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rfc:rfc4586

Network Working Group C. Burmeister Request for Comments: 4586 R. Hakenberg Category: Informational A. Miyazaki

                                                             Panasonic
                                                                J. Ott
                                     Helsinki University of Technology
                                                               N. Sato
                                                           S. Fukunaga
                                                                   Oki
                                                             July 2006
                      Extended RTP Profile for
    Real-time Transport Control Protocol (RTCP)-Based Feedback:
              Results of the Timing Rule Simulations

Status of This Memo

 This memo provides information for the Internet community.  It does
 not specify an Internet standard of any kind.  Distribution of this
 memo is unlimited.

Copyright Notice

 Copyright (C) The Internet Society (2006).

Abstract

 This document describes the results achieved when simulating the
 timing rules of the Extended RTP Profile for Real-time Transport
 Control Protocol (RTCP)-Based Feedback, denoted AVPF.  Unicast and
 multicast topologies are considered as well as several protocol and
 environment configurations.  The results show that the timing rules
 result in better performance regarding feedback delay and still
 preserve the well-accepted RTP rules regarding allowed bit rates for
 control traffic.

Burmeister, et al. Informational [Page 1] RFC 4586 Timing Rules Simulation Results July 2006

Table of Contents

 1. Introduction ....................................................3
 2. Timing Rules of the Extended RTP Profile for RTCP-Based
    Feedback ........................................................4
 3. Simulation Environment ..........................................5
    3.1. Network Simulator Version 2 ................................5
    3.2. RTP Agent ..................................................5
    3.3. Scenarios ..................................................5
    3.4. Topologies .................................................6
 4. RTCP Bit Rate Measurements ......................................6
    4.1. Unicast ....................................................7
    4.2. Multicast .................................................10
    4.3. Summary of the RTCP Bit Rate Measurements .................10
 5. Feedback Measurements ..........................................11
    5.1. Unicast ...................................................11
    5.2. Multicast .................................................12
         5.2.1. Shared Losses vs. Distributed Losses ...............13
 6. Investigations on "l" ..........................................14
    6.1. Feedback Suppression Performance ..........................16
    6.2. Loss Report Delay .........................................18
    6.3. Summary of "l" Investigations .............................18
 7. Applications Using AVPF ........................................19
    7.1. NEWPRED Implementation in NS2 .............................19
    7.2. Simulation ................................................21
         7.2.1. Simulation A - Constant Packet Loss Rate ...........21
         7.2.2. Simulation B - Packet Loss Due to Congestion .......23
    7.3. Summary of Application Simulations ........................24
 8. Summary ........................................................24
 9. Security Considerations ........................................25
 10. Normative References ..........................................26
 11. Informative References ........................................26

Burmeister, et al. Informational [Page 2] RFC 4586 Timing Rules Simulation Results July 2006

1. Introduction

 The Real-time Transport Protocol (RTP) is widely used for the
 transmission of real-time or near real-time media data over the
 Internet.  While it was originally designed to work well for
 multicast groups in very large scales, its scope is not limited to
 that.  More and more applications use RTP for small multicast groups
 (e.g., video conferences) or even unicast (e.g., IP telephony and
 media streaming applications).
 RTP comes together with its companion protocol Real-time Transport
 Control Protocol (RTCP), which is used to monitor the transmission of
 the media data and provide feedback of the reception quality.
 Furthermore, it can be used for loose session control.  Having the
 scope of large multicast groups in mind, the rules regarding when to
 send feedback were carefully restricted to avoid feedback explosion
 or feedback-related congestion in the network.  RTP and RTCP have
 proven to work well in the Internet, especially in large multicast
 groups, which is shown by their widespread usage today.
 However, the applications that transmit the media data only to small
 multicast groups or unicast may benefit from more frequent feedback.
 The source of the packets may be able to react to changes in the
 reception quality, which may be due to varying network utilization
 (e.g., congestion) or other changes.  Possible reactions include
 transmission rate adaptation according to a congestion control
 algorithm or the invocation of error resilience features for the
 media stream (e.g., retransmissions, reference picture selection,
 NEWPRED, etc.).
 As mentioned before, more frequent feedback may be desirable to
 increase the reception quality, but RTP restricts the use of RTCP
 feedback.  Hence it was decided to create a new extended RTP profile,
 which redefines some of the RTCP timing rules, but keeps most of the
 algorithms for RTP and RTCP, which have proven to work well.  The new
 rules should scale from unicast to multicast, where unicast or small
 multicast applications have the most gain from it.  A detailed
 description of the new profile and its timing rules can be found in
 [1].
 This document investigates the new algorithms by the means of
 simulations.  We show that the new timing rules scale well and behave
 in a network-friendly manner.  Firstly, the key features of the new
 RTP profile that are important for our simulations are roughly
 described in Section 2.  After that, we describe in Section 3 the
 environment that is used to conduct the simulations.  Section 4
 describes simulation results that show the backwards compatibility to
 RTP and that the new profile is network-friendly in terms of used

Burmeister, et al. Informational [Page 3] RFC 4586 Timing Rules Simulation Results July 2006

 bandwidth for RTCP traffic.  In Section 5, we show the benefit that
 applications could get from implementing the new profile.  In Section
 6, we investigated the effect of the parameter "l" (used to calculate
 the T_dither_max value) upon the algorithm performance, and finally,
 in Section 7, we show the performance gain we could get for a special
 application, namely, NEWPRED in [6] and [7].

2. Timing Rules of the Extended RTP Profile for RTCP-Based Feedback

 As said above, RTP restricts the usage of RTCP feedback.  The main
 restrictions on RTCP are as follows:
  1. RTCP messages are sent in compound packets, i.e., every RTCP packet

contains at least one sender report (SR) or receiver report (RR)

   message and a source description (SDES) message.
  1. The RTCP compound packets are sent in time intervals (T_rr), which

are computed as a function of the average packet size, the number

   of senders and receivers in the group, and the session bandwidth
   (5% of the session bandwidth is used for RTCP messages; this
   bandwidth is shared between all session members, where the senders
   may get a larger share than the receivers.)
  1. The average minimum interval between two RTCP packets from the same

source is 5 seconds.

 We see that these rules prevent feedback explosion and scale well to
 large multicast groups.  However, they do not allow timely feedback
 at all.  While the second rule scales also to small groups or unicast
 (in this cases the interval might be as small as a few milliseconds),
 the third rule may prevent the receivers from sending feedback
 timely.
 The timing rules to send RTCP feedback from the new RTP profile [1]
 consist of two key components.  First, the minimum interval of 5
 seconds is abolished.  Second, receivers get one chance during every
 other of their (now quite small) RTCP intervals to send an RTCP
 packet "early", i.e., not according to the calculated interval, but
 virtually immediately.  It is important to note that the RTCP
 interval calculation is still inherited from the original RTP
 specification.
 The specification and all the details of the extended timing rules
 can be found in [1].  Rather than describing the algorithms here, we
 reference the original specification [1].  Therefore, we use also the
 same variable names and abbreviations as in [1].

Burmeister, et al. Informational [Page 4] RFC 4586 Timing Rules Simulation Results July 2006

3. Simulation Environment

 This section describes the simulation testbed that was used for the
 investigations and its key features.  The extensions to the simulator
 that were necessary are roughly described in the following sections.

3.1. Network Simulator Version 2

 The simulations were conducted using the network simulator version 2
 (ns2).  ns2 is an open source project, written in a combination of
 Tool Command Language (TCL) and C++.  The scenarios are set up using
 TCL.  Using the scripts, it is possible to specify the topologies
 (nodes and links, bandwidths, queue sizes, or error rates for links)
 and the parameters of the "agents", i.e., protocol configurations.
 The protocols themselves are implemented in C++ in the agents, which
 are connected to the nodes.  The documentation for ns2 and the newest
 version can be found in [4].

3.2. RTP Agent

 We implemented a new agent, based on RTP/RTCP.  RTP packets are sent
 at a constant packet rate with the correct header sizes.  RTCP
 packets are sent according to the timing rules of [2] and [3], and
 also its algorithms for group membership maintenance are implemented.
 Sender and receiver reports are sent.
 Further, we extended the agent to support the extended profile [1].
 The use of the new timing rules can be turned on and off via
 parameter settings in TCL.

3.3. Scenarios

 The scenarios that are simulated are defined in TCL scripts.  We set
 up several different topologies, ranging from unicast with two
 session members to multicast with up to 25 session members.
 Depending on the sending rates used and the corresponding link
 bandwidths, congestion losses may occur.  In some scenarios, bit
 errors are inserted on certain links.  We simulated groups with
 RTP/AVP agents, RTP/AVPF agents, and mixed groups.
 The feedback messages are generally NACK messages as defined in [1]
 and are triggered by packet loss.

Burmeister, et al. Informational [Page 5] RFC 4586 Timing Rules Simulation Results July 2006

3.4. Topologies

 Mainly, four different topologies are simulated to show the key
 features of the extended profile.  However, for some specific
 simulations we used different topologies.  This is then indicated in
 the description of the simulation results.  The main four topologies
 are named after the number of participating RTP agents, i.e., T-2,
 T-4, T-8, and T-16, where T-2 is a unicast scenario, T-4 contains
 four agents, etc.  Figure 1 below illustrates the main topologies.
                                                 A5
                                   A5            |   A6
                                  /              |  /
                                 /               | /--A7
                                /                |/
                  A2          A2-----A6          A2--A8
                 /           /                  /        A9
                /           /                  /        /
               /           /                  /        /---A10
 A1-----A2   A1-----A3   A1-----A3-----A7   A1------A3<
               \           \                  \        \---A11
                \           \                  \        \
                 \           \                  \        A12
                  A4          A4-----A8          A4--A13
                                                 |\
                                                 | \--A14
                                                 |  \
                                                 |  A15
                                                A16
     T-2         T-4            T-8               T-16
                    Figure 1: Simulated topologies

4. RTCP Bit Rate Measurements

 The new timing rules allow more frequent RTCP feedback for small
 multicast groups.  In large groups, the algorithm behaves similarly
 to the normal RTCP timing rules.  While it is generally good to
 have more frequent feedback, it cannot be allowed at all to
 increase the bit rate used for RTCP above a fixed limit, i.e., 5%
 of the total RTP bandwidth according to RTP.  This section shows
 that the new timing rules keep RTCP bandwidth usage under the 5%
 limit for all investigated scenarios, topologies, and group sizes.
 Furthermore, we show that mixed groups (some members using
 AVP, some AVPF) can be allowed and that each session member behaves

Burmeister, et al. Informational [Page 6] RFC 4586 Timing Rules Simulation Results July 2006

 fairly according to its corresponding specification.  Note that
 other values for the RTCP bandwidth limit may be specified using
 the RTCP bandwidth modifiers as in [10].

4.1. Unicast

 First we measured the RTCP bandwidth share in the unicast topology
 T-2.  Even for a fixed topology and group size, there are several
 protocol parameters that are varied to simulate a large range of
 different scenarios.  We varied the configurations of the agents
 in the sense that the agents may use AVP or AVPF.  Thereby it
 is possible that one agent uses AVP and the other AVPF in one RTP
 session.  This is done to test the backwards compatibility of the
 AVPF profile.
 Next, we consider scenarios where no losses occur.  In this case,
 both RTP session members transmit the RTCP compound packets at
 regular intervals, calculated as T_rr, if they use AVPF, and
 use a minimum interval of 5 seconds (on average) if they implement
 AVP.  No early packets are sent, because the need to send early
 feedback is not given.  Still it is important to see that not more
 than 5% of the session bandwidth is used for RTCP and that AVP and
 AVPF members can coexist without interference.  The results can
 be found in Table 1.

Burmeister, et al. Informational [Page 7] RFC 4586 Timing Rules Simulation Results July 2006

     |         |      |      |      |      | Used RTCP Bit Rate |
     | Session | Send | Rec. | AVP  | AVPF | (% of session bw)  |
     |Bandwidth|Agents|Agents|Agents|Agents|  A1  |  A2  | sum  |
     +---------+------+------+------+------+------+------+------+
     |  2 Mbps |  1   |  2   |  -   | 1,2  | 2.42 | 2.56 | 4.98 |
     |  2 Mbps | 1,2  |  -   |  -   | 1,2  | 2.49 | 2.49 | 4.98 |
     |  2 Mbps |  1   |  2   |  1   |  2   | 0.01 | 2.49 | 2.50 |
     |  2 Mbps | 1,2  |  -   |  1   |  2   | 0.01 | 2.48 | 2.49 |
     |  2 Mbps |  1   |  2   | 1,2  |  -   | 0.01 | 0.01 | 0.02 |
     |  2 Mbps | 1,2  |  -   | 1,2  |  -   | 0.01 | 0.01 | 0.02 |
     |200 kbps |  1   |  2   |  -   | 1,2  | 2.42 | 2.56 | 4.98 |
     |200 kbps | 1,2  |  -   |  -   | 1,2  | 2.49 | 2.49 | 4.98 |
     |200 kbps |  1   |  2   |  1   |  2   | 0.06 | 2.49 | 2.55 |
     |200 kbps | 1,2  |  -   |  1   |  2   | 0.08 | 2.50 | 2.58 |
     |200 kbps |  1   |  2   | 1,2  |  -   | 0.06 | 0.06 | 0.12 |
     |200 kbps | 1,2  |  -   | 1,2  |  -   | 0.08 | 0.08 | 0.16 |
     | 20 kbps |  1   |  2   |  -   | 1,2  | 2.44 | 2.54 | 4.98 |
     | 20 kbps | 1,2  |  -   |  -   | 1,2  | 2.50 | 2.51 | 5.01 |
     | 20 kbps |  1   |  2   |  1   |  2   | 0.58 | 2.48 | 3.06 |
     | 20 kbps | 1,2  |  -   |  1   |  2   | 0.77 | 2.51 | 3.28 |
     | 20 kbps |  1   |  2   | 1,2  |  -   | 0.58 | 0.61 | 1.19 |
     | 20 kbps | 1,2  |  -   | 1,2  |  -   | 0.77 | 0.79 | 1.58 |
           Table 1: Unicast simulations without packet loss
 We can see that in configurations where both agents use the new
 timing rules each of them uses, at most, about 2.5% of the session
 bandwidth for RTP, which sums up to 5% of the session bandwidth for
 both.  This is achieved regardless of the agent being a sender or a
 receiver.  In the cases where agent A1 uses AVP and agent A2 AVPF,
 the total RTCP session bandwidth decreases.  This is because agent A1
 can send RTCP packets only with an average minimum interval of 5
 seconds.  Thus, only a small fraction of the session bandwidth is
 used for its RTCP packets.  For a high-bit-rate session (session
 bandwidth = 2 Mbps), the fraction of the RTCP packets from agent A1
 is as small as 0.01%.  For smaller session bandwidths, the fraction
 increases because the same amount of RTCP data is sent.  The
 bandwidth share that is used by RTCP packets from agent A2 is not
 different from what was used, when both agents implemented the AVPF.
 Thus, the interaction of AVP and AVPF agents is not problematic in
 these scenarios at all.
 In our second unicast experiment, we show that the allowed RTCP
 bandwidth share is not exceeded, even if packet loss occurs.  We
 simulated a constant byte error rate (BYER) on the link.  The byte
 errors are inserted randomly according to a uniform distribution.

Burmeister, et al. Informational [Page 8] RFC 4586 Timing Rules Simulation Results July 2006

 Packets with byte errors are discarded on the link; hence the
 receiving agents will not see the loss immediately.  The agents
 detect packet loss by a gap in the sequence number.
 When an AVPF agent detects a packet loss, the early feedback
 procedure is started.  As described in AVPF [1], in unicast
 T_dither_max is always zero, hence an early packet can be sent
 immediately if allow_early is true.  If the last packet was already
 an early one (i.e., allow_early = false), the feedback might be
 appended to the next regularly scheduled receiver report.  The
 max_feedback_delay parameter (which we set to 1 second in our
 simulations) determines if that is allowed.
 The results are shown in Table 2, where we can see that there is no
 difference in the RTCP bandwidth share, whether or not losses occur.
 This is what we expected, because even though the RTCP packet size
 grows and early packets are sent, the interval between the packets
 increases and thus the RTCP bandwidth stays the same.  Only the RTCP
 bandwidth of the agents that use the AVP increases slightly.  This is
 because the interval between the packets is still 5 seconds (in
 average), but the packet size increased because of the feedback that
 is appended.
     |         |      |      |      |      | Used RTCP Bit Rate |
     | Session | Send | Rec. | AVP  | AVPF | (% of session bw)  |
     |Bandwidth|Agents|Agents|Agents|Agents|  A1  |  A2  | sum  |
     +---------+------+------+------+------+------+------+------+
     |  2 Mbps |  1   |  2   |  -   | 1,2  | 2.42 | 2.56 | 4.98 |
     |  2 Mbps | 1,2  |  -   |  -   | 1,2  | 2.49 | 2.49 | 4.98 |
     |  2 Mbps |  1   |  2   |  1   |  2   | 0.01 | 2.49 | 2.50 |
     |  2 Mbps | 1,2  |  -   |  1   |  2   | 0.01 | 2.48 | 2.49 |
     |  2 Mbps |  1   |  2   | 1,2  |  -   | 0.01 | 0.02 | 0.03 |
     |  2 Mbps | 1,2  |  -   | 1,2  |  -   | 0.01 | 0.01 | 0.02 |
     |200 kbps |  1   |  2   |  -   | 1,2  | 2.42 | 2.56 | 4.98 |
     |200 kbps | 1,2  |  -   |  -   | 1,2  | 2.50 | 2.49 | 4.99 |
     |200 kbps |  1   |  2   |  1   |  2   | 0.06 | 2.50 | 2.56 |
     |200 kbps | 1,2  |  -   |  1   |  2   | 0.08 | 2.49 | 2.57 |
     |200 kbps |  1   |  2   | 1,2  |  -   | 0.06 | 0.07 | 0.13 |
     |200 kbps | 1,2  |  -   | 1,2  |  -   | 0.09 | 0.08 | 0.17 |
     | 20 kbps |  1   |  2   |  -   | 1,2  | 2.42 | 2.57 | 4.99 |
     | 20 kbps | 1,2  |  -   |  -   | 1,2  | 2.52 | 2.51 | 5.03 |
     | 20 kbps |  1   |  2   |  1   |  2   | 0.58 | 2.54 | 3.12 |
     | 20 kbps | 1,2  |  -   |  1   |  2   | 0.83 | 2.43 | 3.26 |
     | 20 kbps |  1   |  2   | 1,2  |  -   | 0.58 | 0.73 | 1.31 |
     | 20 kbps | 1,2  |  -   | 1,2  |  -   | 0.86 | 0.84 | 1.70 |
             Table 2: Unicast simulations with packet loss

Burmeister, et al. Informational [Page 9] RFC 4586 Timing Rules Simulation Results July 2006

4.2. Multicast

 Next, we investigated the RTCP bandwidth share in multicast
 scenarios; i.e., we simulated the topologies T-4, T-8, and T-16 and
 measured the fraction of the session bandwidth that was used for RTCP
 packets.  Again we considered different situations and protocol
 configurations (e.g., with or without bit errors, groups with AVP
 and/or AVPF agents, etc.).  For reasons of readability, we present
 only selected results.  For a documentation of all results, see [5].
 The simulations of the different topologies in scenarios where no
 losses occur (neither through bit errors nor through congestion) show
 a similar behavior as in the unicast case.  For all group sizes, the
 maximum RTCP bit rate share used is 5.06% of the session bandwidth in
 a simulation of 16 session members in a low-bit-rate scenario
 (session bandwidth = 20 kbps) with several senders.  In all other
 scenarios without losses, the RTCP bit rate share used is below that.
 Thus, the requirement that not more than 5% of the session bit rate
 should be used for RTCP is fulfilled with reasonable accuracy.
 Simulations where bit errors are randomly inserted in RTP and RTCP
 packets and the corrupted packets are discarded give the same
 results.  The 5% rule is kept (at maximum 5.07% of the session
 bandwidth is used for RTCP).
 Finally, we conducted simulations where we reduced the link bandwidth
 and thereby caused congestion-related losses.  These simulations are
 different from the previous bit error simulations, in that the losses
 occur more in bursts and are more correlated, also between different
 agents.  The correlation and "burstiness" of the packet loss is due
 to the queuing discipline in the routers we simulated; we used simple
 FIFO queues with a drop-tail strategy to handle congestion.  Random
 Early Detection (RED) queues may enhance the performance, because the
 burstiness of the packet loss might be reduced; however, this is not
 the subject of our investigations, but is left for future study.  The
 delay between the agents, which also influences RTP and RTCP packets,
 is much more variable because of the added queuing delay.  Still the
 RTCP bit rate share used does not increase beyond 5.09% of the
 session bandwidth.  Thus, also for these special cases the
 requirement is fulfilled.

4.3. Summary of the RTCP Bit Rate Measurements

 We have shown that for unicast and reasonable multicast scenarios,
 feedback implosion does not happen.  The requirement that at maximum
 5% of the session bandwidth is used for RTCP is fulfilled for all
 investigated scenarios.

Burmeister, et al. Informational [Page 10] RFC 4586 Timing Rules Simulation Results July 2006

5. Feedback Measurements

 In this section we describe the results of feedback delay
 measurements, which we conducted in the simulations.  Therefore, we
 use two metrics for measuring the performance of the algorithms;
 these are the "mean waiting time" (MWT) and the number of feedback
 packets that are sent, suppressed, or not allowed.  The waiting time
 is the time, measured at a certain agent, between the detection of a
 packet loss event and the time when the corresponding feedback is
 sent.  Assuming that the value of the feedback decreases with its
 delay, we think that the mean waiting time is a good metric to
 measure the performance gain we could get by using AVPF instead of
 AVP.
 The feedback an RTP/AVPF agent wants to send can be either sent or
 not sent.  If it was not sent, this could be due to feedback
 suppression (i.e., another receiver already sent the same feedback)
 or because the feedback was not allowed (i.e., the max_feedback_delay
 was exceeded).  We traced for every detected loss, if the agent sent
 the corresponding feedback or not and if not, why.  The more feedback
 was not allowed, the worse the performance of the algorithm.
 Together with the waiting times, this gives us a good hint of the
 overall performance of the scheme.

5.1. Unicast

 In the unicast case, the maximum dithering interval T_dither_max is
 fixed and set to zero.  This is because it does not make sense for a
 unicast receiver to wait for other receivers if they have the same
 feedback to send.  But still feedback can be delayed or might not be
 permitted to be sent at all.  The regularly scheduled packets are
 spaced according to T_rr, which depends in the unicast case mainly on
 the session bandwidth.
 Table 3 shows the mean waiting times (MWTs) measured in seconds for
 some configurations of the unicast topology T-2.  The number of
 feedback packets that are sent or discarded is listed also (feedback
 sent (sent) or feedback discarded (disc)).  We do not list suppressed
 packets, because for the unicast case feedback suppression does not
 apply.  In the simulations, agent A1 was a sender and agent A2 was a
 pure receiver.

Burmeister, et al. Informational [Page 11] RFC 4586 Timing Rules Simulation Results July 2006

     |         |       |          Feedback Statistics          |
     | Session |       |       AVP         |       AVPF        |
     |Bandwidth|  PLR  | sent |disc| MWT   | sent |disc| MWT   |
     +---------+-------+------+----+-------+------+----+-------+
     |  2 Mbps | 0.001 |  781 |  0 | 2.604 |  756 |  0 | 0.015 |
     |  2 Mbps | 0.01  | 7480 |  0 | 2.591 | 7548 |  2 | 0.006 |
     |  2 Mbps | cong. |   25 |  0 | 2.557 | 1741 |  0 | 0.001 |
     | 20 kbps | 0.001 |   79 |  0 | 2.472 |   74 |  2 | 0.034 |
     | 20 kbps | 0.01  |  780 |  0 | 2.605 |  709 | 64 | 0.163 |
     | 20 kbps | cong. |  780 |  0 | 2.590 |  687 | 70 | 0.162 |
       Table 3: Feedback statistics for the unicast simulations
 From the table above we see that the mean waiting time can be
 decreased dramatically by using AVPF instead of AVP.  While the
 waiting times for agents using AVP is always around 2.5 seconds (half
 the minimum interval average), it can be decreased to a few ms for
 most of the AVPF configurations.
 In the configurations with high session bandwidth, normally all
 triggered feedback is sent.  This is because more RTCP bandwidth is
 available.  There are only very few exceptions, which are probably
 due to more than one packet loss within one RTCP interval, where the
 first loss was by chance sent quite early.  In this case, it might be
 possible that the second feedback is triggered after the early packet
 was sent, but possibly too early to append it to the next regularly
 scheduled report, because of the limitation of the
 max_feedback_delay.  This is different for the cases with a small
 session bandwidth, where the RTCP bandwidth share is quite low and
 T_rr thus larger.  After an early packet was sent, the time to the
 next regularly scheduled packet can be very high.  We saw that in
 some cases the time was larger than the max_feedback_delay, and in
 these cases the feedback is not allowed to be sent at all.
 With a different setting of max_feedback_delay, it is possible to
 have either more feedback that is not allowed and a decreased mean
 waiting time or more feedback that is sent but an increased waiting
 time.  Thus, the parameter should be set with care according to the
 application's needs.

5.2. Multicast

 In this section, we describe some measurements of feedback statistics
 in the multicast simulations.  We picked out certain characteristic
 and representative results.  We considered the topology T-16.
 Different scenarios and applications are simulated for this topology.
 The parameters of the different links are set as follows.  The agents
 A2, A3, and A4 are connected to the middle node of the multicast

Burmeister, et al. Informational [Page 12] RFC 4586 Timing Rules Simulation Results July 2006

 tree, i.e., agent A1, via high bandwidth and low-delay links.  The
 other agents are connected to the nodes 2, 3, and 4 via different
 link characteristics.  The agents connected to node 2 represent
 mobile users.  They suffer in certain configurations from a certain
 byte error rate on their access links and the delays are high.  The
 agents that are connected to node 3 have low-bandwidth access links,
 but do not suffer from bit errors.  The last agents, which are
 connected to node 4, have high bandwidth and low delay.

5.2.1. Shared Losses vs. Distributed Losses

 In our first investigation, we wanted to see the effect of the loss
 characteristic on the algorithm's performance.  We investigate the
 cases where packet loss occurs for several users simultaneously
 (shared losses) or totally independently (distributed losses).  We
 first define agent A1 to be the sender.  In the case of shared
 losses, we inserted a constant byte error rate on one of the middle
 links, i.e., the link between A1 and A2.  In the case of distributed
 losses, we inserted the same byte error rate on all links downstream
 of A2.
 These scenarios are especially interesting because of the feedback
 suppression algorithm.  When all receivers share the same loss, it is
 only necessary for one of them to send the loss report.  Hence if a
 member receives feedback with the same content that it has scheduled
 to be sent, it suppresses the scheduled feedback.  Of course, this
 suppressed feedback does not contribute to the mean waiting times.
 So we expect reduced waiting times for shared losses, because the
 probability is high that one of the receivers can send the feedback
 more or less immediately.  The results are shown in the following
 table.
     |     |                Feedback Statistics                |
     |     |  Shared Losses          |  Distributed Losses     |
     |Agent|sent|fbsp|disc|sum | MWT |sent|fbsp|disc|sum | MWT |
     +-----+----+----+----+----+-----+----+----+----+----+-----+
     |  A2 | 274| 351|  25| 650|0.267|   -|   -|   -|   -|    -|
     |  A5 | 231| 408|  11| 650|0.243| 619|   2|  32| 653|0.663|
     |  A6 | 234| 407|   9| 650|0.235| 587|   2|  32| 621|0.701|
     |  A7 | 223| 414|  13| 650|0.253| 594|   6|  41| 641|0.658|
     |  A8 | 188| 443|  19| 650|0.235| 596|   1|  32| 629|0.677|
        Table 4: Feedback statistics for multicast simulations
 Table 4 shows the feedback statistics for the simulation of a large
 group size.  All 16 agents of topology T-16 joined the RTP session.
 However, only agent A1 acts as an RTP sender; the other agents are
 pure receivers.  Only 4 or 5 agents suffer from packet loss, i.e.,

Burmeister, et al. Informational [Page 13] RFC 4586 Timing Rules Simulation Results July 2006

 A2, A5, A6, A7, and A8 for the case of shared losses and A5, A6, A7,
 and A8 in the case of distributed losses.  Since the number of
 session members is the same for both cases, T_rr is also the same on
 the average.  Still the mean waiting times are reduced by more than
 50% in the case of shared losses.  This proves our assumption that
 shared losses enhance the performance of the algorithm, regardless of
 the loss characteristic.
 The feedback suppression mechanism seems to be working quite well.
 Even though some feedback is sent from different receivers (i.e.,
 1150 loss reports are sent in total and only 650 packets were lost,
 resulting in loss reports being received on the average 1.8 times),
 most of the redundant feedback was suppressed.  That is, 2023 loss
 reports were suppressed from 3250 individual detected losses, which
 means that more than 60% of the feedback was actually suppressed.

6. Investigations on "l"

 In this section, we want to investigate the effect of the parameter
 "l" on the T_dither_max calculation in RTP/AVPF agents.  We
 investigate the feedback suppression performance as well as the
 report delay for three sample scenarios.
 For all receivers, the T_dither_max value is calculated as
 T_dither_max = l * T_rr, with l = 0.5.  The rationale for this is
 that, in general, if the receiver has no round-trip time (RTT)
 estimation, it does not know how long it should wait for other
 receivers to send feedback.  The feedback suppression algorithm would
 certainly fail if the time selected is too short.  However, the
 waiting time is increased unnecessarily (and thus the value of the
 feedback is decreased) in case the chosen value is too large.
 Ideally, the optimum time value could be found for each case, but
 this is not always feasible.  On the other hand, it is not dangerous
 if the optimum time is not used.  A decreased feedback value and a
 failure of the feedback suppression mechanism do not hurt the network
 stability.  We have shown for the cases of distributed losses that
 the overall bandwidth constraints are kept in any case and thus we
 could only lose some performance by choosing the wrong time value.
 On the other hand, a good measure for T_dither_max is the RTCP
 interval T_rr.  This value increases with the number of session
 members.  Also, we know that we can send feedback at least every
 T_rr.  Thus, increasing T_dither max beyond T_rr would certainly make
 no sense.  So by choosing T_rr/2, we guarantee that at least
 sometimes (i.e., when a loss is detected in the first half of the
 interval between two regularly scheduled RTCP packets) we are allowed
 to send early packets.  Because of the randomness of T_dither, we
 still have a good chance of sending the early packet in time.

Burmeister, et al. Informational [Page 14] RFC 4586 Timing Rules Simulation Results July 2006

 The AVPF profile specifies that the calculation of T_dither_max, as
 given above, is common to session members having an RTT estimation
 and to those not having it.  If this were not so, participants using
 different calculations for T_dither_max might also have very
 different mean waiting times before sending feedback, which
 translates into different reporting priorities.  For example, in a
 scenario where T_rr = 1 s and the RTT = 100 ms, receivers using the
 RTT estimation would, on average, send more feedback than those not
 using it.  This might partially cancel out the feedback suppression
 mechanism and even cause feedback implosion.  Also note that, in a
 general case where the losses are shared, the feedback suppression
 mechanism works if the feedback packets from each receiver have
 enough time to reach each of the other ones before the calculated
 T_dither_max seconds.  Therefore, in scenarios of very high bandwidth
 (small T_rr), the calculated T_dither_max could be much smaller than
 the propagation delay between receivers, which would translate into a
 failure of the feedback suppression mechanism.  In these cases, one
 solution could be to limit the bandwidth available to receivers (see
 [10]) such that this does not happen.  Another solution could be to
 develop a mechanism for feedback suppression based on the RTT
 estimation between senders.  This will not be discussed here and may
 be the subject of another document.  Note, however, that a really
 high bandwidth media stream is not that likely to rely on this kind
 of error repair in the first place.
 In the following, we define three representative sample scenarios.
 We use the topology from the previous section, T-16.  Most of the
 agents contribute only little to the simulations, because we
 introduced an error rate only on the link between the sender A1 and
 the agent A2.
 The first scenario represents those cases, where losses are shared
 between two agents.  One agent is located upstream on the path
 between the other agent and the sender.  Therefore, agent A2 and
 agent A5 see the same losses that are introduced on the link between
 the sender and agent A2.  Agents A6, A7, and A8 do not join the RTP
 session.  From the other agents, only agents A3 and A9 join.  All
 agents are pure receivers, except A1, which is the sender.
 The second scenario also represents cases where losses are shared
 between two agents, but this time the agents are located on different
 branches of the multicast tree.  The delays to the sender are roughly
 of the same magnitude.  Agents A5 and A6 share the same losses.
 Agents A3 and A9 join the RTP session, but are pure receivers and do
 not see any losses.
 Finally, in the third scenario, the losses are shared between two
 agents, A5 and A6.  The same agents as in the second scenario are

Burmeister, et al. Informational [Page 15] RFC 4586 Timing Rules Simulation Results July 2006

 active.  However, the delays of the links are different.  The delay
 of the link between agents A2 and A5 is reduced to 20 ms and between
 A2 and A6 to 40 ms.
 All agents beside agent A1 are pure RTP receivers.  Thus, these
 agents do not have an RTT estimation to the source.  T_dither_max is
 calculated with the above given formula, depending only on T_rr and
 l, which means that all agents should calculate roughly the same
 T_dither_max.

6.1. Feedback Suppression Performance

 The feedback suppression rate for an agent is defined as the ratio of
 the total number of feedback packets not sent out of the total number
 of feedback packets the agent intended to send (i.e., the sum of sent
 and not sent).  The reasons for not sending a packet include: the
 receiver already saw the same loss reported in a receiver report
 coming from another session member or the max_feedback_delay
 (application-specific) was surpassed.
 The results for the feedback suppression rate of the agent Af that is
 further away from the sender are depicted in Table 5.  In general, it
 can be seen that the feedback suppression rate increases as l
 increases.  However there is a threshold, depending on the
 environment, from which the additional gain is not significant
 anymore.
                |      |  Feedback Suppression Rate  |
                |  l   | Scen. 1 | Scen. 2 | Scen. 3 |
                +------+---------+---------+---------+
                | 0.10 |  0.671  |  0.051  |  0.089  |
                | 0.25 |  0.582  |  0.060  |  0.210  |
                | 0.50 |  0.524  |  0.114  |  0.361  |
                | 0.75 |  0.523  |  0.180  |  0.370  |
                | 1.00 |  0.523  |  0.204  |  0.369  |
                | 1.25 |  0.506  |  0.187  |  0.372  |
                | 1.50 |  0.536  |  0.213  |  0.414  |
                | 1.75 |  0.526  |  0.215  |  0.424  |
                | 2.00 |  0.535  |  0.216  |  0.400  |
                | 3.00 |  0.522  |  0.220  |  0.405  |
                | 4.00 |  0.522  |  0.220  |  0.405  |
  Table 5: Fraction of feedback that was suppressed at agent (Af) of
    the total number of feedback messages the agent wanted to send
 Similar results can be seen in Table 6 for the agent An that is
 nearer to the sender.

Burmeister, et al. Informational [Page 16] RFC 4586 Timing Rules Simulation Results July 2006

                |      |  Feedback Suppression Rate  |
                |  l   | Scen. 1 | Scen. 2 | Scen. 3 |
                +------+---------+---------+---------+
                | 0.10 |  0.056  |  0.056  |  0.090  |
                | 0.25 |  0.063  |  0.055  |  0.166  |
                | 0.50 |  0.116  |  0.099  |  0.255  |
                | 0.75 |  0.141  |  0.141  |  0.312  |
                | 1.00 |  0.179  |  0.175  |  0.352  |
                | 1.25 |  0.206  |  0.176  |  0.361  |
                | 1.50 |  0.193  |  0.193  |  0.337  |
                | 1.75 |  0.197  |  0.204  |  0.341  |
                | 2.00 |  0.207  |  0.207  |  0.368  |
                | 3.00 |  0.196  |  0.203  |  0.359  |
                | 4.00 |  0.196  |  0.203  |  0.359  |
  Table 6: Fraction of feedback that was suppressed at agent (An) of
    the total number of feedback messages the agent wanted to send
 The rate of feedback suppression failure is depicted in Table 7.  The
 trend of additional performance increase is not significant beyond a
 certain threshold.  Dependence on the scenario is noticeable here as
 well.
                |      |Feedback Suppr. Failure Rate |
                |  l   | Scen. 1 | Scen. 2 | Scen. 3 |
                +------+---------+---------+---------+
                | 0.10 |  0.273  |  0.893  |  0.822  |
                | 0.25 |  0.355  |  0.885  |  0.624  |
                | 0.50 |  0.364  |  0.787  |  0.385  |
                | 0.75 |  0.334  |  0.679  |  0.318  |
                | 1.00 |  0.298  |  0.621  |  0.279  |
                | 1.25 |  0.289  |  0.637  |  0.267  |
                | 1.50 |  0.274  |  0.595  |  0.249  |
                | 1.75 |  0.274  |  0.580  |  0.235  |
                | 2.00 |  0.258  |  0.577  |  0.233  |
                | 3.00 |  0.282  |  0.577  |  0.236  |
                | 4.00 |  0.282  |  0.577  |  0.236  |
         Table 7: The ratio of feedback suppression failures.
 Summarizing the feedback suppression results, it can be said that in
 general the feedback suppression performance increases as l
 increases.  However, beyond a certain threshold, depending on
 environment parameters such as propagation delays or session
 bandwidth, the additional increase is not significant anymore.  This
 threshold is not uniform across all scenarios; a value of l=0.5 seems
 to produce reasonable results with acceptable (though not optimal)
 overhead.

Burmeister, et al. Informational [Page 17] RFC 4586 Timing Rules Simulation Results July 2006

6.2. Loss Report Delay

 In this section, we show the results for the measured report delay
 during the simulations of the three sample scenarios.  This
 measurement is a metric of the performance of the algorithms, because
 the value of the feedback for the sender typically decreases with the
 delay of its reception.  The loss report delay is measured as the
 time at the sender between sending a packet and receiving the first
 corresponding loss report.
                |      |   Mean Loss Report Delay    |
                |  l   | Scen. 1 | Scen. 2 | Scen. 3 |
                +------+---------+---------+---------+
                | 0.10 |  0.124  |  0.282  |  0.210  |
                | 0.25 |  0.168  |  0.266  |  0.234  |
                | 0.50 |  0.243  |  0.264  |  0.284  |
                | 0.75 |  0.285  |  0.286  |  0.325  |
                | 1.00 |  0.329  |  0.305  |  0.350  |
                | 1.25 |  0.351  |  0.329  |  0.370  |
                | 1.50 |  0.361  |  0.363  |  0.388  |
                | 1.75 |  0.360  |  0.387  |  0.392  |
                | 2.00 |  0.367  |  0.412  |  0.400  |
                | 3.00 |  0.368  |  0.507  |  0.398  |
                | 4.00 |  0.368  |  0.568  |  0.398  |
     Table 8: The mean loss report delay, measured at the sender.
 As can be seen from Table 8, the delay increases, in general, as l
 increases.  Also, a similar effect as for the feedback suppression
 performance is present: beyond a certain threshold, the additional
 increase in delay is not significant anymore.  The threshold is
 environment dependent and seems to be related to the threshold, where
 the feedback suppression gain would not increase anymore.

6.3. Summary of "l" Investigations

 We have shown experimentally that the performance of the feedback
 suppression mechanisms increases as l increases.  The same applies
 for the report delay, which also increases as l increases.  This
 leads to a threshold where both the performance and the delay do not
 increase any further.  The threshold is dependent upon the
 environment.
 So finding an optimum value of l is not possible because it is always
 a trade-off between delay and feedback suppression performance.  With
 l=0.5, we think that a trade-off was found that is acceptable for
 typical applications and environments.

Burmeister, et al. Informational [Page 18] RFC 4586 Timing Rules Simulation Results July 2006

7. Applications Using AVPF

 NEWPRED is one of the error resilience tools, which is defined in
 both ISO/IEC MPEG-4 visual part and ITU-T H.263.  NEWPRED achieves
 fast error recovery using feedback messages.  We simulated the
 behavior of NEWPRED in the network simulator environment as described
 above and measured the waiting time statistics, in order to verify
 that the extended RTP profile for RTCP-based feedback (AVPF) [1] is
 appropriate for the NEWPRED feedback messages.  Simulation results,
 which are presented in the following sections, show that the waiting
 time is small enough to get the expected performance of NEWPRED.

7.1. NEWPRED Implementation in NS2

 The agent that performs the NEWPRED functionality, called NEWPRED
 agent, is different from the RTP agent we described above.  Some of
 the added features and functionalities are described in the following
 points:
 Application Feedback
    The "Application Layer Feedback Messages" format is used to
    transmit the NEWPRED feedback messages.  Thereby the NEWPRED
    functionality is added to the RTP agent.  The NEWPRED agent
    creates one NACK message for each lost segment of a video frame,
    and then assembles multiple NACK messages corresponding to the
    segments in the same video frame into one Application Layer
    Feedback Message.  Although there are two modes, namely, NACK mode
    and ACK mode, in NEWPRED [6][7], only NACK mode is used in these
    simulations.  In this simulation, the RTP layer doesn't generate
    feedback messages.  Instead, the decoder (NEWPRED) generates a
    NACK message when the segment cannot be decoded because the data
    hasn't arrived or loss of reference picture has occurred.  Those
    conditions are detected in the decoder with frame number, segment
    number, and existence of reference pictures in the decoder.
 The parameters of NEWPRED agent are as follows:
      f: Frame Rate(frames/sec)
    seg: Number of segments in one video frame
     bw: RTP session bandwidth(kbps)
 Generation of NEWPRED's NACK Messages
    The NEWPRED agent generates NACK messages when segments are lost.

Burmeister, et al. Informational [Page 19] RFC 4586 Timing Rules Simulation Results July 2006

    a. The NEWPRED agent generates multiple NACK messages per one
       video frame when multiple segments are lost.  These are
       assembled into one Feedback Control Information (FCI) message
       per video frame.  If there is no lost segment, no message is
       generated and sent.
    b. The length of one NACK message is 4 bytes.  Let num be the
       number of NACK messages in one video frame (1 <= num <= seg).
       Thus, 12+4*num bytes is the size of the low-delay RTCP feedback
       message in a compound RTCP packet.
 Measurements
    We defined two values to be measured:
  1. Recovery time

The recovery time is measured as the time between the detection

      of a lost segment and reception of a recovered segment.  We
      measured this "recovery time" for each lost segment.
  1. Waiting time

The waiting time is the additional delay due to the feedback

      limitation of RTP.
 Figure 2 depicts the behavior of a NEWPRED agent when a loss occurs.
 The recovery time is approximated as follows:
    (Recovery time) = (Waiting time) +
                      (Transmission time for feedback message) +
                      (Transmission time for media data)
 Therefore, the waiting time is derived as follows:
    (Waiting time) = (Recovery time) - (Round-trip delay), where
    (Round-trip delay ) = (Transmission time for feedback message) +
                          (Transmission time for media data)

Burmeister, et al. Informational [Page 20] RFC 4586 Timing Rules Simulation Results July 2006

      Picture Reference                            |: Picture Segment
               ____________________                %: Lost Segment
              /_    _    _    _    \
             v/ \  / \  / \  / \    \
             v   \v   \v   \v   \    \
 Sender   ---|----|----|----|----|----|---|------------->
                  \    \                 ^ \
                   \    \               /   \
                    \    \             /     \
                     \    v           /       \
                      \    x         /         \
                       \   Lost     /           \
                        \    x     /             \
 _____
                         v    x   / NACK          v
 Receiver ---------------|----%===-%----%----%----|----->
                              |-a-|               |
                              |-------  b  -------|
                        a: Waiting time
                        b: Recover time (%: Video segments are lost)
 Figure 2: Relation between the measured values at the NEWPRED agent

7.2. Simulation

 We conducted two simulations (Simulation A and Simulation B).  In
 Simulation A, the packets are dropped with a fixed packet loss rate
 on a link between two NEWPRED agents.  In Simulation B, packet loss
 occurs due to congestion from other traffic sources, i.e., ftp
 sessions.

7.2.1. Simulation A - Constant Packet Loss Rate

 The network topology used for this simulation is shown in Figure 3.
                Link 1         Link 2        Link 3
      +--------+      +------+       +------+      +--------+
      | Sender |------|Router|-------|Router|------|Receiver|
      +--------+      +------+       +------+      +--------+
               10(msec)       x(msec)       10(msec)
       Figure 3: Network topology that is used for Simulation A
 Link1 and link3 are error free, and each link delay is 10 msec.
 Packets may get dropped on link2.  The packet loss rates (Plr) and
 link delay (D) are as follows:

Burmeister, et al. Informational [Page 21] RFC 4586 Timing Rules Simulation Results July 2006

    D [ms] = {10, 50, 100, 200, 500}
    Plr    = {0.005, 0.01, 0.02, 0.03, 0.05, 0.1, 0.2}
 Session bandwidth, frame rate, and the number of segments are shown
 in Table 9.
             +------------+----------+-------------+-----+
             |Parameter ID| bw(kbps) |f (frame/sec)| seg |
             +------------+----------+-------------+-----+
             | 32k-4-3    |     32   |      4      |  3  |
             | 32k-5-3    |     32   |      5      |  3  |
             | 64k-5-3    |     64   |      5      |  3  |
             | 64k-10-3   |     64   |     10      |  3  |
             | 128k-10-6  |    128   |     10      |  6  |
             | 128k-15-6  |    128   |     15      |  6  |
             | 384k-15-6  |    384   |     15      |  6  |
             | 384k-30-6  |    384   |     30      |  6  |
             | 512k-30-6  |    512   |     30      |  6  |
             | 1000k-30-9 |   1000   |     30      |  9  |
             | 2000k-30-9 |   2000   |     30      |  9  |
             +------------+----------+-------------+-----+
            Table 9: Parameter sets of the NEWPRED agents
 Figure 4 shows the key values of the result (packet loss rate vs.
 mean of waiting time).
 When the packet loss rate is 5% and the session bandwidth is 32 kbps,
 the waiting time is around 400 msec, which is just allowable for
 reasonable NEWPRED performance.
 When the packet loss rate is less than 1%, the waiting time is less
 than 200 msec.  In such a case, the NEWPRED allows as much as
 200-msec additional link delay.
 When the packet loss rate is less than 5% and the session bandwidth
 is 64 kbps, the waiting time is also less than 200 msec.
 In 128-kbps cases, the result shows that when the packet loss rate is
 20%, the waiting time is around 200 msec.  In cases with more than
 512-kbps session bandwidth, there is no significant delay.  This
 means that the waiting time due to the feedback limitation of RTCP is
 negligible for the NEWPRED performance.

Burmeister, et al. Informational [Page 22] RFC 4586 Timing Rules Simulation Results July 2006

    +------------------------------------------------------------+
    |           | Packet Loss Rate =                             |
    | Bandwidth | 0.005| 0.01 | 0.02 | 0.03 | 0.05 |0.10  |0.20  |
    |-----------+------+------+------+------+------+------+------|
    |       32k |130-  |200-  |230-  |280-  |350-  |470-  |560-  |
    |           |   180|   250|   320|   390|   430|   610|   780|
    |       64k | 80-  |100-  |120-  |150-  |180-  |210-  |290-  |
    |           |   130|   150|   180|   190|   210|   300|   400|
    |      128k | 60-  | 70-  | 90-  |110-  |130-  |170-  |190-  |
    |           |    70|    80|   100|   120|   140|   190|   240|
    |      384k | 30-  | 30-  | 30-  | 40-  | 50-  | 50-  | 50-  |
    |           |    50|    50|    50|    50|    60|    70|    90|
    |      512k | < 50 | < 50 | < 50 | < 50 | < 50 | < 50 | < 60 |
    |           |      |      |      |      |      |      |      |
    |     1000k | < 50 | < 50 | < 50 | < 50 | < 50 | < 50 | < 55 |
    |           |      |      |      |      |      |      |      |
    |     2000k | < 30 | < 30 | < 30 | < 30 | < 30 | < 35 | < 35 |
    +------------------+------+------+------+------+------+------+
                 Figure 4: The result of simulation A

7.2.2. Simulation B - Packet Loss Due to Congestion

 The configurations of link1, link2, and link3 are the same as in
 Simulation A except that link2 is also error-free, regarding bit
 errors.  However, in addition, some FTP agents are deployed to
 overload link2.  See Figure 5 for the simulation topology.
                 Link1         Link2          Link3
      +--------+      +------+       +------+      +--------+
      | Sender |------|Router|-------|Router|------|Receiver|
      +--------+    /|+------+       +------+|\    +--------+
              +---+/ |                       | \+---+
            +-|FTP|+---+                   +---+|FTP|-+
            | +---+|FTP| ...               |FTP|+---+ | ...
            +---+  +---+                   +---+  +---+
             FTP Agents                      FTP Agents
              Figure 5: Network Topology of Simulation B
 The parameters are defined as for Simulation A with the following
 values assigned:
    D[ms] ={10, 50, 100, 200, 500} 32 FTP agents are deployed at each
    edge, for a total of 64 FTP agents active.

Burmeister, et al. Informational [Page 23] RFC 4586 Timing Rules Simulation Results July 2006

 The sets of session bandwidth, frame rate, and the number of segments
 are the same as in Simulation A (Table 9).
 We provide the results for the cases with 64 FTP agents, because
 these are the cases where packet losses could be detected to be
 stable.  The results are similar to those for Simulation A except for
 a constant additional offset of 50..100 ms.  This is due to the delay
 incurred by the routers' buffers.

7.3. Summary of Application Simulations

 We have shown that the limitations of RTP AVPF profile do not
 generate such high delay in the feedback messages that the
 performance of NEWPRED is degraded for sessions from 32 kbps to 2
 Mbps.  We could see that the waiting time increases with a decreasing
 session bandwidth and/or an increasing packet loss rate.  The cause
 of the packet loss is not significant; congestion and constant packet
 loss rates behave similarly.  Still we see that for reasonable
 conditions and parameters the AVPF is well suited to support the
 feedback needed for NEWPRED.  For more information about NEWPRED, see
 [8] and [9].

8. Summary

 The new RTP profile AVPF was investigated regarding performance and
 potential risks to the network stability.  Simulations were conducted
 using the network simulator ns2, simulating unicast and several
 differently sized multicast topologies.  The results were shown in
 this document.
 Regarding the network stability, it was important to show that the
 new profile does not lead to any feedback implosion or use more
 bandwidth than it is allowed.  We measured the bandwidth that was
 used for RTCP in relation to the RTP session bandwidth.  We have
 shown that, more or less exactly, 5% of the session bandwidth is used
 for RTCP, in all considered scenarios.  Other RTCP bandwidth values
 could be set using the RTCP bandwidth modifiers [10].  The scenarios
 included unicast with and without errors, differently sized multicast
 groups, with and without errors or congestion on the links.  Thus, we
 can say that the new profile behaves in a network-friendly manner in
 the sense that it uses only the allowed RTCP bandwidth, as defined by
 RTP.
 Secondly, we have shown that receivers using the new profile
 experience a performance gain.  This was measured by capturing the
 delay that the sender sees for the received feedback.  Using the new
 profile, this delay can be decreased by orders of magnitude.

Burmeister, et al. Informational [Page 24] RFC 4586 Timing Rules Simulation Results July 2006

 In the third place, we investigated the effect of the parameter "l"
 on the new algorithms.  We have shown that there does not exist an
 optimum value for it but only a trade-off can be achieved.  The
 influence of this parameter is highly environment-specific and a
 trade-off between performance of the feedback suppression algorithm
 and the experienced delay has to be met.  The recommended value of
 l=0.5 given in this document seems to be reasonable for most
 applications and environments.

9. Security Considerations

 This document describes the simulation work carried out to verify the
 correct working of the RTCP timing rules specified in the AVPF
 profile [1].  Consequently, security considerations concerning these
 timing rules are described in that document.

Burmeister, et al. Informational [Page 25] RFC 4586 Timing Rules Simulation Results July 2006

10. Normative References

 [1]  Ott, J., Wenger, S., Sato, N., Burmeister, C., and J. Rey,
      "Extended RTP Profile for Real-time Transport Control Protocol
      (RTCP)-Based Feedback (RTP/AVPF)", RFC 4585, July 2006.

11. Informative References

 [2]  Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
      "RTP: A Transport Protocol for Real-Time Applications", STD 64,
      RFC 3550, July 2003.
 [3]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
      Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
 [4]  Network Simulator Version 2 - ns-2, available from
      http://www.isi.edu/nsnam/ns.
 [5]  C. Burmeister, T. Klinner, "Low Delay Feedback RTCP - Timing
      Rules Simulation Results".  Technical Report of the Panasonic
      European Laboratories, September 2001, available from:
      http://www.informatik.uni-bremen.de/~jo/misc/
      SimulationResults-A.pdf.
 [6]  ISO/IEC 14496-2:1999/Amd.1:2000, "Information technology -
      Coding of audio-visual objects - Part2: Visual", July 2000.
 [7]  ITU-T Recommendation, H.263.  Video encoding for low bitrate
      communication.  1998.
 [8]  S. Fukunaga, T. Nakai, and H. Inoue, "Error Resilient Video
      Coding by Dynamic Replacing of Reference Pictures", IEEE Global
      Telecommunications Conference (GLOBECOM), pp.1503-1508, 1996.
 [9]  H. Kimata, Y. Tomita, H. Yamaguchi, S. Ichinose, T. Ichikawa,
      "Receiver-Oriented Real-Time Error Resilient Video Communication
      System: Adaptive Recovery from Error Propagation in Accordance
      with Memory Size at Receiver", Electronics and Communications in
      Japan, Part 1, vol. 84, no. 2, pp.8-17, 2001.
 [10] Casner, S., "Session Description Protocol (SDP) Bandwidth
      Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556,
      July 2003.

Burmeister, et al. Informational [Page 26] RFC 4586 Timing Rules Simulation Results July 2006

Authors' Addresses

 Carsten Burmeister
 Panasonic R&D Center Germany GmbH
 Monzastr. 4c
 D-63225 Langen, Germany
 EMail: carsten.burmeister@eu.panasonic.com
 Rolf Hakenberg
 Panasonic R&D Center Germany GmbH
 Monzastr. 4c
 D-63225 Langen, Germany
 EMail: rolf.hakenberg@eu.panasonic.com
 Akihiro Miyazaki
 Matsushita Electric Industrial Co., Ltd
 1006, Kadoma, Kadoma City, Osaka, Japan
 EMail: miyazaki.akihiro@jp.panasonic.com
 Joerg Ott
 Helsinki University of Technology, Networking Laboratory
 PO Box 3000, 02015 TKK, Finland
 EMail: jo@acm.org
 Noriyuki Sato
 Oki Electric Industry Co., Ltd.
 1-16-8 Chuo, Warabi, Saitama 335-8510 Japan
 EMail: sato652@oki.com
 Shigeru Fukunaga
 Oki Electric Industry Co., Ltd.
 2-5-7 Hommachi, Chuo-ku, Osaka 541-0053 Japan
 EMail: fukunaga444@oki.com

Burmeister, et al. Informational [Page 27] RFC 4586 Timing Rules Simulation Results July 2006

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Burmeister, et al. Informational [Page 28]

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