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applications MOQ Mailing List Internet-Draft QUIC This document describes use cases and requirements that guide the specification of a simple, low-latency media delivery solution for ingest and distribution, using either the QUIC protocol or WebTransport. Note to Readers RFC Editor: please remove this section before publication Source code and issues for this draft can be found at https://github.com/fiestajetsam/draft-gruessing-moq-requirements. Discussion of this draft should take place on the IETF Media Over QUIC (MoQ) mailing list, at https://www.ietf.org/mailman/listinfo/moq.

Introduction This document describes use cases and requirements that guide the specification of a simple, low-latency media delivery solution for ingest and distribution , using either the QUIC protocol or WebTransport .

Note for MOQ Working Group participants This version of the document is intended to provide the MOQ working group with a starting point for work on the "Use Cases and Requirements document" milestone. The update implements the work plan described in . The authors intend to request MOQ working group adoption after IETF 115, so the working group can begin to focus on these topics in earnest.

Terminology The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 when, and only when, they appear in all capitals, as shown here.

Use Cases Informing This Proposal Our goal in this section is to understand the range of use cases that are in scope for "Media Over QUIC" . For each use case in this section, we also describe

• the number of senders or receiver in a given session transmitting distinct streams,
• whether a session has bi-directional flows of media from senders and receivers, and
• the worst-case expected RTT requirements.

It is likely that we should add other characteristics, as we come to understand them.

Interactive Media The use cases described in this section have one particular attribute in common - the target latency for these cases are on the order of one or two RTTs. In order to meet those targets, it is not possible to rely on protocol mechanisms that require multiple RTTs to function effectively. For example,

• When the target latency is on the order of one RTT, it makes sense to use FEC and codec-level packet loss concealment , rather than selectively retransmitting only lost packets. These mechanisms use more bytes, but do not require multiple RTTs in order to recover from packet loss.
• When the target latency is on the order of one RTT, it is impossible to use congestion control schemes like BBR , since BBR has probing mechanisms that rely on temporarily inducing delay, but these mechanisms can then amortize the consequences of induced delay over multiple RTTs.

This may help to explain why interactive use cases have typically relied on protocols such as RTP , which provide low-level control of packetization and transmission, and make no provision for retransmission.

Gaming

Attribute	Value
Senders/Receivers	One to One
Bi-directional	Yes
Latency	Ull-50

Where media is received, and user inputs are sent by the client. This may also include the client receiving other types of signaling, such as triggers for haptic feedback. This may also carry media from the client such as microphone audio for in-game chat with other players.

Remote Desktop

Attribute	Value
Senders/Receivers	One to One
Bi-directional	Yes
Latency	Ull-50

Where media is received, and user inputs are sent by the client. Latency requirements with this use case are marginally different than the gaming use case. This may also include signalling and/or transmitting of files or devices connected to the user's computer.

Video Conferencing/Telephony

Attribute	Value
Senders/Receivers	Many to Many
Bi-directional	Yes
Latency	Ull-50 to Ull-200

Where media is both sent and received; This may include audio from both microphone(s) or other inputs, or may include "screen sharing" or inclusion of other content such as slide, document, or video presentation. This may be done as client/server, or peer to peer with a many to many relationship of both senders and receivers. The target for latency may be as large as Ull-200 for some media types such as audio, but other media types in this use case have much more stringent latency targets.

Hybrid Interactive and Live Media For the video conferencing/telephony use case, there can be additional scenarios where the audience greatly outnumbers the concurrent active participants, but any member of the audience could participate. As this has a much larger total number of participants - as many as Live Media Streaming , but with the bi-directionality of conferencing, this should be considered a "hybrid".

Live Media The use cases in this section, unlike the use cases described in , still have "humans in the loop", but these humans expect media to be "responsive", where the responsiveness is more on the order of 5 to 10 RTTs. This allows the use of protocol mechanisms that require more than one or two RTTs - as noted in , end-to-end recovery from packet loss and congestion avoidance are two such protocol mechanisms that can be used with Live Media. To illustrate the difference, the responsiveness expected with videoconferencing is much greater than watching a video, even if the video is being produced "live" and sent to a platform for syndication and distribution.

Live Media Ingest

Attribute	Value
Senders/Receivers	One to One
Bi-directional	No
Latency	Ull-200 to Ultra-Low

Where media is received from a source for onwards handling into a distribution platform. The media may comprise of multiple audio and/or video sources. Bitrates may either be static or set dynamically by signaling of connection information (bandwidth, latency) based on data sent by the receiver.

Live Media Syndication

Attribute	Value
Senders/Receivers	One to One
Bi-directional	No
Latency	Ull-200 to Ultra-Low

Where media is sent onwards to another platform for further distribution. The media may be compressed down to a bitrate lower than source, but larger than final distribution output. Streams may be redundant with failover mechanisms in place.

Live Media Streaming

Attribute	Value
Senders/Receivers	One to Many
Bi-directional	No
Latency	Ull-200 to Ultra-Low

Where media is received from a live broadcast or stream. This may comprise of multiple audio or video outputs with different codecs or bitrates. This may also include other types of media essence such as subtitles or timing signalling information (e.g. markers to indicate change of behaviour in client such as advertisement breaks). The use of "live rewind" where a window of media behind the live edge can be made available for clients to playback, either because the local player falls behind edge or because the viewer wishes to play back from a point in the past.

Requirements for Protocol Work Our goal in this section is to understand the requirements that result from the use cases described in . *Note: the initial high-level organization for this section is taken from Suhas Nandakumar's presentation, "Progressing MOQ" , at the October 2022 MOQ virtual interim meeting, which was in turn taken from the MOQ working group charter . We think this is a reasonable starting point. We won't be surprised to see the high-level structure change a bit as things develop, but we didn't want to have this section COMPLETELY blank when we request working group adoption. TODO: Describe overall, high level requirements that we previously stated in earlier versions of this document.

Common Publication Protocol for Media Ingest and Distribution Many of the use cases have bi-directional flows of media, with clients both sending and receiving media concurrently, thus the protocol should have a unified approach in connection negotiation and signalling to send and received media both at the start and ongoing in the lifetime of a session including describing when flow of media is unsupported (e.g. a live media server signalling it does not support receiving from a given client).

Client Media Request Protocol In the initiation of a session both client and server must perform negotiation in order to agree upon a variety of details before media can move in any direction:

• Is the client authenticated and subsequently authorised to initiate a connection?
• What media is available, and for each what are the parameters such as codec, bitrate, and resolution etc?
• Is sending of media from a client permitted? If so, what media is accepted?

Re-negotiation in an existing protocol should be supported to allow changes in what is being sent of received.

Naming and Addressing Media Resources As multiple streams of media may be available for concurrent sending such as multiple camera views or audio tracks, a means of both identifying the technical properties of each resource (codec, bitrate, etc) as well as a useful identification for playback should be part of the protocol. A base level of optional metadata e.g. the known language of an audio track or name of participant's camera should be supported, but further extended metadata of the contents of the media or its ontology should not be supported.

Packaging Media Packaging of media describes how encapsulation of media to carry the raw media will work. There are at a high level two approaches to this:

• Within the protocol itself, where the protocol defines the carrying for each media encoding the ancillary data required for decoding the media.
• A common encapsulation format such as ISOBMFF which defines a generic method for all media and handles ancillary decode information.

The working group must agree on which approach should be taken to the packaging of media, taking into consideration the various technical trade offs that each provide.

End-to-end Security End-to-end security describes the use of encryption of the media stream(s) to provide confidentiality in the presence of unauthorized intermediates or observers and prevent or restrict ability to decrypt the media without authorization. Generally, there are three aspects of end-to-end media security:

• Media Rights Management, which refers to the authorization of receivers to decode a media stream.
• Sender-to-Receiver Media Security, which refers to the ability of media senders and receivers to transfer media while protected from authorized intermediates and observers, and
• Node-to-node Media Security, which refers to security when authorized intermediaries are needed to transform media into a form acceptable to authorized receivers. For example, this might refer to a video transcoder between the media sender and receiver.

**Note: "Node-to-node" refers to a path segment connecting two MOQ nodes, that makes up part of the end-to-end path between the MOQ sender and ultimate MOQ receiver. The working group must agree on a number of details here, and perhaps the first question is whether the MOQ protocol makes any provision for "node-to-node" media security, or simply treats authorized transcoders as MOQ receivers. If that's the decision all MOQ media security is "sender-to-receiver", but some "ends" may not be either senders or ultimate receivers, from a certain point of view.

IANA Considerations This document makes no requests of IANA.

Security Considerations As this document is intended to guide discussion and consensus, it introduces no security considerations of its own.

References Normative References In many standards track documents several words are used to signify the requirements in the specification. These words are often capitalized. This document defines these words as they should be interpreted in IETF documents. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. RFC 2119 specifies common key words that may be used in protocol specifications. This document aims to reduce the ambiguity by clarifying that only UPPERCASE usage of the key words have the defined special meanings. Informative References This memorandum describes RTP, the real-time transport protocol. RTP provides end-to-end network transport functions suitable for applications transmitting real-time data, such as audio, video or simulation data, over multicast or unicast network services. RTP does not address resource reservation and does not guarantee quality-of- service for real-time services. The data transport is augmented by a control protocol (RTCP) to allow monitoring of the data delivery in a manner scalable to large multicast networks, and to provide minimal control and identification functionality. RTP and RTCP are designed to be independent of the underlying transport and network layers. The protocol supports the use of RTP-level translators and mixers. Most of the text in this memorandum is identical to RFC 1889 which it obsoletes. There are no changes in the packet formats on the wire, only changes to the rules and algorithms governing how the protocol is used. The biggest change is an enhancement to the scalable timer algorithm for calculating when to send RTCP packets in order to minimize transmission in excess of the intended rate when many participants join a session simultaneously. [STANDARDS-TRACK] This document describes a framework for using Forward Error Correction (FEC) codes with applications in public and private IP networks to provide protection against packet loss. The framework supports applying FEC to arbitrary packet flows over unreliable transport and is primarily intended for real-time, or streaming, media. This framework can be used to define Content Delivery Protocols that provide FEC for streaming media delivery or other packet flows. Content Delivery Protocols defined using this framework can support any FEC scheme (and associated FEC codes) that is compliant with various requirements defined in this document. Thus, Content Delivery Protocols can be defined that are not specific to a particular FEC scheme, and FEC schemes can be defined that are not specific to a particular Content Delivery Protocol. [STANDARDS-TRACK] This document defines the Opus interactive speech and audio codec. Opus is designed to handle a wide range of interactive audio applications, including Voice over IP, videoconferencing, in-game chat, and even live, distributed music performances. It scales from low bitrate narrowband speech at 6 kbit/s to very high quality stereo music at 510 kbit/s. Opus uses both Linear Prediction (LP) and the Modified Discrete Cosine Transform (MDCT) to achieve good compression of both speech and music. [STANDARDS-TRACK] This document defines the core of the QUIC transport protocol. QUIC provides applications with flow-controlled streams for structured communication, low-latency connection establishment, and network path migration. QUIC includes security measures that ensure confidentiality, integrity, and availability in a range of deployment circumstances. Accompanying documents describe the integration of TLS for key negotiation, loss detection, and an exemplary congestion control algorithm. Google Google Google Google Google This document specifies the BBR congestion control algorithm. BBR ("Bottleneck Bandwidth and Round-trip propagation time") uses recent measurements of a transport connection's delivery rate, round-trip time, and packet loss rate to build an explicit model of the network path. BBR then uses this model to control both how fast it sends data and the maximum volume of data it allows in flight in the network at any time. Relative to loss-based congestion control algorithms such as Reno [RFC5681] or CUBIC [RFC8312], BBR offers substantially higher throughput for bottlenecks with shallow buffers or random losses, and substantially lower queueing delays for bottlenecks with deep buffers (avoiding "bufferbloat"). BBR can be implemented in any transport protocol that supports packet-delivery acknowledgment. Thus far, open source implementations are available for TCP [RFC793] and QUIC [RFC9000]. This document specifies version 2 of the BBR algorithm, also sometimes referred to as BBRv2 or bbr2. Facebook Facebook Facebook Facebook RUSH is an application-level protocol for ingesting live video. This document describes the protocol and how it maps onto QUIC. Discussion Venues This note is to be removed before publishing as an RFC. Discussion of this document takes place on the mailing list (), which is archived at . Source for this draft and an issue tracker can be found at https://github.com/afrind/draft-rush. Twitch Meta Cisco This document defines the core behavior for Warp, a segmented live media transport protocol over QUIC. Media is split into segments based on the underlying media encoding and transmitted independently over QUIC streams. QUIC streams are prioritized based on the delivery order, allowing less important segments to be starved or dropped during congestion. cisco Cisco This specification outlines the design for a media delivery protocol over QUIC. It aims at supporting multiple application classes with varying latency requirements including ultra low latency applications such as interactive communication and gaming. It is based on a publish/subscribe metaphor where entities publish and subscribe to data that is sent through, and received from, relays in the cloud. The information subscribed to is named such that this forms an overlay information centric network. The relays allow for efficient large scale deployments. cisco Cisco Private Octopus Inc. Recently new use cases have emerged requiring higher scalability of media delivery for interactive realtime applications and much lower latency for streaming applications and a combination thereof. draft-jennings-moq-arch specifies architectural aspects of QuicR, a media delivery protocol based on publish/subscribe metaphor and Relay based delivery tree, that enables a wide range of realtime applications with different resiliency and latency needs. This specification defines the protocol aspects of the QuicR media delivery architecture.

Acknowledgements The authors would like to thank several authors of individual drafts that fed into the "Media Over QUIC" charter process:

• Kirill Pugin, Alan Frindell, Jordi Cenzano, and Jake Weissman (,
• Luke Curley (), and
• Cullen Jennings and Suhas Nandakumar (), together with Christian Huitema ().

We would also like to thank Suhas Nandakumar for his presentation, "Progressing MOQ" , at the October 2022 MOQ virtual interim meeting. We used his outline as a starting point for the Requirements section (). James Gruessing would also like to thank Francesco Illy and Nicholas Book for their part in providing the needed motivation.