SIP Server Based VoIP Technology
Ashwin Ronald .S –
MSc Telecommunication and Electronic Engineering
Department of Arts, Computing, Engineering and Sciences (ACES)
this coursework report, the objective is to implement the Voice over Internet Protocol (VoIP)
call using TrueConf SIP Server and provide key results based on the analysis of
the quality of VoIP call. Voice over
Internet Protocol (VoIP)
is a technology for the delivery of voice
communications over Internet
Protocol (IP) networks, such as the
Internet. Voice over Internet Protocol has several advantages over PSTN systems.
There are, however, some issues related to call quality and
security, particularly over Wi-Fi, to be solved before VoIP is truly accepted
as a PSTN/PBX replacement. Therefore, this report aims to present a brief
introduction on the basic principles and technologies of SIP based VoIP
networks, their advantages and disadvantages, and analysis of the
wired/wireless VoIP calls established using TrueConf SIP Server.
of Operation of VoIP Systems
Internet Protocol (VoIP) as the name suggests is the
technology that is slowly replacing the PSTN telephony system, unlike PSTN
network, VoIP systems are used to make calls between two end users by
transmitting the voice packets via Internet rather than traditional telephony
lines. It is responsible for delivering voice communications over Internet
Protocol (IP) networks. The basic principle of operation of VoIP telephone
calls are very similar to traditional telephony. The steps involved are signalling, setting up of the channel,
conversion of analog signals to digital signals, and encoding. The digital information is packetized, and the packetized
data are transmitted as IP packets over a packet-switched network instead of
being transmitted over a circuit-switched network. The media streams are
transported using special media delivery protocols that encode audio and video
with audio codecs, and video
codecs (The term codec is the short form for “encoder/decoder”). The media streams are encoded at the transmitter
end, sent over the internet and decoded at the receiver end. The codecs must
match at each end of the communication link. There are numerous codecs that
optimize the based on the requirements of the application and also based on
available network bandwidth. Some implementation of such codecs usually relies
on narrowband and compressed
speech, while others support high-fidelity stereo codecs. Some popular codecs include µ-law and a-law versions of G.711, G.722, an open source voice codec known as iLBC,
and many others (“Voice over IP”, 4th December 2017, para. 1 &
Voice over Internet Protocol
telephony has numerous advantages over PSTN telephony systems. To make
makes calls using VoIP technology, it just requires internet connection.
Whereas, PSTN requires dedicated telephone lines from the service provider. For
the additional features like music on hold, caller ID, call waiting and
conferencing, the user must pay extra for PSTN networks, but these features are
available for free for VoIP users. Another disadvantage of PSTN network over
VoIP is it is very complicated to expand and upgrade PSTN capabilities and may
involve several additions of hardware and adding lines. But software upgrades
and additional bandwidth is all it takes for upgrading the VoIP. Making
calls using VoIP is cheap and easy with proper usage of bandwidth.
in Implementation of VoIP Systems
“Voice over Internet
Protocol (VoIP) has become a popular alternative to traditional public-switched
telephone network (PSTN) networks that provides advantages of low cost and
flexible advanced digital features. The flexibility of the VoIP system and the
convergence of voice and data networks brings with it additional security risks”
(Butcher, Li & Guo, 2007). Therefore, it is necessary to analyse the
challenges involved in implementing the VoIP systems. Some of the major
drawbacks of VoIP systems are as follows:
Quality of Service (QoS) –
The key parameter
in VoIP network’s operation is Quality of service (QoS). A VoIP application is
much more sensitive to delays than its traditional data counterparts. VoIP
suffers Jitter also referred to nonuniform delays, it can cause packets to
arrive and processed out of sequence. The protocol used for transferring
packets is Real-Time Transport Protocol (RTP), it is based on UDP, therefore
the packets which arrive out of sequence cannot be reassembled at the receiver
end, that is at the transport level, but must be reordered at the application
level, introducing significant overhead. The jitter can be controlled by using
buffers by network designers and implement QoS-supporting network elements
(especially routers) that let VoIP packets “play through” when larger data
packets are scheduled ahead of them. Another issue that must be dealt with is
the packet loss. However, VoIP packets are small and the loss of very minute
amount of speech packets caused in VoIP networks are not noticed by the human
user (Walsh & Kuhn, 2005).
call capacity –
One of the most
important aspect to take into consideration during deployment of Voice over
Internet Protocol (VoIP) network is the bandwidth consumption. If this factor is ignored then it will severely limit
the reliability of the VoIP system and place a huge burden on the network
infrastructure and VoIP Server in question (Soroyewun
& Obiniyi, 2015). A link which has high bandwidth may be able to
carry large volumes of data packets, but the network links with low bandwidth
can cause packets loss and other QoS problems. Hence, proper bandwidth
reservation and allocation is essential to VoIP quality. Sharing data and voice
on the same wires, which is one of the great attractions of VoIP, is also a
potential challenge for implementation.
of CODEC –
conversion of analog voice signals to digital packets that are sent over
internet are done by the codecs in the VoIP systems. Therefore, choice of codec
is an important function. It is necessary to have higher priority voice packets
to enable them to pass through firewalls. This is often necessary to ensure
that audio does not get dropped during calls. The way the audio is compressed
is the major difference between the various types of VoIP codecs used by providers.
Unfortunately, for all circumstances there is no perfect codec that is
acceptable for all users. At the end of the day, the aspect that affects the
final choice of codec is the quality. Some high-quality codecs require
licensing fees whereas others are open source, some offer better quality while
others use less bandwidth.
major security threat in wireless communication of VoIP systems is
eavesdropping. A third party can obtain names, password, phone
numbers and private information through eavesdropping,
allowing them to gain control over voicemail, calling plan, call forwarding and
billing information. This subsequently leads to service theft. VoIP systems
consists of softphones and software that are vulnerable to viruses and malware
attacks. Another security threat is call tampering.
TrueConf SIP Server
TrueConf is a Russian company that produces software-based video
conferencing and unified
communications solutions. Company’s
applications are designed for usage in conference or meeting rooms, at
workplaces, and on mobile devices. The company is headquartered in Moscow.
TrueConf is a member of AVIXA, the Audio-visual and Integrated Experience
Association. TrueConf’s software architecture is based on Scalable
Video Coding (SVC). During a multipoint video
conference each conference participant receives an individual set of video
streams adjusted in accordance with bandwidth, network connection, and
endpoint’s capabilities. As opposed to SVC technology, traditional video
conferencing based on Multipoint Control Units (MCUs) performs transcoding server-side, which is not
scalable thus unsuitable for large deployments. TrueConf considers SVC-based
architecture its competitive advantage as it significantly lowers
infrastructure costs and easily adapts to any necessary device (“TrueConf”, 5th
October 2017, para. 1 & 2). TrueConf server is best used for video
conferencing which enables up to 25 participants. However, in free version of
TrueConf server up to 6 participants can engage in video conferencing. In
short, TrueConf Session Initiation Protocol (SIP) server was chosen because it
is easier to install, setup and configure on Windows Operating Systems, and it
is specifically designed for more enhanced feature that is, video conferencing.
Setup and Configuration of TrueConf SIP Server
TrueConf Server is shipped as a software
installation package that contains the server side and client applications for
popular platforms. After the installation package is downloaded, it is launched
to begin the installation. TrueConf Web Manager port is determined during the
server installation. Since we were installing the server behind the firewall,
to complete the registration TCP port 4310 was used to access from inside to
internet and port 80 was used between server and client applications. To
complete the registration, the free registration key was obtained. On top of
the activation window sign appeared on the server status stating that the
server is running and registered when the server
had been successfully registered. Once the server has been configured, the
TrueConf client software is installed on the client machine. Then internal and
external addresses must be specified. Internal addresses and ports are used for clients to
contact this server. By default, the server uses all IP addresses of machine on
default TrueConf Server port 4307. When default settings are on, current
connections are displayed in this column. External
addresses are the ports and IP addresses or DNS names, which help
client applications to connect to the server (“TrueConf Server Administrator Guide”, n.d.). TrueConf Server has
built-in gateway for SIP protocols interoperability.
Practical Implementation of
Server application has a dedicated softphone for making video calls. The video
call between two end-users can be established by just entering the user ID or
the IP address of the other user in the address bar as shown in figure 1. And
figure 2 shows the TrueConf softphone with video call in progress.
The five main functions of the TrueConf SIP server in
implementation of the VoIP calls are first the TrueConf SIP server determines
the user location and the type of end system that will be used by the session.
Second function of the SIP server is to determine the user availability. The
user availability status can be informed by the users to the system, that they
are available or can inform that they are busy and not to be disturbed. The
third SIP function is the user capabilities function. This function is
important because various devices can have numerous capabilities. For example,
computer can do more things than a phone, therefore, the user capabilities
function allows the SIP to determine the media being used and of the parameters
being associated with it. The fourth function is the session setup. This is
responsible for the call connection. The session between the caller and the
recipient of the call will be established. The fifth and final function of the
TrueConf SIP server is the session management. The users will be allowed to end
a call, add multiple users to make a conference call or make modifications to
the session parameters by this function (“An
introduction to SIP and SIP functions”, n.d.).
Results and Drawbacks
The VoIP video calls using TrueConf SIP server were
successfully made through wired connection as well as through the wireless
connection via Wi-Fi. The sample packets were captured and analysed with the
observer. Problems with VoIP audio quality are always due to network delay,
jitter and packet loss. Observer helps in tracking network factors that affect
quality and reports call quality scores, which measure the overall VoIP network
health. Figure 3 shows the traffic summary of the call made through wired
connection using codec G.711.
Figure 4 and
figure 5 below, shows the statistics of the overall call quality of the call
made through wired connection using codec G.711. The total number of VoIP
packets were 32,955. And the sample packets show how signalling, call set up/take down,
data transfer occur during a typical session, duration of the call and the
average jitter rate.
Figure 6 shows the traffic summary of the call made
through wireless connection using codec G.711.
Figure 7 and figure 8 below, shows the statistics of
the overall call quality of the call made through wired connection using codec
G.711. The total
number of VoIP packets were 34,345.
In this project, two
types of codecs were used – G.711 and G.722.
Codec G.711 offers narrow-band voice and requires
low processing, and has the typical bit-rates of 32-64 kbps. G.711
is currently used in a wide domain of applications,
it performs best in local
networks, therefore best suited for wired networks. Codec G.722 offers wideband voice and requires low DSP processing.
provides improved speech quality due to a wider speech bandwidth of
50–7000 Hz compared to narrowband speech coders like G.711. However, 48-64 kbps bit rate per
channel is required. This codec is best suited for calls over wireless
VoIP has numerous
advantages and upper hand with respect to budget scalability in comparison with
the present PSTN telephony systems. The overall costs of the VoIP networks over
the PSTN networks which are circuit-switched networks is reduced by allowing
the lines to be shared with other users and services. However, because of
majorly being software based network it is exposed to the increasing threat of
cyber-attacks, therefore, vulnerable to malware attacks such as viruses. The
VoIP voice packets must travel across numerous types of networks which are
often heterogeneous in nature. Thus, the Quality of Service (QoS) is degraded
while the voice packets pass across such kinds of networks. The Quality of Service
of the VoIP networks can be evaluated based on parameters such as jitter,
end-to-end delay and Mean Opinion Score (MOS). The quality of the voice over a VoIP
networks is measured using the Mean Opinion Score (MOS) (Miraz, Molvi, Ganie, Ali &
Hussein, 2017). For the calls made
over Wi-Fi networks, the MOS scores decreases with time and
increased level of VoIP traffic. Whereas, there are better MOS scores for the
calls made over wired networks.
this coursework, we implemented the VoIP calls using TrueConf SIP server. The
video calls were established both on wired and wireless based networks. The
overview of the principle of operation of VoIP, advantages and major drawbacks
were presented in this report. Next the overview of TrueConf SIP server, its
setup and configuration process, and its function was presented. Then the voice
packets captured during laboratory investigations were analysed and was presented
to show how signalling, call is setup and terminated, and data is transferred
during a typical session. Finally, difficulties in making secure video calls at acceptable QoS levels over Wi-Fi networks
based on the lab results using different codecs (G.711 and G.722) were
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