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technologies are becoming more popular every day. In this paper VoIP technology
through satellite communications will be considered. Some standards in
infrastructures, protocols and codecs will also be discussed. More specific,
GEO (Geostationary) satellite will be used as a medium. Finally, it will be decided what parameters could make VoIP technology
to operate efficiently and with better quality taking into consideration the
specificities of a satellite TCP/IP and UDP/IP link.



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Voice communication has been a dominant element in analogue
and digital exchanges for years. The invention of telephone in 1876 was the
door step to the world of telecommunications. Human speech, which is an
analogue wave signal, was carried in long distances through analogue networks,
enabling people from different parts of the world to communicate. The
continuous research and the effort to overcome the disadvantages of analogue
signal such as the noise, the cost and the capacity of the network, led to the
development of digital transmission networks. The conversion of the human
speech into digital signals since 1950s, using 0s and 1s, eliminated many
drawbacks of analogue networks because the communication was more efficient due
to simplicity. Telecommunications technology has entered its third wave with
VoIP (SHERBURNE, Phil and Fitzgerald, Cary, 2004).



Voice over Internet Protocol is a technology based on
Internet Protocol (IP). More specifically, VoIP allows the transmission of
voice using a broadband Internet connection instead of a regular (or analog)
phone line (BLAKE, Errol A, 2007). Users are able to connect in any kind of
IP-based networks such as Internet, Intranets or local networks (LAN). The
signal of the voice is digitized; compressed and converted in IP packets and is
finally transmitted over IP networks. At the receiving side the signal is
reassembled and decoded. Signaling protocols are used to create and end calls,
transmit the required data, locate users and define the characteristics of a
call (KOURKOULI, Maria and Charismoglou, Alexandros, 2009).

   The two standards in
VoIP telephony mostly used today are the SIP (Session Initiation Protocol) and
the H.323 (BEACHELL, Ronald L et al., 2000). SIP (Session Initiation Protocol):
“is an application-layer signaling protocol that is used to establish, modify
and terminate application sessions” (FESSI, Ali et al., 2007). H.323 by ITU
(International Telecommunications Union): It is a system design which describes
the procedures that allow multimedia applications to take place in a packet
switched network (TOGA, James and Ott, Jörg, 1999).



In satellite communications systems different parameters
are used in comparison with the simple telecommunications systems. For this
reason, those systems are more complicated. These parameters will be located
and analyzed, especially those regarding GEO (Geostationary) satellite systems.

Few of the parameters are the propagation delay, BER (Bit
Error Rate), free space loss and Doppler.

A typical RTT (Round Trip Time) for a typical GEO
(Geostationary) satellite system is between 479 and 558 milliseconds (BEACHELL,
Ronald L et al., 2000). This could lead to data aknowledgement timeout if TCP
protocol is used with SIP (Session Initiation Protocol) for example.

Satellite networks can also be characterized as bandwidth
and latency asymmetric.

Satellite network topology is another important factor for
VoIP over satellite performance. All packets are routed through the satellite
gateway. Existing GEO satellites cannot perform switching or routing until
today. That means that if a VoIP call is made from one satellite terminal A to
another satellite terminal B, the traffic will be doubled as it should pass the
satellite twice. So host A will transmit the VoIP data packets to the satellite
and then to the satellite gateway. In order to deliver these packets, this
traffic should again pass through satellite and downloaded from host B.

A satellite network capable for VoIP telephone calls should
take into consideration some parameters such as Latency, Jitter, Packet loss,
QoS (Quality of service) and traffic prioritization, compression technologies
and required bandwidth per VoIP call (Voice over IP (VoIP) over satellite,



Latency is metering the time that packets need to reach
their destination. This delay is caused because of the distance between earth
and GEO satellites and also because of the fact that the signal is transferred
with the speed of light. Packet latency is translated to voice delay in VoIP

There are three delay types on satellite part of the
communication. Firstly, propagation delay, secondly transmission delay and finally
queuing delay. The most important one is the propagation delay which is roughly
300 ms for one way communication. The problem of different delay values can be
effectively faced by placing Doppler buffers (HENDERSON, Thomas R and Katz,
Randy H, 1999).



Jitter is measuring how late or early a signal is
transmitted to the recipient. The packets are sent at equal intervals from the
transmitter but are received at irregular intervals. This can result to voice
quality degradation (HANCOCK, Johnnie, 2004).



High average of packet loss due to the satellite
environment is also a significant problem in addition to the usage of UDP (User
Datagram Protocol), as there is no packet resending or correction. Packet loss
would lead to voice interuptions at the receiver’s end and the speech quality
would be degraded.

In order to reduce voice interuption, low BER (Bit Error
Rate) should be achieved (Voice over IP (VoIP) over satellite, 2008).

According to Degermark
et al. the BER (Bit Error Rate) can be reduced with header compression. With
UDP packet header compression the transmitted bits over the link are reduced,
so the quality of service for over wireless communication is improved.


and traffic prioritization

Congested packet switched network could delay, loose or
send out of sequence the packets of a VoIP call that is active. While other
services using other protocols may not be affected, the VoIP call will have
lower quality.

For Standard plan users, a better performance on this level
could be achieved by applying relevant algorithm on satellites. Sunggu Choi et al. 2007 have proposed a
MAC layer PRB (Physical resource block) algorithm for VoIP calls over 3G mobile
communications. This algorithm would be activated dynamically when a VoIP call
is occurring on a given time, giving priority on VoIP packets. A similar
adoption in satellite communications could help “Standard users”. However,
commercially is still difficult to apply such algorithm.



Voice encoding is also an important factor for VoIP over
satellite communications. The most important codec attribute is the compression
provided, leading to low bit rate.

Common codecs in satellite communications are IMBE
(Improved Multi-Band Excitation) and AMBE (Advanced Multi-Band Excitation) from
DVSI (Digital Voice System Inc.) providing rates from 2 kbps to 9.6 kbps
(MCCREE, Alan, 2005).

ITU (International Telecommunications Union) has proposed
G.729 codec providing rates of 8 kbps (NextGen Datacom Inc, 2009).


bandwidth per VoIP call

For proper network design it would be useful to know the
available bandwidth per call, in order to select the proper codec and the
compression that would be applied. In this chapter, the satellite
communications parameters that affect the VoIP calls were analyzed and several
parameters that could improve the quality were considered.



In this paper some standards and protocols were described
and implemented for VoIP over satellite communications. The parameters and
whims of using such a link for VoIP were also discussed. Concluding and further
to the references used, traffic prioritization, header compression, Doppler
Buffer, Jitter Buffer seem to improve VoIP performance although at the same
time delay is increased. What is more, SIP standard and AMBE 3000 are regarded
as the more appropriate to be used via satellite link. A further step of this
work would be to test the VoIP over a satellite link and configure all the
above parameters.


SHERBURNE, Phil and Cary FITZGERALD. 2004. You Don’t Know
Jack About VOIP. QUEUE. 2(6), pp.31-38.


BLAKE, Errol A. 2007. Network security: VoIP security on
data network–a guide. In: Proceedings of the 4th annual conference on
Information security curriculum development. Kennesaw: InfoSecCD ’07.


KOURKOULI, Maria and Alexandros CHARISMOGLOU. 2009. VoIP
and voice transmission over packet switched networks. online. Accessed 2
January 2012. Available from World Wide Web:


BEACHELL, Ronald L, Terje BøHLER, Peter D HOLMES et al.
2000. IP MultiMedia over Satellite. Oslo: Norsk Negresental.


FESSI, Ali, Heiko NIEDERMAYER, Holger KINKELIN, and Georg
CARLE. 2007. A Cooperative SIP Infrastructure for Highly Reliable
Telecommunication Services. In: Proceedings of the 1st international conference
on Principles, systems and applications of IP telecommunications.


TOGA, James and Jörg OTT. 1999. ITU-T standardization
activities for interactive multimedia communications on packet-based networks:
H.323 and related recommendations. Computer Networks: The International Journal
of Computer and Telecommunications Networking – Special issue on Internet
telephony. 31(3), pp.205-223.


HENDERSON, Thomas R and Randy H KATZ. 1999. Transport
Protocols for Internet-Compatible Satellite Networks. Selected Areas in
Communications. 17(2), pp.326-344.


HANCOCK, Johnnie. 2004. Jitter—Understanding it,
Measuring It, Eliminating It Part 1: Jitter Fundamentals. High Frequency
Electronics, April, pp.44-50.


DEGERMARK, Mikael, Mathias ENGAN, Björn NORDGREN, and
Stephen PINK. 1996. Low-loss TCP/IP header compression for wireless networks.
In: Proceedings of the 2nd annual international conference on Mobile computing
and networking. New York: ACM, pp.1-14.


CHOI, Sunggu, Kyungkoo JUN, Yeonseung SHIN et al. 2007.
MAC Scheduling Scheme for VoIP Traffic Service in 3G LTE. In: Vehicular
Technology Conference. Koea: VTC-2007 Fall. 2007 IEEE 66th, pp.1441-1445.


Lexington. NextGen Datacom Inc. 2009. online. Accessed 27 Dec 2012.
Available from World Wide Web:


NextGen Datacom Inc. 2009. online. Accessed 27 Dec
2012. Available from World Wide Web:

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