| Internet-Draft | RTP over QUIC (RoQ) | July 2023 |
| Ott, et al. | Expires 27 January 2024 | [Page] |
This document specifies a minimal mapping for encapsulating Real-time Transport Protocol (RTP) and RTP Control Protocol (RTCP) packets within the QUIC protocol. This mapping is called RTP over QUIC (RoQ). It also discusses how to leverage state from the QUIC implementation in the endpoints, in order to reduce the need to exchange RTCP packets and how to implement congestion control and rate adaptation without relying on RTCP feedback.¶
This note is to be removed before publishing as an RFC.¶
Discussion of this document takes place on the Audio/Video Transport Core Maintenance Working Group mailing list (avt@ietf.org), which is archived at https://mailarchive.ietf.org/arch/browse/avt/.¶
Source for this draft and an issue tracker can be found at https://github.com/mengelbart/rtp-over-quic-draft.¶
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 https://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 27 January 2024.¶
Copyright (c) 2023 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 (https://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 include Revised BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Revised BSD License.¶
This document specifies a minimal mapping for encapsulating Real-time Transport Protocol (RTP) [RFC3550] and RTP Control Protocol (RTCP) [RFC3550] packets within the QUIC protocol ([RFC9000]). This mapping is called RTP over QUIC (RoQ). It also discusses how to leverage state from the QUIC implementation in the endpoints, in order to reduce the need to exchange RTCP packets, and how to implement congestion control and rate adaptation without relying on RTCP feedback.¶
The Real-time Transport Protocol (RTP) [RFC3550] 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.¶
This specification describes an application usage of QUIC ([RFC9308]). As a baseline, the specification does not expect more than a standard QUIC implementation as defined in [RFC8999], [RFC9000], [RFC9001], and [RFC9002], providing a secure end-to-end transport that is also expected to work well through NATs and firewalls. Beyond this baseline, real-time applications can benefit from QUIC extensions such as unreliable QUIC datagrams [RFC9221], which provides additional desirable properties for real-time traffic (e.g., no unnecessary retransmissions, avoiding head-of-line blocking).¶
From time to time, someone asks the reasonable question, "why should anyone implement and deploy RoQ"? This reasonable question deserves a better answer than "because we can". Upon reflection, the following motivations seem useful to state.¶
The motivations in this section are in no particular order, and this reflects the reality that not all implementers and deployers would agree on "the most important motivations".¶
Although application-level mechanisms to encrypt RTP and RTCP payloads have existed since the introduction of Secure Real-time Transport Protocol (SRTP) [RFC3711], encryption of RTP and RTCP header fields and contributing sources has only been defined recently (in Cryptex [RFC9335], and both SRTP and Cryptex are optional capabilities for RTP.¶
This is in sharp contrast to "always-on" transport-level encryption in the QUIC protocol, using Transport Layer Security (TLS 1.3) as described in [RFC9001]. QUIC implementations always authenticate the entirety of each packet, and encrypt as much of each packet as is practical, even switching from "long headers", which expose more QUIC header fields needed to establish a connection, to "short headers", which only expose the absolute minimum QUIC header fields needed to identify the connection to the receiver, so that the QUIC payload is presented to the right QUIC application [RFC8999].¶
When RTP is carried directly over UDP, as is commonly done, the underlying UDP protocol provides no transport services beyond path multiplexing using UDP ports. All congestion avoidance behavior is up to the RTP application itself, and if anything goes wrong with the application resulting in an RTP sender failing to recognize that it is contributing to path congestion, the "worst case" response is to invoke RTP "circuit breaker" procedures [RFC8083], resulting in "ceasing transmission", as described in Section 4.5 of [RFC8083]. Because RTCP-based circuit breakers only detect long-lived congestion, a response based on these mechanisms will not happen quickly.¶
In contrast, when RTP is carried over QUIC, QUIC implementations maintain their own estimates of key transport parameters needed to detect and respond to possible congestion, and these are independent of any measurements RTP senders and receivers are maintaining. The result is that even if an RTP sender continues to "send", QUIC congestion avoidance procedures (for example, the procedures defined in [RFC9002]) will cause the RTP packets to be buffered and only placed on the network path as part of a response to detected loss. This happens without any action being requied on the part of RTP senders.¶
While the effect of QUIC's response to congestion means that some RTP packets will arrive at the receiver later than a user of the RTP flow might prefer, it is still preferable to "ceasing transmission" completely until the RTP sender has a reason to believe that restarting the flow will not result in congestion.¶
Moreover, when a single QUIC connection is used to multiplex both RTP-RTCP and non-RTP packets as described in Section 1.2.5, the QUIC connection will still be Internet-safe, with no coordination required.¶
RTP makes use of a large number of RTP-specific feedback mechanisms because when RTP is carried directly over UDP, there is no other way to receive feedback. Some of these mechanisms are specific to the type of media RTP is sending, but others can be mapped from underlying QUIC implementations that are using this feedback to perform rate adaptation for any QUIC connection, regardless of the application reflected in the QUIC STREAM [RFC9000] and DATAGRAM [RFC9221] frames. This is described in (much) more detail in Section 6 on rate adaptation, and in Section 7 on replacing RTCP and RTP header extensions with QUIC feedback.¶
One word of caution is in order - RTP implementations may rely on at least some minimal periodic RTCP feedback, in order to determine that an RTP flow is still active, and is not causing sustained congestion (as described in [RFC8083], but since this "periodicity" is measured in seconds, the impact of this "duplicate" feedback on path bandwidth utilization is likely close to zero.¶
The minimum Path MTU supported by conformant QUIC implementations is 1200 bytes [RFC9000], and in addition, QUIC implementations allow senders to use either DPLPMTUD ([RFC8899]) or PMTUD ([RFC1191], [RFC8201]) to determine the actual MTU size that the receiver and path between sender and receiver support, which can be even larger.¶
This is especially useful in certain conferencing topologies, where otherwise senders have no choice but to use the lowest path MTU for all conference participants, but even in point-to-point RTP sessions, this also allows senders to piggyback audio media in the same UDP packet as video media, for example, and also allows QUIC receivers to piggyback QUIC ACK frames on any QUIC frames being transmitted in the other direction.¶
In order to conserve ports, especially at NATs and Firewalls, this specification defines a flow identifier, so that multiple RTP flows, RTCP flows, and non-RTP flows can be distinguished even if they are carried on the same QUIC connection. This is described in more detail in Section 5.1.¶
Although there is much interest in multiplexing flows on a single QUIC connection as described in Section 1.2.5, QUIC also provides the capability of establishing and validating multiple paths for a single QUIC connection [RFC9000]. Once multiple paths have been validated, a sender can migrate from one path to another with no additional signaling, allowing an endpoint to move from one endpoint address to another without interruption, as long as only a single path is in active use at any point in time.¶
Connection migration may be desireable for a number of reasons, but to give one example, this allows a sender to distinguish between more costly cellular paths and less costly WiFi paths, with no action required from the application.¶
In addition to connection migration as described in Section 1.2.6, the capability of validating multiple paths for simultaneous active use is under active development in the IETF [I-D.draft-ietf-quic-multipath]. We don't discuss Multipath QUIC further in this document, because the specification hasn't been approved yet, but it's one example of ways that RTP, a mature protocol, can exploit new transport capabilities as they become available.¶
This document defines a mapping for RTP and RTCP over QUIC, called RoQ, and describes ways to reduce the amount of RTCP traffic by leveraging state information readily available within a QUIC endpoint. This document also describes different options for implementing congestion control and rate adaptation for RoQ.¶
This specification focuses on providing a secure encapsulation of RTP packets for transmission over QUIC. The expected usage is wherever RTP is used to carry media packets, allowing QUIC in place of other transport protocols such as TCP, UDP, SCTP, DTLS, etc. That is, we expect RoQ to be used in contexts in which a signaling protocol is used to announce or negotiate a media encapsulation and the associated transport parameters (such as IP address, port number). RoQ is not intended as a stand-alone media transport, although QUIC transport parameters could be statically configured.¶
The above implies that RoQ is targeted at peer-to-peer operation; but it may also be used in client-server-style settings, e.g., when talking to a conference server as described in RFC 7667 ([RFC7667]), or, if RoQ is used to replace RTSP ([RFC7826]), to a media server.¶
Moreover, this document describes how a QUIC implementation and its API can be extended to improve efficiency of the RoQ protocol operation.¶
RoQ does not impact the usage of RTP Audio Video Profiles (AVP) ([RFC3551]), or any RTP-based mechanisms, even though it may render some of them unnecessary, e.g., Secure Real-Time Transport Prococol (SRTP) ([RFC3711]) might not be needed, because end-to-end security is already provided by QUIC, and double encryption by QUIC and by SRTP might have more costs than benefits. Nor does RoQ limit the use of RTCP-based mechanisms, even though some information or functions obtained by using RTCP mechanisms may also be available from the underlying QUIC implementation by other means.¶
Between two (or more) endpoints, RoQ supports multiplexing multiple RTP-based media streams within a single QUIC connection and thus using a single (destination IP address, destination port number, source IP address, source port number, protocol) 5-tuple.. We note that multiple independent QUIC connections may be established in parallel using the same destination IP address, destination port number, source IP address, source port number, protocol) 5-tuple., e.g. to carry different media channels. These connections would be logically independent of one another.¶
This document does not attempt to enhance QUIC for real-time media or define a replacement for, or evolution of, RTP. Work to map other media transport protocols to QUIC is under way elsewhere in the IETF.¶
RoQ is designed for use with point-to-point connections, because QUIC itself is not defined for multicast operation. The scope of this document is limited to unicast RTP/RTCP, even though nothing would or should prevent its use in multicast setups once QUIC supports multicast.¶
RoQ does not define congestion control and rate adaptation algorithms for use with media. However, Section 6 discusses options for how congestion control and rate adaptation could be performed at the QUIC and/or at the RTP layer, and how information available at the QUIC layer could be exposed via an API for the benefit of RTP layer implementation.¶
Editor's note: Need to check whether Section 6 will also describe the QUIC interface that's being exposed, or if that ends up somewhere else in the document.¶
RoQ does not define prioritization mechanisms when handling different media as those would be dependent on the media themselves and their relationships. Prioritization is left to the application using RoQ.¶
This document does not cover signaling for session setup. SDP for RoQ is defined in separate documents such as [I-D.draft-dawkins-avtcore-sdp-rtp-quic], and can be carried in any signaling protocol that can carry SDP, including the Session Initiation Protocol (SIP) ([RFC3261]), Real-Time Protocols for Browser-Based Applications (RTCWeb) ([RFC8825]), or WebRTC-HTTP Ingestion Protocol (WHIP) ([I-D.draft-ietf-wish-whip]).¶
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 [RFC2119] [RFC8174] when, and only when, they appear in all capitals, as shown here.¶
Editor's note: the list of terms below will almost certainly grow in size as the specification matures.¶
The following terms are used:¶
A mechanism to limit the aggregate amount of data that has been sent over a path to a receiver, but has not been acknowledged by the receiver. This prevents a sender from overwhelming the capacity of a path between a sender and a receiver, causing some outstanding data to be discarded before the receiver can receive the data and acknowledge it to the sender.¶
Datagrams exist in UDP as well as in QUIC's unreliable datagram extension. If not explicitly noted differently, the term datagram in this document refers to a QUIC Datagram as defined in [RFC9221].¶
A congestion control algorithm that aims at keeping queues, and thus the latency, at intermediary network elements as short as possible. Delay-based congestion control algorithms use, for example, an increasing one-way delay as a signal of congestion.¶
A QUIC server or client that participates in an RoQ session.¶
A congestion control algorithm that uses packet loss, or an Explicit Congestion Notification (ECN) that is interpreted as loss (as in [RFC3168]), as a signal for congestion. Loss-based congestion control algorithms allow senders to send data on a path until packets are dropped by intermediary network elements, which the algorithm treats as a signal of congestion.¶
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.¶
A software component of an application's QUIC implementation that implements a congestion control algorithm.¶
A congestion control mechanism that helps a sender determine and adjust its sending rate, in order to maximize the amount of information that is sent to a receiver, without causing queues to build beyond a reasonable amount, causing "buffer bloat" and "jitter". Rate adapation is one way to accomplish congestion control for real-time media, especially when a sender has multiple media streams to the receiver, because the sum of all sending rates for media streams must not be high enough to cause congestion on the path these media streams share between sender and receiver.¶
An endpoint that receives media in RTP packets and may send or receive RTCP packets.¶
A software component of an application's RTP implementation that implements a congestion control algorithm.¶
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 Section 1.3 of [RFC9000] to illustrate the order and size of fields.¶
This document introduces a mapping of the Real-time Transport Protocol (RTP) to the QUIC transport protocol. RoQ allows the use of QUIC streams and QUIC datagrams to transport real-time data, and thus, the QUIC implementation MUST support QUIC's 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.¶
[RFC3550] specifies that RTP sessions need to be transmitted on different transport addresses to allow multiplexing between them. RoQ 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 Section 2.1 of [I-D.draft-ietf-masque-h3-datagram]. RoQ uses a flow identifier to replace the network address and port number to multiplex many RTP sessions over the same QUIC connection.¶
A rate adaptation algorithm can be plugged in to adapt the media bitrate to the available bandwidth. This document does not mandate any specific rate adaptation algorithm, because the desired response to congestion can be application and codec-specific. For example, adjusting quantization in response to congestion may work well in many cases, but if what's being shared is video that includes text, maintaining readability is important.¶
As of this writing, the IETF has produced two Experimental-track rate adaptation specifications, Network-Assisted Dynamic Adaptation (NADA) [RFC8698] and Self-Clocked Rate Adaptation for Multimedia (SCReAM) [RFC8298]. These rate adaptation 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 rate adaptation algorithm to avoid the additional overhead of the RTCP stream. This is discussed in more detail in Section 7.¶
RoQ only supports some of the RTP topologies described in [RFC7667]. Most notably, due to QUIC [RFC9000] being a purely IP unicast protocol at the time of writing, RoQ cannot be used as a transport protocol for any of the paths that rely on IP multicast in several multicast topologies (e.g., Topo-ASM, Topo-SSM, Topo-SSM-RAMS).¶
Some "multicast topologies" can include IP unicast paths (e.g., Topo-SSM, Topo-SSM-RAMS). In these cases, the unicast paths can use RoQ.¶
RTP supports different types of translators and mixers. Whenever a middlebox such as a translator or a mixer needs to access the content of RTP/RTCP-packets, the QUIC connection has to be terminated at that middlebox.¶
RoQ streams (see Section 5.2) can support much larger RTP packet sizes than other transport protocols such as UDP can, which can lead to problems with transport translators which translate from RoQ to RTP over a different transport protocol. A similar problem can occur if a translator needs to translate from RTP over UDP to RoQ datagrams, where the MTU of a QUIC datagram may be smaller than the MTU of a UDP datagram. In both cases, the translator may need to rewrite the RTP packets to fit into the smaller MTU of the other protocol. Such a translator may need codec-specific knowledge to packetize the payload of the incoming RTP packets in smaller RTP packets.¶
Additional details are provided in the following table.¶
| RFC 7667 Section | Shortcut Name | RTP over QUIC? | Comments |
|---|---|---|---|
| 3.1 | Topo-Point-to-Point | yes | |
| 3.2.1.1 | Topo-PtP-Relay | yes | Note-NAT |
| 3.2.1.2 | Topo-Trn-Translator | yes | Note-MTU Note-SEC |
| 3.2.1.3 | Topo-Media-Translator | yes | Note-MTU |
| 3.2.2 | Topo-Back-To-Back | yes | Note-SEC Note-MTU Note-MCast |
| 3.3.1 | Topo-ASM | no | Note-MCast |
| 3.3.2 | Topo-SSM | partly | Note-MCast Note-UCast-MCast |
| 3.3.3 | Topo-SSM-RAMS | partly | Note-MCast Note-MCast-UCast |
| 3.4 | Topo-Mesh | yes | Note-MCast |
| 3.5.1 | Topo-PtM-Trn-Translator | possibly | Note-MCast Note-MTU Note-Topo-PtM-Trn-Translator |
| 3.6 | Topo-Mixer | possibly | Note-MCast Note-Topo-Mixer |
| 3.6.1 | Media-Mixing-Mixer | partly | Note-Topo-Mixer |
| 3.6.2 | Media-Switching-Mixer | partly | Note-Topo-Mixer |
| 3.7 | Selective Forwarding Middlebox | yes | Note-MCast Note-Topo-Mixer |
| 3.8 | Topo-Video-switch-MCU | yes | Note-MTU Note-MCast Note-Topo-Mixer |
| 3.9 | Topo-RTCP-terminating-MCU | yes | Note-MTU Note-MCast Note-Topo-Mixer |
| 3.10 | Topo-Split-Terminal | yes | Note-MCast |
| 3.11 | Topo-Asymmetric | Possibly | Note-Warn, Note-MCast, Note-MTU |
Supported, but may require MTU adaptation.¶
Note that RoQ provides mandatory security, and other RTP transports do not. Section 10 describes strategies to prevent the inadvertent disclosure of RTP sessions to unintended third parties.¶
The topology refers to a Distribution Source, which receives and relays RTP from a number of different media senders via unicast before relaying it to the receivers via multicast. QUIC can be used between the senders and the Distribution Source.¶
The topology refers to a Burst Source or Retransmission Source, which retransmits RTP to receivers via unicast. QUIC can be used between the Retransmission Source and the receivers.¶
Supports unicast paths between RTP sources and translators.¶
Supports unicast paths between RTP senders and mixers.¶
Quote from [RFC7667]: This topology is so problematic and it is so easy to get the RTCP processing wrong, that it is NOT RECOMMENDED to implement this topology.¶
QUIC requires the use of ALPN [RFC7301] tokens during connection setup. RoQ uses "rtp-mux-quic" as ALPN token in the TLS handshake (see also Section 11.¶
Note that the use of a given RTP profile is not reflected in the ALPN token even though it could be considered part of the application usage. This is simply because different RTP sessions, which may use different RTP profiles, may be carried within the same QUIC connection.¶
Editor's note: "rtp-mux-quic" indicates that RTP and other protocols may be multiplexed on the same QUIC connection using a flow identifier as described in Section 5. Applications are responsible for negotiation of protocols in use by appropriate use of a signaling protocol such as SDP.¶
Editor's note: This implies that applications cannot use RoQ as specified in this document over WebTransport.¶
RFC Editor's note: Please remove this section prior to publication of a final version of this document.¶
RoQ uses the token "rtp-mux-quic" to identify itself in ALPN.¶
Only implementations of the final, published RFC can identify themselves as "rtp-mux-quic". Until such an RFC exists, implementations MUST NOT identify themselves using this string.¶
Implementations of draft versions of the protocol MUST add the string "-" and the corresponding draft number to the identifier. For example, draft-ietf-avtcore-rtp-over-quic-04 is identified using the string "rtp-mux-quic-04".¶
Non-compatible experiments that are based on these draft versions MUST append the string "-" and an experiment name to the identifier.¶
This section describes the encapsulation of RTP/RTCP packets in QUIC.¶
QUIC supports two transport methods: streams [RFC9000] and datagrams [RFC9221]. This document specifies mappings of RTP to both of the transport modes. Senders MAY combine both modes by sending some RTP/RTCP packets over the same or different QUIC streams and others in QUIC datagrams.¶
Section 5.1 introduces a multiplexing mechanism that supports multiplexing RTP, RTCP, and, with some constraints, other non-RTP protocols. Section 5.2 and Section 5.3 explain the specifics of mapping RTP to QUIC streams and QUIC datagrams, respectively.¶
RoQ uses flow identifiers to multiplex different RTP, RTCP, and non-RTP data streams on a single QUIC connection. A flow identifier is a QUIC variable-length integer as described in Section 16 of [RFC9000]. Each flow identifier is associated with a stream of RTP packets, RTCP packets, or a data stream of a non-RTP protocol.¶
In a QUIC connection using the ALPN token defined in Section 4, every QUIC datagram and every QUIC stream MUST start with a flow identifier. A peer MUST NOT send any data in a datagram or stream that is not associated with the flow identifier which started the datagram or stream.¶
RTP and RTCP packets of different RTP sessions MUST use distinct flow identifiers. If peers wish to send multiple types of media in a single RTP session, they MAY do so by following [RFC8860].¶
A single RTP session MAY be associated with one or two flow identifiers. Thus, it is possible to send RTP and RTCP packets belonging to the same session using different flow identifiers. RTP and RTCP packets of a single RTP session MAY use the same flow identifier (following the procedures defined in [RFC5761], or they MAY use different flow identifiers.¶
The association between flow identifiers and data streams MUST be negotiated using appropriate signaling. Applications MAY send data using flow identifiers not associated with any RTP or RTCP stream. If a receiver cannot associate a flow identifier with any RTP/RTCP or non-RTP stream, it MAY drop the data stream.¶
There are different use cases for sharing the same QUIC connection between RTP and non-RTP data streams. Peers might use the same connection to exchange signaling messages or exchange data while sending and receiving media streams. The semantics of non-RTP datagrams or streams are not in the scope of this document. Peers MAY use any protocol on top of the encapsulation described in this document.¶
Flow identifiers introduce some overhead in addition to the header overhead of RTP/RTCP and QUIC. QUIC variable-length integers require between one and eight bytes depending on the number expressed. Thus, it is advisable to use low numbers first and only use higher ones if necessary due to an increased number of flows.¶
To send RTP/RTCP packets over QUIC streams, a sender MUST open a new unidirectional QUIC stream. Streams are unidirectional because there is no synchronous relationship between sent and received RTP/RTCP packets. A RoQ sender MAY open new QUIC streams for different packets using the same flow identifier, for example, to avoid head-of-line blocking.¶
A receiver MUST be prepared to receive RTP packets on any number of QUIC streams (subject to its limit on parallel open streams) and SHOULD not make assumptions which RTP sequence numbers are carried in which streams.¶
Note: A sender may or may not decide to discontinue using a lower stream number after starting packet transmission on a higher stream number.¶
Figure 1 shows the encapsulation format for RoQ Streams.¶
Payload {
Flow Identifier (i),
RTP/RTCP Payload(..) ...,
}
Flow identifier to demultiplex different data flows on the same QUIC connection.¶
The payload in a QUIC stream starts with the flow identifier followed by one or more RTP/RTCP payloads. All RTP/RTCP payloads sent on a stream MUST belong to the RTP session with the same flow identifier.¶
Each payload begins with a length field indicating the length of the RTP/RTCP packet, followed by the packet itself, see Figure 2.¶
RTP/RTCP Payload {
Length(i),
RTP/RTCP Packet(..),
}
A QUIC variable length integer (see Section 16 of [RFC9000]) describing the length of the following RTP/RTCP packets in bytes.¶
The RTP/RTCP packet to transmit.¶
QUIC uses RESET_STREAM and STOP_SENDING frames to terminate the sending part of a stream and to request termination of an incoming stream by the sending peer respectively.¶
A RoQ sender MAY use RESET_STREAM if it knows that a packet, which was not yet successfully and completely transmitted, is no longer needed.¶
A RoQ receiver that is no longer interested in reading a certain partition of the media stream MAY signal this to the sending peer using a STOP_SENDING frame.¶
When a RoQ sender receives a STOP_SENDING frame for the last open stream available to send RTP/RTCP-data, the RoQ sender MUST open one or more new QUIC streams before sending new media frames. Any media frame that has already been sent on the QUIC stream that received the STOP_SENDING frame, MUST NOT be sent again on the new QUIC stream(s).¶
Note that an RTP receiver cannot request a reset of only a particular media frame because the sending QUIC implementation might already have sent data for the following frame on the same stream. In that case, STOP_SENDING and the following RESET_STREAM would also drop the following media frame and thus lead to unintentionally skipping one or more frames.¶
Editor's note: A receiver cannot cancel a certain frame but still receive retransmissions for a frame the was following on the same stream using STOP_SENDING, because STOP_SENDING does not include an offset which would allow signaling where retransmissions should continue.¶
Editor's note: Instead of using RESET_STREAM and STOP_SENDING frames, RoQ senders and receivers might benefit from negotiating the use of the QUIC extensions defined in [I-D.draft-ietf-quic-reliable-stream-reset-01] and [I-D.draft-thomson-quic-enough-00]. These extensions provide two new frame types, the CLOSE_STREAM and the ENOUGH frame, equivalent to RESET_STREAM and STOP_SENDING, respectively, but with an additional offset. The offset indicates the point to which data will be reliably retransmitted, while everything following might be dropped. Using CLOSE_STREAM and ENOUGH instead of RESET_STREAM and STOP_SENDING could prevent accidentally stopping retransmissions for preceding frames.¶
A translator that translates between two endpoints, both connected via QUIC, MUST forward RESET_STREAM frames received from one end to the other unless it forwards the RTP packets on QUIC datagrams.¶
Large RTP packets sent on a stream will be fragmented into smaller QUIC frames. The QUIC frames are transmitted reliably and in order such that a receiving application can read a complete RTP packet from the stream as long as the stream is not closed with a RESET_STREAM frame. No retransmission has to be implemented by the application since QUIC frames lost in transit are retransmitted by QUIC.¶
Opening new streams for new packets MAY implicitly limit the number of packets concurrently in transit because the QUIC receiver provides an upper bound of parallel streams, which it can update using QUIC MAX_STREAMS frames. The number of packets that have to be transmitted concurrently depends on several factors, such as the number of RTP streams within a QUIC connection, the bitrate of the media streams, and the maximum acceptable transmission delay of a given packet. Receivers are responsible for providing senders enough credit to open new streams for new packets anytime. As an example, consider a conference scenario with 20 participants. Each participant receives audio and video streams of every other participant from a central server. If the sender opens a new QUIC stream for every frame at 30 frames per second video and 50 frames per second audio, it will open 1520 new QUIC streams per second. A receiver must provide at least that many credits for opening new unidirectional streams to the server every second. In addition, the receiver should also consider the requirements of protocols into account that are multiplexed with RTP, including RTCP and data streams. These considerations may also be relevant when implementing signaling since it may be necessary to inform the receiver about how fast and how much stream credits it will have to provide to the media-sending peer.¶
Senders can also transmit RTP packets in QUIC datagrams. QUIC datagrams are an extension to QUIC described in [RFC9221]. QUIC datagrams preserve frame boundaries. Thus, a single RTP packet can be mapped to a single QUIC datagram without additional framing. Senders SHOULD consider the header overhead associated with QUIC datagrams and ensure that the RTP/RTCP packets, including their payloads, flow identifier, QUIC, and IP headers, will fit into path MTU.¶
Figure 3 shows the encapsulation format for RoQ Datagrams.¶
Payload {
Flow Identifier (i),
RTP/RTCP Packet (..),
}
Flow identifier to demultiplex different data flows on the same QUIC connection.¶
The RTP/RTCP packet to transmit.¶
RoQ senders need to be aware that QUIC uses the concept of QUIC frames. Different kinds of QUIC frames are used for different application and control data types. A single QUIC packet can contain more than one QUIC frame, including, for example, QUIC stream or datagram frames carrying application data and acknowledgement frames carrying QUIC acknowledgements, as long as the overall size fits into the MTU. One implication is that the number of packets a QUIC stack transmits depends on whether it can fit acknowledgement and datagram frames in the same QUIC packet. Suppose the application creates many datagram frames that fill up the QUIC packet. In that case, the QUIC stack might have to create additional packets for acknowledgement- (and possibly other control-) frames. The additional overhead could, in some cases, be reduced if the application creates smaller RTP packets, such that the resulting datagram frame can fit into a QUIC packet that can also carry acknowledgement frames.¶
Since QUIC datagrams are not retransmitted on loss (see also Section 7.3 for loss signaling), if an application wishes to retransmit lost RTP packets, the retransmission has to be implemented by the application. RTP retransmissions can be done in the same RTP session or a separate RTP session [RFC4588] and the flow identifier MUST be set to the flow identifier of the RTP session in which the retransmission happens.¶
Like any other application on the internet, RoQ applications need a mechanism to perform congestion control to avoid overloading the network. While any generic congestion controller can protect the network, this document takes advantage of the opportunity to use rate adaptation mechanisms that are designed to provide superior user experiences for real-time media applications.¶
A wide variety of rate adaptation algorithms for real-time media have been developed (for example, "Google Congestion Controller" [I-D.draft-ietf-rmcat-gcc]). The IETF has defined two algorithms in two Experimental RFCs (e.g. SCReAM [RFC8298] and NADA [RFC8698]). These rate adaptation algorithms for RTP are specifically tailored for real-time transmissions at low latencies, but this section would apply to any rate adaptation algorithm that meets the requirements described in "Congestion Control Requirements for Interactive Real-Time Media" [RFC8836].¶
This document defines two architectures for congestion control and bandwidth estimation for RoQ, depending on whether most rate adaptation is performed within a QUIC implementation at the transport layer, as described in Section 6.1, or within an RTP application layer, as described in Section 6.2, but this document does not mandate any specific congestion control or rate adaptation algorithm for either QUIC or RTP.¶
This document also gives guidance about avoiding problems with "nested" congestion controllers, in Section 6.3.¶
This document also discusses congestion control implications of using shared or multiple separate QUIC connections to send and receive multiple independent data streams, in Section 6.4.¶
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 (e.g. SCReAM and NADA) provide such pacing mechanisms. The use of congestion controllers which don't provide a pacing mechanism is out of scope of this document.¶
QUIC is a congestion controlled transport protocol. Senders are required to employ some form of congestion control. The default congestion control specified for QUIC in [RFC9002] is similar to TCP NewReno [RFC6582], but senders are free to choose any congestion control algorithm as long as they follow the guidelines specified in Section 3 of [RFC8085], and QUIC implementors make use of this freedom.¶
If a QUIC implementation is to perform rate adaptation in a way that accommodates real-time media, one way for the implementation to recognize that it is carrying real-time media is to be explicitly told that this is the case. This document defines a new "TLS Application-Layer Protocol Negotiation (ALPN) Protocol ID", as described in Section 4, that a QUIC implementation can use as a signal to choose a real-time media-centric rate controller, but this is not required for ROQ deployments.¶
If congestion control is to be applied at the transport layer, it is RECOMMENDED that the QUIC Implementation uses a congestion controller that keeps queueing delays short to keep the transmission latency for RTP and RTCP packets as low as possible, such as the IETF-defined SCReAM [RFC8298] and NADA [RFC8698] algorithms.¶
Many low latency congestion control algorithms depend on detailed arrival time feedback to estimate the current one-way delay between sender and receiver. QUIC does not provide arrival timestamps in its acknowledgments. The QUIC implementations of the sender and receiver can use an extension to add this information to QUICs acknowledgment frames, e.g. [I-D.draft-smith-quic-receive-ts] or [I-D.draft-huitema-quic-ts].¶
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 estimate is not available from the QUIC congestion controller, the sender can either implement an alternative bandwidth estimation at the application layer as described in Section 6.2 or a receiver can feedback the observed bandwidth through RTCP, e.g., using [I-D.draft-alvestrand-rmcat-remb].¶
RTP itself does not specify a congestion control algorithm, but [RFC8888] defines an RTCP feedback message intended to enable rate adaptation for interactive real-time traffic using RTP, and successful rate adaptation will accomplish congestion control as well.¶
The rate adaptation algorithms for RTP are specifically tailored for real-time
transmissions at low latencies, as described in Section 6. The
available rate adaptation algorithms 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.¶
If an application cannot access a bandwidth estimation from the QUIC layer, or the QUIC implementation does not support a delay-based, low-latency congestion control algorithm, the application 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 Section 6 (NADA, SCReAM). The bandwidth estimation algorithm typically needs some feedback on the transmission performance. This feedback can be collected following the guidelines in Section 7.¶
Because QUIC is a congestion-controlled transport, as described in Section 6.1, and RTP applications can also perform congestion control and rate adaptation, as described in Section 6.2, implementers should be aware of the possibility that these "nested" congestion control loops, where both controllers are managing rate adaptation for the same packet stream independently, may deliver problematic performance. Because this document is describing a specific case (media transport), we can provide some guidance to avoid the worst possible problems.¶
(BBR) "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."¶
Because RTP has so often used UDP as its underlying transport protocol, and receiving little or no feedback from UTP, RTP implementations rely on feedback from the RTP Control Protocol (RTCP) so that RTP senders and receivers can exchange control information to monitor connection statistics and to identify and synchronize streams.¶
Because QUIC provides its own transport-level feedback, at least some RTP transport level feedback can be replaced with current QUIC feedback [rfc9000]. In adition, RTP-level feedback that is not available in QUIC by default can be replaced with generally useful QUIC extensions. Examples of these extentions include:¶
When statistics contained in RTCP packets are also available from QUIC, or can be derived from statistics available from QUIC, it is desireable to provide these statistics at only one protocol layer. This avoids consumption of bandwidth to deliver duplicated control information. Because QUIC relies on certain frames being sent, it is not possible to supress QUIC signaling in favor of RTCP signaling, so if bandwidth is to be conserved, this must be accomplished by surpressing RTCP signaling in favor of QUIC signalling.¶
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, and can make use of new QUIC extensions that become available over time.¶
Most statements about "QUIC" in Section 7 are applicable to both RTP encapsulated in QUIC streams and RTP encapsulated in QUIC datagrams. The differences are described in Section 7.1 and Section 7.2.¶
Editor's Note: Additional discussion of bandwidth minimization could go in this section, or in an earlier proposed section on motivations for defining and deploying RoQ.¶
While RoQ places no restrictions on applications sending RTCP, this document assumes that the reason an implementor chooses to support RoQ is to obtain benefits beyond what's available when RTP uses UDP as its underlying transport layer. It is RECOMMENDED to expose relevant information from the QUIC layer to the application instead of exchanging additional RTCP packets, where applicable.¶
Section 7.3 and Section 7.4 discuss what information can be exposed from the QUIC connection layer to reduce the RTCP overhead.¶
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 using RoQ.¶
A more complete analysis of RTCP Control Packet Types (in Section 7.5), Generic RTP Feedback (RTPFB) (in Section 7.6), Payload-specific RTP Feedback (PSFB) (in Section 7.7), Extended Reports (in Section 7.8), and RTP Header Extensions (in Section 7.9), including the information that cannot be mapped from QUIC.¶
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.¶
When QUIC Streams are used, the application should be aware that the direct mapping proposed below may not reflect the real characteristics of the network. RTP packet loss can seem lower than actual packet loss due to QUIC's automatic retransmissions. Similarly, timing information might be incorrect due to retransmissions.¶
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.¶
Considerations for mapping QUIC feedback into Receiver Reports (PT=201,
Name=RR, [RFC3550]) are:¶
Considerations for mapping QUIC feedback into Negative Acknowledgments
(PT=205, FMT=1, Name=Generic NACK, [RFC4585]) are:¶
Considerations for mapping QUIC feedback into ECN Feedback (PT=205, FMT=8,
Name=RTCP-ECN-FB, [RFC6679]) are:¶
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 QUIC connection supports
ECN, the reporting of ECN counts SHOULD be done using QUIC acknowledgments,
rather than RTCP ECN feedback reports.¶
Considerations for mapping QUIC feedback into Congestion Control Feedback
(PT=205, FMT=11, Name=CCFB, [RFC8888]) are:¶
Considerations for mapping QUIC feedback into Extended Reports (PT=207,
Name=XR, [RFC3611]) are:¶
While Section 7.3.1 presented some RTCP packets that can be replaced by QUIC features, QUIC cannot replace all of the defined RTCP packet types. This mostly affects RTCP packet types which carry control information that is to be interpreted by the RTP application layer, rather than the underlying transport protocol itself.¶
PT=200, Name=SR, [RFC3550]) are similar to Receiver
Reports, as described in Section 7.3.1. They are sent by media senders and
additionally contain an 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. Nor can a QUIC receiver calculate the
packet or octet counts, since it does not know about lost datagrams. Thus,
sender reports are required in RoQ to synchronize streams at the
receiver. The sender reports SHOULD not contain any receiver report blocks, as
the information can be inferred from the QUIC transport as explained in
Section 7.3.1.¶
In addition to carrying transmission statistics, RTCP packets can contain application layer control information, that cannot directly be mapped to QUIC. Examples of this information may include:¶
PT=202, Name=SDES), Bye (PT=203, Name=BYE) and
Application (PT=204, Name=APP) packet types from [RFC3550], or¶
PT=206) defined in
[RFC4585], used to control the codec behavior of the sender.¶
Since QUIC does not provide any kind of application layer control messaging, QUIC feedback cannot be mapped into these RTCP packet types. If the RTP application needs this information, the RTCP packet types are used in the same way as they would be used over any other transport protocol.¶
Several but not all of these control packets and their attributes can be mapped from QUIC, as described in Section 7.3.1. "Mappable from QUIC" has one of three values: "yes", "QUIC extension required", and "no".¶
| Name | Shortcut | PT | Defining Document | Mappable from QUIC | Comments |
|---|---|---|---|---|---|
| SMPTE time-code mapping | SMPTETC | 194 | [RFC5484] | no | |
| Extended inter-arrival jitter report | IJ | 195 | [RFC5450] | QUIC extension required | IJ was introduced to improve jitter reports when RTP packets are not sent at the time indicated by their RTP timestamp. Jitter can be calculated using QUIC timestamps, because QUIC timestamps are added when the QUIC packet is actually sent. |
| Sender Reports | SR | 200 | [RFC3550] | partly | - NTP timestamps cannot be replaced by QUIC and are required for synchronization (but see note below) - packet and octet counts cannot be provided by QUIC - see below for RRs contained in SRs |
| Receiver Reports | RR | 201 | [RFC3550] | possibly | - Fraction Lost/Cumulative Lost/Highest Sequence Number received can directly be inferred from QUIC ACKs - Interarrival Jitter/Last SR need a QUIC timestamp extension. Using QUIC ts is slightly different because it ignores transmission offsets from RTP timestamps, but that seems like a good thing (see IJ above) |
| Source description | SDES | 202 | [RFC3550] | no | |
| Goodbye | BYE | 203 | [RFC3550] | possibly | using QUIC CONNECTION_CLOSE frame? |
| Application-defined | APP | 204 | [RFC3550] | no | |
| Generic RTP Feedback | RTPFB | 205 | [RFC4585] | partly | see table below |
| Payload-specific | PSFB | 205 | [RFC4585] | see table below | |
| extended report | XR | 207 | [RFC3611] | partly | see table below |
| AVB RTCP packet | AVB | ||||
| Receiver Summary Information | RSI | 209 | [RFC5760] | ||
| Port Mapping | TOKEN | 210 | [RFC6284] | no? | |
| IDMS Settings | IDMS | 211 | [RFC7272] | no | |
| Reporting Group Reporting Sources | RGRS | 212 | [RFC8861] | ||
| Splicing Notification Message | SNM | 213 | [RFC8286] | no |
| Name | Long Name | Document | Mappable from QUIC | Comments |
|---|---|---|---|---|
| Generic NACK | Generic negative acknowledgement | [RFC4585] | possibly | Using QUIC ACKs |
| TMMBR | Temporary Maximum Media Stream Bit Rate Request | [RFC5104] | no | |
| TMMBN | Temporary Maximum Media Stream Bit Rate Notification | [RFC5104] | no | |
| RTCP-SR-REQ | RTCP Rapid Resynchronisation Request | [RFC6051] | no | |
| RAMS | Rapid Acquisition of Multicast Sessions | [RFC6285] | no | |
| TLLEI | Transport-Layer Third-Party Loss Early Indication | [RFC6642] | no? | no way of telling QUIC peer "don't ask for retransmission", but QUIC would not ask that anyway, only RTCP NACK? |
| RTCP-ECN-FB | RTCP ECN Feedback | [RFC6679] | partly | QUIC does not provide info about duplicates |
| PAUSE-RESUME | Media Pause/Resume | [RFC7728] | no | |
| DBI | Delay Budget Information (DBI) | [_3GPP-TS-26.114] | ||
| CCFB | RTP Congestion Control Feedback | [RFC8888] | possibly | - ECN/ACK natively in QUIC - timestamps require QUIC timestamp extension |
Because QUIC is a generic transport protocol, QUIC feedback cannot replace the following Payload-specific RTP Feedback (PSFB) feedback.¶
| Name | Long Name | Document |
|---|---|---|
| PLI | Picture Loss Indication | [RFC4585] |
| SLI | Slice Loss Indication | [RFC4585] |
| RPSI | Reference Picture Selection Indication | [RFC4585] |
| FIR | Full Intra Request Command | [RFC5104] |
| TSTR | Temporal-Spatial Trade-off Request | [RFC5104] |
| TSTN | Temporal-Spatial Trade-off Notification | [RFC5104] |
| VBCM | Video Back Channel Message | [RFC5104] |
| PSLEI | Payload-Specific Third-Party Loss Early Indication | [RFC6642] |
| ROI | Video region-of-interest (ROI) | [_3GPP-TS-26.114] |
| LRR | Layer Refresh Request Command | [I-D.draft-ietf-avtext-lrr-07] |
| AFB | Application Layer Feedback | [RFC4585] |
| TSRR | Temporal-Spatial Resolution Request | [I-D.draft-ietf-avtcore-rtcp-green-metadata] |
| TSRN | Temporal-Spatial Resolution Notification | [I-D.draft-ietf-avtcore-rtcp-green-metadata] |
| Name | Document | Mappable from QUIC | Comments |
|---|---|---|---|
| Loss RLE Report Block | [RFC3611] | yes | QUIC ACKs |
| Duplicate RLE Report Block | [RFC3611] | no | |
| Packet Receipt Times Report Block | [RFC3611] | possibly | - Could be replaced by QUIC timestamps. - Would not use RTP timestamps. - Only if QUIC timestamps for every packet is included (e.g. draft-smith-quic-receive-ts but not draft-huitema-quic-ts) |
| Receiver Reference Time Report Block | [RFC3611] | possibly | QUIC timestamps |
| DLRR Report Block | [RFC3611] | possibly | QUIC ACKs and QUIC timestamps. In general, however, it seems to be useful only to calculate RTT, which is natively available in QUIC. |
| Statistics Summary Report Block | [RFC3611] | partly | - loss and jitter as described in other reports above. - TTL/HL/Duplicates not available in QUIC |
| VoIP Metrics Report Block | [RFC3611] | no | as in other reports above, only loss and RTT available |
| RTCP XR | [RFC5093] | no | |
| Texas Instruments Extended VoIP Quality Block | |||
| Post-repair Loss RLE Report Block | [RFC5725] | no | |
| Multicast Acquisition Report Block | [RFC6332] | no | |
| IDMS Report Block | [RFC7272] | no | |
| ECN Summary Report | [RFC6679] | partly | QUIC does not provide info about duplicates |
| Measurement Information Block | [RFC6776] | no | |
| Packet Delay Variation Metrics Block | [RFC6798] | no | QUIC timestamps may be used to achieve the same goal? |
| Delay Metrics Block | [RFC6843] | no | QUIC has RTT and can provide timestamps for one-way delay, but no way of informing peers about end-to-end statistics when QUIC is only used on one segment of the path. |
| Burst/Gap Loss Summary Statistics Block | [RFC7004] | QUIC ACKs? | |
| Burst/Gap Discard Summary Statistics Block | [RFC7004] | no | |
| Frame Impairment Statistics Summary | [RFC7004] | no | |
| Burst/Gap Loss Metrics Block | [RFC6958] | QUIC ACKs? | |
| Burst/Gap Discard Metrics Block | [RFC7003] | no | |
| MPEG2 Transport Stream PSI-Independent Decodability Statistics Metrics Block | [RFC6990] | no | |
| De-Jitter Buffer Metrics Block | [RFC7005] | no | |
| Discard Count Metrics Block | [RFC7002] | no | |
| DRLE (Discard RLE Report) | [RFC7097] | no | |
| BDR (Bytes Discarded Report) | [RFC7243] | no | |
| RFISD (RTP Flows Initial Synchronization Delay) | [RFC7244] | no | |
| RFSO (RTP Flows Synchronization Offset Metrics Block) | [RFC7244] | no | |
| MOS Metrics Block | [RFC7266] | no | |
| LCB (Loss Concealment Metrics Block) | [RFC7294], Section 4.1 | no | |
| CSB (Concealed Seconds Metrics Block) | [RFC7294], Section 4.1 | no | |
| MPEG2 Transport Stream PSI Decodability Statistics Metrics Block | [RFC7380] | no | |
| Post-Repair Loss Count Metrics Report Block | [RFC7509] | no | |
| Video Loss Concealment Metric Report Block | [RFC7867] | no | |
| Independent Burst/Gap Discard Metrics Block | [RFC8015] | no |
Like the payload-specific feedback packets, QUIC cannot directly replace the control information in the following header extensions. RoQ does not place restrictions on sending any RTP header extensions. However, some extensions, such as Transmission Time offsets [RFC5450] are used to improve network jitter calculation, which can be done in QUIC if a timestamp extension is used.¶
| Extension URI | Description | Reference | QUIC |
|---|---|---|---|
| urn:ietf:params:rtp-hdrext:toffset | Transmission Time offsets | [RFC5450] | no |
| urn:ietf:params:rtp-hdrext:ssrc-audio-level | Audio Level | [RFC6464] | no |
| urn:ietf:params:rtp-hdrext:splicing-interval | Splicing Interval | [RFC8286] | no |
| urn:ietf:params:rtp-hdrext:smpte-tc | SMPTE time-code mapping | [RFC5484] | no |
| urn:ietf:params:rtp-hdrext:sdes | Reserved as base URN for RTCP SDES items that are also defined as RTP compact header extensions. | [RFC7941] | no |
| urn:ietf:params:rtp-hdrext:ntp-64 | Synchronisation metadata: 64-bit timestamp format | [RFC6051] | no |
| urn:ietf:params:rtp-hdrext:ntp-56 | Synchronisation metadata: 56-bit timestamp format | [RFC6051] | no |
| urn:ietf:params:rtp-hdrext:encrypt | Encrypted extension header element | [RFC6904] | no, but maybe irrelevant? |
| urn:ietf:params:rtp-hdrext:csrc-audio-level | Mixer-to-client audio level indicators | [RFC6465] | no |
| urn:3gpp:video-orientation:6 | Higher granularity (6-bit) coordination of video orientation (CVO) feature, see clause 6.2.3 | [3GPP TS 26.114, version 12.5.0] | probably not(?) |
| urn:3gpp:video-orientation | Coordination of video orientation (CVO) feature, see clause 6.2.3 | [3GPP TS 26.114, version 12.5.0] | probably not(?) |
| urn:3gpp:roi-sent | Signalling of the arbitrary region-of-interest (ROI) information for the sent video, see clause 6.2.3.4 | [3GPP TS 26.114, version 13.1.0] | probably not(?) |
| urn:3gpp:predefined-roi-sent | Signalling of the predefined region-of-interest (ROI) information for the sent video, see clause 6.2.3.4 | [3GPP TS 26.114, version 13.1.0] | probably not(?) |
| Extension URI | Description | Reference | QUIC |
|---|---|---|---|
| urn:ietf:params:rtp-hdrext:sdes:cname | Source Description: Canonical End-Point Identifier (SDES CNAME) | [RFC7941] | no |
| urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id | RTP Stream Identifier | [RFC8852] | no |
| urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id | RTP Repaired Stream Identifier | [RFC8852] | no |
| urn:ietf:params:rtp-hdrext:sdes:CaptId | CLUE CaptId | [RFC8849] | no |
| urn:ietf:params:rtp-hdrext:sdes:mid | Media identification | [RFC9143] | no |
The mapping described in the previous sections poses some interface requirements on the QUIC implementation. Although a basic mapping should work without any of these requirements most of the optimizations regarding rate adaptation and RTCP mapping require certain functionalities to be exposed to the application. The following to sections contain a list of information that is required by an application to implement different optimizations (Section 8.1) and functions that a QUIC implementation SHOULD expose to an application (Section 8.2).¶
Each item in the following list can be considered individually. Any exposed information or function can be used by RoQ regardless of whether the other items are available. Thus, RoQ does not depend on the availability of all of the listed features but can apply different optimizations depending on the functionality exposed by the QUIC implementation.¶
This section provides a list of items that an application might want to export from an underlying QUIC implementation. It is thus RECOMMENDED that a QUIC implementation exports the listed items to the application.¶
This sections lists functions that a QUIC implementation SHOULD expose to an application to implement different features of the mapping described in the previous sections of this document.¶
RTP sessions are characterized by a continuous flow of packets into either or both directions. A connection migration may lead to pausing media transmission until reachability of the peer under the new address is validated. This may lead to short breaks in media delivery in the order of RTT and, if RTCP is used for RTT measurements, may cause spikes in observed delays. Application layer congestion control mechanisms (and also packet repair schemes such as retransmissions) need to be prepared to cope with such spikes.¶
If a QUIC connection is established via a signaling channel, this signaling may have involved Interactive Connectivity Establishment (ICE) exchanges to determine and choose suitable (IP address, port number) pairs for the QUIC connection. Subsequent address change events may be noticed by QUIC via its connection migration handling and/or at the ICE or other application layer, e.g., by noticing changing IP addresses at the network interface. This may imply that the two signaling and data "layers" get (temporarily) out of sync.¶
Editor's Note: It may be desirable that the API provides an indication of connection migration event for either case.¶
For repeated connections between peers, the initiator of a QUIC connection can use 0-RTT data for both QUIC streams and datagrams. As such packets are subject to replay attacks, applications shall carefully specify which data types and operations are allowed. 0-RTT data may be beneficial for use with RoQ to reduce the risk of media clipping, e.g., at the beginning of a conversation.¶
This specification defines carrying RTP on top of QUIC and thus does not finally define what the actual application data are going to be. RTP typically carries ephemeral media contents that is rendered and possibly recorded but otherwise causes no side effects. Moreover, the amount of data that can be carried as 0-RTT data is rather limited. But it is the responsibility of the respective application to determine if 0-RTT data is permissible.¶
Editor's Note: Since the QUIC connection will often be created in the context of an existing signaling relationship (e.g., using WebRTC or SIP), specific 0-RTT keying material could be exchanged to prevent replays across sessions. Within the same connection, replayed media packets would be discarded as duplicates by the receiver.¶
RoQ is subject to the security considerations of RTP described in Section 9 of [RFC3550] and the security considerations of any RTP profile in use.¶
The security considerations for the QUIC protocol and datagram extension described in Section 21 of [RFC9000], Section 9 of [RFC9001], Section 8 of [RFC9002] and Section 6 of [RFC9221] also apply to RoQ.¶
Note that RoQ provides mandatory security, and other RTP transports do not. In order to prevent the inadvertent disclosure of RTP sessions to unintended third parties, RTP topologies described in Section 3.1 that include middleboxes supporting both RoQ and non-RoQ paths MUST forward RTP packets on non-RoQ paths using a secure AVP profile ([RFC3711], [RFC4585], or another AVP profile providing equivalent RTP-level security), whether or not RoQ senders are using a secure AVP profile for those RTP packets.¶
This document creates a new registration for the identification of RoQ in the "TLS Application-Layer Protocol Negotiation (ALPN) Protocol IDs" registry [RFC7301].¶
The "rtp-mux-quic" string identifies RoQ:¶
The following is a list of QUIC protocol extensions that might be beneficial for RoQ, but are not required by RoQ.¶
A version of QUIC receive timestamps can be helpful for improved jitter calculations and congestion control.¶
An experimental implementation of the mapping described in this document can be found on Github. The application implements the RoQ Datagrams mapping and implements SCReAM congestion control at the application layer. It can optionally disable the builtin QUIC congestion control (NewReno). The endpoints only use RTCP for congestion control feedback, which can optionally be disabled and replaced by the QUIC connection statistics as described in Section 7.3.¶
Experimental results of the implementation can be found on Github, too.¶
Early versions of this document were similar in spirit to [I-D.draft-hurst-quic-rtp-tunnelling], although many details differ. The authors would like to thank Sam Hurst for providing his thoughts about how QUIC could be used to carry RTP.¶
The authors would like to thank Bernard Aboba, David Schinazi, Lucas Pardue, Sergio Garcia Murillo, Spencer Dawkins, and Vidhi Goel for their valuable comments and suggestions contributing to this document.¶