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journal homepage: www.elsevier.com/locate/comnet
a r t i c l e
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Article history:
Received 7 July 2016
Revised 6 October 2016
Accepted 5 November 2016
Available online 8 November 2016
Keywords:
TCP Wave
Future internet
TCP performance
a b s t r a c t
Current and future communication scenarios are very different from those in which TCP was conceived
and developed, bringing new protocol requirements. Performance optimization is usually pursued with
patching of the traditional TCP implementations. Following an alternative approach, TCP Wave has been
designed with the primary goal to satisfy new requirements coming from current networks, such as adaptation to bandwidth/delay changes (due to mobility, dynamic switching, handover), ecient management
of spurious losses as well as link interruption, optimal exploitation of very high link capacity and ecient
transmission of small objects, as in most of Web applications (Web browsing, sensor apps, SCADA, M2M,
etc.), irrespective of the underlying link characteristics. Protocol fairness, TCP friendliness, congestion and
ow control and error recovery are also guaranteed. TCP Wave replaces the window-based transmission
paradigm of the standard TCP with a burst-based transmission, the ACK-clock scheduling with a selfmanaged timer and the RTT-based congestion control loop with an Ack-based Capacity and Congestion
Estimation (ACCE) module. As a valuable study case, the novel TCP Wave capabilities has been validated
on a generic NS-3 simulation testbed where all the most challenging events impacting on transport protocol can occur.
2016 Elsevier B.V. All rights reserved.
1. Introduction
TCP [1] is the traditional protocol conceived to provide a reliable transport service for most of Internet applications, operating
with no knowledge of either underlying network or trac characteristics. TCP was designed with a primary goal: to allow an efcient and reliable management of long connections (mostly long
le transfers) over terrestrial links characterized by small delays,
limited bandwidth and quasi error-free channels [2]. The communication environment had the above characteristics when Internet
was born. TCP eciently satised such requirements, adopting the
TCP window-based transmission, which is still the standard for Internet communications although enhanced and upgraded several
times. Whenever new challenging scenarios emerged (i.e., errorprone wireless links, broadband networks, large delay link), one
or more TCP algorithms (i.e. error recovery, congestion control,
Slow Start, Congestion Avoidance, etc.) have been modied, keeping the overall protocol architecture unchanged (i.e.[39]). The
main TCP transmission principles preserved along several decades
are the window-based transmission paradigm, the ACK-clocked
transmission, the dependency between congestion control and er-
Corresponding author.
E-mail address: roseti@ing.uniroma2.it (C. Roseti).
http://dx.doi.org/10.1016/j.comnet.2016.11.002
1389-1286/ 2016 Elsevier B.V. All rights reserved.
123
Westwood [7]). In fact, most of these protocols deal with modifying the window growth function of TCP in a more scalable/rapid
fashion. Bandwidth utilization results improved, but a number
of issues remain as major challenges: friendliness with existing
TCP trac, fairness among ows experiencing different RTT, a fair
bandwidth sharing between long and short transfers.
One of the most successful upgrade of TCP is TCP Cubic [22].
It is currently used as default in the Linux kernels (2.6.19 and
above) and it is very important within the proposed work, since
it can be considered a relevant attempt to decouple cwnd increase
and the RTT. In fact, the cwnd growth mainly depends on the
time elapsed from the last congestion event, although still ACKtriggered. Undoubtedly, TCP Cubic demonstrated an improvement
on the achievement of the optimal transmission rate on a wide
set of channels (including long latency ones), but its clocking and
reactive nature (not proactive) to congestion events leave wellknown open issues: start up behavior is still slow in getting available bandwidth penalizing short transfers, which are dominant in
the current Internet; fairness among ows is achieved over a time
strongly dependent on the underlying link characteristics, leading
to unfair situations impairing short transfers. Last, but not least,
TCP Cubic design as well as other TCP versions, does not address
at all possible dynamic link changes during a connection establishment.
Recently, several new TCP proposals aimed to specically address end-to-end link variability and/or diversity and belong to the
second problem category.
OpenTCP variant [23], for instance, has been specically proposed for SDN-based Data Center networks. It utilizes information
provided by the underlying SDN network state (i.e., topology and
routing information) and trac information, in a sort of cross-layer
interactions. Depending on such feedback and congestion control
policies, dened by the network operator, OpenTCP decides on a
specic set of adaptations for standard TCP. In more details, a Congestion Control Agent (CCA) is installed as a module in the endsystems with the aim to manage update massages sent by SDN
controllers and modify TCP stack accordingly. TCP behavior can be
modied leveraging on two main options: by modifying feedback
provided to TCP sources, i.e. using ECN bits in the ACK segments,
or taking advantage from today extensible TCP implementations,
which allow triggering a new congestion control scheme. OpenTCP
relies on a computation cycle including data collection, feedback
generation and TCP adaptation. Timescale must be much slower
than experienced RTT to make network dynamics stable, while TCP
scheme changes are obviously on a per-ow basis. Extending its
application to non-SDN context would require much more effort
and modication in-network, with additional signaling protocols.
In [24], Multipath TCP (MPTCP) has been presented as a transport protocol conguration able to exploit multi-path capability
to manage transfers between end-systems pairs through different network interfaces and then following different routes. MPTCP
benets include better resource utilization, better throughput and
smoother reaction to failures. On the other hand, TCP behavior
over each path heritages issues already discussed for all the aforementioned TCP versions.
Last, but not least, TCP Noordwijk [25] can also be considered
as a specic-scenario transport protocol, since it has been originally designed to operate between satellite Performance Enhancing
Proxies (PEPs). Since TCP Wave inherits the basics sender architecture from TCP Noordwijk, its detailed description is provided in the
next section.
3. TCP Noordwijk: pioneer of the burst-based transmission
TCP Noordwijk [2527] is the rst TCP-based protocol that
overcomes the window-based transmission paradigm introducing
124
125
Table 1
TCP Noordwijk limitations tackled by TCP Wave.
TCP Noordwijk function
Weakness description
Flow Control
Assumption of transmission buffers large enough to always include fresh data on the next scheduled BURST. Flow
Control is receiver-driven only.
Tested and validated with Selective ACK (SACK) option enabled only, not interoperable with common Operating
Systems (O.S.s).
Loss recovery through SACK notication only.
RTO timer management inherited from standard TCP, not suitable for link disconnections.
Triggering of the Adjustment Mode algorithm leverages on the raw RTT measurements, then affected by possible
inconsistent samples.
Tailored to transmit at the maximum bottleneck speed irrespective of presence of competing ows.
Tailored to perform an abrupt rate decrease waiting for the congestion reduction. Its duration its unpredictable and
depends on the nature of the competing trac.
Stability Factor (SF) denes the timing for transmission parameter changes. Potential loss of responsiveness to sudden
changes.
ACK management
Fast Retransmit
Retransmission Time Out (RTO)
Switching between rate control modes
Tracking mode algorithm
Adjustment mode algorithm
Stability Factor method
126
derived from the same cumulative ACK, so that they lead to the
computation of an ACK train dispersion close to 0. For all these
reasons, TCP Wave introduces a set of lters for both RTT and
Ack Train Dispersion calculations. As far as the RTT is concerned,
the procedure of splitting the cumulative ACKs has been enhanced
with the following lter:
if (bytesAcked > 2 * segmentSize)
Time RTT= avgRTT;
Basically, TCP Wave sender performs only a single ACCE update
every cumulative ACK.
In addition, the following lter is implemented to lter altered
Ack Train Dispersion sample:
if ((LastAckTime - FirstAckTime) > avgRTT
&& (LastAckTime - FirstAckTime) == 0)
ackTrainDispersion=minTrainDispersion;
This algorithm discards ACK train dispersion samples higher
than measured RTT or equal to 0, which are expected during retransmission phases.
4.4. Enhanced Retransmission Time Out (RTO) mechanism
Retransmission Time Out (RTO) indicates how long TCP senders
must wait for an ACK before performing a retransmission action
and it works in a tight combination with Fast Retransmit (FR) algorithm of TCP. RTO is usually set proportionally to RTT measurements, which are computed once per burst in TCP Wave. When
a lost packet is retransmitted through FR algorithm (Section 4.3),
the resulting RTT value is inated because RTT is measured against
the original packet transmission. Obviously, in this case the RTT increase is not a symptom of congestion. Differently from standard
TCP, all the RTT measurements including those calculated during
retransmissions, are sent to the underlying RTT estimator, which
calculates the round-trip time variation (RTTVAR) and sets the RTO
value accordingly. This expedient ensures that the estimated RTO
value is always consistent with the RTT value seen by TCP Wave,
also during retransmission, avoiding unnecessary and unwanted
RTO expirations. The second aspect tackled by TCP Wave for the
RTO management is related to the actions to perform upon RTO
expiration (RTO recovery). Without receivers feedback for a time
that exceeds the estimated RTO, the sender assumes that all segments that were previously sent, but not yet acknowledged, are
lost. Thus, data recovery is performed starting from the rst unacknowledged segment until the last segment that has been sent
when the RTO is expired. Because RTO usually indicates a serious problem in the network, either severe congestion or even
temporary disconnection, TCP Wave sender behavior during the
RTO recovery is quite conservative: the sender resets the BURST
size and the TxTime to the initial values BURST0 and TxTime0 respectively, assuming that network conditions are not well known
yet. Then transmission restarts in bursts from the latest Acknowledged packet before the RTO expiration. If cumulative ACKs related to packets belonging to multiple bursts are received during
RTO recovery, TCP Wave sender accounts them for jumping in the
sequence number space, and to update ACK-based statistics only
once. This produces gaps between the retransmitted bursts, reducing the amount of data unnecessarily retransmitted. In the meanwhile, BURST size and TxTime are still dynamically updated upon
the reception of regular ACK trains.
4.5. Switching between tracking and adjustment mode
In order to avoid effect of unexpected RTT spikes due to several network dynamics (i.e. processing glitches, link layer retransmissions, etc.), TCP Wave considers an average RTT value to compute RTT. In addition, an Exponentially Weighted Moving Average
k =
RT Tk minRT Tk
RT Tk
Thus, the lter is very reactive when RTT is close to the absolute minimum measured value minRTTk , updated at each received
ACK train. In other words, smaller RTT samples greatly inuences
the new average RTT estimation, while larger RTT samples have a
minor effect. Denitively, RTT, used as condition to switch between Tracking and mode, is computed as follows:
T xRate =
BURST0
T xT ime
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128
SF =
avgRT T
ACKT rainDispersion
RateINIT =
BURST0
T xT ime0
RateMIN =
BURSTmin
T xT ime0
BURST0
BURST0
Ratetrack
T xT ime +
T xT ime
This also implies a better capability to achieve a fair and stable bandwidth sharing in case of multiple TCP Wave ows;
2. Triggering later the Adjustment mode, where RTTmin is reset,
TCP Wave is slower in experiencing possible network changes
and in general has a gentler interaction with competing ows
(e.g. TCP NewReno ows).
Typical values for are in the range 0.1 0.3 s.
5. Simulation setup
TCP Wave has been implemented on the Network Simulator NS3 [29]. A new module tcp-wave.cc has been added at the opensource software package allowing the conguration of a TCP Wave
sender object to be mounted on a generic sender node for transporting application trac. The reference conguration of the simulation scenario is depicted in Fig. 3. A variable number of source
nodes are connected to the Router 1 through point-to-point links,
characterized by a low latency. Each source nodes generates a congurable amount of raw data over reliable connections exploiting
either TCP or TCP Wave services. Then, Router 1 is connected to
the Router 2 through the bottleneck link, which is shared among
all the running data ows. Finally, data ows generated are terminated at separated Sink agents, each directly connected to Router 2.
If not explicitly reported , link parameters will be those included
in Fig. 3.
As far as the initial TCP Wave parameters are concerned, the
following default values are considered:
1.
2.
3.
4.
5.
MT U = 400 bytes
BURST0 = 10 T CP packets;
T XT ime0 = 0.5 s;
BURSTmin = 3 T CP packets;
= 0.15 s;
129
130
Fig. 6. Details on Wave effect, associated to the smooth increase and decrease of TxTime as function of the sequence number.
131
Table 2
Transmission rate summary (kbit/s).
Flow
Flow
Flow
Flow
Total
1
2
3
4
Int. 1
Int. 2
Int. 3
Int. 4
Int. 5
Int. 6
Int. 7
1991
1991
955
832
1789
680
697
610
1987
546
509
498
447
20 0 0
668
674
615
1957
887
868
1755
1930
1930
132
link. As shown in Fig. 11, TxTime is updated about 0.5 s after the
bandwidth change, a time interval corresponding to the minimum
RTT envisaged in the congured simulation scenario. Because of
the bandwidth increases, the only apparent effect on transmission
dynamic is a sudden shortening of the TxTime, while burst size is
kept constant since the protocol remains in Tracking mode.
When halving bandwidth at 10 s, instead, transmission with
the old settings causes a temporary but inevitable RTT increase.
TCP Wave suddenly reacts to RTT increase by adjusting actual TxTime according to the tracking mode algorithm, which depends on
RTT. When RTT is brought back to its minimum value, transmission continues by using the optimal TxTime computed in the meanwhile, considering new measured values of ACK train dispersion.
Fig. 13 reports the RTT variations over the simulation time.
Bandwidth increase at 5 s does not lead any ination of the RTT,
while the bandwidth reduction at 10 s causes a temporary increase
of RTT, which doubles its value for a time interval of about 1 s (2
physical RTTs). Afterwards, TCP Wave is able to both achieve new
optimal transmission rate and empty the bottleneck buffer, bringing back RTT to its minimum value.
To cover all possible scenarios, an additional simulation setup
envisages a simultaneous change of both bandwidth and delay.
This scenario is very challenging for any transport protocol, since
it causes a fully changing of any time reference. Specically, at 5 s,
bandwidth changes from 2 to 10 Mbit/s and physical delay of the
bottleneck link increases of 50 ms (RTT is expected to increase of
100 ms, +20%). The result in terms of transmission dynamic, shown
in Fig. 14, is similar to the one experienced with only bandwidth
change. TCP Wave is able to adjust its transmission parameters to
match the new available capacity in the bottleneck, being resilient
to the RTT change.
The link delay increase makes the reception of ACK trains,
bringing new channel information, delayed. During the rst RTT
after delay-bandwidth change, a temporary glitch slightly increases
RTT above the new minimum value (see Fig. 15).
6.4. Flow control
In general, Transmit/Receive (Tx-Rx) buffer size should be at
least larger than bandwidth-delay product to guarantee a full exploitation of the available capacity. In case this condition is not respected, meaningful consequences may affect dynamic scenarios,
133
134
where network characteristics can change during a connection lifetime and end-system settings can not be intrinsically optimized for
all the network characteristics.
Rate regulation according to end-point buffering capability is in
charge of the Flow Control mechanism. TCP Wave envisages a Flow
Control scheme acting on the actual burst size. The burst size, as
computed by the rate control component, is corrected on the basis
of the actual amount of fresh data in the TxBuffer (data not yet
transmitted in previous bursts). If the amount of fresh data is
smaller than the computed BURST value, the denitive BURST size
is reduced accordingly or even set to 0 waiting for reception of
ACK trains, which allow to cancel corresponding ACKed segment
from TxBuffer, leaving space for new application data.
In order to validate the effectiveness of the TCP Wave Flow Control, a critical scenario has been reproduced setting overall capacity to 4 Mbit/s. With 4 Mbit/s and an RTT of about 500 ms, the
achievement of the maximum throughput requires an amount of
data in ight (and buffered in the TxBuffer) at any time equal to
about 250 kBytes. Then, when using TX-RX buffers smaller than
135
136
related operations and in these cases TCP Wave must face an RTO
occurrence. This scenario has been reproduced by forcing a 2 s link
outage (between 30 s and 32 s) during a TCP Wave connection
running, and the simulation output is shown in Fig. 20. At 30 s,
although link is off, TCP Wave continues to transmit new bursts
until RTO expires (after about 1 s). Therefore, TCP Wave starts retransmission of all the unacknowledged segments at 31 s, restoring
default parameters. In the simulated scenario, the rst two retransmitted bursts are lost again due to the outage persistence. On the
contrary, the third burst successfully arrives at the destination and
triggers a cumulative ACKs covering also some segments received
before the link outage and whose ACKs were lost. This cumulative ACK resets RTO timer, refreshes transmission parameters and
causes a jump in the burst sequence. As a result, after 1 RTT from
the link reactivation, TCP Wave transmits at the maximum rate. Of
course, all the segments proactively transmitted during outage are
retransmitted and RTO phase is concluded just after the recovery
of the last segment transmitted upon the RTO expiration.
6.6. RTT-Fairness
A very interesting network setup envisages a TCP Wave highlatency connection sharing the bottleneck link capacity with a TCP
Wave low-latency one. To this purpose the reference scenario in
Fig. 3 has been modied setting the bottleneck delay (Router1
Router2 link) to 5 ms, while only the delay on the link between
Node2 and Router1 has been increased to 240 ms. Denitively, TCP
ows coming from Node1 experience a minimum RTT of about
20 ms, and TCP ows from Node2 experience a minimum RTT of
about 490 ms.
As largely reported in literature, such a scenario drastically impairs traditional TCP performance, since ACK-clocked cwnd operations are sensitive to the experienced RTT: the higher is RTT the
slower is cwnd growth. As a consequence, bottleneck capacity is
mostly got by the low-RTT ows. Several TCP-based proposals have
solved, or at least mitigated, such a problem using some equationbased enhancements to the TCP congestion control. This is the case
of both TCP Cubic [22] and TCP Hybla [6]. TCP Cubic algorithms
positively impact on fairness as reaction of losses, while TCP Hybla
137
adopts an adaptation factor to the traditional TCP control formulas, using a reference RTT of some dozens of milliseconds. The aim
of the next simulation is to demonstrate that TCP Wave presents
an optimal RTT-fairness capability as well. Simulation envisages a
TCP Wave transfer from Node1 to sink1 lasting 60 s. In the middle of this transfer, Node2 accesses shared link to transfer data to
the sink 2. Figs. 21 and 22 show BURST transmission evolution and
RTT, respectively, for both TCP Wave connections. The overall behavior is similar to that observed when competing ows see the
same RTT. In the rst 30 s, the low-latency connection gets all the
available bandwidth. When also the high latency connection starts,
both immediately transmit at the same rate. Average throughput
along the whole simulation is 1.965 Mbit/s. Then, both full utilization and optimal fairness are conrmed.
7. TCP Friendliness
To evaluate TCP Wave applicability in current networks, it is
of paramount importance to evaluate its friendliness with standard
TCP, based on the Additive Increase Multiple Decrease (AIMD) con-
gestion control. In fact, in short or medium-terms scenarios, coexistence with standard TCP ows must be assumed. As a target, a
fair and stable sharing of the bottleneck capacity must be provided
at the steady state. Traditionally, a new TCP-like transport protocol
is TCP-friendly when, at the steady state, it gets an amount of
the bottleneck capacity equal or lower than that exploited by standard TCP. Herein, TCP NewReno implementation has been taken as
standard protocol reference and then considered for simulations.
To test friendliness, a simulation scenario relying on the basic
setup shown in Fig. 3 was congured. The only variations regard
the MTU size and bandwidth in access link between TCP-based
sources and Router 1. MTU has been increased up to 10 0 0 bytes
in order to improve on TCP New Reno congestion control, which
increases rate on a packet-basis. Instead, bandwidth of the access links has been increased from 2 Mbit/s to 10 Mbit/s in order
to emphasize buffering dynamics at the bottleneck link between
Router 1 and Router 2.
Tests envisage a TCP Wave and a TCP New Reno connection simultaneously running over a very long time (1 h) and competing
for the bottleneck capacity. For the sake of result clearness, bulk
138
data transfers are performed over both TCP connections. Therefore, the long-term interaction between the two protocols can be
fully characterized, avoiding any limitation coming from application methods and congurations. Test outputs address a number
of parameters (and plots) useful for an exhaustive analysis of the
TCP Wave fairness as well as the overall protocol eciency:
Transmission Rate plot - Useful to show the smoothed bandwidth sharing over time of the two competing ows. This
output represents the main friendliness indicator; as much
throughput of the two ows is similar and close to half of the
available bottleneck bandwidth as more TCP Wave can be considered friendly with TCP New Reno.
Instantaneous Rate detail plot - this plot is a ner zoom of the
previous one, aimed to highlight dynamics of the interaction
between TCP Wave and TCP New Reno.
Re-transmission detail plot - A cycle of the interaction between
TCP New Reno and TCP Wave terminates with a buffer overow, which is the consequence of the reactive approach of TCP
NewReno congestion control. Basically, TCP NewReno increases
more and more its cwnd until losses are experienced. This plot
focuses on how TCP Wave manages the recovery of multiple
losses up on a buffer overow.
Cwnd/Equivalent Cwnd plot - Congestion Window (cwnd) is the
main parameter for all the standard TCP versions to regulate congestion control operations. The higher is the cwnd the
higher is the transmission rate. Then,
cwndopt [MT U] =
bandwidth[bytes/s] RT T [s]
MT U[bytes]
cwndeq [MT U ] =
BU RST [MT U ]
RT T [s]
T xT ime[s]
RTT - Round Trip Time provides a meter of the overall network conditions. When RTT is higher than its minimum value,
139
Fig. 23. average rate and overall throughput with = 150 ms.
rectly demonstrates how TCP Wave avoids inter-protocol starvation, although competing for bandwidth with a greedy protocol
that tends to increase its rate more and more, irrespective of growing congestion. In fact, the Adjustment Rate algorithm allows TCP
Wave to periodically reset its transmission parameters contrasting
the greedy TCP NewReno standard behaviour.
To further clarify such interaction details, Fig. 24 shows a snapshot of the instantaneous throughput over the rst 50 s of simulation. TCP NewReno rate is computes as cwnd/RTT and presents
enough small oscillations due to the variation of the experienced
RTT, which is periodically inated upon the transmission of TCP
Wave bursts. To opposite, TCP Wave burst transmission leads to a
completely different trend based on succession of periods where
rate is equal to the overall bottleneck capacity, at the beginning
of the Tracking Mode, with periods where rate is taken down to a
minimum value as envisaged by the Adjustment Mode algorithm;
eventually, both transfer with a similar average rate as shown in
Fig. 23. As a conclusion, the TCP NewReno cycle is bounded be-
140
Fig. 25. Zoom on TCP Wave error recovery during buffer overow.
141
8. Conclusion
Future Internet must face up new challenges not approached by
traditional transport protocol design. In particular, TCP has been
designed with assumptions not valid any longer in current networks, such as low latency, no channel packet losses, no dynamic
link handover, etc. In this context, TCP Wave proposes a new approach to provide a reliable transport services over today networks.
New requirements and characteristics of current networks are addressed through an innovative burst-based transmission paradigm,
as an alternative of the traditional TCP window-based approach.
This paper presents the new protocol characteristics and methods implemented in the Network Simulator NS-3. In particular, the
most challenging network events are tested through a exible simulation setup in order to validate TCP Wave suitability to transmission scenarios not envisaged at all by traditional TCP implementations. In fact, simulation setup was designed with the aim to reproduce the most challenging dynamics faced in most innovative
networks: heterogeneous networks, leaky links, link handover, SDN
changes, multi-protocol bottleneck competition, etc. Simulation results conrmed how TCP Wave is suitable for the use in these new
networks. In particular, achievements highlighted two main protocol characteristics: i) the ability to quickly match the available
bandwidth, useful in case of short transfers as most of Internet
ows, and ii) the ability to converge to a fair and stable steadystate in case of longer ows either stand-alone or competing with
other ows. Last, but not least, TCP Wave ows resulted friendly
with standard TCP, allowing a fair bottleneck bandwidth sharing in
case of competition.
References
[1] M. Allman, V. Paxson, E. Blanton, TCP Congestion Control, Network Working
Group RFC 5681, 2009.
[2] W. Stevens, TCP/IP Illustrated, Addison Wesley, vol 1 (1994).
[3] G. Huston, TCP in a wireless world, IEEE Internet Comput. 5 (2) (2001) 8284.
[4] M. Luglio, C. Roseti, F. Zampognaro, Performance evaluation of TCP-based applications over DVB-RCS DAMA schemes, Int. J. Satell. Commun. Netw. 27 (3)
(2009) 163191.
[5] C. Barakat, E. Altman, W. Dabbous, On TCP performance in a heterogeneous
network: a survey, IEEE Commun. Mag. 38 (1) (20 0 0) 4046.
[6] C. Caini, R. Firrincieli, A new transport protocol proposal for internet via satellite: the TCP hybla, in: Proceedings of ESA ASMS conference 2003, 2003.
[7] C. Cassetti, M. Gerla, S. Mascolo, M. Sanadidi, R. Wang, TCP Westwood: end
to-end congestion control for wired/wireless networks, Wireless Netw. J. (8)
(2002) 467479.
[8] F. Belli, M. Luglio, C. Roseti, F. Zampognaro, Evaluation of TCP performance over
emulated DVB-RCS scenario with multiple RCSTs, in: Proceedings of IWSSC09
- 2009 International Workshop on Satellite and Space Communications, 2009,
pp. 424428.
[9] M. Luglio, C. Roseti, F. Zampognaro, Improving performance of TCP-based applications over DVB-RCS links, in: Proceedings of 2009 IEEE International Conference on Communications, ICC 2009, 2009.
[10] G.T. 36.331, LTE; evolved universal terrestrial radio access (E-UTRA; radio resource control (RRC); protocol specication, ETSI TS 136 331 v10.7.0 (201211)(2012).
[11] ETSI, Digital video broadcasting (DVB); second generation DVB interactive
satellite system (RCS2); part 2: lower layers for satellite standard, Draft ETSI
EN 301 545-2 V1.1.1(2011).
[12] F. Zampognaro, C. Roseti, M. Luglio, A. Detti, A. Caponi, Mobile-PEP: satellite
terminal handover preserving service continuity, The proceedings of Twelfth
International Symposium on Wireless Communication Systems, 2015.
[13] S. Salsano, N. Blefari-Melazzi, A. Detti, G. Morabito, L. Veltri, Information centric networking over SDN and openow: architectural aspects and experiments
on the OFELIA testbed, Comput. Netw. 57 (16) (2013) 32073221.
[14] C. Ververidis, P. Fragoudis, Y. Thomas, V. Siris, F.A. I. Andrikopoulos, C. Baudoin,
M. Guta, Experimenting with services over an information-centric integrated
satellite terrestrial network, IEEE Future Netw. Mobile Summit (2013).
[15] T. Lakshman, U. Madhow, The performance of TCP/IP for networks with high
bandwidth-delay products and random loss, IEEE/ACM Trans. Netw. 5 (3)
(1997) 336350.
[16] M. Luglio, C. Roseti, F. Zampognaro, F. Belli, An emulation platform for IP-based
satellite networks, in: Proceedings of 27th IET and AIAA International Communications Satellite Systems Conference, ICSSC 20 09, 20 09.
[17] Satlabs, Interoperable PEP (I-PEP) transport extensions and session framework
for satellite communications: air interface specication, Satlabs specs (2005).
[18] T. Henderson, S. Floyd, A. Gurtov, Y. Nishida, The newreno modication to
TCPs fast recovery algorithm, Internet Engineering Task Force (IETF), RFC
6582(2012).
[19] C. Jin, D. Wei, S. Low, FAST TCP: motivation, architecture, algorithms, performance, INFOCOM 2004. Twenty-third Annu. Joint Conf. IEEE Comput. Commun.
Soc. 4 (2004) 24902501.
[20] S. Floyd, Highspeed TCP for large congestion windows, Network Working
Group, RFC 3649, 2003.
[21] Y. Iyer, S. Gandham, S. Venkatesan, STCP: a generic transport layer protocol for
142
[22]
[23]
[24]
[25]
143
Ahmed Abdelsalam received his B.Sc. degree from University of Alexandria - Egypt in 20 0 0. He received his M.Sc. in Information Security from
Royal Holloway University of London United Kingdom in 2010. Ahmed received his Ph.D. in Space Systems and Technologies from University of
Rome Tor Vergata - Italy in 2015. Ahmed Abdelsalam is now a Postdoc Researcher at the University of Rome Tor Vergata, Italy. He is involved in
research activities related to Satellite Systems, supporting some research funded projects. His research interests include network security of both
wired and wireless networks, networking protocols, and future internet.
Luglio received the Laurea degree in Electronic Engineering at University of Rome Tor Vergata. He received the Ph.D. degree in telecommunications. At present he is associate professor of telecommunication at University of Rome Tor Vergata. He works on designing satellite systems
for multimedia services both mobile and xed. He teaches Satellite Telecommunications and Telecommunications basics. He is member of the
steering committee and of the executive board of NITEL Consortium. He has several years of experience in project management, teaching and
leading many R&D activities within research institutes, universities and private companies.
Cesare Roseti graduated cum laude in 2003 in Telecommunication Engineering at University of Rome Tor Vergata. In 2003 and 2004, he was a
visiting student at Computer Science Department of University of California, Los Angeles (UCLA). From August to December 2005 he worked at the
TEC-SWS division of the European Space Agency (Noordwijk, The Netherlands). He received the Ph.D. degree in Space systems and technologies in
20 07. In 20 09, he achieved II-level master on Homeland Security at the University of Bologna. He is currently assistant professor at the University
of Rome Tor Vergata. He teaches Broadcast technologies and he is assistant professor in the Satellite Communications class, at the University
of Rome Tor Vergata. Since 2010, he teaches Principles and methods for the risk analysis at the II-level master on Homeland security (Campus
Biomedico, Rome). He is involved in national and international research projects in the telecommunication eld, and he is author of about 60
scientic publications.
Francesco Zampognaro received his Ph.D. in Space Systems and Technologies from University of Rome Tor Vergata in 2010 and his M.Sc. in
Telecommunication Engineering from University of Rome La Sapienza in 2004. He was Young Graduate Trainee for 1 year in 2006 at ESA/ESTEC
in the EUI/Telecommunications department. He is contract-professor in Space Systems at University Marconi, and was awarded a Research Grant
in Satellite Broadband multi-beam simulations, for which activities are still in progress. He is involved in research activities related to Satellite
Systems, supporting many research funded projects. His main interests are the study and simulation of Satellite Systems, in particular DVB-RCS,
covering protocols optimisation, services provision and QoS, security, resource allocation and integrated/hybrid architectures.