TSVWG K. Carlberg
Internet-Draft G11
Intended Status: Informational P. O'Hanlon
Expires: April 4, 2013 UCL
Oct 4, 2012
Reactions to Signaling from ECN Support for RTP/RTCP
<draft-carlberg-tsvwg-ecn-reactions-03.txt>
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Abstract
This document presents various responses to Congestion Experience (CE)
notifications by real time applications that have negotiated end-to-end
support of Explicit Congestion Notification (ECN). This document is a
follow-on effort of [rfc6679], which specifies the signaling used to
provide ECN support for RTP/RTCP flows.
1. Introduction
This document presents various responses to Congestion Experience (CE)
notifications by real time applications that have negotiated end-to-end
support of Explicit Congestion Notification (ECN). [rfc6679] defines
the signaling for support of ECN by RTP based sessions, and also covers
the case where a set of nodes do not respond to CE notifications. A
more detailed discussion about how back-off algorithms can be achieved
and supported for specific applications is viewed as out of scope of
that document and may be addressed by a companion document.
1.1 Background
ECN is a mechanism used to explicitly signal the presence of congestion
without relying on packet loss. It was initially designed using a dual
layer signaling model; negotiation and feedback at the transport layer,
and downstream notification of congestion at the network layer. For IP,
a new two bit field was used to both indicate the successful negotiated
support for ECN signaling, as well as indicate the presence of
congestion via the CE flag. In the case of TCP [rfc3168], a new TCP
header flag was defined that provides upstream end-to-end indication of
congestion occurring somewhere along the downstream path.
There should be no difference in congestion response if ECN-CE marks or
packet drops are detected. However it is noted that there MAY be other
reactions to ECN-CE specified in the future. Such an alternative
reaction MUST be specified and considered to be safe for deployment
under any restrictions specified. We specify such an alternative in
this document.
With respect to ECN for TCP, [rfc3168] specifies an indication of
congestion, but it does so once per Round Trip Time (RTT). [rfc6679] is
an effort that proposes a finer grained notification reflecting a more
accurate indication of the number of ECN marked packets received within
one RTT.
1.2 Terminology and Abbreviations
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
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document are to be interpreted as described in RFC2119 [RFC2119].
2. Issues
The initial discussions and presentation of [rfc6679] produced a
consensus that the specification of signaling was to be done within the
AVTcore working group, and any subsequent discussion on end-to-end
reactions to the signaling would be accomplished in the Transport
Services (TSV) working group. This draft satisfies the latter effort.
Another issue that needs to be recognized is that the reactions to CE in
the context of [rfc6679] are the responsibility of the application.
This is in contrast to ECN support for TCP, where explicit signaled
feedback of, and reaction to, CE is kept transparent to the application.
The issue of placing the feedback responsibility in the application is
that each application needs to add specific support for that reaction.
On the other hand, multiple reactions may be considered by the
application. For this reason, [rfc6679] states the need for a default
congestion control reaction that MUST be supported. Section 3 through 5
expands on this topic.
3. Congestion Control Algorithms
The transport of any data flow across the Internet produces a need for
some form of congestion control to attain a suitable share of the
capacity of the path through a network. Most of the existing work on
realtime congestion control algorithms has been rooted in TCP-friendly
approaches but with smoother adaptation cycles. TCP congestion control
is unsuitable for interactive media for a number of reasons including
the fact that it is loss-based so it maximises the latency on a path, it
changes its transmit rate to quickly for multimedia, and favours
reliability over timeliness. In the case of real time media transport,
one requires:
Smoother rate variation: (than for bulk data) to accommodate
the underlying media flow's characteristics.
Low latency: Maintaining latencies sufficient to be usable, where
150ms one way delay is understood to be a good target
[ITU.G114.2003].
Fairness: The algorithm must be fair to both itself and other flows
3.1 TCP Friendly Rate Control (TFRC)
TFRC has a smoother response to congestion than TCP-like approaches,
thus making it more suitable for real-time interactive multimedia
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applications. It has been cited in a number of other documents within
the IETF for use with UDP and media flows [rfc3714, bcp145] and is
seeing full and partial deployment in related solutions such as
Empathy/Farsight, and GoogleTalk [goog1].
However it should be noted that TFRC is only recommended for real-time
media use with ECN response. TFRC is not recommended for non-ECN paths
due to its loss based operation which leads to full queues with
maximised latencies. It is assumed that ECN markings will usually occur
with lower queue occupancy and thus lower latency. However it is
understood that ECN marks may not provide for sufficiently low
latencies in some situations so other congestion control solutions
would be preferable.
[rfc4342] specifies the profile for TFRC for use in the Datagram
Congestion Control Protocol (DCCP) [rfc4340] for a half connection. A
DCCP half connection is defined as application data sent downstream with
corresponding acknowledgements sent upstream. These half-connections
can be realized in the form of one-way pre-recoded media, one-way live
media, or two-way interactive. A perceived drawback in this profile
concerns its application to interactive media that use small packets.
[RFC4828] is an experimental protocol defining a variation of TFRC used
to address this drawback and achieve the same bandwidth as a TCP flow
using packets of size 1500 bytes.
[rfc6679] is an standard that specifies how RTP flows can
be supported using the RTP/AVPF profile and the general RTP header
extension mechanism.
3.2 Related Work
3.2.1 3GPP
Outside of this previous and on-going work with TFRC, it is understood
that some parties have issues with the behavior of TFRC under certain
conditions. A notable mention of this is made in the 3GPP's document on
IP Multimedia Subsystem (IMS) Media handling and interaction [TR26.114],
where it is mentioned:
"Note that for IMS networks, which normally have nonzero packet loss and
fairly long round-trip delay, the amount of bitrate reduction specified
in RFC 3448 is generally too restrictive for video and may, if used as
specified, result in very low video bitrates already at (for IMS)
moderate packet loss rates."
Though it is unclear exactly what the 3GPP community consider as too
restrictive and whether some alteration of the response may be suitable.
It should be noted that the 3GPP document only referred to an older
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version of TFRC defined in [RFC3448]. Given that the current version
of TFRC [RFC5348] has made significant changes to the idle and data-
limited responses it is unclear whether their assessment is relevant
to current TFRC implementations.
Furthermore the specification [TR26.114] only outlines a rudimentary
approach to congestion control, providing an example of a 60% back-off
reaction to loss within an RTCP reporting period. The proposed signalling
employs Temporary Maximum Media Stream Bit Rate Request (TMMBR)
[RFC5104] and Codec Mode Request (CMR) [RFC4867] for video and audio
respectively, which would only provide for very basic rate control
if used as specified. We note that [TR26.114] specifies terminal
behavior, while [TS36.300] specifies base station behaviour, though
neither specify any standardised congestion control approach.
It is understood that there are a number of proprietary and patented
approaches that provide more sophisticated response in the case of
3G/LTE, but since these are neither endorsed nor standardized this
document advocates a standardized approach such as TFRC.
We also acknowledge that there are many congestion control algorithms
available for implementers to choose from, with a subset that are
specifically suited to real time media transmission. However, given a
variety of real time applications and their various characteristics
(sender-only broadcast, interactive unicast, etc), we need to expand the
notion of how back-off can be achieved. Hence, the focus needs to be on
an output that would resemble the characteristics of TFRC.
Within the RTCweb Working Group the need for a more media friendly
congestion control mechanism has been made apparent. Currently, TFRC is
perceived as having deficiencies (e.g. its loss-based design, lack of
cross-stream congestion control functionality etc) that make it an
incomplete or insufficient solution for the envisioned RTCWEB media
flows. The RTP Media Congestion Avoidance Techniques (rmcat) working
group has now been formed which aims to lead to the formation of a
working group on these issues. The group aims to develop one or more
congestion control algorithms, associated extensions, and evaluation
criteria. Furthermore it has been proposed that certain practices, such
as 'circuit-breaker' conditions, to provide operational limits on
congestion control algorithms, and feedback messages, may be tackled in
other groups such as AVTCORE and AVTEXT respectively.
Thus there is some movement to attempt to develop new algorithms better
suited to media transport, but these efforts will clearly take a
considerable time to reach fruition. Whilst TFRC has some perceived
issues it still provides the best existing solution for media transport.
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3.3 ECN response
As mentioned above and in accordance to [rfc3168], the actual response
to the reception of an ECN-CE marked packet MUST normally be the same as
that of a lost packet. However there are a number of contexts where one
may also be interested in more varied approaches. We expand on this in
Section 5 below.
4. Application Layer Congestion Response
Whilst the congestion control algorithm may decide to alter the rate at
which the application should operate, in the case of media applications
this process is not as straightforward as the case of bulk data. The
different media engines and codecs in use may only have limited
adaptation ranges, thus, this limitation needs to be a consideration
when adapting the rate. Furthermore the application needs to be aware
of the capability of the specific codecs in terms of their ability to
switch configuration mid-stream (without loss of fidelity), which may
impose further limits on the modes of operation.
One approach for achieving a lower generation of data is through reduced
sampling of the media (e.g., voice or video). In the case of video,
this may also involve slower frame rates. Specific recommendations that
describe how applications should respond to congestion in the context of
supporting the algorithmic characteristics of a congestion control
algorithm are outside the scope of this document.
5. Other Reactions
In addition to the activation of congestion control algorithm, other
reactions can be used or leveraged by an application in response to CE.
We divide these other potential reactions into two categories: signaling
and fault tolerance. We note that these other reactions are considered
symmetric because they require downstream peer support. We also point
out that activation of other reactions represents an example of an
on-demand and as-needed approach in responding to CE.
5.1 Signaling
5.1.1 RSVP
The resource Reservation Protocol (RSVP) can be used to signal a desired
set of path characteristics (e.g., bandwidth, delay) in response to CE
feedback [rfc2205]. Its operation is based on the use of PATH messages
sent downstream hop-by-hop from the source to a destination that specify
requested forwarding characteristics. In return, the destination sends
a hop-by-hop RESV message upstream towards the source confirming the
resources that have been reserved for that flow.
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[rfc3181] defines a priority policy element that specifies both an
allocation and defending priority. This dual specification supports the
use of preemption of existing reservations. [draft-priority-rsvp] is a
work-in-progress that defines a new policy element that only conveys
priority during reservation establishment. This latter effort also
presents several reservation models, including one that describes
engineered resources set aside for priority users.
5.1.2 Differentiated Services
Unlike RSVP and its use of a separate signaling mechanism to reserve
resources, Differentiated Services (diff-serv) uses code points within
the IP header to convey the forwarding behavior of that packet
[rfc2474]. This may range from various drop precedence values to a code
point that signifies low delay and low loss (i.e., characteristics
attributed to real time flows).
As in the case of RSVP, applications could rely on the reception of CE
feedback to initiate a subsequent setting of diff-serv code points to
provide additional protection or explicit association of forwarding
characteristics of a given flow of packets. In addition, the setting of
diff-serv code points would be done on an as-needed basis in reaction to
CE feedback. Recommendations concerning specific diff-serv values are
outside the scope of this document.
5.2 Fault Tolerance
Fault tolerance is another category of reactions that may be used by
applications in response to CE feedback. In some cases, these efforts
may contribute to an increase in traffic load in order to add protection
and resiliency to a flow.
Redundant Transmissions: This approach is based on a source sending
duplicate payloads that can be used to compensate for lost packets.
Given that ECN marks the packet and forwards it towards the destination
(instead of dropping it), this approach can be considered extreme in
terms of being network unfriendly. Its positive value may emerge in
cases where a path has several downstream congestion points. However,
its actions of producing redundant packets still associates a high
measure of greedy use of resources.
Application Layer Forward Error Correction (FEC): This approach also
adds additional overhead to the flow in order to compensate for
potential packet loss. And as the case of redundant transmissions, the
value of this approach is probably better realized when there exists
multiple downstream congestion points. However, the impact of the
overhead is minimized by having one (or a few) additional packet(s) used
to compensate for the loss of a set of packets.
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Codec Swapping: This approach involves changing codecs to either reduce
load or achieve an improvement in compensating for lost packets.
5.3 Alternative Reaction for Emergency Communications
As mentioned in [rtp-ecn], the default reaction on the reception of
these ECN-CE marked packets MUST be to provide the congestion control
algorithm with a congestion notification that triggers the algorithm to
react as if packet loss had occurred. There MAY be an alternative
reaction if it is considered safe for deployment. An example of the
need for an alternative reaction would be the case of Emergency
Telecommunications Service (ETS) [rfc3689, rfc4190], where an
improvement in QoS or a higher probability of session establishment and
forwarding of traffic is of high interest.
It is proposed that certain authorized ETS flows may be permitted to
employ either a substantially less aggressive back-off algorithm than
the default algorithm, or some level of exemption from reacting to ECN
marked packets. This alternative reaction will benefit these flows as
the marks would normally be considered as equivalent to lost packets,
which would effectively increase the loss level, which in turn will
generally result in the reduction of flow rate. This applies to all
flows that utilize some form of the rate control that is inversely
proportional to the loss rate, which includes TCP-like algorithms or
equation-based approaches.
Simulations of the use of ECN exemption with TFRC and have found that it
has limited effect on the normal flows with low numbers of exempt
flows. A half-dumbbell network was used with a RED router queue
configured using the settings recommended by Sally Floyd. The candidate
flows are 1Mbit/s each with a backhaul 100Mbit/s link. In the standard
case where 1% of flows would be exempt the remaining flows achieve
99.99% of the bandwidth that they would achieve without the presence of
the exempt flows. This is what would be expected from the simple
calculation of the allocation, given that the exempt flows achieve their
full rate (1Mbit/s); With 100 normal plus 1 exempt flow, assuming that
the except flow uses 1Mbit/s, the remaining capacity is 99Mbit/s which
is divided between the 100 normal flows. Whilst when 101 normal flows
are run over the 100Mbit/s link they would have to share it evenly, so
it works out thus: ((99/100)/(100/101))*100=99.99%. In the case of 5%
exempt flows then the proportion is very slightly lower at
((95/100)/(100/105))*100=99.75%. Both these calculations are borne out
in the simulation runs.
The level of exemption employed can be altered in a number of ways. Two
simple approaches would be to either set a threshold number of ECN
marked packets that could be considered as a loss, and another approach
would be to set a percentage threshold of ECN marked packet that would
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be considered as a loss.
It should be noted that in the simulations the end-to-end delay of the
packets within the flows was monitored and the relative delay of the
exempt flows apparently rises somewhat when exemption is
enacted. However what is actually occurring is that the 'normal' flows
are reducing their throughput and are thus reducing their latency
somewhat. There is normally some limited latency when using loss-based
techniques such as TFRC because it fills the queues to ascertain the
link capacity and maintains that level of delay throughout a
session. However the level of latency is clearly limited by the queue
sizes in the network and on media specific links these queue sizes are
typically quite small, so the resulting latency is limited.
Furthermore in the case where media flows employing TFRC, or any other
congestion control algorithm (e.g. delay-based), are sharing a
bottleneck link with TCP flows then the queues will be filled by the TCP
flows and the latency will be kept near or at a their maximum despite
any other flows.
6. IANA Considerations
This document requires no actions from IANA.
7. Security Considerations
The reliance on accurate and un-modified RTCP information means that
SRTP needs to be used, or any other mechanism that helps prevent
modification of RTCP feedback packets.
8. Acknowledgements
TBD
9. References
9.1 Normative
[rfc2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[rfc2205] Braden, B., et. al., "Resource ReSerVation Protocol (RSVP)
-- Version 1 Functional Specification", RFC 2205, September
1997
[rfc2209] Braden, R., L. Zhang, "Resource Reservation Protocol
(RSVP) -- Version 1 Message Processing Rules", RFC2209
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September 1997
[rfc2474] Nichols, K., et. al., "Definition of the Differentiated
Services Field in the IPv4 and IPv6 Headers", RFC 2474,
December 1998
[rfc3168] Ramakrishnan, K,. et. al., "The Addition of Explicit
Congestion Notification (ECN) to IP", RFC 3168,
September, 2001
[rfc3181] Herzog, S., "Signaled Preemption Priority Policy Element",
RFC 3181, October 2001
[rfc3448] Handley, M., et. al., "TCP Friendly Rate Control (TFRC):
Protocol Specification", RFC 3448, January 2003
[rfc4867] Sjoberg, J., et. al., "RTP Payload Format and File Storage
Format for the AMR and AMR-WB Audio Codecs", RFC 4867,
April 2007
[rfc5104] Wenger, S., et. al., "Codec Control Messages in the RTP
Audio-Visual Profile with Feedback (AVPF)", RFC 5104,
February 2008
[rfc6679] Westerlund, M., et. al., "Explicit Congestion
Notification (ECN) for RTP over UDP", RFC 6679,
IETF, July 2012
9.2 Informative
[draft-rtp-tfrc] Gharai, L., C. Perkins, "RTP with TCP Friendly Rate
Control", work-in-progress, Sept 2011
[Goog1] http://code.google.com/apis/talk/call_signaling.html
[tr26.114] "IMS; Multimedia telephony; Media Handling and
Interaction", 3GPP, version 10, April 2011
[ts36.300] "E-UTRA and E-UTRAN Overall Description, Stage 2",
3GPP, Release 10, September, 2011
[RFC4340] Kohler, E., et. al, Datagram Congestion Control
Protocol (DCCP), RFC4340, March 2006
[RFC4342] Floyd, S., et. al., "Profile for DCCP Congestion
Control ID 3: TFRC", RFC 4342, March 2006
[RFC4828] Floyd, S., E. Kohler, "TFRC: The Small Packet
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Variant", RFC 4828, April 2007
[rfc3689] Carlberg, K., Atkinson, R., "General Requirements for
Emergency Telecommunications Service (ETS)", RFC 3689,
February 2004
[rfc4190] Carlberg, K. et, al., "Framework for Supporting
Emergency Telecommunications Service (ETS) in
IP Telephony", RFC 4190, November 2005
[rfc3714] Floyd, S., Kempf, J., "IAB Concerns Regarding Congestion
Control for Voice Traffic in the Internet", RFC 3714,
March 2004
[bcp145] Eggert, L., Fairhurst, G., "Unicast UDP Usage Guidelines
for Application Designers", RFC 5405, BCP 145, November 2008
[ITU.G114.2003]
International Telecommunications Union, "One-way
transmission time", ITU-T Recommendation G.707, May 2003.
Author's Addresses
Piers O'Hanlon
University of Oxford
Oxford Internet Institute
1 St Giles
Oxford OX1 3JS
United Kingdom
Email: piers.ohanlon@oii.ox.ac.uk
Ken Carlberg
G11
1600 Clarendon Blvd
Arlington VA
USA
Email: carlberg@g11.org.uk
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