Wave and Equation Based Rate Control (WEBRC) Building Block
draft-ietf-rmt-bb-webrc-04

Versions: 00 01 02 03 04 rfc3738                            Experimental
                                                        IPR declarations
Internet Engineering Task Force                                   RMT WG
INTERNET-DRAFT                                  M. Luby/Digital Fountain
draft-ietf-rmt-bb-webrc-04.txt               V. K Goyal/Digital Fountain
                                                         9 December 2002
                                                      Expires: June 2003


          Wave and Equation Based Rate Control building block



Status of this Document

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

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                                Abstract


     Wave and Equation Based Rate Control provides rate and
     congestion control for data delivery.  Wave and Equation Based
     Rate Control is specifically designed to support protocols
     using IP multicast.  It provides multiple-rate, congestion-
     controlled delivery to receivers, i.e., different receivers
     joined to the same session may be receiving packets at
     different rates depending on the bandwidths of their



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     individual connections to the sender and on competing traffic
     along these connections.  Wave and Equation Based Rate Control
     requires no feedback from receivers to the sender, i.e., it is
     a completely receiver-driven congestion control protocol.
     Thus, it is designed to scale to potentially massive numbers
     of receivers attached to a session from a single sender.
     Furthermore, because each individual receiver adjusts to the
     available bandwidth between the sender and that receiver,
     there is the potential to deliver data to each individual
     receiver at the fastest possible rate for that receiver, even
     in a highly heterogeneous network architecture, using a single
     sender.







































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                           Table of Contents


     1. Introduction. . . . . . . . . . . . . . . . . . . . . .   4
     2. Rationale . . . . . . . . . . . . . . . . . . . . . . .   6
     3. Functionality . . . . . . . . . . . . . . . . . . . . .   7
      3.1. Sender Operation . . . . . . . . . . . . . . . . . .   9
       3.1.1. Sender inputs and initialization. . . . . . . . .   9
       3.1.2. Sending packets to the session. . . . . . . . . .  11
      3.2. Receiver Operation . . . . . . . . . . . . . . . . .  12
       3.2.1. Receiver inputs and initialization. . . . . . . .  13
       3.2.2. Receiver measurements and calculations. . . . . .  14
        3.2.2.1. Average loss probability . . . . . . . . . . .  14
        3.2.2.2. Average round-trip time. . . . . . . . . . . .  16
        3.2.2.3. Rate Equation. . . . . . . . . . . . . . . . .  17
        3.2.2.4. Epochs . . . . . . . . . . . . . . . . . . . .  17
        3.2.2.5. Average reception rate . . . . . . . . . . . .  17
        3.2.2.6. Slow start . . . . . . . . . . . . . . . . . .  19
        3.2.2.7. Target rate. . . . . . . . . . . . . . . . . .  20
       3.2.3. Receiver events . . . . . . . . . . . . . . . . .  20
        3.2.3.1. Packet reception . . . . . . . . . . . . . . .  21
        3.2.3.2. First packet after join. . . . . . . . . . . .  21
        3.2.3.3. Time slot change . . . . . . . . . . . . . . .  21
        3.2.3.4. Loss event . . . . . . . . . . . . . . . . . .  22
        3.2.3.5. Epoch change . . . . . . . . . . . . . . . . .  22
        3.2.3.6. Join the next higher layer . . . . . . . . . .  22
        3.2.3.7. Join timeout . . . . . . . . . . . . . . . . .  23
        3.2.3.8. Exceptional timeouts . . . . . . . . . . . . .  24
     4. Applicability Statement . . . . . . . . . . . . . . . .  24
      4.1. Environmental Requirements and
      Considerations. . . . . . . . . . . . . . . . . . . . . .  24
     5. Packet Header Fields. . . . . . . . . . . . . . . . . .  26
      5.1. Short Format Congestion Control Information. . . . .  27
      5.2. Long Format Congestion Control Information . . . . .  28
     6. Requirements From Other Building Blocks . . . . . . . .  29
     7. Security Considerations . . . . . . . . . . . . . . . .  29
     8. IANA Considerations . . . . . . . . . . . . . . . . . .  30
     9. Intellectual Property Issues. . . . . . . . . . . . . .  30
     10. References . . . . . . . . . . . . . . . . . . . . . .  31
     11. Authors' Addresses . . . . . . . . . . . . . . . . . .  32
     12. Full Copyright Statement . . . . . . . . . . . . . . .  33










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1.  Introduction

This document specifies Wave and Equation Based Rate Control (WEBRC).
WEBRC is a congestion control building block that is designed to be
massively scalable when used with the IP multicast network service.
WEBRC is also suitable as the basis for unicast congestion control, but
this is outside the scope of this document.  WEBRC is designed to
compete fairly with TCP and similar congestion-controlled sessions.
WEBRC can be used as a congestion control protocol for any type of data
delivery, including reliable content delivery and streaming delivery.

WEBRC is a receiver-driven congestion control protocol in the spirit of
[3] and [18]. This means that all measurements and decisions to raise or
lower the reception rate are made by each individual receiver, and these
decisions are acted upon by sending join and leave messages for channels
to the network.  A receiver using WEBRC adjusts its reception rate
without regard for other concerns such as reliability.  This is
different from TCP, where the congestion control protocol and the
reliability protocol are intricately interwoven.

WEBRC takes the same basic equation-based approach as TFRC [7]. In
particular, each WEBRC receiver measures parameters that are plugged
into a TCP-like equation to compute the receiver target reception rate,
and adjusts its reception rate up and down to closely approximate the
target reception rate.  The sender sends packets to multiple channels;
one channel is called the base channel and the remaining channels are
called wave channels.  Each wave channel follows the same pattern of
packet rate transmission spread out over equally spaced intervals of
time.  The pattern of each wave is that it starts at a high rate that
decreases gradually and continually over a long interval of time.
(Picture an infinite sequence of waves.)  The receiver increases its
reception rate by joining the next wave channel earlier in the descent
of the wave than it joined the previous wave channel, and the receiver
decreases its reception rate by joining the next wave channel later in
the descent of the wave than it joined the previous wave channel.

The wave channels are ordered at each point in time from a lowest layer
to a highest layer.  At each point in time, the lowest layer is the wave
channel that has the smallest rate among all active wave channels and
the highest layer is the wave channel that has the highest rate.
Because waves are dynamically becoming active and quiescent over time,
the designation of which wave channel is at which layer changes
dynamically over time.  In addition to being joined to the base channel,
at each point in time a receiver is joined to a consecutive set of
layers starting at the lowest layer and proceeding towards the highest.

WEBRC introduces a natural notion of a multicast round-trip time (MRTT).
An MRTT is measured individually by each receiver and averaged as a



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substitute for conventional unicast round-trip time (RTT).  Because the
throughput of a TCP session depends strongly on RTT, having some measure
of RTT is essential in making the WEBRC equation-based rate control
protocol ``TCP-friendly.''  The use of the MRTT also helps to coordinate
and equalize the reception rates of proximate receivers joined to a
session behind a bottleneck link.  This implies that packets for the
session that flow through the bottleneck link are on average sent to
almost all downstream receivers, and thus the efficiencies of multicast
are realized.  Furthermore, WEBRC is designed to be massively scalable
in the sense that the sender is insensitive to the number of receivers
joined to a multicast session.

WEBRC is designed for applications that use a fixed packet size and vary
their packet reception rates in response to congestion.  WEBRC is
designed to be reasonably fair when competing for bandwidth with TCP
flows, where a flow is ``reasonably fair'' if its reception rate is
generally within a factor of two of the reception rate of a TCP flow
under the same conditions.  However WEBRC has a much lower variation of
throughput over time compared to TCP, which makes it more suitable for
applications such as telephony or streaming media where a relatively
smooth rate is of importance.  The penalty of having smoother throughput
than TCP while competing fairly for bandwidth is that WEBRC responds
more slowly than TCP to changes in available bandwidth.

The receiver measures and performs the calculation of congestion control
parameters (e.g., the average loss probability, the average MRTT) and
makes decisions on how to increase or decrease its rate based on these
parameters.  The receiver-based approach is well suited to an
application where the sender is handling many concurrent connections and
therefore WEBRC is suitable as a building block for multicast congestion
control.

The paper [15] and technical report [14] provide much of the rationale
and intuition for the WEBRC design and describe some preliminary
simulations.

This document describes a building block as defined in RFC3048 [19].
This document describes a congestion control building block that
conforms to RFC2357 [16]. This document is a product of the IETF RMT WG
and follows the general guidelines provided in RFC3269 [9].  The key
words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD",
"SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are
to be interpreted as described in RFC2119 [2].








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2.  Rationale

WEBRC provides congestion control for massively scalable protocols using
the IP multicast network service.  The congestion control that WEBRC
provides is common to a variety of applications, including reliable
content delivery and streaming applications.

WEBRC is designed to provide congestion control for all packets that are
sent to a session.  A session comprises multiple channels originating at
a single sender that are used for some period of time to carry packets
pertaining to the transmission of one or more objects that can be of
interest to receivers.  The logic behind defining a session as
originating from a single sender is that this is the right granularity
to regulate packet traffic via congestion control.  The rationale for
providing congestion control that uses multiple channels within the same
session is that this allows the data on the channels to be layered,
which in turn allows each receiver to control its reception rate by
joining and leaving channels during its participation in the session.
There are advantages to layered data for streaming, where the most
important data can be sent to the lower layers and incrementally
valuable data to the higher layers.  For reliable content delivery, as
described in [12], an application can send in packets encoded data
generated from an object in such a way that the arrival of enough
packets by a receiver is sufficient to reliably reconstruct the original
object.  A primary advantage of WEBRC is that each receiver controls it
reception rate independent of other receivers.  Thus, for example, a
receiver with a slow connection to the sender does not slow down the
receivers with faster connections.

There are coding techniques that provide massively scalable reliability
and asynchronous delivery which are compatible with WEBRC, e.g., as
described in [10]. When combined the result is a massively scalable,
reliable, asynchronous content delivery protocol that is network
friendly.  WEBRC also provides congestion control that is suitable for
streaming applications.

WEBRC avoids using techniques that are not massively scalable.  For
example, WEBRC does not provide any mechanisms for sending information
from receivers to senders, although this does not rule out protocols
that both use WEBRC and that send information from receivers to senders.

WEBRC provides congestion control that can be tuned for different
applications that may have differing application requirements. For
example, a content delivery protocol may aggressively strive to use all
available bandwidth between receivers and the sender, and thus to
maintain fairness it must drastically reduce its rate when there is
competing traffic.  On the other hand, a streaming delivery protocol may
strive to maintain a constant rate instead of trying to use all



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available bandwidth, and thus it may not reduce its rate as fast when
there is competing traffic.

WEBRC does not provide any support beyond congestion control, and thus
WEBRC is to be combined with other building blocks to provide a complete
protocol instantiation.  For example, WEBRC does not provide any means
that can be used to identify which session each received packet belongs
to.  As another example, WEBRC does not provide support for identifying
which object each packet is carrying information about.



3.  Functionality

A WEBRC session comprises a logically related set of channels
originating from a single sender that are used for some period of time
to carry data packets with a header carrying WEBRC Congestion Control
Information.  When packets are received, they are first checked to see
that they belong to the appropriate session before WEBRC is applied.  A
session label defined by a protocol instantiation may be carried in each
packet to identify to which session the packet belongs.  For example, if
LCT [11] is being used with the session, then the sender IP address
together with the Transport Session Identifier supported by LCT would be
used to determine which session a received packet belongs to.  The
particular details of how this filtering is performed it outside the
scope of this document.  In the remainder of this document, references
to channels are always within the scope of a single session.

A channel can be uniquely identified at the network layer by a (sender
IP address, multicast group address) pair, and this is the address to
which the receiver sends messages to join and leave the channel.  The
channels used by a WEBRC session are mapped uniquely to consecutive
channel numbers.  In each packet sent to a channel, the channel number
that corresponds to the channel is carried in the WEBRC Congestion
Control Information.  A WEBRC receiver uses the channel number to
determine which channel within a session a packet is received from.

At the sender, time is partitioned into time slots, each of duration TSD
seconds.  There are a fixed number T of time slot indices associated
with a session.  As time progresses, the current time slot index
increases by one modulo T each TSD seconds.  The current time slot index
CTSI is carried in the WEBRC Congestion Control Information.  This
allows receivers to perform very coarse-grained synchronization within a
session.

WEBRC congestion control is achieved by having the sender send packets
associated with a given session to several different channels.
Individual receivers dynamically join and leave these channels according



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to the network congestion they experience.  These congestion control
adjustments are performed at each receiver independently of all other
receivers, without any impact on the sender.  A packet sequence number
is carried in the WEBRC Congestion Control Information.  The packet
sequence numbers are consecutively numbered per channel and are used by
receivers to measure packet loss.

The channels associated with a session consist of one base channel and T
wave channels.  The packet rate for each channel varies over time.  For
the base channel, packets are sent to the channel at a low rate BCR_P at
the beginning of a time slot and this rate gradually decreases to
P*BCR_P at the end of the time slot, where P < 1 is a constant defined
later.  This pattern for the base channel repeats over each time slot.
For each wave channel i, packets are sent to channel i at a rate that
first increases very quickly to a high rate and then decreases over time
by a fixed fraction P per time slot until a rate of BCR_P is reached at
the end of time slot i.  Then, for a period of time called the quiescent
period, no packets are sent to wave channel i, and thereafter the whole
cycle repeats itself, where the duration of the cycle is T*TSD seconds.
Thus, the wave channels are going through the same cyclic pattern of
packet rate transmission spaced out evenly by TSD seconds.

Before joining a session, the receivers MUST obtain enough of the
session description to start the session.  This MUST include the
relevant session parameters needed by a receiver to participate in the
session and perform WEBRC congestion control.  The session description
is determined by the sender and is typically communicated to the
receivers out of band.  How receivers obtain the session description is
outside the scope of this document.

When a receiver initiates a session, it first joins the base channel.
The packets in the base channel help the receiver orient itself in terms
of what the current time slot index is, which in turn allows the
receiver to know the relative rates on the wave channels.  The receiver
remains joined to the base channel for the duration of its participation
in the session.

At each point in time the active (non-quiescent) wave channels are
ordered into layers, where the lowest layer is the active wave channel
whose wave is nearest to completion and the highest layer is the active
wave channel whose wave is furthest from completion.  (This is almost
the same as saying that the lowest layer has the lowest rate and the
highest layer has the highest rate.  The possible deviation from this is
due to the optional non-exponential beginnings of the waves as described
in [6].) Each time a wave channel becomes active, it is the highest
layer.  At the end of each time slot the lowest layer wave channel
becomes quiescent, and thus all active wave channels move down a layer
at this point in time.  At each point in time a receiver is joined to



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the base channel and a consecutive set of layers starting with the
lowest.  Each time a receiver joins a wave channel it joins the lowest
layer not yet joined.  A receiver always leaves the lowest layer when it
becomes quiescent.

After joining a session the receiver adjusts its rate upwards by joining
wave channels in sequence, starting with the lowest layer and moving
towards the highest.  The rates on the active wave channels are
decreasing with time, so the receiver adjusts its rate downwards simply
by refraining from joining additional wave channels.  Since the layer
ordering among the channels changes dynamically over time depending on
the current time slot index, it is important that the receiver
continually monitor the current time slot index contained in received
packets.  The reception rate at the receiver is determined by how early
each wave channel is joined by the receiver: the earlier the receiver
joins a channel with respect to when its wave started, the higher the
reception rate.

Once the receiver is joined to a wave channel, the receiver remains
joined to the wave channel until the channel goes quiescent, at which
point the receiver MUST leave the channel.

The way the receiver adjusts its reception rate is inspired by TFRC [7].
The receiver at all points in time maintains a target reception rate,
and the receiver is allowed to join the next wave channel if after
joining its anticipated reception rate from all the layers it is joined
to would be at most its target reception rate.  The target rate is
continually updated based on a set of measured parameters.  The primary
parameters are an estimate LOSSP of the average loss probability and an
estimate ARTT of the average multicast round-trip time.



3.1.  Sender Operation

The sender operation is by design much simpler than the receiver
operation.



3.1.1.  Sender inputs and initialization

The primary input to the sender for the session is SR_b.  SR_b is an
upper bound to the sender transmission rate in bits per second at any
point in time (with some reasonable granularity) in aggregate to all
channels.  Naturally, this is then also the maximum rate in bits per
second that any  receiver could receive data from the session at any
point in time.  It is RECOMMENDED that the sender transmission rate in



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aggregate to all channels be made constant as described in [6]. It is
also RECOMMENDED that the session description indicate whether the
aggregate transmission rate is constant, unless there is no ambiguity.

The secondary inputs to the sender are listed below.  These inputs are
secondary because their values will generally be fixed to default values
that will not change, because they will be derived from SR_b, or because
they are chosen based on non-WEBRC considerations.

  o LENP_B is the length of packets in bytes sent to the session.  The
    value of LENP_B depends on the complete protocol, but in general
    this SHOULD be set to as high a value as possible without exceeding
    the MTU size for the network that would cause fragmentation.

  o BCR_P is the transmission rate on the base channel at the beginning
    of a time slot in packets per second.  The default value for BCR_P
    is 1.

  o TSD is the time slot duration measured in seconds.  The RECOMMENDED
    value for TSD is 10.

  o QD is the minimum quiescent period duration measured in seconds.
    The RECOMMENDED value for QD is 300.

  o P is the multiplicative drop in every channel rate over each time
    slot.  The default value for P is 0.75.

  o N is the duration in time slots for each wave.  N is also the number
    of wave channels active at any time.  (A wave channel is called
    active when it is not quiescent.)  A sender may choose any value
    that allows it to produce waves that substantially follow the
    required exponential shape described in Section 3.1.2. A RECOMMENDED
    mechanism for relating N to SR_b, BCR_P and P is described in [6].

>From these inputs the following fixed sender parameters can be derived
as follows.

  o SR_P = SR_b/(8*LENP_B) is the sender transmission rate in packets
    per second.

  o BCR_b = 8*LENP_B*BCR_P is the rate of the base channel at the
    beginning of a time slot in bits per second.

  o L = ceil(BCR_P*TSD*(P-1)/log(P)) is the number of base channel
    packets sent in each time slot.

  o Q = ceil(QD/TSD) is the number of quiescent time slots per cycle for
    a wave channel.



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  o T = N + Q is the total number of time slots in a cycle.  T is also
    the total number of wave channels.

  o For the base channel CN = T and for the wave channels CN =
    0,1,...,T-1.  The sender has the description of the channels
    assigned to the session and the mapping between the channels and the
    CNs.

  o C = TSD*T is the total duration of a cycle in seconds.



3.1.2.  Sending packets to the session

The sender keeps track of the current time slot index CTSI.  The value
of CTSI is incremented by 1 modulo T each TSD seconds.  The value of
CTSI is placed into each packet in the format described in Section 5.
For each packet sent to the session, the sender also places the channel
number CN of the channel into the packets in the format described in
Section 5. Recall that CN = T for the base channel and CN = 0,1,...,T-1
for the wave channels.

For each packet sent to the session, the sender calculates a packet
sequence number PSN and places it into the packet.  The value of PSN is
scoped by CN, and the value of PSN is consecutively increasing within
each channel.  Furthermore, for each wave channel, the last packet sent
before the channel becomes quiescent must have the maximum possible PSN
value.  When the short format for Congestion Control Information is used
(see Section 5.1), this implies that for any wave channel the last PSN
value sent to the channel just before the channel becomes quiescent is
2^16-1 = 65 535.  Similarly, when the long format for Congestion Control
Information is used (see Section 5.2), the PSN for the final packet of
any wave is 2^32-1 = 4 294 967 295.  The PSN of the initial packet of a
wave thus depends on TSD, P, BCR_P and SR_P.  For the base channel, the
first packet of each time slot has a PSN congruent to zero modulo L.
Hence, instead of 2^16 - 1 or 2^32 - 1 being the highest PSN used
(depending on the choice of short format or long format Congestion
Control Information), the highest PSN is one less than the largest
multiple of L that does not exceed 2^16 (short format) or 2^32 (long
format).  The format for the PSN within packets is described in Section
5.

The rate at which packets are sent to the base channel starts at BCR_P
packets per second at the beginning of each time slot and decreases
exponentially to P*BCR_P at the end of that time slot.






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The packet rate for the wave channels is more complicated.  Each wave
channel carries a sequence of waves separated by quiescent periods.  On
each wave channel each wave is active during N time slots followed by a
quiescent period of Q time slots.  The waves on wave channel i end at
the ends of time slots with CTSI i.  Therefore wave channel i is active
during time slots i-N+1 modulo T, i-N+2 modulo T, ..., i and is
quiescent for time slots i+1 modulo T, i+2 modulo T, ..., i+Q modulo T.
Wave channel i first becomes active within time slot i-N+1 modulo T at a
point in time that may depend on the value of SR_b.


For a substantial fraction of the duration of a wave ending at the end
of a wave, the packet rate MUST decrease exponentially by a factor of P
per TSD seconds, with a rate of BCR_P at the end of the last active time
slot.  At the beginning of each wave, the rate MAY deviate from this
exponential form so that the total sending rate in aggregate to all of
the channels is constant.  A RECOMMENDED design for the beginnings of
waves to achieve this goal is described in [6].


3.2.  Receiver Operation

The bulk of the complexity in WEBRC is in the receiver operation.  For
ease of explanation, suppose for the moment that during the reception
there is no packet loss and packets are arriving at exactly the rate at
which they were sent.  The sender transmission rate to the channels is
designed so that the receiver reception rate behaves as follows.

Upon entering a session, the receiver immediately joins the base
channel.  When the receiver wants to increase its rate, it joins
consecutive layers starting with the lowest and moving towards the
highest. (Recall that the designations of lowest to highest change as
waves become active and quiescent.)  When the receiver wants to maintain
its current reception rate and it is already joined to the lowest NWC
layers, if the receiver joins channel i-1+NWC modulo T sometime during
time slot i then the receiver joins channel i+NWC modulo T TSD seconds
later in time slot i+1.  When the lowest layer becomes quiescent the
receiver leaves the channel.

Suppose the receiver wants to decrease its rate till it is joined to
just the base channel.  Assume that a receiver is joined to the lowest
NWC < N-2 layers at the beginning of time slot i, i.e., wave channels i,
i+1 modulo T,..., i+NWC-1 modulo T.  Then, the aggregate packet
reception rate of the receiver over the next NWC time slots will behave
as follows if the receiver does not join any wave channels during this
time.  At the beginning of time slot i the receiver reception rate is
BCR_P*(1 + (1/P) + (1/P)^2 + ... + (1/P)^NWC).  Then the receiver
reception rate decreases by a factor of P over the duration of each time



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slot, and at the end of each time slot the reception rate decreases by
an additive amount of P*BCR_P.  At the end of time slot i+NWC-1 mod T,
the receiver reception rate is BCR_P*(1+P), and at the beginning of time
slot i+NWC mod T the receiver is joined only to the base channel and its
reception rate is BCR_P.



3.2.1.  Receiver inputs and initialization

Before joining a session the receiver MUST know the mapping between the
CNs and the channels.  Upon joining the session or shortly thereafter,
it SHOULD have the values of LENP_B, BCR_P, TSD, P, N, L, Q and T.  Some
of these values may be computed or measured once the receiver has joined
the session.  For example, the receiver MAY obtain LENP_B and T from the
first packet received from the base channel, and the receiver MAY
measure BCR_P once it is joined to the base channel.  The values of P, Q
and TSD MAY be fixed to default values built into the receiver that do
not change from session to session, and the value of N MAY be computed
as T-Q.  The receiver SHOULD know whether the sender is employing a
technique to produce constant aggregate rate as described in [6].

When a receiver first joins a session, it MUST first join just the base
channel and start receiving packets to determine the current time slot
index.  If during the course of the session the receiver continually
loses a high fraction of the packets from the base channel even when the
receiver is only joined to the base channel, the receiver SHOULD leave
the session.

The receiver MAY also have other individually set parameters that may be
used to determine its behavior.  One such parameter is MRR_b:

  o MRR_b is the maximum receiver reception rate in bits per second.
    This may be used to determine the maximum reception rate this
    receiver is willing to reach.  Thus, the maximum reception rate that
    the receiver can possibly achieve in the session is the minimum of
    SR_b and MRR_b.  A recommended value of MRR_b for a receiver is the
    bandwidth capacity of the last link to the receiver.  MRR_P is the
    maximum receiver reception rate in packets per second, i.e., MRR_P =
    MRR_b/(8*LENP_B).



3.2.2.  Receiver measurements and calculations

As outlined in the introduction, the way a receiver adjusts its
reception rate is inspired by TFRC [7]. The receiver at all points in
time maintains a target reception rate, and the receiver is allowed to



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join the next wave channel if joining would increase its reception rate
to at most its target reception rate.  The target rate is continually
updated based on a set of measured parameters.

Two primary parameters are the estimate LOSSP of the average loss
probability and the estimate ARTT of the average MRTT.  Both LOSSP and
ARTT are moving averages of measurements based on discrete events.   For
many of the other estimates calculated by WEBRC, using an exponentially
weighted moving average (EWMA) with a fixed averaging fraction is
sufficient.  However, the calculations of LOSSP and ARTT require a more
general and sophisticated filtering approach.



3.2.2.1.  Average loss probability

The design of TFRC [7] reflects that, because the average packet loss
probability can vary by orders of magnitude, any estimate of the average
loss probability based on either a fixed number of packets or on a fixed
period of time with a fixed averaging fraction will be poor.  In TFRC
the average is estimated from the numbers of packets between beginnings
of loss events, and the number of loss events used is fixed.

The estimate LOSSP of the average loss probability of the receiver is
maintained in a manner somewhat similar to that described in TFRC [7].
The WEBRC receiver estimates the inverse of the average loss probability
by applying two EWMA filters to the packet reception measurements, a
time-based filter with smoothing constant 0 < Nu < 1 and a loss-based
filter with smoothing constant 0 < Delta < 1.  The recommended values
for the smoothing constants are Nu = 0.3 and Delta = 0.3.  The reason
for the time-based filter is that the loss events in WEBRC are bursty;
they typically occur just after a new wave has been joined.  To smooth
out this burstiness, the time-based filter is applied to the packet
reception measurements at the end of each epoch to smooth out the bursty
loss events over a few time slot durations.  Intuitively, the time-based
filter averages packet reception events such that the events are
smoothed out over an interval of time proportional to TSD/Nu seconds.
The loss-based filter, similar to what is suggested in TFRC, is applied
to the output of the time-based filter to produce the estimate of the
inverse of the average loss probability.  Intuitively, the loss-based
filter averages loss events such that each loss event is averaged in
with weight Delta.

As described later, LOSSP is initialized at the end of slow start and
occasionally reset due to other events.  Let W and X be counts of
packets, let Y be a count of loss events and let Z be the long-term
estimate of the inverse of the average loss probability.  Whenever the
value of LOSSP is initialized or reset, the values of W, X, Y and Z are



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also initialized or reset.

Recall that TSD is the duration of a time slot.  The epoch length EL is
the duration of time between decisions to adjust the reception rate.
Generally EL is much smaller than TSD, and the RECOMMENDED values are EL
= 0.5 seconds and TSD = 10 seconds.

Define G = Nu*EL/TSD as the amount of time-based smoothing to perform at
the end of each epoch.  The update rules for W, Y, Z and LOSSP are the
following:


  o At the end of each epoch, adjust X, Y and Z and compute LOSSP as
    follows:

        Z = Z*(1-Delta)^(G*Y) + X/(Y+1)*(1-(1-Delta)^(Y+1))

        X = X*(1-G)

        Y = Y*(1-G)

        Z1 = Z*(1-Delta)^Y + X/(Y+1)*(1-(1-Delta)^(Y+1))

        Z2 = Z*(1-Delta)^(Y+1) + (X+W+1)/(Y+2)*(1-(1-Delta)^(Y+2))

        LOSSP = 1/max{Z1,Z2,1}


  o For each packet event (whether it is a received packet or a lost
    packet), W = W + 1


  o At the beginning of each loss event, update X, Y and Z as follows:

        X = X + W

        W = 0

        Y = Y + 1


The intuition behind these update rules is the following.  If just loss-
filtering were used to update Z, then Z would be decreased by a
multiplicative amount 1 - Delta for each loss event and Z would be
increased by an additive amount Delta for each packet.  To smooth out
loss events over more than one time slot, these adjustments are filtered
into Z over time, at the rate of a fraction G at the end of each epoch.
Thus, the variables X and Y are counts of the portions of the packets



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and loss events, respectively, that have not yet been filtered into the
long-term memory Z.  W is the count of packets since the last loss event
started.  This explains why W is increased by one for each packet and Y
is increased by one for each loss event.  At the end of each epoch a
fraction G of both X and Y are filtered into Z according to the loss-
filter rule described above, and then the same fraction G is removed
from both X and Y to account for the fact that this portion has been
filtered into Z.  The LOSSP calculation combines the short-term history
(X,Y) with the long-term history Z and also allows the arrivals since
the last loss W to have some influence.  The value of Z2 is what Z1
would become were the next packet to be lost.


To reset the loss calculation to a value LOSSP = a, the state variables
are set as follows:

        W = 0

        X = 0

        Y = 0

        Z = 1/a



3.2.2.2.  Average round-trip time

The receiver maintains an average round-trip time, ARTT, as a
measurement-based filter of MRTT measurements using a smoothing constant
0 < Alpha < 1.  The RECOMMENDED value for Alpha is 0.25.

Each time the receiver joins a channel (either the base channel upon
entering a session or wave channels continually), it makes a measurement
of the multicast round-trip time MRTT as follows.  Let V be an auxiliary
variable that is used that keep track of the average of the square of
the MRTT measurements.  When the receiver sends the join for the channel
it records the current time JoinTime and sets a Boolean variable JOINING
to true.  When the first packet is received from the channel the
receiver records the current time FirstTime and resets the value of
JOINING to false.  If it is the base channel that has been joined, ARTT
is set to FirstTime-JoinTime and V is set to ARTT*ARTT.  Otherwise, the
value of MRTT is set to (FirstTime - JoinTime) - log(1/P)/2/(1-P)/BCR_P
* P^NWC.  (Note that this value can be negative.)  Then, ARTT is updated
as follows.  Let Omega = Alpha*ARTT*ARTT/V, and at the Kth MRTT
measurement let Rho = Omega/(Omega+(1-Omega)*(1-(1-Omega)^K)).  (Note
that as K grows Rho approaches Omega.)  Then, V is updated to
(1-Rho)*V+Rho*MRTT*MRTT and ARTT is updated to



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max{P*ARTT,(1-Rho)*ARTT+Rho*MRTT}.

Usually ARTT is updated to the second term in the max, and in this case
ARTT is the EWMA of the previous value of ARTT and the new MRTT, with a
weighting on the new MRTT that as K grows is proportional to the square
of the previous ARTT divided by the previous average V of the square of
the MRTT.  Thus, if there is not much variance in the previous MRTTs
relative to the square of their average then the new MRTT will be
filtered into ARTT with a high weight, whereas  if there is a lot of
variance in the previous MRTTs relative to the square of their average
then the new MRTT will be filtered into ARTT with a low weight.  The
intuitive rationale for this is that in general the number of
measurements needed to compute a meaningful average for a random
variable is proportional to its variance divided by the square of its
average; see, e.g., [4]. By making the weight factor depend on previous
measurements in this way, the appropriate weight to use to average the
new MRTT into the ARTT self-adjusts automatically to the variability in
the measurements.



3.2.2.3.  Rate Equation

The receiver calculates the reception rate REQN based on the TCP
equation as follows: REQN = 1/(ARTT*sqrt{LOSSP}(0.816 +
7.35*LOSSP*(1+32*LOSSP^2))).  This equation comes from TFRC [7].


3.2.2.4.  Epochs

The receiver makes decisions on whether or not to join another wave
channel at equally spaced units of time called epochs.  The duration of
an epoch in seconds, EL, is set to be a small fraction of TSD, so that
decisions to join a channel can be made at a much finer granularity than
TSD.  A standard setting is EL = TSD/20.  Thus, with the recommended
setting of TSD = 10, it is RECOMMENDED that EL = 0.5.



3.2.2.5.  Average reception rate

There are two averaged reception rates maintained by the receiver:
TRR_P, the true reception rate, and ARR_P, the anticipated reception
rate.  These are used for different purposes and thus are calculated
quite differently.  Recommended values for the filtering weights Beta
and Zeta are provided at the end of this subsection.

In start-up mode, the true reception rate TRR_P is used to ensure that



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the receiver does not increase its reception rate too quickly above its
current reception rate.  In the transition from start-up mode to normal
operation and in normal operation, TRR_P is used in setting the slow
start rate.  TRR_P is calculated based on the measurement of RR_P, where
RR_P is the receiver reception rate in packets per second measured at
the beginning of an epoch averaged over the epoch that just ended.
TRR_P is initialized to BCR_P + k*log(P)/TSD when the first base channel
packet of the session arrives, where k is the PSN of the packet reduced
modulo L.  TRR_P is updated to (1-Zeta)*TRR_P + Zeta*RR_P at the
beginning of each epoch after RR_P is measured for the previous epoch.

The anticipated reception rate ARR_P is the receiver's estimate of the
total instantaneous rate of the currently joined channels.  It is used
to compare against the target rate to decide whether or not the receiver
should increase its reception rate by joining the next higher unjoined
layer.  ARR_P is calculated based on a measurement IRR_P and on the
number of joined wave channels NWC.  The ideal reception rate IRR_P is
the reception rate in packets per second including both received and
lost packets; like RR_P, it is measured at the beginning of the epoch
and averaged over the previous epoch.  ARR_P, IRR_P and NWC are updated
as follows:

  o NWC is initialized to 0.

  o When the first base channel packet arrives, ARR_P is set to BCR_P +
    k*log(P)/TSD, where k is the PSN of the packet reduced modulo L.

  o At the beginning of each epoch, IRR_P is measured over the previous
    epoch and then ARR_P is updated to P^(EL/TSD)*(1-Beta)*ARR_P +
    Beta*IRR_P.

  o When a join is made to the next higher unjoined layer, NWC is
    updated to NWC+1 and then ARR_P is multiplicatively increased by the
    factor ((1/P)^(NWC+1)-1)/((1/P)^NWC-1).  (Joins happen at epoch
    boundaries; this adjustment is in addition to the adjustment above.)

  o Each time a next time slot index is detected, ARR_P is additively
    increased by (1-P)*BCR_P to account for the change in rate on the
    base channel.  In addition, the bottom layer in the previous time
    slot has just gone quiescent and thus a message to leave this layer
    has been sent, ARR_P is additively decreased by BCR_P and NWC is
    decremented by 1.

Consider for the moment what happens if Beta = 0 and ARR_P is an
accurate estimate of the total rate of the joined channels.  The
adjustments to ARR_P upon joining and leaving wave channels, with the
passage of epochs, and with the detection of time slot changes will then
cause ARR_P to remain an accurate estimate.  In practice, Beta MUST be



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positive; allowing an influence of IRR_P prevents ARR_P from drifting
away from being an accurate estimate of the total joined rate.

The motivation for separate estimates TRR_P and ARR_P is as follows.
ARR_P is needed for comparison with the TFRC-inspired target rate
because there is no lag before it reflects the potential rate increase
resulting from joining the next higher layer and because it measures the
total possible impact on the network since it also includes lost
packets.  TRR_P is needed because it reflects the rate of data arriving
at the receiver and this is used to ensure that there is not a large gap
between the joined rate and the receiving rate.

The recommended values for Beta and Zeta depend on whether the receiver
is in start-up mode (SSR_P = infinity).  In start-up mode, it is
RECOMMENDED that Beta = (1 - P^(0.25))/2 and Zeta = sqrt(P)/(1 +
sqrt(P)).  In normal operation, it is RECOMMENDED that Beta = 1 -
(P/(1+P))^(EL/TSD) and Zeta = 2*EL/(4+TSD).



3.2.2.6.  Slow start

WEBRC uses a slow start mechanism to quickly ramp up its rate at both
the beginning of the session and in the middle of a session when the
rate drops precipitously.  To enact this, the receiver maintains the
following parameters:

  o SSMINR_P is the minimum allowed slow start threshold rate in packets
    per second.  The recommended value for SSMINR_P is
    BCR_P*(1+1/P+1/P^2).

  o SSR_P is the slow start threshold rate in packets per second.  It is
    adjusted at the beginning of loss events as described in Section
    3.2.3.4. SSR_P is initialized to infinity and is first set to a
    finite value when the receiver leaves the initial start-up period as
    described below.

At the beginning of a session, the receiver cannot compute a meaningful
target rate from its measurements.  Thus, it uses SSR_P = infinity until
one of the following events causes an end to this start-up mode:

  o A packet loss is detected.  In this case the value of SSR_P is
    updated to max{SSMINR_P, P*TRR_P} as with the beginning of any other
    loss event.

  o A sharp increase in MRTT is detected.  While SSR_P = infinity the
    receiver MUST compute, in the notation of Section 3.2.2.2,
    differences in successive measurements of (FirstTime-JoinTime) from



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    successive waves and MUST set SSR_P to max{SSMINR_P, P*TRR_P} when a
    large increase in (FirstTime-JoinTime) is observed.  It is
    RECOMMENDED that an increase in (FirstTime-JoinTime) be considered
    large if it exceeds (1-P^(NWC+1))/(P*log(P)) / ARR_P.

  o The maximum reception rate is reached.  When SSR_P = infinity, if
    (P^(-NWC-2)-1)/(P^(-NWC-1)-1)*ARR_P exceeds MRR_P or SR_P, the
    receiver MUST set SSR_P to max{SSMINR_P, TRR_P}.

  o TRR_P is not increasing consistent with the last join of a wave
    channel.  While SSR_P = infinity, it is RECOMMENDED that the
    receiver wait at least one full epoch after the first packet of a
    wave is received before joining the next wave.  If the TRR_P after
    that full epoch is greatly below ARR_P the receiver SHOULD NOT join
    and SHOULD then set SSR_P to max{SSMINR_P, TRR_P}.  It is
    RECOMMENDED that TRR_P be considered greatly below ARR_P if TRR_P <
    c * ARR_P - 2/EL, where c = Zeta + (1-Zeta)*(P^(-EL/TSD))*(Zeta +
    (1-Zeta)*sqrt(P)*(P^(-EL/TSD)))/g with g =
    (P^(-NWC-1)-1)/(P^(-NWC)-1).

In any of these four cases, the variables associated with LOSSP are
reset to make REQN, calculated as in Section 3.2.2.3 with the current
value of ARTT, equal TRR_P.



3.2.2.7.  Target rate

In typical operation, SSR_P has a finite value and the target rate TRATE
is computed as TRATE = min{max{SSR_P, REQN}, MRR_P}.  When SSR_P =
infinity, TRATE is computed as TRATE = min{4*TRR_P, MRR_P}.



3.2.3.  Receiver events

There are various receiver events, some of which are triggered by the
passing of time on the receiver, and others by events such as packet
reception, detection of packet loss, reception of a first packet from a
channel, and exceptional time-outs.



3.2.3.1.  Packet reception

Most packet reception events require the receiver to merely register the
reception for later calculation of RR_P and IRR_P (see Section 3.2.2.5)
and increment W for later calculation of LOSSP (see Section 3.2.2.1).



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Additional actions, described in the following three subsections, are
required if the packet is the first packet received in response to a
join operation, the CTSI of the packet indicates a time slot change, or
the CN and PSN of the packet indicate a packet loss.



3.2.3.2.  First packet after join

When channel i is the most recently joined channel and the Boolean
variable JOINING is true, the reception of a packet with PSN = i is a
special event because it is the first packet received in response to the
most recent join.  MRTT is calculated and ARTT and V are updated as
described in Section 3.2.2.2, and JOINING is set to false.  The first
received packet of the session furthermore necessitates initialization
of ARR_P and TRR_P as described in Section 3.2.2.5.


3.2.3.3.  Time slot change

This is an event that is triggered by the reception of a packet with a
CTSI value that is one larger modulo T than the previous CTSI value.
When a packet with a new CTSI = i is received, a leave is sent for the
lowest layer in the previous time slot, i.e., wave channel i-1 modulo T,
NWC is updated to NWC-1, and ARR_P is updated to ARR_P - P*BCR_P as
described in Section 3.2.2.5. If the channel for which the leave is sent
is also the most recently joined wave channel and JOINING is true, then
JOINING is set to false.

It is possible due to packet reordering for some packets from the
previous time slot to be received after packets from the current time
slot.  It is RECOMMENDED that measures be put into place to handle this
situation appropriately, i.e., to not trigger a time slot change in this
situation.  One simple mechanism for this is as follows: Compute the
difference i-j modulo T, where i is the CTSI of the received packet and
j is the current CTSI of the receiver.  A difference of zero is, of
course, not a time slot change.  In addition, a very large difference,
for example a difference larger than T-Q/2, should also not trigger a
time slot change.



3.2.3.4.  Loss event

Each time the receiver detects a lost packet (based on the sequence
numbers in the packets scoped by the channel number), the receiver
records the start of a new loss event, and sets a Boolean variable
LOSS_EVENT to true that will automatically reset to false after ARTT



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seconds.  All subsequent packet loss for a period of ARTT seconds is
considered as part of the same loss event.  When a start of a loss event
is detected, the value of SSR_P is updated to max{SSMINR_P, P*TRR_P}.

It is RECOMMENDED that the receiver account for simple misordering of
packets without inferring a loss.



3.2.3.5.  Epoch change

This is an event that is triggered by the passage of time at the
receiver, which occurs each EL seconds.  When this happens, TRR_P and
ARR_P are computed as described in Section 3.2.2.5. Immediately after
these updates, a decision is made about whether to join the next higher
layer as described in Section 3.2.3.6.


3.2.3.6.  Join the next higher layer

At the beginning of each epoch, after updating the values of ARR_P and
TRR_P as described in Section 3.2.2.5, the receiver decides whether or
not to join the next higher layer as follows:

  o If the first base channel packet has not yet arrived the receiver
    does not join.

  o If there is a loss event in progress (LOSS_EVENT = true) the
    receiver does not join.

  o If a join of a channel is in progress (JOINING = true), the receiver
    does not join.

  o If NWC = N the receiver does not join.

  o If the receiver is employing the OPTIONAL rule described in Section
    3.2.2.6, SSR_P = infinity, and a full epoch has not passed since the
    first packet arrival on the most recently joined wave channel then
    the receiver does not join.

  o If the receiver is employing the OPTIONAL rule described in Section
    3.2.2.6, SSR_P = infinity, and a full epoch has passed since the
    first packet arrival on the most recently joined wave channel, then
    the receiver checks if TRR_P is greatly below ARR_P as described in
    Section 3.2.2.6. If TRR_P is greatly below ARR_P the receiver does
    not join.





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  o The receiver calculates REQN as described in Section 3.2.2.3.

  o The receiver calculates TRATE as described in Section 3.2.2.7.

  o If the sender is not sending at constant aggregate rate and TRATE <
    ARR_P*((1/P)^{NWC+2}-1)/((1/P)^{NWC+1}-1), the receiver does not
    join.  If the sender is sending at constant aggregate rate and
    neither TRATE >= ARR_P*((1/P)^{NWC+2}-1)/((1/P)^{NWC+1}-1) nor TRATE
    >= SR_P is true, the receiver does not join.

  o If the sender is producing constant aggregate rate and TRATE >=
    SR_P, the receiver joins the next wave channel.  Otherwise if SSR_P
    is finite the receiver MAY apply one additional OPTIONAL check
    before deciding to join.

    It is RECOMMENDED that the receiver not join if the value of RR_P is
    not sufficiently lower than the maximum value of RR_P observed since
    the last join.  It is RECOMMENDED that RR_P is sufficiently low to
    allow a join if RR_P <= max{RRmax-2/EL,P*RRmax}, where RRmax is the
    maximum measured RR_P since the last join.

    If the receiver does not join because RR_P is not sufficiently small
    then a value of LOSSP is calculated so as to make the value of the
    REQN equation given in Section 3.2.2.3 evaluate to
    ARR_P*((1/P)^(NWC+2)-1)/((1/P)^(NWC+1)-1) with respect to the
    current value of ARR_P.  Then, the variables associated with LOSSP
    are reset based on this calculated value of LOSSP as described at
    the end of Section 3.2.2.1.

Suppose the receiver has decided to join and CTSI = i.  The receiver
joins the next higher wave channel, i.e., the wave channel with CN =
i+NWC modulo T, increments NWC by 1, and then updates ARR to
ARR_P*((1/P)^{NWC+1}-1)/((1/P)^NWC-1) as described in Section 3.2.2.5.
The time of the join is recorded for use in updating ARTT as described
in Section 3.2.2.2.


3.2.3.7.  Join timeout

When no packet arrives in response to the join of channel for a long
period of time, the join times out.  The receiver sets JOINING to false,
updates ARR to ARR_P*((1/P)^NWC-1)/((1/P)^{NWC+1}-1), and then
decrements NWC by 1.

The RECOMMENDED threshold for a join timeout is max{2*V/ARTT,10*ARTT}
seconds.





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3.2.3.8.  Exceptional timeouts

These are timeouts when the packet reception behavior is far from what
it should be and these MUST trigger the receiver to leave the session.
Exceptional timeouts include

  o No packets are received for a long period.  A RECOMMENDED threshold
    is max{10,TSD} seconds.

  o There is no change in time slot index for a long period.  A
    RECOMMENDED threshold is max{20,2*TSD} seconds.



4.  Applicability Statement

WEBRC is intended to be a congestion control scheme that can be used in
a complete protocol instantiation that delivers objects and streams
(both reliable content delivery and streaming of multimedia
information).  WEBRC is most applicable for delivery of objects or
streams of substantial length, i.e., objects or streams that range in
length from hundreds of kilobytes to many gigabytes, and whose transfer
time is on the order of tens of seconds or more.


4.1.  Environmental Requirements and Considerations


WEBRC can be used with both multicast and unicast networks.  However,
the scope of this document is limited to multicast.  WEBRC requires
connectivity between a sender and receivers, but does not require
connectivity from receivers to the sender.

WEBRC inherently works with all types of networks, including LANs, WANs,
Intranets, the Internet, asymmetric networks, wireless networks, and
satellite networks.  Thus, the inherent raw scalability of WEBRC is
unlimited.  However, in some network environments varying reception
rates to receivers may not be advantageous.  For example, the network
may have a dedicated fixed amount of bandwidth allocated to the session
and there may be no effective way for receivers to dynamically vary the
set of channels they are joined to, e.g., in a satellite network.

Receivers join and leave channels using the appropriate multicast join
and leave messages.  For IPv4 multicast, IGMP messages are used by
receivers to join and leave channels.  For IPv6, MLDv2 messages are used
by receivers to join and leave channels.  This is the only dependency of
WEBRC on the IP version.




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WEBRC requires receivers to be able to uniquely identify and demultiplex
packets associated with a session in order to effectively perform
congestion control over all packets associated with the session.  How
receivers achieve this is outside the scope of this document.

WEBRC is presumed to be used with an underlying network or transport
service that is a ``best effort'' service that does not guarantee packet
reception, packet reception order, and which does not have any support
for flow or congestion control.  For example, the Any-Source Multicast
(ASM) model of IP multicast as defined in RFC1112 [5] is such a best
effort network service.  While the basic service provided by RFC1112 is
largely scalable, providing congestion control or reliability should be
done carefully to avoid severe scalability limitations, especially in
the presence of heterogeneous sets of receivers.

There are currently two models of multicast delivery, the Any-Source
Multicast (ASM) model as defined in RFC1112 [5] and the Source-Specific
Multicast (SSM) model as defined in [8]. WEBRC works with both multicast
models, but in a slightly different way with somewhat different
environmental concerns.  When using ASM, a sender S sends packets to a
multicast group G, and the WEBRC channel address consists of the pair
(S,G), where S is the IP address of the sender and G is a multicast
group address.  When using SSM, a sender S sends packets to an SSM
channel (S,G), and the WEBRC channel address coincides with the SSM
channel address.

A sender can locally allocate unique SSM channel addresses, and this
makes allocation of channel addresses easy with SSM.  To allocate
channel addresses using ASM, the sender must uniquely chose the ASM
multicast group address across the scope of the group, and this makes
allocation of WEBRC channel addresses more difficult with ASM.  This is
an issue for WEBRC because several channels are used per session.

WEBRC channels and SSM channels coincide, and thus the receiver will
only receive packets sent to the requested WEBRC channel.  With ASM, the
receiver joins a channel by joining a multicast group G, and all packets
sent to G, regardless of the sender, may be received by the receiver.
Thus, SSM has compelling security advantages over ASM for prevention of
denial of service attacks.  In either case, receivers SHOULD use
mechanisms to filter out packets from unwanted sources.

WEBRC assumes that the packet route between the sender and a particular
receiver is the same for all channels associated with a session.  For
SSM this assumption is true because the multicast tree is a shortest
path tree from each receiver to the sender and generally this path
changes infrequently.  For ASM there are some issues that if not
properly considered may invalidate this assumption.  With ASM, the
packet route between the sender and receivers may initially be through



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the Rendezvous Point (RP) and then switch over to the shortest path to
the sender as packets start flowing in a channel.  The first issue is
that the RP may not be the same for all channels associated with a
session, and thus the first packets sent to the channels may follow a
route that depends on the RP of the channel.  This depends on the RP
configuration for the sender.  If the sender registers all channels
associated with the session with the same RP then the assumption is
true, but if the sender registers different channels with different RPs
then the assumption may not be true.  Thus, it is RECOMMENDED that the
sender register all channels associated with a session with the same RP.
Another issue is that when the channel switches over from the RP to the
sender-based tree then the route to the receivers may vary within a
channel.  Furthermore, this may cause either the receipt of duplicate
packets at receivers or loss of packets depending on the smoothness of
the switchover.  Thus, it is RECOMMENDED that the RP be placed as close
as possible to the sender.  The best location for the RP is that it be
the first-hop router closest to the sender, in which case the path to
the sender and the path to the RP is the same for each receiver and the
problems mentioned above are eliminated.  The consequences of this
assumption not being true are that the receiver reaction to congestion
may not be appropriate.  Generally, the WEBRC receiver will act
conservatively and reduce its reception rate too much if this assumption
is not true, but there can be cases where the receivers will act
inappropriately.



5.  Packet Header Fields

Packets sent to a session using WEBRC MUST include Congestion Control
Information fields as specified in this section. This document specifies
short and long formats for the Congestion Control Information, and it is
RECOMMENDED that protocol instantiations use one of these two formats.
Other formats for the Congestion Control Information fields MAY be used
by protocol instantiations, but all protocol instantiations are REQUIRED
to use these fields in a format that is compatible with the
interpretations of these fields.  Thus, if a protocol does use a
different format for the fields in the Congestion Control Information
then it MUST specify the lengths and positions of these fields within
the packet header.

All integer fields are carried in "big-endian" or "network order"
format, that is, most significant byte (octet) first.  All constants,
unless otherwise specified, are expressed in base ten.







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5.1.  Short Format Congestion Control Information

The short format for the Congestion Control Information is shown in Fig.
1.  The total length of the short format is 32-bits.


  0                   1                   2                   3
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |      CTSI     | Channel Number|    Packet Sequence Number     |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


Fig. 1 - Short format for Congestion Control Information


The function of each field in the Congestion Control Information is the
following.


  Current Time Slot Index (CTSI): 8 bits

      CTSI indicates the index of the current time slot.  This must be
      sent in each packet within the session.  The Current Time Slot
      Index increases by one modulo T each TSD seconds at the sender,
      where T is the number of time slots associated with the session
      and TSD is the time slot duration.  Note that T is also the number
      of wave channels associated with the session, and thus T MUST be
      at most 255.


  Channel Number (CN): 8 bits

      CN is the channel number that this packet belongs to.  CN for the
      base channel is T, and the CNs for the wave channels are 0 through
      T-1.  Thus, T+1 channels in total are used, and thus T MUST be at
      most 255.


  Packet Sequence Number (PSN): 16 bits

      The PSN of each packet is scoped by its CN value.  The sequence
      numbers of consecutive packets sent to the base channel are
      numbered consecutively modulo 2^16.  The same sequence of PSNs are
      used for each wave channel in each cycle.  The sequence numbers of
      consecutive packets sent to a wave channel are numbered
      consecutively modulo 2^16 within each cycle, ending with the last
      packet sent to the channel before the channel goes quiescent with



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      PSN = 2^16-1.



5.2.  Long Format Congestion Control Information

The long format for the Congestion Control Information is shown in Fig.
2.  The total length of the long format is 64-bits.


  0                   1                   2                   3
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |             CTSI              |        Channel Number         |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |                     Packet Sequence Number                    |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


Fig. 2 - Long format for Congestion Control Information


The meaning of each field for the long format is the same as for the
short format, the only difference is that each field is twice as long.


  Current Time Slot Index (CTSI): 16 bits

      CTSI indicates the index of the current time slot.  This must be
      sent in each packet within the session.  The Current Time Slot
      Index increases by one modulo T each TSD seconds at the sender,
      where T is the number of time slots associated with the session
      and TSD is the time slot duration.  Note that T is also the number
      of wave channels associated with the session, and thus T MUST be
      at most 65 535.


  Channel Number (CN): 16 bits

      CN is the channel number that this packet belongs to.  CN for the
      base channel is T, and the CNs for the wave channels are 0 through
      T-1.  Thus, T+1 channels in total are used, and thus T MUST be at
      most 65 535.


  Packet Sequence Number (PSN): 32 bits





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      The PSN of each packet is scoped by its CN value.  The sequence
      numbers of consecutive packets sent to the base channel are
      numbered consecutively modulo 2^32.  The same sequence of PSNs are
      used for each wave channel in each cycle.  The sequence numbers of
      consecutive packets sent to a wave channel are numbered
      consecutively modulo 2^32 within each cycle, ending with the last
      packet sent to the channel before the channel goes quiescent with
      PSN = 2^32-1.



6.  Requirements From Other Building Blocks

As described in RFC3048 [19], WEBRC is a building block that is intended
to be used, in conjunction with other building blocks, to help specify a
protocol instantiation.

WEBRC does not provide higher level session support, e.g., how receivers
obtain the necessary session description and how the receivers
demultiplex received packets based on their session.  There is support
provided by other building blocks that can be used in conjunction with
WEBRC to provide some of this support.  For example, LCT [11] can
provide some of the higher level in-band session support that may be
needed by receivers, and the WEBRC Congestion Control Information (CCI)
required in each packet can be carried in the CCI field of the LCT
header [11].

WEBRC does not provide any type of reliability, and in particular does
not provide support for retransmission of loss packets.  Reliability can
be added by independent means, such as by the use of FEC codes as
described in [12] and specified in the FEC building block [13].


7.  Security Considerations

WEBRC can be subject to denial-of-service attacks by attackers that try
to confuse the congestion control mechanism for receivers by injecting
forged packets into the multicast stream.  This attack most adversely
affects network elements and receivers downstream of the attack, and
much less significantly the rest of the network and other receivers.
Because of this and because of the potential attacks due to the use of
FEC described above, it is RECOMMENDED that Reverse Path Forwarding
checks be enabled in all network routers and switches along the path
from the sender to receivers to limit the possibility of a bad agent
injecting forged packets into the multicast tree data path.

It is also RECOMMENDED that packet authentication be used to
authenticate each packet immediately upon receipt before the receiver



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performs any WEBRC actions based upon its receipt.  Unfortunately, there
are currently no practical multicast packet authentication schemes that
offer instant packet authentication upon receipt.  However, TESLA [17]
can be used to authenticate each packet a few seconds after receipt.
Thus, TESLA could be used in conjunction with WEBRC to authenticate
packets and for example terminate the session upon detection of a forged
packet.  However, it is RECOMMENDED that the normal WEBRC receiver
responses to received packets are allowed to occur immediately and are
not delayed by the TESLA authentication process.  This is because the
overall WEBRC performance would be greatly degraded if the receiver
delayed its WEBRC response to packet receipt for several seconds.

A receiver with an incorrect or corrupted implementation of WEBRC may
affect health of the network in the path between the sender and the
receiver, and may also affect the reception rates of other receivers
joined to the session.  It is therefore RECOMMENDED that receivers be
required to identify themselves as legitimate before they receive the
session description needed to join the session.

Another vulnerability of WEBRC is the potential of receivers obtaining
an incorrect session description for the session.  The consequences of
this could be that legitimate receivers with the wrong session
description are unable to correctly receive the session content, or that
receivers inadvertently try to receive at a much higher rate than they
are capable of, thereby disrupting traffic in portions of the network.
To avoid these problems, it is RECOMMENDED that measures be taken to
prevent receivers from accepting incorrect session descriptions, e.g.,
by using source authentication to ensure that receivers only accept
legitimate session descriptions from authorized senders.



8.  IANA Considerations

No information in this specification is subject to IANA registration.



9.  Intellectual Property Issues

The IETF has been notified of intellectual property rights claimed in
regard to some or all of the specification contained in this document.
For more information consult the online list of claimed rights.








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10.  References


[1] S. Bradner, ``The Internet Standards Process -- Revision 3,''
RFC2026, October 1996.

[2] S. Bradner, ``Key words for use in RFCs to Indicate Requirement
Levels,'' RFC2119, March 1997.

[3] J.W. Byers, G. Horn, M. Luby, M. Mitzenmacher, W. Shaver.  ``FLID-
DL: Congestion control for layered multicast,'' IEEE J. on Selected
Areas in Communications, Special Issue on Network Support for Multicast
Communication, Vol. 20, No. 8, October 2002, pp. 1558-1570.

[4] P. Dagum, R. Karp, M. Luby, and S. Ross, ``An optimal algorithm for
Monte Carlo estimation,'' SIAM J. Comput., 29(5):1484-1496, April 2000.

[5] S. Deering, ``Host Extensions for IP Multicasting,'' RFC1112, August
1989.

[6] V. K Goyal, ``On WEBRC Wave Design and Server Implementation,''
Digital Fountain Technical Report no. DF2002-09-001, September 2002,
available at http://www.digitalfountain.com/technology/.

[7] M. Handley, J. Padhye, S. Floyd, and J. Widmer, ``TCP Friendly Rate
Control (TFRC): Protocol Specification,'' Internet Draft draft-ietf-
tsvwg-tfrc-05, October 2002, a work in progress.

[8] H. W. Holbrook, ``A Channel Model for Multicast,'' Ph.D.
Dissertation, Stanford University, Department of Computer Science,
Stanford, California, August 2001.

[9] R. Kermode and L. Vicisano, ``Author Guidelines for Reliable
Multicast Transport (RMT) Building Blocks and Protocol Instantiation
documents'', RFC3269, April 2002.

[10] M. Luby, J. Gemmell, L. Vicisano, L. Rizzo, and J. Crowcroft,
``Asynchronous Layered Coding protocol instantiation,'' Internet Draft
draft-ietf-rmt-pi-alc-08, April 2002, a work in progress.

[11] M. Luby, J. Gemmell, L. Vicisano, L. Rizzo, M. Handley, and J.
Crowcroft, ``Layered Coding Transport building block,'' Internet Draft
draft-ietf-rmt-bb-lct-04.txt, February 2002, a work in progress.

[12] M. Luby, J. Gemmell, L. Vicisano, L. Rizzo, M. Handley, and J.
Crowcroft, ``The Use of Forward Error Correction in Reliable
Multicast,'' Internet Draft draft-ietf-rmt-info-fec-03.txt, September
2002, a work in progress.



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[13] M. Luby, J. Gemmell, L. Vicisano, L. Rizzo, M. Handley, and J.
Crowcroft, ``Forward Error Correction building block,'' Internet Draft
draft-ietf-rmt-bb-fec-07.txt, September 2002, a work in progress.

[14] M. Luby and V. K Goyal, ``Wave and Equation Based Rate Control
Using Multicast Round Trip Time: Extended Report,'' Digital Fountain
Technical Report no. DF2002-07-001, September 2002, available at
http://www.digitalfountain.com/technology/.

[15] M. Luby, V. K Goyal, S. Skaria, and G. B. Horn, ``Wave and Equation
Based Rate Control Using Multicast Round Trip Time,'' Proc. ACM SIGCOMM
2002, Pittsburgh, PA,  August 2002, pp. 191-214.

[16] A. Mankin, A. Romanow, S. Bradner, and V. Paxson, ``IETF Criteria
for Evaluating Reliable Multicast Transport and Application Protocols,''
RFC2357, June 1998.

[17] A. Perrig, R. Canetti, D. Song, and J. D. Tygar, ``Efficient and
Secure Source Authentication for Multicast,'' Network and Distributed
System Security Symposium, NDSS 2001, pp. 35-46, February 2001.

[18] L. Vicisano, L. Rizzo, and J. Crowcroft, "TCP-like Congestion
Control for Layered Multicast Data Transfer", Proc. IEEE Infocom '98,
San Francisco, CA, March-April 1998, pp. 996-1003.

[19] B. Whetten, L. Vicisano, R. Kermode, M. Handley, S. Floyd, and M.
Luby, ``Reliable Multicast Transport Building Blocks for One-to-Many
Bulk-Data Transfer,'' RFC3048, January 2001.



11.  Authors' Addresses

   Michael Luby
   luby@digitalfountain.com
   Digital Fountain
   39141 Civic Center Drive, Suite 300
   Fremont, CA, USA, 94538

   Vivek K Goyal
   vivek@digitalfountain.com
   Digital Fountain
   39141 Civic Center Drive, Suite 300
   Fremont, CA, USA, 94538







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12.  Full Copyright Statement

Copyright (C) The Internet Society (2002).  All Rights Reserved.

This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it or
assist in its implementation may be prepared, copied, published and
distributed, in whole or in part, without restriction of any kind,
provided that the above copyright notice and this paragraph are included
on all such copies and derivative works. However, this document itself
may not be modified in any way, such as by removing the copyright notice
or references to the Internet Society or other Internet organizations,
except as needed for the purpose of developing Internet standards in
which case the procedures for copyrights defined in the Internet
languages other than English.

The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.

This document and the information contained herein is provided on an "AS
IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK
FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT
LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT
INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR
FITNESS FOR A PARTICULAR PURPOSE."


























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