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Versions: 00 01 02 RFC 3714

Internet Engineering Task Force                      Sally Floyd, Editor
INTERNET DRAFT                                       James Kempf, Editor
draft-iab-congestion-00.txt                                    June 2003

IAB Concerns Regarding Congestion Control for Voice Traffic in the Internet

                          Status of this Memo

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet- Drafts as reference
   material or to cite them other than as "work in progress."

   The list of current Internet-Drafts can be accessed at

   The list of Internet-Draft Shadow Directories can be accessed at


   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 on underprovisioned links in the
   Internet. The concerns are particularly relevant in light of the
   absence of a widespread QoS deployment in the Internet, and the
   likelihood that this situation will not change much in the near term.

   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].

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1.  Introduction

   There is a widely held assumption among many in the area of Internet
   telephony that QoS of some form will ultimately become broadly
   deployed and available for regulating real-time traffic. Many
   circuit-switched providers studying the possibility of deploying IP
   telephony speak of the need for end-to-end QoS. The assumption seems
   to be that Internet telephony will serve as the application to drive
   the deployment of QoS; at the moment, very few public access ISPs
   have deployed QoS, although some corporate intranets have done so.

   While the assumption of IP telephony driving the deployment of end-
   to-end QoS might turn out to be correct, past history of Internet
   technology deployment indicates this is not very likely.  The
   deployment of technologies requiring change to the Internet
   infrastructure is subject to a wide range of commercial as 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.  For example, class-based queueing at a router could
   give limited priority to VoIP traffic on a queue, based on port
   number ranges for UDP traffic.  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.  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 tend to be
   overprovisioned in the core but underprovisioned on the last hop

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   links.  If an IP telephony call runs completely over the Internet,
   the connection is likely to traverse underprovisioned 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 less likely to be overprovisioned, making congestion
   limitation an especially important topic for developing countries.

   Given the possible deployment of IP telephony over underprovisioned
   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 telephony
   traffic on best-effort networks.  We consider best-effort telephony
   connections that have a minimum sending rate and that compete
   directly with other best-effort traffic on underprovisioned links,
   and address the specific question of whether such traffic should be
   required to terminate in the face of a persistent, high packet drop
   rate, when reducing the sending rate is not a viable alternative.

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 overprovisioned 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%
   [JAK].  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 voice packets
   could be delayed up to 90 ms, which might cause an increase in the

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   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.  Nevertheless, VoIP usage
   over underprovisioned 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 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, universal
   overprovisioning for Internet links is likely to be impractical or

   However, this does not mean that universal overprovisioning is
   acceptable as the only solution to congestion even in environments
   where overprovisioning is the typical state of affairs.  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 (in this case, in the form of
   congested links wasting scarce bandwidth carrying packets that will
   only be dropped downstream).  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 the generally-overprovisioned network.

   Best-effort traffic in the Internet does not include mechanisms for
   call acceptance or rejection.  Instead, the 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.  One part of this approach is to consider the appropriate
   response when a codec is at its minimum data rate, and the packet

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   drop rate experienced by the flow in the network 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 overprovisioning 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
   potential problem in VoIP networks when both ends of the VoIP call
   are on an underprovisioned broadband connection such as DSL, or when
   the call traverses an underprovisioned 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

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   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

   It seems clear from this simple example that the quality of every
   VoIP traffic flow accessible over an underprovisioned link 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].  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.

   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 are 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 congested link.

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   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 with more
   aggressive RTO settings sending more than a TCP implementation with
   less aggressive RTO settings.

   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].
   (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 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 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

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   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.

   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,
   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 judgement
   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.  (E.g.
   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 protocol specification, does not discuss
   congestion control [RFC1890].  The revised draft of "RTP Profile for

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   Audio and Video Conferences with Minimal Control" [SC03] currently
   under development in the IETF discusses congestion control in Section
   2.  [SC03] currently 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."

     "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 [SC03] says that receivers "SHOULD" monitor packet loss.
   [SC03] 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, since "danger to the Internet" is
   one of two criteria given in RFC 2119 for the use of "MUST"

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

     "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

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     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.

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, 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 an underprovisioned
   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.

   As one IETF effort, 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

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   speech encoding modes with 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 2367 recommends that AMR or
   AMR-WB applications using the RTP payload format specified in RFC
   3267 use congestion control, though no specific mechanism is
   considered.  RFC 3267 gives "Equation-Based Congestion Control for
   Unicast Applications" as an example of a congestion control mechanism
   suitable for real-time flows [FHPW00].

   Unfortunately, the AMR and AMR-WB codecs are covered by IPR and
   require licensing, which may serve as a deterrent for low margin
   manufacturers and service providers in the developing world. 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 hence are ill-suited for
   Internet use, even with the addition of FEC on top.

5.0.  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,

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   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.

   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
   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 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.  This would be a less appropriate response for
   a network path where the limitation was 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 not correct, then
   the sending rate in pps could contribute to congestion even when the
   sending rate in Bps was quite moderate.  While the ideal would be to
   have a transport protocol that was able to detect whether the
   bottleneck links along the path were limited in Bps or in pps, and to
   respond appropriately when the limitation was in pps, such an ideal
   is hard to achieve, and 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 reserve 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

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   considering the minimum sending rate of telephony traffic.

   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).

   Restated, the question is, 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 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 that 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.

   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.

   One argument in response to this proposal 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

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   might also just keep on talking, with the occasional noise getting
   though, for minutes or longer waiting for a short period of clarity.
   Another argument against this proposal is that nobody really benefits
   from VoIP connections being terminated 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 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 this 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 would be somewhat different.
   First, 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.

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

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   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 and fourth columns give the average sending
   rate from simulations with random packet drops.  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 give a packet drop rate of
   between 35% and 40% 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, 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 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
   drop rate.  The simulations in the last 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%.
   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 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.  For this section, consider a codec

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   with a minimum sending rate is 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 with
   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 last
   column of Table 1, for an RTO of 2 RTT, this corresponds to a packet
   drop rate between 20 and 25%.  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 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 last column of Table 2, which shows that a sending
   rate 0.066 ppr corresponds to a packet drop rate of roughly 30%.
   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 when the persistent packet drop rate exceeded 30%.

5.3. 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 was greater than 25% or N back-to-back RTCP
   reports were missing, the application could assume that the network
   is too congested, and terminate.

6. Constraints on VoIP Systems

   Ultimately, attempting to run VoIP on underprovisioned, 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

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   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 when the steady-state packet
   drop rate significantly exceeds the specified drop rate.

   (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

   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.

8.  Acknowledgements

   We thank Brian Adamson, Ran Atkinson, Alan Duric, Mark Handley, Orion
   Hodson, Geoff Huston, Eddie Kohler, Jon Peterson, 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

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   in Section 6.0 resulted from a session at the whiteboard with Mark

9.  References

   Normative References

   [RFC2988] V. Paxson and M. Allman, Computing TCP's Retransmission
   Timer, RFC 2988, November 2000.

   [RFC3267] J. Sjoberg, M. Westerlund, A. Lakaniemi, 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

   Informative References

   [A02] Ran Atkinson, An ISP Reality Check, Presentation to ieprep,
   55th IETF Meeting, November 2002.  URL

   [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.

   [DCCP03] Eddie Kohler, Mark Handley, Sally Floyd, and Jitendra
   Padhye, Datagram Congestion Control Protocol (DCCP), internet-draft
   draft-ietf-dccp-spec-01.txt, work in progress, March 2003.  URL

   [E2E] The end2end-interest mailing list, URL

   [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

   [FWD] Free World Dialup, URL "www.pulver.com/fwd/".

   [IEPREP02] Internet Emergency Preparedness (ieprep), Minutes, 55th
   IETF Meeting, November 2002.  URL

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   [ILBRC] S.V. Andersen, et. al., Internet Low Bit Rate Codec, draft-
   ietf-avt-ilbc-codec-01.txt, internet draft, work in progress, March

   [G.114] Recommendation G.114 - One-way Transmission Time, ITU, May
   2003.  URL "http://www.itu.int/itudoc/itu-

   [IVOX] The Interactive VOice eXchange, URL

   [Jacobson88] V. Jacobson, Congestion Avoidance and Control, ACM
   SIGCOMM '88, August 1988.

   [JAK] Maximum 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.

   [JS00]  Wenyu Jiang and Henning Schulzrinne, Modeling of Packet Loss
   and Delay and Their Effect on Real-Time Multimedia Service Quality,
   NOSSDAV, 2000.  URL

   [JS02] Wenyu Jiang and Henning Schulzrinne, Comparison and
   Optimization of Packet Loss Repair Methods on VoIP Perceived Quality
   under Bursty Loss, NOSSDAV, 2002.  URL

   [JS03] Wenyu Jiang, Kazummi Koguchi, and Henning Schulzrinne, QoS
   Evaluation of VoIP End-points, ICC 2003.  URL

   [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 Communciations", 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.

   [PCC] Joerg Widmer, Martin Mauve, and Jan Peter Damm. Probabilistic
   Congestion Control for Non-Adaptable Flows. Technical Report 3/2001,

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   Department of Mathematics and Computer Science, University of
   Mannheim.  URL "http://www.informatik.uni-

   [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

   [RFC1890] H. Schulzrinne, RTP Profile for Audio and Video Conferences
   with Minimal Control, RFC 1890, January 1996.

   [RFC2119] S. Bradner, Key words for use in RFCs to Indicate
   Requirement Levels, RFC 2119, 1997.

   [RFC2581] Allman, M., Paxson, V. and W. Stevens, "TCP Congestion
   Control", RFC 2581, April 1999.

   [RFC2914] S. Floyd, Congestion Control Principles, RFC 2914,
   September 2000.

   [RFC2990] G. Huston, Next Steps for the IP QoS Architecture, November

   [RFC3042] Allman, M., Balakrishnan, H., and Floyd, S., Enhancing
   TCP's Loss Recovery Using Limited Transmit, RFC 3042, January 2001.

   [RFC3168] K. Ramakrishnan, S. Floyd, and D. Black, The Addition of
   Explicit Congestion Notification (ECN) to IP, RFC 3168, September

   [RFC3448] Handley, M., Floyd, S., Pahdye, J., and Widmer, J., TCP
   Friendly Rate Control (TFRC): Protocol Specification, RFC 3448,
   Proposed Standard, January 2003.

   [RTTWeb] Web Page on Round-Trip Times in the Internet, URL

   [S03] H. Schulzrinne, private communication, 2003.

   [SC03] Schulzrinne and Casner, RTP Profile for Audio and Video
   Conferences with Minimal Control, draft-ietf-avt-profile-new-13.txt,
   internet-draft, work in progress, March 2003.  URL

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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-

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, but this is not necessarily the case with packet drops.  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).

   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 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 and fourth

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   columns, with ECN and packet drops respectively.  (The simulation
   scripts are available from http://www.icir.org/floyd/VoIP/sims.)
   Each simulation computes the 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.

   Drop Rate p     Equation (1)  Sims (ECN)    Sims (Drops)
   -----------     -----------   ----------    -------------
         0.1       2.42           2.92          2.32
         0.2        .89           1.82          0.82
         0.25       .55           1.52           .44
         0.3        .35           1.24           .22
         0.35       .23           .99            .11
         0.4        .16           .75            .054
         0.45       .11           .55            .029
         0.5        .10           .37            .018
         0.55       .060          .25            .011
         0.6        .045          .15            .0068
         0.65       .051          .              .0034
         0.7        .041          .06            .0022
         0.75       .034          .04            .0011
         0.8        .028          .027           .00072
         0.85       .023          .015           .00034
         0.9        .020          .011           .00010
         0.95       .017          .0079          .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.

   The table below shows the average sending rate S, for TCP connections
   with the RTO set to 10 RTT.

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   Drop Rate p     Equation (1)  Sims (ECN)    Sims (Drops)
   -----------     -----------   ----------    -------------
    0.1              0.97        2.92           1.64
    0.2              0.23        1.82            .31
    0.25             0.13         .88            .13
    0.3              0.08         .61            .059
    0.35             0.056        .41            .029
    0.4              0.040        .28            .014
    0.45             0.029        .18            .0080
    0.5              0.021        .11            .0053
    0.55             0.016        .077           .0030
    0.6              0.013        .045           .0018
    0.65             0.010        .              .0013
    0.7              0.0085       .018
    0.75             0.0069       .012           .00071
    0.8              0.0057       .0082          .00030
    0.85             0.0046       .0047          .00014
    0.9              0.0041       .0034          .000025
    0.95             0.0035       .0024          .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.

10.1.  Sending Rates with ECN

   One of the reasons to look at sending rates with ECN is that this
   avoids retransmitting packets, and therefore a successfully-
   transmitted packet is more likely to result in a backed-off
   retransmit timer.  ECN allows routers to explicitly notify end-nodes
   of congestion by ECN-marking instead of dropping packets [RFC3168].
   The third column of Tables 1 and 2 shows the results of simulations
   that use ECN instead of packet drops.

   Packet Mark Rate p     Sending Rate S (in ppr)
   ------------------     -----------------------
         0.16                  2
         0.33                  1.5

   Table 6: Sending Rate as a Function of the Packet Mark Rate

   For example, with an ECN marking rate p of 1/3, assuming a fixed,
   periodic pattern of packet marks, a TCP connection would send three
   packets every two round-trip times, as follows:

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   Round-Trip Time     Packets Sent     Packets Marked
   ---------------     ------------     --------------
           1                1             0
           2                2             1 (second packet)
           3                1             0
           4                2             1 (second packet)

   Table 7: Sending rate of 1.5 pkts/rtt with marking rate of 0.33.

   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.

   Round-Trip Time     Packets Sent     Packets Marked
   ---------------     ------------     --------------
           1                1             0
           2                2             0
           3                3             1 (second packet)
           4                1             0
           5                2             0
           6                3             1 (second packet)

   Table 8: Sending rate of 2 pkts/rtt with marking rate of 0.16.

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.


   Internet Architecture Board
   EMail:  iab@iab.org

   IAB Membership at time this document was undertaken:

   Harald Alvestrand (IETF chair)
   Ran Atkinson

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   Rob Austein
   Fred Baker
   Leslie Daigle (IAB chair)
   Sally Floyd
   Ted Hardie
   Geoff Huston (IAB Executive Director)
   Charlie Kaufman
   James Kempf
   Vern Paxson (IRTF chair)
   Eric Rescorla
   Mike St. Johns

   This draft was created in June 2003.

IAB                           Informational                    [Page 25]

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