.. include:: replace.txt
.. highlight:: cpp

TCP models in ns-3
------------------

This chapter describes the TCP models available in |ns3|.

Generic support for TCP
***********************

|ns3| was written to support multiple TCP implementations. The implementations
inherit from a few common header classes in the ``src/network`` directory, so that
user code can swap out implementations with minimal changes to the scripts.

There are two important abstract base classes:

* class :cpp:class:`TcpSocket`:  This is defined in
  ``src/internet/model/tcp-socket.{cc,h}``. This class exists for hosting TcpSocket
  attributes that can be reused across different implementations. For instance,
  the attribute ``InitialCwnd`` can be used for any of the implementations
  that derive from class :cpp:class:`TcpSocket`.
* class :cpp:class:`TcpSocketFactory`:  This is used by the layer-4 protocol
  instance to create TCP sockets of the right type.

There are presently three implementations of TCP available for |ns3|.

* a natively implemented TCP for ns-3
* support for the `Network Simulation Cradle (NSC) <http://www.wand.net.nz/~stj2/nsc/>`__
* support for `Direct Code Execution (DCE) <https://www.nsnam.org/overview/projects/direct-code-execution/>`__

It should also be mentioned that various ways of combining virtual machines
with |ns3| makes available also some additional TCP implementations, but
those are out of scope for this chapter.

ns-3 TCP
********

In brief, the native |ns3| TCP model supports a full bidirectional TCP with
connection setup and close logic.  Several congestion control algorithms
are supported, with NewReno the default, and Westwood, Hybla, HighSpeed,
Vegas, Scalable, Veno, Binary Increase Congestion Control (BIC), Yet Another
HighSpeed TCP (YeAH), Illinois, H-TCP, Low Extra Delay Background Transport
(LEDBAT) and TCP Low Priority (TCP-LP) also supported. The model also supports
Selective Acknowledgements (SACK). Multipath-TCP is not yet supported in the |ns3|
releases.

Model history
+++++++++++++

Until the ns-3.10 release, |ns3| contained a port of the TCP model from `GTNetS
<http://www.ece.gatech.edu/research/labs/MANIACS/GTNetS/index.html>`_. 
This implementation was substantially rewritten by Adriam Tam for ns-3.10.
In 2015, the TCP module has been redesigned in order to create a better 
environment for creating and carrying out automated tests. One of the main 
changes involves congestion control algorithms, and how they are implemented.

Before ns-3.25 release, a congestion control was considered as a stand-alone TCP
through an inheritance relation: each congestion control (e.g. TcpNewReno) was
a subclass of TcpSocketBase, reimplementing some inherited methods. The
architecture was redone to avoid this inheritance,
the fundamental principle of the GSoC proposal was avoiding this inheritance,
by making each congestion control a separate class, and making an interface
to exchange important data between TcpSocketBase and the congestion modules.
For instance, similar modularity is used in Linux.

Along with congestion control, Fast Retransmit and Fast Recovery algorithms
have been modified; in previous releases, these algorithms were demanded to
TcpSocketBase subclasses. Starting from ns-3.25, they have been merged inside
TcpSocketBase. In future releases, they can be extracted as separate modules,
following the congestion control design.

Usage
+++++

In many cases, usage of TCP is set at the application layer by telling
the |ns3| application which kind of socket factory to use.

Using the helper functions defined in ``src/applications/helper`` and
``src/network/helper``, here is how one would create a TCP receiver::

  // Create a packet sink on the star "hub" to receive these packets
  uint16_t port = 50000;
  Address sinkLocalAddress(InetSocketAddress (Ipv4Address::GetAny (), port));
  PacketSinkHelper sinkHelper ("ns3::TcpSocketFactory", sinkLocalAddress);
  ApplicationContainer sinkApp = sinkHelper.Install (serverNode);
  sinkApp.Start (Seconds (1.0));
  sinkApp.Stop (Seconds (10.0));

Similarly, the below snippet configures OnOffApplication traffic source to use
TCP::

  // Create the OnOff applications to send TCP to the server
  OnOffHelper clientHelper ("ns3::TcpSocketFactory", Address ());

The careful reader will note above that we have specified the TypeId of an
abstract base class :cpp:class:`TcpSocketFactory`. How does the script tell
|ns3| that it wants the native |ns3| TCP vs. some other one?  Well, when
internet stacks are added to the node, the default TCP implementation that is
aggregated to the node is the |ns3| TCP.  This can be overridden as we show
below when using Network Simulation Cradle. So, by default, when using the |ns3|
helper API, the TCP that is aggregated to nodes with an Internet stack is the
native |ns3| TCP.

To configure behavior of TCP, a number of parameters are exported through the
|ns3| attribute system. These are documented in the `Doxygen
<http://www.nsnam.org/doxygen/classns3_1_1_tcp_socket.html>` for class
:cpp:class:`TcpSocket`.  For example, the maximum segment size is a
settable attribute.

To set the default socket type before any internet stack-related objects are
created, one may put the following statement at the top of the simulation
program:: 

  Config::SetDefault ("ns3::TcpL4Protocol::SocketType", StringValue ("ns3::TcpNewReno")); 

For users who wish to have a pointer to the actual socket (so that
socket operations like Bind(), setting socket options, etc. can be
done on a per-socket basis), Tcp sockets can be created by using the 
``Socket::CreateSocket()`` method.  The TypeId passed to CreateSocket()
must be of type :cpp:class:`ns3::SocketFactory`, so configuring the underlying 
socket type must be done by twiddling the attribute associated with the
underlying TcpL4Protocol object.  The easiest way to get at this would be 
through the attribute configuration system.  In the below example,
the Node container "n0n1" is accessed to get the zeroth element, and a socket is
created on this node::

  // Create and bind the socket...
  TypeId tid = TypeId::LookupByName ("ns3::TcpNewReno");
  Config::Set ("/NodeList/*/$ns3::TcpL4Protocol/SocketType", TypeIdValue (tid));
  Ptr<Socket> localSocket =
    Socket::CreateSocket (n0n1.Get (0), TcpSocketFactory::GetTypeId ());

Above, the "*" wild card for node number is passed to the attribute
configuration system, so that all future sockets on all nodes are set to 
NewReno, not just on node 'n0n1.Get (0)'.  If one wants to limit it to just 
the specified node, one would have to do something like::

  // Create and bind the socket...
  TypeId tid = TypeId::LookupByName ("ns3::TcpNewReno");
  std::stringstream nodeId;
  nodeId << n0n1.Get (0)->GetId ();
  std::string specificNode = "/NodeList/" + nodeId.str () + "/$ns3::TcpL4Protocol/SocketType";
  Config::Set (specificNode, TypeIdValue (tid));
  Ptr<Socket> localSocket =
    Socket::CreateSocket (n0n1.Get (0), TcpSocketFactory::GetTypeId ()); 

Once a TCP socket is created, one will want to follow conventional socket logic
and either connect() and send() (for a TCP client) or bind(), listen(), and
accept() (for a TCP server).
Please note that applications usually create the sockets they use automatically,
and so is not straightforward to connect direcly to them using pointers. Please
refer to the source code of your preferred application to discover how and when
it creates the socket.

TCP Socket interaction and interface with Application layer
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^

In the following there is an analysis on the public interface of the TCP socket,
and how it can be used to interact with the socket itself. An analysis of the
callback fired by the socket is also carried out. Please note that, for
the sake of clarity, we will use the terminology "Sender" and "Receiver" to clearly
divide the functionality of the socket. However, in TCP these two roles can be
applied at the same time (i.e. a socket could be a sender and a receiver at the
same time): our distinction does not lose generality, since the following
definition can be applied to both sockets in case of full-duplex mode.

----------

**TCP state machine (for commodity use)**

.. _fig-tcp-state-machine:

.. figure:: figures/tcp-state-machine.*
   :align: center

   TCP State machine

In ns-3 we are fully compliant with the state machine depicted in 
Figure :ref:`fig-tcp-state-machine`.

----------

**Public interface for receivers (e.g. servers receiving data)**

*Bind()*
  Bind the socket to an address, or to a general endpoint. A general endpoint
  is an endpoint with an ephemeral port allocation (that is, a random port
  allocation) on the 0.0.0.0 IP address. For instance, in current applications,
  data senders usually binds automatically after a *Connect()* over a random
  port. Consequently, the connection will start from this random port towards
  the well-defined port of the receiver. The IP 0.0.0.0 is then translated by
  lower layers into the real IP of the device.

*Bind6()*
  Same as *Bind()*, but for IPv6.

*BindToNetDevice()*
  Bind the socket to the specified NetDevice, creating a general endpoint.

*Listen()*
  Listen on the endpoint for an incoming connection. Please note that this
  function can be called only in the TCP CLOSED state, and transit in the
  LISTEN state. When an incoming request for connection is detected (i.e. the
  other peer invoked *Connect()*) the application will be signaled with the
  callback *NotifyConnectionRequest* (set in *SetAcceptCallback()* beforehand).
  If the connection is accepted (the default behavior, when the associated
  callback is a null one) the Socket will fork itself, i.e. a new socket is
  created to handle the incoming data/connection, in the state SYN_RCVD. Please
  note that this newly created socket is not connected anymore to the callbacks
  on the "father" socket (e.g. DataSent, Recv); the pointer of the newly
  created socket is provided in the Callback *NotifyNewConnectionCreated* (set
  beforehand in *SetAcceptCallback*), and should be used to connect new
  callbacks to interesting events (e.g. Recv callback). After receiving the ACK
  of the SYN-ACK, the socket will set the congestion control, move into
  ESTABLISHED state, and then notify the application with
  *NotifyNewConnectionCreated*.

*ShutdownSend()*
  Signal a termination of send, or in other words revents data from being added
  to the buffer. After this call, if buffer is already empty, the socket
  will send a FIN, otherwise FIN will go when buffer empties. Please note
  that this is useful only for modeling "Sink" applications. If you have
  data to transmit, please refer to the *Send()* / *Close()* combination of
  API.

*GetRxAvailable()*
  Get the amount of data that could be returned by the Socket in one or multiple
  call to Recv or RecvFrom. Please use the Attribute system to configure the
  maximum available space on the receiver buffer (property "RcvBufSize").

*Recv()*
  Grab data from the TCP socket. Please remember that TCP is a stream socket,
  and it is allowed to concatenate multiple packets into bigger ones. If no data
  is present (i.e. *GetRxAvailable* returns 0) an empty packet is returned.
  Set the callback *RecvCallback* through *SetRecvCallback()* in order to have
  the application automatically notified when some data is ready to be read.
  It's important to connect that callback to the newly created socket in case
  of forks.

*RecvFrom()*
  Same as Recv, but with the source address as parameter.

-------------------

**Public interface for senders (e.g. clients uploading data)**

*Connect()*
  Set the remote endpoint, and try to connect to it. The local endpoint should
  be set before this call, or otherwise an ephemeral one will be created. The
  TCP then will be in the SYN_SENT state. If a SYN-ACK is received, the TCP will
  setup the congestion control, and then call the callback
  *ConnectionSucceeded*.

*GetTxAvailable()*
  Return the amount of data that can be stored in the TCP Tx buffer. Set this
  property through the Attribute system ("SndBufSize").

*Send()*
  Send the data into the TCP Tx buffer. From there, the TCP rules will decide
  if, and when, this data will be transmitted. Please note that, if the tx
  buffer has enough data to fill the congestion (or the receiver) window, dynamically
  varying the rate at which data is injected in the TCP buffer does not have any
  noticeable effect on the amount of data transmitted on the wire, that will
  continue to be decided by the TCP rules.

*SendTo()*
  Same as *Send()*.

*Close()*
  Terminate the local side of the connection, by sending a FIN (after all data
  in the tx buffer has been transmitted). This does not prevent the socket in
  receiving data, and employing retransmit mechanism if losses are detected. If
  the application calls *Close()* with unread data in its rx buffer, the socket
  will send a reset. If the socket is in the state SYN_SENT, CLOSING, LISTEN,
  FIN_WAIT_2, or LAST_ACK, after that call the application will be notified with
  *NotifyNormalClose()*. In other cases, the notification is delayed
  (see *NotifyNormalClose()*).

-----------------------------------------

**Public callbacks**

These callbacks are called by the TCP socket to notify the application of
interesting events. We will refer to these with the protected name used in
socket.h, but we will provide the API function to set the pointers to these
callback as well.

*NotifyConnectionSucceeded*: *SetConnectCallback*, 1st argument
  Called in the SYN_SENT state, before moving to ESTABLISHED. In other words, we
  have sent the SYN, and we received the SYN-ACK: the socket prepare the
  sequence numbers, send the ACK for the SYN-ACK, try to send out more data (in
  another segment) and then invoke this callback. After this callback, it
  invokes the NotifySend callback.

*NotifyConnectionFailed*: *SetConnectCallback*, 2nd argument
  Called after the SYN retransmission count goes to 0. SYN packet is lost
  multiple time, and the socket give up.

*NotifyNormalClose*: *SetCloseCallbacks*, 1st argument
  A normal close is invoked. A rare case is when we receive an RST segment (or a
  segment with bad flags) in normal states. All other cases are:
  - The application tries to *Connect()* over an already connected socket
  - Received an ACK for the FIN sent, with or without the FIN bit set (we are in LAST_ACK)
  - The socket reaches the maximum amount of retries in retransmitting the SYN (*)
  - We receive a timeout in the LAST_ACK state
  - Upon entering the TIME_WAIT state, before waiting the 2*Maximum Segment Lifetime seconds to finally deallocate the socket.

*NotifyErrorClose*: *SetCloseCallbacks*, 2nd argument
  Invoked when we send an RST segment (for whatever reason) or we reached the
  maximum amount of data retries.

*NotifyConnectionRequest*: *SetAcceptCallback*, 1st argument
  Invoked in the LISTEN state, when we receive a SYN. The return value indicates
  if the socket should accept the connection (return true) or should ignore it
  (return false).

*NotifyNewConnectionCreated*: *SetAcceptCallback*, 2nd argument
  Invoked when from SYN_RCVD the socket passes to ESTABLISHED, and after setting
  up the congestion control, the sequence numbers, and processed the incoming
  ACK. If there is some space in the buffer, *NotifySend* is called shortly
  after this callback. The Socket pointer, passed with this callback, is the
  newly created socket, after a Fork().

*NotifyDataSent*: *SetDataSentCallback*
  The Socket notifies the application that some bytes has been transmitted on
  the IP level. These bytes could still be lost in the node (traffic control
  layer) or in the network.

*NotifySend*: *SetSendCallback*
  Invoked if there is some space in the tx buffer when entering the ESTABLISHED
  state (e.g. after the ACK for SYN-ACK is received), after the connection
  succeeds (e.g. after the SYN-ACK is received) and after each new ack (i.e.
  that advances SND.UNA).

*NotifyDataRecv*: *SetRecvCallback*
  Called when in the receiver buffere there are in-order bytes, and when in
  FIN_WAIT_1 or FIN_WAIT_2 the socket receive a in-sequence FIN (that can carry
  data).


Congestion Control Algorithms
+++++++++++++++++++++++++++++
Here follows a list of supported TCP congestion control algorithms. For an
academic peer-reviewed paper on these congestion control algorithms, see
http://dl.acm.org/citation.cfm?id=2756518 .

New Reno
^^^^^^^^
New Reno algorithm introduces partial ACKs inside the well-established Reno
algorithm. This and other modifications are described in RFC 6582. We have two
possible congestion window increment strategy: slow start and congestion
avoidance. Taken from RFC 5681:

  During slow start, a TCP increments cwnd by at most SMSS bytes for
  each ACK received that cumulatively acknowledges new data.  Slow
  start ends when cwnd exceeds ssthresh (or, optionally, when it
  reaches it, as noted above) or when congestion is observed.  While
  traditionally TCP implementations have increased cwnd by precisely
  SMSS bytes upon receipt of an ACK covering new data, we RECOMMEND
  that TCP implementations increase cwnd, per Equation :eq:`newrenocongavoid`,
  where N is the number of previously unacknowledged bytes acknowledged
  in the incoming ACK.

.. math:: cwnd += min (N, SMSS)
   :label: newrenocongavoid

During congestion avoidance, cwnd is incremented by roughly 1 full-sized
segment per round-trip time (RTT), and for each congestion event, the slow
start threshold is halved.

High Speed
^^^^^^^^^^
TCP HighSpeed is designed for high-capacity channels or, in general, for
TCP connections with large congestion windows.
Conceptually, with respect to the standard TCP, HighSpeed makes the
cWnd grow faster during the probing phases and accelerates the
cWnd recovery from losses.
This behavior is executed only when the window grows beyond a
certain threshold, which allows TCP Highspeed to be friendly with standard
TCP in environments with heavy congestion, without introducing new dangers
of congestion collapse.

Mathematically:

.. math::  cWnd = cWnd + \frac{a(cWnd)}{cWnd}

The function a() is calculated using a fixed RTT the value 100 ms (the
lookup table for this function is taken from RFC 3649). For each congestion
event, the slow start threshold is decreased by a value that depends on the
size of the slow start threshold itself. Then, the congestion window is set
to such value.

.. math::   cWnd = (1-b(cWnd)) \cdot cWnd

The lookup table for the function b() is taken from the same RFC.
More information at: http://dl.acm.org/citation.cfm?id=2756518

Hybla
^^^^^
The key idea behind TCP Hybla is to obtain for long RTT connections the same
instantaneous transmission rate of a reference TCP connection with lower RTT.
With analytical steps, it is shown that this goal can be achieved by
modifying the time scale, in order for the throughput to be independent from
the RTT. This independence is obtained through the use of a coefficient rho.

This coefficient is used to calculate both the slow start threshold
and the congestion window when in slow start and in congestion avoidance,
respectively.

More information at: http://dl.acm.org/citation.cfm?id=2756518

Westwood
^^^^^^^^
Westwood and Westwood+ employ the AIAD (Additive Increase/Adaptive Decrease)·
congestion control paradigm. When a congestion episode happens,·
instead of halving the cwnd, these protocols try to estimate the network's
bandwidth and use the estimated value to adjust the cwnd.·
While Westwood performs the bandwidth sampling every ACK reception,·
Westwood+ samples the bandwidth every RTT.

More information at: http://dl.acm.org/citation.cfm?id=381704 and
http://dl.acm.org/citation.cfm?id=2512757

Vegas
^^^^^
TCP Vegas is a pure delay-based congestion control algorithm implementing a
proactive scheme that tries to prevent packet drops by maintaining a small
backlog at the bottleneck queue. Vegas continuously samples the RTT and computes
the actual throughput a connection achieves using Equation (1) and compares it
with the expected throughput calculated in Equation (2). The difference between
these 2 sending rates in Equation (3) reflects the amount of extra packets being
queued at the bottleneck.

.. math::

   actual &= \frac{cWnd}{RTT}        \\
   expected &= \frac{cWnd}{BaseRTT}  \\
   diff &= expected - actual

To avoid congestion, Vegas linearly increases/decreases its congestion window
to ensure the diff value fall between the two predefined thresholds, alpha and
beta. diff and another threshold, gamma, are used to determine when Vegas
should change from its slow-start mode to linear increase/decrease mode.
Following the implementation of Vegas in Linux, we use 2, 4, and 1 as the
default values of alpha, beta, and gamma, respectively, but they can be
modified through the Attribute system.

More information at: http://dx.doi.org/10.1109/49.464716

Scalable
^^^^^^^^
Scalable improves TCP performance to better utilize the available bandwidth of
a highspeed wide area network by altering NewReno congestion window adjustment
algorithm.  When congestion has not been detected, for each ACK received in an
RTT, Scalable increases its cwnd per:

.. math::  cwnd = cwnd + 0.01

Following Linux implementation of Scalable, we use 50 instead of 100 to account
for delayed ACK.

On the first detection of congestion in a given RTT, cwnd is reduced based on
the following equation:

.. math::  cwnd = cwnd - ceil(0.125 \cdot cwnd)

More information at: http://dl.acm.org/citation.cfm?id=956989

Veno
^^^^

TCP Veno enhances Reno algorithm for more effectively dealing with random
packet loss in wireless access networks by employing Vegas's method in
estimating the backlog at the bottleneck queue to distinguish between
congestive and non-congestive states.

The backlog (the number of packets accumulated at the bottleneck queue) is
calculated using Equation (1):

.. math::
   N &= Actual \cdot (RTT - BaseRTT) \\
     &= Diff \cdot BaseRTT

where:

.. math::
   Diff &= Expected - Actual \\
        &= \frac{cWnd}{BaseRTT} - \frac{cWnd}{RTT}

Veno makes decision on cwnd modification based on the calculated N and its
predefined threshold beta.

Specifically, it refines the additive increase algorithm of Reno so that the
connection can stay longer in the stable state by incrementing cwnd by
1/cwnd for every other new ACK received after the available bandwidth has
been fully utilized, i.e. when N exceeds beta.  Otherwise, Veno increases
its cwnd by 1/cwnd upon every new ACK receipt as in Reno.

In the multiplicative decrease algorithm, when Veno is in the non-congestive
state, i.e. when N is less than beta, Veno decrements its cwnd by only 1/5
because the loss encountered is more likely a corruption-based loss than a
congestion-based.  Only when N is greater than beta, Veno halves its sending
rate as in Reno.

More information at: http://dx.doi.org/10.1109/JSAC.2002.807336

Bic
^^^

In TCP Bic the congestion control problem is viewed as a search
problem. Taking as a starting point the current window value
and as a target point the last maximum window value
(i.e. the cWnd value just before the loss event) a binary search
technique can be used to update the cWnd value at the midpoint between
the two, directly or using an additive increase strategy if the distance from
the current window is too large.

This way, assuming a no-loss period, the congestion window logarithmically
approaches the maximum value of cWnd until the difference between it and cWnd
falls below a preset threshold. After reaching such a value (or the maximum
window is unknown, i.e. the binary search does not start at all) the algorithm
switches to probing the new maximum window with a 'slow start' strategy.

If a loss occur in either these phases, the current window (before the loss)
can be treated as the new maximum, and the reduced (with a multiplicative
decrease factor Beta) window size can be used as the new minimum.

More information at: http://ieeexplore.ieee.org/xpl/articleDetails.jsp?arnumber=1354672

YeAH
^^^^

YeAH-TCP (Yet Another HighSpeed TCP) is a heuristic designed to balance various
requirements of a state-of-the-art congestion control algorithm:


1. fully exploit the link capacity of high BDP networks while inducing a small number of congestion events
2. compete friendly with Reno flows
3. achieve intra and RTT fairness
4. robust to random losses
5. achieve high performance regardless of buffer size

YeAH operates between 2 modes: Fast and Slow mode.  In the Fast mode when the queue
occupancy is small and the network congestion level is low, YeAH increments
its congestion window according to the aggressive STCP rule.  When the number of packets
in the queue grows beyond a threshold and the network congestion level is high, YeAH enters
its Slow mode, acting as Reno with a decongestion algorithm.  YeAH employs Vegas' mechanism
for calculating the backlog as in Equation :eq:`q_yeah`.  The estimation of the network congestion
level is shown in Equation :eq:`l_yeah`.

.. math::  Q = (RTT - BaseRTT) \cdot \frac{cWnd}{RTT}
   :label: q_yeah

.. math::  L = \frac{RTT - BaseRTT}{BaseRTT}
   :label: l_yeah

To ensure TCP friendliness, YeAH also implements an algorithm to detect the presence of legacy
Reno flows.  Upon the receipt of 3 duplicate ACKs, YeAH decreases its slow start threshold
according to Equation (3) if it's not competing with Reno flows.  Otherwise,  the ssthresh is
halved as in Reno:

.. math::  ssthresh = min(max(\frac{cWnd}{8}, Q), \frac{cWnd}{2})

More information: http://www.csc.lsu.edu/~sjpark/cs7601/4-YeAH_TCP.pdf

Illinois
^^^^^^^^

TCP Illinois is a hybrid congestion control algorithm designed for
high-speed networks.  Illinois implements a Concave-AIMD (or C-AIMD)
algorithm that uses packet loss as the primary congestion signal to
determine the direction of window update and queueing delay as the
secondary congestion signal to determine the amount of change.

The additive increase and multiplicative decrease factors (denoted as
alpha and beta, respectively) are functions of the current average queueing
delay da as shown in Equations (1) and (2).  To improve the protocol
robustness against sudden fluctuations in its delay sampling,
Illinois allows the increment of alpha to alphaMax
only if da stays below d1 for a some (theta) amount of time.

.. math::
   alpha &=
   \begin{cases}
      \quad alphaMax              & \quad \text{if } da <= d1 \\
      \quad k1 / (k2 + da)        & \quad \text{otherwise} \\
   \end{cases} \\
   \\
   beta &=
   \begin{cases}
      \quad betaMin               & \quad \text{if } da <= d2 \\
      \quad k3 + k4 \, da         & \quad \text{if } d2 < da < d3 \\
      \quad betaMax               & \quad \text{otherwise}
   \end{cases}
			     
where the calculations of k1, k2, k3, and k4 are shown in the following:

.. math::

   k1 &= \frac{(dm - d1) \cdot alphaMin \cdot alphaMax}{alphaMax - alphaMin} \\
   \\
   k2 &= \frac{(dm - d1) \cdot alphaMin}{alphaMax - alphaMin} - d1 \\
   \\
   k3 &= \frac{alphaMin \cdot d3 - alphaMax \cdot d2}{d3 - d2} \\
   \\
   k4 &= \frac{alphaMax - alphaMin}{d3 - d2}

Other parameters include da (the current average queueing delay), and
Ta (the average RTT, calculated as sumRtt / cntRtt in the implementation) and
Tmin (baseRtt in the implementation) which is the minimum RTT ever seen.
dm is the maximum (average) queueing delay, and Tmax (maxRtt in the
implementation) is the maximum RTT ever seen.

.. math::

   da &= Ta - Tmin

   dm &= Tmax - Tmin

   d_i &= eta_i \cdot dm

Illinois only executes its adaptation of alpha and beta when cwnd exceeds a threshold
called winThresh. Otherwise, it sets alpha and beta to the base values of 1 and 0.5,
respectively.

Following the implementation of Illinois in the Linux kernel, we use the following
default parameter settings:

* alphaMin = 0.3      (0.1 in the Illinois paper)
* alphaMax = 10.0
* betaMin = 0.125
* betaMax = 0.5
* winThresh = 15      (10 in the Illinois paper)
* theta = 5
* eta1 = 0.01
* eta2 = 0.1
* eta3 = 0.8

More information: http://www.doi.org/10.1145/1190095.1190166

H-TCP
^^^^^

H-TCP has been designed for high BDP (Bandwidth-Delay Product) paths. It is 
a dual mode protocol. In normal conditions, it works like traditional TCP 
with the same rate of increment and decrement for the congestion window. 
However, in high BDP networks, when it finds no congestion on the path 
after ``deltal`` seconds, it increases the window size based on the alpha 
function in the following:

.. math::

        alpha(delta)=1+10(delta-deltal)+0.5(delta-deltal)^2 

where ``deltal`` is a threshold in seconds for switching between the modes and 
``delta`` is the elapsed time from the last congestion. During congestion, 
it reduces the window size by multiplying by beta function provided 
in the reference paper.  The calculated throughput between the last two 
consecutive congestion events is considered for beta calculation. 

The transport ``TcpHtcp`` can be selected in the program 
``examples/tcp/tcp-variants/comparison`` to perform an experiment with H-TCP,
although it is useful to increase the bandwidth in this example (e.g.
to 20 Mb/s) to create a higher BDP link, such as

::

  ./waf --run "tcp-variants-comparison --transport_prot=TcpHtcp --bandwidth=20Mbps --duration=10"

More information (paper):  http://www.hamilton.ie/net/htcp3.pdf

More information (Internet Draft):  https://tools.ietf.org/html/draft-leith-tcp-htcp-06

LEDBAT
^^^^^^

Low Extra Delay Background Transport (LEDBAT) is an experimental delay-based 
congestion control algorithm that seeks to utilize the available bandwidth on
an end-to-end path while limiting the consequent increase in queueing delay 
on that path. LEDBAT uses changes in one-way delay measurements to limit 
congestion that the flow itself induces in the network.

As a first approximation, the LEDBAT sender operates as shown below:

on receipt of an ACK:

.. math::
       currentdelay = acknowledgement.delay
       basedelay = min (basedelay, currentdelay)
       queuingdelay = currentdelay - basedelay
       offtarget = (TARGET - queuingdelay) / TARGET
       cWnd += GAIN * offtarget * bytesnewlyacked * MSS / cWnd

``TARGET`` is the maximum queueing delay that LEDBAT itself may introduce in the
network, and ``GAIN`` determines the rate at which the cwnd responds to changes in 
queueing delay;  ``offtarget`` is a normalized value representing the difference between
the measured current queueing delay and the predetermined TARGET delay. offtarget can 
be positive or negative; consequently, cwnd increases or decreases in proportion to 
offtarget.

Following the recommendation of RFC 6817, the default values of the parameters are:

* TargetDelay = 100
* baseHistoryLen = 10
* noiseFilterLen = 4
* Gain = 1

To enable LEDBAT on all TCP sockets, the following configuration can be used:

::

  Config::SetDefault ("ns3::TcpL4Protocol::SocketType", TypeIdValue (TcpLedbat::GetTypeId ()));

To enable LEDBAT on a chosen TCP socket, the following configuration can be used:

::

  Config::Set ("$ns3::NodeListPriv/NodeList/1/$ns3::TcpL4Protocol/SocketType", TypeIdValue (TcpLedbat::GetTypeId ()));

The following unit tests have been written to validate the implementation of LEDBAT:

* LEDBAT should operate same as NewReno during slow start
* LEDBAT should operate same as NewReno if timestamps are disabled
* Test to validate cwnd increment in LEDBAT

In comparison to RFC 6817, the scope and limitations of the current LEDBAT
implementation are:

* It assumes that the clocks on the sender side and receiver side are synchronised
* In line with Linux implementation, the one-way delay is calculated at the sender side by using the timestamps option in TCP header
* Only the MIN function is used for noise filtering 

More information about LEDBAT is available in RFC 6817: https://tools.ietf.org/html/rfc6817

TCP-LP
^^^^^^

TCP-Low priority is a delay based congestion control protocol in which the low
priority data utilizes only the excess bandwidth available on an end-to-end path.
TCP-LP uses one way delay measurements as an indicator of congestion as it does
not influence cross-traffic in the reverse direction.

On acknowledgement:

.. math::

  One way delay = Receiver timestamp - Receiver timestamp echo reply
  Smoothed one way delay = 7/8 * Old Smoothed one way delay + 1/8 * one way delay
  If smoothed one way delay > owdMin + 15 * (owdMax - owdMin) / 100
    if LP_WITHIN_INF
      cwnd = 1
    else
      cwnd = cwnd / 2
    Inference timer is set

where owdMin and owdMax are the minimum and maximum one way delays experienced
throughout the connection, LP_WITHIN_INF indicates if TCP-LP is in inference
phase or not

More information (paper): http://cs.northwestern.edu/~akuzma/rice/doc/TCP-LP.pdf

Validation
++++++++++

The following tests are found in the ``src/internet/test`` directory.  In
general, TCP tests inherit from a class called :cpp:class:`TcpGeneralTest`,
which provides common operations to set up test scenarios involving TCP
objects.  For more information on how to write new tests, see the
section below on :ref:`Writing-tcp-tests`.

* **tcp:** Basic transmission of string of data from client to server
* **tcp-bytes-in-flight-test:** TCP correctly estimates bytes in flight under loss conditions
* **tcp-cong-avoid-test:** TCP congestion avoidance for different packet sizes
* **tcp-datasentcb:** Check TCP's 'data sent' callback
* **tcp-endpoint-bug2211-test:** A test for an issue that was causing stack overflow
* **tcp-fast-retr-test:** Fast Retransmit testing
* **tcp-header:** Unit tests on the TCP header
* **tcp-highspeed-test:** Unit tests on the Highspeed congestion control
* **tcp-htcp-test:** Unit tests on the H-TCP congestion control
* **tcp-hybla-test:** Unit tests on the Hybla congestion control
* **tcp-vegas-test:** Unit tests on the Vegas congestion control
* **tcp-veno-test:** Unit tests on the Veno congestion control
* **tcp-scalable-test:** Unit tests on the Scalable congestion control
* **tcp-bic-test:** Unit tests on the BIC congestion control
* **tcp-yeah-test:** Unit tests on the YeAH congestion control
* **tcp-illinois-test:** Unit tests on the Illinois congestion control
* **tcp-ledbat-test:** Unit tests on the LEDBAT congestion control
* **tcp-lp-test:** Unit tests on the TCP-LP congestion control
* **tcp-option:** Unit tests on TCP options
* **tcp-pkts-acked-test:** Unit test the number of time that PktsAcked is called
* **tcp-rto-test:** Unit test behavior after a RTO timeout occurs
* **tcp-rtt-estimation-test:** Check RTT calculations, including retransmission cases
* **tcp-slow-start-test:** Check behavior of slow start
* **tcp-timestamp:** Unit test on the timestamp option
* **tcp-wscaling:** Unit test on the window scaling option
* **tcp-zero-window-test:** Unit test persist behavior for zero window conditions
* **tcp-close-test:** Unit test on the socket closing: both receiver and sender have to close their socket when all bytes are transferred

Several tests have dependencies outside of the ``internet`` module, so they
are located in a system test directory called ``src/test/ns3tcp``.  Three
of these six tests involve use of the Network Simulation Cradle, and are
disabled if NSC is not enabled in the build.  

* **ns3-tcp-cwnd:** Check to see that ns-3 TCP congestion control works against liblinux2.6.26.so implementation
* **ns3-tcp-interoperability:** Check to see that ns-3 TCP interoperates with liblinux2.6.26.so implementation
* **ns3-tcp-loss:** Check behavior of ns-3 TCP upon packet losses
* **nsc-tcp-loss:** Check behavior of NSC TCP upon packet losses
* **ns3-tcp-no-delay:** Check that ns-3 TCP Nagle"s algorithm works correctly and that it can be disabled
* **ns3-tcp-socket:** Check that ns-3 TCP successfully transfers an application data write of various sizes
* **ns3-tcp-state:** Check the operation of the TCP state machine for several cases
 
Several TCP validation test results can also be found in the
`wiki page <http://www.nsnam.org/wiki/New_TCP_Socket_Architecture>`_ 
describing this implementation.

Writing a new congestion control algorithm
++++++++++++++++++++++++++++++++++++++++++

Writing (or porting) a congestion control algorithms from scratch (or from
other systems) is a process completely separated from the internals of
TcpSocketBase.

All operations that are delegated to a congestion control are contained in
the class TcpCongestionOps. It mimics the structure tcp_congestion_ops of
Linux, and the following operations are defined:

.. code-block:: c++

  virtual std::string GetName () const;
  virtual uint32_t GetSsThresh (Ptr<const TcpSocketState> tcb, uint32_t bytesInFlight);
  virtual void IncreaseWindow (Ptr<TcpSocketState> tcb, uint32_t segmentsAcked);
  virtual void PktsAcked (Ptr<TcpSocketState> tcb, uint32_t segmentsAcked,const Time& rtt);
  virtual Ptr<TcpCongestionOps> Fork ();
  virtual void CwndEvent (Ptr<TcpSocketState> tcb, const TcpSocketState::TcpCaEvent_t event);

The most interesting methods to write are GetSsThresh and IncreaseWindow.
The latter is called when TcpSocketBase decides that it is time to increase
the congestion window. Much information is available in the Transmission
Control Block, and the method should increase cWnd and/or ssThresh based
on the number of segments acked.

GetSsThresh is called whenever the socket needs an updated value of the
slow start threshold. This happens after a loss; congestion control algorithms
are then asked to lower such value, and to return it.

PktsAcked is used in case the algorithm needs timing information (such as
RTT), and it is called each time an ACK is received.

CwndEvent is used in case the algorithm needs the state of socket during different
congestion window event.

TCP SACK and non-SACK
+++++++++++++++++++++
To avoid code duplication and the effort of maintaining two different versions
of the TCP core, namely RFC 6675 (TCP-SACK) and RFC 5681 (TCP congestion control),
we have merged RFC 6675 in the current code base. If the receiver supports the
option, the sender bases its retransmissions over the received SACK information.
However, in the absence of that option, the best it can do is to follow the RFC
5681 specification (on Fast Retransmit/Recovery) and employing NewReno
modifications in case of partial ACKs.

A similar concept is used in Linux with the function tcp_add_reno_sack.
Our implementation resides in the TcpTxBuffer class that implements a scoreboard
through two different lists of segments. TcpSocketBase actively uses the API
provided by TcpTxBuffer to query the scoreboard; please refer to the Doxygen
documentation (and to in-code comments) if you want to learn more about this
implementation.

Current limitations
+++++++++++++++++++

* TcpCongestionOps interface does not contain every possible Linux operation
* Fast retransmit / fast recovery are bound with TcpSocketBase, thereby preventing easy simulation of TCP Tahoe

.. _Writing-tcp-tests:

Writing TCP tests
+++++++++++++++++

The TCP subsystem supports automated test
cases on both socket functions and congestion control algorithms. To show
how to write tests for TCP, here we explain the process of creating a test
case that reproduces a bug (#1571 in the project bug tracker).

The bug concerns the zero window situation, which happens when the receiver can
not handle more data. In this case, it advertises a zero window, which causes
the sender to pause transmission and wait for the receiver to increase the
window.

The sender has a timer to periodically check the receiver's window: however, in
modern TCP implementations, when the receiver has freed a "significant" amount
of data, the receiver itself sends an "active" window update, meaning that
the transmission could be resumed. Nevertheless, the sender timer is still
necessary because window updates can be lost.

.. note::
   During the text, we will assume some knowledge about the general design
   of the TCP test infrastructure, which is explained in detail into the
   Doxygen documentation. As a brief summary, the strategy is to have a class
   that sets up a TCP connection, and that calls protected members of itself.
   In this way, subclasses can implement the necessary members, which will
   be called by the main TcpGeneralTest class when events occour. For example,
   after processing an ACK, the method ProcessedAck will be invoked. Subclasses
   interested in checking some particular things which must have happened during
   an ACK processing, should implement the ProcessedAck method and check
   the interesting values inside the method. To get a list of available methods,
   please check the Doxygen documentation.

We describe the writing of two test case, covering both situations: the
sender's zero-window probing and the receiver "active" window update. Our focus
will be on dealing with the reported problems, which are:

* an ns-3 receiver does not send "active" window update when its receive buffer
  is being freed;
* even if the window update is artificially crafted, the transmission does not
  resume.

However, other things should be checked in the test:

* Persistent timer setup
* Persistent timer teardown if rWnd increases

To construct the test case, one first derives from the TcpGeneralTest class:

The code is the following:

.. code-block:: c++

   TcpZeroWindowTest::TcpZeroWindowTest (const std::string &desc)
      : TcpGeneralTest (desc)
   {
   }

Then, one should define the general parameters for the TCP connection, which
will be one-sided (one node is acting as SENDER, while the other is acting as
RECEIVER):

* Application packet size set to 500, and 20 packets in total (meaning a stream
  of 10k bytes)
* Segment size for both SENDER and RECEIVER set to 500 bytes
* Initial slow start threshold set to UINT32_MAX
* Initial congestion window for the SENDER set to 10 segments (5000 bytes)
* Congestion control: NewReno

We have also to define the link properties, because the above definition does
not work for every combination of propagation delay and sender application behavior.

* Link one-way propagation delay: 50 ms
* Application packet generation interval: 10 ms
* Application starting time: 20 s after the starting point

To define the properties of the environment (e.g. properties which should be
set before the object creation, such as propagation delay) one next implements
ehe method ConfigureEnvironment:

.. code-block:: c++

   void
   TcpZeroWindowTest::ConfigureEnvironment ()
   {
     TcpGeneralTest::ConfigureEnvironment ();
     SetAppPktCount (20);
     SetMTU (500);
     SetTransmitStart (Seconds (2.0));
     SetPropagationDelay (MilliSeconds (50));
   }

For other properties, set after the object creation, one can use 
ConfigureProperties ().
The difference is that some values, such as initial congestion window
or initial slow start threshold, are applicable only to a single instance, not
to every instance we have. Usually, methods that requires an id and a value
are meant to be called inside ConfigureProperties (). Please see the doxygen
documentation for an exhaustive list of the tunable properties.

.. code-block:: c++

   void
   TcpZeroWindowTest::ConfigureProperties ()
   {
     TcpGeneralTest::ConfigureProperties ();
     SetInitialCwnd (SENDER, 10);
   }

To see the default value for the experiment, please see the implementation of
both methods inside TcpGeneralTest class.

.. note::
   If some configuration parameters are missing, add a method called
   "SetSomeValue" which takes as input the value only (if it is meant to be
   called inside ConfigureEnvironment) or the socket and the value (if it is
   meant to be called inside ConfigureProperties).

To define a zero-window situation, we choose (by design) to initiate the connection
with a 0-byte rx buffer. This implies that the RECEIVER, in its first SYN-ACK,
advertises a zero window. This can be accomplished by implementing the method
CreateReceiverSocket, setting an Rx buffer value of 0 bytes (at line 6 of the
following code):

.. code-block:: c++
   :linenos:
   :emphasize-lines: 6,7,8

   Ptr<TcpSocketMsgBase>
   TcpZeroWindowTest::CreateReceiverSocket (Ptr<Node> node)
   {
     Ptr<TcpSocketMsgBase> socket = TcpGeneralTest::CreateReceiverSocket (node);

     socket->SetAttribute("RcvBufSize", UintegerValue (0));
     Simulator::Schedule (Seconds (10.0),
                          &TcpZeroWindowTest::IncreaseBufSize, this);

     return socket;
   }

Even so, to check the active window update, we should schedule an increase
of the buffer size. We do this at line 7 and 8, scheduling the function
IncreaseBufSize.

.. code-block:: c++

   void
   TcpZeroWindowTest::IncreaseBufSize ()
   {
     SetRcvBufSize (RECEIVER, 2500);
   }

Which utilizes the SetRcvBufSize method to edit the RxBuffer object of the
RECEIVER. As said before, check the Doxygen documentation for class TcpGeneralTest
to be aware of the various possibilities that it offers.

.. note::
   By design, we choose to mantain a close relationship between TcpSocketBase
   and TcpGeneralTest: they are connected by a friendship relation. Since
   friendship is not passed through inheritance, if one discovers that one
   needs to access or to modify a private (or protected) member of TcpSocketBase,
   one can do so by adding a method in the class TcpGeneralSocket. An example
   of such method is SetRcvBufSize, which allows TcpGeneralSocket subclasses
   to forcefully set the RxBuffer size.

   .. code-block:: c++

      void
      TcpGeneralTest::SetRcvBufSize (SocketWho who, uint32_t size)
      {
        if (who == SENDER)
          {
            m_senderSocket->SetRcvBufSize (size);
          }
        else if (who == RECEIVER)
          {
            m_receiverSocket->SetRcvBufSize (size);
          }
        else
          {
            NS_FATAL_ERROR ("Not defined");
          }
      }

Next, we can start to follow the TCP connection:

#. At time 0.0 s the connection is opened sender side, with a SYN packet sent from
   SENDER to RECEIVER
#. At time 0.05 s the RECEIVER gets the SYN and replies with a SYN-ACK
#. At time 0.10 s the SENDER gets the SYN-ACK and replies with a SYN.

While the general structure is defined, and the connection is started,
we need to define a way to check the rWnd field on the segments. To this aim,
we can implement the methods Rx and Tx in the TcpGeneralTest subclass,
checking each time the actions of the RECEIVER and the SENDER. These methods are
defined in TcpGeneralTest, and they are attached to the Rx and Tx traces in the
TcpSocketBase. One should write small tests for every detail that one wants to ensure during the
connection (it will prevent the test from changing over the time, and it ensures
that the behavior will stay consistent through releases). We start by ensuring that
the first SYN-ACK has 0 as advertised window size:

.. code-block:: c++

   void
   TcpZeroWindowTest::Tx(const Ptr<const Packet> p, const TcpHeader &h, SocketWho who)
   {
     ...
     else if (who == RECEIVER)
       {
         NS_LOG_INFO ("\tRECEIVER TX " << h << " size " << p->GetSize());

         if (h.GetFlags () & TcpHeader::SYN)
           {
             NS_TEST_ASSERT_MSG_EQ (h.GetWindowSize(), 0,
                                    "RECEIVER window size is not 0 in the SYN-ACK");
           }
       }
       ....
    }

Pratically, we are checking that every SYN packet sent by the RECEIVER has the
advertised window set to 0. The same thing is done also by checking, in the Rx
method, that each SYN received by SENDER has the advertised window set to 0.
Thanks to the log subsystem, we can print what is happening through messages.
If we run the experiment, enabling the logging, we can see the following:

.. code-block:: bash

   ./waf shell
   gdb --args ./build/utils/ns3-dev-test-runner-debug --test-name=tcp-zero-window-test --stop-on-failure --fullness=QUICK --assert-on-failure --verbose
   (gdb) run

   0.00s TcpZeroWindowTestSuite:Tx(): 0.00	SENDER TX 49153 > 4477 [SYN] Seq=0 Ack=0 Win=32768 ns3::TcpOptionWinScale(2) ns3::TcpOptionTS(0;0) size 36
   0.05s TcpZeroWindowTestSuite:Rx(): 0.05	RECEIVER RX 49153 > 4477 [SYN] Seq=0 Ack=0 Win=32768 ns3::TcpOptionWinScale(2) ns3::TcpOptionTS(0;0) ns3::TcpOptionEnd(EOL) size 0
   0.05s TcpZeroWindowTestSuite:Tx(): 0.05	RECEIVER TX 4477 > 49153 [SYN|ACK] Seq=0 Ack=1 Win=0 ns3::TcpOptionWinScale(0) ns3::TcpOptionTS(50;0) size 36
   0.10s TcpZeroWindowTestSuite:Rx(): 0.10	SENDER RX 4477 > 49153 [SYN|ACK] Seq=0 Ack=1 Win=0 ns3::TcpOptionWinScale(0) ns3::TcpOptionTS(50;0) ns3::TcpOptionEnd(EOL) size 0
   0.10s TcpZeroWindowTestSuite:Tx(): 0.10	SENDER TX 49153 > 4477 [ACK] Seq=1 Ack=1 Win=32768 ns3::TcpOptionTS(100;50) size 32
   0.15s TcpZeroWindowTestSuite:Rx(): 0.15	RECEIVER RX 49153 > 4477 [ACK] Seq=1 Ack=1 Win=32768 ns3::TcpOptionTS(100;50) ns3::TcpOptionEnd(EOL) size 0
   (...)

The output is cut to show the threeway handshake. As we can see from the headers,
the rWnd of RECEIVER is set to 0, and thankfully our tests are not failing.
Now we need to test for the persistent timer, which sould be started by
the SENDER after it receives the SYN-ACK. Since the Rx method is called before
any computation on the received packet, we should utilize another method, namely
ProcessedAck, which is the method called after each processed ACK. In the
following, we show how to check if the persistent event is running after the
processing of the SYN-ACK:

.. code-block:: c++

   void
   TcpZeroWindowTest::ProcessedAck (const Ptr<const TcpSocketState> tcb,
                                    const TcpHeader& h, SocketWho who)
   {
     if (who == SENDER)
       {
         if (h.GetFlags () & TcpHeader::SYN)
           {
             EventId persistentEvent = GetPersistentEvent (SENDER);
             NS_TEST_ASSERT_MSG_EQ (persistentEvent.IsRunning (), true,
                                    "Persistent event not started");
           }
       }
    }

Since we programmed the increase of the buffer size after 10 simulated seconds,
we expect the persistent timer to fire before any rWnd changes. When it fires,
the SENDER should send a window probe, and the receiver should reply reporting
again a zero window situation. At first, we investigates on what the sender sends:

..  code-block:: c++
    :linenos:
    :emphasize-lines: 1,6,7,11

      if (Simulator::Now ().GetSeconds () <= 6.0)
        {
          NS_TEST_ASSERT_MSG_EQ (p->GetSize () - h.GetSerializedSize(), 0,
                                 "Data packet sent anyway");
        }
      else if (Simulator::Now ().GetSeconds () > 6.0 &&
               Simulator::Now ().GetSeconds () <= 7.0)
        {
          NS_TEST_ASSERT_MSG_EQ (m_zeroWindowProbe, false, "Sent another probe");

          if (! m_zeroWindowProbe)
            {
              NS_TEST_ASSERT_MSG_EQ (p->GetSize () - h.GetSerializedSize(), 1,
                                     "Data packet sent instead of window probe");
              NS_TEST_ASSERT_MSG_EQ (h.GetSequenceNumber(), SequenceNumber32 (1),
                                     "Data packet sent instead of window probe");
              m_zeroWindowProbe = true;
            }
        }

We divide the events by simulated time. At line 1, we check everything that
happens before the 6.0 seconds mark; for instance, that no data packets are sent,
and that the state remains OPEN for both sender and receiver.

Since the persist timeout is initialized at 6 seconds (excercise left for the
reader: edit the test, getting this value from the Attribute system), we need
to check (line 6) between 6.0 and 7.0 simulated seconds that the probe is sent.
Only one probe is allowed, and this is the reason for the check at line 11.

.. code-block:: c++
   :linenos:
   :emphasize-lines: 6,7

   if (Simulator::Now ().GetSeconds () > 6.0 &&
       Simulator::Now ().GetSeconds () <= 7.0)
     {
       NS_TEST_ASSERT_MSG_EQ (h.GetSequenceNumber(), SequenceNumber32 (1),
                              "Data packet sent instead of window probe");
       NS_TEST_ASSERT_MSG_EQ (h.GetWindowSize(), 0,
                              "No zero window advertised by RECEIVER");
     }

For the RECEIVER, the interval between 6 and 7 seconds is when the zero-window
segment is sent.

Other checks are redundant; the safest approach is to deny any other packet
exchange between the 7 and 10 seconds mark.

.. code-block:: c++

   else if (Simulator::Now ().GetSeconds () > 7.0 &&
            Simulator::Now ().GetSeconds () < 10.0)
     {
       NS_FATAL_ERROR ("No packets should be sent before the window update");
     }

The state checks are performed at the end of the methods, since they are valid
in every condition:

.. code-block:: c++

   NS_TEST_ASSERT_MSG_EQ (GetCongStateFrom (GetTcb(SENDER)), TcpSocketState::CA_OPEN,
                          "Sender State is not OPEN");
   NS_TEST_ASSERT_MSG_EQ (GetCongStateFrom (GetTcb(RECEIVER)), TcpSocketState::CA_OPEN,
                          "Receiver State is not OPEN");

Now, the interesting part in the Tx method is to check that after the 10.0
seconds mark (when the RECEIVER sends the active window update) the value of
the window should be greater than zero (and precisely, set to 2500):

.. code-block:: c++

   else if (Simulator::Now().GetSeconds() >= 10.0)
     {
       NS_TEST_ASSERT_MSG_EQ (h.GetWindowSize(), 2500,
                              "Receiver window not updated");
     }

To be sure that the sender receives the window update, we can use the Rx
method:

.. code-block:: c++
   :linenos:
   :emphasize-lines: 5

   if (Simulator::Now().GetSeconds() >= 10.0)
     {
       NS_TEST_ASSERT_MSG_EQ (h.GetWindowSize(), 2500,
                              "Receiver window not updated");
       m_windowUpdated = true;
     }

We check every packet after the 10 seconds mark to see if it has the
window updated. At line 5, we also set to true a boolean variable, to check
that we effectively reach this test.

Last but not least, we implement also the NormalClose() method, to check that
the connection ends with a success:

.. code-block:: c++

   void
   TcpZeroWindowTest::NormalClose (SocketWho who)
   {
     if (who == SENDER)
       {
         m_senderFinished = true;
       }
     else if (who == RECEIVER)
       {
         m_receiverFinished = true;
       }
   }

The method is called only if all bytes are transmitted successfully. Then, in
the method FinalChecks(), we check all variables, which should be true (which
indicates that we have perfectly closed the connection).

.. code-block:: c++

   void
   TcpZeroWindowTest::FinalChecks ()
   {
     NS_TEST_ASSERT_MSG_EQ (m_zeroWindowProbe, true,
                            "Zero window probe not sent");
     NS_TEST_ASSERT_MSG_EQ (m_windowUpdated, true,
                            "Window has not updated during the connection");
     NS_TEST_ASSERT_MSG_EQ (m_senderFinished, true,
                            "Connection not closed successfully (SENDER)");
     NS_TEST_ASSERT_MSG_EQ (m_receiverFinished, true,
                            "Connection not closed successfully (RECEIVER)");
   }

To run the test, the usual way is

.. code-block:: bash

   ./test.py -s tcp-zero-window-test

   PASS: TestSuite tcp-zero-window-test
   1 of 1 tests passed (1 passed, 0 skipped, 0 failed, 0 crashed, 0 valgrind errors)

To see INFO messages, use a combination of ./waf shell and gdb (really useful):

.. code-block:: bash


    ./waf shell && gdb --args ./build/utils/ns3-dev-test-runner-debug --test-name=tcp-zero-window-test --stop-on-failure --fullness=QUICK --assert-on-failure --verbose

and then, hit "Run".

.. note::
   This code magically runs without any reported errors; however, in real cases,
   when you discover a bug you should expect the existing test to fail (this
   could indicate a well-written test and a bad-writted model, or a bad-written
   test; hopefull the first situation). Correcting bugs is an iterative
   process. For instance, commits created to make this test case running without
   errors are 11633:6b74df04cf44, (others to be merged).

Network Simulation Cradle
*************************

The `Network Simulation Cradle (NSC) <http://www.wand.net.nz/~stj2/nsc/>`_ is a
framework for wrapping real-world network code into simulators, allowing
simulation of real-world behavior at little extra cost. This work has been
validated by comparing situations using a test network with the same situations
in the simulator. To date, it has been shown that the NSC is able to produce
extremely accurate results.  NSC supports four real world stacks: FreeBSD,
OpenBSD, lwIP and Linux. Emphasis has been placed on not changing any of the
network stacks by hand. Not a single line of code has been changed in the
network protocol implementations of any of the above four stacks. However, a
custom C parser was built to programmatically change source code.

NSC has previously been ported to |ns2| and OMNeT++, and was 
was added to |ns3| in September 2008 (ns-3.2 release).  This section 
describes the |ns3| port of NSC and how to use it.

To some extent, NSC has been superseded by the Linux kernel support within 
`Direct Code Execution (DCE) <http://www.nsnam.org/docs/dce/manual/singlehtml/index.html>`__.  However, NSC is still available through the bake build
system.  NSC supports Linux kernels 2.6.18 and 2.6.26, but newer
versions of the kernel have not been ported.  

Prerequisites
+++++++++++++

Presently, NSC has been tested and shown to work on these platforms:
Linux i386 and Linux x86-64.  NSC does not support powerpc.  Use on
FreeBSD or OS X is unsupported (although it may be able to work).

Building NSC requires the packages flex and bison.  

Configuring and Downloading
+++++++++++++++++++++++++++

As of ns-3.17 or later, NSC must either be downloaded separately from
its own repository, or downloading when using the 
`bake build system <http://www.nsnam.org/docs/tutorial/html/getting-started.html#downloading-ns3-using-bake>`_ of 
|ns3|.  

For ns-3.17 or later releases, when using bake, one must configure NSC as 
part of an "allinone" configuration, such as:

.. sourcecode:: bash

  $ cd bake
  $ python bake.py configure -e ns-allinone-3.19
  $ python bake.py download
  $ python bake.py build

Instead of a released version, one may use the ns-3 development version
by specifying "ns-3-allinone" to the configure step above.

NSC may also be downloaded from 
`its download site <http://research.wand.net.nz/software/nsc.php>`_ 
using Mercurial:

.. sourcecode:: bash

  $ hg clone https://secure.wand.net.nz/mercurial/nsc

Prior to the ns-3.17 release, NSC was included in the allinone tarball and
the released version did not need to be separately downloaded.

Building and validating
+++++++++++++++++++++++

NSC may be built as part of the bake build process; alternatively, one
may build NSC by itself using its build system; e.g.:

.. sourcecode:: bash

  $ cd nsc-dev
  $ python scons.py

Once NSC has been built either manually or through the bake system, change
into the |ns3| source directory and try running the following configuration:

.. sourcecode:: bash

  $ ./waf configure

If NSC has been previously built and found by waf, then you will see:

.. sourcecode:: bash

  Network Simulation Cradle     : enabled

If NSC has not been found, you will see:

.. sourcecode:: bash

  Network Simulation Cradle     : not enabled (NSC not found (see option --with-nsc))

In this case, you must pass the relative or absolute path to the NSC libraries
with the "--with-nsc" configure option; e.g.

.. sourcecode:: bash

  $ ./waf configure --with-nsc=/path/to/my/nsc/directory

For |ns3| releases prior to the ns-3.17 release, using the ``build.py`` 
script in ns-3-allinone directory, NSC will be built by default unless the 
platform does not support it. To explicitly disable it when building |ns3|, 
type:

.. sourcecode:: bash

  $ ./waf configure --enable-examples --enable-tests --disable-nsc

If waf detects NSC, then building |ns3| with NSC is performed the same way
with waf as without it.  Once |ns3| is built, try running the following 
test suite:

.. sourcecode:: bash

    $ ./test.py -s ns3-tcp-interoperability

If NSC has been successfully built, the following test should show up 
in the results:

.. sourcecode:: text

    PASS TestSuite ns3-tcp-interoperability

This confirms that NSC is ready to use.

Usage
+++++

There are a few example files.  Try:

.. sourcecode:: bash

    $ ./waf --run tcp-nsc-zoo
    $ ./waf --run tcp-nsc-lfn

These examples will deposit some ``.pcap`` files in your directory,
which can be examined by tcpdump or wireshark.

Let's look at the ``examples/tcp/tcp-nsc-zoo.cc`` file for some typical
usage. How does it differ from using native |ns3| TCP? There is one main
configuration line, when using NSC and the |ns3| helper API, that needs to be
set::

  InternetStackHelper internetStack;

  internetStack.SetNscStack ("liblinux2.6.26.so");
  // this switches nodes 0 and 1 to NSCs Linux 2.6.26 stack.
  internetStack.Install (n.Get(0));
  internetStack.Install (n.Get(1));


The key line is the ``SetNscStack``.  This tells the InternetStack
helper to aggregate instances of NSC TCP instead of native |ns3| TCP
to the remaining nodes.  It is important that this function be called
**before** calling the ``Install()`` function, as shown above.

Which stacks are available to use? Presently, the focus has been on
Linux 2.6.18 and Linux 2.6.26 stacks for |ns3|. To see which stacks
were built, one can execute the following find command at the |ns3| top level
directory:

.. sourcecode:: bash

    $ find nsc -name "*.so" -type f 
    nsc/linux-2.6.18/liblinux2.6.18.so
    nsc/linux-2.6.26/liblinux2.6.26.so

This tells us that we may either pass the library name liblinux2.6.18.so or
liblinux2.6.26.so to the above configuration step.

Stack configuration
+++++++++++++++++++

NSC TCP shares the same configuration attributes that are common across TCP
sockets, as described above and documented in `Doxygen
<http://www.nsnam.org/doxygen/classns3_1_1_tcp_socket.html>`_

Additionally, NSC TCP exports a lot of configuration variables into the 
|ns3| attributes system, via a `sysctl <http://en.wikipedia.org/wiki/Sysctl>`_-like interface. In the ``examples/tcp/tcp-nsc-zoo`` example, you
can see the following configuration::


  // this disables TCP SACK, wscale and timestamps on node 1 (the attributes 
    represent sysctl-values).
  Config::Set ("/NodeList/1/$ns3::Ns3NscStack<linux2.6.26>/net.ipv4.tcp_sack", 
    StringValue ("0"));
  Config::Set ("/NodeList/1/$ns3::Ns3NscStack<linux2.6.26>/net.ipv4.tcp_timestamps", 
  StringValue ("0"));
  Config::Set ("/NodeList/1/$ns3::Ns3NscStack<linux2.6.26>/net.ipv4.tcp_window_scaling", 
  StringValue ("0"));

These additional configuration variables are not available to native |ns3| TCP.

Also note that default values for TCP attributes in |ns3| TCP may differ from the nsc TCP implementation.  Specifically in |ns3|:

  1) TCP default MSS is 536
  2) TCP Delayed Ack count is 2 
		
Therefore when making comparisons between results obtained using nsc and |ns3| TCP, care must be taken to ensure these values are set appropriately.  See /examples/tcp/tcp-nsc-comparision.cc for an example.

NSC API
+++++++

This subsection describes the API that NSC presents to |ns3| or any other
simulator. NSC provides its API in the form of a number of classes that are
defined in ``sim/sim_interface.h`` in the nsc directory.

* **INetStack** INetStack contains the 'low level' operations for the operating
  system network stack, e.g. in and output functions from and to the network
  stack (think of this as the 'network driver interface'. There are also
  functions to create new TCP or UDP sockets.
* **ISendCallback** This is called by NSC when a packet should be sent out to
  the network. This simulator should use this callback to re-inject the packet
  into the simulator so the actual data can be delivered/routed to its
  destination, where it will eventually be handed into Receive() (and eventually
  back to the receivers NSC instance via INetStack->if_receive() ).
* **INetStreamSocket** This is the structure defining a particular connection
  endpoint (file descriptor). It contains methods to operate on this endpoint,
  e.g. connect, disconnect, accept, listen, send_data/read_data, ...
* **IInterruptCallback** This contains the wakeup callback, which is called by
  NSC whenever something of interest happens. Think of wakeup() as a replacement
  of the operating systems wakeup function: Whenever the operating system would
  wake up a process that has been waiting for an operation to complete (for
  example the TCP handshake during connect()), NSC invokes the wakeup() callback
  to allow the simulator to check for state changes in its connection endpoints. 

ns-3 implementation
+++++++++++++++++++

The |ns3| implementation makes use of the above NSC API, and is implemented as
follows.

The three main parts are:

* :cpp:class:`ns3::NscTcpL4Protocol`:  a subclass of Ipv4L4Protocol (and two nsc
  classes: ISendCallback and IInterruptCallback)
* :cpp:class:`ns3::NscTcpSocketImpl`: a subclass of TcpSocket 
* :cpp:class:`ns3::NscTcpSocketFactoryImpl`:  a factory to create new NSC
  sockets

``src/internet/model/nsc-tcp-l4-protocol`` is the main class. Upon
Initialization, it loads an nsc network stack to use (via dlopen()). Each
instance of this class may use a different stack. The stack (=shared library) to
use is set using the SetNscLibrary() method (at this time its called indirectly
via the internet stack helper). The nsc stack is then set up accordingly (timers
etc). The NscTcpL4Protocol::Receive() function hands the packet it receives
(must be a complete tcp/ip packet) to the nsc stack for further processing.  To
be able to send packets, this class implements the nsc send_callback method.
This method is called by nsc whenever the nsc stack wishes to send a packet out
to the network. Its arguments are a raw buffer, containing a complete TCP/IP
packet, and a length value. This method therefore has to convert the raw data to
a Ptr<Packet> usable by |ns3|. In order to avoid various ipv4 header issues,
the nsc ip header is not included. Instead, the tcp header and the actual
payload are put into the Ptr<Packet>, after this the Packet is passed down to
layer 3 for sending the packet out (no further special treatment is needed in
the send code path).

This class calls ``ns3::NscTcpSocketImpl`` both from the nsc wakeup() callback
and from the Receive path (to ensure that possibly queued data is scheduled for
sending).

``src/internet/model/nsc-tcp-socket-impl`` implements the nsc socket interface.
Each instance has its own nscTcpSocket. Data that is Send() will be handed to
the nsc stack via m_nscTcpSocket->send_data(). (and not to nsc-tcp-l4, this is
the major difference compared to |ns3| TCP). The class also queues up data that
is Send() before the underlying descriptor has entered an ESTABLISHED state.
This class is called from the nsc-tcp-l4 class, when the nsc-tcp-l4 wakeup()
callback is invoked by nsc. nsc-tcp-socket-impl then checks the current
connection state (SYN_SENT, ESTABLISHED, LISTEN...) and schedules appropriate
callbacks as needed, e.g. a LISTEN socket will schedule Accept to see if a new
connection must be accepted, an ESTABLISHED socket schedules any pending data
for writing, schedule a read callback, etc.

Note that ``ns3::NscTcpSocketImpl`` does not interact with nsc-tcp directly:
instead, data is redirected to nsc. nsc-tcp calls the nsc-tcp-sockets of a node
when its wakeup callback is invoked by nsc. 

Limitations
+++++++++++

* NSC only works on single-interface nodes; attempting to run it on a
  multi-interface node will cause a program error.  
* Cygwin and OS X PPC are not supported; OS X Intel is not supported but may work
* The non-Linux stacks of NSC are not supported in |ns3|
* Not all socket API callbacks are supported

For more information, see `this wiki page <http://www.nsnam.org/wiki/Network_Simulation_Cradle_Integration>`_.
