TrueConfSIP Server Based VoIP TechnologyAshwin Ronald .S -26035789MSc Telecommunication and Electronic EngineeringDepartment of Arts, Computing, Engineering and Sciences (ACES)Sheffield HallamUniversity AbstractInthis 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 ofthe quality of VoIP call. Voice overInternet Protocol (VoIP)is a technology for the delivery of voicecommunications over InternetProtocol (IP) networks, such as theInternet.
Voice over Internet Protocol has several advantages over PSTN systems.There are, however, some issues related to call quality andsecurity, particularly over Wi-Fi, to be solved before VoIP is truly acceptedas a PSTN/PBX replacement. Therefore, this report aims to present a briefintroduction on the basic principles and technologies of SIP based VoIPnetworks, their advantages and disadvantages, and analysis of thewired/wireless VoIP calls established using TrueConf SIP Server. Principleof Operation of VoIP SystemsVoice overInternet Protocol (VoIP) as the name suggests is thetechnology that is slowly replacing the PSTN telephony system, unlike PSTNnetwork, VoIP systems are used to make calls between two end users bytransmitting the voice packets via Internet rather than traditional telephonylines. It is responsible for delivering voice communications over InternetProtocol (IP) networks. The basic principle of operation of VoIP telephonecalls 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 packetizeddata are transmitted as IP packets over a packet-switched network instead ofbeing transmitted over a circuit-switched network.
The media streams aretransported using special media delivery protocols that encode audio and videowith audio codecs, and videocodecs (The term codec is the short form for “encoder/decoder”). The media streams are encoded at the transmitterend, sent over the internet and decoded at the receiver end. The codecs mustmatch at each end of the communication link. There are numerous codecs thatoptimize the based on the requirements of the application and also based onavailable network bandwidth. Some implementation of such codecs usually relieson narrowband and compressedspeech, 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 &2).Voice over Internet Protocoltelephony has numerous advantages over PSTN telephony systems. To makemakes calls using VoIP technology, it just requires internet connection.Whereas, PSTN requires dedicated telephone lines from the service provider. Forthe additional features like music on hold, caller ID, call waiting andconferencing, the user must pay extra for PSTN networks, but these features areavailable for free for VoIP users.
Another disadvantage of PSTN network overVoIP is it is very complicated to expand and upgrade PSTN capabilities and mayinvolve several additions of hardware and adding lines. But software upgradesand additional bandwidth is all it takes for upgrading the VoIP. Makingcalls using VoIP is cheap and easy with proper usage of bandwidth. Challengesin Implementation of VoIP Systems”Voice over InternetProtocol (VoIP) has become a popular alternative to traditional public-switchedtelephone network (PSTN) networks that provides advantages of low cost andflexible advanced digital features. The flexibility of the VoIP system and theconvergence of voice and data networks brings with it additional security risks”(Butcher, Li & Guo, 2007). Therefore, it is necessary to analyse thechallenges involved in implementing the VoIP systems.
Some of the majordrawbacks of VoIP systems are as follows:a) Quality of Service (QoS) – The key parameterin VoIP network’s operation is Quality of service (QoS). A VoIP application ismuch more sensitive to delays than its traditional data counterparts. VoIPsuffers Jitter also referred to nonuniform delays, it can cause packets toarrive and processed out of sequence. The protocol used for transferringpackets is Real-Time Transport Protocol (RTP), it is based on UDP, thereforethe packets which arrive out of sequence cannot be reassembled at the receiverend, that is at the transport level, but must be reordered at the applicationlevel, introducing significant overhead. The jitter can be controlled by usingbuffers by network designers and implement QoS-supporting network elements(especially routers) that let VoIP packets “play through” when larger datapackets are scheduled ahead of them.
Another issue that must be dealt with isthe packet loss. However, VoIP packets are small and the loss of very minuteamount of speech packets caused in VoIP networks are not noticed by the humanuser (Walsh & Kuhn, 2005).b) Bandwidth/concurrentcall capacity – One of the mostimportant aspect to take into consideration during deployment of Voice overInternet Protocol (VoIP) network is the bandwidth consumption. If this factor is ignored then it will severely limitthe reliability of the VoIP system and place a huge burden on the networkinfrastructure and VoIP Server in question (Soroyewun& Obiniyi, 2015). A link which has high bandwidth may be able tocarry large volumes of data packets, but the network links with low bandwidthcan cause packets loss and other QoS problems. Hence, proper bandwidthreservation and allocation is essential to VoIP quality.
Sharing data and voiceon the same wires, which is one of the great attractions of VoIP, is also apotential challenge for implementation.c) Choiceof CODEC – Theconversion of analog voice signals to digital packets that are sent overinternet are done by the codecs in the VoIP systems. Therefore, choice of codecis an important function. It is necessary to have higher priority voice packetsto enable them to pass through firewalls. This is often necessary to ensurethat audio does not get dropped during calls. The way the audio is compressedis the major difference between the various types of VoIP codecs used by providers.Unfortunately, for all circumstances there is no perfect codec that isacceptable for all users.
At the end of the day, the aspect that affects thefinal choice of codec is the quality. Some high-quality codecs requirelicensing fees whereas others are open source, some offer better quality whileothers use less bandwidth.d) Security– Themajor security threat in wireless communication of VoIP systems iseavesdropping. A third party can obtain names, password, phonenumbers and private information through eavesdropping,allowing them to gain control over voicemail, calling plan, call forwarding andbilling information. This subsequently leads to service theft. VoIP systemsconsists of softphones and software that are vulnerable to viruses and malwareattacks. Another security threat is call tampering.
TrueConf SIP ServerTrueConf is a Russian company that produces software-based videoconferencing and unifiedcommunications solutions. Company’sapplications are designed for usage in conference or meeting rooms, atworkplaces, and on mobile devices. The company is headquartered in Moscow.TrueConf is a member of AVIXA, the Audio-visual and Integrated ExperienceAssociation. TrueConf’s software architecture is based on ScalableVideo Coding (SVC). During a multipoint videoconference each conference participant receives an individual set of videostreams adjusted in accordance with bandwidth, network connection, andendpoint’s capabilities. As opposed to SVC technology, traditional videoconferencing based on Multipoint Control Units (MCUs) performs transcoding server-side, which is notscalable thus unsuitable for large deployments.
TrueConf considers SVC-basedarchitecture its competitive advantage as it significantly lowersinfrastructure costs and easily adapts to any necessary device (“TrueConf”, 5thOctober 2017, para. 1 & 2). TrueConf server is best used for videoconferencing which enables up to 25 participants. However, in free version ofTrueConf server up to 6 participants can engage in video conferencing. Inshort, TrueConf Session Initiation Protocol (SIP) server was chosen because itis easier to install, setup and configure on Windows Operating Systems, and itis specifically designed for more enhanced feature that is, video conferencing.Setup and Configuration of TrueConf SIP ServerTrueConf Server is shipped as a softwareinstallation package that contains the server side and client applications forpopular platforms. After the installation package is downloaded, it is launchedto begin the installation. TrueConf Web Manager port is determined during theserver installation.
Since we were installing the server behind the firewall,to complete the registration TCP port 4310 was used to access from inside tointernet and port 80 was used between server and client applications. Tocomplete the registration, the free registration key was obtained. On top ofthe activation window sign appeared on the server status stating that theserver is running and registered when the serverhad been successfully registered. Once the server has been configured, theTrueConf client software is installed on the client machine. Then internal andexternal addresses must be specified. Internal addresses and ports are used for clients tocontact this server.
By default, the server uses all IP addresses of machine ondefault TrueConf Server port 4307. When default settings are on, currentconnections are displayed in this column. Externaladdresses are the ports and IP addresses or DNS names, which helpclient applications to connect to the server (“TrueConf Server Administrator Guide”, n.d.
). TrueConf Server hasbuilt-in gateway for SIP protocols interoperability. Practical Implementation ofVoIP CallsThe TrueConfServer application has a dedicated softphone for making video calls. The videocall between two end-users can be established by just entering the user ID orthe IP address of the other user in the address bar as shown in figure 1. Andfigure 2 shows the TrueConf softphone with video call in progress.Fig.
1Fig.2The five main functions of the TrueConf SIP server inimplementation of the VoIP calls are first the TrueConf SIP server determinesthe 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. Theuser availability status can be informed by the users to the system, that theyare available or can inform that they are busy and not to be disturbed. Thethird SIP function is the user capabilities function. This function isimportant because various devices can have numerous capabilities.
For example,computer can do more things than a phone, therefore, the user capabilitiesfunction allows the SIP to determine the media being used and of the parametersbeing associated with it. The fourth function is the session setup. This isresponsible for the call connection.
The session between the caller and therecipient of the call will be established. The fifth and final function of theTrueConf SIP server is the session management. The users will be allowed to enda call, add multiple users to make a conference call or make modifications tothe session parameters by this function (“Anintroduction to SIP and SIP functions”, n.d.). Results and DrawbacksThe VoIP video calls using TrueConf SIP server weresuccessfully made through wired connection as well as through the wirelessconnection via Wi-Fi. The sample packets were captured and analysed with theobserver.
Problems with VoIP audio quality are always due to network delay,jitter and packet loss. Observer helps in tracking network factors that affectquality and reports call quality scores, which measure the overall VoIP networkhealth. Figure 3 shows the traffic summary of the call made through wiredconnection using codec G.711.Fig.
3Figure 4 andfigure 5 below, shows the statistics of the overall call quality of the callmade through wired connection using codec G.711. The total number of VoIPpackets 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 theaverage jitter rate. Fig.4 Fig.5Figure 6 shows the traffic summary of the call madethrough wireless connection using codec G.711.
Fig.6Figure 7 and figure 8 below, shows the statistics ofthe overall call quality of the call made through wired connection using codecG.711. The totalnumber of VoIP packets were 34,345.Fig.7Fig.8In this project, twotypes of codecs were used – G.711 and G.
722. Codec G.711 offers narrow-band voice and requireslow processing, and has the typical bit-rates of 32-64 kbps. G.711is currently used in a wide domain of applications,it performs best in localnetworks, therefore best suited for wired networks.
Codec G.722 offers wideband voice and requires low DSP processing.Itprovides improved speech quality due to a wider speech bandwidth of50–7000 Hz compared to narrowband speech coders like G.711.
However, 48-64 kbps bit rate perchannel is required. This codec is best suited for calls over wirelessnetworks.VoIP has numerousadvantages and upper hand with respect to budget scalability in comparison withthe present PSTN telephony systems. The overall costs of the VoIP networks overthe PSTN networks which are circuit-switched networks is reduced by allowingthe lines to be shared with other users and services. However, because ofmajorly being software based network it is exposed to the increasing threat ofcyber-attacks, therefore, vulnerable to malware attacks such as viruses. TheVoIP voice packets must travel across numerous types of networks which areoften heterogeneous in nature. Thus, the Quality of Service (QoS) is degradedwhile the voice packets pass across such kinds of networks.
The Quality of Serviceof 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 VoIPnetworks is measured using the Mean Opinion Score (MOS) (Miraz, Molvi, Ganie, Ali &Hussein, 2017). For the calls madeover Wi-Fi networks, the MOS scores decreases with time andincreased level of VoIP traffic.
Whereas, there are better MOS scores for thecalls made over wired networks. ConclusionInthis coursework, we implemented the VoIP calls using TrueConf SIP server. Thevideo calls were established both on wired and wireless based networks. Theoverview of the principle of operation of VoIP, advantages and major drawbackswere presented in this report. Next the overview of TrueConf SIP server, itssetup and configuration process, and its function was presented. Then the voicepackets captured during laboratory investigations were analysed and was presentedto show how signalling, call is setup and terminated, and data is transferredduring a typical session. Finally, difficulties in making secure video calls at acceptable QoS levels over Wi-Fi networksbased on the lab results using different codecs (G.
711 and G.722) werediscussed. References 1 Voice over IP. (4th December2017). In Wikipedia. RetrievedDecember 19, 2017, from https://en.
wikipedia.org/wiki/Voice_over_IP 2 Butcher,D., Li, X., & Guo, J. (2007). Security challenge and defense in VoIPinfrastructures.
IEEE Transactions on Systems, Man, and Cybernetics,Part C (Applications and Reviews), 37(6), 1152-1162. 3 Walsh, T.J., & Kuhn, D.
R. (2005). Challenges in securing voice over IP. IEEESecurity & Privacy, 3(3), 44-49.
4 Soroyewun,M. B., & Obiniyi, A. A (2015).
Empirical Study of Achievable BandwidthCapacity of VoIP Infrastructure in an Intranet with Open Source Tools. 5 TrueConf. (5th October 2017).In Wikipedia.
Retrieved December 19,2017, from https://en.wikipedia.org/wiki/TrueConf 6 TrueConf Server Administrator Guide.
(n.d.). Retrieved December 19, 2017, from https://trueconf.com/support/online-help/server-help.
html 7 An introduction to SIP and SIP functions.(n.d.). Retrieved January 4, 2018, from http://searchunifiedcommunications.
techtarget.com/tip/An-introduction-to-SIP-part-1 8 Miraz, M.H., Molvi, S.
A., Ganie, M. A., Ali, M., & Hussein, A. H. (2017).
Simulation and Analysis of Quality of Service (QoS) Parameters of Voice over IP(VoIP) Traffic through Heterogeneous Networks. arXiv preprintarXiv:1708.01572. 9 Mohammed,M. H., & Abdullah, W.
A. N. PERFORMANCE ANALYSIS OF VoIP OVER WIRED ANDWIRELESS NETWORKS: NETWORK IMPLEMENTATION IN ADEN UNIVERSITY.