]> National Institute of Information and Communications Technology

4-2-1 Nukui-Kitamachi Koganei Tokyo 184-8795 Japan asaeda@nict.go.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 two RTP Control Protocol (RTCP) Extended Report (XR) Blocks and associated with SDP parameters that allow the reporting of synchronization delay and offset metrics for use in a range of RTP applications.

This draft defines two new block types to augment those defined in , for use in a range of RTP applications. The first new block type supports reporting of Initial Synchronization Delay to establish multimedia session. Information is recorded about time difference between the start of RTP sessions and the time the RTP receiver acquires all components of RTP sessions in the multimedia session . The second new block type supports reporting of the relative synchronization offset time of two arbitrary streams (e.g., between audio and video streams), with the same RTCP CNAME included in RTCP SDES packets . 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. These metrics belong to the class of transport level metrics defined in .

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. In addition, the following terms are defined: A multimedia session comprises a set of concurrent RTP sessions among a common group of participants, using one RTP session for each media type. The initial synchronization Delay is the average time for receiver to synchronize all components of a multimedia session . Synchronization between two media streams must be maintained to ensure satisfactory QoE. Two media streams can be of the same media type belonging to one RTP session or in different media types belonging to one multimedia session. The Synchronization Offset is the relative time difference of the two media streams that need to be synchronized.

The report blocks defined in this document could be used by dedicated network monitoring applications. When joining each session in layered video sessions or the multimedia session, a receiver may not synchronize playout across the multimedia session or layered video session until RTCP SR packets have been received on all components of RTP sessions. The component RTP session are referred to as each RTP session for each media type in multimedia session or separate RTP session for each layer in the layered video session. 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, the number of members, and the number of senders in the session. The RTP flow Initial synchronization delay block can be used to report the initial synchronization delay to receive all the RTP streams belonging to the same multimedia session or layered video session. 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 based on the in-band mapping of RTP and NTP-format timestamps or 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 carried in different RTP sessions, with the same RTCP CNAME. These streams may be not synchronized with each other. For example, one audio stream and one video stream belong to the same session and audio stream are transmitted lag behind video stream for multiple tens of milliseconds . The RTP Flows Synchronization Offset block can be used to report such synchronization offset between video stream and audio stream.

This block is sent by RTP receivers and reports Initial synchronization delay beyond the information carried in the standard RTCP packet format. Information is recorded about time difference between the start of RTP sessions and the time the RTP receiver acquires all components of RTP sessions .

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=RFISD | Reserved | Block length=2 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of Source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Initial Synchronization Delay | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The RTP Flows Initial Synchronization Delay Report Block is identified by the constant <RFISD>. 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. The constant 2, in accordance with the definition of this field in Section 3 of RFC 3611. The SSRC of the media source SHALL be set to the value of the SSRC identifier carried in any arbitrary component of RTP sessions belonging to the same multimedia session. The average delay, expressed in units of 1/65536 seconds, from the RTCP packets received on all of the components RTP sessions to the beginning of session . The value is calculated based on received RTCP SR packets or the RTP header extension containing in-band mapping of RTP and NTP-format timestamps . If there is no packet loss, the initial synchronization delay is expected to be equal to the average time taken to receive the first RTCP packet in the RTP session with the longest RTCP reporting interval or the average time taken to receive the first RTP header extension containing in-band mapping of RTP and NTP-format timestamps. If the measurement is unavailable, the value of this field with all bits set to 1 SHOULD be reported.

In the RTP multimedia sessions, there can be an arbitrary number of Media streams and each media stream (e.g., audio stream or video stream) is sent in a separate RTP stream. The receiver associates RTP streams to be synchronized by means of RTCP CNAME contained in the RTCP Source Description (SDES) packets . This block is sent by RTP receivers and reports synchronization offset of the arbitrary two RTP streams that needs to be synchronized in the RTP multimedia session. Information is recorded about the relative time difference between two media streams with the same CNAME. In two media streams, one media stream is the reference stream, which can be chosen as the arbitrary stream with minimum delay according to the common criterion defined in section 6.2.2.1 of .

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=RFSO |L| Reserved | Block length=4 | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of source | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | SSRC of reference | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Synchronization Offset, most significant word | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | Synchronization Offset, least significant word | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

The RTP Flow General Synchronization Offset Report Block is identified by the constant <RFSO>. This field is used to indicate whether the reporting stream lead before the reference stream or lag behind the reference stream. The value of this field is set to 1, if the reporting stream lead before the reference stream, 0 otherwise. 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. The constant 4, in accordance with the definition of this field in Section 3 of RFC 3611. The SSRC of the media source SHALL be set to the value of the SSRC identifier of the reporting RTP stream to which the XR relates. The SSRC of the reference stream SHALL be set to the value of the SSRC identifier of the reference RTP stream to which the XR relates. The synchronization offset of the reporting RTP stream relative to the reference RTP stream with the same CNAME. The calculation of Synchronization Offset is similar to Difference D calculation in the RFC3550. That is to say,if Si is the RTP timestamp from the reporting RTP packet i, and Ri is the time of arrival in RTP timestamp units for reporting RTP packet i, Sj is the RTP timestamp from the reference RTP packet j, and Rj is the time of arrival in RTP timestamp units for reference RTP packet j, then the value of the synchronization offset D may be expressed as D(i,j) = (Rj - Ri) - (Sj - Si) = (Rj - Sj) - (Ri - Si) The value of the synchronization offset is represented using a 64-bit NTP-format timestamp as defined in , which is 64-bit unsigned fixed-point number with the integer part in the first 32 bits and the fractional part in the last 32 bits. If the measurement is unavailable, the value of this field with all bits set to 1 SHOULD be reported.

defines the use of SDP (Session Description Protocol) for signaling the use of XR blocks. XR blocks MAY be used without prior signaling.

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-delay / RTP-flows-syn-offset RTP-flows-init-syn-delay = "RTP-flows-init-syn-delay" RTP-flow-syn-offset = "RTP-flows-syn-offset"

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

When SDP is used in offer-answer context, the SDP Offer/Answer usage defined in applies.

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> RFSO RTP Flows Synchronization Offset Metrics Block <RFSO> This document also registers two new SDP parameters for the "rtcp-xr" attribute in the RTCP XR SDP Parameters Registry: "RTP-flows-init-syn-delay" "RTP-flows-syn-offset" 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, Kevin Gross, Jing Zhao, Fernando Boronat Seguí, 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 &rfc3611; &rfc2119; &rfc4566; &rfc5234; &rfc3550; This memo outlines how RTP sessions are synchronized, and discusses how rapidly such synchronisation can occur. We show that most RTP sessions can be synchronized 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. BBF Forum Huawei ITU-T

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

The following are the major changes compared to previous version: Editorial change based on comments raised on the list.