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INTERNET-DRAFT Strategies for Streaming Media Using TFRC  October 2005

   Strategies for Streaming Media Using TFRC
   Internet Draft                                             T. Phelan
   Document: draft-ietf-dccp-tfrc-media-01.txt           Sonus Networks
   Expires: May 2006                                       October 2005


                Strategies for Streaming Media Applications
                      Using TCP-Friendly Rate Control


Status of this Memo

   By submitting this Internet-Draft, each author represents that any
   applicable patent or other IPR claims of which he or she is aware
   have been or will be disclosed, and any of which he or she becomes
   aware will be disclosed, in accordance with Section 6 of BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
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   This Internet-Draft will expire on May 24, 2006.


Copyright Notice

   Copyright (C) The Internet Society (2005).


Abstract

   This document discusses strategies for using streaming media
   applications with unreliable congestion-controlled transport
   protocols such as the Datagram Congestion Control Protocol (DCCP) or
   the RTP Profile for TCP Friendly Rate Control.  Of particular
   interest is how media streams, which have their own transmit rate
   requirements, can be adapted to the varying and sometimes conflicting
   transmit rate requirements of congestion control protocols such as
   TCP-Friendly Rate Control (TFRC).

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Table of Contents

   1. Introduction...................................................3
   2. TFRC Basics....................................................3
   3. Streaming Media Applications...................................5
      3.1 Stream Switching...........................................6
      3.2 Media Buffers..............................................7
      3.3 Variable Rate Media Streams................................7
   4. Strategies for Streaming Media Applications....................8
      4.1 First Strategy -- One-way Pre-recorded Media...............8
         4.1.1 Strategy 1............................................8
         4.1.2 Issues With Strategy 1................................9
      4.2 Second Try -- One-way Live Media..........................10
         4.2.1 Strategy 2...........................................10
         4.2.2 Issues with Strategy 2...............................12
      4.3 One More Time -- Two-way Interactive Media................12
         4.3.1 Strategy 3...........................................13
         4.3.2 Issues with Strategy 3...............................14
   5. Security Considerations.......................................14
   6. IANA Considerations...........................................14
   7. Thanks........................................................14
   8. Informative References........................................15
   9. Author's Address..............................................16


























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1. Introduction

   The canonical streaming media application emits fixed-sized (often
   small) packets at a regular interval.  It relies on the network to
   deliver the packets to the receiver in roughly the same regular
   interval.  Often, the transmitter operates in a fire-and-forget mode,
   receiving no indications of packet delivery and never changing its
   mode of operation.  This often holds true even if the packets are
   encapsulated in the Real-time Transport Protocol (RTP) and the RTP
   Control Protocol (RTCP) [RFC 3550] is used to get receiver
   information; it's rare that the RTCP reports trigger changes in the
   transmitted stream.

   The IAB has expressed concerns over the stability of the Internet if
   these applications become too popular with regard to TCP-based
   applications [RFC 3714].  They suggest that media applications should
   monitor their packet loss rate, and abort if they exceed certain
   thresholds.  Unfortunately, up until this threshold is reached, the
   network, the media applications, and the other applications are all
   experiencing considerable duress.

   TCP-Friendly Rate Control (TFRC, [RFC 3448]) offers an alternative to
   the [RFC 3714] method.  The key differentiator of TFRC, relative to
   the Additive Increase Multiplicative Decrease (AIMD) method used in
   TCP and SCTP, is its smooth response to packet loss.  TFRC has been
   implemented as one of the "pluggable" congestion control algorithms
   for the Datagram Congestion Control Protocol (DCCP, [DCCP] and
   [CCID3]) and as a profile for RTP [RTP-TFRC].

   This document explores issues to consider and strategies to employ
   when adapting or creating streaming media applications to use
   transport protocols using TFRC for congestion control.  The approach
   here is one of successive refinement.  Strategies are described and
   their strengths and weaknesses are explored.  New strategies are then
   presented that improve on the previous ones and the process iterates.
   The intent is to illuminate the issues, rather than to jump to
   solutions, in order to provide guidance to application designers.

2. TFRC Basics

   AIMD congestion control algorithms, such DCCP's CCID2 [CCID2] or
   TCP's SACK-based control [RFC 3517], use a congestion window (the
   maximum number of packets or segments in flight) to limit the
   transmitter.  The congestion window is increased by one for each
   acknowledged packet, or for each window of acknowledged packets,
   depending on the phase of operation.  If any packet is dropped (or
   ECN-marked [ECN]; for simplicity in the rest of the document assume

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   that "dropped" equals "dropped or ECN-marked"), the congestion window
   is halved.  This produces a characteristic saw-tooth wave variation
   in throughput, where the throughput increases linearly up to the
   network capacity and then drops abruptly (roughly shown in Figure 1).

                 |
                 |      /|    /|    /|    /|    /
                 |     / |   / |   / |   / |   /
       Throughput|    /  |  /  |  /  |  /  |  /
                 |   /   | /   | /   | /   | /
                 |  /    |/    |/    |/    |/
                 |
                  ----------------------------------
                                 Time

   Figure 1: Characteristic throughput for AIMD congestion control.


   On the other hand, with TCP-Friendly Rate Control (TFRC), the
   immediate response to packet drops is less dramatic.  To compensate
   for this TFRC is less aggressive in probing for new capacity after a
   loss.  TFRC uses a version of the TCP throughput equation to compute
   a maximum transmit rate, taking a weighted history of loss events as
   input (more weight is given to more recent losses).  The
   characteristic throughput graph for a TFRC connection looks like a
   flattened sine wave (extremely roughly shown in Figure 2).

                 |
                 |    --        --        --
                 |   /  \      /  \      /  \
       Throughput|  /    \    /    \    /    \
                 | /      \  /      \  /      \
                 |-        --        --        -
                 |
                  ----------------------------------
                                 Time

   Figure 2: Characteristic throughput for TFRC congestion control.


   In addition to this high-level behavior, there are several details of
   TFRC operation that, at first blush at least, seem at odds with
   common media stream transmission practices.  Some particular
   considerations are:

    o  Slow Start -- A connection starts out with a transmission rate of
       up to four packets per round trip time (RTT).  After the first
       RTT, the rate is doubled each RTT until a packet is lost.  At
       this point the transmission rate is halved and we enter the
       equation-based phase of operation.  It's likely that in many
       situations the initial transmit rate is slower than the lowest

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       bit rate encoding of the media.  This will require the
       application to deal with a ramp up period.

    o  Capacity Probing and Lost Packets -- If the application transmits
       for some time at the maximum rate that TFRC will allow without
       packet loss, TFRC will continuously raise the allowed rate until
       a packet is lost.  This means that, in many circumstances, if an
       application wants to transmit at the maximum possible rate,
       packet loss will not be an exceptional event, but will happen
       routinely in the course of probing for more capacity.

    o  Idleness Penalty -- TFRC follows a "use it or lose it" policy.
       If the transmitter goes idle for a few RTTs, as it would if, for
       instance, silence suppression were being used, the transmit rate
       returns to two packets per RTT, and then doubles every RTT until
       the previous rate is achieved.  This can make restarting after a
       silence suppression interval problematic.

    o  Contentment Penalty -- TFRC likes to satisfy greed.  If you are
       transmitting at the maximum allowed rate, TFRC will try to raise
       that rate.  However, if your application is transmitting below
       the maximum allowed rate, the maximum allowed rate will not be
       increased higher than twice the current transmit rate, no matter
       how long it has been since the last increase.  This can create
       problems when attempting to shift to a higher rate encoding, or
       with video codecs that vary the transmission rate with the amount
       of movement in the image.

    o  Packet Rate, not Bit Rate -- TFRC controls the rate that packets
       may enter the network, not bytes.  To respond to a lowered
       transmit rate you must reduce the packet transmission rate.
       Making the packets smaller while still keeping the same packet
       rate will not be effective.

    o  Smooth Variance of Transmit Rate -- The strength and purpose of
       TFRC (over AIMD Congestion Control) is that it smoothly decreases
       the transmission rate in response to recent packet loss events,
       and smoothly increases the rate in the absence of loss events.
       This smoothness is somewhat at odds with most media stream
       encodings, where the transition from one rate to another is often
       a step function.

3. Streaming Media Applications

   While all streaming media applications have some characteristics in
   common (e.g. data must arrive at the receiver at some minimum rate
   for reasonable operation), other characteristics (e.g. tolerance of
   end-to-end delay) vary considerably from application to application.
   For the purposes of this document, it's useful to divide streaming
   media applications into three subtypes:


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    o  One-way pre-recorded media
    o  One-way live media
    o  Two-way interactive media

   The relevant difference, as far as this discussion goes, between
   recorded and live media is that recorded media can be transmitted as
   fast as the network allows (assuming adequate buffering at the
   receiver) -- it could be viewed as a special file transfer operation.
   Live media can't be transmitted faster than the rate at which it's
   encoded.

   The difference between one-way and two-way media is the sensitivity
   to delay.  For one-way applications, delays from encoding at the
   sender to playout at the receiver of several or even tens of seconds
   are acceptable.  For two-way applications delays from encoding to
   playout of as little as 150 to 200 ms are often problematic [XTIME].

   While delay sensitivity is most problematic when dealing with two-way
   conversational applications such as telephony, it is also apparent in
   nominally one-way applications when certain user interactions are
   allowed, such as program switching ("channel surfing") or fast
   forward/skip.  Arguably, these user interactions have turned the one-
   way application into a two-way application -- there just isn't the
   same sort of data flowing in both directions.

3.1 Stream Switching

   The discussion here assumes that media transmitters are able to
   provide their data in a number of encodings with various bit rate
   requirements and are able to dynamically change between these
   encodings with low overhead.  It also assumes that switching back and
   forth between coding rates does not cause excessive user annoyance.

   Given the current state of codec art, these are big assumptions.  As
   a practical matter, continuous shifts between higher and lower
   quality levels can greatly annoy users, much more so than one shift
   to a lower quality level and then staying there.  The algorithms
   given below indicate methods for returning to higher bandwidth
   encodings, but, because of the bad user perception of shifting
   quality, many media applications may choose to never invoke these
   methods.

   Also, the algorithms and results described here hold even if the
   media sources can only supply media at one rate.  Obviously the
   statements about switching encoding rates don't apply, and an
   application with only one encoding rate behaves as if it is
   simultaneously at its minimum and maximum rate.

   For convenience in the discussion below, assume that all media
   streams have two encodings, a high bit rate and a low bit rate,
   unless otherwise indicated.

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3.2 Media Buffers

   The strategies below make use of the concept of a media buffer.  A
   media buffer is a first-in-first-out queue of media data.  The buffer
   is filled by some source of data (the encoder or the network) and
   drained by some sink (the network or the playout device).  It
   provides rate and jitter compensation between the source and the
   sink.

   Media buffer contents are measured in seconds of media play time, not
   bytes or packets.  Media buffers are completely application-level
   constructs and are separate from transport-layer transmit and receive
   queues.

3.3 Variable Rate Media Streams

   The canonical media codec encodes its media as a constant rate bit
   stream.  As the technology has progressed from its time-division
   multiplexing roots, this constant rate stream has become not so
   constant.  Voice codecs often employ silence suppression (also called
   Voice Activity Detection, or VAD), where the stream (in at least one
   direction) goes totally idle for sometimes several seconds while one
   side listens to what the other side has to say.  When the one side
   wants to start talking again, the codec resumes sending immediately
   at its "constant" rate.

   Video codecs similarly employ what could be called "stillness"
   suppression, but is instead called motion compensation.  Often these
   codecs effectively transmit the changes from one video frame to
   another.  When there is little change from frame to frame (as when
   the background is constant and a talking head is just moving its
   lips) the amount of information to send is small.  When there is a
   major motion, or change of scene, much more information must be sent.
   For some codecs, the variation from the minimum rate to the maximum
   rate can be a factor of ten [MPEG4].  Unlike voice codecs, though,
   video codecs typically never go completely idle.

   These abrupt jumps in transmission rate are problematic for any
   congestion control algorithm.  A basic tenet of all existing
   algorithms assumes that increases in transmission rate must be
   gradual and smooth to avoid damaging other connections in the
   network.  In TFRC, the transmission rate in a Round Trip Time (RTT)
   can never be more than twice the rate actually delivered to the
   receiver in the previous RTT.

   TFRC uses a maximum rate of two packets per RTT after an idle period.
   This rate might support immediate restart of voice data after a
   silence period, at least when the RTT is in the suitable range for
   two-way media.  More problematic are the factor of ten variations in
   some video codecs.  In some circumstances, TFRC allows an application


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   to double its transmit rate over one RTT (assuming no recent packet
   loss events), but an immediate ten times increase is not possible.

4. Strategies for Streaming Media Applications

   This section covers a number of strategies that can be used by
   streaming media applications.  Each strategy is applicable to one or
   more types of streaming media.

4.1 First Strategy -- One-way Pre-recorded Media

   The first strategy is suitable for use with pre-recorded media, and
   takes advantage of the fact that the data for pre-recorded media can
   be transferred to the receiver as fast as the network will allow it,
   assuming that the receiver has sufficient buffer space.

4.1.1 Strategy 1

   Assume a recorded program resides on a media server, and the server
   and its clients are capable of stream switching between two encoding
   rates, as described in section 3.1.

   The client (receiver) implements a media buffer as a playout buffer.
   This buffer is potentially big enough to hold the entire recording.
   The playout buffer has three thresholds: a low threshold, a playback
   start threshold, and a high threshold, in order of increasing size.
   These values will typically be in the several to tens of seconds
   range.  The buffer is filled by data arriving from the network and
   drained at the decoding rate necessary to display the data to the
   user.  Figure 3 shows this schematically.

                             high threshold
                                 |  playback start threshold
                                 |    |  low threshold
   +-------+                     |    |    |
   | Media |  transmit at    +---v----v----v--+
   | File  |---------------->| Playout buffer |-------> display
   |       |  TFRC max rate  +----------------+ drain at
   +-------+                 fill at network    decode rate
                             arrival rate

   Figure 3: One-way pre-recorded media.


   During the connection the server needs to be able to determine the
   depth of data in the playout buffer.  This could be provided by
   direct feedback from the client to the server, or the server could
   estimate its depth (e.g. the server knows how much data has been
   sent, and how much time has passed).



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   To start the connection, the server begins transmitting data in the
   high bit rate encoding as fast as TFRC allows.  Since TFRC is in slow
   start, this is probably too slow initially, but eventually the rate
   should increase to fast enough (assuming sufficient capacity in the
   network path).  As the client receives data from the network it adds
   it to the playout buffer.  Once the buffer depth reaches the playback
   start threshold, the receiver begins draining the buffer and playing
   the contents to the user.

   If the network has sufficient capacity, TFRC will eventually raise
   the transmit rate to greater than necessary to keep up with or exceed
   the decoding rate, the playout buffer will back up as necessary, and
   the entire program will eventually be transferred.

   If the TFRC transmit rate never gets fast enough, or loss events make
   TFRC drop the rate, the receiver will drain the playout buffer faster
   than it is filled.  When the playout buffer drops below the low
   threshold the server switches to the low bit rate encoding.  Assuming
   that the network has a bit more capacity than the low bit rate
   requires, the playout buffer will begin filling again.

   When the buffer crosses the high threshold the server may switch back
   to the high encoding rate.  Assuming that the network still doesn't
   have enough capacity for the high bit rate, the playout buffer will
   start draining again.  When it reaches the low threshold the server
   switches again to the low bit rate encoding.  The server will
   oscillate back and forth like this until the connection is concluded.

   If the network has insufficient capacity to support the low bit rate
   encoding, the playout buffer will eventually drain completely, and
   playback will need to be paused until the buffer refills to the
   playback start level.

   Note that, in this scheme, the server doesn't need to explicitly know
   the rate that TFRC has determined; it simply always sends as fast as
   TFRC allows (perhaps alternately reading a chunk of data from disk
   and then blocking on the socket write call until it's transmitted).
   TFRC shapes the stream to the network's requirements, and the playout
   buffer feedback allows the server to shape the stream to the
   application's requirements.

4.1.2 Issues With Strategy 1

   The advantage of this strategy is that it provides insurance against
   an unpredictable future.  Since there's no guarantee that a currently
   supported transmit rate will continue to be supported, the strategy
   takes what the network is willing to give when it's willing to give
   it.  The data is transferred from the server to the client perhaps
   faster than is strictly necessary, but once it's there no network
   problems (or new sources of traffic) can affect the display.


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   Silence suppression can be used with this strategy, since the
   transmitter doesn't actually go idle during the silence -- it just
   gets further ahead.  Variable rate video codecs can also function
   well.  Again, the transmitter will get ahead faster during the
   interpolated frames and fall back during the index frames, but a
   playout buffer of a few seconds is probably sufficient to mask these
   variations.

   One obvious disadvantage, if the client is a "thin" device, is the
   large buffer at the client.  A subtler disadvantage involves the way
   TFRC probes the network to determine its capacity.  Basically, TFRC
   does not have an a priori idea of what the network capacity is; it
   simply gradually increases the transmit rate until packets are lost,
   then backs down.  After a period of time with no losses, the rate is
   gradually increased again until more packets are lost.  Over the long
   term, the transmit rate will oscillate up and down, with packet loss
   events occurring at the rate peaks.

   This means that packet loss will likely be routine with this
   strategy.  For any given transfer, the number of lost packets is
   likely to be small, but non-zero.  Whether this causes noticeable
   quality problems depends on the characteristics of the particular
   codec in use.

4.2 Second Try -- One-way Live Media

   With one-way live media you can only transmit the data as fast as
   it's created, but end-to-end delays of several or tens of seconds are
   usually acceptable.

4.2.1 Strategy 2

   Assume that we have a playout media buffer at the receiver and a
   transmit media buffer at the sender.  The transmit buffer is filled
   at the encoding rate and drained at the TFRC transmit rate.  The
   playout buffer is filled at the network arrival rate and drained at
   the decoding rate.  The playout buffer has a playback start threshold
   and the transmit buffer has a switch encoding threshold and a discard
   data threshold.  These thresholds are on the order of several to tens
   of seconds.  Switch encoding is less than discard data, which is less
   than playback start.  Figure 4 shows this schematically.











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                   discard data
                     |  switch encoding
                     |   |                 playback start
                     |   |                   |
   media   +---------v---v---+          +----v-----------+
   ------->| Transmit buffer |--------->| Playout buffer |---> display
   source  +-----------------+ transmit +----------------+
           fill at             at TFRC rate          drain at
           encode rate                               decode rate

   Figure 4: One-way live media.


   At the start of the connection, the sender places data into the
   transmit buffer at the high encoding rate.  The buffer is drained at
   the TFRC transmit rate, which at this point is in slow-start and is
   probably slower than the encoding rate.  This will cause a backup in
   the transmit buffer.  Eventually TFRC will slow-start to a rate
   slightly above the rate necessary to sustain the encoding rate
   (assuming the network has sufficient capacity).  When this happens
   the transmit buffer will drain and we'll reach a steady state
   condition where the transmit buffer is normally empty and we're
   transmitting at a rate that is probably below the maximum allowed by
   TFRC.

   Meanwhile at the receiver, the playout buffer is filling, and when it
   reaches the playback start threshold playback will start.  After TFRC
   slow-start is complete and the transmit buffer is drained, this
   buffer will reach a steady state where packets are arriving from the
   network at the encoding rate (ignoring jitter) and being drained at
   the (equal) decoding rate.  The depth of the buffer will be the
   playback start threshold plus the maximum depth of the transmit
   buffer during slow start.

   Now assume that network congestion (packet losses) forces TFRC to
   drop its rate to below that needed by the high encoding rate.  The
   transmit buffer will begin to fill and the playout buffer will begin
   to drain.  When the transmit buffer reaches the switch encoding
   threshold, the sender switches to the low encoding rate, and converts
   all of the data in the transmit buffer to low rate encoding.

   Assuming that the network can support the new, lower, rate (and a
   little more) the transmit buffer will begin to drain and the playout
   buffer will begin to fill.  Eventually the transmit buffer will empty
   and the playout buffer will be back to its steady state level.

   At this point (or perhaps after a slight delay) the sender can switch
   back to the higher rate encoding.  If the new rate can't be sustained
   the transmit buffer will fill again, and the playout buffer will
   drain.  When the transmit buffer reaches the switch encoding
   threshold the sender goes back to the lower encoding rate.  This

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   oscillation continues until the stream ends or the network is able to
   support the high encoding rate for the long term.

   If the network can't support the low encoding rate, the transmit
   buffer will continue to fill (and the playout buffer will continue to
   drain).  When the transmit buffer reaches the discard data threshold,
   the sender must discard a chunk of data from the transmit buffer for
   every chunk of data added.  Preferably, the discard should happen
   from the head of the transmit buffer, as these are the stalest data,
   but the application could make other choices (e.g. discard the
   earliest silence in the buffer).  This discard behavior continues
   until the transmit buffer falls below the switch encoding threshold.
   If the playout buffer ever drains completely, the receiver should
   fill the output with suitable material (e.g. silence or stillness).

   Note that this strategy is also suitable for one-way pre-recorded
   media, as long as the transmit buffer is only filled at the encoding
   rate, not at the disk read rate.

4.2.2 Issues with Strategy 2

   Strategy 2 is fairly effective.  There is a limit on the necessary
   size of the playout buffer at the client, so clients with limited
   resources can be supported.  When silence suppression is used or
   motion compensation sends interpolated frames, the transmit rate will
   actually go down, and then must slowly ramp up to return to the
   maximum rates, but this smoothing can often be masked by a playout
   buffer of a few seconds.

   Also, since strategy 2 limits the transmission rate to the maximum
   encoding rate, and therefore doesn't try to get every last bit of
   possible throughput from the network, routine packet loss can be
   avoided (assuming that there's enough network capacity for the
   maximum encoding rate).

4.3 One More Time -- Two-way Interactive Media

   Two-way interactive media is characterized by its low tolerance for
   end-to-end delay, usually requiring less than 200 ms for interactive
   conversation, including jitter buffering at the receiver.  Rate
   adapting buffers will insert too much delay and the slow start period
   is likely to be noticeable ("Hello" clipping).

   This low delay requirement makes using TFRC with variable-rate codecs
   (codecs using silence suppression or motion compensation) highly
   problematic.  The extra delays imposed by the smooth rate increases
   mandated by TFRC are unlikely to be tolerated by the interactive
   applications.

   There are further problems with the usual practice in interactive
   voice applications of using small packets.  In voice applications,

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   the data rate is low enough that waiting to accumulate enough data to
   fill a large packet adds unacceptable delay.  For example, the G.711
   codec generates one byte of data every 125 microseconds.  To
   accumulate enough data for a 1480-byte packet, the encoder would need
   to delay some data by 185 ms, eating up nearly the entire delay
   budget just for packetization.  These considerations can also apply
   to very low rate video.

   The goal of TFRC is fair sharing of a bottleneck, in packets per
   second, with a TCP application using 1480-byte packets.  Applications
   using smaller packets will receive a fair share of packets per
   second, but less than a fair of bytes per second.  With the packet
   sizes typically in use in interactive voice applications (e.g., 80
   bytes of user data for G.711 with 10 ms packetization), it can be
   very difficult to achieve useful byte per second rates when in
   competition with TCP applications.

   Further research is needed to resolve these issues.  The strategy
   below can only be applied to constant rate codecs whose data rate is
   sufficiently large to fill 1480-byte packets within tolerable delay
   limits.

4.3.1 Strategy 3

   To start, the calling party sends an INVITE (loosely using SIP [RFC
   3261] terminology) indicating the IP address and port to use for
   media at its end.  Without informing the called user, the called
   system responds to the INVITE by connecting to the calling party
   media port.  Both end systems then begin exchanging test data, at the
   (slowly increasing) rate allowed by TFRC.  The purpose of this test
   data is to see what rate the connection can be ramped up to.  If a
   minimum acceptable rate cannot be achieved within some time period,
   the call is cleared (conceptually, the calling party hears "fast
   busy" and the called user is never informed of the incoming call).
   Note that once the rate has ramped up sufficiently for the highest
   rate codec there's no need to go further.

   If an acceptable rate can be achieved (in both directions), the
   called user is informed of the incoming call.  The test data is
   continued during this period.  Once the called user accepts the call,
   the test data is replaced by real data at the same rate.

   If congestion is encountered during the call, TFRC will reduce its
   allowed sending rate.  When that rate falls below the codec currently
   in use, the sender switches to a lower rate codec, but should pad its
   transmission out to the allowed TFRC rate.  Note that this padding is
   only necessary if the application wishes to return to the higher
   encoding rate when possible.  If the TFRC rate continues to fall past
   the lowest rate codec, the sender must discard packets to conform to
   that rate.


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   If the network capacity is sufficient to support one of the lower
   rate codecs, eventually the congestion will clear and TFRC will
   slowly increase the allowed transmit rate.  The application should
   increase its transmission padding to keep up with the increasing TFRC
   rate.  The application may switch back to the higher rate codec when
   the TFRC rate reaches a sufficient value.

   An application that did not wish to switch back to the higher
   encoding (perhaps for reasons outlined in section 3.1) would not need
   to pad its transmission out to the TFRC maximum rate.

   Note that the receiver would normally implement a short playout
   buffer (with playback start on the order of 100 ms) to smooth out
   jitter in the packet arrival gaps.

4.3.2 Issues with Strategy 3

   An obvious issue with strategy 3 is the post-dial call connection
   delay imposed by the slow-start ramp up.  This is perhaps less of an
   issue for two-way video applications, where post-dial delays of
   several seconds are accepted practice.  For telephony applications,
   however, post-dial delays significantly greater than a second are a
   problem, given that users have been conditioned to that behavior by
   the public telephone network.  On the other hand, the four packets
   per RTT initial transmit rate allowed by DCCP's CCID3 in some
   circumstance is likely to be sufficient for many telephony
   applications, and the ramp up will be very quick.

   As was stated in section 4.3, this strategy is only suitable for use
   with constant-rate codecs with fast enough data rates to tolerate
   using large packets.

5. Security Considerations

   There are no security considerations for this document.  Security
   consideration for TFRC and the protocols implementing TFRC are
   discussed in their defining documents.

6. IANA Considerations

   There are no IANA actions required for this document.

7. Thanks

   Thanks to the AVT working group, especially Philippe Gentric and
   Brian Rosen, for comments on the earlier version of this document.






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8. Informative References

   [DCCP]      E. Kohler, M. Handley, S. Floyd, J. Padhye, Datagram
               Congestion Control Protocol (DCCP), February 2004, draft-
               ietf-dccp-spec-06.txt, work in progress.

   [CCID2]     S. Floyd, E. Kohler, Profile for DCCP Congestion Control
               2: TCP-Like Congestion Control, February 2004, draft-
               ietf-dccp-ccid2-05.txt, work in progress.

   [CCID3]     S. Floyd, E. Kohler, J. Padhye, Profile for DCCP
               Congestion Control 3: TFRC Congestion Control, February
               2004, draft-ietf-dccp-ccid3-04.txt, work in progress.

   [RFC 3448]  M. Handley, S. Floyd, J. Padhye, J. Widmer, TCP Friendly
               Rate Control (TFRC): Protocol Specification, RFC 3448.

   [RFC 3714]  S. Floyd, J, Kempf, IAB Concerns Regarding Congestion for
               Voice Traffic in the Internet, March 2004, RFC 3714.

   [RFC 3261]  J. Rosenberg, et al, SIP: Session Initiation Protocol,
               June 2002, RFC 3261

   [RFC 3517]  E. Blanton, M. Allman, K. Fall, L. Wang, A Conservative
               Selective Acknowledgment (SACK)-based Loss Recovery
               Algorithm for TCP, April 2003, RFC 3517

   [RFC 3550]  H. Schulzrinne, S. Casner, R. Frederick, V. Jacobson,
               RTP: A Transport Protocol for Real-Time Applications,
               July 2003, RFC 3550

   [XTIME]     ITU-T: Series G: Transmission Systems and Media, Digital
               Systems and Networks, Recommendation G.114, One-way
               Transmission Time, May 2000

   [ECN]       K. Ramakrishnan, S. Floyd, D. Black, The Addition of
               Explicit Congestion Notification (ECN) to IP, September
               2001, RFC 3168

   [MPEG4]     ISO/IEC International Standard 14496 (MPEG-4),
               Information technology - Coding of audio-visual objects,
               January 2000

   [RTP-TFRC]  L. Gharai, RTP Profile for TCP-Friendly Rate Control,
               October 2004, draft-ietf-avt-tfrc-profile-03.txt, work in
               progress






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9. Author's Address

   Tom Phelan
   Sonus Networks
   5 Carlisle Rd.
   Chelmsford, MA USA 01824
   Phone: +1-978-614-8456
   Email: tphelan@sonusnet.com












































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