Subject: [Iccrg] A Wireless Channel Model Based Rate Control (WMRC) Scheme (WAS Re: present a draft)
From: [email-masked] (Lachlan Andrew)
Date: Mon Nov 22 22:03:14 2010

Greetings Dong,

Thanks for the update.  My concern wasn't that the draft lacks
implementation details.  It is that I think this type of result is
better published in a journal or presented at a conference.  It is a
solution that works well for one particular model of a wireless
channel, but wireless channels are very hard to model accurately.

If you want this protocol standardized by the IETF, I would recommend
publishing several papers evaluating it under a wide range of wireless
settings (WiFi, 3G, LTE, WiMax, satellite, Zigbee) and showing that
the algorithm is robust.  Then summarize the results in a
recommendation to the IETF.

Cheers,
Lachlan

On 22 November 2010 21:27:11 UTC+11, Y Dong <[email-masked]> wrote:
>
> It seems the main concern is its being ?too theoretical?. So we added a
> section in the modified draft (see attachment): Section 6 Packet Processing
> Protocol to give more details of the sender and receiver behavior for
> simulation /experimental test.
>
> As for the "fairness" issue, we have tested the inter-flow fairness of the
> proposed algorithm by simulation. And we are trying to test its performance
> in real networks with wireless access, e.g. The IEEE 802.11 based Wlan or 3G
> networks.
>
> We would be grateful to all especially Ingemar and Lachlan for any further
> comments.
>
> ---------- Forwarded message ----------
> From: Lachlan Andrew <[email-masked]>
>
> Although I'm a theorist at heart, on a cursory glance I also feel that
> this may be too theoretical for an IETF standard. ?There are many
> possible channel models (for many possible channels), and
> standardizing transmission based on one should be done with caution.
>
> IETF documents are mainly useful for issues that need agreement to be
> implemented. ?The main reason congestion control algorithms (as
> distinct from protocols) need agreement is for "fairness". ?Fairness
> in systems with different channels is a very complicated issue, which
> doesn't seem to be addressed in this draft.
>
> $0.02,
> Lachlan

-- 
Lachlan Andrew? Centre for Advanced Internet Architectures (CAIA)
Swinburne University of Technology, Melbourne, Australia
<http://caia.swin.edu.au/cv/landrew>
Ph +61 3 9214 4837

From: [email-masked] (Y Dong)
Date: Tue Nov 23 11:07:06 2010
Subject: {SPAM?} Re: [Iccrg] A Wireless Channel Model Based Rate Control (WMRC) Scheme (WAS Re: present a draft)
Message-ID: <00a201cb8a2f$d33939b0$188f0a0a@759aa7dc016a46c>

Skipped content of type multipart/alternative-------------- next part --------------

                                                                     

Datagram Congestion Control		                Yu-ning Dong
Internet Draft	                                        Hai-tao Zhao
Intended status: Experimental    Nanjing Univ. of Posts and Telecom.
Expires: May 2011                                  November 19, 2010

A Wireless Channel Model Based Rate Control (WMRC) Scheme in RTP/UDP
for Real Time Multimedia Transmissions over Wired-Wireless Networks
                        (WMRC-RTP/UDP)
                draft-dong-wmrc-rtpcontrol-01.txt

Status of this Memo

   This document is an Internet-Draft and is subject to all provisions
   of section 3 of RFC 3667. By submitting this Internet-Draft, each
   author represents that any applicable patent or other IPR claims of
   which he or she is aware have been or will be disclosed, and any of
   which he or she become aware will be disclosed, in accordance with
   RFC 3668.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-Drafts.
   
   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."
   
   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt.
   
   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.
   
   This Internet-Draft will expire on May 19, 2011.

Copyright Notice

   Copyright (c) 2010 IETF Trust and the persons identified as the
   document authors. All rights reserved.

Abstract

   This document introduces a Wireless Channel Model Based rate control
   scheme for improving the behavior of real time multimedia streams

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    with TFRC [1] in wireless-wired heterogeneous networks. Based on
    wireless channel characteristics, the method can first identify the
    level of packet losses of two different types by sending large and
    small packets alternately, then adopt different adaptive rate
    control strategies to increase the network throughput and decrease
    congestion packet loss rate, to improve transmission quality of
    real-time multimedia stream. The proposed method is compared with
    previously reported algorithms [2-3] by simulation. It is shown from
    the simulation results in different network topology environments,
    the performance the proposed algorithm is better than existing
    algorithms in the aspects of network bandwidth utilization and
    congestion packet loss control. Parts of this method published in
    the IEEE Wireless Communications and Networking Conference (WCNC2007)
    and in a pending China patent (access number: CN101686100).

Table of Contents

   1.  Introduction. . . . . . . . . . . . . . . . . . . . . .  2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Packet Loss Discrimination. . . . . . . . . . . . . . .  3
   4.  Adaptation to Network State Changes . . . . . . . . . .  7
   5.  WMRC Behavior Description . . . . . . . . . . . . . . .  8
       5.1. Adaptive Rate Control Mechanism. . . . . . . . . .  8
       5.2. WMRC Specific Implementation Steps . . . . . . . .  9
   6. Packet Processing Protocol . . . . . . . . . . . . . . . 10
       6.1. Sender Initialization. . . . . . . . . . . . . . . 10
       6.2. Receiver Behavior When a Data Packet Is Received . 10
       6.3. Sender Behavior When a Feedback Packet Is Received.10
   7.  Security Considerations . . . . . . . . . . . . . . . . 12
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . 12
   9.  Conclusions . . . . . . . . . . . . . . . . . . . . . . 12
   10.  References . . . . . . . . . . . . . . . . . . . . . . 13
       10.1. Informative References. . . . . . . . . . . . . . 13
   11. Acknowledgments . . . . . . . . . . . . . . . . . . . . 14

1.  Introduction

   This document describes a wireless channel model based packet loss
   discrimination method that can differentiate wireless random bit
   error packet loss from congestion packet loss over wireless-wired
   networks. This work focuses on how to obtain the dynamic change
   characteristics of wireless communication network from the
   transport/application layer, and to achieve a better end-to-end
   quality of service by making adaptive adjustment according to these
   changes. The proposed scheme shows to be more accurate than existing

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   methods in estimating current network status by means of a wireless  
   channel model and statistical analysis of large and small packets
   loss rates, and its performance basically not affected by the
   variation of network topology and the competition flows. The realtime
   multimedia transmission protocol can carry out performance
   optimization based on the packet loss discrimination results. For
   example, if only the wireless losses are reported, the source and
   channel coding ratio can be adjusted to increase the data protection
   instead of reducing sending rate.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 
   document are to be interpreted as described in RFC-2119 [RFC2119].

3.  Packet Loss Discrimination

   Previous research indicates that the packet loss probability caused
   by random bit errors is related to the packet size in wireless
   environment, namely, the larger the packet size, the higher the
   packet loss probability. The congestion packet losses (drop-tail
   router), however, are generally independent of the packet size.
   Based on this observation, one can send small probing packets
   regularly to the channel to distinguish from the large-size video
   packets, and estimate current main cause of packet losses from their
   feedback information. However, channel bandwidth utilization will be
   reduced by these probing packets. Therefore, we use small-size video
   packets instead of probing packet to improve channel bandwidth
   utilization. A method for identifying packet loss reason from the
   packet loss rates of large and small packets is developed below.
    
   In order to identify between congestion packet losses and wireless
   fading losses, the sender node sends large and small packets 
   alternately and the statistics of the lost large and small packets 
   over a period is calculated at the receiver end. For ease of analysis,
   let us define the following variables:

      Nps = the number of lost small packets;

      Npsc = the number of lost small packets due to congestions;

      Npsb = the number of lost small packets due to bit errors;

      Npl = the number of lost large packets;

      Nplc = the number of lost large packets due to congestions;

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      Nplb = the number of lost large packets due to bit errors.

   According to traditional communication theory, the wireless packet
   loss rate varies exponentially with the channel bit error rate r. In
   some cases such as uniform distribution with a very small r, the
   variations of packet loss rate with r can be approximated by a
   linear function. For the wired channel, as mentioned above, the
   numbers of large and small congestion packet losses in the given 
   period would approximately be equal, namely Nplc=Npsc. Thus, we have 
   the following equations:

      Npl = Nplc + Nplb (1)

      Nps = Npsc + Npsb (2)

   Let B denote the ratio of Nplb to Npsb in certain wireless channel
   condition. Obviously, B is a function of the channel bit error rate 
   r and can be expressed as:

      B(r) = Nplb / Npsb (3)

   Thus, equation (1) can be rewritten as:   

      Npl = Npsc + B(r) * Npsb (4)

   At the receiver end, one can obtain the statistical values of Npl and 
   Nps over a time interval. If B is known, one can get the values of 
   Nplc (=Npsc), Nplb and Npsb by solving above equations, namely, the 
   congestion loss rate and random erroneous loss rate of large and 
   small packets respectively. One can then know current congestion 
   level of the wired networks from Nplc and Npsc, and fading condition 
   of the wireless link from Nplb and Npsb.
   
   The problem now is how to know the value of B, and there seems very
   few works have been done on this issue. In [4], B was assumed to be a
   constant, dependent on the sizes of large and small packets. This
   however is not always true according to our experimental results.
   Therefore, by taking this assumption in solving equations (2) through
   (4) for other variables, the applicable scope of the obtained values
   will probably be rather limited.
   
   To analyze the relations between packet loss rates and the packet
   sizes, a group of experimental tests in wireless channels have been
   carried out by using a modified Jakes Rayleigh fading model [5] in
   different channel BERs (Bit Error Rates) and packet sizes. Linear,
   quadratic and exponential curves are used to fit the obtained

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   simulation data where the exponential fitting has the largest fitting
   errors.

   Let G denote packet length, and assume the random bit error of a
   wireless channel is subject to uniform distribution and any two bit
   error events are uncorrelated. The packet loss error rate P of
   wireless link can thus be computed as

      P=1-(1-r)^G (5)
 
   If r is very small, by Taylor expansion in r = 0 and omitting higher
   order terms, the linear (first-order) and second-order approximation
   of equation (5) can be obtained as follows,

   Linear approximation (first order): P=r*G (6)

   Second-order approximation: P=r*G*(1-r*G/2) (7)

   If the linear approximation is used, we have,

      Pl/Ps = Gl/Gs = b (8)

   Where Pl and Ps are the loss rate of large and small packets
   respectively; Gl and Gs are the lengths of large and small packet
   respectively; b denotes the ratio of the large packet length to small
   packet length. B in this condition is a constant (=b), which is
   frequently used in previous literatures. However, when r is not small
   enough, the above equation is not applicable. At this time, one can
   consider second-order approximation and the ratio can be obtained
   from equation (7) as,

                 2-r*b*Gs
      Pl/Ps = b*----------- = b*A(r) (9)
                  2-r*Gs
   
   Where A(r) = (2-r*b*Gs)/ (2-r*Gs) (10)

   When b = 2, the above equation becomes,

      A(r) = (1-r*Gs)/ (1-r*Gs/2) (11)
 
   From above analysis, we can see that, when r is not small enough,
   Pl/Ps is not a constant but determined by A. In most practical
   settings, the reasonable value scope of A in equation (11) is 0.6~1
   on condition that r<4/7*Gs (In fact, this is not a necessary
   condition). For example, when Gs=4000bits (500Bytes), then r<1.4*10^-
   4, and when Gs£½800bits (100Bytes), r<7.1*10^-4. That is to say, the

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   bit error rate is kept in a small or medium value range. Otherwise,
   the second-order approximation (equation (7)) may not be applicable.
   A(r) monotonically decreases for a given value G with the increase of
   r.

   On the other hand, let us define the following variable,
   
      &N = Npl - Nps = Nplc + Nplb - Npsc - Npsb = Nplb - Npsb = &Nb
      (Since Nplc =Npsc) (12)

   From equation (12), we can know that the difference between the
   number of lost large and small packets is equal to the difference
   between the number of lost large and small packets due to bit errors.
   The wireless packet loss rate P increases linearly with packet size
   under certain r ranges, namely,

      &P=k*&G (13)
 
   Where &P=Pl-Ps, &G=Gl-Gs and k (k>0) is an r-related constant. Since
   &P can be calculated from equation (12), and &G is known in advance,
   constant k can be obtained, that reflects the degree of link bit
   errors.

   The slope of packet loss rate curve decreases with packet length
   increase when BER is relatively high; while the curve slope almost
   keeps stable and uncorrelated to packet length when BER is small,
   namely, the packet loss rate varies linearly with packet length under
   small BER conditions. Therefore, one can estimate the current level
   of r from k within the scope of r<1.8*10^-4.

   As for congestion packet losses in wired networks, the adaptive rate
   control mechanism TFRC[1], decreases the sending rate when high
   packet loss rate is reported, and allows the loss rate to decrease
   quickly (assuming the video stream shares the network bandwidth with
   other TCP-friendly traffic) [6]. According to our NS2 simulation
   results, the packet loss rate can normally drop by more than 50%
   within 4 RTT (Round Trip Time) time.

   Based on above analyses, we propose the following computation
   strategy:

   1) When the packet loss rate rises to an unacceptable level, the rate
   control mechanism will decrease the sending rate. Thus, (1) If this
   high loss rate is due to congestions of the wired link, then as
   discussed above, the loss rate will drop considerably within several
   RTTs; (2) If however, the loss rate does not show any obvious drop
   within the time interval, one may regard present packet losses most

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   probably due to the bit errors of the wireless link rather than
   network congestions.

   2) If present packet losses are indeed mainly due to bit errors of
   the wireless link, we may assume the number of congestion packet
   losses Npsc in (2) and (4) much less than the number of erroneous
   packet losses Npsb, and Nplc << Nplb. Equations (2) and (4) can then
   be simplified as:

      Nps = Npsb (14)

      Npl = B(r)*Npsb (15)

   From above equations, one can estimate the value of B, and@the
   same time identify the main cause of current packet losses. Note that
   in our method, there is no assumption of the linear correlation of
   the erroneous packet loss rate with the packet size.

   3) A linear prediction with error correction method is adopted to
   estimate the value of B. If present packet losses are mainly due to
   bit errors of the wireless link, then

      B(t) = d*B(t¨C1) + (1-d)*&B(t) (B0 is the initial value) (16a)
      
      Otherwise, B(t)=B0 (16b)

   Where B0=Gl/Gs; &B(t)=Npl/Nps is the prediction error corrective
   value; d is a weighting coefficient (0<d<1); t£½1£¬2£¬¡, denotes
   sampling time with sampling interval of q*RTT (0<q<=1, a constant).
   
   As mentioned above, B is no longer a constant when wireless random
   bit errors are high. Therefore, we adopt the linear prediction with
   error correction (corrective value &B(t)) method to gradually track
   the value of B as shown in equation (16a). The weighting coefficient
   d determines tracking speed of B(t) to the real value of B that is
   not a constant, and the selection of d should compromise between
   tracking speed and stability of B. Equation (16b) represents the case
   of low BER of wireless link, where packet losses are mainly due to
   network congestion and B can be approximated by a constant (see
   equation (8)).

4. Adaptation to Network State Changes

   For timely response to the change of network status, a finite length
   history record based large and small packet loss statistics is used.
   In a sliding window, Nps and Npl record the lost number of small and
   large packets respectively. When a new packet loss occurs, the oldest

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   recorded one will be removed from the record queue, and the
   statistical value will be updated correspondingly. The main concern
   here is the selection of the queue length: on one hand, if the queue
   is shorter, the response to the change of network status will be
   quick, but may cause stability problem; on the other hand, if the
   queue is too long, the sending rate will be relatively stable, but
   the response to network state changes may be slow. Therefore, the
   length of lost packets record queue represents a compromise between
   response time and control stability.

   In order to track the change of the network states effectively, we
   introduce another statistic to help determine the current network
   states. Let,
 
      Dec = 2*(B*Ps-Pl) / ((B-1)*(Ps+Pl)) (17)

      And, Pc = (Npsc+Nplc) / (Nps+Npl) (18)

   From equations (1)-(4), one can see that the statistic Dec is equal
   to Pc in a statistical sense, and reflects the ratio of the number of
   congestion packet losses to the number of total lost packets.
   Therefore, the value of Dec also reflects the congestion/wireless
   packet loss status of current networks. The larger the value is, the
   more serious the congestion packet losses. The values of Pl and Ps in
   equation (17) are approximated by the reciprocal of the average
   length of intervals of packet loss events [RFC 5348, section 5] for
   large and small packets respectively.

   The advantage of calculating Dec by equation (17) is: if no packet
   loss happens for a long time, although no new packet loss event
   occurs, the interval between packet loss events gets longer, and the
   congestion reduction can be reflected timely by the weighting
   coefficient of sliding window. This means the weight of loss events
   that occur more recently is larger, and that of older loss events is
   smaller. In this way, we can track current network status more
   quickly, and at the same time achieve good control stability.

5. WMRC Behavior Description

5.1.  Adaptive Rate Control Mechanism

   To satisfy the TCP friendliness requirement, an adaptive rate control
   based on TCP throughput model is used in this paper. The used TCP
   throughput mathematical model is as follows [1]:

                                       TU
      rate= ------------------------------------------------------- (19)
            RTT*sqrt(2*p/3) + (4*tout*(3*sqrt(3*p/8)*p*(1+32*p^2)))

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   Where, TU denotes data packet size, RTT denotes round trip time, tout
   denotes retransmission timeout time (tout = 4 RTTs), P denotes packet
   loss event rate[1]. Equation (19) works well in wired IP environment,
   but in wireless environment, the performance of conventional TCP/TFRC
   is often unsatisfactory. Therefore, we modify this model a bit by
   removing the number of lost packets due to wireless bit errors in the
   calculation of P. Thus the capability to differentiate the packet
   loss types will be essential.

   In the WMRC algorithm, when the receiver detects a packet loss event,
   it calculates the Dec first, then judges whether it is a congestion
   packet loss by comparing the value of Dec with a given threshold Dth
   (0<Dth<1). If it is judged as a congestion packet loss (Dec > Dth),
   this packet loss event will be used in the calculation of packet loss
   rate P in Equation (19); otherwise, it won't be. Thus, the packet
   loss rate P obtained from the RTCP feedback packets at the sender
   reflects only the congestion packet loss rate, not including the
   wireless packet losses. And the calculated sending rate will be fit
   the current network situation well. Then, the situation of excessive
   restriction of the sending rate will not happen. The smaller the
   value of Dth is, the greater the possibility of judging the current
   packet loss as the congestion loss; conversely, the greater the
   possibility of judging it as the wireless random packet loss. We
   generally set Dth=0.8 after running a number of experimental tests.

5.2. WMRC Specific Implementation Steps

   The proposed scheme is based on RTP/UDP transport layer protocol [7],
   and works in following steps:

   1) Initialization: after setting the initial sending rate, the
   parameters of a, b, d, h, q, Gl, Gs, Dth, and the length of record
   queue of packet losses, the algorithm enters a slow start stage.

   2) When the media streaming gets into a stable state when packet loss
   occurs, the receiver end computes the statistical values of Npl, Nps,
   Pl, Ps, RTT, and P (recording the congestion packet loss rate only).

   3) The receiver end calculates the value of Dec by equation (17) for
   every packet loss event, and estimates the reason of the current
   packet loss. If Dec > Dth, the packet loss is judged as the
   congestion loss, otherwise, the wireless packet loss. If it is a
   congestion packet loss, it will be used to update the packet loss
   rate P; otherwise it will not be. At every sampling time point t, B(t)
   is updated by equation (16) according to the nature of the current

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   packet loss. At every certain time interval, the receiver informs the
   sender the RTT and packet loss rate P by RTCP feedback packets.

   4) If there are packet losses occurred during the feedback period,
   when the sender receives the RTCP feedback packets, rate control will
   be carried out according to the equation (19). Otherwise, a MIMD(a, b)
   congestion control scheme [8] will be adopted.
   
6. WMRC Packet Processing Protocol

6.1. Sender Initialization

   Sender parameter settings: a=0.2, b=0.1, d=0.9, h=0.5, q=0.5,
   Gl=1016Bytes, Gs=508Bytes, and Dth=0.8. A loss queue of 32 packets is
   used in WMRC. The initial slow-start phase will basically be the same
   as that of TFRC [1,section 4].

6.2. Receiver Behavior When a Data Packet Is Received

   At the receiver end, if there are losses of large packets, one 
   Updates the loss queue by Npl+1; otherwise (there are losses of small
   packets), Nps+1, and simultaneously calculates Dec by Equation (17).
   The values of Pl and Ps are calculated based on the length of 
   intervals of (large and small) packet loss events in a similar way as 
   TFRC [RFC 5348, section 5]. Then B is updated by B(t) = d*B(t-1)+(1-
   d)*&B(t). If Dec is greater than a fixed threshold Dth, we judge the 
   current packet loss type as congestion loss and record this packet 
   loss event (that will be used in the computation of the packet loss
   event rate p); otherwise the current packet loss type is judged as
   random loss and we do not record this loss event. At every h*RTT
   (0<h<=1) time, the receiver end feeds back the loss event rate p and
   round trip time RTT to the sender with a feedback packet.

6.3. Sender Behavior When a Feedback Packet Is Received
   
   The algorithm of WMRC will decrease the sender's sending rate when
   high packet loss rates are reported, expecting the loss rates to drop
   quickly (assuming the multimedia stream shares the network bandwidth
   with other TCP-friendly traffic). We propose the following sending
   strategy of the WMRC algorithm at the sender end.

   When the sender receives the feedback packet, it will carry out rate
   control according to (19) with the packet loss rate p and round trip
   time RTT. Sometimes the calculated rate may not suit the current
   networks state due to the change of round trip time. If there is no
   packet loss reported in this feedback round, or the calculated rate

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   is 1.5 times more than the current sending rate, the sender's rate
   control will be based on the MIMD (a, b) algorithm (described below). 

   The MIMD(a, b) congestion control algorithm can be briefly described
   as: a and b are the parameters of MIMD(a, b) model to increase or
   decrease the sending rate respectively, and 0<a, b<1. Let R denote
   the current sending rate, if no packet losses occurred in this
   control period, the sending rate will be increased to (a+1)*R,
   otherwise, decreased to (1-b)*R.

   The whole adaptive rate control scheme is described in Algorithm 1.
   -------------------------------------------------------------------
   Algorithm 1: The adaptive rate control scheme
   -------------------------------------------------------------------
   1. Initialization
   2. Updating Nps, Npl
   3. For {receive a feedback packet} do
   4. Compute Dec by equations (17) and (1)-(4)
   5. If Dec>Dth
   6. The current packet loss is the congestion loss, and updates the
      packet loss rate P
   7. Else then
   8. The current packet loss is the wireless loss, and does not update the
      packet loss rate P
   9. End else
   10.End if
   11. At a certain time interval, updating RTT and P, and compute Rate
       by equation (19)
   12.  If no packet loss or Rate > 1.5 Rcurrent
   13.  The rate control be done by the MIMD algorithm
   14.  Else then
   15.  The next transmission rate Rn = Rate
   16.  End else
   17.  End if
   18. End for
   -------------------------------------------------------------------

7. Security Considerations
 
   WMRC is not a transport protocol in its own right, but a congestion
   control mechanism that is intended to be used in conjunction with a
   transport protocol. Therefore security primarily needs to be
   considered in the context of a specific transport protocol and its
   authentication mechanisms. Congestion control mechanisms can
   potentially be exploited to create denial of service. This may occur

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   through spoofed feedback. Thus any transport protocol that uses WMRC
   should take care to ensure that feedback is only accepted from the
   receiver of the data. The precise mechanism to achieve this will
   however depend on the transport protocol itself.

   In addition, congestion control mechanisms may potentially be
   manipulated by a greedy receiver that wishes to receive more than its
   fair share of network bandwidth. A receiver might do this by claiming
   to have received packets that in fact were lost due to congestion.
   Possible defenses against such a receiver would normally include some
   form of nonce that the receiver must feed back to the sender to prove
   receipt. However, the details of such a nonce would depend on the
   transport protocol, and in particular on whether the transport
   protocol is reliable or unreliable.

   We expect that protocols incorporating large/small packet with WMRC
   will also want to incorporate feedback from the receiver to the
   sender using packet loss discrimination. The packet loss
   discrimination is a modification to TFRC that distinguishes the loss
   packets from congestion loss or wireless random error.

8. IANA Considerations

   There are no IANA actions required for this document.

9. Conclusions

   This document presents a wireless channel model based rate control
   scheme WMRC for wireless multimedia transmission control over hybrid
   networks. This scheme can detect the network status and differentiate
   packet loss types (wireless or congestion losses) by means of a
   wireless channel model and a special packet sending scheme with
   different packet sizes. It can adapt to the dynamic change of the
   networks and control the sending rate effectively. Theoretical
   analysis and detailed implementation of the proposed scheme are given.
   At the same time, it should be noted that due to the use of large and
   small packets alternately in the proposed algorithm, the application
   layer needs to pack data into two different-size packets for
   transmission, which may increase the overhead of packing process.

10. References
 
10.1. Informative References
 
   [1] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
   Friendly Rate Control (TFRC): Protocol Specification", RFC 5348,
   September 2008.

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   [2] Song Cen, et al, ¡°End-to-End Differentiation of Congestion and
   Wireless Losses,¡± IEEE/ACM Transactions on Networking, vol. 11,
   no. 5, pp. 703-717, October 2003.

   [3] Min Kyu Park, Kue-Hwan Sihn, Jun Ho Jeong, ¡°A Statistical
   Method of Packet Loss Type Discrimination in Wired-Wireless
   Networks,¡± Proc. IEEE CCNC, 2006, pp. 458-462.

   [4] C. L. Lee, et al, ¡°On the Use of Loss History for Performance
   Improvement of TCP over Wireless Networks¡±, IEICE Trans.
   Commun. , vol. E85-B, no. 11, pp. 2457-2467, 2002.

   [5] H.-J. Lee, H.-J. Byun, and J.-T. Lim, ¡°TCP-friendly congestion
   control for streaming real-time applications over wireless
   networks¡±, IET Commun., vol. 2, no. 1, pp. 159¨C163, 2008.

   [6] S. Floyd and J. Padhye, ¡°Equation-Based congestion control for
   unicast applications¡±, Proc. ACM SIGCOMM¡¯00, 2000, pp. 43-56.

   [7] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
   "RTP: A Transport Protocol for Real-Time Applications", RFC
   1889, January 1996.

   [8] Yu-ning Dong and Meng-yue Chen, ¡°Real time video transmission
   control in wireless-wired IP networks¡±, Proc. IEEE Wireless
   Communications and Networking Conference (WCNC2007), Hong Kong,
   Mar. 2007, pp. 3687-3691.

11. Acknowledgments

   The authors would like to acknowledge feedback and discussions on
   equation-based congestion control with a wide range of people,
   including members of the Wireless Communication Research Group and
   the End-to-End Research Group. Thanks are given to the National
   Natural Science Foundation of China (No.60972038), the Jiangsu
   Province Universities Natural Science Research Key Grant Project
   (07KJA51006), and Jiangsu Province Graduate Innovative Research Plan
   (CX09B_149Z) for their financial support.

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Authors¡¯ Addresses

   Yu-ning Dong
   Nanjing University of Posts and Telecommunications PO Box 166
   66 New Mo-fan-ma-lu Road, Nanjing, Jiangsu, 210003
   China

   Email: [email-masked]

   Hai-tao Zhao
   Nanjing University of Posts and Telecommunications PO Box 54
   66 New Mo-fan-ma-lu Road, Nanjing, Jiangsu, 210003
   China

   Email: [email-masked]

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From: [email-masked] (=?GB2312?B?1dS6o8zO?=)
Date: Tue Nov 16 09:34:39 2010
Subject: [Iccrg] Present an modified thought of a draft on ICCRG work group
Message-ID: <[email-masked]>

Skipped content of type multipart/alternative-------------- next part --------------

                                                                     

Datagram Congestion Control		                Yu-ning Dong
Internet Draft	                                        Hai-tao Zhao
Intended status: Experimental    Nanjing Univ. of Posts and Telecom.
Expires: May 2011                                  November 15, 2010

A Wireless Channel Model Based Rate Control (WMRC) Scheme in RTP/UDP
for Real Time Multimedia Transmissions over Wired-Wireless Networks
                        (WMRC-RTP/UDP)
                draft-dong-wmrc-rtpcontrol-01.txt

Status of this Memo

   This document is an Internet-Draft and is subject to all provisions
   of section 3 of RFC 3667. By submitting this Internet-Draft, each
   author represents that any applicable patent or other IPR claims of
   which he or she is aware have been or will be disclosed, and any of
   which he or she become aware will be disclosed, in accordance with
   RFC 3668.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-Drafts.
   
   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress."
   
   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt.
   
   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html.
   
   This Internet-Draft will expire on May 15, 2011.

Copyright Notice

   Copyright (c) 2010 IETF Trust and the persons identified as the
   document authors. All rights reserved.

Abstract

   This document introduces a Wireless Channel Model Based rate control
   scheme for improving the behavior of real time multimedia streams

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    with TFRC [1] in wireless-wired heterogeneous networks. Based on
    wireless channel characteristics, the method can first identify the
    level of packet losses of two different types by sending large and
    small packets alternately, then adopt different adaptive rate
    control strategies to increase the network throughput and decrease
    congestion packet loss rate, to improve transmission quality of
    real-time multimedia stream. The proposed method is compared with
    previously reported algorithms [2-3] by simulation. It is shown from
    the simulation results in different network topology environments,
    the performance the proposed algorithm is better than existing
    algorithms in the aspects of network bandwidth utilization and
    congestion packet loss control. Parts of this method published in
    the IEEE Wireless Communications and Networking Conference (WCNC2007)
    and in a pending China patent (access number: CN101686100).

Table of Contents

   1.  Introduction. . . . . . . . . . . . . . . . . . . . . .  2
   2.  Terminology . . . . . . . . . . . . . . . . . . . . . .  3
   3.  Packet Loss Discrimination. . . . . . . . . . . . . . .  3
   4.  Adaptation to Network State Changes . . . . . . . . . .  7
   5.  WMRC Behavior Description . . . . . . . . . . . . . . .  8
       5.1. Adaptive Rate Control Mechanism. . . . . . . . . .  8
       5.2. WMRC Specific Implementation Steps . . . . . . . .  9
   6. Packet Processing Protocol . . . . . . . . . . . . . . . 10
       6.1. Receiver Behavior When a Data Packet Is Received . 10
       6.2. Sender Behavior When a Feedback Packet Is Received.10
   7.  Security Considerations . . . . . . . . . . . . . . . . 12
   8.  IANA Considerations . . . . . . . . . . . . . . . . . . 12
   9.  Conclusions . . . . . . . . . . . . . . . . . . . . . . 12
   10.  References . . . . . . . . . . . . . . . . . . . . . . 13
       10.1. Informative References. . . . . . . . . . . . . . 13
   11. Acknowledgments . . . . . . . . . . . . . . . . . . . . 14

1.  Introduction

   This document describes a wireless channel model based packet loss
   discrimination method that can differentiate wireless random bit
   error packet loss from congestion packet loss over wireless-wired
   networks. This work focuses on how to obtain the dynamic change
   characteristics of wireless communication network from the
   transport/application layer, and to achieve a better end-to-end
   quality of service by making adaptive adjustment according to these
   changes. The proposed scheme shows to be more accurate than existing
   methods in estimating current network status by means of a wireless

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   channel model and statistical analysis of large and small packets
   loss rates, and its performance basically not affected by the
   variation of network topology and the competition flows. The realtime
   multimedia transmission protocol can carry out performance
   optimization based on the packet loss discrimination results. For
   example, if only the wireless losses are reported, the source and
   channel coding ratio can be adjusted to increase the data protection
   instead of reducing sending rate.

2.  Terminology

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this 
   document are to be interpreted as described in RFC-2119 [RFC2119].

3.  Packet Loss Discrimination

   Previous research indicates that the packet loss probability caused
   by random bit errors is related to the packet size in wireless
   environment, namely, the larger the packet size, the higher the
   packet loss probability. The congestion packet losses (drop-tail
   router), however, are generally independent of the packet size.
   Based on this observation, one can send small probing packets
   regularly to the channel to distinguish from the large-size video
   packets, and estimate current main cause of packet losses from their
   feedback information. However, channel bandwidth utilization will be
   reduced by these probing packets. Therefore, we use small-size video
   packets instead of probing packet to improve channel bandwidth
   utilization. A method for identifying packet loss reason from the
   packet loss rates of large and small packets is developed below.
    
   In order to identify between congestion packet losses and wireless
   fading losses, the sender node sends large and small packets 
   alternately and the statistics of the lost large and small packets 
   over a period is calculated at the receiver end. For ease of analysis,
   let us define the following variables:

      Nps = the number of lost small packets;

      Npsc = the number of lost small packets due to congestions;

      Npsb = the number of lost small packets due to bit errors;

      Npl = the number of lost large packets;

      Nplc = the number of lost large packets due to congestions;

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      Nplb = the number of lost large packets due to bit errors.

   According to traditional communication theory, the wireless packet
   loss rate varies exponentially with the channel bit error rate r. In
   some cases such as uniform distribution with a very small r, the
   variations of packet loss rate with r can be approximated by a
   linear function. For the wired channel, as mentioned above, the
   numbers of large and small congestion packet losses in the given 
   period would approximately be equal, namely Nplc=Npsc. Thus, we have 
   the following equations:

      Npl = Nplc + Nplb (1)

      Nps = Npsc + Npsb (2)

   Let B denote the ratio of Nplb to Npsb in certain wireless channel
   condition. Obviously, B is a function of the channel bit error rate 
   r and can be expressed as:

      B(r) = Nplb / Npsb (3)

   Thus, equation (1) can be rewritten as:   

      Npl = Npsc + B(r) * Npsb (4)

   At the receiver end, one can obtain the statistical values of Npl and 
   Nps over a time interval. If B is known, one can get the values of 
   Nplc (=Npsc), Nplb and Npsb by solving above equations, namely, the 
   congestion loss rate and random erroneous loss rate of large and 
   small packets respectively. One can then know current congestion 
   level of the wired networks from Nplc and Npsc, and fading condition 
   of the wireless link from Nplb and Npsb.
   
   The problem now is how to know the value of B, and there seems very
   few works have been done on this issue. In [4], B was assumed to be a
   constant, dependent on the sizes of large and small packets. This
   however is not always true according to our experimental results.
   Therefore, by taking this assumption in solving equations (2) through
   (4) for other variables, the applicable scope of the obtained values
   will probably be rather limited.
   
   To analyze the relations between packet loss rates and the packet
   sizes, a group of experimental tests in wireless channels have been
   carried out by using a modified Jakes Rayleigh fading model [5] in
   different channel BERs (Bit Error Rates) and packet sizes. Linear,
   quadratic and exponential curves are used to fit the obtained

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   simulation data where the exponential fitting has the largest fitting
   errors.

   Let G denote packet length, and assume the random bit error of a
   wireless channel is subject to uniform distribution and any two bit
   error events are uncorrelated. The packet loss error rate P of
   wireless link can thus be computed as

      P=1-(1-r)^G (5)
 
   If r is very small, by Taylor expansion in r = 0 and omitting higher
   order terms, the linear (first-order) and second-order approximation
   of equation (5) can be obtained as follows,

   Linear approximation (first order): P=r*G (6)

   Second-order approximation: P=r*G*(1-r*G/2) (7)

   If the linear approximation is used, we have,

      Pl/Ps = Gl/Gs = b (8)

   Where Pl and Ps are the loss rate of large and small packets
   respectively; Gl and Gs are the lengths of large and small packet
   respectively; b denotes the ratio of the large packet length to small
   packet length. B in this condition is a constant (=b), which is
   frequently used in previous literatures. However, when r is not small
   enough, the above equation is not applicable. At this time, one can
   consider second-order approximation and the ratio can be obtained
   from equation (7) as,

                 2-r*b*Gs
      Pl/Ps = b*----------- = b*A(r) (9)
                  2-r*Gs
   
   Where A(r) = (2-r*b*Gs)/ (2-r*Gs) (10)

   When b = 2, the above equation becomes,

      A(r) = (1-r*Gs)/ (1-r*Gs/2) (11)
 
   From above analysis, we can see that, when r is not small enough,
   Pl/Ps is not a constant but determined by A. In most practical
   settings, the reasonable value scope of A in equation (11) is 0.6~1
   on condition that r<4/7*Gs (In fact, this is not a necessary
   condition). For example, when Gs=4000bits (500Bytes), then r<1.4*10^-
   4, and when Gs??800bits (100Bytes), r<7.1*10^-4. That is to say, the

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   bit error rate is kept in a small or medium value range. Otherwise,
   the second-order approximation (equation (7)) may not be applicable.
   A(r) monotonically decreases for a given value G with the increase of
   r.

   On the other hand, let us define the following variable,
   
      &N = Npl ?C Nps = Nplc + Nplb ?C Npsc ?C Npsb = Nplb ?C Npsb = &Nb
      (Since Nplc =Npsc) (12)

   From equation (12), we can know that the difference between the
   number of lost large and small packets is equal to the difference
   between the number of lost large and small packets due to bit errors.
   The wireless packet loss rate P increases linearly with packet size
   under certain r ranges, namely,

      &P=k*&G (13)
 
   Where &P=Pl-Ps, &G=Gl-Gs and k (k>0) is an r-related constant. Since
   &P can be calculated from equation (12), and &G is known in advance,
   constant k can be obtained, that reflects the degree of link bit
   errors.

   The slope of packet loss rate curve decreases with packet length
   increase when BER is relatively high; while the curve slope almost
   keeps stable and uncorrelated to packet length when BER is small,
   namely, the packet loss rate varies linearly with packet length under
   small BER conditions. Therefore, one can estimate the current level
   of r from k within the scope of r<1.8*10^-4.

   As for congestion packet losses in wired networks, the adaptive rate
   control mechanism TFRC[1], decreases the sending rate when high
   packet loss rate is reported, and allows the loss rate to decrease
   quickly (assuming the video stream shares the network bandwidth with
   other TCP-friendly traffic) [6]. According to our NS2 simulation
   results, the packet loss rate can normally drop by more than 50%
   within 4 RTT (Round Trip Time) time.

   Based on above analyses, we propose the following computation
   strategy:

   1) When the packet loss rate rises to an unacceptable level, the rate
   control mechanism will decrease the sending rate. Thus, (1) If this
   high loss rate is due to congestions of the wired link, then as
   discussed above, the loss rate will drop considerably within several
   RTTs; (2) If however, the loss rate doesn??t show any obvious drop
   within the time interval, one may regard present packet losses most

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   probably due to the bit errors of the wireless link rather than
   network congestions.

   2) If present packet losses are indeed mainly due to bit errors of
   the wireless link, we may assume the number of congestion packet
   losses Npsc in (2) and (4) much less than the number of erroneous
   packet losses Npsb, and Nplc << Nplb. Equations (2) and (4) can then
   be simplified as:

      Nps = Npsb (14)

      Npl = B(r)*Npsb (15)

   From above equations, one can estimate the value of B, and@the
   same time identify the main cause of current packet losses. Note that
   in our method, there is no assumption of the linear correlation of
   the erroneous packet loss rate with the packet size.

   3) A linear prediction with error correction method is adopted to
   estimate the value of B. If present packet losses are mainly due to
   bit errors of the wireless link, then

      B(t) = d*B(t?C1) + (1-d)*&B(t) (B0 is the initial value) (16a)
      
      Otherwise, B(t)=B0 (16b)

   Where B0=Gl/Gs; &B(t)=Npl/Nps is the prediction error corrective
   value; d is a weighting coefficient (0<d<1); t??1??2????, denotes
   sampling time with sampling interval of q?RTT (0<q<=1, a constant).
   
   As mentioned above, B is no longer a constant when wireless random
   bit errors are high. Therefore, we adopt the linear prediction with
   error correction (corrective value &B(t)) method to gradually track
   the value of B as shown in equation (16a). The weighting coefficient
   d determines tracking speed of B(t) to the real value of B that is
   not a constant, and the selection of d should compromise between
   tracking speed and stability of B. Equation (16b) represents the case
   of low BER of wireless link, where packet losses are mainly due to
   network congestion and B can be approximated by a constant (see
   equation (8)).

4. Adaptation to Network State Changes

   For timely response to the change of network status, a finite length
   history record based large and small packet loss statistics is used.
   In a sliding window, Nps and Npl record the lost number of small and
   large packets respectively. When a new packet loss occurs, the oldest

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   recorded one will be removed from the record queue, and the
   statistical value will be updated correspondingly. The main concern
   here is the selection of the queue length: on one hand, if the queue
   is shorter, the response to the change of network status will be
   quick, but may cause stability problem; on the other hand, if the
   queue is too long, the sending rate will be relatively stable, but
   the response to network state changes may be slow. Therefore, the
   length of lost packets record queue represents a compromise between
   response time and control stability.

   In order to track the change of the network states effectively, we
   introduce another statistic to help determine the current network
   states. Let,
 
      Dec = 2*(B*Ps-Pl) / ((B-1)*(Ps+Pl)) (17)

      And, Pc = (Npsc+Nplc) / (Nps+Npl) (18)

   From equations (1)-(4), one can see that the statistic Dec is equal
   to Pc in a statistical sense, and reflects the ratio of the number of
   congestion packet losses to the number of total lost packets.
   Therefore, the value of Dec also reflects the congestion/wireless
   packet loss status of current networks. The larger the value is, the
   more serious the congestion packet losses. The values of Pl and Ps in
   equation (17) are approximated by the reciprocal of the average
   length of intervals of packet loss events [RFC 5348, section 5] for
   large and small packets respectively.

   The advantage of calculating Dec by equation (17) is: if no packet
   loss happens for a long time, although no new packet loss event
   occurs, the interval between packet loss events gets longer, and the
   congestion reduction can be reflected timely by the weighting
   coefficient of sliding window. This means the weight of loss events
   that occur more recently is larger, and that of older loss events is
   smaller. In this way, we can track current network status more
   quickly, and at the same time achieve good control stability.

5. WMRC Behavior Description

5.1.  Adaptive Rate Control Mechanism

   To satisfy the TCP friendliness requirement, an adaptive rate control
   based on TCP throughput model is used in this paper. The used TCP
   throughput mathematical model is as follows [1]:

                                       TU
      rate= ------------------------------------------------------- (19)
            RTT*sqrt(2*p/3) + (4*tout*(3*sqrt(3*p/8)*p*(1+32*p^2)))

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   Where, TU denotes data packet size, RTT denotes round trip time, tout
   denotes retransmission timeout time (tout = 4 RTTs), P denotes packet
   loss event rate[1]. Equation (19) works well in wired IP environment,
   but in wireless environment, the performance of conventional TCP/TFRC
   is often unsatisfactory. Therefore, we modify this model a bit by
   removing the number of lost packets due to wireless bit errors in the
   calculation of P. Thus the capability to differentiate the packet
   loss types will be essential.

   In the WMRC algorithm, when the receiver detects a packet loss event,
   it calculates the Dec first, then judges whether it is a congestion
   packet loss by comparing the value of Dec with a given threshold Dth
   (0<Dth<1). If it is judged as a congestion packet loss (Dec > Dth),
   this packet loss event will be used in the calculation of packet loss
   rate P in Equation (19); otherwise, it won't be. Thus, the packet
   loss rate P obtained from the RTCP feedback packets at the sender
   reflects only the congestion packet loss rate, not including the
   wireless packet losses. And the calculated sending rate will be fit
   the current network situation well. Then, the situation of excessive
   restriction of the sending rate will not happen. The smaller the
   value of Dth is, the greater the possibility of judging the current
   packet loss as the congestion loss; conversely, the greater the
   possibility of judging it as the wireless random packet loss. We
   generally set Dth??0.8 after running a number of experimental tests.

5.2. WMRC Specific Implementation Steps

   The proposed scheme is based on RTP/UDP transport layer protocol [7],
   and works in following steps:

   1) Initialization: after setting the initial sending rate, the
   parameters of a, b, d, h, q, Gl, Gs, Dth, and the length of record
   queue of packet losses, the algorithm enters a slow start stage.

   2) When the media streaming gets into a stable state when packet loss
   occurs, the receiver end computes the statistical values of Npl, Nps,
   Pl, Ps, RTT, and P (recording the congestion packet loss rate only).

   3) The receiver end calculates the value of Dec by equation (17) for
   every packet loss event, and estimates the reason of the current
   packet loss. If Dec > Dth, the packet loss is judged as the
   congestion loss, otherwise, the wireless packet loss. If it is a
   congestion packet loss, it will be used to update the packet loss
   rate P; otherwise it will not be. At every sampling time point t, Bt
   is updated by equation (16) according to the nature of the current

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   packet loss. At every certain time interval, the receiver informs the
   sender the RTT and packet loss rate P by RTCP feedback packets.

   4) If there are packet losses occurred during the feedback period,
   when the sender receives the RTCP feedback packets, rate control will
   be carried out according to the equation (19). Otherwise, a MIMD(a, b)
   congestion control scheme [8] will be adopted.
   
6. WMRC Packet Processing Protocol

6.1. Receiver Behavior When a Data Packet Is Received

   a=0.2, b=0.1, d=0.9, h=0.5, q=0.5, Gl=1016Bytes, Gs=508Bytes, and
   Dth=0.8. A loss queue of 32 packets is used in WMRC.
   
   At the receiver end, if there are loss of large packets, one updates
   loss queue, namely, Npl+1; otherwise, Nps+1, and simultaneously
   calculates Dec by Equation (18). The value of Pl and Ps is given by
   equations (20) and (21) that Nl and Ns is the number of sent large
   and small packets, then we can update B by B(t) = d*B(t?C1) + (1-
   d)*&B(t). When Dec is greater than a fixed threshold Dth, we judge
   that current packet loss type is congestion loss and record the loss
   packets; otherwise current packet loss type is judged as random loss
   and we do not record the loss packets. At every h?RTT (0<h<=1) time,
   the receiver end feeds back the loss event rate p and round trip time
   RTT to the sender with a feedback packet.

      Pl= Npl/Nl+Ns (20)

      Ps= Nps/Nl+Ns (21)

6.2. Sender Behavior When a Feedback Packet Is Received

   The slow-start begins in the exponential growth phase initially with
   a congestion window size (cwnd) of 1 or 2 segments and increases it
   by 1 Segment Size (SS) for each ACK received. This behavior
   effectively doubles the window size each round trip of the network.
   This behavior continues until the congestion window size (cwnd)
   reaches the size of the receiver??s advertised window or until a loss
   occurs. When a loss occurs, half of the current cwnd is saved as a
   Slow Start Threshold (SSThresh) and slow start begins again from its
   initial cwnd. Once the cwnd reaches the SSThresh, the WMRC scheme
   goes into congestion avoidance mode where each ACK increases the cwnd
   by SS*SS/cwnd. This results in a linear increase of the cwnd.
   
   The algorithm of WMRC will decrease the sender??s sending rate when
   high packet loss rates are reported, expecting the loss rates to drop

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   quickly (assuming the multimedia stream shares the network bandwidth
   with other TCP-friendly traffic). We propose the following sending
   strategy of the WMRC algorithm at the sender end.

   When the sender receives the feedback packet, it calculates the
   sending rate with the packet loss rate p and round trip time RTT
   based on TFRC[1] algorithm. The calculated rate may not suit the
   current networks state due to the change of round trip time. If the
   calculated rate is 1.5 times more than the current send rate within
   this time interval, the packet loss rate rises to a high level and
   the sender controls rate based on MIMD (a, b) (described below)
   congestion control algorithm. Otherwise, the rate control mechanism
   of TFRC will decrease the source??s sending rate.

   The MIMD(a, b) congestion control algorithm can be briefly described
   as: a and b are the parameters of MIMD(a, b) model to increase or
   decrease the sending rate respectively, and 0<a, b<1. Let R denote
   the current sending rate, if no packet losses occurred in this
   control period, the sending rate will be increased to (a+1)*R,
   otherwise, decreased to (1-b)*R.

   The whole adaptive rate control scheme is described in Algorithm 1.
   -------------------------------------------------------------------
   Algorithm 1: The adaptive rate control scheme
   -------------------------------------------------------------------
   1. Initialization
   2. Updating Nps, Npl
   3. For {receive a ACK packet} do
   4. Compute Dec by equations (18) and (1)-(4)
   5. If Dec>Dth
   6. The current packet loss is the congestion loss, and updates the
      packet loss rate P
   7. Else then
   8. The current packet loss is the wireless loss, and increase MAC
      retransmission count
   9. End else
   10. End if
   11. At a certain time interval, updating RTT and P, and compute Rate
       by equation (19)
   12. If no packet loss
   13. The next transmission rate Rn=1.2 Rcurrent
   14. Else then
   15. If |Rcurrent-Rate|<=0.5 Rcurrent
   16. The next transmission rate Rn = Rate
   17. Else
   18. The next transmission rate Rn =0.9 Rcurrent

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   19. End else
   20. End if
   21. End else
   22. End if
   23. End for
   -------------------------------------------------------------------

7. Security Considerations
 
   WMRC is not a transport protocol in its own right, but a congestion
   control mechanism that is intended to be used in conjunction with a
   transport protocol. Therefore security primarily needs to be
   considered in the context of a specific transport protocol and its
   authentication mechanisms. Congestion control mechanisms can
   potentially be exploited to create denial of service. This may occur
   through spoofed feedback. Thus any transport protocol that uses WMRC
   should take care to ensure that feedback is only accepted from the
   receiver of the data. The precise mechanism to achieve this will
   however depend on the transport protocol itself.

   In addition, congestion control mechanisms may potentially be
   manipulated by a greedy receiver that wishes to receive more than its
   fair share of network bandwidth. A receiver might do this by claiming
   to have received packets that in fact were lost due to congestion.
   Possible defenses against such a receiver would normally include some
   form of nonce that the receiver must feed back to the sender to prove
   receipt. However, the details of such a nonce would depend on the
   transport protocol, and in particular on whether the transport
   protocol is reliable or unreliable.

   We expect that protocols incorporating large/small packet with WMRC
   will also want to incorporate feedback from the receiver to the
   sender using packet loss discrimination. The packet loss
   discrimination is a modification to TFRC that distinguishes the loss
   packets from congestion loss or wireless random error.

8. IANA Considerations

   There are no IANA actions required for this document.

9. Conclusions

   This document presents a wireless channel model based rate control
   scheme WMRC for wireless multimedia transmission control over hybrid
   networks. This scheme can detect the network status and differentiate
   packet loss types (wireless or congestion losses) by means of a

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   wireless channel model and a special packet sending scheme with
   different packet sizes. It can adapt to the dynamic change of the
   networks and control the sending rate effectively. Theoretical
   analysis and detailed implementation of the proposed scheme are given.
   At the same time, it should be noted that due to the use of large and
   small packets alternately in the proposed algorithm, the application
   layer needs to pack data into two different-size packets for
   transmission, which may increase the overhead of packing process.

10. References
 
10.1. Informative References
 
   [1] Floyd, S., Handley, M., Padhye, J., and J. Widmer, "TCP
   Friendly Rate Control (TFRC): Protocol Specification", RFC 5348,
   September 2008.

   [2] Song Cen, et al, ??End-to-End Differentiation of Congestion and
   Wireless Losses,?? IEEE/ACM Transactions on Networking, vol. 11,
   no. 5, pp. 703-717, October 2003.

   [3] Min Kyu Park, Kue-Hwan Sihn, Jun Ho Jeong, ??A Statistical
   Method of Packet Loss Type Discrimination in Wired-Wireless
   Networks,?? Proc. IEEE CCNC, 2006, pp. 458-462.

   [4] C. L. Lee, et al, ??On the Use of Loss History for Performance
   Improvement of TCP over Wireless Networks??, IEICE Trans.
   Commun. , vol. E85-B, no. 11, pp. 2457-2467, 2002.

   [5] H.-J. Lee, H.-J. Byun, and J.-T. Lim, ??TCP-friendly congestion
   control for streaming real-time applications over wireless
   networks??, IET Commun., vol. 2, no. 1, pp. 159?C163, 2008.

   [6] S. Floyd and J. Padhye, ??Equation-Based congestion control for
   unicast applications??, Proc. ACM SIGCOMM??00, 2000, pp. 43-56.

   [7] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
   "RTP: A Transport Protocol for Real-Time Applications", RFC
   1889, January 1996.

   [8] Yu-ning Dong and Meng-yue Chen, ??Real time video transmission
   control in wireless-wired IP networks??, Proc. IEEE Wireless
   Communications and Networking Conference (WCNC2007), Hong Kong,
   Mar. 2007, pp. 3687-3691.

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11. Acknowledgments

   The authors would like to acknowledge feedback and discussions on
   equation-based congestion control with a wide range of people,
   including members of the Wireless Communication Research Group and
   the End-to-End Research Group. Thanks are given to the National
   Natural Science Foundation of China (No.60972038), the Jiangsu
   Province Universities Natural Science Research Key Grant Project
   (07KJA51006), and Jiangsu Province Graduate Innovative Research Plan
   (CX09B_149Z) for their financial support.

Authors?? Addresses

   Yu-ning Dong
   Nanjing University of Posts and Telecommunications PO Box 166
   66 New Mo-fan-ma-lu Road, Nanjing, Jiangsu, 210003
   China

   Email: [email-masked]

   Hai-tao Zhao
   Nanjing University of Posts and Telecommunications PO Box 54
   66 New Mo-fan-ma-lu Road, Nanjing, Jiangsu, 210003
   China

   Email: [email-masked]

Dong, et. al.            Expires May 15, 2011	               [Page 14]
 
>From [email-masked]  Thu Nov 18 15:10:19 2010
From: [email-masked] (Michael Welzl)
Date: Thu Nov 18 14:59:28 2010
Subject: [Iccrg] Agenda of the (very short) ICCRG meeting @ PFLDNeT
Message-ID: <[email-masked]>

Hi all,

The (tentative) agenda of PFLDNeT is now available - and with it, the  
agenda of the ICCRG meeting which we'll simply run as a session in the  
PFLDNeT context this time because it's so short:
http://pfld.net/2010/technical.php

It would be nice to see some of you there!
(and we can still fit a talk or two on the agenda, just drop us a note)

Cheers,
Michael

From: [email-masked] (Scheffenegger, Richard)
Date: Mon Nov 15 14:04:36 2010
Subject: [Iccrg] Re: [Tmrg] Testbed Study on IW10 vs IW3
In-Reply-To: <[email-masked]>
References: <[email-masked]><[email-masked]> <[email-masked]>
Message-ID: <[email-masked]>

Hi Jerry,

Just one though regarding queue size: 

IMHO, it's probably rare to go from 1Gbps down to 64kbps. I would think
that the link from the provider side to the home user side will go
though consecutive steps of decreasing bottleneck bandwidth. Even if
some intermediate links have higher capacity again, that should in
effect alleviate the burst load on the ultimate bottleneck queue
somewhat. In such a scenario, you may get away with only 10 packets
queue depth at each step, if at least 4 consecutive smaller bottlenecks
are on the path from the sender to the receiver...

Also, Lars seems to do some investigation on home gateways:

>>> On Fri, Jul 9, 2010 at 12:15 AM, Lars Eggert
>> <[email-masked]<mailto:[email-masked]>> wrote:
>>> Hi,
>>>
>>> a quick status update. We now have received over 100
>> donated home gateways, plus a DSLAM. The students are on
>> their summer break, after which we'll start running a
>> significantly expanded set of tests over this much larger
>> population of devices.
>>>
>>> Many of yo have donated boxes and suggested more
>> experiments and better ways of performing our current tests -
>> thank you!
>>>
>>> In case you are attending IETF-78 in Maastricht and would like to
>>> donate a home gateway, simply bring it. (Or contact me now for
>>> shipping details; no cost to you.)
>>>
>>> We're especially interested in devices from outside the EU
>> and North America, or any other model we may not have yet
>> (see
>> http://fit.nokia.com/lars/tmp/2010-hgw-study-devices.txt).
>> And we're still lacking a CMTS for testing cable modems...
>>> 

Perhaps this team could provide some realistic ranges of the queue
depths (down- and upstream) for very small bottleneck pipes too?

Best regards,

Richard Scheffenegger

> -----Original Message-----
> From: Jerry Chu [mailto:[email-masked]] 
> Sent: Montag, 15. November 2010 07:53
> To: Stefan Hirschmann
> Cc: tmrg; [email-masked] list
> Subject: [Iccrg] Re: [Tmrg] Testbed Study on IW10 vs IW3
> 
> On Sun, Nov 14, 2010 at 2:48 PM, Stefan Hirschmann 
> <[email-masked]> wrote:
> > Jerry Chu wrote:
> >>
> >> I believe TCPM is a superset of the two lists above. If 
> not, please 
> >> find my presentation at the TCPM meeting at
> >>
> >> http://www.ietf.org/proceedings/79/slides/tcpm-0.pdf
> >>
> >> Our testbed data can be found at
> >> http://research.csc.ncsu.edu/netsrv/?q=content/iw10
> >
> > Thanks for your data. I missed information about the maximum queue 
> > size in this presentations. Can you add this informations. 
> I believe 
> > the maximum
> 
> See the fourth bullet on slide7. (Did you bother to read the 
> slides at all?)
> 
> > buffer size is for Slow-Start simulations really important.
> 
> Sure. We've tried smaller buffer size. See slides from the 
> previous IETF for the case of 20Mbps. For 64Kbps the numbers 
> turned sour quickly when max qlen goes below 40 packets. We 
> can rerun some numbers if you think it's worthwhile (and you 
> do have the time to read the results). We found out when max 
> qlen was too small not only the numbers were horrible for 
> both IW3 and IW10 but also extremely unstable.
> 
> Jerry
> 
> >
> > Stefan
> >
> 
> _______________________________________________
> Iccrg mailing list
> [email-masked]
> http://oakham.cs.ucl.ac.uk/mailman/listinfo/iccrg
> 

From: [email-masked] (Jerry Chu)
Date: Mon Nov 15 06:42:23 2010
Subject: [Iccrg] Re: [Tmrg] Testbed Study on IW10 vs IW3
In-Reply-To: <[email-masked]>
References: <[email-masked]> <[email-masked]>
Message-ID: <[email-masked]>

On Sun, Nov 14, 2010 at 2:48 PM, Stefan Hirschmann <[email-masked]> wrote:
> Jerry Chu wrote:
>>
>> I believe TCPM is a superset of the two lists above. If not, please find
>> my
>> presentation at the TCPM meeting at
>>
>> http://www.ietf.org/proceedings/79/slides/tcpm-0.pdf
>>
>> Our testbed data can be found at
>> http://research.csc.ncsu.edu/netsrv/?q=content/iw10
>
> Thanks for your data. I missed information about the maximum queue size in
> this presentations. Can you add this informations. I believe the maximum

See the fourth bullet on slide7. (Did you bother to read the slides at all?)

> buffer size is for Slow-Start simulations really important.

Sure. We've tried smaller buffer size. See slides from the previous IETF for
the case of 20Mbps. For 64Kbps the numbers turned sour quickly when max
qlen goes below 40 packets. We can rerun some numbers if you think it's
worthwhile (and you do have the time to read the results). We found out when
max qlen was too small not only the numbers were horrible for both IW3 and
IW10 but also extremely unstable.

Jerry

>
> Stefan
>

From: [email-masked] (Stefan Hirschmann)
Date: Sun Nov 14 22:38:09 2010
Subject: [Iccrg] Re: [Tmrg] Testbed Study on IW10 vs IW3
In-Reply-To: <[email-masked]>
References: <[email-masked]>
Message-ID: <[email-masked]>

Jerry Chu wrote:
> I believe TCPM is a superset of the two lists above. If not, please find my
> presentation at the TCPM meeting at
> 
> http://www.ietf.org/proceedings/79/slides/tcpm-0.pdf
> 
> Our testbed data can be found at
> http://research.csc.ncsu.edu/netsrv/?q=content/iw10

Thanks for your data. I missed information about the maximum queue size 
in this presentations. Can you add this informations. I believe the 
maximum buffer size is for Slow-Start simulations really important.

Stefan

From: [email-masked] (Colin Perkins)
Date: Sun Nov 14 03:46:43 2010
Subject: [Iccrg] CFP: 14th IEEE Global Internet Symposium
Message-ID: <[email-masked]>

14th IEEE Global Internet Symposium

Shanghai, China
15 April 2011

http://www.dcs.gla.ac.uk/conferences/gi2011/

*** Call for Papers ***

The 14th IEEE Global Internet Symposium will be held in conjunction with IEEE Infocom 2011 in Shanghai, China, on 15 April 2011. All relevant dates, location, and travel information are available from the IEEE Infocom 2011 website (http://www.ieee-infocom.org/2011/).

The IEEE Global Internet Symposium aims to provide a forum for researchers and practitioners to present and discuss advances in Internet-related technologies. The focus of the symposium is on experimental systems and on emerging future Internet technologies, and especially on scaling such systems to a global scale. The Program Committee encourages original submissions describing promising work in progress, speculations about the future of the Internet, and progressive position papers (which should be clearly marked as such).

The proceedings of the 14th IEEE Global Internet Symposium will be published on-line through the IEEE Xplore Digital Library after IEEE Infocom 2011 concludes.

*** Topics ***

Authors are invited to submit papers on issues, especially scaling issues, related to current and future Internet technology.  Topics  of interest include, but not limited to, the following: 

- Novel applications and new paradigms
- Next-generation network architectures
- Handling Internet dynamics/heterogeneity (by applications and/or the network)
- P2P networking and overlay networks
- Large-scale distributed Internet applications 
- Privacy and/or security issues in the Internet 
- Anomaly, intrusion and attack detection 
- Routing (unicast, multicast, anycast, etc.) 
- Flow management (fairness/sharing, congestion control, differentiated services, etc.) 
- The Internet and wireless/mobile devices, as well as intermittent connectivity 
- Traffic measurement, analysis, modeling, and visualization 
- Provisioning, monitoring, and management of IP services 
- Content networking (caching, content distribution, content routing, content services, load balancing, etc.) 

*** Important Dates ***

Paper submission: 20 December 2010
Notification of acceptance: 4 February 2011
Symposium: 15 April 2011 

*** Submission Instructions ***

Submitted manuscripts must be formatted in standard IEEE camera-ready format (double-column, 10-pt font) and must be submitted via EDAS (http://edas.info/newPaper.php?c=9011) as PDF files formatted for 8.5x11-inch paper. The manuscripts must be no longer than 6 pages. The Programme Committee reserves the right to not review papers that violate these formatting rules.

Submitted papers must not have been previously published, or be under consideration for publication elsewhere. All submitted papers will be reviewed and judged on originality, technical correctness, relevance, and quality of presentation. All accepted papers must be presented at the symposium by one of the authors.

*** Technical Programme Committee Co-chairs ***

Dina Papagiannaki (Intel Labs, Pittsburgh)
Colin Perkins (University of Glasgow)

The Technical Programme Committee Co-chairs can be reached via email to <[email-masked]>.

*** Technical Programme Committee ***

Nabeel Ahmed (MIT)
Marcelo Bagnulo (University Carlos III of Madrid)
Saleem Bhatti (University of St Andrews)
Jaideep Chandrashekar (Intel)
Constantine Dovrolis (Georgia Tech)
Lars Eggert (Nokia Research Center & Helsinki University of Technology)
Ted Faber (USC/ISI)
Gorry Fairhurst (University of Aberdeen)
Christos Gkantsidis (Microsoft Research Cambridge)
Sergey Gorinsky (IMDEA Networks)
Janardhan Iyengar (Franklin and Marshall College)
Joerg Liebeherr (University of Toronto)
Yan Luo (University of Massachusetts Lowell)
Evangelos Markatos (Forth)
Dan Massey (Colorado State University)
Rodney Van Meter (Keio University)
Laurent Mathy (Lancaster University)
Joerg Ott (Helsinki University of Technology)
Christos Papadopoulos (Colorado State University)
Craig Partridge (BBN Technologies)
Peter Reiher (UCLA)
Georgos Siganos (Telefonica Research)
Joe Touch (USC/ISI)
Joerg Widmer (DOCOMO Euro-Labs)
Lars Wolf (Technische Universitaet Braunschweig)
Tilman Wolf (University of Massachusetts)

-/-

From: [email-masked] (Lars Eggert)
Date: Sun Nov  7 04:32:56 2010
Subject: [Iccrg] review of high-speed proposals?
In-Reply-To: <[email-masked]>
References: <[email-masked]> <[email-masked]> <[email-masked]> <[email-masked]> <[email-masked]> <AANLkTi=SpFhoE9-DK1+hk=[email-masked]> <[email-masked]>
Message-ID: <[email-masked]>

Hi,

to reiterate, in order to keep the ball rolling: the IETF cannot start the process of publishing these IDs as Experimental until they have actually been submitted to the WG. Author action is required before anything can happen in the TCPM WG...

Lars
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>From [email-masked]  Thu Nov 11 11:15:16 2010
From: [email-masked] (Jerry Chu)
Date: Thu Nov 11 11:05:32 2010
Subject: [Iccrg] Testbed Study on IW10 vs IW3
Message-ID: <[email-masked]>

I believe TCPM is a superset of the two lists above. If not, please find my
presentation at the TCPM meeting at

http://www.ietf.org/proceedings/79/slides/tcpm-0.pdf

Our testbed data can be found at
http://research.csc.ncsu.edu/netsrv/?q=content/iw10

FYI,

Jerry

From: [email-masked] (Michael Welzl)
Date: Mon Nov  1 09:13:31 2010
Subject: [Iccrg] ICCRG meeting @ Pfldnet
In-Reply-To: <[email-masked]>
References: <[email-masked]> <[email-masked]>
Message-ID: <[email-masked]>

Hi,

This is to remind you that we'll have an ICCRG meeting co-located with  
Pfldnet:
http://pfld.net/2010/

Please send Wes and me a note (title, expected duration of the talk)  
if you want to give a presentation. Thanks!

Cheers,
Michael

On Aug 30, 2010, at 11:56 AM, Michael Welzl wrote:

> Dear all,
>
> I would like to remind you of this opportunity for submitting a paper.
>
> At the same time, I'd like to give an update on upcoming ICCRG  
> meetings:
> - we will not meet at the upcoming Beijing IETF
> - we will, as announced previously and below, meet at this workshop.  
> If you have something you'd like to present at this meeting, send me  
> and Wes an email please, stating the title and expected duration of  
> your talk.
>
> Cheers,
> Michael
>
>
> On Aug 15, 2010, at 2:39 PM, Michael Welzl wrote:
>
>> Hi all,
>>
>> Please consider the CFP below, definitely of interest to the group.  
>> As usual we'll co-locate a meeting with this,
>> this time on 29. November 2010.
>>
>> Cheers,
>> Michael
>>
>>
>> CALL FOR PAPERS
>>
>> PFLDNeT 2010
>>
>> The 8th International Workshop on Protocols for Future,
>> Large-Scale and Diverse Network Transports (PFLDNeT)
>>
>> Franklin & Marshall College, Lancaster, Pennsylvania, USA
>> November 28-29, 2010
>> Web page: http://pfld.net/2010
>>
>>
>> Scope:
>>
>> The Internet continues to evolve along several dimensions, allowing
>> more and more end systems to communicate in increasingly diverse  
>> ways.
>> At one end of the performance spectrum, the Internet protocols  
>> provide
>> communication facilities for extremely-high-speed special-use
>> networks. At the other end of the performance spectrum, the Internet
>> contains very low-power and low-bandwidth networks that cater to
>> infrequent, bursty communication. Enabling efficient and high-
>> performance end-to-end communication across such a diverse
>> internetwork is a difficult problem, which is not solved by current
>> transport layer protocols.  The need to support an application base
>> that grows more and more dissimilar adds additional challenges.
>>
>> The 8th International Workshop on Protocols for Future, Large-scale &
>> Diverse Network Transports (PFLDNeT) brings together researchers and
>> practitioners from all continents to exchange their ideas and
>> experiences in the area of transport issues for modern communication
>> networks. The workshop provides theorists, experimentalists and
>> technologists with a focused, highly interactive opportunity to
>> present, discuss and exchange experience on leading research,
>> development and future directions in transport and application
>> protocols for networks that are increasingly growing in size,
>> heterogeneity and dynamicity of interaction.
>>
>> PFLDNeT 2010 solicits papers that further the research on end-to-end
>> communication protocols for todays and tomorrows Internet in all its
>> diversity along the continuum from specialized grid networks, optical
>> transports, wireless connections, to lossy and low-power networks.  A
>> specific focus of the workshop lies on transport protocols for the
>> efficient end-to-end transfer of data for a diverse set of
>> applications and application-layer protocols.
>>
>> Now approaching its eighth instantiation, the PFLDneT workshop has
>> broadened its focus over the years from protocols targeted at  
>> specific
>> fast, long-distance networks (the original expansion of the PFLDneT
>> acronym) into a venue where all kinds of new ideas relating to
>> end-to-end transport protocols for diverse network scenarios are  
>> being
>> discussed first.
>>
>> The previous International Workshops on Protocols for Fast, Long-
>> Distance Networks held at CERN (2003), Argonne (2004), Lyon (2005),
>> Nara (2006), Marina del Rey (2007), Manchester (2008), and Tokyo
>> (2009) were very successful in bringing together many researchers  
>> from
>> all over the world, including North America, Europe and Asia, who are
>> working on these problems.  PFLDNeT 2010 will continue this  
>> tradition,
>> and provide a perfect forum for researchers in this area to exchange
>> ideas and experience.
>>
>> As in previous years, a meeting of the IRTF Internet Congestion
>> Control Research Group (ICCRG) will be co-located with PFLDNeT, on
>> November 29, 2010.
>>
>>
>> Important Dates and Relevant Event Information:
>>
>> Abstract submission: September 20, 2010
>> Position paper submission: September 27, 2010
>> Notification of acceptance: October 25, 2010
>> Final camera ready submission: November 14, 2010
>> Workshop: November 28-29, 2010
>> IRTF ICCRG meeting (co-located): November 29, 2010
>>
>> Note: PFLDNeT 2010 will be immediately before CoNEXT 2010, and its  
>> venue
>> is located only a short drive or train ride from Philadelphia,  
>> making it
>> easy to attend both events in one trip.
>>
>>
>> Topics:
>>
>> PFLDNeT 2010 covers all aspects related to transport protocols for  
>> the
>> current and future Internet, including, but not limited to:
>>
>> - Transport protocol development
>> - Enhancements to TCP and other transports
>> - Innovative congestion control mechanisms
>> - Novel data transport protocols designed for new networks and  
>> applications
>> - Transport services for data center networks and grids
>> - Transport services for wireless and sensor networks
>> - Explicit signaling protocols: optimization criteria and  
>> deployment strategies
>> - Pacing and shaping of traffic
>> - Parallel transfers and multi-streaming
>> - Performance evaluation
>> - Modeling and simulation-based results
>> - Interaction of transport protocols and network equipment
>> - Experiments on real networks and live measurements
>> - Transport protocol benchmarking
>> - Transport over optical networks
>> - Transport implementation and hardware issues
>> - End system performance
>> - Data replication and striping
>> - Applications with demanding or unusual network performance  
>> requirements
>> - Bulk-data transfer applications
>> - Quality-of-service and scalability issues
>> - Multicast
>>
>>
>> Workshop Organizers:
>>
>> Program Committee Chairs:
>> Bryan Ford, Yale University, USA
>> Injong Rhee, North Carolina State University, USA
>>
>> Steering Committee:
>> Lachlan Andrew, Swinburne University of Technology, Australia
>> Lars Eggert, Nokia Research Center, Finland
>> Richard Hughes-Jones, Univ. of Manchester, UK
>> Katsushi Kobayashi, AIST, Japan
>> Doug Leith, Hamilton Institute, Ireland
>> Injong Rhee, North Carolina State University, USA
>> Pascale Vicat-Blanc, INRIA, France
>> Michael Welzl, University of Oslo, Norway
>>
>> Technical Program Committee (preliminary):
>> Scott Brim, Cisco, USA
>> Dirceu Cavendish, Kyushu Institute of Technology, Japan
>> Stuart Cheshire, Apple, USA
>> Larry Dunn, University of Minnesota, USA
>> Lars Eggert, Nokia Research Center, Finland
>> Ted Faber, USC/ISI, USA
>> Saikat Guha, Microsoft Research, India
>> Sangtae Ha, Princeton University, USA
>> Janardhan Iyengar, Franklin & Marshall College, USA
>> Katsushi Kobayashi, AIST, Japan
>> Aleksandar Kuzmanovic, Northwestern University, USA
>> Preethi Natarajan, Cisco, USA
>> Narasimha Reddy, Texas A&M University, USA
>> Medy Sanadidi, UCLA, USA
>> Pasi Sarolahti, Nokia Research Center, Finland
>> Michael Scharf, Alcatel-Lucent, Germany
>> Hideyuki Shimonishi, NEC, Japan
>> Murari Sridharan, Microsoft, USA
>> Joe Touch, USC/ISI, USA
>> Michael Welzl, University of Oslo, Norway
>> Lisong Xu, University of Nebraska, USA
>>
>> Local Arrangements:
>> Janardhan Iyengar, Franklin & Marshall College, USA
>>
>>
>>
>> _______________________________________________
>> Iccrg mailing list
>> [email-masked]
>> http://oakham.cs.ucl.ac.uk/mailman/listinfo/iccrg
>
>
> _______________________________________________
> Iccrg mailing list
> [email-masked]
> http://oakham.cs.ucl.ac.uk/mailman/listinfo/iccrg