Internet DRAFT - draft-ietf-rmcat-cc-codec-interactions
draft-ietf-rmcat-cc-codec-interactions
Network Working Group M. Zanaty
Internet-Draft Cisco
Intended status: Standards Track V. Singh
Expires: September 19, 2016 Aalto University
S. Nandakumar
Cisco
Z. Sarkar
Ericsson AB
March 18, 2016
Congestion Control and Codec interactions in RTP Applications
draft-ietf-rmcat-cc-codec-interactions-02
Abstract
Interactive real-time media applications that use the Real-time
Transport Protocol (RTP) over the User Datagram Protocol (UDP) must
use congestion control techniques above the UDP layer since it
provides none. This memo describes the interactions and conceptual
interfaces necessary between the application components that relate
to congestion control, specifically the media codec control layer,
and the components dedicated to congestion control functions.
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 19, 2016.
Copyright Notice
Copyright (c) 2016 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
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Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 2
2. Key Words for Requirements . . . . . . . . . . . . . . . . . 3
3. Conceptual Model . . . . . . . . . . . . . . . . . . . . . . 4
4. Implementation Model . . . . . . . . . . . . . . . . . . . . 5
5. Codec - CC Interactions . . . . . . . . . . . . . . . . . . . 6
5.1. Mandatory Interactions . . . . . . . . . . . . . . . . . 6
5.1.1. Allowed Rate . . . . . . . . . . . . . . . . . . . . 6
5.2. Optional Interactions . . . . . . . . . . . . . . . . . . 7
5.2.1. Media Elasticity . . . . . . . . . . . . . . . . . . 7
5.2.2. Startup Ramp . . . . . . . . . . . . . . . . . . . . 7
5.2.3. Delay Tolerance . . . . . . . . . . . . . . . . . . . 8
5.2.4. Loss Tolerance . . . . . . . . . . . . . . . . . . . 8
5.2.5. Forward Error Correction . . . . . . . . . . . . . . 8
5.2.6. Probing for Available Bandwidth . . . . . . . . . . . 8
5.2.7. Throughput Sensitivity . . . . . . . . . . . . . . . 9
5.2.8. Rate Stability . . . . . . . . . . . . . . . . . . . 9
6. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . 9
7. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 9
8. References . . . . . . . . . . . . . . . . . . . . . . . . . 9
8.1. Normative References . . . . . . . . . . . . . . . . . . 9
8.2. Informative References . . . . . . . . . . . . . . . . . 11
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 12
1. Introduction
Interactive real-time media applications most commonly use RTP
[RFC3550] over UDP [RFC0768]. Since UDP provides no form of
congestion control, which is essential for any application deployed
on the Internet, these RTP applications have historically implemented
one of the following options at the application layer to address
their congestion control requirements.
o For media with relatively low packet rates and bit rates, such as
many speech codecs, some applications use a simple form of
congestion control that stops transmission permanently or
temporarily after observing significant packet loss over a
significant period of time, similar to the RTP circuit breakers
[I-D.ietf-avtcore-rtp-circuit-breakers].
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o Some applications have no explicit congestion control, despite the
clear requirements in RTP and its profiles AVP [RFC3551] and AVPF
[RFC4585], under the expectation that users will terminate media
flows that are significantly impaired by congestion (in essence,
human circuit breakers).
o For media with substantially higher packet rates and bit rates,
such as many video codecs, various non-standard congestion control
techniques are often used to adapt transmission rate based on
receiver feedback.
o Some experimental applications use standardized techniques such as
TCP-Friendly Rate Control (TFRC) [RFC5348]. However, for various
reasons, these have not been widely deployed.
The RTP Media Congestion Avoidance Techniques (RMCAT) working group
was chartered to standardize appropriate and effective congestion
control for RTP applications. It is expected such applications will
migrate from the above historical solutions to the RMCAT solution(s).
The RMCAT requirements [I-D.ietf-rmcat-cc-requirements] include low
delay, reasonably high throughput, fast reaction to capacity changes
including routing or interface changes, stability without over-
reaction or oscillation, fair bandwidth sharing with other instances
of itself and TCP flows, sharing information across multiple flows
when possible [I-D.welzl-rmcat-coupled-cc], and performing as well or
better in networks which support Active Queue Management (AQM),
Explicit Congestion Notification (ECN), or Differentiated Services
Code Points (DSCP).
In order to meet these requirements, interactions are necessary
between the application's congestion controller, the RTP layer, media
codecs, other components, and the underlying UDP/IP network stack.
This memo attempts to present a conceptual model of the various
interfaces based on a simplified application decomposition. This
memo discusses interactions between the congestion control and codec
control layer in a typical RTP Application.
Note that RTP can also operate over other transports with integrated
congestion control such as TCP [RFC5681] and DCCP [RFC4340], but that
is beyond the scope of RMCAT and this memo.
2. Key Words for Requirements
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 [RFC2119].
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3. Conceptual Model
It is useful to decompose an RTP application into several components
to facilitate understanding and discussion of where congestion
control functions operate, and how they interface with the other
components. The conceptual model in Figure 1 consists of the
following components.
+----------------------------+
| +-----Config-----+ |
| | | | |
| | Codec | |
| | | | | | |
| APP +---RTP | RTCP---+ |
| | | | | |
| | | | | |
| +---Congestion-------|---Shared
| Control | State
+----------------------------+
|
+----------------------------+
| Network UDP |
| Stack | |
| IP |
+----------------------------+
Figure 1
o APP: Application containing one or more RTP streams and the
corresponding media codecs and congestion controllers. For
example, a WebRTC browser.
o Config: Configuration specified by the application that provides
the media and transport parameters, RTP and RTCP parameters and
extensions, and congestion control parameters. For example, a
WebRTC Javascript application may use the 'constraints' API to
affect the media configuration, and SDP applications may negotiate
the media and transport parameters with the remote peer. This
determines the initial static configuration negotiated in session
establishment. The dynamic state may differ due to congestion or
other factors, but still must conform to limits established in the
config.
o Codec: Media encoder/decoder or other source/sink for the RTP
payload. The codec may be, for example, a simple monaural audio
format, a complex scalable video codec with several dependent
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layers, or a source/sink with no live encoding/decoding such as a
mixer which selectively switches and forwards streams rather than
mixes media.
o RTP: Standard RTP stack functions, including media packetization /
de-packetization and header processing, but excluding existing
extensions and possible new extensions specific to congestion
control (CC) such as absolute timestamps or relative transmission
time offsets in RTP header extensions. RTCP: Standard RTCP
functions, including sender reports, receiver reports, extended
reports, circuit breakers [I-D.ietf-avtcore-rtp-circuit-breakers],
feedback messages such as NACK [RFC4585] and codec control
messages such as TMMBR [RFC5104], but excluding existing
extensions and possible new extensions specific to congestion
control (CC) such as REMB [I-D.alvestrand-rmcat-remb] (for
receiver-side CC), ACK (for sender-side CC), absolute and/or
relative timestamps (for sender-side or receiver-side CC), etc.
o Congestion Control: All functions directly responsible for
congestion control, including possible new RTP/RTCP extensions,
send rate computation (for sender-side CC), receive rate
computation (for receiver-side CC), other statistics, and control
of the UDP sockets including packet scheduling for traffic
shaping/pacing.
o Shared State: Storage and exchange of congestion control state for
multiple flows within the application and beyond it.
o Network Stack: The platform's underlying network functions,
usually part of the Operating System (OS), containing the UDP
socket interface and other network functions such as ECN, DSCP,
physical interface events, interface-level traffic shaping and
packet scheduling, etc. This is usually part of the Operating
System, often within the kernel; however, user-space network
stacks and components are also possible.
4. Implementation Model
There are advantages and drawbacks to implementing congestion control
in the application layer. It avoids platform dependencies and allows
for rapid experimentation, evolution and optimization for each
application. However, it also puts the burden on all applications,
which raises the risks of improper or divergent implementations. One
motivation of this memo is to mitigate such risks by giving proper
guidance on how the application components relating to congestion
control should interact.
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Another drawback of congestion control in the application layer is
that any decomposition, including the one presented in Figure 1, is
purely conceptual and illustrative, since implementations have
differing designs and decompositions. Conversely, this can be viewed
as an advantage to distribute congestion control functions wherever
expedient without rigid interfaces. For example, they may be
distributed within the RTP/RTCP stack itself, so the separate
components in Figure 1 are combined into a single RTP+RTCP+CC
component as shown in Figure 2.
+----------------------------+
| +-----Config |
| | | |
| | Codec |
| APP | | |
| +---RTP+RTCP+CC------|---Shared
+----------------------------+ State
|
+----------------------------+
| Network UDP |
| Stack | |
| IP |
+----------------------------+
Figure 2
5. Codec - CC Interactions
The following subsections identify the necessary interactions between
the Codec and congestion control (CC) layer interfaces that needs to
be considered important.
5.1. Mandatory Interactions
5.1.1. Allowed Rate
Allowed Rate (from CC to Codec): The max transmit rate allowed over
the next time interval. The time interval may be specified or may
use a default. The rate may be specified in bytes or packets or
both. The rate must never exceed permanent limits established in
session signaling such as the SDP bandwidth attribute [RFC4566] nor
temporary limits in RTCP such as TMMBR [RFC5104] or REMB
[I-D.alvestrand-rmcat-remb]. This is the most important interface
among all components, and is always required in any RMCAT solution.
In the simplest possible solution, it may be the only CC interface
required.
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5.2. Optional Interactions
This section identifies certain advanced interactions that if
implemented by an RMCAT solution shall provide more granular control
over the congestion control state and the encoder behavior. As of
today, these interactions are optional to implement and future
evaluations of the existing/upcoming codecs might result in
considering some or all of these as Mandatory interactions.
5.2.1. Media Elasticity
Media Elasticity (from Codec to CC): Many live media encoders are
highly elastic, often able to achieve any target bit rate within a
wide range, by adapting the media quality. For example, a video
encoder may support any bit rate within a range of a few tens or
hundreds of kbps up to several Mbps, with rate changes registering as
fast as the next video frame, although there may be limitations in
the frequency of changes. Other encoders may be less elastic,
supporting a narrower rate range, coarser granularity of rate steps,
slower reaction to rate changes, etc. Other media, particularly some
audio codecs, may be fully inelastic with a single fixed rate. CC
can beneficially use codec elasticity, if provided, to plan Allowed
Rate changes, especially when there are multiple flows sharing CC
state and bandwidth.
5.2.2. Startup Ramp
Startup Ramp (from Codec to CC, and from CC to Codec): Startup is an
important moment in a conversation. Rapid rate adaptation during
startup is therefore important. The codec should minimize its
startup media rate as much as possible without adversely impacting
the user experience, and support a strategy for rapid rate ramp. The
CC should allow the highest startup media rate as possible without
adversely impacting network conditions, and also support rapid rate
ramp until stabilizing on the available bandwidth. Startup can be
viewed as a negotiation between the codec and the CC. The
specification of the ramp may take a number of forms depending on the
interface to the codec; for example, a percentage bit rate increase
per RTT (or other time interval), or an increased transmit window (in
number of packets and/or octets allowed outstanding) are all
potential forms. The codec requests a startup rate and ramp, and the
CC responds with the allowable parameters which may be lower/slower.
The RMCAT requirements also include the possibility of bandwidth
history to further accelerate or even eliminate startup ramp time.
While this acceleration or elimination in ramp time is beneficial to
the session user experience when bandwidth is sufficient, it can be
detrimental if significant congestion results (the user experience of
this session and all other sessions traversing the point of
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congestion may unnecessarily degrade). Therefore, it is recommended
that use of potentially stale congestion state for acceleration or
elimination in ramp up be limited to topologies or deployments
believed to have sufficient bandwidth margin or those in which the
potential transient congestion risk is acceptable. Note that startup
can often commence before user interaction or conversation to reduce
the chance of clipped media.
5.2.3. Delay Tolerance
Delay Tolerance (from Codec to CC): An ideal CC will always minimize
delay and target zero. However, real solutions often need a real
non-zero delay tolerance. The codec should provide an absolute delay
tolerance, perhaps expressed as an impairment factor to mix with
other metrics.
5.2.4. Loss Tolerance
Loss Tolerance (from Codec to CC): An ideal CC will always minimize
packet loss and target zero. However, real solutions often need a
real non-zero loss tolerance. The codec should provide an absolute
loss tolerance, perhaps expressed as an impairment factor to mix with
other metrics. Note this is unrecoverable post-repair loss after
retransmission or forward error correction.
5.2.5. Forward Error Correction
Forward Error Correction (FEC): Simple FEC schemes like XOR Parity
codes [RFC5109] may not handle consecutive or burst loss well. More
complex FEC schemes like Reed-Solomon [RFC6865] or Raptor [RFC6330]
codes are more effective at handling bursty loss. The sensitivity to
packet loss therefore depends on the media (source) encoding as well
as the FEC (channel) encoding, and this sensitivity may differ for
different loss patterns like random, periodic, or consecutive loss.
Expressing this sensitivity to the congestion controller may help it
choose the right balance between optimizing for throughput versus low
loss.
5.2.6. Probing for Available Bandwidth
FEC can also be used to probe for additional available bandwidth, if
the application desires a higher target rate than the current rate.
FEC is preferable to synthetic probes since any contribution to
congestion by the FEC probe will not impact the post-repair loss rate
of the source media flow while synthetic probes may adversely affect
the loss rate. Note that any use of FEC or retransmission must
ensure that the total flow of all packets including FEC,
retransmission and original media never exceeds the Allowed Rate.
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5.2.7. Throughput Sensitivity
Throughput Sensitivity (from Codec to CC): An ideal CC will always
maximize throughput. However, real solutions often need a trade-off
between throughput and other metrics such as delay or loss. The
codec should provide throughput sensitivity, perhaps expressed as an
impairment factor (for low throughputs) to mix with other metrics.
5.2.8. Rate Stability
Rate Stability (from Codec to CC): The CC algorithm must strike a
balance between rate stability and fast reaction to changes in
available bandwidth. The codec should provide its preference for
rate stability versus fast and frequent reaction to rate changes,
perhaps expressed as an impairment factor (for high rate variance
over short timescales) to mix with other metrics.
6. Acknowledgements
The RMCAT design team discussions contributed to this memo. The
authors would also like to thank Michael Ramalho for reviewing and
suggesting text improvements.
7. IANA Considerations
This memo includes no request to IANA.
8. References
8.1. Normative References
[I-D.alvestrand-rmcat-remb]
Alvestrand, H., "RTCP message for Receiver Estimated
Maximum Bitrate", draft-alvestrand-rmcat-remb-03 (work in
progress), October 2013.
[I-D.ietf-avtcore-rtp-circuit-breakers]
Perkins, C. and V. Varun, "Multimedia Congestion Control:
Circuit Breakers for Unicast RTP Sessions", draft-ietf-
avtcore-rtp-circuit-breakers-14 (work in progress), March
2016.
[I-D.ietf-mmusic-sdp-bundle-negotiation]
Holmberg, C., Alvestrand, H., and C. Jennings,
"Negotiating Media Multiplexing Using the Session
Description Protocol (SDP)", draft-ietf-mmusic-sdp-bundle-
negotiation-27 (work in progress), February 2016.
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[I-D.ietf-rmcat-cc-requirements]
Jesup, R. and Z. Sarker, "Congestion Control Requirements
for Interactive Real-Time Media", draft-ietf-rmcat-cc-
requirements-09 (work in progress), December 2014.
[I-D.welzl-rmcat-coupled-cc]
Welzl, M., Islam, S., and S. Gjessing, "Coupled congestion
control for RTP media", draft-welzl-rmcat-coupled-cc-05
(work in progress), June 2015.
[RFC0768] Postel, J., "User Datagram Protocol", STD 6, RFC 768, DOI
10.17487/RFC0768, August 1980,
<http://www.rfc-editor.org/info/rfc768>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, DOI 10.17487/
RFC2119, March 1997,
<http://www.rfc-editor.org/info/rfc2119>.
[RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V.
Jacobson, "RTP: A Transport Protocol for Real-Time
Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550,
July 2003, <http://www.rfc-editor.org/info/rfc3550>.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and
Video Conferences with Minimal Control", STD 65, RFC 3551,
DOI 10.17487/RFC3551, July 2003,
<http://www.rfc-editor.org/info/rfc3551>.
[RFC4340] Kohler, E., Handley, M., and S. Floyd, "Datagram
Congestion Control Protocol (DCCP)", RFC 4340, DOI
10.17487/RFC4340, March 2006,
<http://www.rfc-editor.org/info/rfc4340>.
[RFC4566] Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
Description Protocol", RFC 4566, DOI 10.17487/RFC4566,
July 2006, <http://www.rfc-editor.org/info/rfc4566>.
[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, DOI
10.17487/RFC4585, July 2006,
<http://www.rfc-editor.org/info/rfc4585>.
[RFC5104] Wenger, S., Chandra, U., Westerlund, M., and B. Burman,
"Codec Control Messages in the RTP Audio-Visual Profile
with Feedback (AVPF)", RFC 5104, DOI 10.17487/RFC5104,
February 2008, <http://www.rfc-editor.org/info/rfc5104>.
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[RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error
Correction", RFC 5109, DOI 10.17487/RFC5109, December
2007, <http://www.rfc-editor.org/info/rfc5109>.
[RFC5348] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification", RFC
5348, DOI 10.17487/RFC5348, September 2008,
<http://www.rfc-editor.org/info/rfc5348>.
[RFC5450] Singer, D. and H. Desineni, "Transmission Time Offsets in
RTP Streams", RFC 5450, DOI 10.17487/RFC5450, March 2009,
<http://www.rfc-editor.org/info/rfc5450>.
[RFC5506] Johansson, I. and M. Westerlund, "Support for Reduced-Size
Real-Time Transport Control Protocol (RTCP): Opportunities
and Consequences", RFC 5506, DOI 10.17487/RFC5506, April
2009, <http://www.rfc-editor.org/info/rfc5506>.
[RFC5681] Allman, M., Paxson, V., and E. Blanton, "TCP Congestion
Control", RFC 5681, DOI 10.17487/RFC5681, September 2009,
<http://www.rfc-editor.org/info/rfc5681>.
[RFC5761] Perkins, C. and M. Westerlund, "Multiplexing RTP Data and
Control Packets on a Single Port", RFC 5761, DOI 10.17487/
RFC5761, April 2010,
<http://www.rfc-editor.org/info/rfc5761>.
[RFC6330] Luby, M., Shokrollahi, A., Watson, M., Stockhammer, T.,
and L. Minder, "RaptorQ Forward Error Correction Scheme
for Object Delivery", RFC 6330, DOI 10.17487/RFC6330,
August 2011, <http://www.rfc-editor.org/info/rfc6330>.
[RFC6865] Roca, V., Cunche, M., Lacan, J., Bouabdallah, A., and K.
Matsuzono, "Simple Reed-Solomon Forward Error Correction
(FEC) Scheme for FECFRAME", RFC 6865, DOI 10.17487/
RFC6865, February 2013,
<http://www.rfc-editor.org/info/rfc6865>.
8.2. Informative References
[RFC2818] Rescorla, E., "HTTP Over TLS", RFC 2818, DOI 10.17487/
RFC2818, May 2000,
<http://www.rfc-editor.org/info/rfc2818>.
[RFC3552] Rescorla, E. and B. Korver, "Guidelines for Writing RFC
Text on Security Considerations", BCP 72, RFC 3552, DOI
10.17487/RFC3552, July 2003,
<http://www.rfc-editor.org/info/rfc3552>.
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Authors' Addresses
Mo Zanaty
Cisco
Email: mzanaty@cisco.com
Varun Singh
Aalto University
Email: varun@comnet.tkk.fi
Suhas Nandakumar
Cisco
Email: snandaku@cisco.com
Zaheduzzaman Sarker
Ericsson AB
Email: zaheduzzaman.sarker@ericsson.com
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