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | begin_seq | end_seq | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | lost_frames | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | dup frames | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | partial_lost_frames | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | partial_dup_frames | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | key frames impairement proportion | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Block type (BT): 8 bits The Transport Statistics Summary Report Block is identified by the constant . Picture type indicator (T): 1 bits Picture types used in the different video algorithms compose of key-frame and derivation frame. This field is used to indicate the frame type to be reported. Bits set to 0 if the lost_frames field or dup_frames field contain a key_frame report or reference frame report, 1 if the lost_frames field and dup_frames field contain other derivation frame report. P: 1 bit Bit set to 1 if the partial_lost_frames field or the partial_dup_ frames field contains a report, 0 otherwise. Rsd.: 3 bits This field is reserved for future definition. In the absence of such a definition, the bits in this field MUST be set to zero and MUST be ignored by the receiver. Block length: 16 bits The constant 5, in accordance with the definition of this field in Section 3 of RFC 3611 [RFC3611]. Wu, et al. Expires September 3, 2011 [Page 11] Internet-Draft RTCP XR Quality Report Blocks March 2011 SSRC of source: 32 bits As defined in Section 4.1 of RFC 3611 [RFC3611]. begin_seq: 16 bits As defined in Section 4.1 of RFC 3611 [RFC3611]. end_seq: 16 bits As defined in Section 4.1 of RFC 3611 [RFC3611]. lost_frames: 32 bits Number of lost_frames in the above sequence number interval. dup_frames: 32 bits Number of dup_frames in the above sequence number interval. partial lost_frames: 32 bits Number of partial lost_frames in the above sequence number interval. partial dup_frames: 32 bits Number of partial_dup_frames in the above sequence number interval. key frames impairment proportion:32bits The proportion of key frame impaired by packet loss,discard and duplication. 5.2. Transport Stream Loss and Discard Metrics Block This block reports Loss and Discard metrics statistics beyond the information carried in the standard RTCP packet format. The block reports separately on packets lost on the IP channel, and those that have been received but then discarded by the receiving jitter buffer. It is very useful to distinguish between packets lost by the network and those discarded due to jitter. Both have equal effect on the quality of the video stream, however, having separate counts helps identify the source of quality degradation. These fields MUST be populated, and MUST be set to zero if no packets have been received. Implementations MUST provide values for all the fields defined here. For certain metrics, if the value is undefined or unknown, then the specified default or unknown field value MUST be provided. The block is encoded as six 32-bit words: Wu, et al. Expires September 3, 2011 [Page 12] Internet-Draft RTCP XR Quality Report Blocks March 2011 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | BT=TBD |T | reserved | block length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Loss rate | Discard rate | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block type (BT): 8 bits A Transport Stream Metrics Report Block is identified by the constant . Picture type indicator (T): 1 bits Picture types used in the different video algorithms compose of key-frame and derivation frame. This field is used to indicate the picture type to be reported. Bits set to 0 if the Loss rate field and discard rate field contain a Key_frame report or reference frame report, 1 if the Loss rate field and discard rate field contain other derivation frame reports. reserved: 6 bits This field is reserved for future definition. In the absence of such a definition, the bits in this field MUST be set to zero and MUST be ignored by the receiver. block length: 16 bits The constant 1, in accordance with the definition of this field in Section 3 of RFC 3611 [RFC3611]. SSRC of source: 32 bits The SSRC of the RTP data packet source being reported upon by this report block. in accordance with the definition of this field in Section 3 of RFC 3611 [RFC3611]. Loss rate: 8 bits The fraction of RTP data packets from the source lost since the beginning of reception, expressed as a fixed point number with the binary point at the left edge of the field. This value is calculated by dividing the total number of lost packets containing specified frame (e.g., Key frame) (after the effects of applying any error protection such as FEC) by the total number of packets expected, multiplying the result of the division by 256, limiting the maximum value to 255 (to avoid overflow), and taking the integer part. The numbers of duplicated packets and discarded packets do not enter into this calculation. Since receivers cannot be required to maintain unlimited buffers, a receiver MAY categorize late-arriving packets as lost. The degree of lateness Wu, et al. Expires September 3, 2011 [Page 13] Internet-Draft RTCP XR Quality Report Blocks March 2011 that triggers a loss SHOULD be significantly greater than that which triggers a discard. Discard rate: 8 bits The fraction of RTP data packets from the source that have been discarded since the beginning of reception, due to late or early arrival, under-run or overflow at the receiving jitter buffer. This value is expressed as a fixed point number with the binary point at the left edge of the field. It is calculated by dividing the total number of discarded packets containing specified frame (e.g., Key Frame) (excluding duplicate packet discards) by the total number of packets expected, multiplying the result of the division by 256, limiting the maximum value to 255 (to avoid overflow), and taking the integer part. 5.3. Transport Stream Burst Metrics Block This block reports Burst metrics statistics beyond the information carried in the standard RTCP packet format. It reports on the combined effect of losses and discards, as both have equal effect on video quality. In order to properly assess the quality of a video stream, it is desirable to consider the degree of burstiness of packet loss RFC 3357 [RFC3357]. Following the one-way loss pattern sample metrics discussed in [RFC3357], a measure of the spacing between consecutive network packet loss or error events, is a "loss distance". The loss distance metric captures the spacing between the loss periods. The duration of a loss or error event (e.g. and how many packets are lost in that duration) is a "loss period", the loss period metric captures the frequency and length (burstiness) of loss once it starts. Delay reports include the transit delay between RTP end points and the end system processing delays, both of which contribute to the user perceived delay. Implementations MUST provide values for all the fields defined here. For certain metrics, if the value is undefined or unknown, then the specified default or unknown field value MUST be provided. The block is encoded as six 32-bit words: Wu, et al. Expires September 3, 2011 [Page 14] Internet-Draft RTCP XR Quality Report Blocks March 2011 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | BT=TBD | Reserved | block length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Loss Distance | Loss Period | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Max Loss Duration | Reserved. | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ block type (BT): 8 bits A Transport Stream Metrics Report Block is identified by the constant . reserved: 8 bits This field is reserved for future definition. In the absence of such a definition, the bits in this field MUST be set to zero and MUST be ignored by the receiver. block length: 16 bits The constant 2, in accordance with the definition of this field in Section 3 of RFC 3611 [RFC3611]. SSRC of source: 32 bits The SSRC of the RTP data packet source being reported upon by this report block. in accordance with the definition of this field in Section 3 of RFC 3611 [RFC3611]. Loss Distance: 16 bits The mean duration, expressed in milliseconds, of the loss intervals that have occurred since the beginning of reception [DSLF]. The duration of each loss distance is calculated based upon the frames that mark the beginning and end of that period. It is equal to the timestamp of the end frame, plus the duration of the end frame, minus the timestamp of the beginning frame. If the actual values are not available, estimated values MUST be used. If there have been no burst periods, the burst duration value MUST be zero. Loss Period: 16 bits The mean duration, expressed in milliseconds, of the burst loss periods that have occurred since the beginning of reception [DSLF]. The duration of each period is calculated based upon the frame that marks the end of the prior burst loss and the frame that marks the beginning of the subsequent burst loss. It is equal to the timestamp of the subsequent burst frame, minus the Wu, et al. Expires September 3, 2011 [Page 15] Internet-Draft RTCP XR Quality Report Blocks March 2011 timestamp of the prior burst packet, plus the duration of the prior burst packet. If the actual values are not available, estimated values MUST be used. In the case of a gap that occurs at the beginning of reception, the sum of the timestamp of the prior burst packet and the duration of the prior burst packet are replaced by the reception start time. In the case of a gap that occurs at the end of reception, the timestamp of the subsequent burst packet is replaced by the reception end time. If there have been no gap periods, the gap duration value MUST be zero. Max Loss Duration of a single error: 16 bits The maximum loss duration, expressed in milliseconds, of the loss periods that have occurred since the beginning of reception. The recommended max loss duration is specified as less than 16 ms in [DSLF], which provides a balance between interleaver depth protection from xDSL errors induced by impulse noise, delay added to other applications and video service QoE requirements to reduce visible impairments. Reserved: 16 bits All bits SHALL be set to 0 by the sender and SHALL be ignored on reception. block length: 16 bits The constant 2, in accordance with the definition of this field in Section 3 of RFC 3611 [RFC3611]. 5.4. Synthetical Multimedia Quality Metrics Block This block reports the multimedia application performance or quality metrics beyond the information carried in the standard RTCP packet format. Information is recorded about multimedia application QoE metric which is expressed as a MOS ("Mean Opinion Score"), MOS is on a scale from 1 to 5, in which 5 represents excellent and 1 represents unacceptable. MOS scores are usually obtained using subjective testing or using objective algorithm to estimate the multimedia quality. However Subjective testing is not suitable for measuring the multimedia quality since the results may vary from test to test. Therefore using objective algorithm to calculate MOS scores is recommended. ITU-T recommendation [G.1082][P.NAMS][P.NBAMS] defines a methodology for verifying the performance of QoE estimation algorithms for video and audio. Hence this document recommends vendors and implementers to use International Telecommunication Union (ITU)-specified methodologies to measure parameters when possible. The report block contents are dependent upon a series of flag bits carried in the first part of the header. Not all parameters need to Wu, et al. Expires September 3, 2011 [Page 16] Internet-Draft RTCP XR Quality Report Blocks March 2011 be reported in each block. Flags indicate which are and which are not reported. The fields corresponding to unreported parameters MUST be present, but are set to zero. The receiver MUST ignore any Perceptual Quality Metrics Block with a non-zero value in any field flagged as unreported. The Synthetical Multimedia Quality Metrics Block has the following format: 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | BT=TBD |I| MC | Rsd.| block length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | MOS Value | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Block type (BT): 8 bits The Perceptual Quality Metrics Block is identified by the constant . Interval Metric flag (I): 1 bit This field is used to indicate whether the Basic Loss/Discard metrics are Interval or Cumulative metrics, that is, whether the reported values applies to the most recent measurement interval duration between successive metrics reports (I=1) (the Interval Duration) or to the accumulation period characteristic of cumulative measurements (I=0) (the Cumulative Duration). MoS Class (MC): 4 bits This field is used to indicate the MOS type to be reported. The MOS type is defined as follows: 0000 MOS-A - Audio Quality MOS [G.107][P.564]. 0001 MOS-V - Video Quality MOS [P.NAMS][P.NBAMS]. 0010 MOS-AV - Audio-Video Quality MOS[P.NAMS][P.NBAMS]. 0100~1111 - Reserved Rsd.: 7 bits This field is reserved for future definition. In the absence of such a definition, the bits in this field MUST be set to zero and MUST be ignored by the receiver. Wu, et al. Expires September 3, 2011 [Page 17] Internet-Draft RTCP XR Quality Report Blocks March 2011 SSRC of source: 32 bits As defined in Section 4.1 of [RFC3611]. MOS Value: Variable Length The estimated mean opinion score for Audio Qulity, Video Quality or Audio-Video quality is defined as including the effects of delay and other effects that would affect Audio-Video quality [G.1082][P.NAMS][P.NBAMS]. It is expressed as an integer in the range 10 to 50, corresponding to MOS x 10, as for MOS. A value of 127 indicates that this parameter is unavailable. Values other than 127 and the valid range defined above MUST NOT be sent and MUST be ignored by the receiving system. 6. SDP Signaling Six new parameters are defined for the six report blocks defined in this document to be used with Session Description Protocol (SDP) [RFC4566] using the Augmented Backus-Naur Form (ABNF) [RFC5234]. They have the following syntax within the "rtcp-xr" attribute [RFC3611]: Wu, et al. Expires September 3, 2011 [Page 18] Internet-Draft RTCP XR Quality Report Blocks March 2011 rtcp-xr-attrib = "a=rtcp-xr:" [xr-format *(SP xr-format)] CRLF xr-format = RTP-flows-init-syn / RTP-flows-general-syn / multimedia-quality-metrics / transport-stream-loss-metrics / transport-stream-burst-metrics / transport-stat-summary / layered-stream-stat-metrics RTP-flows-init-syn = "RTP-flows-init-syn" ["=" max-size] max-size = 1*DIGIT ; maximum block size in octets RTP-flow-general-syn = "RTP-flows-general-syn" ["=" max-size] max-size = 1*DIGIT ; maximum block size in octets transport-stream-burst-metrics = "transport-stream-burst-metrics" ["=" max-size] max-size = 1*DIGIT ; maximum block size in octets transport-stream-loss-metrics = "transport-stream-loss-metrics" ["=" stat-flag *("," stat-flag)] stat-flag = "key Frame loss and duplication" / "derivation Frame loss and duplication" transport-stream-stat-summary = "transport-stream-stat-summary" ["=" stat-flag *("," stat-flag)] stat-flag = "key Frame loss and duplication" / "derivation Frame loss and duplication" layered-stream-stat-metrics = "layered-stream-stat-metrics" ["=" stat-flag *("," stat-flag)] stat-flag = "base layer packet" / "enhancment layer packet" multimedia-quality-metrics = "multimedia-quality-metrics" ["=" stat-flag *("," stat-flag)] stat-flag = "Interval Metrics" /"Cumulative metrics" Refer to Section 5.1 of RFC 3611 [RFC3611] for a detailed description and the full syntax of the "rtcp-xr" attribute. Wu, et al. Expires September 3, 2011 [Page 19] Internet-Draft RTCP XR Quality Report Blocks March 2011 7. IANA Considerations New report block types for RTCP XR are subject to IANA registration. For general guidelines on IANA allocations for RTCP XR, refer to Section 6.2 of [RFC3611]. This document assigns six new block type values in the RTCP XR Block Type Registry: Name: RFISD Long Name: RTP Flows Initial Synchronization Delay Value Reference: Section 4.1 Name: RFGSO Long Name: RTP Flows General Synchronization Offset Metrics Block Value Reference: Section 4.2 Name: TSSS Long Name: Transport Stream Statistics Summary Value Reference: Section 5.1 Name: LSSM Value Long Name: Layered Stream Statistics Metrics Reference: Section 4.3 Name: TSLDM Long Name: Transport Stream Loss and Discard Metrics Value Reference: Section 5.2 Name: TSBM Long Name: Transport Stream Burst Metrics Value Reference: Section 5.3 Name: SMQM Long Name: Synthetical Multimedia Quality Metric Value Reference: Section 5.4 This document also registers seven SDP [RFC4566] parameters for the "rtcp-xr" attribute in the RTCP XR SDP Parameters Registry: Wu, et al. Expires September 3, 2011 [Page 20] Internet-Draft RTCP XR Quality Report Blocks March 2011 * "RTP-flows-init-syn" * "RTP-flows-general-syn" * "multimedia-quality-metrics" * "transport-stream-loss-metrics" * "transport-stream-burst-metrics" * "transport-stat-summary" * "layered-stream-stat-metrics" The contact information for the registrations is: Qin Wu sunseawq@huawei.com 101 Software Avenue, Yuhua District Nanjing, JiangSu 210012 China 8. Security Considerations The new RTCP XR report blocks proposed in this document introduces no new security considerations beyond those described in [RFC3611]. 9. Acknowledgements The authors would like to thank Bill Ver Steeg, David R Oran, Ali Begen,Colin Perkins, Roni Even,Youqing Yang, Wenxiao Yu and Yinliang Hu for their valuable comments and suggestions on this document. 10. References 10.1. Normative References [I-D.ietf-avt-rtp-svc] Wenger, S., Wang, Y., Schierl, T., and A. Eleftheriadis, "RTP Payload Format for Scalable Video Coding", draft-ietf-avt-rtp-svc-27 (work in progress), February 2011. [ISO-IEC.13818-1.2007] International Organization for Standardization, "Information technology - Generic coding of moving pictures and associated audio information: Systems", ISO International Standard 13818-1, October 2007. [RFC2119] Bradner, S., "Key words for use in RFCs to Indicate Requirement Levels", BCP 14, RFC 2119, March 1997. Wu, et al. Expires September 3, 2011 [Page 21] Internet-Draft RTCP XR Quality Report Blocks March 2011 [RFC2250] Hoffman, D., Fernando, G., Goyal, V., and M. Civanlar, "RTP Payload Format for MPEG1/MPEG2 Video", RFC 2250, January 1998. [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, July 2003. [RFC3611] Friedman, T., Caceres, R., and A. Clark, "RTP Control Protocol Extended Reports (RTCP XR)", RFC 3611, November 2003. [RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session Description Protocol", RFC 4566, July 2006. [RFC5234] Crocker, D. and P. Overell, "Augmented BNF for Syntax Specifications: ABNF", STD 68, RFC 5234, January 2008. [RFC6051] Perkins, C. and T. Schierl, "Rapid Synchronisation of RTP Flows", RFC 6051, November 2010. 10.2. Informative References [DSLF] Rahrer, T., Ed., Fiandra, Ed., and Wright, Ed., "Triple- play Services Quality of Experience (QoE) Requirements", DSL Forum Technical Report TR-126, December 2006. [G.107] ITU-T, "The E Model, a computational model for use in transmission planning", ITU-T Recommendation G.107, April 2009. [G.1082] ITU-T, "Measurement-based methods for improving the robustness of IPTV performance", ITU-T Recommendation G.1082, April 2009. [I-D.ietf-pmol-metrics-framework-02] Clark, A., "Framework for Performance Metric Development". [IEEE] IEEE, "Human Perception of Jitter and Media Synchronization", IEEE Journal on Selected Areas in Communications Vol. 14, No. 1, January 1996. [P.564] ITU-T, "Conformance testing for narrowband Voice over IP transmission quality assessment models", ITU-T Recommendation P.564, July 2006. [P.NAMS] ITU-T, "Non-intrusive parametric model for the Assessment of performance of Multimedia Streaming", ITU-T Wu, et al. Expires September 3, 2011 [Page 22] Internet-Draft RTCP XR Quality Report Blocks March 2011 Recommendation P.NAMS, November 2009. [P.NBAMS] ITU-T, "non-intrusive bit-stream model for assessment of performance of multimedia streaming", ITU-T Recommendation P.NBAMS, November 2009. [RFC3357] Koodli, R. and R. Ravikanth, "One-way Loss Pattern Sample Metrics", RFC 3357, August 2002. Authors' Addresses Qin Wu Huawei 101 Software Avenue, Yuhua District Nanjing, Jiangsu 210012 China Email: sunseawq@huawei.com Glen Zorn Network Zen 77/440 Soi Phoomjit, Rama IV Road Phra Khanong, Khlong Toie Bangkok 10110 Thailand Phone: +66 (0) 87 502 4274 Email: gwz@net-zen.net Roland Schott Deutsche Telekom Laboratories Deutsche-Telekom-Allee 7 Darmstadt 64295 Germany Email: Roland.Schott@telekom.de Wu, et al. Expires September 3, 2011 [Page 23]