Internet DRAFT - draft-jesup-rtp-congestion-reqs
draft-jesup-rtp-congestion-reqs
Network Working Group R. Jesup
Internet-Draft Mozilla
Intended status: Informational H. Alvestrand
Expires: September 5, 2012 Google
March 4, 2012
Congestion Control Requirements For Real Time Media
draft-jesup-rtp-congestion-reqs-00
Abstract
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 by low-delay, semi-reliable data delivery,
are different from the requirements for bulk transfer like FTP or
bursty transfers like Web pages, and the TCP algorithms are not
suitable for this traffic.
This document attempts to describe a set of requirements that can be
used to evaluate other congestion control mechanisms in order to
figure out their fitness for this purpose.
Requirements Language
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [RFC2119].
Status of this Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
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This Internet-Draft will expire on September 5, 2012.
Copyright Notice
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 3
2. Requirements . . . . . . . . . . . . . . . . . . . . . . . . . 3
3. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 6
4. Security Considerations . . . . . . . . . . . . . . . . . . . . 6
5. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 6
6. References . . . . . . . . . . . . . . . . . . . . . . . . . . 7
6.1. Normative References . . . . . . . . . . . . . . . . . . . 7
6.2. Informative References . . . . . . . . . . . . . . . . . . 7
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 7
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1. Introduction
The traditional TCP congestion control requirements were developed in
order to promote efficient 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 fast as possible" mode, 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 doing real time interactive media, the requirements are
different; one needs to provide the data continuously, within a very
limited time window (no more than 100s of milliseconds end-to-end
delay), the sources of data may be able to adapt the amount of data
that needs sending within fairly wide margins, and 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 useless.
One particular protocol portofolio being developed for this use case
is WebRTC [I-D.ietf-rtcweb-overview], where one envisions sending
multiple RTP-based flows between two peers, in conjunction with data
flows, all at the same time, without having special arrangements with
the intervening service providers.
Given that this use case is the focus of this document, use cases
involving noninteractive media such as YouTube-like video streaming,
and use cases using multicast/broadcast-type technologies, are out of
scope.
The terminology defined in [I-D.ietf-rtcweb-overview] is used in this
memo.
2. Requirements
1. The congestion control algorithm must attempt to provide low-
delay transit for real-time traffic, even when faced with
intermediate bottlenecks and competing flows.
A. It should also deal well with routing changes and interface
changes (WiFi to 3G data, etc) which may radically change
the bandwidth available.
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2. The algorithm must be fair to other flows, both realtime flows
(such as other instances of itself), and TCP flows, both long-
lived and bursts such as the traffic generated by a typical web
browsing session. Note that 'fair' is a rather hard-to-define
term.
A. The algorithm must not overreact to short-term bursts (such
as web-browsing) which can quickly saturate a local-
bottleneck router or link, but also clear quickly, and
should recover quickly when the burst ends.
3. The algorithm should merge information across multiple RTP
streams between the same endpoints, whether or not they're
multiplexed on the same ports, in order to allow congestion
control of the set of streams together instead of as multiple
independent streams. This allows better overall bandwidth
management, faster response to changing conditions, and fairer
sharing of bandwidth with other network users.
A. If possible, it should also share information and adaptation
with other non-RTP flows between the same endpoints, such as
a WebRTC data channel
4. The algorithm should not require any special support from
network elements (ECN, etc). As much as possible, it should
leverage existing information about the incoming flows to
provide feedback to the sender. Examples of this information
are the packet arrival times, packet timestamps, packet sizes,
packet losses. Extra information could be added to the packets
to provide more detailed information on actual send times (as
opposed to sampling times), but should not be required.
A. When signals such as ECN are available, it is good if they
can be utilized.
5. Since the assumption here is a set of RTP streams, the
backchannel typically should be done via RTCP; the alternative
would be to include it in a reverse RTP channel using header
extensions.
A. In order to react sufficiently quickly, the AVPF/SAVPF RTP
profile[RFC4585] must be used
B. Note that in some cases, backchannel messages may be delayed
until the RTCP channel can be allocated enough bandwidth,
even under AVPF rules. This may also imply allowing a
higher maximum percentage for RTCP data.
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C. Note that RTCP is of course unreliable
D. Bandwidth for the feedback messages should be minimized
(such as via RFC 5506 [RFC5506]to allow RTCP without SR/RR)
E. Header extensions would avoid the RTCP timing rules issues,
and allow the application to allocate bandwidth as needed
for the congestion algorithm.
F. Backchannel data should be minimized to avoid taking too
much reverse-channel bandwidth (since this will often be
used in a bidirectional set of flows). In areas of
stability, backchannel data may be sent more infrequently so
long as algorithm stability and fairness are maintained.
When the channel is unstable or has not yet reached
equilibrium after a change, backchannel feedback may be more
frequent and use more reverse-channel bandwidth.
6. It should attempt to avoid bandwidth 'collapse' when facing a
long-lived saturating TCP flow or flows. (I.e. a classic delay-
sensitive algorithm will reduce bandwidth to keep delay down
until the TCP flow has all the bandwidth). See the Cx-TCP
algorithm discussed in a recent Transactions On Networking
[cx-tcp] for an example of a delay-sensitive congestion-control
algorithm that transitions to a loss-based mode when competing
with TCP flows - at the cost of increased delay.
7. The algorithm should be stable and low-delay when faced with
active queue management (AQM) in the channel.
8. The algorithm should quickly adapt to initial network conditions
at the start of a flow; the adaptation may be faster than
adaptation later in a flow. This should occur both if the
initial bandwidth is above or below the bottleneck bandwidth.
A. it should allow for both slow-start operation (adapt up) and
history-based startup (start at a point expected to be at or
below channel bandwidth from historical information, which
may need to adapt down quickly if the initial guess is
wrong). Starting too low and/or adapting up too slowly can
cause a critical point in a personal communication to be
poor ("Hello!").
9. Where possible, the algorithm should leverage and piggyback on
other RTCP communications, such as SR/RR, rctp-fb PLI, RPSI, SLI
or application-specific NACK messages (such as for loss
information).
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10. It should be evaluated in how it works both with backbone-router
bottlenecks, (asymmetric) local-loop bottlenecks, and local-lan
(WiFi/etc) bottlenecks.
11. The algorithm should sense the unexpected lack of backchannel
information as a possible indication of a channel overuse
problem and react accordingly to avoid burst events causing a
congestion collapse.
12. It should be stable if the RTP streams are halted or
discontinuous (VAD/DTX); after a resumption of RTP data it may
adapt more quickly (similar to the start of a flow).
3. IANA Considerations
This document makes no request of IANA.
Note to RFC Editor: this section may be removed on publication as an
RFC.
4. Security Considerations
An attacker with the ability to delete, delay or insert messages in
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 percieved available bandwidth
are concievable, and need to be evaluated.
Algorithm designers SHOULD consider the possibility of malicious on-
path attackers.
5. Acknowledgements
This document is the result of discussions in various fora of the
WebRTC effort, in particular on the rtp-congestion@alvestrand.no
mailing list. Many people contributed their thoughts to this.
6. References
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6.1. Normative References
[I-D.ietf-rtcweb-overview]
Alvestrand, H., "Overview: Real Time Protocols for Brower-
based Applications", draft-ietf-rtcweb-overview-00 (work
in progress), June 2011.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC4585] Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
"Extended RTP Profile for Real-time Transport Control
Protocol (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585,
July 2006.
6.2. Informative References
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, April 2009.
[cx-tcp] Budzisz, L., Stanojevic, R., Schlote, A., Baker, F., and
R. Shorten, "On the Fair Coexistence of Loss- and Delay-
Based TCP", December 2011.
Authors' Addresses
Randell Jesup
Mozilla
USA
Email: randell-ietf@jesup.org
Harald Alvestrand
Google
Kungsbron 2
Stockholm 11122
Sweden
Email: harald@alvestrand.no
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