RFC 8836 | RTP Media Congestion Control Requir | January 2021 |
Jesup & Sarker | Informational | [Page] |
Congestion control is needed for all data transported across the Internet, in order to promote fair usage and prevent congestion collapse. The requirements for interactive, point-to-point real-time multimedia, which needs low-delay, semi-reliable data delivery, are different from the requirements for bulk transfer like FTP or bursty transfers like web pages. Due to an increasing amount of RTP-based real-time media traffic on the Internet (e.g., with the introduction of the Web Real-Time Communication (WebRTC)), it is especially important to ensure that this kind of traffic is congestion controlled.¶
This document describes a set of requirements that can be used to evaluate other congestion control mechanisms in order to figure out their fitness for this purpose, and in particular to provide a set of possible requirements for a real-time media congestion avoidance technique.¶
This document is not an Internet Standards Track specification; it is published for informational purposes.¶
This document is a product of the Internet Engineering Task Force (IETF). It represents the consensus of the IETF community. It has received public review and has been approved for publication by the Internet Engineering Steering Group (IESG). Not all documents approved by the IESG are candidates for any level of Internet Standard; see Section 2 of RFC 7841.¶
Information about the current status of this document, any errata, and how to provide feedback on it may be obtained at https://www.rfc-editor.org/info/rfc8836.¶
Copyright (c) 2021 IETF Trust and the persons identified as the document authors. All rights reserved.¶
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Most of today's TCP congestion control schemes were developed with a focus on a use of the Internet for reliable bulk transfer of non-time-critical data, such as transfer of large files. They have also been used successfully to govern the reliable transfer of smaller chunks of data in as short a time as possible, such as when fetching web pages.¶
These algorithms have also been used for transfer of media streams that are viewed in a non-interactive manner, such as "streaming" video, where having the data ready when the viewer wants it is important, but the exact timing of the delivery is not.¶
When handling real-time interactive media, the requirements are different. One needs to provide the data continuously, within a very limited time window (no more delay than hundreds of milliseconds end-to-end). In addition, the sources of data may be able to adapt the amount of data that needs sending within fairly wide margins, but they can be rate limited by the application -- even not always having data to send. They may tolerate some amount of packet loss, but since the data is generated in real time, sending "future" data is impossible, and since it's consumed in real time, data delivered late is commonly useless.¶
While the requirements for real-time interactive media differ from the requirements for the other flow types, these other flow types will be present in the network. The congestion control algorithm for real-time interactive media must work properly when these other flow types are present as cross traffic on the network.¶
One particular protocol portfolio being developed for this use case is WebRTC [RFC8825], where one envisions sending multiple flows using the Real-time Transport Protocol (RTP) [RFC3550] between two peers, in conjunction with data flows, all at the same time, without having special arrangements with the intervening service providers. As RTP does not provide any congestion control mechanism, a set of circuit breakers, such as those described in [RFC8083], are required to protect the network from excessive congestion caused by non-congestion-controlled flows. When the real-time interactive media is congestion controlled, it is recommended that the congestion control mechanism operate within the constraints defined by these circuit breakers when a circuit breaker is present and that it should not cause congestion collapse when a circuit breaker is not implemented.¶
Given that this use case is the focus of this document, use cases involving non-interactive media such as video streaming and those using multicast/broadcast-type technologies, are out of scope.¶
The terminology defined in [RFC8825] is used in this memo.¶
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in BCP 14 [RFC2119].¶
The congestion control algorithm MUST attempt to provide as-low-as-possible-delay transit for interactive real-time traffic while still providing a useful amount of bandwidth. There may be lower limits on the amount of bandwidth that is useful, but this is largely application specific, and the application may be able to modify or remove flows in order to allow some useful flows to get enough bandwidth. For example, although there might not be enough bandwidth for low-latency video+audio, there could be enough for audio only.¶
The algorithm MUST be fair to other flows, both real-time flows (such as other instances of itself) and TCP flows, both long-lived flows and bursts such as the traffic generated by a typical web-browsing session. Note that "fair" is a rather hard-to-define term. It SHOULD be fair with itself, giving a fair share of the bandwidth to multiple flows with similar RTTs, and if possible to multiple flows with different RTTs.¶
The algorithm SHOULD NOT starve competing TCP flows and SHOULD, as best as possible, avoid starvation by TCP flows.¶
The algorithm SHOULD adapt as quickly as possible to initial network conditions at the start of a flow. This SHOULD occur whether the initial bandwidth is above or below the bottleneck bandwidth.¶
The algorithm SHOULD be stable if the RTP streams are halted or discontinuous (for example, when using Voice Activity Detection).¶
Where possible, the algorithm SHOULD merge information across multiple RTP streams sent between two endpoints when those RTP streams share a common bottleneck, whether or not those streams are multiplexed onto the same ports. This will allow congestion control of the set of streams together instead of as multiple independent streams. It will also allow better overall bandwidth management, faster response to changing conditions, and fairer sharing of bandwidth with other network users.¶
The algorithm SHOULD NOT require any special support from network elements to be able to convey congestion-related information. As much as possible, it SHOULD leverage available information about the incoming flow to provide feedback to the sender. Examples of this information are the packet arrival times, acknowledgements and feedback, packet timestamps, packet losses, and Explicit Congestion Notification (ECN) [RFC3168]; all of these can provide information about the state of the path and any bottlenecks. However, the use of available information is algorithm dependent.¶
Since the assumption here is a set of RTP streams, the backchannel typically SHOULD be done via the RTP Control Protocol (RTCP) [RFC3550]; instead, one alternative would be to include it in a reverse-RTP channel using header extensions.¶
Flows managed by this algorithm and flows competing against each other at a bottleneck may have different Differentiated Services Code Point (DSCP) [RFC5865] markings depending on the type of traffic or may be subject to flow-based QoS. A particular bottleneck or section of the network path may or may not honor DSCP markings. The algorithm SHOULD attempt to leverage DSCP markings when they're available.¶
Among the existing congestion control mechanisms, TCP Friendly Rate Control (TFRC) [RFC5348] is the one that claims to be suitable for real-time interactive media. TFRC is an equation-based congestion control mechanism that provides a reasonably fair share of bandwidth when competing with TCP flows and offers much lower throughput variations than TCP. This is achieved by a slower response to the available bandwidth change than TCP. TFRC is designed to perform best with applications that have a fixed packet size and do not have a fixed period between sending packets.¶
TFRC detects loss events and reacts to congestion-caused loss by reducing its sending rate. It allows applications to increase the sending rate until loss is observed in the flows. As noted in IAB/IRTF report [RFC7295], large buffers are available in the network elements, which introduce additional delay in the communication. It becomes important to take all possible congestion indications into consideration. Looking at the current Internet deployment, TFRC's biggest deficiency is that it only considers loss events as a congestion indication.¶
A typical real-time interactive communication includes live-encoded audio and video flow(s). In such a communication scenario, an audio source typically needs a fixed interval between packets and needs to vary the segment size of the packets instead of the packet rate in response to congestion; therefore, it sends smaller packets. A variant of TFRC, Small-Packet TFRC (TFRC-SP) [RFC4828], addresses the issues related to such kind of sources. A video source generally varies video frame sizes, can produce large frames that need to be further fragmented to fit into path Maximum Transmission Unit (MTU) size, and has an almost fixed interval between producing frames under a certain frame rate. TFRC is known to be less optimal when using such video sources.¶
There are also some mismatches between TFRC's design assumptions and how the media sources in a typical real-time interactive application work. TFRC is designed to maintain a smooth sending rate; however, media sources can change rates in steps for both rate increase and rate decrease. TFRC can operate in two modes: i) bytes per second and ii) packets per second, where typical real-time interactive media sources operate on bit per second. There are also limitations on how quickly the media sources can adapt to specific sending rates. Modern video encoders can operate in a mode in which they can vary the output bitrate a lot depending on the way they are configured, the current scene they are encoding, and more. Therefore, it is possible that the video source will not always output at an allowable bitrate. TFRC tries to increase its sending rate when transmitting at the maximum allowed rate, and it increases only twice the current transmission rate; hence, it may create issues when the video sources vary their bitrates.¶
Moreover, there are a number of studies on TFRC that show its limitations, including TFRC's unfairness to low statistically multiplexed links, oscillatory behavior, performance issues in highly dynamic loss-rate conditions, and more [CH09].¶
Looking at all these deficiencies, it can be concluded that the requirements for a congestion control mechanism for real-time interactive media cannot be met by TFRC as defined in the standard.¶
This document has no IANA actions.¶
An attacker with the ability to delete, delay, or insert messages into the flow can fake congestion signals, unless they are passed on a tamper-proof path. Since some possible algorithms depend on the timing of packet arrival, even a traditional, protected channel does not fully mitigate such attacks.¶
An attack that reduces bandwidth is not necessarily significant, since an on-path attacker could break the connection by discarding all packets. Attacks that increase the perceived available bandwidth are conceivable and need to be evaluated. Such attacks could result in starvation of competing flows and permit amplification attacks.¶
Algorithm designers should consider the possibility of malicious on-path attackers.¶
This document is the result of discussions in various fora of the WebRTC effort, in particular on the <[email protected]> mailing list. Many people contributed their thoughts to this.¶