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rfc:rfc4410

Network Working Group M. Pullen Request for Comments: 4410 F. Zhao Category: Experimental George Mason Univ

                                                              D. Cohen
                                                      Sun Microsystems
                                                         February 2006
           Selectively Reliable Multicast Protocol (SRMP)

Status of This Memo

 This memo defines an Experimental Protocol for the Internet
 community.  It does not specify an Internet standard of any kind.
 Discussion and suggestions for improvement are requested.
 Distribution of this memo is unlimited.

Copyright Notice

 Copyright (C) The Internet Society (2006).

Abstract

 The Selectively Reliable Multicast Protocol (SRMP) is a transport
 protocol, intended to deliver a mix of reliable and best-effort
 messages in an any-to-any multicast environment, where the best-
 effort traffic occurs in significantly greater volume than the
 reliable traffic and therefore can carry sequence numbers of reliable
 messages for loss detection.  SRMP is intended for use in a
 distributed simulation application environment, where only the latest
 value of reliable transmission for any particular data identifier
 requires delivery.  SRMP has two sublayers: a bundling sublayer
 handling message aggregation and congestion control, and a
 Selectively Reliable Transport (SRT) sublayer.  Selection between
 reliable and best-effort messages is performed by the application.

Pullen, et al. Experimental [Page 1] RFC 4410 Selectively Reliable Multicast Protocol February 2006

Table of Contents

 1. Introduction ....................................................3
    1.1. Terminology ................................................3
 2. Protocol Description ............................................4
 3. Message Formats .................................................6
    3.1. Bundle Message Format: .....................................6
    3.2. Bundle Header Format .......................................7
    3.3. Feedback Message Format ....................................9
    3.4. SRT Mode 0 Header Format ..................................10
    3.5. SRT Mode 1 Header Format ..................................11
    3.6. SRT Mode 2 Header Format ..................................11
    3.7. SRT NACK Format ...........................................12
    3.8. User-Configurable Parameters ..............................13
 4. TFMCC Operation ................................................13
    4.1. TCP Rate Prediction Equation for TFMCC ....................13
    4.2. Bundling ..................................................13
    4.3. Congestion Control ........................................14
    4.4. Any-Source Multicast ......................................14
    4.5. Multiple Sources ..........................................14
    4.6. Bundle Size ...............................................15
    4.7. Data Rate Control .........................................15
    4.8. Mode 1 Loss Detection .....................................16
         4.8.1. Sending a Negative Acknowledgement .................16
    4.9. Unbundling ................................................17
    4.10. Heartbeat Bundle .........................................17
 5. SRT Operation ..................................................17
    5.1. Mode 0 Operation ..........................................18
         5.1.1. Sending Mode 0 Messages ............................18
         5.1.2. Receiving Mode 0 Messages ..........................18
    5.2. Mode 1 Operation ..........................................18
         5.2.1. Sending Mode 1 Data Messages .......................19
         5.2.2. Receiving Mode 1 Data Messages .....................19
         5.2.3. Sending a Negative Acknowledgement .................20
         5.2.4. Receiving a Negative Acknowledgement ...............21
    5.3. Mode 2 Operation ..........................................21
         5.3.1. Sending Mode 2 Data Messages .......................21
         5.3.2. Receiving Mode 2 Data Messages .....................22
         5.3.3. Sending a Positive Acknowledgement .................23
         5.3.4. Receiving a Positive Acknowledgement ...............23
 6. RFC 2357 Analysis ..............................................23
    6.1. Scalability ...............................................23
    6.2. Congestion ................................................24
 7. Security Considerations ........................................25
 8. List of Acronyms Used ..........................................26
 9. Contributions ..................................................27
 10. References ....................................................27

Pullen, et al. Experimental [Page 2] RFC 4410 Selectively Reliable Multicast Protocol February 2006

1. Introduction

 There is no viable generic approach to achieving reliable transport
 over multicast networks.  Existing successful approaches require that
 the transport protocol take advantage of special properties of the
 traffic in a way originally proposed by Cohen [10].  The protocol
 described here is applicable to real-time traffic containing a mix of
 two categories of messages: a small fraction requiring reliable
 delivery, mixed with a predominating flow of best-effort messages.
 This sort of traffic is associated with distributed virtual
 simulation (RFC 2502 [4]) and also with some forms of distributed
 multimedia conferencing.  These applications typically have some data
 that changes rarely, or not at all, so the best efficiency will be
 achieved by transmitting that data reliably (the external appearance
 of a simulated vehicle is an excellent example).  They also require
 real-time transmission of a best-effort stream (for example, the
 position and orientation of the vehicle).  There is no value to
 reliable transmission of this stream because typically new updates
 arrive faster than loss identification and retransmission could take
 place.  By piggy-backing the sequence number (SN) of the latest
 reliable transmission on each bundle of traffic, the reliable and
 best-effort traffic can co-exist synergistically.  This approach is
 implemented in the Selectively Reliable Multicast Protocol (SRMP).
 The IETF has conducted a successful working group on Reliable
 Multicast Transport (RMT) that has produced RFCs 2357 [6], 2887 [11],
 and 3450 through 3453 [12 - 15], which define building block
 protocols for reliable multicast.  Selectively reliable multicast is
 similar in spirit to these protocols and in fact uses one of them,
 TCP-Friendly Multicast Congestion Control (TFMCC).  This document
 provides the basis for specifying SRMP with TFMCC for use on an
 experimental basis.  Key requirements of the RMT process that is
 carried forward here are specified in RFC 2357 [6].  These generally
 relate to scalability and congestion control, and are addressed in
 section 6 of this document.

1.1. Terminology

 In this document, the key words "MUST", "MUST NOT", "REQUIRED",
 "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
 and "OPTIONAL" are to be interpreted as described in RFC 2119 [1] and
 indicate requirement levels for compliant implementations.

Pullen, et al. Experimental [Page 3] RFC 4410 Selectively Reliable Multicast Protocol February 2006

2. Protocol Description

 The Selectively Reliable Multicast Protocol (SRMP) has two major
 components: Selectively Reliable Transport (SRT) and a "bundling
 sublayer" that implements TCP-Friendly Multicast Congestion Control
 (TFMCC), as proposed by Widmer and Handley [2], in order to meet the
 requirements of RFC 2357 [6] for congestion avoidance.
 SRMP is capable of reliable message delivery over multicast networks,
 when the messages to be delivered reliably represent a fraction of a
 larger, associated best-effort flow and only the latest reliable
 message must be delivered.  The basic strategy for SRMP is to trade
 as little network capacity as possible for reliability by buffering
 the most recently sent reliable message at each sender and piggy-
 backing its sequence number on associated best-effort messages.  For
 this purpose, three modes of sending are defined:
 o  Mode 0 messages.  These will be delivered best-effort; if lost, no
    retransmission will be done.
 o  Mode 1 messages.  When a Mode 1 message loss is detected, the
    receiver will send back a NACK to the sender, where SRMP will
    retransmit the latest reliable message from that sender.  Senders
    define data identifiers (dataIDs), allowing multiple reliable
    message streams to be supported.  Mode 1 messages may be up to
    131,071 bytes long; SRMP provides for segmentation and reassembly,
    but only for the latest Mode 1 message for any given
    <sourceAddress, multicastAddress, dataID>.
 o  Mode 2 messages.  Through Mode 2 messages, SRMP provides for a
    lightweight, reliable, connectionless peer-to-peer unicast
    transaction exchange between any two members of the multicast
    group.  This is a unicast message requiring positive
    acknowledgement (ACK).
    | Application   |
    -----------------       ----------
    |      SRT      |
    -----------------   ->     SRMP
    |Bundling(TFMCC)|
    -----------------       ----------
    |      UDP      |
 The bundling sublayer is transparent to the Selectively Reliable
 Transport (SRT) sublayer.  It implements congestion control both by
 dropping Mode 0 messages at the source when needed and by bundling
 multiple short messages that are presented by applications within a
 short time window.  It also performs NACK suppression.

Pullen, et al. Experimental [Page 4] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 A bundling sublayer data unit is called a bundle.  A bundle is made
 up of a bundle header and one or more Mode 0 and Mode 1 SRMP
 messages.  Retransmission of Mode 1 messages does not imply
 retransmission of the original bundle; the retransmitted message
 becomes part of a new bundle.
 The TFMCC layer's behavior follows the mechanism described by Widmer
 and Handley.  This is an equation-based multicast congestion control
 mechanism: in a multicast group, each receiver determines its loss
 rate with regard to the sender, and calculates a desired source
 sending rate based on an equation that models the steady-state
 sending rate of TCP.  A distributed feedback suppression mechanism
 restricts feedback to those receivers likely to report the lowest
 desired rates.  Congestion control is achieved by dropping best-
 effort (Mode 0) messages at random.  For example, in distributed
 simulation, Mode 0 messages are part of a stream of state updates for
 dynamic data such as geographic location; therefore, the application
 can continue to function (with lower fidelity) when they are dropped.
 As described by its authors, TFMCC's congestion control mechanism
 works as follows:
 o  Each receiver measures the loss event rate and its Round-Trip Time
    (RTT) to the sender.
 o  Each receiver then uses this information, together with an
    equation for TCP throughput, to derive a TCP-friendly sending
    rate.
 o  Through a distributed feedback suppression mechanism, only a
    subset of the receivers is allowed to give feedback to prevent a
    feedback implosion at the sender.  The feedback mechanism ensures
    that receivers reporting a low desired transmission rate have a
    high probability of sending feedback.
 o  Receivers whose feedback is not suppressed report the calculated
    transmission rate back to the sender in so-called receiver
    reports.  The receiver reports serve two purposes: they inform the
    sender about the appropriate transmit rate, and they allow the
    receivers to measure their RTT.
 o  The sender selects the receiver that reports the lowest rate as
    the current limiting receiver (CLR).  Whenever feedback with an
    even lower rate reaches the sender, the corresponding receiver
    becomes the CLR and the sending rate is reduced to match that
    receiver's calculated rate.  The sending rate increases when the
    CLR reports a calculated rate higher than the current sending
    rate.

Pullen, et al. Experimental [Page 5] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 TFMCC was intended for fixed-size packets with variable rate.  SRMP
 applies it to variable-size SRMP messages that are mostly the same
 size because the best-effort updates typically all represent the same
 sort of simulation information and are grouped into bundles of size
 just under one MTU during periods of heavy network activity.  Future
 developments in TFMCC for variable-size messages will be of high
 value for inclusion in SRMP if, as expected, they prove to be
 appropriate for the types of traffic SRMP is intended to support.
 SRMP is intended for general use under applications that need its
 services and may exist in parallel instances on the same host.  The
 UDP port is therefore established ad hoc from available application
 ports; accordingly, it would not be appropriate to have a well-known
 port for SRMP.

3. Message Formats

3.1. Bundle Message Format:

  1. ——————————————————————-

| bundle header | SRT Message 0 | SRT message 1 | SRT message 2 |…

  1. ——————————————————————-
 A bundle is an aggregation of multiple SRMP messages destined for the
 same multicast address.  A bundle can contain only Mode 0 and Mode 1
 messages; Mode 2 messages are exchanged using unicast addresses.
 SRMP identifies the sender and receiver using their 32-bit Sender_ID,
 which may be an IPv4 address.  For use with IPv6, a user group will
 need to establish a unique identifier per host.  There is no
 requirement for this identifier to be unique in the Internet; it only
 needs to be unique in the communicating group.

Pullen, et al. Experimental [Page 6] RFC 4410 Selectively Reliable Multicast Protocol February 2006

3.2. Bundle Header Format

    0              8              16             24             32
    +--------------+--------------+--------------+--------------+
    |Version| Type |fb_nr | flag  |        bundle_SN            |
    +--------------+--------------+--------------+--------------+
    |                       Sender_ID                           |
    +--------------+--------------+--------------+--------------+
    |                       Receiver_ID                         |
    +--------------+--------------+--------------+--------------+
    |       Sender_Timestamp      |    Receiver_Timestamp       |
    +--------------+--------------+--------------+--------------+
    |            x_supp           |            R_max            |
    +--------------+--------------+--------------+--------------+
    |  DSN_count   |   padding    |           Length            |
    +--------------+--------------+--------------+--------------+
    |     0 to 255 DSN: <dataID, SN, NoSegs> of this sender     |
    +-----------------------------------------------------------+
 Version:
    4 bits   currently 0010
 Type:
    4 bits   0000 - indicates bundle
 fb_nr:
    4 bits   feedback round, range 0-15
 flag:
    4 bits   0001 Is_CLR
             other bits reserved
 bundle_SN:
    16 bits   range 0-65535
 Sender_Timestamp:
    16 bits   Representing the time that the bundle was sent out (in
              milliseconds) based on the sender's local clock.
 Receiver_Timestamp:
    16 bits   Echo of the Receiver_Time_Stamp field (in milliseconds)
              of the receiver feedback message.  If the sender has
              time delay between receiving the feedback and echoing
              the timestamp, it MUST adjust the Receiver_Timestamp
              value to compensate.

Pullen, et al. Experimental [Page 7] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 Receiver_ID
    32 bits   Unique identifier for the receiver within the multicast
              group.  IPv4 addresses may be used.
 Sender_ID:
    32 bits   Unique identifier for the sender within the multicast
              group.  IPv4 addresses may be used.
 X_supp:
    16 bits   The suppression rate corresponding to the sender, in
              bits/s.  Only those receivers whose desired rate is less
              than the suppression rate, or whose RTT is larger than
              R_max, may send feedback information to the sender.  The
              suppression rate is represented as a 16-bit floating
              point value with 8 bits for the unsigned exponent and 8
              bits for the unsigned mantissa.
 R_max:
    16 bits   The maximum of the RTTs of all receivers, in
              milliseconds.  The Maximum RTT should be represented as
              a 16-bit floating point value with 8 bits for the
              unsigned exponent and 8 bits for the unsigned mantissa.
 DSN_count:
    8 bits    The count of DSN blocks following the header.
 Length:
    16 bits   Range from 0~65535.  The total length of the bundle
              in octets (including the header).
 DSN:
    32 bits   There can be up to 256 of these in a header.  An SRMP
              implementation MUST support a minimum of 1.  Each DSN
              consists of three fields:
    dataID:
       16 bits   A unique number associated with a particular data
                 element on the sending host, used to identify a
                 Mode 1 message.
    SN:
       9 bits    Sequence number associated with a particular Mode 1
                 transmission of a particular dataID.
    NoSegs:
       7 bits    Number of segments, if the dataID was long enough
                 to require segmentation; otherwise 0x0.

Pullen, et al. Experimental [Page 8] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 Note that the number of DSNs reflects the number of different Mode 1
 DataIDs being supported at this time by this instance of SRMP, and is
 not the count of SRMP messages bundled in this transmission.
 Note also that 16-bit timestamps will wrap around in 65536
 milliseconds.  This should not be a problem unless an RTT is greater
 than 65 seconds. If a timestamp is less than its predecessor
 (treating the 16 bits as an unsigned integer), its value must be
 increased by 65536 for comparisons against the predecessor.

3.3. Feedback Message Format

    0              8              16             24             32
    +--------------+--------------+--------------+--------------+
    |Version| Type | fb_nr| flag  |             X_r             |
    +--------------+--------------+--------------+--------------+
    |       Sender_Timestamp      |    Receiver_Timestamp       |
    +--------------+--------------+--------------+--------------+
    |                       Sender_ID                           |
    +--------------+--------------+--------------+--------------+
    |                      Receiver_ID                          |
    +--------------+--------------+--------------+--------------+
 Version:
    4 bits   currently 0010
 Type:
    4 bits   value 0001
 fb_nr:
    4 bits   current feedback round of the sender
 flag:
    4 bits
       0001 - have_RTT
       0010 - have_loss
       0100 - receiver_leave
       other values reserved
 X_r:
    16 bits   desired sending rate X_r in bits/s, calculated by the
              receiver to be TCP-friendly, 16 bit floating point
              value with 8 bits for the unsigned exponent and 8 bits
              for the unsigned mantissa.

Pullen, et al. Experimental [Page 9] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 Sender_Timestamp:
    16 bits   Echo of the Sender_Timestamp in bundle header.  If the
              receiver has time delay between receiving the bundle and
              echoing the timestamp, it MUST adjust the
              Sender_Timestamp value correspondently.
 Receiver_Timestamp:
    16 bits   The time when the feedback message was sent out from the
              receiver.
 Receiver_ID:
    32 bits   Unique identifier for the receiver within the multicast
              group.  IPv4 addresses may be used.  (Identifies the
              receiver that sends the feedback message).
 Sender_ID:
    32 bits   Unique identifier for the sender within the multicast
              group.  IPv4 addresses may be used.  (Identifies the
              sender that is the destination of the current feedback
              message.)

3.4. SRT Mode 0 Header Format

    0              8              16             24             32
    +--------------+--------------+--------------+--------------+
    |Version| Type | 000 |  00000000  |        Length           |
    +--------------+--------------+--------------+--------------+
 Version:
    4 bits   currently 0010
 Type:
    4 bits   0000
 Mode:
    3 bits   000
 Padding:
    8 bits   00000000
 Length:
    11 bits  Length of the payload data in octets (does not include
             the header).

Pullen, et al. Experimental [Page 10] RFC 4410 Selectively Reliable Multicast Protocol February 2006

3.5. SRT Mode 1 Header Format

    0              8              16             24             32
    +--------------+--------------+--------------+--------------+
    |Version| Type | 001 |  SegNo    |            Length        |
    +--------------+--------------+--------------+--------------+
    |                            DSN                            |
    +--------------+--------------+--------------+--------------+
 Version:
    4 bits   currently 0010
 Type:
    4 bits   0000
 Mode:
    3 bits   001
 SegNo:
    7 bits   The index number of this segment.
 Length:
    14 bits   Length of the payload data in octets (does not include
              the header).
 DSN:
    32 bits   Same as in the bundle header.  Note that this contains
              NoSegs, whereas SegNo is a separate element.

3.6. SRT Mode 2 Header Format

    0              8              16             24             32
    +--------------+--------------+--------------+--------------+
    |Version| Type |010 |  00000  |            Length           |
    +--------------+--------------+--------------+--------------+
    |                            SN                             |
    +--------------+--------------+--------------+--------------+
 Version:
    4 bits   currently 0010
 Type:
    4 bits   0010
 Mode:
    3 bits   010

Pullen, et al. Experimental [Page 11] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 Padding:
    5 bits   00000
 Length:
    16 bits  Length of the payload data in octets (does not the
             include header).
 SN:
    32 bits   Same as in bundle header.

3.7. SRT NACK Format

    0              8              16             24             32
    +--------------+--------------+--------------+--------------+
    |Version| Type |111 |  00000  |          reserved           |
    +--------------+--------------+--------------+--------------+
    |                            DSN                            |
    +--------------+--------------+--------------+--------------+
    |                      Sender Address                       |
    +--------------+--------------+--------------+--------------+
 Version:
    4 bits   currently 0010
 Type:
    4 bits   0010
 Mode:
    3 bits   111
 Padding:
    5 bits   00000
 Reserved:
    16 bits
 DSN:
    32 bits  sequence number
 Sender Address:
    The IP address of the sender of the message being NACKed.

Pullen, et al. Experimental [Page 12] RFC 4410 Selectively Reliable Multicast Protocol February 2006

3.8. User-Configurable Parameters

 Name                 Minimum Value   Recommended Value       Units
 DSN_Max                 1                 32                messages
 dataID_Timeout         none              none                 ms
 Segment_Timeout         50                250                 ms
 Bundle_Timeout          1                 10                  ms
 Heartbeat_Interval      1                none                 s
 Mode2_Max               1                none               messages
 ACK_Threshold          none         worst RTT in group        ms

4. TFMCC Operation

4.1. TCP Rate Prediction Equation for TFMCC

 The RECOMMENDED throughput equation for SRMP is a slightly simplified
 version of the throughput equation for Reno TCP from [5]:
                                    8*s
    X = ------------------------------------------------------   (1)
          R * (sqrt(2*p/3) + (3*sqrt(6*p) * p * (1+32*p^2)))
 (the formula may be simplified for implementation), where
    X is the transmit rate in bits/second.
    s is the message size in bytes.
    R is the round-trip time in seconds.
    p is the loss event rate, between 0.0 and 1.0, of the number of
      loss events as a fraction of the number of messages transmitted.
 In the future, different TCP formulas may be substituted for this
 equation.  The requirement is that the throughput equation be a
 reasonable approximation of the sending rate of TCP for conformant
 TCP congestion control.

4.2. Bundling

 Multiple SRMP messages will be encapsulated into a bundle.  When a
 new SRMP message (Mode 0 or Mode 1) arrives, the SRMP daemon will try
 to add the new message into the current bundle.
 The SRMP daemon MUST keep a timer, which will be reset when the first
 SRMP message is added into the bundle.  After Bundle_Timeout, the
 timer will time out, and the current bundle should be transmitted

Pullen, et al. Experimental [Page 13] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 immediately.  A new bundle will then be initialized to hold new SRMP
 messages.  Bundle_Timeout SHALL NOT be less than 1 ms.  The
 recommended value is 10 ms.
 Also, the bundle length MUST NOT exceed LENGTH_MAX.  If adding a new
 SRMP message will produce a greater length, the SRMP daemon MUST
 initialize a new bundle for the new SRMP messages, and the current
 bundle should be transmitted immediately.  The recommended value for
 LENGTH_MAX is 1454 bytes (Ethernet MTU minus IP and UDP header
 lengths).
 In a bundle, there may exist multiple SRMP messages with the same
 dataID.  In this case, only the latest version of that dataID is
 useful.  SRMP may check for duplicate dataIDs in the same bundle and
 delete all but the latest one.  If a Mode 1 message appears in the
 outgoing bundle, then the corresponding DSN should not appear in the
 bundle header.
 The bundle header contains the DSN <dataID,SN,NoSegs> for Mode 1
 messages from this sender.  The absolute maximum number of DSN is
 255; however, an implementation may apply a user-specified DSN_Max,
 no smaller than 1.  An implementation may support a user-defined
 dataID_Timeout, after which a given dataID will not be announced in
 the bundle header unless a new Mode 1 message has been sent.  If the
 sender has more dataIDs sent (and not timed out) than will fit in the
 bundle header, the DSNs MUST be announced on a round-robin basis,
 with the exception that no bundle header will announce a DSN for a
 Mode 1 message contained within that bundle.  If a duplicate DSN is
 received, it may be silently discarded.

4.3. Congestion Control

 The congestion control mechanism operates as described in [7].

4.4. Any-Source Multicast

 SRMP uses the Any-Source Multicast Mode.  Each sender will determine
 its maximum RTT, suppression data rate, and sending rate with respect
 to each sender.  Each receiver will measure its RTT and desired rate
 to each sender in the group, and send feedback to every sender by
 sending to the multicast group.

4.5. Multiple Sources

 Under SRMP, each group member in a multicast group is a sender as
 well as a receiver.  Each receiver may need to participate in TFMCC
 information exchange with all senders.  Thus, when a receiver sends a

Pullen, et al. Experimental [Page 14] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 feedback message, it must identify to which source the message should
 be sent using the "Sender ID" field in the header.
 The feedback is multicast to the group.  Depending on the network
 situation, senders may select different receivers to provide
 feedback.  Feedback messages from receivers that are not among those
 selected by the local TFMCC to provide feedback should be silently
 discarded.

4.6. Bundle Size

 TFMCC is designed for traffic with a fixed message size.  The maximum
 bundle size (including header) for SRMP is set to a configurable
 maximum, typically 1454 bytes (Ethernet MTU minus IP and UDP header
 lengths).  The bundle size will be used in a TCP throughput equation,
 to get a desired source rate.  However, in SRMP, the message size is
 variable because:
 1. After bundle time out, the current bundle will not wait for new
    SRMP messages.  This happens with sources sending at a slow rate.
 2. In long messages, there is no further space in the current bundle
    for new SRMP messages.  This will happen with sources sending at a
    high rate or sending messages with a length over half of the
    bundle payload size.
 The case 1 bundle size is likely to be much smaller than that of case
 2.
 Therefore, in SRMP, the mean value of the 10 most recent bundles'
 sizes will be used as the bundle size in the TCP throughput equation.
 This mean value is independent from the network condition and
 reflects current activity of the source.

4.7. Data Rate Control

 Each host will have a single instance of SRMP supporting all of its
 applications.  Thus, the sender's source rate is the sum of the rates
 of all the clients of the same multicast group.
 If the source rate is larger than the sender's desired transmission
 rate, it is the sender's responsibility to do traffic shaping.  Any
 method that conforms to the target sending rate may be used.  The
 RECOMMENDED method is to randomly discard enough Mode 0 messages to
 meet the target rate.

Pullen, et al. Experimental [Page 15] RFC 4410 Selectively Reliable Multicast Protocol February 2006

4.8. Mode 1 Loss Detection

 Bundle header processing includes checking each DSN in the bundle
 header and scheduling a NACK for each DSN bearing a dataID for which
 some application has indicated interest, if the SN/SegNo in that DSN
 indicates that a NACK is needed.  NACKs are sent in bundles and may
 be bundled with data messages.  A NACK is required if:
 o  the SN is one or more greater (mod 512) than the latest received
    Mode 1 message for that dataID, or
 o  the SegNo has not been received, some segment of the <dataID,SN>
    has been received, and a user-defined Segment_Timeout, which SHALL
    NOT be less than 50 ms, has expired since receipt of the first
    SegNo for the <dataID,SN>.
 The bundling sublayer will pass the DSN list in any received bundle
 header to the SRT sublayer.  It also will suppress NACKs in outgoing
 bundles, as described in the next section.

4.8.1. Sending a Negative Acknowledgement

 Negative acknowledgements are used by SRMP for multicast messages in
 order to avoid the congestion of an "ACK implosion" at the original
 sender that would likely occur if positive acknowledgements were used
 instead.  However, with a large multicast group spread out over a
 congested wide-area network, there is the potential for enough
 members of the multicast group to fail to receive the message and
 generate NACKs to cause considerable congestion at the original
 sender despite the use of negative acknowledgements instead of
 positive acknowledgements.  For this reason, SRMP uses a NACK
 suppression mechanism to reduce the number of NACKs generated in
 response to any single lost message.
 The NACK suppression mechanism uses the Bundle_Timeout to distribute
 NACKs over an appropriate time window.  This assumes that the user
 has selected a bundle timeout appropriate for the needs of the
 application for real-time responsiveness.
 When the bundling sublayer is ready to send a bundle, it removes from
 the bundle any NACKs for which a response has been sent by another
 member of the multicast group within the NACK_Repeat_Timeout window.
 If the original Bundle_Timeout has not expired, transmission of the
 bundle may then be delayed until the original Bundle_Timeout expires
 or the bundle is full, whichever happens first.

Pullen, et al. Experimental [Page 16] RFC 4410 Selectively Reliable Multicast Protocol February 2006

4.9. Unbundling

 After a receiver completes congestion control processing on a bundle,
 it parses the bundle into SRT messages and sends these to the SRT
 sublayer.

4.10. Heartbeat Bundle

 SRMP implementations may support a user-defined Heartbeat_Interval,
 which SHALL NOT be less than one second.  At the end of each
 heartbeat interval, if the sender has not sent any bundle, an empty
 bundle will be sent in order to trigger Mode 1 loss detection.

5. SRT Operation

 SRMP operates in three distinct transmission modes in order to
 deliver varying levels of reliability: Mode 0 for multicast data that
 does not require reliable transmission, Mode 1 for data that must be
 received reliably by all members of a multicast group, and Mode 2 for
 data that must be received reliably by a single dynamically
 determined member of a multicast group.
 Mode 0 operates as a pure best-effort service.  Mode 1 operates with
 negative acknowledgements only, triggered by bundle arrivals that
 indicate loss of a Mode 1 message.  Mode 2 uses a positive
 acknowledgement for each message to provide reliability and low
 latency.  Mode 2 is used where a transaction between two members of a
 multicast group is needed.  Because there can be many members in such
 a group, use of a transaction protocol, with reliability achieved by
 SRMP retransmission, avoids the potentially large amount of
 connection setup and associated state that would be required if each
 pair of hosts in the group established a separate TCP connection.
 Use of SRMP anticipates that only a small fraction of messages will
 require reliable multicast, and a comparably small fraction will
 require reliable unicast.  This is due to a property of distributed
 virtual simulation: the preponderance of messages consist of state
 update streams for object attributes such as position and
 orientation.  SRMP is unlikely to provide effective reliable
 multicast if the traffic does not have this property.
 In SRMP, "dataID" is used to associate related messages with each
 other.  Typically, all messages with the same dataID are associated
 with the same application entity.  All the messages with the same
 dataID must be transmitted in the same mode.  Among all the messages
 with the same dataID, the latest version  will obsolete all older
 messages.

Pullen, et al. Experimental [Page 17] RFC 4410 Selectively Reliable Multicast Protocol February 2006

5.1. Mode 0 Operation

 Mode 0 is for multicast messages that do not require reliable
 transmission because they are part of a real-time stream of data that
 is periodically updated with high frequency.  Any such message is
 very likely to have been superceded by a more recent update before
 retransmission could be completed.

5.1.1. Sending Mode 0 Messages

 When an application requests transmission of Mode 0 data, a
 destination multicast group must be provided to SRMP along with the
 data to be sent.  After verifying the data length and multicast
 group, the following steps MUST be performed by the SRT sublayer:
 1. An SRT message MUST be generated with the following
    characteristics:
    the version is set to the current version, the message type is set
    to 0x0, the mode is set to 0x0.  User data is included after the
    message header.  If the message cannot be generated as described
    above, the user data is discarded and the error MUST be reported
    to the application.
 2. If step 1 was completed without error, the newly generated message
    MUST be sent to the bundling sublayer.  The implementation MUST
    report to the application whether the message was ultimately
    accepted by UDP.

5.1.2. Receiving Mode 0 Messages

 When a Mode 0 message is received by SRMP, it MUST be processed as
 follows: after verifying the version, message type, and destination
 multicast address fields, the user data MUST be delivered to all
 applications that are associated with the multicast group in the
 message.  If the SRMP receiver has never received any Mode 1 messages
 before the Mode 0 message is received, the Mode 0 message should be
 silently discarded.
 It is RECOMMENDED that the following information be provided to the
 receiving applications: message body, multicast address.

5.2. Mode 1 Operation

 Mode 1 is for multicast data that requires reliable transmission.  A
 Mode 1 message can be either a data message or a NACK.  Mode 1 data
 messages are expected to be part of a data stream.  This data stream
 is likely to contain Mode 0 messages as well (see section 5.1.1), but

Pullen, et al. Experimental [Page 18] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 it is possible for a data stream to be comprised solely of Mode 1
 messages.

5.2.1. Sending Mode 1 Data Messages

 After the data length, dataID, and destination multicast group are
 verified, SRT MUST take the following steps:
 1. If the message will not fit in an empty bundle with DSN_Max DSN in
    the header, the message MUST be segmented.  The remaining steps
    pertain to each segment of the message.  Each segment receives a
    unique SegNo, starting with 0 and ending with (NoSegs-1).
 2. An SRT message is generated with the following characteristics:
    the version is set to 0x02, the message type is set to 0x0, the
    transmission mode is set to 0x01, the SN is set equal to the SN of
    the most recently sent Mode 1 complete message of the same dataID,
    incremented by 1 modulo 512.  If no such Mode 1 message exists,
    the SN is set to 0x0.
 3. The newly generated message (all segments) must then be buffered,
    replacing any formerly buffered Mode 1 message of the same dataID,
    destination multicast address.  If the message cannot be buffered,
    the user data is discarded and the error is reported to the
    application.
 4. If step 2 was completed without error, the newly generated message
    is sent to the TFMCC sublayer.

5.2.2. Receiving Mode 1 Data Messages

 When a Mode 1 data message is received by SRT, it will be processed
 as follows (assuming that the version field has already been verified
 to be 0x02):
 1. The destination address MUST be verified to be a valid IP
    multicast address on which this instance of SRMP is a member.  If
    this is not the case, the message should be silently discarded.
 2. The destination address MUST be verified to be one for which some
    application has indicated interest.  Otherwise, the message should
    be silently discarded.
 3. The SN, SegNo, source_ip_address, and the body of the received
    message MUST be buffered, and the user data MUST then be delivered
    to all applications that have indicated interest in the multicast
    group of the received message.

Pullen, et al. Experimental [Page 19] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 4. When a new DSN value is received with NoSegs greater than zero, a
    timer should be set for Segment_Timeout, after which a NACK should
    be sent to the bundling sublayer and the timer should be restarted
    for Segment_Timeout.
 5. If NoSegs in the received message is not 0, a reassembly process
    MUST be started.  Each segment MUST be buffered.  If receipt of
    the current message completes the segment, the reassembled message
    MUST be released to the application and the Segment_Timeout timer
    cancelled.
 6. If a new DSN is received before all segments of the previous DSN
    are received, the segments that have been received should be
    dropped silently.
 7. It is RECOMMENDED that the following information be provided to
    the receiving applications: message body, dataID,
    source_ip_address, multicast_group address.
 8. When a client signs on to a new multicast group, all locally
    buffered Mode 1 messages related to that multicast group should be
    delivered to the client immediately.

5.2.3. Sending a Negative Acknowledgement

 Whenever a bundle is received, the bundling sublayer will forward the
 DSN list from the bundle header to the SRT sublayer.  The SRT
 sublayer will examine buffered values of <SenderID,dataID,SN,SegNo>
 to determine whether a NACK is required.  If so, it will generate a
 NACK message and send it to the bundling sublayer.  The NACK message
 will have version set to 0x2, message type set to 0x2, and
 transmission mode set to 0x7.  dataID, SN, and destination address
 are set to that of the Mode 1 message for which the NACK is being
 sent.  If a NACK has been received from any member of the destination
 multicast group for the Mode 1 message in question within the NACK
 threshold, no NACK is generated.
 For segmented messages, there are two possible types of NACKs:
 o  Based on the DSN list in the bundle header, the SRT implementation
    may determine that an entire segmented Mode 1 message was lost.
    In this case, the NACK MUST carry SegNo=0x7F (all in one field).
 o  Based on the Segment Timeout, the SRT implementation may determine
    that one or more segments of a message have not been delivered.
    In this case, a NACK will be sent for each missing segment.

Pullen, et al. Experimental [Page 20] RFC 4410 Selectively Reliable Multicast Protocol February 2006

5.2.4. Receiving a Negative Acknowledgement

 When a NACK is received by SRT, it MUST be processed as follows,
 after verifying the multicast address, dataID, source IP address, and
 transmission mode:
 1. If this instance of SRT's most recent Mode 1 message of the dataID
    indicated in the NACK has an SN newer than the SN in the NACK,
    that message (which is buffered) should be immediately
    retransmitted to the multicast address indicated in the received
    NACK.  If the most recent Mode 1 message has an SN equal to the SN
    indicated in the NACK, and if the SegNo field in the NACK contains
    0x7F, all segments of the buffered Mode 1 message MUST be
    retransmitted; if the SegNo has some other value, only the
    indicated segment should be retransmitted.
 2. Whether or not step 1 results in the retransmission of a message,
    the event of receiving the NACK and the (local machine) time at
    which the NACK was received should be buffered.  Each instance of
    SRT MUST buffer the number of NACKs that have been received for
    each dataID-multicast address pair, since the most recent Mode 1
    message of the same pair was received and the time at which the
    most recent of these NACKs was received.

5.3. Mode 2 Operation

 Mode 2 is for infrequent reliable transaction-oriented communication
 between two dynamically determined members of a multicast group.  TCP
 could be used for such communication, but there would be unnecessary
 overhead and delay in establishing a stream-oriented connection for a
 single exchange of data, whereas there is already an ongoing stream
 of best-effort data between the hosts that require Mode 2
 transmission.  An example is a Distributed Interactive Simulation
 (DIS) collision PDU.

5.3.1. Sending Mode 2 Data Messages

 When an application requests transmission of Mode 2 data, a dataID
 and a destination unicast IP address MUST be provided to SRT along
 with the data to be sent.  After verifying the data length, dataID,
 and destination address, SRT MUST perform the following steps:
 1. An SRT message is generated with the following characteristics:
    the version is set to 0x02, the message type is set to 0x02, the
    transmission mode is set to 0x2, the dataID is set to the
    application-provided value, and the destination address is set to
    the application-provided IP address.  The SN is set equal to the
    SN of the most recently sent Mode 2 message of the same dataID

Pullen, et al. Experimental [Page 21] RFC 4410 Selectively Reliable Multicast Protocol February 2006

    incremented by 1 modulo 65536.  If no such Mode 1 message exists,
    it is set to 0x0.
 2. The newly generated message is buffered.  This new message does
    not replace any formerly buffered Mode 2 messages.  An
    implementation MUST provide a Mode 2 message buffer that can hold
    one or more Mode 2 messages. Mode 2 messages are expected to be
    infrequent (less than 1 percent of total traffic), but it is still
    strongly RECOMMENDED that an implementation provide a buffer of
    user-configurable size Mode2_Max that can hold more than a single
    Mode 2 message.  If the message cannot be buffered, the user data
    is discarded and the error MUST be reported to the application.
    If the message can be buffered, it should be sent to UDP
    immediately after being buffered.
 3. If step 2 was completed without error, the newly generated message
    MUST be sent to the IP address contained in its destination
    address field, encapsulated within a UDP datagram.  If the UDP
    interface on the sending system reports an error to SRT when the
    attempt to send the SRT message is made, an implementation may
    attempt to resend the message any finite number of times.
    However, every implementation MUST provide a mode in which no
    retries are attempted.  Implementations should default to this
    latter mode of operation.  The implementation MUST report to the
    application whether the message was ultimately accepted by UDP.
 4. If some user-configurable "ACK_Threshold" (which should be greater
    than the worst-case round-trip time for the multicast group)
    elapses without receipt of an ACK for the Mode 2 message, it is
    retransmitted.  An implementation may define a maximum number of
    retransmissions to be attempted before the Mode 2 message is
    removed from the buffer.

5.3.2. Receiving Mode 2 Data Messages

 When a Mode 2 data message is received by SRT, it should be processed
 as follows after verifying version, dataID, sender address, and SN:
 1. For Mode 2 messages, the sequence number field is used to
    associate the required positive acknowledgement with a specific
    Mode 2 message.  If the message passes verification, the
    encapsulated user data is delivered to all applications that have
    indicated interest in the dataID and multicast address of the
    received message, regardless of the value of the SN field.
 2. Additionally, an ACK MUST be sent to the host from which the Mode
    2 data message originated.  See section 5.3.3. below for details.

Pullen, et al. Experimental [Page 22] RFC 4410 Selectively Reliable Multicast Protocol February 2006

5.3.3. Sending a Positive Acknowledgement

 A positive acknowledgement (ACK) is triggered by the receipt of a
 Mode 2 data message.  To send an ACK, a new SRT message is generated
 with version set to 0x02, message type set to 0x2, and transmission
 mode set to 0x2.  The dataID and SN are those of the Mode 2 data
 message being acknowledged.  The destination address field is set to
 the source IP address from which the data message was received.
 Since Mode 2 data messages are unicast, there is little concern about
 an ACK implosion causing excessive congestion at the original sender,
 so no suppression mechanism is necessary.

5.3.4. Receiving a Positive Acknowledgement

 When an ACK is received by SRT, after verifying the transmission
 mode, dataID, and source IP address against outstanding Mode 2
 transmission, SRT MUST remove the pending transmission from its
 buffer.

6. RFC 2357 Analysis

 This section provides answers to the questions posed by RFC 2357 for
 reliable multicast protocols, which are quoted.

6.1. Scalability

 "How scalable is the protocol to the number of senders or receivers
 in a group, the number of groups, and wide dispersion of group
 members?"
 SRMP is intended to scale at least to hundreds of group members.  It
 has been designed not to impose limitations on the scalability of the
 underlying multicast network.  No problems have been identified in
 its mechanisms that would preclude this on uncongested networks.
 "Identify the mechanisms which limit scalability and estimate those
 limits."
 There is a practical concern with use of TFMCC, in that the receiver
 with the most congested path constrains delivery to the entire group.
 Distributed virtual simulation requires data delivery at rates
 perceived as continuous by humans.  Therefore, it may prove necessary
 to assign such receivers to different, lower-fidelity groups as a
 practical means of sustaining performance to the majority of
 participating hosts.  SRMP does not have a mechanism to support such
 pruning at this time.

Pullen, et al. Experimental [Page 23] RFC 4410 Selectively Reliable Multicast Protocol February 2006

6.2. Congestion

 "How does the protocol protect the Internet from congestion?  How
 well does it perform?  When does it fail?  Under what circumstances
 will the protocol fail to perform the functions needed by the
 applications it serves?  Is there a congestion control mechanism?
 How well does it perform?  When does it fail?"
 Both simulations and tests indicate that SRMP with TFMCC displays
 backoff comparable to that of TCP under conditions of significant
 packet loss.  The mechanism fails in a network-friendly way, in that
 under severe congestion, it reduces sending of the best-effort
 traffic to a very small rate that typically is unsatisfactory to
 support a virtual simulation.  This is possible because the reliable
 traffic typically is a small percentage of the overall traffic and
 SRMP is NACK oriented, with NACK suppression, so that reliable
 traffic loss adds little traffic to the total.  If the traffic mix
 assumption is not met, the reliable traffic (which does not back off
 under increased RTT) could produce a higher level of traffic than a
 comparable TCP connection.  However, levels of reliable traffic this
 large are not in the intended application domain of SRMP.
 "Include a description of trials and/or simulations which support the
 development of the protocol and the answers to the above questions."
 SRMP has been simulated using a discrete event simulator developed
 for academic use [8].  The design assumptions were validated by the
 results.  It also has been emulated in a LAN-based cluster and
 application-tested in a wide-area testbed under its intended traffic
 mix (distributed virtual simulation) and using a traffic generator
 with losses emulated by random dropping of packets [9].
 "Include an analysis of whether the protocol has congestion avoidance
 mechanisms strong enough to cope with deployment in the Global
 Internet, and if not, clearly document the circumstances in which
 congestion harm can occur.  How are these circumstances to be
 prevented?"
 Because it provides sending backoff comparable to TCP, SRMP is able
 to function as well as TCP for congestion avoidance, even in the
 Global Internet.  The only way an SRMP sender can generate congestion
 is to use the protocol for unintended purposes, for example, reliable
 transmission of a large fraction of the traffic.  Doing this would
 produce unsatisfactory results for the application, as SRMP's
 mechanism for providing reliability will not function well if the
 best-effort traffic does not constitute the majority of the total
 traffic.

Pullen, et al. Experimental [Page 24] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 "Include a description of any mechanisms which contain the traffic
 within limited network environments."
 SRMP has no such mechanisms, as it is intended for use over the open
 Internet.
 "Reliable multicast protocols must include an analysis of how they
 address a number of security and privacy concerns."
 See section 7 below.

7. Security Considerations

 As a transport protocol, SRMP is subject to denial of service by
 hostile third parties sending conflicting values of its parameters on
 the multicast address.  SRMP could attempt to protect itself from
 this sort of behavior.  However, it can be shielded from such attacks
 by traffic authentication at the network layer, as described below.
 A comparable level of authentication also could be obtained by a
 message using MD5, or a similar message hash in each bundle, and
 using the SRMP bundle header to detect duplicate transmissions from a
 given host.  However, this would duplicate the function of existing
 network layer authentication protocols.
 Specific threats that can be eliminated by packet-level
 authentication are as follows:
 a. Amplification attack: SRMP receivers could be manipulated into
    sending large amounts of NACK traffic, which could cause network
    congestion or overwhelm the processing capabilities of a sender.
    This could be done by sending them faked traffic indicating that a
    reliable transmission has been lost.  SRMP's NACK suppression
    limits the effect of such manipulation.  However, true protection
    requires authentication of each bundle.
 b. Denial-of-service attack: If an SRMP sender accepts a large number
    of forged NACKs, it will flood the multicast group with repair
    messages.  This attack also is stopped by per-bundle
    authentication.
 c. Replay attack: The attacker could copy a valid, authenticated
    bundle containing a NACK and send it repeatedly to the original
    sender of the NACKed data.  Protection against this attack
    requires a sequence number per transmission per source host.  The
    SRMP bundle header sequence number would satisfy this need.
    However, the SN also can be applied at a lower layer.

Pullen, et al. Experimental [Page 25] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 d. Reverse path forwarding attack (spoofing): If checks are not
    enabled in all network routers and switches along the path from
    each sender to all receivers, forged packets can be injected into
    the multicast tree data path to manipulate the protocol into
    sending a large volume of repairs.  Packet-level authentication
    can eliminate this possibility.
 e. Inadvertent errors: A receiver with an incorrect or corrupted
    implementation of TFMCC could respond with values of RTT that
    might stimulate a TFMCC sender to create or increase congestion in
    the path to that sender.  It is therefore RECOMMENDED that
    receivers be required to identify themselves as legitimate before
    they receive the Session Description needed to join the session.
    How receivers identify themselves as legitimate is outside the
    scope of this document.
 The required authentication could become part of SRMP or could be
 accomplished by a lower layer protocol.  In any case, it needs to be
 (1) scalable and (2) not very computationally demanding so it can be
 performed with minimal delay on a real-time virtual simulation
 stream.  Public-key encryption meets the first requirement but not
 the second.  Using the IPsec Authentication Header (AH) (RFC 4302
 [3]) meets the second requirement using symmetric-key cryptography.
 See RFC 4302 [3] for guidance on multicast per-packet authentication.
 In practice, users of distributed simulation are likely to work over
 a (possibly virtual) private network and thus will not need special
 authentication for SRMP.

8. List of Acronyms Used

 ACK   - positive acknowledgement
 AH    - Authentication Header
 CLR   - current limiting receiver
 IPSEC - Internet Protocol Security
 MTU   - maximum transmission unit
 NACK  - negative acknowledgement
 RTT   - round-trip time
 SA    - security association
 SRMP  - Selectively Reliable Multicast Protocol
 SRT   - Selectively Reliable Transport
 TFMCC - TCP-Friendly Multicast Congestion Control

Pullen, et al. Experimental [Page 26] RFC 4410 Selectively Reliable Multicast Protocol February 2006

9. Contributions

 We gratefully acknowledge the significant contributions of two
 people without whom this RFC would not have been developed.
 Vincent Laviano created the first specification and implementation
 of SRMP (at that time called SRTP).  Babu Shanmugam employed SRMP
 in a sizable distributed virtual simulation environment, where he
 revised the implementation and helped revise the design to support
 distributed virtual simulation workload effectively.

10. References

10.1. Normative References

 [1]  Bradner, S., "Key words for use in RFCs to Indicate Requirement
      Levels", BCP 14, RFC 2119, March 1997.
 [2]  J. Widmer, M. Handley, Extending Equation-Based Congestion
      Control to Multicast Applications, ACM SIGCOMM Conference, San
      Diego, August 2001.  <http://www.sigcomm.org/sigcomm2001/p22-
      widmer.pdf>
 [3]  Kent, S., "IP Authentication Header", RFC 4302, December 2005.

10.2. Informative References

 [4]  Pullen, M., Myjak, M., and C. Bouwens, "Limitations of Internet
      Protocol Suite for Distributed Simulation the Large Multicast
      Environment", RFC 2502, February 1999.
 [5]  J. Padhye, V. Firoiu, D. Towsley and J. Kurose, "Modeling TCP
      Throughput: A Simple Model and its Empirical Validation",
      Proceedings of ACM SIGCOMM 1998.
 [6]  Mankin, A., Romanow, A., Bradner, S., and V. Paxson, "IETF
      Criteria for Evaluating Reliable Multicast Transport and
      Application Protocols", RFC 2357, June 1998.
 [7]  Floyd, S., "Congestion Control Principles", BCP 41, RFC 2914,
      September 2000.
 [8]  J. M. Pullen, "The Network Workbench: Network Simulation
      Software for Academic Investigation of Internet Concepts,"
      Computer Networks Vol 32 No 3 pp 365-378, March 2000.

Pullen, et al. Experimental [Page 27] RFC 4410 Selectively Reliable Multicast Protocol February 2006

 [9]  J. M. Pullen, R. Simon, F. Zhao and W. Chang, "NGI-FOM over
      RTI-NG and SRMP: Lessons Learned," Proceedings of the IEEE Fall
      Simulation Interoperability Workshop, paper 03F-SIW-111,
      Orlando, FL, September 2003.
 [10] D. Cohen, "NG-DIS-PDU: The Next Generation of DIS-PDU (IEEE-
      P1278)", 10th Workshop on Standards for Interoperability of
      Distributed Simulations, March 1994.
 [11] Handley, M., Floyd, S., Whetten, B., Kermode, R., Vicisano, L.,
      and M. Luby, "The Reliable Multicast Design Space for Bulk Data
      Transfer", RFC 2887, August 2000.
 [12] Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., and J.
      Crowcroft, "Asynchronous Layered Coding (ALC) Protocol
      Instantiation", RFC 3450, December 2002.
 [13] Luby, M., Gemmell, J., Vicisano, L., Rizzo, L., Handley, M., and
      J. Crowcroft, "Layered Coding Transport (LCT) Building Block",
      RFC 3451, December 2002.
 [14] Luby, M., Vicisano, L., Gemmell, J., Rizzo, L., Handley, M., and
      J. Crowcroft, "Forward Error Correction (FEC) Building Block",
      RFC 3452, December 2002.
 [15] Luby, M., Vicisano, L., Gemmell, J., Rizzo, L., Handley, M., and
      J. Crowcroft, "The Use of Forward Error Correction (FEC) in
      Reliable Multicast", RFC 3453, December 2002.

Pullen, et al. Experimental [Page 28] RFC 4410 Selectively Reliable Multicast Protocol February 2006

Authors' Addresses

 J. Mark Pullen
 C4I Center
 George Mason University
 Fairfax, VA 22030
 USA
 EMail: mpullen@gmu.edu
 Fei Zhao
 C4I Center
 George Mason University
 Fairfax, VA 22030
 USA
 EMail: fzhao@netlab.gmu.edu
 Danny Cohen
 Sun Microsystems
 M/S UMPK16-160
 16 Network Circle
 Menlo Park, CA 94025
 USA
 EMail: danny.cohen@sun.com

Pullen, et al. Experimental [Page 29] RFC 4410 Selectively Reliable Multicast Protocol February 2006

Full Copyright Statement

 Copyright (C) The Internet Society (2006).
 This document is subject to the rights, licenses and restrictions
 contained in BCP 78, and except as set forth therein, the authors
 retain all their rights.
 This document and the information contained herein are provided on an
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Acknowledgement

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 Administrative Support Activity (IASA).

Pullen, et al. Experimental [Page 30]

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