Internet DRAFT - draft-burmeister-avt-rtcp-feedback-sim
draft-burmeister-avt-rtcp-feedback-sim
Internet Draft C. Burmeister
draft-burmeister-avt-rtcp-feedback-sim-06.txt R. Hakenberg
Expires: October 2004 A. Miyazaki
Matsushita
J. Ott
University of Bremen TZI
N. Sato
S. Fukunaga
Oki
April 2004
Extended RTP Profile for RTCP-based Feedback
- Results of the Timing Rule Simulations -
Status of this Memo
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Copyright Notice
Copyright (C) The Internet Society (2004). All Rights
Reserved.
Abstract
This document describes the results achieved when simulating the
timing rules of the Extended RTP Profile for RTCP-based Feedback,
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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.
Table of Contents
1 Introduction
2 Timing rules of the extended RTP profile for RTCP-based feedback
3 Simulation Environment
4 RTCP Bit Rate Measurements
5 Feedback Measurements
6 Investigations on "l"
7 Applications Using AVPF
8 Summary
9 Security Considerations
10 Informative References
11 IPR Notices
12 Authors' Address
13 Full Copyright Statement
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 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 its 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
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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 3. After that, we describe the
environment that is used to conduct the simulations in Section 4.
Section 5 describes simulation results that show the backwards
compatibility to RTP and that the new profile is network-friendly
in terms of used bandwidth for RTCP traffic. In Section 6, we
show the benefit that applications could get from implementing the
new profile. In Section 7 we investigated the effect of the
parameter "l" (used to calculate the T_dither_max value) upon the
algorithm performance and finally in Section 8 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:
- 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.
- 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.)
- The average minimum interval between two RTCP packets from the
same source
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is 5 seconds.
We see that these rules prevent feedback explosion and scale well
to large multicast groups. However, they 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 once during their
(now quite small) RTCP interval the chance 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]. We shall describe the algorithms here, but
rather reference these from the original specification where
needed. Therefore we use also the same variable names and
abbreviations as in [1].
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
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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.
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. The figures below illustrate the main
topologies.
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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, i.e. some members using
AVP some AVPF, can be allowed and that each session member behaves
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 which 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 the AVP or AVPF. Thereby it
is possible that one agent uses AVP and the other AVPF in one RTP
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session. This is done to test the backwards compatibility of the
AVPF profile.
First 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 the AVPF, and
use a minimum interval of 5s (in average) if they implement the
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 co-exist without interference. The results can
be found in table 1.
| | | | | | 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 is decreased. This is due
to the fact that 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
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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 an uniform distribution.
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 losses occur or
not. 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 |
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| 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.
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 = 20kbps) 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 research. 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.
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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.
5 Feedback Measurements
In this chapter 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 the 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 due to the fact that 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 (MWT) 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
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suppression does not apply. In the simulations, agent A1 was a
sender and agent A2 a pure receiver.
| | | 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
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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 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, that 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
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agents are pure receivers. Only 4 or 5 agents suffer from packet
loss, i.e. 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 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
however 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 to send the
early packet in time.
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RTP/AVPF Profile Timing Rules Simulation Results April 2004
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 an scenario where T_rr = 1s 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 object 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 represents also 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
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RTP/AVPF Profile Timing Rules Simulation Results April 2004
active. However, the delays of the links are different. The
delay of the link between agent A2 and A5 is reduced to 20ms and
between A2 and A6 to 40ms.
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 10. In
general it can be seen that the feedback suppression rate
increases with an increasing l. 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 10: 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 for the agent that is nearer to the
sender in Table 11.
| | Feedback Suppression Rate |
| l | Scen. 1 | Scen. 2 | Scen. 3 |
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RTP/AVPF Profile Timing Rules Simulation Results April 2004
+------+---------+---------+---------+
| 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 11: 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 12.
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 12: The ratio of feedback suppression failures.
Summarizing the feedback suppression results, it can be said that
in general the feedback suppression performance increases with an
increasing l. 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.
6.2 Loss Report Delay
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RTP/AVPF Profile Timing Rules Simulation Results April 2004
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 13: The mean loss report delay, measured at the sender.
As can be seen from Table 13 the delay increases in general with
an increasing value of l. 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 with an increasing value of l.
The same applies for the report delay, which increases also with
an increasing l. This leads to a threshold where both the
performance and the delay does 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 tradeoff was found that
is acceptable for typical applications and environments.
7 Applications Using AVPF
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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.
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.
a. The NEWPRED agent generates multiple NACK messages per
one video frame when multiple segments are lost. These
are assembled into one 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.
Measurements
We defined two values to be measured:
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RTP/AVPF Profile Timing Rules Simulation Results April 2004
- 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.
- Waiting time
The waiting time is the additional delay due to the feedback
limitation of RTP.
Fig.1 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)
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)
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RTP/AVPF Profile Timing Rules Simulation Results April 2004
Fig.1: 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 Fig.2.
Link 1 Link 2 Link 3
+--------+ +------+ +------+ +--------+
| Sender |------|Router|-------|Router|------|Receiver|
+--------+ +------+ +------+ +--------+
10(msec) x(msec) 10(msec)
Fig2. 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:
D [ms] = {10, 50, 100, 200, 500}
Plr = {0.005, 0.01, 0.02, 0.03, 0.05, 0.1, 0.2}
Session band width, frame rate and the number of segments are
shown in Table 14
+------------+----------+-------------+-----+
|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 |
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RTP/AVPF Profile Timing Rules Simulation Results April 2004
+------------+----------+-------------+-----+
Table 14: Parameter sets of the NEWPRED agents
Figure3 shows the packet loss rate vs. mean of waiting time. A
plotted line represents a parameter ID ( "[session bandwidth] -
[frame rate] - [the number of segments] - [link2 delay]" ). E.g.
384k-15-9-100 means the session of 384kbps session bandwidth, 15
frames per second, 9 segments per frame and 100msec link delay.
When the packet loss rate is 5% and the session bandwidth is
32kbps, the waiting time is around 400msec, which is just
allowable for reasonable NEWPRED performance.
When the packet loss rate is less than 1%, the waiting time is
less than 200msec. In such a case, the NEWPRED allows as much as
200msec additional link delay.
When the packet loss rate is less than 5% and the session
bandwidth is 64kbps, the waiting time is also less than 200msec.
In 128kbps cases, the result shows that when the packet loss rate
is 20%, the waiting time is around 200msec. In cases with more
than 512kbps session bandwidth, there is no significant delay.
This means that the waiting time due to the feedback limitation of
RTCP is neglectable for the NEWPRED performance.
+------------------------------------------------------------+
| | 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 |
+------------------+------+------+------+------+------+------+
Fig. 3 The result of simulation A
7.2.2. Simulation B - Packet Loss due to Congestion
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RTP/AVPF Profile Timing Rules Simulation Results April 2004
The configuration 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 4 for the simulation topology.
Link1 Link2 Link3
+--------+ +------+ +------+ +--------+
| Sender |------|Router|-------|Router|------|Receiver|
+--------+ /|+------+ +------+|\ +--------+
+---+/ | | \+---+
+-|FTP|+---+ +---+|FTP|-+
| +---+|FTP| ... |FTP|+---+ | ...
+---+ +---+ +---+ +---+
FTP Agents FTP Agents
Fig4. 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.
The sets of session bandwidth, frame rate, the number of
segments
are the same as in Simulation A (Table 14)
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 the Simulation A except for a
constant additional offset of 50..100ms. 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 32kbps to
2Mbps. 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
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RTP/AVPF Profile Timing Rules Simulation Results April 2004
that for reasonable conditions and parameters the AVPF is well
suited to support the feedback needed for NEWPRED.
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, 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 as it is allowed. Thus 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,
different sized multicast groups, with and without errors or
congestion on the links. Thus we can say that the new profile
behaves network-friendly 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.
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 the draft 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.
10 Informative References
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RTP/AVPF Profile Timing Rules Simulation Results April 2004
1 J. Ott, S. Wenger, N. Sato, C. Burmeister, and J. Rey, "Extended
RTP Profile for RTCP-based Feedback", Internet Draft, draft-
ietf-avt-rtcp-feedback-07.txt, Work in Progress, June 2003.
2 H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, " RTP
- A Transport Protocol for Real-time Applications, RFC 3550,
July 2003.
3 H. Schulzrinne, S. Casner, "RTP Profile for Audio and Video
Conferences with Minimal Control", 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 S. Casner, "Session Description Protocol (SDP) Bandwidth
Modifiers for RTP Control Protocol (RTCP) Bandwidth", RFC 3556,
July 2003.
11 IPR Notices
The IETF takes no position regarding the validity or scope of any
intellectual property or other rights that might be claimed to
pertain to the implementation or use of the technology described
in this document or the extent to which any license under such
rights might or might not be available; neither does it represent
that it has made any effort to identify any such rights.
Information on the IETF's procedures with respect to rights in
standards-track and standards-related documentation can be found
Burmeister et al. Expires October 2004 24
RTP/AVPF Profile Timing Rules Simulation Results April 2004
in BCP 11 [13]. Copies of claims of rights made available for
publication and any assurances of licenses to be made available,
or the result of an attempt made to obtain a general license or
permission for the use of such proprietary rights by implementers
or users of this specification can be obtained from the IETF
Secretariat.
The IETF invites any interested party to bring to its attention
any copyrights, patents or patent applications, or other
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12 Authors' Address
Carsten Burmeister
Panasonic European Laboratories GmbH
Monzastr. 4c, 63225 Langen, Germany
mailto: burmeister@panasonic.de
Rolf Hakenberg
Panasonic European Laboratories GmbH
Monzastr. 4c, 63225 Langen, Germany
mailto: hakenberg@panasonic.de
Akihiro Miyazaki
Matsushita Electric Industrial Co., Ltd
1006, Kadoma, Kadoma City, Osaka, Japan
mailto: akihiro@isl.mei.co.jp
Joerg Ott
Universitaet Bremen TZI
MZH 5180, Bibliothekstr. 1, 28359 Bremen, Germany
{sip,mailto}: jo@tzi.uni-bremen.de
Noriyuki Sato
Oki Electric Industry Co., Ltd.
1-16-8 Chuo, Warabi, Saitama 335-8510 Japan
mailto: sato652@oki.com
Shigeru Fukunaga
Oki Electric Industry Co., Ltd.
2-5-7 Honmachi, Chuo-ku, Osaka 541-0053 Japan
mailto: fukunaga444@oki.com
13 Full Copyright Statement
"Copyright (C) The Internet Society (2004). All Rights Reserved.
Burmeister et al. Expires October 2004 25
RTP/AVPF Profile Timing Rules Simulation Results April 2004
This document and translations of it may be copied and furnished
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copied, published and distributed, in whole or in part, without
restriction of any kind, provided that the above copyright notice
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The limited permissions granted above are perpetual and will
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Burmeister et al. Expires October 2004 26
