]> Technical University Munich

ott@in.tum.de

Technical University Munich

mathis.engelbart@gmail.com

ART AVTCORE Internet-Draft This document specifies a minimal mapping for encapsulating RTP and RTCP packets within QUIC. It also discusses how to leverage state from the QUIC implementation in the endpoints to reduce the exchange of RTCP packets. Discussion Venues Source for this draft and an issue tracker can be found at .

Introduction The Real-time Transport Protocol (RTP) is generally used to carry real-time media for conversational media sessions, such as video conferences, across the Internet. Since RTP requires real-time delivery and is tolerant to packet losses, the default underlying transport protocol has been UDP, recently with DTLS on top to secure the media exchange and occasionally TCP (and possibly TLS) as a fallback. With the advent of QUIC and, most notably, its unreliable DATAGRAM extension , another secure transport protocol becomes available. QUIC and its DATAGRAMs combine desirable properties for real-time traffic (e.g., no unnecessary retransmissions, avoiding head-of-line blocking) with a secure end-to-end transport that is also expected to work well through NATs and firewalls. Moreover, with QUIC's multiplexing capabilities, reliable and unreliable transport connections as, e.g., needed for WebRTC, can be established with only a single port used at either end of the connection. This document defines a mapping of how to carry RTP over QUIC. The focus is on RTP and RTCP packet mapping and on reducing the amount of RTCP traffic by leveraging state information readily available within a QUIC endpoint. This document also briefly touches upon how to signal media over QUIC using the Session Description Protocol (SDP) . The scope of this document is limited to unicast RTP/RTCP. Note that this draft is similar in spirit to but differs in numerous ways from .

Terminology and Notation 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. The following terms are used:

Congestion Controller:

QUIC specifies a congestion controller in , but the specific requirements for interactive real-time media lead to the development of dedicated congestion control algorithms. In this document, the term congestion controller refers to these algorithms dedicated to real-time applications.

Datagram:

Datagrams exist in UDP as well as in QUICs unreliable datagram extension. If not explicitly noted differently, the term datagram in this document refers to a QUIC Datagram as defined in .

Endpoint:

A QUIC server or client that participates in an RTP over QUIC session.

Frame:

A QUIC frame as defined in .

Media Encoder:

An entity that is used by an application to produce a stream of encoded media, which can be packetized in RTP packets to be transmitted over QUIC.

Receiver:

An endpoint that receives media in RTP packets and may send or receive RTCP packets.

Sender:

An endpoint that sends media in RTP packets and may send or receive RTCP packets.

Packet diagrams in this document use the format defined in to illustrate the order and size of fields.

Protocol Overview This document introduces a mapping of the Real-time Transport Protocol (RTP) to the QUIC transport protocol. RTP over QUIC allows the use of QUIC streams and unreliable QUIC datagrams to transport real-time data, and thus, the QUIC implementation MUST support QUICs unreliable datagram extension, if RTP packets should be sent over QUIC datagrams. Since datagram frames cannot be fragmented, the QUIC implementation MUST also provide a way to query the maximum datagram size so that an application can create RTP packets that always fit into a QUIC datagram frame. specifies that RTP sessions need to be transmitted on different transport addresses to allow multiplexing between them. RTP over QUIC uses a different approach to leverage the advantages of QUIC connections without managing a separate QUIC connection per RTP session. QUIC does not provide demultiplexing between different flows on datagrams but suggests that an application implement a demultiplexing mechanism if required. An example of such a mechanism are flow identifiers prepended to each datagram frame as described in . RTP over QUIC uses a flow identifier to replace the network address and port number to multiplex many RTP sessions over the same QUIC connection. A congestion controller can be plugged in to adapt the media bitrate to the available bandwidth. This document does not mandate any congestion control algorithm. Some examples include Network-Assisted Dynamic Adaptation (NADA) and Self-Clocked Rate Adaptation for Multimedia (SCReAM) . These congestion control algorithms require some feedback about the network's performance to calculate target bitrates. Traditionally this feedback is generated at the receiver and sent back to the sender via RTCP. Since QUIC also collects some metrics about the network's performance, these metrics can be used to generate the required feedback at the sender-side and provide it to the congestion controller to avoid the additional overhead of the RTCP stream.

Encapsulation QUIC supports two transport methods: reliable streams and unreliable datagrams . This document specifies a mapping of RTP to both of the transport modes. The encapsulation format for RTP over QUIC is described in . and explain the specifics of mapping of RTP to QUIC streams and QUIC datagrams respectively.

RTP over QUIC Payload Format

Flow Identifier:

Flow identifier to demultiplex different data flows on the same QUIC connection.

RTP/RTCP Packet:

The RTP/RTCP packet to transmit.

For multiplexing different RTP and other data streams on the same QUIC connection, each RTP/RTCP packet is prefixed with a flow identifier. A flow identifier is a QUIC variable-length integer which must be unique per stream. RTP and RTCP packets of a single RTP session MAY be sent using the same flow identifier (following the procedures defined in , or they MAY be sent using different flow identifiers. The respective mode of operation MUST be indicated using the appropriate signaling, e.g., when using SDP as discussed in . RTP and RTCP packets of different RTP sessions MUST be sent using different flow identifiers. Differentiating RTP/RTCP packets of different RTP sessions from non-RTP/RTCP datagrams is the responsibility of the application by means of appropriate use of flow identifiers and the corresponding signaling.

QUIC Streams An application MUST open a new QUIC stream for each Application Data Unit (ADU). Each ADU MUST be encapsulated in a single RTP packet and the application MUST not send more than one RTP packet per stream. Opening a new stream for each packet adds implicit framing to RTP packets, allows to receive packets without strict ordering and gives an application the possibility to cancel certain packets. Large RTP packets sent on a stream will be fragmented in smaller QUIC frames, that are transmitted reliably and in order, such that a receiving application can read a complete packet from the stream. No retransmission has to be implemented by the application, since QUIC frames that are lost in transit are retransmitted by the QUIC connection. If it is known to either the sender or the receiver, that a packet, which was not yet successfully and completely transmitted, is no longer needed, either side can close the stream.

• Editor's Note: We considered adding a framing like the one described in to send multiple RTP packets on one stream, but we don't think it is worth the additional overhead only to reduce the number of streams. Moreover, putting multiple ADUs into a single stream would also require defining policies when to use the same (and which) stream and when to open a new one.

QUIC Datagrams RTP packets can be sent in QUIC datagrams. QUIC datagrams are an extension to QUIC described in . QUIC datagrams preserve frame boundaries, thus a single RTP packet can be mapped to a single QUIC datagram, without the need for an additional framing. Senders SHOULD consider the header overhead associated with QUIC datagrams and ensure that the RTP/RTCP packets, including their payloads, QUIC, and IP headers, will fit into path MTU. If an application wishes to retransmit lost RTP packets, the retransmission has to be implemented by the application by sending a new datagram for the RTP packet, because QUIC datagrams are not retransmitted on loss (see also for loss signaling).

RTCP The RTP Control Protocol (RTCP) allows RTP senders and receivers to exchange control information to monitor connection statistics and to identify and synchronize streams. Many of the statistics contained in RTCP packets overlap with the connection statistics collected by a QUIC connection. To avoid using up bandwidth for duplicated control information, the information SHOULD only be sent at one protocol layer. QUIC relies on certain control frames to be sent. In general, applications MAY send RTCP without any restrictions. This document specifies a baseline for replacing some of the RTCP packet types by mapping the contents to QUIC connection statistics. Future documents can extend this mapping for other RTCP format types. It is RECOMMENDED to expose relevant information from the QUIC layer to the application instead of exchanging addtional RTCP packets, where applicable. This section discusses what information can be exposed from the QUIC connection layer to reduce the RTCP overhead and which type of RTCP messages cannot be replaced by similar feedback from the transport layer. The list of RTCP packets in this section is not exhaustive and similar considerations SHOULD be taken into account before exchanging any other type of RTCP control packets.

• TODO: Define parameters for SDP to signal RTCP vs. QUIC feedback. Could use RTCP by default and add parameters for "can use QUIC statistics for X".

Transport Layer Feedback This section explains how some of the RTCP packet types which are used to signal reception statistics can be replaced by equivalent statistics that are already collected by QUIC. The following list explains how this mapping can be achieved for the individual fields of different RTCP packet types. QUIC Datagrams are ack-eliciting packets, which means, that an acknowledgment is triggered when a datagram frame is received. Thus, a sender can assume that an RTP packet arrived at the receiver or was lost in transit, using the QUIC acknowledgments of QUIC Datagram frames. In the following, an RTP packet is regarded as acknowledged, when the QUIC Datagram frame that carried the RTP packet, was acknowledged. For RTP packets that are sent over QUIC streams, an RTP packet can be considered acknowledged, when all frames which carried fragments of the RTP packet were acknowledged. Some of the transport layer feedback that can be implemented in RTCP contains information that is not included in QUIC by default, but can be added via QUIC extensions. One important example are arrival timestamps, which are not part of QUIC's default acknowledgment frames, but can be added using or . Another extension, that can improve the precision of the feedback from QUIC is , which allows a sender to control the delay of acknowledgments sent by the receiver.

• Receiver Reports (PT=201, Name=RR, )
  • Fraction lost: The fraction of lost packets can be directly infered from QUIC's acknowledgments. The calculation SHOULD include all packets up to the acknowledged RTP packet with the highest RTP sequence number. Later packets SHOULD be ignored, since they may still be in flight, unless other QUIC packets that were sent after the datagram frame, were already acknowledged.
  • Cumulative lost: Similar to the fraction of lost packets, the cumulative loss can be infered from QUIC's acknowledgments including all packets up to the latest acknowledged packet.
  • Highest Sequence Number received: The highest sequence number received is the sequence number of all RTP packets that were acknowledged.
  • Interarrival jitter: If QUIC acknowledgments carry timestamps as described in one of the extensions referenced above, senders can infer from QUIC acks the interarrival jitter from the arrival timestamps.
  • Last SR: Similar to RTP arrival times, the arrival time of RTCP Sender Reports can be inferred from QUIC acknowledgments, if they include timestamps.
  • Delay since last SR: This field is not required when the receiver reports are entirely replaced by QUIC feedback.
• Negative Acknowledgments (PT=205, FMT=1, Name=Generic NACK, )
  • The generic negative acknowledgment packet contains information about packets which the receiver considered lost. recommends to use this feature only, if the underlying protocol cannot provide similar feedback. QUIC does not provide negative acknowledgments, but can detect lost packets through acknowledgments.
• ECN Feedback (PT=205, FMT=8, Name=RTCP-ECN-FB, )
  • ECN feedback packets report the count of observed ECN-CE marks. defines two RTCP reports, one packet type (with PT=205 and FMT=8) and a new report block for the extended reports which are listed below. QUIC supports ECN reporting through acknowledgments. If the connection supports ECN, the reporting of ECN counts SHOULD be done using QUIC acknowledgments.
• Congestion Control Feedback (PT=205, FMT=11, Name=CCFB, )
  • RTP Congestion Control Feedback contains acknowledgments, arrival timestamps and ECN notifications for each received packet. Acknowledgments and ECNs can be infered from QUIC as described above. Arrival timestamps can be added through extended acknowledgment frames as described in or .
• Extended Reports (PT=207, Name=XR, )
  • Extended Reports offer an extensible framework for a variety of different control messages. Some of the standard report blocks which can be implemented in extended reports such as loss RLE or ECNs can be implemented in QUIC, too. For other report blocks, it SHOULD be evaluated individually, if the contained information can be transmitted using QUIC instead.

Application Layer Repair and other Control Messages While the previous section presented some RTCP packet that can be replaced by QUIC features, QUIC cannot replace all of the available RTCP packet types. This mostly affects RTCP packet types which carry control information that is to be interpreted by the application layer instead of the transport itself. Sender Reports (PT=200, Name=SR, ) are similar to Receiver Reports. They are send by media senders and additionally contain a NTP and a RTP timestamp and the number of packets and octets transmitted by the sender. The timestamps can be used by a receiver to synchronize streams. QUIC cannot provide a similar control information, since it does not know about RTP timestamps. A QUIC receiver can also not calculate the packet or octet counts, since it does not know about lost datagrams. Thus, sender reports are required in RTP over QUIC to synchronize streams at the receiver. The sender reports SHOULD not contain any receiver report blocks, as the information can be infered from the QUIC transport as explained in the previous section. Next to carrying transmission statistics, RTCP packets can contain application layer control information, that cannot directly be mapped to QUIC. This includes for example the Source Description (PT=202, Name=SDES), Bye (PT=203, Name=BYE) and Application (PT=204, Name=APP) packet types from or many of the payload specific feedback messages (PT=206) defined in , which can for example be used to control the codec behavior of the sender. Since QUIC does not provide any kind of application layer control messaging, these RTCP packet types SHOULD be used in the same way as they would be used over any other transport protocol.

Congestion Control Like any other application on the internet, RTP over QUIC needs to perform congestion control to avoid overloading the network. QUIC is a congestion controlled transport protocol. Senders are required to employ some form of congestion control. The default congestion control specified for QUIC is an alogrithm similar to TCP NewReno, but senders are free to choose any congestion control algorithm as long as they follow the guidelines specified in . RTP does not specify a congestion controller, but provides feedback formats for congestion control (e.g. ) as well as different congestion control algorithms in separate RFCs (e.g. and ). The congestion control algorithms for RTP are specifically tailored for real-time transmissions at low latencies. The available congestion control algorithms for RTP expose a target_bitrate that can be used to dynamically reconfigure media codecs to produce media at a rate that can be sent in real-time under the observed network conditions. This section defines two architectures for congestion control and bandwidth estimation for RTP over QUIC, but it does not mandate any specific congestion control algorithm to use. It is assumed that the congestion controller in use provides a pacing mechanism to determine when a packet can be sent to avoid bursts. The currently proposed congestion control algorithms for real-time communications provide such pacing mechanisms. The use of congestion controllers which don't provide a pacing mechanism is out of scope of this document.

• TODO:: Add considerations for bandwidth shares when a QUIC connection is shared between RTP and non-RTP streams?

Congestion Control at the QUIC layer If congestion control is to be applied at the transport layer, it is RECOMMENDED to configure the QUIC Implementation to use a delay-based real-time congestion control algorithm instead of a loss-based algorithm. The currently available delay-based congestion control algorithms depend on detailed arrival time feedback to estimate the current one-way delay between sender and receiver. Since QUIC does not provide arrival timestamps in its acknowledgments, the QUIC implementations of the sender and receiver MUST use an extension to add this information to QUICs acknowledgment frames, e.g. . If congestion control is done by the QUIC implementation, the application needs a mechanism to query the currently available bandwidth to adapt media codec configurations. The employed congestion controller of the QUIC connection SHOULD expose such an API to the application. If a current bandwidth estimation is not available from the QUIC congestion controller, the sender can either implement an alternative bandwidth estimation at the application layer as described in or a receiver can feedback the observed bandwidth through RTCP, e.g., using .

• Editor's note: An alternative to the hard requirement to use a timestamp extension could be to use RTCP, but that would mean, that an application has to negotiate RTCP congestion control feedback which would then have to be passed to the QUIC congestion controller.

• Editor's note: How can a QUIC connection be shared with non-RTP streams, when SCReAM/NADA/GCC is used as congestion controller? Can these algorithms be adapted to allow different streams including non-real-time streams? Do they even have to be adapted or should this just work?

Congestion Control at the Application Layer If an application cannot access a bandwidth estimation from the QUIC layer, or the QUIC implementation does not support a dealy-based, low-latency congestion control algorithm, it can alternatively implement a bandwidth estimation algorithm at the application layer. Calculating a bandwidth estimation at the application layer can be done using the same bandwidth estimation algorithms as described in (NADA, SCReAM). The bandwidth estimation algorithm typically needs some feedback on the transmission performance. This feedback can be collected following the guidelines in . If the application implements full congestion control rather than just a bandwidth estimation at the application layer using a congestion controller that satisfies the requirements of , and the connection is only used to send real-time media which is subject to the application layer congestion control, it is RECOMMENDED to disable any other congestion control that is possibly running at the QUIC layer. Disabling the additional congestion controllers helps to avoid any interference between the different congestion controllers.

SDP Signalling

• Editor's note: See also .

QUIC is a connection-based protocol that supports connectionless transmissions of DATAGRAM frames within an established connection. As noted above, demultiplexing DATAGRAMS intended for different purposes is up to the application using QUIC. There are several necessary steps to carry out jointly between the communicating peers to enable RTP over QUIC:

1. The protocol identifier for the m= lines MUST be "QUIC/RTP", combined as per with the respective audiovisual profile: for example, "QUIC/RTP/AVP".
2. The peers need to decide whether to establish a new QUIC connection or whether to re-use an existing one. In case of establishing a new connection, the initiator and the responder (client and server) need to be determined. Signaling for this step MUST follow on SDP attributes for connection-oriented media for the a=setup, a=connection, and a=fingerprint attributes. They MUST use the appropriate protocol identification as per 1.
3. The peers must provide a means for identifying RTP sessions carried in QUIC DATAGRAMS. To enable using a common transport connection for one, two, or more media sessions in the first place, the BUNDLE grouping framework MUST be used . All media sections belonging to a bundle group, except the first one, MUST set the port in the m= line to zero and MUST include the a=bundle-only attribute. For disambiguating different RTP session, a reference needs to be provided for each m= line to allow associating this specific media session with a flow identifier. This could be achieved following different approaches:
   • Simply reusing the a=extmap attribute and relying on RTP header extensions for demultiplexing different media packets carried in QUIC DATAGRAM frames.
   • Defining a variant or different flavor of the a=extmap attribute that binds media sessions to flow identifiers used in QUIC DATAGRAMS.
   • Editor's note: It is likely preferable to use multiplexing using QUIC DATAGRAM flow identifiers because this multiplexing mechanisms will also work across RTP and non-RTP media streams.
   In either case, the corresponding identifiers MUST be treated independently for each direction of transmission, so that an endpoint MAY choose its own identifies and only uses SDP to inform its peer which RTP sessions use which identifiers. To this end, SDP MUST be used to indicate the respective flow identifiers for RTP and RTCP of the different RTP sessions (for which we borrow inspiration from ).
4. The peers MUST agree, for each RTP session, whether or not to apply RTP/RTCP multiplexing. If multiplexing RTP and RTCP shall take place on the same flow identifier, this MUST be indicated using the attribute a=rtcp-mux.

A sample session setup offer (liberally borrowed and extended from and could look as follows:

SDP Offer m=video 0 QUIC/RTP/AVP 31 32 b=AS:1000 a=bundle-only a=mid:bar a=rtcp-mux a=rtpmap:31 H261/90000 a=rtpmap:32 MPV/90000 a=extmap:2 urn:ietf:params: ]]>

Signaling details to be worked out.

Discussion

Flow Identifier suggests to use flow identifiers to multiplex different streams on QUIC Datagrams, which is implemented in , but it is unclear how applications can combine RTP over QUIC with other data streams using the same QUIC connections. If the non-RTP data streams use the same flow identifies, too and the application can make sure, that flow identifiers are unique, there should be no problem. Flow identifiers could be problematic, if different specifications for RTP and non-RTP data streams over QUIC mandate different incompatible flow identifiers.

Impact of Connection Migration

Security Considerations TBD

IANA Considerations This document has no IANA actions.

References Normative 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 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. This document defines an extension to the QUIC transport protocol to add support for sending and receiving unreliable datagrams over a QUIC connection. This memo defines the Session Description Protocol (SDP). SDP is intended for describing multimedia sessions for the purposes of session announcement, session invitation, and other forms of multimedia session initiation. This document obsoletes RFC 4566. 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. This document describes loss detection and congestion control mechanisms for QUIC. Google LLC Cloudflare This document describes HTTP Datagrams, a convention for conveying multiplexed, potentially unreliable datagrams inside an HTTP connection. In HTTP/3, HTTP Datagrams can be conveyed natively using the QUIC DATAGRAM extension. When the QUIC DATAGRAM frame is unavailable or undesirable, they can be sent using the Capsule Protocol, a more general convention for conveying data in HTTP connections. HTTP Datagrams and the Capsule Protocol are intended for use by HTTP extensions, not applications. This document describes Network-Assisted Dynamic Adaptation (NADA), a novel congestion control scheme for interactive real-time media applications such as video conferencing. In the proposed scheme, the sender regulates its sending rate, based on either implicit or explicit congestion signaling, in a unified approach. The scheme can benefit from Explicit Congestion Notification (ECN) markings from network nodes. It also maintains consistent sender behavior in the absence of such markings by reacting to queuing delays and packet losses instead. This memo describes a rate adaptation algorithm for conversational media services such as interactive video. The solution conforms to the packet conservation principle and uses a hybrid loss-and-delay- based congestion control algorithm. The algorithm is evaluated over both simulated Internet bottleneck scenarios as well as in a Long Term Evolution (LTE) system simulator and is shown to achieve both low latency and high video throughput in these scenarios. This memo discusses issues that arise when multiplexing RTP data packets and RTP Control Protocol (RTCP) packets on a single UDP port. It updates RFC 3550 and RFC 3551 to describe when such multiplexing is and is not appropriate, and it explains how the Session Description Protocol (SDP) can be used to signal multiplexed sessions. [STANDARDS-TRACK] This memo defines a method for framing Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) packets onto connection-oriented transport (such as TCP). The memo also defines how session descriptions may specify RTP streams that use the framing method. [STANDARDS-TRACK] Magic Leap, Inc. Google LLC This document defines an extension to the QUIC transport protocol which supports reporting multiple packet receive timestamps using a new ACK_RECEIVE_TIMESTAMPS frame type. Discussion Venues This note is to be removed before publishing as an RFC. Discussion of this document takes place on the QUIC Working Group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/browse/quic/. Source for this draft and an issue tracker can be found at https://github.com/wcsmith/draft-quic-receive-ts. Private Octopus Inc. The TIMESTAMP frame can be added to Quic packets when one way delay measurements are useful. The timestamp is set to the number of microseconds from the beginning of the node's epoch to the time at which the packet is sent. The draft defines the "enable_timestamp" transport parameter for negotiating the use of this extension frame, and the TIMESTAMP frame. Fastly Google This document describes a QUIC extension for an endpoint to control its peer's delaying of acknowledgements. Note to Readers Discussion of this draft takes place on the QUIC working group mailing list (quic@ietf.org), which is archived at https://mailarchive.ietf.org/arch/search/?email_list=quic. Source code and issues list for this draft can be found at https://github.com/quicwg/ack-frequency. Working Group information can be found at https://github.com/quicwg. Real-time media streams that use RTP are, to some degree, resilient against packet losses. Receivers may use the base mechanisms of the Real-time Transport Control Protocol (RTCP) to report packet reception statistics and thus allow a sender to adapt its transmission behavior in the mid-term. This is the sole means for feedback and feedback-based error repair (besides a few codec-specific mechanisms). This document defines an extension to the Audio-visual Profile (AVP) that enables receivers to provide, statistically, more immediate feedback to the senders and thus allows for short-term adaptation and efficient feedback-based repair mechanisms to be implemented. This early feedback profile (AVPF) maintains the AVP bandwidth constraints for RTCP and preserves scalability to large groups. [STANDARDS-TRACK] This memo specifies how Explicit Congestion Notification (ECN) can be used with the Real-time Transport Protocol (RTP) running over UDP, using the RTP Control Protocol (RTCP) as a feedback mechanism. It defines a new RTCP Extended Report (XR) block for periodic ECN feedback, a new RTCP transport feedback message for timely reporting of congestion events, and a Session Traversal Utilities for NAT (STUN) extension used in the optional initialisation method using Interactive Connectivity Establishment (ICE). Signalling and procedures for negotiation of capabilities and initialisation methods are also defined. [STANDARDS-TRACK] An effective RTP congestion control algorithm requires more fine-grained feedback on packet loss, timing, and Explicit Congestion Notification (ECN) marks than is provided by the standard RTP Control Protocol (RTCP) Sender Report (SR) and Receiver Report (RR) packets. This document describes an RTCP feedback message intended to enable congestion control for interactive real-time traffic using RTP. The feedback message is designed for use with a sender-based congestion control algorithm, in which the receiver of an RTP flow sends back to the sender RTCP feedback packets containing the information the sender needs to perform congestion control. This document defines the Extended Report (XR) packet type for the RTP Control Protocol (RTCP), and defines how the use of XR packets can be signaled by an application if it employs the Session Description Protocol (SDP). XR packets are composed of report blocks, and seven block types are defined here. The purpose of the extended reporting format is to convey information that supplements the six statistics that are contained in the report blocks used by RTCP's Sender Report (SR) and Receiver Report (RR) packets. Some applications, such as multicast inference of network characteristics (MINC) or voice over IP (VoIP) monitoring, require other and more detailed statistics. In addition to the block types defined here, additional block types may be defined in the future by adhering to the framework that this document provides. The User Datagram Protocol (UDP) provides a minimal message-passing transport that has no inherent congestion control mechanisms. This document provides guidelines on the use of UDP for the designers of applications, tunnels, and other protocols that use UDP. Congestion control guidelines are a primary focus, but the document also provides guidance on other topics, including message sizes, reliability, checksums, middlebox traversal, the use of Explicit Congestion Notification (ECN), Differentiated Services Code Points (DSCPs), and ports. Because congestion control is critical to the stable operation of the Internet, applications and other protocols that choose to use UDP as an Internet transport must employ mechanisms to prevent congestion collapse and to establish some degree of fairness with concurrent traffic. They may also need to implement additional mechanisms, depending on how they use UDP. Some guidance is also applicable to the design of other protocols (e.g., protocols layered directly on IP or via IP-based tunnels), especially when these protocols do not themselves provide congestion control. This document obsoletes RFC 5405 and adds guidelines for multicast UDP usage. This document proposes an RTCP message for use in experimentally- deployed congestion control algorithms for RTP-based media flows. It also describes an absolute-value timestamp option for use in bandwidth estimatoin. This document specifies how to establish secure connection-oriented media transport sessions over the Transport Layer Security (TLS) protocol using the Session Description Protocol (SDP). It defines the SDP protocol identifier, 'TCP/TLS'. It also defines the syntax and semantics for an SDP 'fingerprint' attribute that identifies the certificate that will be presented for the TLS session. This mechanism allows media transport over TLS connections to be established securely, so long as the integrity of session descriptions is assured. This document obsoletes RFC 4572 by clarifying the usage of multiple fingerprints. This specification defines a new Session Description Protocol (SDP) Grouping Framework extension called 'BUNDLE'. The extension can be used with the SDP offer/answer mechanism to negotiate the usage of a single transport (5-tuple) for sending and receiving media described by multiple SDP media descriptions ("m=" sections). Such transport is referred to as a BUNDLE transport, and the media is referred to as bundled media. The "m=" sections that use the BUNDLE transport form a BUNDLE group. This specification defines a new RTP Control Protocol (RTCP) Source Description (SDES) item and a new RTP header extension. This specification updates RFCs 3264, 5888, and 7941. This document provides a general mechanism to use the header extension feature of RTP (the Real-time Transport Protocol). It provides the option to use a small number of small extensions in each RTP packet, where the universe of possible extensions is large and registration is decentralized. The actual extensions in use in a session are signaled in the setup information for that session. This document obsoletes RFC 5285. The Session Description Protocol (SDP) is used to describe the parameters of media streams used in multimedia sessions. When a session requires multiple ports, SDP assumes that these ports have consecutive numbers. However, when the session crosses a network address translation device that also uses port mapping, the ordering of ports can be destroyed by the translation. To handle this, we propose an extension attribute to SDP. Informative References QUIC is a UDP-based transport protocol for stream-orientated, congestion-controlled, secure, multiplexed data transfer. RTP carries real-time data between endpoints, and the accompanying control protocol RTCP allows monitoring and control of the transfer of such data. With RTP and RTCP being agnostic to the underlying transport protocol, it is possible to multiplex both the RTP and associated RTCP flows into a single QUIC connection to take advantage of QUIC features such as low-latency setup and strong TLS-based security. Tencent America LLC This document describes these new SDP "proto" attribute values: "QUIC", "QUIC/RTP/SAVP", "QUIC/RTP/AVPF", and "QUIC/RTP/SAVPF", and describes how SDP Offer/Answer can be used to set up an RTP connection using QUIC as a transport protocol. These proto values are necessary to allow the use of QUIC as an underlying transport protocol for applications such as SIP and WebRTC that commonly use SDP as a session signaling protocol to set up RTP connections.

Acknowledgments TODO acknowledge.