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NADA: A Unified Congestion Control Scheme for Real-Time Media
draft-ietf-rmcat-nada-00

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This is an older version of an Internet-Draft that was ultimately published as RFC 8698.
Authors Xiaoqing Zhu , Rong Pan , Michael A. Ramalho , Sergio Mena de la Cruz , Charles Ganzhorn , Paul Jones , Stefano D'Aronco
Last updated 2015-04-29
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draft-ietf-rmcat-nada-00
Network Working Group                                     X. Zhu, R. Pan
Internet Draft                                    M. A. Ramalho, S. Mena
Intended Status: Informational                  C. Ganzhorn, P. E. Jones
Expires: October 29, 2015                                  Cisco Systems
                                                            S. De Aronco
                                Ecole Polytechnique Federale de Lausanne
                                                          April 28, 2015

     NADA: A Unified Congestion Control Scheme for Real-Time Media
                        draft-ietf-rmcat-nada-00

Abstract

   Network-Assisted Dynamic Adaptation (NADA) is a novel congestion
   control scheme for interactive real-time media applications, such as
   video conferencing. In NADA, the sender regulates its sending rate
   based on either implicit or explicit congestion signaling in a
   consistent manner. As one example of explicit signaling, NADA can
   benefit from explicit congestion notification (ECN) markings from
   network nodes. It also maintains consistent sender behavior in the
   absence of explicit signaling by reacting to queuing delay and packet
   loss.  

   This document describes the overall system architecture for NADA, as
   well as recommended behavior at the sender and the receiver.

Status of this Memo

   This Internet-Draft is submitted to IETF in full conformance with the
   provisions of BCP 78 and BCP 79.

   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/1id-abstracts.html

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html
 

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Copyright and License Notice

   Copyright (c) 2012 IETF Trust and the persons identified as the
   document authors. All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document. Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document. Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.

Table of Contents

   1. Introduction  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   2. Terminology . . . . . . . . . . . . . . . . . . . . . . . . . .  3
   3. System Model  . . . . . . . . . . . . . . . . . . . . . . . . .  3
   4. NADA Receiver Behavior  . . . . . . . . . . . . . . . . . . . .  4
     4.1 Estimation of one-way delay and queuing delay  . . . . . . .  4
     4.2 Estimation of packet loss/marking ratio  . . . . . . . . . .  5
     4.3 Non-linear warping of delay  . . . . . . . . . . . . . . . .  6
     4.4 Aggregating congestion signals . . . . . . . . . . . . . . .  7
     4.5 Estimating receiving rate  . . . . . . . . . . . . . . . . .  7
     4.6 Sending periodic feedback  . . . . . . . . . . . . . . . . .  7
     4.7 Discussions on delay metrics . . . . . . . . . . . . . . . .  8
   5. NADA Sender Behavior  . . . . . . . . . . . . . . . . . . . . .  9
     5.1 Reference rate calculation . . . . . . . . . . . . . . . . . 10
       5.1.1 Accelerated ramp up  . . . . . . . . . . . . . . . . . . 10
       5.1.2. Gradual rate update . . . . . . . . . . . . . . . . . . 11
     5.2 Video encoder rate control . . . . . . . . . . . . . . . . . 12
     5.3 Rate shaping buffer  . . . . . . . . . . . . . . . . . . . . 12
     5.4 Adjusting video target rate and sending rate . . . . . . . . 12
   6. Incremental Deployment  . . . . . . . . . . . . . . . . . . . . 13
   7. Implementation Status . . . . . . . . . . . . . . . . . . . . . 13
   8. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 14
   9. References  . . . . . . . . . . . . . . . . . . . . . . . . . . 14
     9.1  Normative References  . . . . . . . . . . . . . . . . . . . 14
     9.2  Informative References  . . . . . . . . . . . . . . . . . . 14
   Appendix A. Network Node Operations  . . . . . . . . . . . . . . . 15
     A.1 Default behavior of drop tail  . . . . . . . . . . . . . . . 16
     A.2 ECN marking  . . . . . . . . . . . . . . . . . . . . . . . . 16
     A.3 PCN marking  . . . . . . . . . . . . . . . . . . . . . . . . 16
   Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 17

 

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1. Introduction

   Interactive real-time media applications introduce a unique set of
   challenges for congestion control. Unlike TCP, the mechanism used for
   real-time media needs to adapt quickly to instantaneous bandwidth
   changes, accommodate fluctuations in the output of video encoder rate
   control, and cause low queuing delay over the network. An ideal
   scheme should also make effective use of all types of congestion
   signals, including packet loss, queuing delay, and explicit
   congestion notification (ECN) [RFC3168] markings. 

   Based on the above considerations, this document describes a scheme
   called network-assisted dynamic adaptation (NADA). The NADA design
   benefits from explicit congestion control signals (e.g., ECN
   markings) from the network, yet also operates when only implicit
   congestion indicators (delay and/or loss) are available. In addition,
   it supports weighted bandwidth sharing among competing video flows. 

   This documentation describes the overall system architecture,
   recommended designs at the sender and receiver, as well as expected
   network node operations. The signaling mechanism consists of standard
   RTP timestamp [RFC3550] and standard RTCP feedback reports. 

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. System Model

   The overall system consists of the following elements: 

        * Source media stream, in the form of consecutive raw video
        frames and audio samples;

        * Media encoder with rate control capabilities. It takes the
        source media stream and encodes it to an RTP stream at a target
        bit rate R_v. Note that the actual output rate from the encoder
        R_o may fluctuate around the target R_v. Also, the encoder can
        only change its rate at rather coarse time intervals, e.g., once
        every 0.5 seconds.  

        * RTP sender, responsible for calculating the target bit rate
        R_n based on network congestion indicators (delay, loss, or ECN
        marking reports from the receiver), for updating the video
        encoder with a new target rate R_v, and for regulating the
 

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        actual sending rate R_s accordingly. A rate shaping buffer is
        employed to absorb the instantaneous difference between video
        encoder output rate R_v and sending rate R_s. The buffer size
        L_s, together with R_n, influences the calculation of actual
        sending rate R_s and video encoder target rate R_v. The RTP
        sender also generates RTP timestamp in outgoing packets.

        * RTP receiver, responsible for measuring and estimating end-to-
        end delay based on sender RTP timestamp. In the presence of
        packet loss and ECN markings, it keeps track of packet loss and
        ECN marking ratios. It calculates the equivalent delay x_n that
        accounts for queuing delay, ECN marking, and packet loss, as
        well as the derivative (i.e., rate of change) of this congestion
        signal as x'_n. The receiver feeds both pieces of information
        (x_n and x'_n) back to the sender via periodic RTCP reports. 

        * Network node, with several modes of operation. The system can
        work with the default behavior of a simple drop tail queue.  It
        can also benefit from advanced AQM features such as RED-based
        ECN marking, and PCN marking using a token bucket algorithm.
        Note that network node operation is out of scope for the design
        of NADA. 

In the following, we will elaborate on the respective operations at the
NADA receiver and sender.

4. NADA Receiver Behavior

The receiver continuously monitors end-to-end per-packet statistics in
terms of delay, loss, and/or ECN marking ratios. It then aggregates all
forms of congestion indicators into the form of an equivalent delay and
periodically reports this back to the sender. In addition, the receiver
tracks the receiving rate of the flow and includes that in the feedback
message.

4.1 Estimation of one-way delay and queuing delay

The delay estimation process in NADA follows a similar approach as in
earlier delay-based congestion control schemes, such as LEDBAT
[RFC6817]. NADA estimates the forward delay as having a constant base
delay component plus a time varying queuing delay component. The base
delay is estimated as the minimum value of one-way delay observed over a
relatively long period (e.g., tens of minutes), whereas the individual
queuing delay value is taken to be the difference between one-way delay
and base delay.  

 

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In mathematical terms, for packet n arriving at the receiver, one-way
delay is calculated as:  

                         z_n = t_r,n - t_s,n, 

where t_s,n and t_r,n are sender and receiver timestamps, respectively.
A real-world implementation should also properly handle practical issues
such as wrap-around in the value of z_n, which are omitted from the
above simple expression for brevity. 

The base delay, d_f, is estimated as the minimum value of previously
observed z_n's over a relatively long period. This assumes that the
drift between sending and receiving clocks remains bounded by a small
value.

Correspondingly, the queuing delay experienced by the packet n is
estimated as: 

                           d_n = z_n - d_f. 

The individual sample values of queuing delay should be further filtered
against various non-congestion-induced noise, such as spikes due to
processing "hiccup" at the network nodes. We denote the resulting
queuing delay value as d_hat_n. 

Our current implementation employs a simple 5-point median filter over
per-packet queuing delay estimates, followed by an exponential smoothing
filter. We have found such relatively simple treatment to suffice in
guarding against processing delay outliers observed in wired
connections. For wireless connections with a higher packet delay
variation (PDV), more sophisticated techniques on de-noising, outlier
rejection, and trend analysis may be needed. 

Like other delay-based congestion control schemes, performance of NADA
depends on the accuracy of its delay measurement and estimation module.
Appendix A in [RFC6817] provides an extensive discussion on this aspect.

4.2 Estimation of packet loss/marking ratio

The receiver detects packet losses via gaps in the RTP sequence numbers
of received packets. It then calculates instantaneous packet loss ratio
as the ratio between the number of missing packets over the number of
total transmitted packets in the given time window (e.g., during the
most recent 500ms). This instantaneous value is passed over an
exponential smoothing filter, and the filtered result is reported back
to the sender as the observed packet loss ratio p_L. 

 

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We note that more sophisticated methods in packet loss ratio
calculation, such as that adopted by TFRC [Floyd-CCR00], will likely be
beneficial. These alternatives are currently under investigation. 

Estimation of packet marking ratio p_M, when ECN is enabled at
bottleneck network nodes along the path, will follow the same procedure
as above. Here it is assumed that ECN marking information at the IP
header are somehow passed along to the transport layer by the receiving
endpoint.

4.3 Non-linear warping of delay

In order for a delay-based flow to hold its ground and sustain a
reasonable share of bandwidth in the presence of a loss-based flow
(e.g., loss-based TCP), it is important to distinguish between different
levels of observed queuing delay. For instance, a moderate queuing delay
value below 100ms is likely self-inflicted or induced by other delay-
based flows, whereas a high queuing delay value of several hundreds of
milliseconds may indicate the presence of a loss-based flow that does
not refrain from increased delay. 

Inspired by the delay-adaptive congestion window backoff policy in
[Budzisz-TON11] -- the work by itself is a window-based congestion
control scheme with fair coexistence with TCP -- we devise the following
non-linear warping of estimated queuing delay value: 

        d_tilde_n = (d_hat_n),  if d_hat_n < d_th; 

                         (d_max - d_hat_n)^4
        d_tilde_n = d_th --------------------, if  d_th<d_hat_n<d_max; 
                          (d_max - d_th)^4

        d_tilde_n = 0,   if d_hat_n > d_max. 

Here, the queuing delay value is unchanged when it is below the first
threshold d_th; it is discounted following a non-linear curve when its
value falls between d_th and d_max; above d_max, the high queuing delay
value no longer counts toward congestion control.  

When queuing delay is in the range (0, d_th), NADA operates in pure
delay-based mode if no losses/markings are present. When queuing delay
exceeds d_max, NADA reacts to loss/marking only. In between d_th and
d_max, the sending rate will converge and stabilize at an operating
point with a fairly high queuing delay and non-zero packet loss ratio. 

In our current implementation d_th is chosen as 50ms and d_max is chosen
as 400ms. The impact of the choice of d_th and d_max will be
investigated in future work. 
 

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4.4 Aggregating congestion signals

The receiver aggregates all three forms of congestion signal in terms of
an equivalent delay: 

             x_n = d_tilde_n + p_M*d_M + p_L*d_L,      (1) 

where d_M is a prescribed fictitious delay value associated with ECN
markings (e.g., d_M = 200 ms), and d_L is a prescribed fictitious delay
value associated with packet losses (e.g., d_L = 1 second). By
introducing a large fictitious delay penalty for ECN marking and packet
loss, the proposed scheme leads to low end-to-end actual delay in the
presence of such events. 

While the value of d_M and d_L are fixed and predetermined in the
current design, a scheme for automatically tuning these values based on
desired bandwidth sharing behavior in the presence of other competing
loss-based flows (e.g., loss-based TCP) is being studied.

In the absence of ECN marking from the network, the value of x_n falls
back to the observed queuing delay d_n for packet n when queuing delay
is low and no packets are lost over a lightly congested path. In that
case the algorithm operates in purely delay-based mode. 

4.5 Estimating receiving rate

Estimation of receiving rate of the flow is fairly straightforward. NADA
maintains a recent observation window of 500ms, and simply divides the
total size of packets arriving during that window over the time span.
The receiving rate is denoted as R_r. 

4.6 Sending periodic feedback 

Periodically, the receiver feeds back a tuple of the most recent values
of <d_hat_n, x_n, x'_n, R_r> in RTCP feedback messages to aid the sender
in its calculation of target rate. The queuing delay value d_hat_n is
included along with the composite congestion signal x_n so that the
sender can decide whether the network is truly underutilized (see Sec.
6.1.1 Accelerated ramp-up).

The value of x'_n corresponds to the derivative (i.e., rate of change)
of the composite congestion signal:

                x_n - x_(n-k)
        x'_n = ---------------,           (2) 
                  delta

 

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where the interval between consecutive RTCP feedback messages is denoted
as delta. The packet indices corresponding to the current and previous
feedback are n and (n-k), respectively. 

The choice of target feedback interval needs to strike the right balance
between timely feedback and low RTCP feedback message counts. Through
simulation studies and frequency-domain analysis, it was determined that
a feedback interval below 250ms will not break up the feedback control
loop of the NADA congestion control algorithm. Thus, it is recommended
to use a target feed interval of 100ms. This will result in a feedback
bandwidth of 16Kbps with 200 bytes per feedback message, less than 0.1%
overhead for a 1Mbps flow.

4.7 Discussions on delay metrics

The current design works with relative one-way-delay (OWD) as the main
indication of congestion. The value of the relative OWD is obtained by
maintaining the minimum value of observed OWD over a relatively long
time horizon and subtract that out from the observed absolute OWD value.
Such an approach cancels out the fixed difference between the sender and
receiver clocks. It has been widely adopted by other delay-based
congestion control approaches such as LEDBAT [RFC6817]. As discussed in
[RFC6817], the time horizon for tracking the minimum OWD needs to be
chosen with care: it must be long enough for an opportunity to observe
the minimum OWD with zero queuing delay along the path, and sufficiently
short so as to timely reflect "true" changes in minimum OWD introduced
by route changes and other rare events.

The potential drawback in relying on relative OWD as the congestion
signal is that when multiple flows share the same bottleneck, the flow
arriving late at the network experiencing a non-empty queue may
mistakenly consider the standing queuing delay as part of the fixed path
propagation delay. This will lead to slightly unfair bandwidth sharing
among the flows. 

Alternatively, one could move the per-packet statistical handling to the
sender instead and use RTT in lieu of OWD, assuming that per-packet ACKs
are available. The main drawback of this latter approach is that the
scheme will be confused by congestion in the reverse direction.

Note that the choice of either delay metric (relative OWD vs. RTT)
involves no change in the proposed rate adaptation algorithm at the
sender. Therefore, comparing the pros and cons regarding which delay
metric to adopt can be kept as an orthogonal direction of
investigation.

 

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5. NADA Sender Behavior

Figure 1 provides a detailed view of the NADA sender. Upon receipt of an
RTCP report from the receiver, the NADA sender updates its calculation
of the reference rate R_n. It further adjusts both the target rate for
the live video encoder R_v and the sending rate R_s over the network
based on the updated value of R_n, as well as the size of the rate
shaping buffer.

In the following, we describe these modules in further detail, and
explain how they interact with each other.  

                    --------------------             
                    |                  |            
                    |  Reference Rate  | <---- RTCP report
                    |  Calculator      |  
                    |                  |           
                    --------------------
                            |
                            | R_n
                            |
                --------------------------
               |                          |
               |                          |
              \ /                        \ /
    --------------------           -----------------
    |                  |           |               |
    |  Video Target    |           | Sending Rate  |
    |  Rate Calculator |           | Calculator    |
    |                  |           |               |            
    --------------------           -----------------
       |        /|\                   /|\      |
    R_v|         |                     |       |  
       |         -----------------------       |       
       |                     |                 | R_s 
    ------------             |L_s              |
    |          |             |                 |
    |          |  R_o    --------------       \|/
    |  Encoder |---------->   | | | | | --------------->          
    |          |              | | | | |     video packets
    ------------         --------------         
                         Rate Shaping Buffer     

                     Figure 1 NADA Sender Structure

 

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5.1 Reference rate calculation

The sender initializes the reference rate R_n as R-min by default, or to
a value specified by the upper-layer application. [Editor's note: should
proper choice of starting rate value be within the scope of the CC
solution? ]

The reference rate R_n is calculated based on receiver feedback
information regarding queuing delay d_tilde_n, composite congestion
signal x_n, its derivative x'_n, as well as the receiving rate R_r. The
sender switches between two modes of operation:

        * Accelerated ramp up: if the reported queuing delay is close to
        zero and both values of x_n and x'_n are close to zero,
        indicating empty queues along the path of the flow and,
        consequently, underutilized network bandwidth; or 

        * Gradual rate update: in all other conditions, whereby the
        receiver reports on a standing or increasing/decreasing queue
        and/or composite congestion signal. 

5.1.1 Accelerated ramp up

In the absence of a non-zero congestion signal to guide the sending rate
calculation, the sender needs to ramp up its estimated bandwidth as
quickly as possible without introducing excessive queuing delay. Ideally
the flow should inflict no more than T_th milliseconds of queuing delay
at the bottleneck during the ramp-up process. A typical value of T_th is
50ms.

Note that the sender will be aware of any queuing delay introduced by
its rate increase after at least one round-trip time. In addition, the
bottleneck bandwidth C is greater than or equal to the receive rate R_r
reported from the most recent "no congestion" feedback message. The rate
R_n is updated as follows: 

                              T_th
    gamma = min [gamma_0, ---------------]      (3)
                           RTT_0+delta_0

    R_n  =  (1+gamma) R_r                     (4)

                                    

In (3) and (4), the multiplier gamma for rate increase is upper-bounded
by a fixed ratio gamma_0 (e.g., 20%), as well as a ratio which depends
 

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on T_th, base RTT as measured during the non-congested phase, and target
ACK interval delta_0.  The rationale behind this is that the rate
increase multiplier should decrease with the delay in the feedback
control loop, and that RTT_0 + delta_0 provides a worst-case estimate of
feedback control delay when the network is not congested. 

5.1.2. Gradual rate update

When the receiver reports indicate a standing congestion level, NADA
operates in gradual update mode, and calculates its reference rate as: 

                   kappa * delta_s           
    R_n  <-- R_n + ---------------- * (theta-(R_n-R_min)*x_hat) (5) 
                      tau_o^2

where

    theta = w*(R_max - R_min)*x_ref.    (6)

    x_hat = x_n + eta*tau_o* x'_n       (7)

In (5), delta_s refers to the time interval between current and previous
rate updates. Note that delta_s is the same as the RTCP report interval
at the receiver (see delta from (2)) when the backward path is un-
congested.

In (6), R_min and R_max denote the content-dependent rate range the
encoder can produce. The weighting factor reflecting a flow's priority
is w. The reference congestion signal x_ref is chosen so that the
maximum rate of R_max can be achieved when x_hat = w*x_ref. 

Proper choice of the scaling parameters eta and kappa in (5) and (7) can
ensure system stability so long as the RTT falls below the upper bound
of tau_o. The recommended default value of tau_o is chosen as 500ms. 

For both modes of operations, the final reference rate R_n is clipped
within the range of [R_min, R_max]. Note also that the sender does not
need any explicit knowledge of the management scheme inside the network.
Rather, it reacts to the aggregation of all forms of congestion
indications (delay, loss, and explicit markings) via the composite
congestion signals x_n and x'_n from the receiver in a coherent manner. 

 

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5.2 Video encoder rate control

The video encoder rate control procedure has the following
characteristics:

    * Rate changes can happen only at large intervals, on the order of
    seconds.

    * The encoder output rate may fluctuate around the target rate R_v. 

    * The encoder output rate is further constrained by video content
    complexity. The range of the final rate output is [R_min, R_max].
    Note that it is content-dependent and may vary over time.

The operation of the live video encoder is out of the scope of the
design for the congestion control scheme in NADA. Instead, its behavior
is treated as a black box. 

5.3 Rate shaping buffer

A rate shaping buffer is employed to absorb any instantaneous mismatch
between encoder rate output R_o and regulated sending rate R_s. The size
of the buffer evolves from time t-tau to time t as: 

              L_s(t) = max [0, L_s(t-tau)+(R_o-R_s)*tau]. 

A large rate shaping buffer contributes to higher end-to-end delay,
which may harm the performance of real-time media communications.
Therefore, the sender has a strong incentive to constrain the size of
the shaping buffer. It can either deplete it faster by increasing the
sending rate R_s, or limit its growth by reducing the target rate for
the video encoder rate control R_v. 

5.4 Adjusting video target rate and sending rate

The target rate for the live video encoder is updated based on both the
reference rate R_n and the rate shaping buffer size L_s, as follows:

                               L_s
        R_v = R_n - beta_v * -------.       (8) 
                              tau_v

Similarly, the outgoing rate is regulated based on both the reference
rate R_n and the rate shaping buffer size L_s, such that:

                               L_s
        R_s = R_n + beta_s * -------.       (9) 
                              tau_v
 

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In (8) and (9), the first term indicates the rate calculated from
network congestion feedback alone. The second term indicates the
influence of the rate shaping buffer. A large rate shaping buffer nudges
the encoder target rate slightly below -- and the sending rate slightly
above -- the reference rate R_n. 

Intuitively, the amount of extra rate offset needed to completely drain
the rate shaping buffer within the same time frame of encoder rate
adaptation tau_v is given by L_s/tau_v. The scaling parameters beta_v
and beta_s can be tuned to balance between the competing goals of
maintaining a small rate shaping buffer and deviating the system from
the reference rate point.

6. Incremental Deployment

One nice property of NADA is the consistent video endpoint behavior
irrespective of network node variations. This facilitates gradual,
incremental adoption of the scheme. 

To start off with, the encoder congestion control mechanism can be
implemented without any explicit support from the network, and relies
solely on observed one-way delay measurements and packet loss ratios as
implicit congestion signals. 

When ECN is enabled at the network nodes with RED-based marking, the
receiver can fold its observations of ECN markings into the calculation
of the equivalent delay. The sender can react to these explicit
congestion signals without any modification.

Ultimately, networks equipped with proactive marking based on token
bucket level metering can reap the additional benefits of zero standing
queues and lower end-to-end delay and work seamlessly with existing
senders and receivers. 

7. Implementation Status

The NADA scheme has been implemented in the ns-2 simulation platform
[ns2]. Extensive simulation evaluations of an earlier version of the
draft are documented in [Zhu-PV13]. Evaluation results of the current
draft over several test cases in [I-D.draft-sarker-rmcat-eval-test] have
been presented at recent IETF meetings [IETF-90][IETF-91]. 

The scheme has also been implemented and evaluated in a lab setting as
described in [IETF-90]. Preliminary evaluation results of NADA in
single-flow and multi-flow scenarios have been presented in [IETF-91]. 

 

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8. IANA Considerations

There are no actions for IANA.

9. References

9.1  Normative References

   [RFC2119]  Bradner, S., "Key words for use in RFCs to Indicate
              Requirement Levels", BCP 14, RFC 2119, March 1997.

   [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
              of Explicit Congestion Notification (ECN) to IP",
              RFC 3168, September 2001.

   [RFC3550]  Schulzrinne, H., Casner, S., Frederick, R., and V.
              Jacobson, "RTP: A Transport Protocol for Real-Time
              Applications", STD 64, RFC 3550, July 2003.

9.2  Informative References

   [RFC3168]  Ramakrishnan, K., Floyd, S., and D. Black, "The Addition
              of Explicit Congestion Notification (ECN) to IP",
              RFC 3168, September 2001.

   [RFC2309]  Braden, B., Clark, D., Crowcroft, J., Davie, B., Deering,
              S., Estrin, D., Floyd, S., Jacobson, V., Minshall, G.,
              Partridge, C., Peterson, L., Ramakrishnan, K., Shenker,
              S., Wroclawski, J., and L. Zhang, "Recommendations on
              Queue Management and Congestion Avoidance in the
              Internet", RFC 2309, April 1998.

   [RFC6817] Shalunov, S.,  Hazel, G., Iyengar, J., and Kuehlewind, M.,
              "Low Extra Delay Background Transport (LEDBAT)", RFC 6817,
              December 2012

   [Floyd-CCR00] Floyd, S., Handley, M., Padhye, J., and Widmer, J., 
              "Equation-based Congestion Control for Unicast
              Applications", ACM SIGCOMM Computer Communications Review,
              vol. 30. no. 4., pp. 43-56, October 2000.

   [Budzisz-TON11] Budzisz, L. et al.,  "On the Fair Coexistence of
              Loss- and Delay-Based TCP", IEEE/ACM Transactions on
              Networking, vol. 19, no. 6, pp. 1811-1824, December 2011. 

 

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   [ns2] "The Network Simulator - ns-2", http://www.isi.edu/nsnam/ns/

   [Zhu-PV13] Zhu, X. and Pan, R., "NADA: A Unified Congestion Control
              Scheme for Low-Latency Interactive Video", in Proc. IEEE
              International Packet Video Workshop (PV'13). San Jose, CA,
              USA. December 2013.

   [I-D.draft-sarker-rmcat-eval-test] Sarker, Z., Singh, V., Zhu, X.,
              and Ramalho, M., "Test Cases for Evaluating RMCAT
              Proposals", draft-sarker-rmcat-eval-test-01 (work in
              progress), June 2014.

   [IETF-90] Zhu, X. et al., "NADA Update: Algorithm, Implementation,
              and Test Case Evalua6on Results", presented at IETF 90,
              https://tools.ietf.org/agenda/90/slides/slides-90-rmcat-
              6.pdf

   [IETF-91] Zhu, X. et al., "NADA Algorithm Update and Test Case
              Evaluations", presented at IETF 91 Interium,
              https://datatracker.ietf.org/meeting/91/agenda/rmcat/

Appendix A. Network Node Operations

              NADA can work with different network queue management
              schemes and does not assume any specific network node
              operation. As an example, this appendix describes three
              variations of queue management behavior at the network
              node, leading to either implicit or explicit congestion
              signals.

              In all three flavors described below, the network queue
              operates with the simple first-in-first-out (FIFO)
              principle. There is no need to maintain per-flow state.
              Such a simple design ensures that the system can scale
              easily with a large number of video flows and high link
              capacity.

              NADA sender behavior stays the same in the presence of all
              types of congestion indicators: delay, loss, ECN marking
              due to either RED/ECN or PCN algorithms. This unified
              approach allows a graceful transition of the scheme as the
              network shifts dynamically between light and heavy
              congestion levels.

 

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A.1 Default behavior of drop tail

              In a conventional network with drop tail or RED queues,
              congestion is inferred from the estimation of end-to-end
              delay and/or packet loss. Packet drops at the queue are
              detected at the receiver, and contributes to the
              calculation of the equivalent delay x_n. No special action
              is required at network node.

A.2 ECN marking

              In this mode, the network node randomly marks the ECN
              field in the IP packet header following the Random Early
              Detection (RED) algorithm [RFC2309]. Calculation of the
              marking probability involves the following steps: 

    * upon packet arrival, update smoothed queue size q_avg as: 

                  q_avg = alpha*q + (1-alpha)*q_avg. 

    The smoothing parameter alpha is a value between 0 and 1. A value of
    alpha=1 corresponds to performing no smoothing at all. 

    * calculate marking probability p as: 

        p = 0, if q < q_lo; 

                   q_avg - q_lo
        p = p_max*--------------, if q_lo <= q < q_hi; 
                   q_hi - q_lo

        p = 1, if q >= q_hi. 

Here, q_lo and q_hi corresponds to the low and high thresholds of queue
occupancy. The maximum marking probability is p_max. 

The ECN markings events will contribute to the calculation of an
equivalent delay x_n at the receiver. No changes are required at the
sender.

A.3 PCN marking

As a more advanced feature, we also envisage network nodes which support
PCN marking based on virtual queues. In such a case, the marking
probability of the ECN bit in the IP packet header is calculated as
follows:
 

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    * upon packet arrival, meter packet against token bucket (r,b); 

    * update token level b_tk; 

    * calculate the marking probability as: 

        p = 0, if b-b_tk < b_lo; 

                    b-b_tk-b_lo
        p = p_max* --------------, if b_lo<= b-b_tk <b_hi; 
                     b_hi-b_lo

        p = 1, if b-b_tk>=b_hi. 

Here, the token bucket lower and upper limits are denoted by b_lo and
b_hi, respectively. The parameter b indicates the size of the token
bucket. The parameter r is chosen as r=gamma*C, where gamma<1 is the
target utilization ratio and C designates link capacity. The maximum
marking probability is p_max. 

The ECN markings events will contribute to the calculation of an
equivalent delay x_n at the receiver. No changes are required at the
sender. The virtual queuing mechanism from the PCN marking algorithm
will lead to additional benefits such as zero standing queues. 

Authors' Addresses

Xiaoqing Zhu
Cisco Systems, 
12515 Research Blvd.,
Austin, TX 78759, USA
Email: xiaoqzhu@cisco.com

Rong Pan
Cisco Systems
510 McCarthy Blvd, 
Milpitas, CA 95134, USA
Email: ropan@cisco.com

Michael A. Ramalho     
6310 Watercrest Way Unit 203          
Lakewood Ranch, FL, 34202, USA
Email: mramalho@cisco.com
 

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Sergio Mena de la Cruz
Cisco Systems
EPFL, Quartier de l'Innovation, Batiment E
Ecublens, Vaud 1015, Switzerland
Email:  semena@cisco.com

Charles Ganzhorn
7900 International Drive 
International Plaza, Suite 400 
Bloomington, MN 55425, USA
Email: charles.ganzhorn@gmail.com

Paul E. Jones
7025 Kit Creek Rd.
Research Triangle Park, NC 27709, USA
Email: paulej@packetizer.com

Stefano D'Aronco
EPFL STI IEL LTS4 
ELD 220 (Batiment ELD), Station 11 
CH-1015 Lausanne, Switzerland
Email: stefano.daronco@epfl.ch

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