]> Keio University

Graduate School of Media and Governance 5322 Endo Fujisawa Kanagawa 252-0882 Japan asaeda@wide.ad.jp

Huawei Technologies Co., Ltd.

101 Software Avenue, Yuhua District Nanjing Jiangsu 210012 China Rachel@huawei.com

Huawei Technologies Co., Ltd.

101 Software Avenue, Yuhua District Nanjing Jiangsu 210012 China sunseawq@huawei.com

This document defines an RTCP XR Report Block and associated SDP parameters that allows the reporting of synchronization delay and offset metrics for use in a range of RTP applications.

This draft defines a new block type to augment those defined in , for use in a range of RTP applications. This new block type supports reporting of Initial synchronization delay. Information is recorded about the difference between the start of RTP sessions and the time the RTP receiver acquires all components of RTP sessions . It also supports reporting of the general Synchronization offset status of an arbitrary number of streams, with the same RTCP CNAME. Information is recorded about the synchronization offset time of each RTP stream relative to the reference RTP stream with the same CNAME and General Synchronization Offset of zero. The metrics belong to the class of transport level metrics defined in (work in progress).

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.

The report blocks defined in this document could be used by dedicated network monitoring 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. 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 playout. For multicast session, such initial synchronization delay varies with the session bandwidth and the number of members, the number of senders in the session. The RTP flow synchronization delay block can be used to report the initial synchronization delay of these RTP streams beyond the information carried in the standard RTCP packet format. In the absence of packet loss, the initial synchronization 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. In an RTP multimedia session, there can be an arbitrary number of streams, with the same RTCP CNAME. The RTP Flows General Synchronization Offset block can be used to report the general Synchronization offset status of these RTP streams beyond the information carried in the standard RTCP packet format. For layered session or multimedia session, the first RTP packet can be chosen as the basic packet of reference RTP stream.

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 . 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 .

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 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The Statistics Summary Report Block is identified by the constant <RFISD>. The constant 2, in accordance with the definition of this field in Section 3 of RFC 3611. The SSRC of the RTP data packet source being reported upon by this report block. (Section 4.1 of ). 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 . 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 .

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.

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 | Reserved | Block length | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | General Synchronization Offset | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The RTP Flow General Synchronization Offset Report Block is identified by the constant <RFGSO>. The constant 3, in accordance with the definition of this field in Section 3 of RFC 3611. The SSRC of the RTP data packet source being reported upon by this report block. (Section 4.1 of ). 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 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.

Two new parameters are defined for the two report blocks defined in this document to be used with Session Description Protocol (SDP) using the Augmented Backus-Naur Form (ABNF) . They have the following syntax within the "rtcp-xr" attribute :

rtcp-xr-attrib = "a=rtcp-xr:" [xr-format *(SP xr-format)] CRLF xr-format = RTP-flows-init-syn / RTP-flows-general-syn 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

Refer to Section 5.1 of RFC 3611 for a detailed description and the full syntax of the "rtcp-xr" attribute.

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. This document assigns two new block type values in the RTCP XR Block Type Registry: RFISD RTP Flows Initial Synchronization Delay <RFISD> RFGSO RTP Flows General Synchronization Offset Metrics Block <RFGSO> This document also registers two new SDP parameters for the "rtcp-xr" attribute in the RTCP XR SDP Parameters Registry: "RTP-flows-init-syn" "RTP-flows-general-syn" The contact information for the registrations is: Qin Wu sunseawq@huawei.com 101 Software Avenue, Yuhua District Nanjing, Jiangsu 210012, China

The new RTCP XR report blocks proposed in this document introduces no new security considerations beyond those described in .

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.

This memo describes an RTP payload format for Scalable Video Coding (SVC) as defined in Annex G of ITU-T Recommendation H.264, which is technically identical to Amendment 3 of ISO/IEC International Standard 14496-10. The RTP payload format allows for packetization of one or more Network Abstraction Layer (NAL) units in each RTP packet payload, as well as fragmentation of a NAL unit in multiple RTP packets. Furthermore, it supports transmission of an SVC stream over a single as well as multiple RTP sessions. The payload format defines a new media subtype name "H264-SVC", but is still backwards compatible to [I-D.ietf-avt-rtp-rfc3984bis] since the base layer, when encapsulated in its own RTP stream, must use the H.264 media subtype name ("H264") and the packetization method specified in [I- D.ietf-avt-rtp-rfc3984bis]. The payload format has wide applicability in videoconferencing, Internet video streaming, and high bit-rate entertainment-quality video, among others.Table of Contents &rfc2250; &rfc3611; &rfc2119; &rfc4566; &rfc5234; &rfc3550; 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. Huawei Telchemy Incorporated Telchemy

Note to the RFC-Editor: please remove this section prior to publication as an RFC.

This document is separated from draft-wu-xrblock-rtcp-xr-quality-monitoring-01 with some editorial changes and focuses on RTP Flow Initial Synchronization Delay and RTP Flows General Synchronization Offset.

Separate Synchronization Delay and Offset Metrics Block into two independent block based on comments on the list.