IAB Concerns Regarding Congestion Control for Voice Traffic in the Internet
RFC 3714
This RFC was published on the Internet Architecture Board (IAB) stream.
This RFC is not endorsed by the IETF and has no formal standing in the
IETF standards process.
| Document | Type |
RFC
- Informational
(March 2004)
Was
draft-iab-congestion
(iab)
|
|
|---|---|---|---|
| Authors | Sally Floyd, James Kempf | ||
| Last updated | 2013-03-02 | ||
| RFC stream | Internet Architecture Board (IAB) | ||
| Formats |
RFC 3714
Network Working Group S. Floyd, Ed.
Request for Comments: 3714 J. Kempf, Ed.
Category: Informational March 2004
IAB Concerns Regarding Congestion Control for
Voice Traffic in the Internet
Status of this Memo
This memo provides information for the Internet community. It does
not specify an Internet standard of any kind. Distribution of this
memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2004). All Rights Reserved.
Abstract
This document discusses IAB concerns about effective end-to-end
congestion control for best-effort voice traffic in the Internet.
These concerns have to do with fairness, user quality, and with the
dangers of congestion collapse. The concerns are particularly
relevant in light of the absence of a widespread Quality of Service
(QoS) deployment in the Internet, and the likelihood that this
situation will not change much in the near term. This document is
not making any recommendations about deployment paths for Voice over
Internet Protocol (VoIP) in terms of QoS support, and is not claiming
that best-effort service can be relied upon to give acceptable
performance for VoIP. We are merely observing that voice traffic is
occasionally deployed as best-effort traffic over some links in the
Internet, that we expect this occasional deployment to continue, and
that we have concerns about the lack of effective end-to-end
congestion control for this best-effort voice traffic.
Table of Contents
1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 2
2. An Example of the Potential for Trouble. . . . . . . . . . . . 4
3. Why are Persistent, High Drop Rates a Problem? . . . . . . . . 6
3.1. Congestion Collapse. . . . . . . . . . . . . . . . . . . 6
3.2. User Quality . . . . . . . . . . . . . . . . . . . . . . 7
3.3. The Amorphous Problem of Fairness. . . . . . . . . . . . 8
4. Current efforts in the IETF. . . . . . . . . . . . . . . . . . 10
4.1. RTP. . . . . . . . . . . . . . . . . . . . . . . . . . . 10
4.2. TFRC . . . . . . . . . . . . . . . . . . . . . . . . . . 11
4.3. DCCP . . . . . . . . . . . . . . . . . . . . . . . . . . 12
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4.4. Adaptive Rate Audio Codecs . . . . . . . . . . . . . . . 12
4.5. Differentiated Services and Related Topics . . . . . . . 13
5. Assessing Minimum Acceptable Sending Rates . . . . . . . . . . 13
5.1. Drop Rates at 4.75 kbps Minimum Sending Rate . . . . . . 17
5.2. Drop Rates at 64 kbps Minimum Sending Rate . . . . . . . 18
5.3. Open Issues. . . . . . . . . . . . . . . . . . . . . . . 18
5.4. A Simple Heuristic . . . . . . . . . . . . . . . . . . . 19
6. Constraints on VoIP Systems . . . . . . . . . . . . . . . . . . 20
7. Conclusions and Recommendations. . . . . . . . . . . . . . . . 20
8. Acknowledgements . . . . . . . . . . . . . . . . . . . . . . . 21
9. References . . . . . . . . . . . . . . . . . . . . . . . . . . 21
9.1. Normative References . . . . . . . . . . . . . . . . . . 21
9.2. Informative References . . . . . . . . . . . . . . . . . 22
10. Appendix - Sending Rates with Packet Drops . . . . . . . . . . 26
11. Security Considerations. . . . . . . . . . . . . . . . . . . . 29
12. IANA Considerations. . . . . . . . . . . . . . . . . . . . . . 29
13. Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . 30
14. Full Copyright Statement . . . . . . . . . . . . . . . . . . . 31
1. Introduction
While many in the telephony community assume that commercial VoIP
service in the Internet awaits effective end-to-end QoS, in reality
voice service over best-effort broadband Internet connections is an
available service now with growing demand. While some ISPs deploy
QoS on their backbones, and some corporate intranets offer end-to-end
QoS internally, end-to-end QoS is not generally available to
customers in the current Internet. Given the current commercial
interest in VoIP on best-effort media connections, it seems prudent
to examine the potential effect of real time flows on congestion. In
this document, we perform such an analysis. Note, however, that this
document is not making any recommendations about deployment paths for
VoIP in terms of QoS support, and is not claiming that best-effort
service can be relied upon to give acceptable performance for VoIP.
This document is also not discussing signalling connections for VoIP.
However, voice traffic is in fact occasionally deployed as best
effort traffic over some links in the Internet today, and we expect
this occasional deployment to continue. This document expresses our
concern over the lack of effective end-to-end congestion control for
this best-effort voice traffic.
Assuming that VoIP over best-effort Internet connections continues to
gain popularity among consumers with broadband connections, the
deployment of end-to-end QoS mechanisms in public ISPs may be slow.
The IETF has developed standards for QoS mechanisms in the Internet
[DIFFSERV, RSVP] and continues to be active in this area [NSIS,COPS].
However, the deployment of technologies requiring change to the
Internet infrastructure is subject to a wide range of commercial as
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well as technical considerations, and technologies that can be
deployed without changes to the infrastructure enjoy considerable
advantages in the speed of deployment. RFC 2990 outlines some of the
technical challenges to the deployment of QoS architectures in the
Internet [RFC2990]. Often, interim measures that provide support for
fast-growing applications are adopted, and are successful enough at
meeting the need that the pressure for a ubiquitous deployment of the
more disruptive technologies is reduced. There are many examples of
the slow deployment of infrastructure that are similar to the slow
deployment of QoS mechanisms, including IPv6, IP multicast, or of a
global PKI for IKE and IPsec support.
Interim QoS measures that can be deployed most easily include
single-hop or edge-only QoS mechanisms for VoIP traffic on individual
congested links, such as edge-only QoS mechanisms for cable access
networks. Such local forms of QoS could be quite successful in
protecting some fraction of best-effort VoIP traffic from congestion.
However, these local forms of QoS are not directly visible to the
end-to-end VoIP connection. A best-effort VoIP connection could
experience high end-to-end packet drop rates, and be competing with
other best-effort traffic, even if some of the links along the path
might have single-hop QoS mechanisms.
The deployment of IP telephony is likely to include best-effort
broadband connections to public-access networks, in addition to other
deployment scenarios of dedicated IP networks, or as an alternative
to band splitting on the last mile of ADSL deployments or QoS
mechanisms on cable access networks. There already exists a
rapidly-expanding deployment of VoIP services intended to operate
over residential broadband access links (e.g., [FWD, Vonage]). At
the moment, many public-access IP networks are uncongested in the
core, with low or moderate levels of link utilization, but this is
not necessarily the case on last hop links. If an IP telephony call
runs completely over the Internet, the connection could easily
traverse congested links on both ends. Because of economic factors,
the growth rate of Internet telephony is likely to be greatest in
developing countries, where core links are more likely to be
congested, making congestion control an especially important topic
for developing countries.
Given the possible deployment of IP telephony over congested best-
effort networks, some concerns arise about the possibilities of
congestion collapse due to a rapid growth in real-time voice traffic
that does not practice end-to-end congestion control. This document
raises some concerns about fairness, user quality, and the danger of
congestion collapse that would arise from a rapid growth in best-
effort telephony traffic on best-effort networks. We consider best-
effort telephony connections that have a minimum sending rate and
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that compete directly with other best-effort traffic on a path with
at least one congested link, and address the specific question of
whether such traffic should be required to terminate, or to suspend
sending temporarily, in the face of a persistent, high packet drop
rate, when reducing the sending rate is not a viable alternative.
The concerns in this document about fairness and the danger of
congestion collapse apply not only to telephony traffic, but also to
video traffic and other best-effort real-time traffic with a minimum
sending rate. RFC 2914 already makes the point that best-effort
traffic requires end-to-end congestion control [RFC2914]. Because
audio traffic sends at such a low rate, relative to video and other
real-time traffic, it is sometimes claimed that audio traffic doesn't
require end-to-end congestion control. Thus, while the concerns in
this document are general, the document focuses on the particular
issue of best-effort audio traffic.
Feedback can be sent to the IAB mailing list at iab@ietf.org, or to
the editors at floyd@icir.org and kempf@docomolabs-usa.com. Feedback
can also be sent to the end2end-interest mailing list [E2E].
2. An Example of the Potential for Trouble
At the November, 2002, IEPREP Working Group meeting in Atlanta, a
brief demonstration was made of VoIP over a shared link between a
hotel room in Atlanta, Georgia, USA, and Nairobi, Kenya. The link
ran over the typical uncongested Internet backbone and access links
to peering points between either endpoint and the Internet backbone.
The voice quality on the call was very good, especially in comparison
to the typical quality obtained by a circuit-switched call with
Nairobi. A presentation that accompanied the demonstration described
the access links (e.g., DSL, T1, T3, dialup, and cable modem links)
as the primary source of network congestion, and described VoIP
traffic as being a very small percentage of the packets in commercial
ISP traffic [A02]. The presentation further stated that VoIP
received good quality in the presence of packet drop rates of 5-40%
[AUT]. The VoIP call used an ITU-T G.711 codec, plus proprietary FEC
encoding, plus RTP/UDP/IP framing. The resulting traffic load over
the Internet was substantially more than the 64 kbps required by the
codec. The primary congestion point along the path of the
demonstration was a 128 kbps access link between an ISP in Kenya and
several of its subscribers in Nairobi. So the single VoIP call
consumed more than half of the access link capacity, capacity that is
shared across several different users.
Note that this network configuration is not a particularly good one
for VoIP. In particular, if there are data services running TCP on
the link with a typical packet size of 1500 bytes, then some voice
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packets could be delayed an additional 90 ms, which might cause an
increase in the end to end delay above the ITU-recommended time of
150 ms [G.114] for speech traffic. This would result in a delay
noticeable to users, with an increased variation in delay, and
therefore in call quality, as the bursty TCP traffic comes and goes.
For a call that already had high delay, such as the Nairobi call from
the previous paragraph, the increased jitter due to competing TCP
traffic also increases the requirements on the jitter buffer at the
receiver. Nevertheless, VoIP usage over congested best-effort links
is likely to increase in the near future, regardless of VoIP's
superior performance with "carrier class" service. A best-effort
VoIP connection that persists in sending packets at 64 Kbps,
consuming half of a 128 Kbps access link, in the face of a drop rate
of 40%, with the resulting user-perceptible degradation in voice
quality, is not behaving in a way that serves the interests of either
the VoIP users or the other concurrent users of the network.
As the Nairobi connection demonstrates, prescribing universal
overprovisioning (or more precisely, provisioning sufficient to avoid
persistent congestion) as the solution to the problem is not an
acceptable generic solution. For example, in regions of the world
where circuit-switched telephone service is poor and expensive, and
Internet access is possible and lower cost, provisioning all Internet
links to avoid congestion is likely to be impractical or impossible.
In particular, an over-provisioned core is not by itself sufficient
to avoid congestion collapse all the way along the path, because an
over-provisioned core can not address the common problem of
congestion on the access links. Many access links routinely suffer
from congestion. It is important to avoid congestion collapse along
the entire end-to-end path, including along the access links (where
congestion collapse would consist of congested access links wasting
scarce bandwidth carrying packets that will only be dropped
downstream). So an over-provisioned core does not by itself
eliminate or reduce the need for end-to-end congestion avoidance and
control.
There are two possible mechanisms for avoiding this congestion
collapse: call rejection during busy periods, or the use of end-to-
end congestion control. Because there are currently no
acceptance/rejection mechanisms for best-effort traffic in the
Internet, the only alternative is the use of end-to-end congestion
control. This is important even if end-to-end congestion control is
invoked only in those very rare scenarios with congestion in
generally-uncongested access links or networks. There will always be
occasional periods of high demand, e.g., in the two hours after an
earthquake or other disaster, and this is exactly when it is
important to avoid congestion collapse.
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Best-effort traffic in the Internet does not include mechanisms for
call acceptance or rejection. Instead, a best-effort network itself
is largely neutral in terms of resource management, and the
interaction of the applications' transport sessions mutually
regulates network resources in a reasonably fair fashion. One way to
bring voice into the best-effort environment in a non-disruptive
manner is to focus on the codec and look at rate adaptation measures
that can successfully interoperate with existing transport protocols
(e.g., TCP), while at the same time preserving the integrity of a
real-time, analog voice signal; another way is to consider codecs
with fixed sending rates. Whether the codec has a fixed or variable
sending rate, we consider the appropriate response when the codec is
at its minimum data rate, and the packet drop rate experienced by the
flow remains high. This is the key issue addressed in this document.
3. Why are Persistent, High Drop Rates a Problem?
Persistent, high packet drop rates are rarely seen in the Internet
today, in the absence of routing failures or other major disruptions.
This happy situation is due primarily to low levels of link
utilization in the core, with congestion typically found on lower-
capacity access links, and to the use of end-to-end congestion
control in TCP. Most of the traffic on the Internet today uses TCP,
and TCP self-corrects so that the two ends of a connection reduce the
rate of packet sending if congestion is detected. In the sections
below, we discuss some of the problems caused by persistent, high
packet drop rates.
3.1. Congestion Collapse
One possible problem caused by persistent, high packet drop rates is
that of congestion collapse. Congestion collapse was first observed
during the early growth phase of the Internet of the mid 1980s
[RFC896], and the fix was provided by Van Jacobson, who developed the
congestion control mechanisms that are now required in TCP
implementations [Jacobson88, RFC2581].
As described in RFC 2914, congestion collapse occurs in networks with
flows that traverse multiple congested links having persistent, high
packet drop rates [RFC2914]. In particular, in this scenario packets
that are injected onto congested links squander scarce bandwidth
since these packets are only dropped later, on a downstream congested
link. If congestion collapse occurs, all traffic slows to a crawl
and nobody gets acceptable packet delivery or acceptable performance.
Because congestion collapse of this form can occur only for flows
that traverse multiple congested links, congestion collapse is a
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potential problem in VoIP networks when both ends of the VoIP call
are on an congested broadband connection such as DSL, or when the
call traverses a congested backbone or transoceanic link.
3.2. User Quality
A second problem with persistent, high packet drop rates concerns
service quality seen by end users. Consider a network scenario where
each flow traverses only one congested link, as could have been the
case in the Nairobi demonstration above. For example, imagine N VoIP
flows sharing a 128 Kbps link, with each flow sending at least 64
Kbps. For simplicity, suppose the 128 Kbps link is the only
congested link, and there is no traffic on that link other than the N
VoIP calls. We will also ignore for now the extra bandwidth used by
the telephony traffic for FEC and packet headers, or the reduced
bandwidth (often estimated as 70%) due to silence suppression. We
also ignore the fact that the two streams composing a bidirectional
VoIP call, one for each direction, can in practice add to the load on
some links of the path. Given these simplified assumptions, the
arrival rate to that link is at least N*64 Kbps. The traffic
actually forwarded is at most 2*64 Kbps (the link bandwidth), so at
least (N-2)*64 Kbps of the arriving traffic must be dropped. Thus, a
fraction of at least (N-2)/N of the arriving traffic is dropped, and
each flow receives on average a fraction 1/N of the link bandwidth.
An important point to note is that the drops occur randomly, so that
no one flow can be expected statistically to present better quality
service to users than any other. Everybody's voice quality therefore
suffers.
It seems clear from this simple example that the quality of best-
effort VoIP traffic over congested links can be improved if each VoIP
flow uses end-to-end congestion control, and has a codec that can
adapt the bit rate to the bandwidth actually received by that flow.
The overall effect of these measures is to reduce the aggregate
packet drop rate, thus improving voice quality for all VoIP users on
the link. Today, applications and popular codecs for Internet
telephony attempt to compensate by using more FEC, but controlling
the packet flow rate directly should result in less redundant FEC
information, and thus less bandwidth, thereby improving throughput
even further. The effect of delay and packet loss on VoIP in the
presence of FEC has been investigated in detail in the literature
[JS00, JS02, JS03, MTK03]. One rule of thumb is that when the packet
loss rate exceeds 20%, the audio quality of VoIP is degraded beyond
usefulness, in part due to the bursty nature of the losses [S03]. We
are not aware of measurement studies of whether VoIP users in
practice tend to hang up when packet loss rates exceed some limit.
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The simple example in this section considered only voice flows, but
in reality, VoIP traffic will compete with other flows, most likely
TCP. The response of VoIP traffic to congestion works best by taking
into account the congestion control response of TCP, as is discussed
in the next subsection.
3.3. The Amorphous Problem of Fairness
A third problem with persistent, high packet drop rates is fairness.
In this document we consider fairness with regard to best-effort VoIP
traffic competing with other best-effort traffic in the Internet.
That is, we are explicitly not addressing the issues raised by
emergency services, or by QoS-enabled traffic that is known to be
treated separately from best-effort traffic at a congested link.
While fairness is a bit difficult to quantify, we can illustrate the
effect by adding TCP traffic to the congested link discussed in the
previous section. In this case, the non-congestion-controlled
traffic and congestion-controlled TCP traffic [RFC2914] share the
link, with the congestion-controlled traffic's sending rate
determined by the packet drop rate experienced by those flows. As in
the previous section, the 128 Kbps link has N VoIP connections each
sending 64 Kbps, resulting in packet drop rate of at least (N-2)/N on
the congested link. Competing TCP flows will experience the same
packet drop rates. However, a TCP flow experiencing the same packet
drop rates will be sending considerably less than 64 Kbps. From the
point of view of who gets what amount of bandwidth, the VoIP traffic
is crowding out the TCP traffic.
Of course, this is only one way to look at fairness. The relative
fairness between VoIP and TCP traffic can be viewed several different
ways, depending on the assumptions that one makes on packet sizes and
round-trip times. In the presence of a fixed packet drop rate, for
example, a TCP flow with larger packets sends more (in Bps, bytes per
second) than a TCP flow with smaller packets, and a TCP flow with a
shorter round-trip time sends more (in Bps) than a TCP flow with a
larger round-trip time. In environments with high packet drop rates,
TCP's sending rate depends on the algorithm for setting the
retransmit timer (RTO) as well, with a TCP implementation having a
more aggressive RTO setting sending more than a TCP implementation
having a less aggressive RTO setting.
Unfortunately, there is no obvious canonical round-trip time for
judging relative fairness of flows in the network. Agreement in the
literature is that the majority of packets on most links in the
network experience round-trip times between 10 and 500 ms [RTTWeb].
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(This does not include satellite links.) As a result, if there was a
canonical round-trip for judging relative fairness, it would have to
be within that range. In the absence of a single representative
round-trip time, the assumption of this paper is that it is
reasonable to consider fairness between a VoIP connection and a TCP
connection with the same round-trip time.
Similarly, there is no canonical packet size for judging relative
fairness between TCP connections. However, because the most common
packet size for TCP data packets is 1460 bytes [Measurement], we
assume that it is reasonable to consider fairness between a VoIP
connection, and a TCP connection sending 1460-byte data packets.
Note that 1460 bytes is considerably larger than is typically used
for VoIP packets.
In the same way, while RFC 2988 specifies TCP's algorithm for setting
TCP's RTO, there is no canonical value for the minimum RTO, and the
minimum RTO heavily affects TCP's sending rate in times of high
congestion [RFC2988]. RFC 2988 specifies that TCP's RTO must be set
to SRTT + 4*RTTVAR, for SRTT the smoothed round-trip time, and for
RTTVAR the mean deviation of recent round-trip time measurements.
RFC 2988 further states that the RTO "SHOULD" have a minimum value of
1 second. However, it is not uncommon in practice for TCP
implementations to have a minimum RTO as low as 100 ms. For the
purposes of this document, in considering relative fairness, we will
assume a minimum RTO of 100 ms.
As an additional complication, TCP connections that use fine-grained
timestamps can have considerably higher sending rates than TCP
connections that do not use timestamps, in environments with high
packet drop rates. For TCP connections with fine-grained timestamps,
a valid round-trip time measurement is obtained when a retransmitted
packet is successfully received and acknowledged by the receiver; in
this case a backed-off retransmit timer can be un-backed-off as well.
For TCP connections without timestamps, a valid round-trip time
measurement is only obtained when the transmission of a new packet is
received and acknowledged by the receiver. This limits the
opportunities for the un-backing-off of a backed-off retransmit
timer. In this document, in considering relative fairness, we use a
TCP connection without timestamps, since this is the dominant use of
TCP in the Internet.
A separate claim that has sometimes been raised in terms of fairness
is that best-effort VoIP traffic is inherently more important that
other best-effort traffic (e.g., web surfing, peer-to-peer traffic,
or multi-player games), and therefore merits a larger share of the
bandwidth in times of high congestion. Our assumption in this
document is that TCP traffic includes pressing email messages,
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business documents, and emergency information downloaded from web
pages, as well as the more recreational uses cited above. Thus, we
do not agree that best-effort VoIP traffic should be exempt from
end-to-end congestion control due to any claims of inherently more
valuable content. (One could equally logically argue that because
email and instant messaging are more efficient forms of communication
than VoIP in terms of bandwidth usage, as a result email and instant
messaging are more valuable uses of scarce bandwidth in times of high
congestion.) In fact, the network is incapable of making a judgment
about the relative user value of traffic. The default assumption is
that all best-effort traffic has equal value to the network provider
and to the user.
We note that this discussion of relative fairness does not in any way
challenge the right of ISPs to allocate bandwidth on congested links
to classes of traffic in any way that they choose. (For example,
administrators rate-limit the bandwidth used by peer-to-peer traffic
on some links in the network, to ensure that bandwidth is also
available for other classes of traffic.) This discussion merely
argues that there is no reason for entire classes of best-effort
traffic to be exempt from end-to-end congestion control.
4. Current efforts in the IETF
There are four efforts currently underway in IETF to address issues
of congestion control for real time traffic: an upgrade of the RTP
specification, TFRC, DCCP, and work on audio codecs.
4.1. RTP
RFC 1890, the original RTP Profile for Audio and Video Control, does
not discuss congestion control [RFC1890]. The revised document on
"RTP Profile for Audio and Video Conferences with Minimal Control"
[RFC3551] discusses congestion control in Section 2. [RFC3551] says
the following:
"If best-effort service is being used, RTP receivers SHOULD
monitor packet loss to ensure that the packet loss rate is within
acceptable parameters. Packet loss is considered acceptable if a
TCP flow across the same network path and experiencing the same
network conditions would achieve an average throughput, measured
on a reasonable timescale, that is not less than the RTP flow is
achieving. This condition can be satisfied by implementing
congestion control mechanisms to adapt the transmission rate (or
the number of layers subscribed for a layered multicast session),
or by arranging for a receiver to leave the session if the loss
rate is unacceptably high."
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"The comparison to TCP cannot be specified exactly, but is
intended as an "order-of-magnitude" comparison in timescale and
throughput. The timescale on which TCP throughput is measured is
the round-trip time of the connection. In essence, this
requirement states that it is not acceptable to deploy an
application (using RTP or any other transport protocol) on the
best-effort Internet which consumes bandwidth arbitrarily and does
not compete fairly with TCP within an order of magnitude."
Note that [RFC3551] says that receivers "SHOULD" monitor packet loss.
[RFC3551] does not explicitly say that the RTP senders and receivers
"MUST" detect and respond to a persistent high loss rate. Since
congestion collapse can be considered a "danger to the Internet" the
use of "MUST" would be appropriate for RTP traffic in the best-effort
Internet, where the VoIP traffic shares a link with other traffic,
since "danger to the Internet" is one of two criteria given in RFC
2119 for the use of "MUST" [RFC2119]. Different requirements may
hold for a private best-effort IP network provisioned solely for
VoIP, where the VoIP traffic does not interact with the wider
Internet.
4.2. TFRC
As mentioned in RFC 3267, equation-based congestion control is one of
the possibilities for VoIP. TCP Friendly Rate Control (TFRC) is the
equation-based congestion control mechanism that has been
standardized in the IETF. The TFRC specification, "TCP Friendly Rate
Control (TFRC): Protocol Specification" [RFC3448], says the
following:
"TFRC ... is reasonably fair when competing for bandwidth with TCP
flows, but has a much lower variation of throughput over time
compared with TCP, making it more suitable for applications such
as telephony or streaming media where a relatively smooth sending
rate is of importance. ... TFRC is designed for applications
that use a fixed packet size, and vary their sending rate in
packets per second in response to congestion. Some audio
applications require a fixed interval of time between packets and
vary their packet size instead of their packet rate in response to
congestion. The congestion control mechanism in this document
cannot be used by those applications; TFRC-PS (for TFRC-
PacketSize) is a variant of TFRC for applications that have a
fixed sending rate but vary their packet size in response to
congestion. TFRC-PS will be specified in a later document."
There is no draft available for TFRC-PS yet, unfortunately, but
several researchers are still working on these issues.
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4.3. DCCP
The Datagram Congestion Control Protocol (DCCP) is a transport
protocol being standardized in the IETF for unreliable flows, with
the application being able to specify either TCP-like or TFRC
congestion control [DCCP03].
DCCP currently has two Congestion Control IDentifiers or CCIDs; these
are CCID 2 for TCP-like congestion control and CCID 3 for TFRC
congestion control. As TFRC-PS becomes available and goes through
the standards process, we would expect DCCP to create a new CCID,
CCID 4, for use with TFRC-PS congestion control.
4.4. Adaptive Rate Audio Codecs
A critical component in the design of any real-time application is
the selection of appropriate codecs, specifically codecs that operate
at a low sending rate, or that will reduce the sending rate as
throughput decreases and/or packet loss increases. Absent this, and
in the absence of the response to congestion recommended in this
document, the real-time application is likely to significantly
increase the risk of Internet congestion collapse, thereby adversely
impacting the health of the deployed Internet. If the codec is
capable of reducing its bit rate in response to congestion, this
improves the scaling of the number of VoIP or TCP sessions capable of
sharing a congested link while still providing acceptable performance
to users. Many current audio codecs are capable of sending at a low
bit rate, in some cases adapting their sending rate in response to
congestion indications from the network.
RFC 3267 describes RTP payload formats for use with the Adaptive
Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) audio
codecs [RFC 3267]. The AMR codec supports eight speech encoding
modes having bit rates between 4.75 and 12.2 kbps, with the speech
encoding performed on 20 ms speech frames, and is able to reduce the
transmission rate during silence periods. The payload format
specified in RFC 3267 includes forward error correction (FEC) and
frame interleaving to increase robustness against packet loss
somewhat. The AMR codec was chosen by the Third Generation
Partnership Project (3GPP) as the mandatory codec for third
generation (3G) cellular systems, and RFC 3267 recommends that AMR or
AMR-WB applications using the RTP payload format specified in RFC
3267 use congestion control, though no specific mechanism is
recommended. RFC 3267 gives "Equation-Based Congestion Control for
Unicast Applications" as an example of a congestion control mechanism
suitable for real-time flows [FHPW00].
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The "Internet Low Bit Rate Codec", iLBC, is an IETF effort to develop
an IPR-free codec for robust voice communication over IP [ILBRC].
The codec is designed for graceful speech quality degradation in the
case of lost packets, and has a payload bit rate of 13.33 kbps for 30
ms frames or 15.20 kbps for 20 ms frames.
There are several unencumbered low-rate codec algorithms in Ivox (the
Interactive VOice eXchange) [IVOX], with plans to add additional
variable rate codecs. For example, LPC2400 (a.k.a. LQ2400) is a 2400
bps LPC based codec with an enhancement to permit "silence
detection". The 2400 bps codec is reported to have a "slight robotic
quality" [A03] (even without the additional complications of packet
loss). The older multirate codec described in [KFK79, KF82] is an
LPC codec that works at two rates, 2.4 kbps and 9.6 kbps, and can
optionally send additional "residual" bits for enhanced quality at a
higher bit rate.
Off-the-shelf ITU-T vocoders such as G.711 were generally designed
explicitly for circuit-switched networks and are not as well-adapted
for Internet use, even with the addition of FEC on top.
4.5. Differentiated Services and Related Topics
The Differentiated Services Working Group [DIFFSERV], which concluded
in 2003, completed standards for the Differentiated Services Field
(DS Field) in the IPv4 and IPv6 Headers [RFC2474], including several
per-hop forwarding behaviors [RFC2597, RFC3246]. The Next Steps in
Signaling Working Group [NSIS] is developing an optimized signalling
protocol for QoS, based in part on earlier work of the Resource
Reservation Setup Protocol Working Group [RSVP]. We do not discuss
these and related efforts further in this document, since this
document concerns only that VoIP traffic that might be carried as
best-effort traffic over some congested link in the Internet.
5. Assessing Minimum Acceptable Sending Rates
Current IETF work in the DCCP and AVT working groups does not
consider the problem of applications that have a minimum sending rate
and are not able to go below that sending rate. This clearly must be
addressed in the TFRC-PS draft. As suggested in the RTP document, if
the loss rate is persistently unacceptably high relative to the
current sending rate, and the best-effort application is unable to
lower its sending rate, then the only acceptable answer is for that
flow to discontinue sending on that link. For a multicast session,
this could be accomplished by the receiver withdrawing from the
multicast group. For a unicast session, this could be accomplished
by the unicast connection terminating, at least for a period of time.
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We can formulate a problem statement for the minimum sending rate in
the following way. Consider a best-effort, adaptive audio
application that is able to adapt down to a minimum sending rate of N
Bps (bytes per second) of application data, sending M packets per
second. Is this a sufficiently low sending rate that the best-effort
flow is never required to terminate due to congestion, or to reduce
its sending rate in packets per second still further? In other words,
is N Bps an acceptable minimum sending rate for the application,
which can be continued in the face of congestion without terminating
or suspending the application?
We assume, generously for VoIP, that the limitation of the network is
in bandwidth in bytes per second (Bps), and not in CPU cycles or in
packets per second (pps). If the limitation in the network is in
bandwidth, this is a limitation in Bps, while if the limitation is in
router processing capacity in packets, this would be a limitation in
pps. We note that TCP sends fixed-size data packets, and reduces its
sending rate in pps when it adapts to network congestion, thus
reducing the load on the forward path both in Bps and in pps. In
contrast, for adaptive VoIP applications, the adaption is sometimes
to keep the same sending rate in pps, but to reduce the packet size,
reducing the sending rate in Bps. This fits the needs of audio as an
application, and is a good response on a network path where the
limitation is in Bps. Such behavior would be a less appropriate
response for a network path where the limitation is in pps.
If the network limitation in fact is in Bps, then all that matters in
terms of congestion is a flow's sending rate on the wire in Bps. If
this assumption of a network limitation in Bps is false, then the
sending rate in pps could contribute to congestion even when the
sending rate in Bps is quite moderate. While the ideal would be to
have a transport protocol that is able to detect whether the
bottleneck links along the path are limited in Bps or in pps, and to
respond appropriately when the limitation is in pps, such an ideal is
hard to achieve. We would not want to delay the deployment of
congestion control for telephony traffic until such an ideal could be
accomplished. In addition, we note that the current TCP congestion
control mechanisms are themselves not very effective in an
environment where there is a limitation along the reverse path in
pps. While the TCP mechanisms do provide an incentive to use large
data packets, TCP does not include any effective congestion control
mechanisms for the stream of small acknowledgement packets on the
reverse path. Given the arguments above, it seems acceptable to us
to assume a network limitation in Bps rather than in pps in
considering the minimum sending rate of telephony traffic.
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Assuming 40-byte packet headers (IP, RTP, and UDP or DCCP), the
application data sending rate of N Bps and M pps translates to a
sending rate on the wire of B = N+40M Bps. If the application uses
additional FEC (Forward Error Correction), the FEC bits must be added
in as well. In our example, we ignore bandwidth adjustments that are
needed to take into account the additional overhead for FEC or the
reduced sending rate for silence periods. We also are not taking
into account the possible role of header compression on congested
edge links, which can reduce significantly the number of bytes used
for headers on those links.
Now, consider an equivalent-rate TCP connection with data packets of
P bytes and a round-trip time of R seconds. Taking into account
header size, such a TCP connection with a sending rate on the wire of
B Bps is sending B/(P+40) pps, or, equivalently, BR/(P+40) ppr
(packets per round-trip time).
Restating the question in terms of the above expressions for VoIP and
TCP: if the best-effort VoIP connection is experiencing a persistent
packet drop rate of D, and is at its minimum sending rate on the wire
of B Bps, when should the application or transport protocol terminate
or suspend the VoIP connection?
One answer to this question is to find the sending rate in ppr for a
TCP connection sending at the same rate on the wire in Bps, and to
use the TCP response function to determine whether a conformant TCP
connection would be able to maintain a sending rate close to that
sending rate with the same persistent drop rate D. If the sending
rate of the VoIP connection is significantly higher than the sending
rate of a conformant TCP connection under the same conditions, and
the VoIP connection is unable to reduce its sending rate on the wire,
then the VoIP connection should terminate or suspend.
As discussed above, there are two reasons for requiring the
application to terminate:
1) Avoiding congestion collapse, given the possibility of multiple
congested links,
2) Fairness for congestion-controlled TCP traffic sharing the
link.
In addition, if an application requires a minimum service level from
the network in order to operate, and that service level is
consistently not achieved, then the application should terminate or
suspend sending.
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One counter-argument is that users will just hang up anyway with a
high packet drop rate so there is no point in enforcing a minimum
acceptable rate. Users might hang up, but they might also just keep
on talking, with the occasional noise getting though, for minutes or
longer waiting for a short period of clarity. Another counter-
argument is that nobody really benefits from VoIP connections being
terminated or suspended when persistent packet drop rates exceed the
allowable packet drop rate for the configured minimum sending rate.
This is untrue, since the termination of these VoIP connections could
allow competing TCP and VoIP traffic to make some progress.
In the next section, we illustrate the approach outlined above for
VoIP flows with minimum sending rates of 4.75 and 64 kbps
respectively, and show that in practice such an approach would not
seem too burdensome for VoIP traffic. This approach implies that the
VoIP traffic would terminate or suspend when the packet drop rate
significantly exceeds 40% for a VoIP flow with a minimum sending rate
of 4.75 kbps. If VoIP is to deliver "carrier quality" or even near
"carrier quality" on best-effort links, conditioning deployment on
the ability to maintain maximum sending rates during periods of
persistent packet drops rates exceeding 40% does not suggest a
service model that will see widespread acceptance among consumers, no
matter what the price differential. Good packet throughput is vital
for the delivery of acceptable VoIP service.
For a VoIP flow that stops sending because its minimum sending rate
is too high for the steady-state packet drop rate, we have not
addressed the question of when a VoIP flow might be able to start
sending again, to see if the congestion on the end-to-end path has
changed. This issue has been addressed in a proposal for
Probabilistic Congestion Control [PCC].
We note that if the congestion indications are in the form of ECN-
marked packets (Explicit Congestion Notification), as opposed to
dropped packets, then the answers about when a flow with a minimum
sending rate would have to stop sending are somewhat different. ECN
allows routers to explicitly notify end-nodes of congestion by ECN-
marking instead of dropping packets [RFC3168]. If packets are ECN-
marked instead of dropped in the network, then there are no concerns
of congestion collapse or of user quality (for the ECN-capable
traffic, at any rate), and what remains are concerns of fairness with
competing flows. Second, in regimes with very high congestion, TCP
has a higher sending rate with ECN-marked than with dropped packets,
in part because of different dynamics in terms of un-backing-off a
backed-off retransmit timer.
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5.1. Drop Rates at 4.75 kbps Minimum Sending Rate
Consider an adaptive audio application with an RTT of R=0.1 seconds
that is able to adapt down to a minimum sending rate of 4.75 kbps
application data, sending M=20 packets per second. This sending rate
translates to N=593 Bps of application data, for a sending rate on
the wire of B=1393 Bps. An equivalent-rate TCP connection with data
packets of P=1460 bytes and a round-trip time of R=0.1 seconds would
be sending BR/(P+40) = 0.09 ppr.
Table 1 in the Appendix looks at the packet drop rate experienced by
a TCP connection with the RTO set to twice the RTT, and gives the
corresponding sending rate of the TCP connection in ppr. The second
column gives the sending rate estimated by the standard analytical
approach, and the third, fourth, and fifth columns give the average
sending rate from simulations with random packet drops or marks. The
sixth column gives the sending rates from experiments on a 4.8-
RELEASE FreeBSD machine. The analytical approaches require an RTO
expressed as a multiple of the RTT, and Table 1 shows the results for
the RTO set to 2 RTT. In the simulations, the minimum RTO is set to
twice the RTT. See the Appendix for more details.
For a sending rate of 0.09 ppr and an RTO set to 2 RTT, Table 1 shows
that the analytical approach gives a corresponding packet drop rate
of roughly 50%, while the simulations in the fifth column and the
experiments in the sixth column give a packet drop rate of between
35% and 40% to maintain a sending rate of 0.09 ppr. (For a reference
TCP connection using timestamps, shown in the fourth column, the
simulations give a packet drop rate of 55% to maintain a sending rate
of 0.09 ppr.) Of the two approaches for determining TCP's
relationship between the sending rate and the packet drop rate, the
analytic approach and the use of simulations, we consider the
simulations to be the most realistic, for reasons discussed in the
Appendix. This suggests a packet drop rate of 40% would be
reasonable for a TCP connection with an average sending rate of 0.09
ppr. As a result, a VoIP connection with an RTT of 0.1 sec and a
minimum sending rate of 4.75 kbps would be required to terminate or
suspend when the persistent packet drop rate significantly exceeds
40%.
These estimates are sensitive to the assumed round-trip time of the
TCP connection. If we assumed instead that the equivalent-rate TCP
connection had a round-trip time of R=0.01 seconds, the equivalent-
rate TCP connection would be sending BR/(P+40) = 0.009 ppr. However,
we have also assumed a minimum RTO for TCP connections of 0.1
seconds, which in this case would mean an RTO of at least 10 RTT.
For this setting of the RTO, we would use Table 2 from the appendix
to determine the average TCP sending rate for a particular packet
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drop rate. The simulations in the fifth column of Table 2 suggest
that a TCP connection with an RTT of 0.01 sec and an RTO of 10 RTT
would be able to send 0.009 ppr with a packet drop rate of 45%. (For
the same TCP connection using timestamps, shown in the fourth column,
the simulations give a packet drop rate of 60-65% to maintain a
sending rate of 0.009 ppr.)
Thus, for a VoIP connection with an RTT of 0.01 sec and a minimum
sending rate of 4.75 kbps, the VoIP connection would be required to
terminate or suspend when the persistent packet drop rate exceeded
45%.
5.2. Drop Rates at 64 kbps Minimum Sending Rate
The effect of increasing the minimum acceptable sending rate to 64
kbps is effectively to decrease the packet drop rate at which the
application should terminate or suspend sending. For this section,
consider a codec with a minimum sending rate of 64 kbps, or N=8000
Bps, and a packet sending rate of M=50 pps. (This would be
equivalent to 160-byte data packets, with 20 ms. per packet.) The
sending rate on the wire is B = N+40M Bps, including headers, or
10000 Bps. A TCP connection having that sending rate, with packets
of size P=1460 bytes and a round-trip time of R=0.1 seconds, sends
BR/(P+40) = 0.66 ppr. From the fifth column of Table 1, for an RTO
of 2 RTT, this corresponds to a packet drop rate between 20 and 25%.
[For a TCP connection using fine-grained timestamps, as shown in the
fourth column of Table 1, this sending rate corresponds to a packet
drop rate between 25% and 35%.] As a result, a VoIP connection with
an RTT of 0.1 sec and a minimum sending rate of 64 kbps would be
required to terminate or suspend when the persistent packet drop rate
significantly exceeds 25%.
For an equivalent-rate TCP connection with a round-trip time of
R=0.01 seconds and a minimum RTO of 0.1 seconds (giving an RTO of 10
RTT), we use the fifth column of Table 2, which shows that a sending
rate of 0.066 ppr corresponds to a packet drop rate of roughly 30%.
[For a TCP connection using fine-grained timestamps, as shown in the
fourth column of Table 2, this sending rate corresponds to a packet
drop rate of roughly 45%.] Thus, for a VoIP connection with an RTT
of 0.01 sec and a minimum sending rate of 64 kbps, the VoIP
connection would be required to terminate or suspend when the
persistent packet drop rate exceeded 30%.
5.3. Open Issues
This document does not attempt to specify a complete protocol. For
example, this document does not specify the definition of a
persistent packet drop rate. The assumption would be that a
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"persistent packet drop rate" would refer to the packet drop rate
over a significant number of round-trip times, e.g., at least five
seconds. Another possibility would be that the time interval for
measuring the persistent drop rate is a function of the lifetime of
the connection, with longer-lived connections using longer time
intervals for measuring the persistent drop rate.
The time period for detecting persistent congestion also affects the
potential synchronization of VoIP sessions all terminating or
suspending at the same time in response to shared congestion. If
flows use some randomization in setting the time interval for
detecting persistent congestion, or use a time interval that is a
function of the connection lifetime, this could help to prevent all
VoIP flows from terminating at the same time.
Another design issue for a complete protocol concerns whether a flow
terminates when the packet drop rate is too high, or only suspends
temporarily. For a flow that suspends temporarily, there is an issue
of how long it should wait before resuming transmission. At the very
least, the sender should wait long enough so that the flow's overall
sending rate doesn't exceed the allowed sending rate for that packet
drop rate.
The recommendation of this document is that VoIP flows with minimum
sending rates should have corresponding configured packet drop rates,
such that the flow terminates or suspends when the persistent packet
drop rate of the flow exceeds the configured rate. If the persistent
packet drop rate increases over time, flows with higher minimum
sending rates would have to suspend sending before flows with lower
minimum sending rates. If VoIP flows terminate when the persistent
packet drop rate is too high, this could lead to scenarios where VoIP
flows with lower minimum sending rates essentially receive all of the
link bandwidth, while the VoIP flows with higher minimum sending
rates are required to terminate. However, if VoIP flows suspend
sending for a time when the persistent packet drop rate is too high,
instead of terminating entirely, then the bandwidth could end up
being shared reasonably fairly between VoIP flows with different
minimum sending rates.
5.4. A Simple Heuristic
One simple heuristic for estimating congestion would be to use the
RTCP reported loss rate as an indicator. For example, if the RTCP-
reported lost rate is greater than 30%, or N back-to-back RTCP
reports are missing, the application could assume that the network is
too congested, and terminate or suspend sending.
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6. Constraints on VoIP Systems
Ultimately, attempting to run VoIP on congested links, even with
adaptive rate codecs and minimum packet rates, is likely to run into
hard constraints due to the nature of real time traffic in heavily
congested scenarios. VoIP systems exhibit a limited ability to scale
their packet rate. If the number of packets decreases, the amount of
audio per packet is greater and error concealment at the receiver
becomes harder. Any error longer than phoneme length, which is
typically 40 to 100 ms depending on the phoneme and speaker, is
unrecoverable. Ideally, applications want sub 30ms packets and this
is what most voice codecs provide. In addition, voice media streams
exhibit greater loss sensitivity at lower data rates. Lower-data
rate codecs maintain more end-to-end state and as a result are
generally more sensitive to loss.
We note that very-low-bit-rate codecs have proved useful, although
with some performance degradation, in very low bandwidth, high noise
environments (e.g., 2.4 kbps HF radio). For example, 2.4 kbps codecs
"produce speech which although intelligible is far from natural
sounding" [W98]. Figure 5 of [W98] shows how the speech quality with
several forms of codecs varies with the bit rate of the codec.
7. Conclusions and Recommendations
In the near term, VoIP services are likely to be deployed, at least
in part, over broadband best-effort connections. Current real time
media encoding and transmission practice ignores congestion
considerations, resulting in the potential for trouble should VoIP
become a broadly deployed service in the near to intermediate term.
Poor user quality, unfairness to other VoIP and TCP users, and the
possibility of sporadic episodes of congestion collapse are some of
the potential problems in this scenario.
These problems can be mitigated in applications that use fixed-rate
codecs by requiring the best-effort VoIP application to specify its
minimum bit throughput rate. This minimum bit rate can be used to
estimate a packet drop rate at which the application would terminate.
This document specifically recommends the following:
(1) In IETF standards for protocols regarding best-effort flows with
a minimum sending rate, a packet drop rate must be specified, such
that the best-effort flow terminates, or suspends sending
temporarily, when the steady-state packet drop rate significantly
exceeds the specified drop rate.
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(2) The specified drop rate for the minimum sending rate should be
consistent with the use of Tables 1 and 2 as illustrated in this
document.
We note that this is a recommendation to the IETF community, as a
specific follow-up to RFC 2914 on Congestion Control Principles.
This is not a specific or complete protocol specification.
Codecs that are able to vary their bit rate depending on estimates of
congestion can be even more effective in providing good quality
service while maintaining network efficiency under high load
conditions. Adaptive variable-bit-rate codecs are therefore
preferable as a means of supporting VOIP sessions on shared use
Internet environments.
Real-time traffic such as VoIP could derive significant benefits from
the use of ECN, where routers may indicate congestion to end-nodes by
marking packets instead of dropping them. However, ECN is only
standardized to be used with transport protocols that react
appropriately to marked packets as indications of congestion. VoIP
traffic that follows the recommendations in this document could
satisfy the congestion-control requirements for using ECN, while VoIP
traffic with no mechanism for terminating or suspending when the
packet dropping and marking rate was too high would not. However, we
repeat that this document is not a complete protocol specification.
In particular, additional mechanisms would be required before it was
safe for applications running over UDP to use ECN. For example,
before using ECN, the sending application would have to ensure that
the receiving application was capable of receiving ECN-related
information from the lower-layer UDP stack, and of interpreting this
ECN information as a congestion indication.
8. Acknowledgements
We thank Brian Adamson, Ran Atkinson, Fred Baker, Jon Crowcroft,
Christophe Diot, Alan Duric, Jeremy George, Mark Handley, Orion
Hodson, Geoff Huston, Eddie Kohler, Simon Leinen, David Meyer, Jean-
Francois Mule, Colin Perkins, Jon Peterson, Mike Pierce, Cyrus
Shaoul, and Henning Schulzrinne for feedback on this document. (Of
course, these people do not necessarily agree with all of the
document.) Ran Atkinson and Geoff Huston contributed to the text of
the document.
The analysis in Section 6.0 resulted from a session at the whiteboard
with Mark Handley. We also thank Alberto Medina for the FreeBSD
experiments showing TCP's sending rate as a function of the packet
drop rate.
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9. References
9.1. Normative References
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119, March 1997.
[RFC2988] Paxson, V. and M. Allman, "Computing TCP's
Retransmission Timer", RFC 2988, November 2000.
[RFC3267] Sjoberg, J., Westerlund, M., Lakaniemi, A. and Q. Xie,
"Real-Time Transport Protocol (RTP) Payload Format and
File Storage Format for the Adaptive Multi-Rate (AMR)
and Adaptive Multi-Rate Wideband (AMR-WB) Audio
Codecs", RFC 3267, June 2002.
9.2. Informative References
[A02] Ran Atkinson, An ISP Reality Check, Presentation to
ieprep, 55th IETF Meeting, November 2002. URL
"http://www.ietf.cnri.reston.va.us/proceedings/
02nov/219.htm#slides".
[A03] Brian Adamson, private communication, June 2003.
[BBFS01] Deepak Bansal, Hari Balakrishnan, Sally Floyd, and
Scott Shenker, Dynamic Behavior of Slowly-Responsive
Congestion Control Algorithms, SIGCOMM 2001.
[COPS] Durham, D., Ed., Boyle, J., Cohen, R., Herzog, S.,
Rajan, R. and A. Sastry, "The COPS (Common Open Policy
Service) Protocol", RFC 2748, January 2000.
[DCCP03] Eddie Kohler, Mark Handley, Sally Floyd, and Jitendra
Padhye, Datagram Congestion Control Protocol (DCCP),
internet-draft Work in Progress, March 2003. URL
"http://www.icir.org/kohler/dcp/".
[DIFFSERV] Differentiated Services (diffserv), Concluded Working
Group, URL
"http://www.ietf.cnri.reston.va.us/html.charters/
OLD/diffserv-charter.html".
[E2E] The end2end-interest mailing list, URL
"http://www.postel.org/mailman/listinfo/end2end-
interest".
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[FHPW00] S. Floyd, M. Handley, J. Padhye, J. Widmer, "Equation-
Based Congestion Control for Unicast Applications", ACM
SIGCOMM 2000.
[FM03] S. Floyd and R. Mahajan, Router Primitives for
Protection Against High-Bandwidth Flows and Aggregates,
internet draft (not yet submitted).
[FWD] Free World Dialup, URL "www.pulver.com/fwd/".
[IEPREP02] Internet Emergency Preparedness (ieprep), Minutes, 55th
IETF Meeting, November 2002. URL
"http://www.ietf.cnri.reston.va.us/proceedings/
02nov/219.htm#cmr".
[ILBRC] S.V. Andersen, et. al., Internet Low Bit Rate Codec,
Work in Progress, March 2003.
[G.114] Recommendation G.114 - One-way Transmission Time, ITU,
May 2003. URL "http://www.itu.int/itudoc/itu-
t/aap/sg12aap/recaap/g.114/".
[IVOX] The Interactive VOice eXchange, URL
"http://manimac.itd.nrl.navy.mil/IVOX/".
[Jacobson88] V. Jacobson, Congestion Avoidance and Control, ACM
SIGCOMM '88, August 1988.
[AUT] The maximum feasible drop rate for VoIP traffic depends
on the codec. These numbers are a range for a variety
of codecs; voice quality begins to deteriorate for many
codecs around a 10% drop rate. Note from authors.
[JS00] Wenyu Jiang and Henning Schulzrinne, Modeling of Packet
Loss and Delay and Their Effect on Real-Time Multimedia
Service Quality, NOSSDAV, 2000. URL
"http://citeseer.nj.nec.com/jiang00modeling.html".
[JS02] Wenyu Jiang and Henning Schulzrinne, Comparison and
Optimization of Packet Loss Repair Methods on VoIP
Perceived Quality under Bursty Loss, NOSSDAV, 2002.
URL "http://www1.cs.columbia.edu/~wenyu/".
[JS03] Wenyu Jiang, Kazummi Koguchi, and Henning Schulzrinne,
QoS Evaluation of VoIP End-points, ICC 2003. URL
"http://www1.cs.columbia.edu/~wenyu/".
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[KFK79] G.S. Kang, L.J. Fransen, and E.L. Kline, "Multirate
Processor (MRP) for Digital Voice Communications", NRL
Report 8295, Naval Research Laboratory, Washington DC,
March 1979.
[KF82] G.S. Kang and L.J. Fransen, "Second Report of the
Multirate Processor (MRP) for Digital Voice
Communications", NRL Report 8614, Naval Research
Laboratory, Washington DC, September 1982.
[Measurement] Web page on "Measurement Studies of End-to-End
Congestion Control in the Internet", URL
"http://www.icir.org/floyd/ccmeasure.html". The
section on "Network Measurements at Specific Sites"
includes measurement data about the distribution of
packet sizes on various links in the Internet.
[MTK03] A. P. Markopoulou, F. A. Tobagi, and M. J. Karam,
"Assessing the Quality of Voice Communications Over
Internet Backbones", IEEE/ACM Transactions on
Networking, V. 11 N. 5, October 2003.
[NSIS] Next Steps in Signaling (nsis), IETF Working Group, URL
"http://www.ietf.cnri.reston.va.us/html.charters/nsis-
charter.html".
[PCC] Joerg Widmer, Martin Mauve, and Jan Peter Damm.
Probabilistic Congestion Control for Non-Adaptable
Flows. Technical Report 3/2001, Department of
Mathematics and Computer Science, University of
Mannheim. URL "http://www.informatik.uni-
mannheim.de/informatik/pi4/projects/
CongCtrl/pcc/index.html".
[PFTK98] J. Padhye, V. Firoiu, D. Towsley, J. Kurose, Modeling
TCP Throughput: A Simple Model and its Empirical
Validation, Tech Report TF 98-008, U. Mass, February
1998.
[RFC896] Nagle, J., "Congestion Control in IP/TCP", RFC 896,
January 1984.
[RFC1890] Schulzrinne, H., "RTP Profile for Audio and Video
Conferences with Minimal Control", RFC 1890, January
1996.
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RFC 3714 IAB Concerns Regarding Congestion Control March 2004
[RFC2474] Nichols, K., Blake, S., Baker, F. and D. Black,
"Definition of the Differentiated Services Field (DS
Field) in the IPv4 and IPv6 Headers", RFC 2474,
December 1998.
[RFC2581] Allman, M., Paxson, V. and W. Stevens, "TCP Congestion
Control", RFC 2581, April 1999.
[RFC2597] Heinanen, J., Baker, F., Weiss, W. and J. Wroclawski,
"Assured Forwarding PHB Group, RFC 2597, June 1999.
[RFC2914] Floyd, S., "Congestion Control Principles", BCP 41, RFC
2914, September 2000.
[RFC2990] Huston, G., "Next Steps for the IP QoS Architecture",
RFC 2990, November 2000.
[RFC3042] Allman, M., Balakrishnan, H. and S., Floyd, "Enhancing
TCP's Loss Recovery Using Limited Transmit", RFC 3042,
January 2001.
[RFC3168] Ramakrishnan, K., Floyd, S. and D. Black, "The Addition
of Explicit Congestion Notification (ECN) to IP", RFC
3168, September 2001.
[RFC3246] Davie, B., Charny, A., Bennet, J.C.R., Benson, K., Le
Boudec, J.Y., Courtney, W., Davari, S., Firoiu, V. and
D. Stiliadis, "An Expedited Forwarding PHB (Per-Hop
Behavior)", RFC 3246, March 2002.
[RFC3448] Handley, M., Floyd, S., Pahdye, J. and J. Widmer, "TCP
Friendly Rate Control (TFRC): Protocol Specification",
RFC 3448, January 2003.
[RSVP] Resource Reservation Setup Protocol (rsvp), Concluded
Working Group, URL
"http://www.ietf.cnri.reston.va.us/html.charters/
OLD/rsvp-charter.html".
[RTTWeb] Web Page on Round-Trip Times in the Internet, URL
"http://www.icir.org/floyd/rtt-questions.html"
[S03] H. Schulzrinne, private communication, 2003.
[RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio
and Video Conferences with Minimal Control", RFC 3551,
July 2003.
Floyd & Kempf Informational [Page 25]
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[Vonage] Vonage, URL "www.vonage.com".
[W98] J. Woodward, Speech Coding, Communications Research
Group, University of Southampton, 1998. URL
"http://www-mobile.ecs.soton.ac.uk/speech_codecs/",
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10. Appendix - Sending Rates with Packet Drops
The standard way to estimate TCP's average sending rate S in packets
per round-trip as a function of the packet drop rate would be to use
the TCP response function estimated in [PFTK98]:
S = 1/(sqrt(2p/3) + K min(1,3 sqrt(3p/8)) p (1 + 32 p^2)) (1)
for acks sent for every data packet, and the RTO set to K*RTT.
The results from Equation (1) are given in the second column in
Tables 1 and 2 below. However, Equation (1) overestimates TCP's
sending rate in the regime with heavy packet drop rates (e.g., of 30%
or more). The analysis behind Equation (1) assumes that once a
single packet is successfully transmitted, TCP's retransmit timer is
no longer backed-off. This might be appropriate for an environment
with ECN, or for a TCP connection using fine-grained timestamps, but
this is not necessarily the case for a non-ECN-capable TCP connection
without timestamps. As specified in [RFC2988], if TCP's retransmit
timer is backed-off, this back-off should only be removed when TCP
successfully transmits a new packet (as opposed to a retransmitted
packet), in the absence of timestamps.
When the packet drop rate is 50% or higher, for example, many of the
successful packet transmissions can be of retransmitted packets, and
the retransmit timer can remain backed-off for significant periods of
time, in the absence of timestamps. In this case, TCP's throughput
is determined largely by the maximum backoff of the retransmit timer.
For example, in the NS simulator the maximum backoff of the
retransmit timer is 64 times the un-backed-off value. RFC 2988
specifies that "a maximum value MAY be placed on RTO provided it is
at least 60 seconds." [Although TCP implementations vary, many TCP
implementations have a maximum of 45 seconds for the backed-off RTO
after dropped SYN packets.]
Another limitation of Equation (1) is that it models Reno TCP, and
therefore underestimates the sending rate of a modern TCP connection
that used SACK and Limited Transmit.
The table below shows estimates of the average sending rate S in
packets per RTT, for TCP connections with the RTO set to 2 RTT for
Equation (1).
These estimates are compared with simulations in the third, fourth,
and fifth columns, with ECN, packet drops for TCP with fine-grained
timestamps, and packet drops for TCP without timestamps respectively.
(The simulation scripts are available from
http://www.icir.org/floyd/VoIP/sims.) Each simulation computes the
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average sending rate over the second half of a 10,000-second
simulation, and for each packet drop rate, the average is given over
50 simulations. For the simulations with very high packet drop
rates, it is sometimes the case that the SYN packet is repeatedly
dropped, and the TCP sender never successfully transmits a packet.
In this case, the TCP sender also never gets a measurement of the
round-trip time.
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The sixth column of Table 1 shows the average sending rate S in
packets per RTT for an experiment using a 4.8-RELEASE FreeBSD
machine. For the low packet drop rates of 0.1 and 0.2, the sending
rate in the simulations is higher than the sending rate in the
experiments; this is probably because the TCP implementation in the
simulations uses Limited Transmit [RFC3042]. With Limited Transmit,
the TCP sender can sometimes avoid a retransmit timeout when a packet
is dropped and the congestion window is small. With high packet drop
rates of 0.65 and 0.7, the sending rate in the simulations is
somewhat lower than the sending rate in the experiments. For these
high packet drop rates, the TCP connections in the experiments would
often abort prematurely, after a sufficient number of successive
packet drops.
We note that if the ECN marking rate exceeds a locally-configured
threshold, then a router is advised to switch from marking to
dropping. As a result, we do not expect to see high steady-state
marking rates in the Internet, even if ECN is in fact deployed.
Drop
Rate p Eq(1) Sims:ECN Sims:TimeStamp Sims:Drops Experiments
------ ----- -------- -------------- ---------- -----------
0.1 2.42 2.92 2.38 2.32 0.72
0.2 .89 1.82 1.26 0.82 0.29
0.25 .55 1.52 .94 .44 0.22
0.35 .23 .99 .51 .11 0.10
0.4 .16 .75 .36 .054 0.068
0.45 .11 .55 .24 .029 0.050
0.5 .10 .37 .16 .018 0.036
0.55 .060 .25 .10 .011 0.024
0.6 .045 .15 .057 .0068 0.006
0.65 .051 . .033 .0034 0.008
0.7 .041 .06 .018 .0022 0.007
0.75 .034 .04 .0099 .0011
0p.8 .028 .027 .0052 .00072
0.85 .023 .015 .0021 .00034
0.9 .020 .011 .0011 .00010
0.95 .017 .0079 .00021 .000037
Table 1: Sending Rate S as a Function of the Packet Drop Rate p,
for RTO set to 2 RTT, and S in packets per RTT.
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The table below shows the average sending rate S, for TCP connections
with the RTO set to 10 RTT.
Drop
Rate p Eq(1) Sims:ECN Sims:TimeStamp Sims:Drops
------ ----- -------- -------------- ----------
0.1 0.97 2.92 1.67 1.64
0.2 0.23 1.82 .56 .31
0.25 0.13 .88 .36 .13
0.3 0.08 .61 .23 .059
0.35 0.056 .41 .15 .029
0.4 0.040 .28 .094 .014
0.45 0.029 .18 .061 .0080
0.5 0.021 .11 .038 .0053
0.55 0.016 .077 .022 .0030
0.6 0.013 .045 .013 .0018
0.65 0.010 . .0082 .0013
0.7 0.0085 .018 .0042
0.75 0.0069 .012 .0025 .00071
0.8 0.0057 .0082 .0014 .00030
0.85 0.0046 .0047 .00057 .00014
0.9 0.0041 .0034 .00026 .000025
0.95 0.0035 .0024 .000074 .000013
Table 2: Sending Rate as a Function of the Packet Drop Rate,
for RTO set to 10 RTT, and S in packets per RTT.
11. Security Considerations
This document does not itself create any new security issues for the
Internet community.
12. IANA Considerations
There are no IANA considerations regarding this document.
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13. Authors' Addresses
Internet Architecture Board
EMail: iab@iab.org
Internet Architecture Board Members
at the time this document was published were:
Bernard Aboba
Harald Alvestrand (IETF chair)
Rob Austein
Leslie Daigle (IAB chair)
Patrik Faltstrom
Sally Floyd
Jun-ichiro Itojun Hagino
Mark Handley
Geoff Huston (IAB Executive Director)
Charlie Kaufman
James Kempf
Eric Rescorla
Mike St. Johns
This document was created in January 2004.
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14. Full Copyright Statement
Copyright (C) The Internet Society (2004). This document is subject
to the rights, licenses and restrictions contained in BCP 78 and
except as set forth therein, the authors retain all their rights.
This document and the information contained herein are provided on an
"AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE
REPRESENTS OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE
INTERNET ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF
THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
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The IETF takes no position regarding the validity or scope of any
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Copies of IPR disclosures made to the IETF Secretariat and any
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to implement this standard. Please address the information to the
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Acknowledgement
Funding for the RFC Editor function is currently provided by the
Internet Society.
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