Int. J. Communications, Network and System Sciences, 2011, 4, 256-265
doi:10.4236/ijcns.2011.44031 Published Online April 2011 (
Copyright © 2011 SciRes. IJCNS
A New Transmission Control Protocol for
Satellite Networks
Liang Yu, Gang Zhou
College of Comp ut er Sci ence, Sichua n U ni versi t y, Chengdu, China
Received February 16, 2011; revise d M arch 16, 2011; accepted March 22, 2011
According to technical statistics, current TCP protocols with approximately 80% Internet applications run on
perform very well on wired networks. However, due to the effects of long propagation delay, great band-
width asymmetry, high sporadic Bit Error Rate (BER) and etc., TCP performance degrades obviously on the
satellite communication networks. To avoid the problems, TP-S, a novel transport control protocol, is intro-
duced for satellite IP networks. Firstly, in order to increase the increment speed of Congestion Window
(cwnd) at the beginning of data transmission, the traditional Slow Start strategy is replaced by a new strategy,
known as Super Start. Secondly, a new packet lost discriminated scheme based on IP packets alternately
sending with different priority is used in the protocol to decouple congestion decision from errors. Thirdly,
bandwidth asymmetry problem is avoided by adopting Modified NACK (M-NACK) in receiving ends,
which is sent periodically. In addition, the sending strategy in routers is also modified along with other’s
changes to support the protocol. Finally, the simulation experiments show that the new protocol can not only
significantly enhance throughput performance, but also reduce sharply bandwidth used in the reverse path as
compared with traditional TCP protocols and those protocols, which are recently proposed for satellite IP
Keywords: Satellite Communication, TCP, Long Propagation Delay, Bandwidth Asymmetry
1. Introduction
With the rapid development of Internet, TCP protocols
has great adjusted and improved. Since Van Jacobson
propose four algorithms [1]: Slow Start, Avoiding Con-
gestion, Fast-retransmission and Fast-restoration, three
TCP protocols TCP-Tahoe, TCP-Reno [2] and TCP-New
Reno [3] have gradually become the current main ver-
sion of Internet. These TCP protocols have their own
traits. TCP-Tahoe protocol hasn’t Fast-restoration algo-
rithm that means it will directly access to the Slow Start
initial state when Fast-retransmission with lost data.
TCP-Reno protocol immediately access to Fast-restora-
tion stage after Fast-retransmission rather than return to
the Slow Start stage. TCP-New Reno protocol has modi-
fied the Fast-restoration algorithms of TCP-Reno proto-
col, and it avoids the Congestion Window and Slow Start
(ssthresh) from halving problem, which caused by many
lost data in Sending Window. Whether TCP-Tahoe or
TCP-New Reno protocol, these protocols are designed to
achieve that the Error Rate should be lower than 10–8 in
wired ground network under the assumption that conges-
tion is the only reason caused data loss and damage. Ob-
viously, the assumption is not suit for the satellite net-
work [4]. Satellite Channel has the following characteris-
tics: long propagation delay (Synchronous satellite chan-
nel Return-Transmission Time (RTT) is about 550 ms),
great bandwidth asymmetry and high sporadic Bit Error
Rate (BER) [5,6] .All these traits have serious impact on
the transmission performance of TCP protocols [7-9].
The paper proposes a novel transmission control pro-
tocol which is suit for satellite network based on the
traits of the satellite network. It not only can increase the
Congestion Window value rapidly after the connection
set up, but also can distinguish the specific reasons for
lost data. Thereby, it adopts related control strategy of
Sending Window. Sending-Client applies cycle Sent-
Response strategy in order to solve reverse link band-
width asymmetry problem. According to simulation and
analysis, the new protocol not only can increase the
throughput for the forward link, but also greatly reduces
L. YU ET AL.257
bandwidth occupancy rate of the reverse link, compared
with the traditional TCP protocols and transmission con-
trol protocol which is proposed according to traits of
satellite links in recent year.
One of the most interesting aspects of TCP is a mecha-
nism for congestion control. Recall that in the Internet,
delay or packet loss is more likely to be caused by con-
gestion than a hardware failure, and that retransmission
can exacerbate the problem of congestion by injecting
additional copies of a packet. To avoid congestion col-
lapse, TCP uses changes in delay as a measure of con-
gestion, and responds to congestion by reducing the rate
at which it retransmits data. TCP’s four intertwined con-
gestion control algorithms are slow start, congestion
avoidance, fast retransmit, and fast recovery.
1.1. Slow Start and Congestion Avoidance
TCP uses a special congestion control mechanism when
starting a new connection or when a message is lost. In-
stead of transmitting enough data to fill the receiver’s
buffer (i.e., the receiver’s window size), TCP begins by
sending a single message containing data. If an ac-
knowledgement arrives without additional loss, TCP
doubles the amount of data being sent and sends two
additional messages. If both acknowledgements arrive,
TCP sends four messages, and so on. The exponential
increase continues until TCP is sending half of the re-
ceiver’s advertised window. When one-half of the origi-
nal window size is reached, TCP slows the rate of in-
crease, and increases the window size linearly as long as
congestion does not occur. The approach is known as
slow start.
TCP’s congestion control mechanisms respond well to
increases in traffic. By backing off quickly, TCP is able
to alleviate congestion. In essence, TCP avoids adding
retransmissions when the Internet becomes congested.
More important, if all TCPs follow the standard, the
congestion control scheme means that all senders back
off when congestion occurs and congestion collapse is
When the load offered to any network is more than it
can handle, congestion builds up. The Internet is no ex-
ception. In this section we will discuss algorithms that
have been developed over the past quarter of a century to
deal with congestion. Although the network layer also
tries to manage congestion, most of the heavy lifting is
done by TCP because the real solution to congestion is to
slow down the data rate.
In theory, congestion can be dealt with by employing a
principle borrowed from physics: the law of conservation
of packets. The idea is to refrain from injecting a new
packet into the network until an old one leaves (i.e., is
delivered). TCP attempts to achieve this goal by dy-
namically manipulating the window size. The first step in
managing congestion is detecting it. In the old days, de-
tecting congestion was difficult. A timeout caused by a
lost packet could have been caused by either noise on a
transmission line or packet discard at a congested router.
Consequently, most transmission timeouts on the Internet
are due to congestion. All the Internet TCP algorithms
assume that timeouts are caused by congestion and mon-
itor timeouts for signs of trouble the way miners watch
their canaries.
The slow start and congestion avoidance algorithms
must be used by a TCP sender to control the amount of
outstanding data being injected into the network. To im-
plement these algorithms, two variables are added to the
TCP per-connection state. The congestion window (cwnd)
is a sender-side limit on the amount of data the sender
can transmit into the network before receiving an ac-
knowledgment (ACK), while the receiver’s advertised
window (rwnd) is a receiver-side limit on the amount of
outstanding data. The minimum of cwnd and rwnd gov-
erns data transmission.
Another state variable, the slow start threshold
(ssthresh), is used to determine whether the slow start or
congestion avoidance algorithm is used to control data
transmission, as discussed below.
Beginning transmission into a network with unknown
conditions requires TCP to slowly probe the network to
determine the available capacity, in order to avoid con-
gesting the network with an inappropriately large burst
of data. The slow start algorithm is used for this purpose
at the beginning of a transfer, or after repairing loss de-
tected by the retransmission timer.
IW, the initial value of cwnd, must be less than or
equal to 2*SMSS bytes and must not be more than 2
We note that a non-standard, experimental TCP exten-
sion allows that a TCP may use a larger initial window
(IW), as defined in Equation (1):
IW = min (4 × SMSS, max (2 × SMSS, 4380 bytes)) (1)
With this extension, a TCP sender may use a 3 or 4
segment initial window, provided the combined size of
the segments does not exceed 4380 bytes. We do not
allow this change as part of the standard defined by this
document. However, we include discussion of (1) in the
remainder of this document as a guideline for those ex-
perimenting with the change, rather than conforming to
the present standards for TCP congestion control.
The initial value of ssthresh may be arbitrarily high
(for example, some implementations use the size of the
advertised window), but it may be reduced in response to
congestion. The slow start algorithm is used when cwnd
< ssthresh, while the congestion avoidance algorithm is
Copyright © 2011 SciRes. IJCNS
used when cwnd > ssthresh. When cwnd and ssthresh are
equal the sender may use either slow start or congestion
During slow start, a TCP increments cwnd by at most
SMSS bytes for each ACK received that acknowledges
new data. Slow start ends when cwnd exceeds ssthresh
(or, optionally, when it reaches it, as noted above) or
when congestion is observed.
During congestion avoidance, cwnd is incremented by
1 full-sized segment per round-trip time (RTT). Conges-
tion avoidance continues until congestion is detected.
One formula commonly used to update cwnd during
congestion avoidance is given in Equation (2):
cwnd += SMSS × SMSS/cwnd (2)
This adjustment is executed on every incoming non-
duplicate ACK. Equation (2) provides an acceptable ap-
proximation to the underlying principle of increasing
cwnd by 1 full-sized segment per RTT. (Note that for a
connection in which the receiver acknowledges every
data segment, (2) proves slightly more aggressive than 1
segment per RTT, and for a receiver acknowledging
every-other packet, (2) is less aggressive.)
Another acceptable way to increase cwnd during con-
gestion avoidance is to count the number of bytes that
have been acknowledged by ACKs for new data. (A
drawback of this implementation is that it requires main-
taining an additional state variable.) When the number of
bytes acknowledged reaches cwnd, then cwnd can be
incremented by up to SMSS bytes. Note that during
congestion avoidance, cwnd must not be increased by
more than the larger of either 1 full-sized segment per
RTT, or the value computed using Equation (2).
When a TCP sender detects segment loss using the re-
transmission timer, the value of ssthresh must be set to
no more than the value given in Equation (3):
ssthresh = max (FlightSize/2, 2 × SMSS) (3)
As discussed above, FlightSize is the amount of out-
standing data in the network.
Furthermore, upon a timeout cwnd must be set to no
more than the loss window, LW, which equals 1 full-
sized segment (regardless of the value of IW). Therefore,
after retransmitting the dropped segment the TCP sender
uses the slow start algorithm to increase the window
from 1 full-sized segment to the new value of ssthresh, at
which point congestion avoidance again takes over.
1.2. Fast Retransmit/Fast Recovery
To handle packet loss, transport protocols use positive
acknowledgement with retransmission. Whenever a frame
arrives intact, the receiving protocol software sends a
small acknowledgement (ACK) message that reports
successful reception, the sender takes responsibility for
ensuring that each packet is transferred successfully.
Whenever it sends a packet, the sending-side protocol
software starts a timer. If an acknowledgement arrives
before the timer expires, the software cancels the timer;
if the timer expires before an acknowledgement arrives,
the software sends another copy of the packet and starts
the timer again. The action of sending a second copy is
known as retransmitting, and the copy is commonly called
a retransmission.
A TCP receiver should send an immediate duplicate
ACK when an out-of-order segment arrives. The purpose
of this ACK is to inform the sender that a segment was
received out-of-order and which sequence number is
expected. From the sender’s perspective, duplicate ACKs
can be caused by a number of network problems. First,
they can be caused by dropped segments. In this case, all
segments after the dropped segment will trigger duplicate
ACKs. Second, duplicate ACKs can be caused by the
re-ordering of data segments by the network (not a rare
event along some network paths. Finally, duplicate
ACKs can be caused by replication of ACK or data seg-
ments by the network. In addition, a TCP receiver should
send an immediate ACK when the incoming segment
fills in all or part of a gap in the sequence space. This
will generate more timely information for a sender re-
covering from a loss through a retransmission timeout, a
fast retransmit, or an experimental loss recovery algo-
rithm, such as NewReno.
The TCP sender should use the “fast retransmit” algo-
rithm to detect and repair loss, based on incoming dupli-
cate ACKs. The fast retransmit algorithm uses the arrival
of 3 duplicate ACKs (4 identical ACKs without the arri-
val of any other intervening packets) as an indication that
a segment has been lost. After receiving 3 duplicate
ACKs, TCP performs a retransmission of what appears
to be the missing segment, without waiting for the re-
transmission timer to expire.
After the fast retransmit algorithm sends what appears
to be the missing segment, the “fast recovery” algorithm
governs the transmission of new data until a non-duplica-
te ACK arrives. The reason for not performing slow start
is that the receipt of the duplicate ACKs not only indi-
cates that a segment has been lost, but also that segments
are most likely leaving the network (although a massive
segment duplication by the network can invalidate this
conclusion). In other words, since the receiver can only
generate a duplicate ACK when a segment has arrived,
that segment has left the network and is in the receiver’s
buffer, so we know it is no longer consuming network
resources. Furthermore, since the ACK “clock” is pre-
served, the TCP sender can continue to transmit new
segments (although transmission must continue using a
Copyright © 2011 SciRes. IJCNS
L. YU ET AL.259
reduced cwnd).
The fast retransmit and fast recovery algorithms are
usually implemented together as follows.
1) When the third duplicate ACK is received, set
ssthresh to no more than the value given in Equa-
tion (3).
2) Retransmit the lost segment and set cwnd to
ssthresh plus 3*SMSS. This artificially “inflates”
the congestion window by the number of segments
(three) that have left the network and which the
receiver has buffered.
3) For each additional duplicate ACK received, in-
crement cwnd by SMSS. This artificially inflates
the congestion window in order to reflect the addi-
tional segment that has left the network.
4) Transmit a segment, if allowed by the new value of
cwnd and the receiver’s advertised window.
5) When the next ACK arrives that acknowledges
new data, set cwnd to ssthresh (the value set in step
1). This is termed “deflating” the window.
This ACK should be the acknowledgment elicited by
the retransmission from step 1, one RTT after the re-
transmission (though it may arrive sooner in the presence
of significant out-of-order delivery of data segments at
the receiver).
Additionally, this ACK should acknowledge all the
intermediate segments sent between the lost segment and
the receipt of the third duplicate ACK, if none of these
were lost.
How to improve the performance of TCP protocols in
satellite network has become the focus. Through analyz-
ing theoretical and experimental, the following are fac-
tors about TCP performance that impacted on by satellite
Long propagation delay
The most throughput of TCP is restricted by RTT
and Winmax. When most sending window value is
64Kbytes and RTT is 550 ms, the most throughput
of TCP is only 1 Mb/s.
The TCP congestion window increases very slowly
in slow start and congestion avoiding stage so that
TCP’s performance is not efficiency.
High Bit Error Rate
Sporadic Bit Error Rate is prime formation in GEO
channel environment. TCP judge the network is in
congestion state when BER occur in channel. So it
not only reduces the efficiency of TCP transmis-
sion but also waste bandwidth available in satellite
When the satellite signal is obscured or multi-path
and eclipse become serious, it will occur unex-
pected Bit Error. All make TCP’s performance un-
Link bandwidth asymmetry
The ACKs congestion that caused by narrow
bandwidth in reverse channel result in throughput
of forward link decline.
The satellite and network working group in Internet
Engineering Task Force (IETF) have established some
Request for Comments (RFC) [10-12] in order to im-
prove TCP protocols performance in satellite network.
Many research organizations and institutions dedicate to
this field, so a large number of new solutions are emerg-
ing. Now many TCP protocols solutions have been pro-
posed that can be sort by TCP protocol modification
program, other layers protocol modification program,
agent design program and specialized transmission con-
trol protocol solutions program.
1) TCP protocol modification program adopts to ex-
pand the largest sending window, expand the initial
sending window, TCP header compression, con-
gestion instructions, T/TCP, multiple TCP connec-
tion, SACK and a NACK to adapt to satellite net-
work data transmission.
2) Other layer protocols modification programs are
concentrate to link layer include FEC and Auto-
Retransmission mechanism. The principle of Auto-
Retransmission mechanism is that it isolates TCP
protocol duplication responses information when
data loss and directly retransmits lost data in link
layer. All of this to make TCP protocol believes
satellite channel is without fault and the delay in-
creased slightly. The shortage is redundant func-
tion between link layer and transport layer and the
design of link layer is complex. In addition, it
maybe results in waited overtime.
3) Agent solution program adopt agent manner (TCP-
Spoofing, TCP-Splitting) [13-15] which refer to
TCP-Spoofing, Snoop-TCP, TCP-Splitting and I-
TCP. The core idea of agent manner is that it sends
fake response information at satellite network gate-
way so that the users can not detect the long pro-
pagation delay in satellite link.
4) Special transmission control protocol solutions in-
Fast Start [16]
Fast start is instead of Slow Start strategy in connec-
tion beginning stage by adopting last link of TCP trans-
mission window size and repeating the latest TCP trans-
mission data rate. In order to avoid network congestion,
the IP packet should be set with low priority, which is
sent at beginning state.
TCP-Peach [17] and TCP-Peach+ [18]
TCP-Peach protocol adopts to send “Dummy” data seg-
ment of redundancy data information to detect the avail-
able bandwidth in start and Retransmission-Restoration
stage. In order to reduce the bandwidth occupied in re-
Copyright © 2011 SciRes. IJCNS
verse link, STP protocol send a periodic “POLL” packet
to inquire the receiver about receiving the data situation.
STP and STP+ [19]
In order to reduce the bandwidth occupied in reverse
link, STP protocol send a periodic “POLL” packet to
inquire the receiver about receiving the data situation.
Through periodic response and NACK, it reduces the
occupied resources in backward bandwidth
XCP [20]
It uses the congestion instruction method and adds the
current congestion window value, the time for data return
and back, the reservation bandwidth. In addition, it de-
mands mid router to add algorithm and modify the res-
ervation bandwidth information.
TCP-Westwood [21]
In order to improve the network bandwidth utilization,
TCP Westwood protocol achieves to estimate end-to-end
available bandwidth through continuous monitoring the
response information.
2. Novel Protocol Module
The Transmission Control Protocol (TCP) is the major
transport protocol in the TCP/IP protocol suite. TCP pro-
vides application programs with a reliable, flow controlled,
full-duplex, stream transport service. After requesting TCP
to establish a connection, an application program can use
the connection to send or receive data; TCP guarantees to
deliver the data in order without duplication. Finally, when
the two applications finish using a connection, they re-
quest that the connection be terminated.
TCP on one computer communicates with TCP on an-
other computer by exchanging messages. All messages
from one TCP to another use the TCP segment format,
including messages that carry data, acknowledgements,
and window advertisements, as well as messages used to
establish and terminate a connection. Each TCP segment
travels in an IP datagram.
In general, transport protocols use a variety of mecha-
nisms to insure reliable service. TCP has a particularly
complex combination of techniques that have proven to
be extremely successful. In addition to a checksum in
each segment, TCP retransmits any message that is lost.
To be useful in the Internet where delays vary over time,
TCP’s retransmission timeout is adaptive TCP measures
the current round-trip delay separately for each connec-
tion, and uses a weighted average of the round- trip time
to choose a timeout for retransmission.
Novel protocol is composed with Super Start of sending
client, congestion avoiding, distinguishing lost, conges-
tion restoration strategy, discarded data strategy of router,
and periodic response strategy of receiving terminal.
2.1. Novel Protocol Module and Architecture
As shown in Figure 1, the sending terminal include fol-
lowing algorithm in novel protocol: super start, conges-
tion avoiding, distinguishing lost, congestion restoration
strategy and etc. The super start strategy has replaced the
slow start strategy in TCP-Reno which is widely applied
in current network and in TCP-New Reno. Distinguish-
ing lost and congestion restoration are new strategies, but
congestion avoiding is same as TCP-Reno, TCP-New
2.2 Super Start Strategy
Super start strategy is shown in Figure 2; the congestion
window value of sending terminal is set half of the re-
ceiving window value of receiving terminal, i.e. cwnd =
rwnd/2. Sending terminal send a TCP data segment every
interval time τ and the priority of sending data segment
are alternative switch. Interval time τ is calculated in
accordance with the following formula:
τ = 2·RTT/cwnd (4)
The return time is estimated in link set up process. At
this stage a priority bit (“pri”) is stand for the priority of
packet in IP header’s TOS segment, besides a state bit
(“start”) is stand for the state of sending data. The IP
packet set with start = 1 in super start stage and the
sending terminal access to congestion avoiding state
from ending super start strategy when receiving first M-
NACK information. The later IP packet is sent with start
= 0, in addition the receiver send an M-NACK every RTT.
The specific algorithm of super start strategy can be
illustrated by following examples. Assuming that TCP
segment begin to send data when t = 0, thereby the re-
ceiver receive first segment and begin to count time after
Super Start
Congestion Avoidance
Congestion RecoveryPacket lost
Loss Distiction
Figure 1. A novel protocol architecture.
Copyright © 2011 SciRes. IJCNS
L. YU ET AL.261
about RTT/2. The receiving start to send the first
M-NACK when t 3·RTT/2, i.e. after one RTT. After
RTT/2, the receiver receives the M-NACK and increase
by the number of cwnd data which are received by re-
ceiver. At the same time it ends the super start stage and
enter to congestion avoiding state.
M-NACK include all state of receiver like hoping to
get the maximum sequence value for segment, hoping to
get the maximum sequence value of segment and packets
lost list.
Super Start()
start = 1;
cwnd = rwnd/2;
τ = RTT/cwnd;
for (i = 1 to rwnd)
if (pri = = 1)
pri = 0;
pri = 1;
start = 0;
wait for ACK;
if (ACK arrives)
cwnd = cwnd + num_received;
τ = RTT/cwnd;
Figure 2. Super start strategy.
2.3. Distinguishing Lost Strategy
The IP packet sent by alternating high and low priority
with start = 0 after accessing to congestion avoiding stage.
The transmitter judge whether data are lost according to
M-NACK information. As shown in Figure 3, when
transmitter finds that data lost, it judges why data loss
according to how many high and low priority packet lost.
When congestion occurs in network, the router firstly
discards the low priority data in queue. Liking Formula
(5), the number of lost low priority packets is same as all
lost packets.
Low_pri = lost_n u m (5)
When there is serious congestion in network and all
low priority packets have been lost, the high priority
packet will begin to discard. As shown in Formula (6),
the number of lost packets is more than half of conges-
tion window value.
Loss Distinction()
if(lost num = = 1)
pri = 1;
if((low-pri = = lost num) or (lost num > cwnd/2))
goto Congestion Recovery();
while(lost num > 0)
lost num = lost num1;
Figure 3. Distinguishing lost strategy.
Lost_num > cwnd/2 (6)
When above two situation occur, the transmitter con-
siders the congestion occurs in network, so it adopts
congestion restoration strategy. As shown in Figure 4,
congestion window value has halved. It recounts interval
time τ and sends lost data in low and high priority inter-
val time. After sending all lost data, it ends congestion
restoration stage and accesses to congestion avoiding stage.
The transmitter considers that data lost due to bit error
except above two situations. As shown in Figure 3, the
transmitter immediately sends lost data and the sending
window value should not be changed. It access to con-
gestion avoiding stage and send new data after sending
all lost data.
When the network congestion situation is not serious
and only a single packet loss, the whole network per-
formance will not obviously decrease. Therefore, when
finding a single packet loss, the transmitter considers that
lost data are caused by bit error and will not reduce con-
gestion window value.
2.4. Router Packet Discarded Strategy
According to the priority (“pri”) and state value (“start”)
of header, it can be sorted by four categories: 1) start = 1,
pri = 0, 2) start = 1, pri = 1, 3) start = 0, pri = 0, 4) start
= 0, pri = 1. When congestion occurs, router will
Copyright © 2011 SciRes. IJCNS
Congestion Recovery()
if((nowlast_time) > = 2*RTT)
cwnd = cwnd/2;
last_time = now;
τ = RTT/cwnd;
for(i = 1 to lost_num)
if(pri = = 1)
pri = 0;
pri = 1;
Figure 4. Congestion restoration strategy.
apply following data discarded strategy like Figure 5.
1) Firstly discarding low priority with high beginning
packets, i.e. start = 1, pri = 0;
2) When the above packets can not be found, high
priority with high beginning packets will be dis-
carded, i.e. start = 1, pri = 1;
3) When high beginning packets are not exist, low
priority with low beginning packets will be dis-
carded, i.e. start = 0, pri = 0;
4) When above all packets are not exist, high priority
with low beginning packets will be discarded, i.e.
start = 0, pri = 1;
Due to the IP data header sent at super start stage with
state value start = 1, router adopts above data discarded
strategy which can avoid from data loss in other link
caused by network congestion when new connection set
3. Simulation and Analysis
In order to compare with TCP-Reno, TCP-New Reno,
TCP-Peach, TCP Westwood, XCP and STP protocols.
We take advantage of the current popular NS-2 network
simulation software to do simulation according to simu-
lation topology proposed in literature user connect to
synchronous satellites through ground gateway which
can aggregate N connections. The capacity of gateway
cache is 50 packets; the bandwidth of ground connection
is 10 Mbits/s, the forward bandwidth (Bforward)of satellite
link is 10 Mbits/s, RTT is 550 ms, every TCP segment
length is 1000 bytes, the receiving window value of re-
ceiver is 64 packets, simulation time is 1000 times than
RTT, which is 550 s.
Packet Drop()
if (start = 1)
if (pri = 0)
drop (packet);
drop (packet);
if (pri = 0)
drop (packet);
drop (packet);
Figure 5. Data discarded strategy.
3.1. The Performance of Single-link Comparison
When only one link connects in network and reverse link
bandwidth (Bbackward) is 10 Mbits/s, it sends files with
different size and data transmission time for every pro-
tocol. From Figure 6, we can see TP-S transmission time
is shortest when file is less than 64 Kbytes and its trans-
mission time is nearly same as TCP-Peach when file is
more than 64 Kbytes. In the other protocols, STP trans-
mission time is longest but TCP Westwood, TCP-NewR-
eno and TCP-Reno all are same. Evidently, super start
strategy of TP-S can quickly increase transmitter’s con-
gestion window, thus the smaller files can be transmitted
with short time.
When only one link connects in network, all the aver-
age throughput of forward link for every protocol is
shown in Figure 7. Bit error is the only reason result in
Figure 6. Compared performance of file transmission with
single-link without bit error.
Copyright © 2011 SciRes. IJCNS
L. YU ET AL.263
Figure 7. Compared throughput of forward link in single-
link with random bit error.
data lost in such situation. It can be seen from graph that
TP-S protocol average throughput is 9.5 Mbits/s or so
which is similar with others when the packet lost rate is
less than 10–4. The throughput of STP protocol is lower.
With packet lost rate is gradually increasing, TCP-Reno,
TCP-NewReno, TCP-Peach, TCP Westwood, XCP and
STP protocols’ throughput decrease rapidly. When packet
lost rate 10–2., the throughput of TCP-Reno, TCP-New R-
eno and STP only are 0.2 Mbits/s, TCP Westwood is 0.3
Mbits/s, XCP is 0.5 Mbits/sTCP-Peach is 0.72 Mbits/s.
But the performance of TP-S degrade slightly, whose
throughput begin to decrease until packet lost rate is
more than 10–3. When the packet lost rate as high as 10–2,
throughput still remain at 0.8 Mbits/s level.
We can see that the occupied backward bandwidth of
TCP-Reno, TCP-NewReno, TCP-Peach, TCP Westwood
and XCP protocols are more than 35 Kbits/s when the
packet lost rate is less than 10–4. With packet lost rate is
gradually increasing, backward bandwidth decrease. All
backward bandwidth are more than 5 Kbit/s when packet
lost rate is as high as 10–2. Due to adopt periodic sending
response information strategy, STP and TP-S protocols
backward bandwidth occupied mainly rely on sending
periods. With packet lost rate growing, backward band-
width of TP-S protocol increase slowly. When the packet
lost rate is as high as 10–2, the backward bandwidth is not
higher than 1 Kbits/s.
3.2. The Performance of Multi-Link Comparison
When the number of connection N > 20, there is not only
link error but also congestion in network. It caused by
network data maximum input rate is more than satellite
forward link bandwidth. How to calculate Maximum
input rate (Inmax) is shown in the following formula:
Inmax = N·rwnd·8/RTT (7)
That can be deduced by above formula, when rwnd =
64RTT = 550 ms and N = 10, forward Link is at the
saturation state, that is the utilization of forward link
bandwidth will be maximum without congestion in net-
work. When the number of connection N = 20 and
backward bandwidth Bbackward = 10 Mbits/s, the average
throughput of forward link for every protocol is shown in
Figure 8. It shows that throughput of TP-S is not the
highest which value is close to 9 Mbits/s but lower than
the TCP Westwood, STP and XCP protocol when packet
lost rate is less than 10–4. But with increasing of packet
lost rate, TP-S throughput has not declined. It has re-
mained at close to 9 Mbits/s level, but the performance
of other agreements are obviouly decreased. When packet
lost rate is 10–2, throughput of TCP-Reno, TCP-NewReno
and STP is only about 3.5 Mbits/s. The throughput of
XCP and TCP Westwood protocols are 4.5 Mbits/s and
5.5 Mbits/s, even if TCP-Peach protocol is only 7.5 Mbits/s.
The situation of backward bandwidth occupied when
connection number N = 20 is shown in Figure 9. When
packet lost rate is less than 10–3, the occupied backward
bandwidth of TCP-Reno, TCP-NewReno, TCP-Peach,
TCP Westwood and XCP protocols are more than 300
Kbits/s. Although backward bandwidth of these proto-
cols continuously decline with increasing of packet lost
rate, backward bandwidth still more than 100 Kbit/s es-
pecially TCP-Peach protocol which backward bandwidth
is nearly 300 Kbits/s, even if the packet lost rate is 10–2.
As a result of adopting periodic sending response infor-
mation strategy, backward bandwidth of STP and TP-S
protocol has not seriously changed with increasing of
packet lost rate. Backward bandwidth of TP-S protocol
occupoes very low with remained at 12 Kbits/s level.
Figure 8. Compared with the performance of forward link
throughput in 20 connections with random error.
Copyright © 2011 SciRes. IJCNS
Figure 9. Compared with the performance of backward
link throughput in 20 connections with random error.
3.3. The Performance of Backward Link
Restricted Bandwidth Comparison
When network connection number N = 10, forward link
is in saturation state and the average throughput of for-
ward link for every protocols are shown in below. As can
be seen from the graph, the performance of TCP-Reno,
TCP-NewReno, TCP-Peach, TCP Westwood and XCP
protocols are impacted by congestion of backward link
response information. When backward link bandwidth is
200 Kbits/s, throughputs of these protocols are only
about 8 Mbits/s. With backward link bandwidth is con-
tinuous declining; the congestion of response informa-
tion become serious and throughput of forward link also
decline. When backward link bandwidth is only 50 Kbits/s,
performance of protocol drop to less than 6 Mbits/s. As
backward link bandwidth which TP-S and STP need is
much less than 50 Kbits/s, especially TP-S protocol only
need 5Kbits/s, the performance of forward link will not
be impacted by backward link bandwidth for TP-S and
STP protocols, throughput of TP-S maintain at 9.5
Mbits/s or so. When network connection number N = 10,
the situation of backward link bandwidth occupied for
every protocols are shown in Figure 10.
4. Conclusions
In order to resolve the problem that TCP protocols has
poor performance in satellite network, the paper pro-
posed a novel satellite network transmission control pro-
tocol. The protocol adopts a novel window growth strat-
egy to accelerate the speed of increasing congestion
window after connection established. In order to distin-
guish specific reasons for data lost, the protocol adopts
interval sending strategy for different priority IP packets
and judges network performance according to data lost.
Figure 10. Compared with throughput of forward link un-
der backward link bandwidth restricted.
In addition, periodic sending response information strat-
egy is used in this protocol to solve bandwidth asymme-
try problem in forward and backward satellite link.
Compared with TCP protocols and some satellite net-
work transmission control protocols proposed in recent
year through simulation, the protocol not only can obvi-
ously enhance the throughput of forward link but also
greatly reduces the bandwidth occupancy rate in back-
ward link. Main feature of the protocol is that it adopts
different priority data interleaving sending principle to
send data. Super start can rapidly enhance sending rate at
the beginning of connection. Data lost judgment and
congestion restoration strategy can effectively distin-
guish the specific reasons for data lost and take related
transmission rate control strategy. Periodic sending re-
sponse strategy can reduce occupancy for backward link
bandwidth to solve satellite link bandwidth asymmetry
problem. Router adopts a simple and easy data discarded
strategy with different priority. The protocol can rapidly
enhance sending rate at the beginning of connection and
effectively distinguish reasons for data lost. It maintains
very high throughputs of forward link in channel envi-
ronment with random and unexpected error. Backward
link bandwidth occupied with rare resource can solve the
asymmetry problem of satellite link bandwidth. Router
algorithm is simple, low design requirements and facility
realization in network.
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Copyright © 2011 SciRes. IJCNS
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