Understanding Voice Over Internet Protocol (UVoIP)


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Voice over Internet Protocol (VoIP) is a technology that is able to pass voice, video and data traffic in the form of packets through an IP network. The IP network itself is a packet-switch based data communication network, so in making calls using an IP network or the Internet. By making calls using VoIP, many advantages can be taken, including in terms of cost, it is clearly cheaper than traditional telephone rates, because the IP network is global. So that international relations can be reduced by up to 70%. In addition, maintenance costs can be reduced because voice and data networks are separate, so that IP Phones can be added, moved and changed. This is because VoIP can be installed on any ethernet and IP address, unlike traditional telephones that must have their own port in the Central or PBX.

Figure 5.32. VoIP
Figure 5.32. VoIP

The rapid development of internet technology is driving towards convergence with other communication technologies. Standardization of communication protocols in VoIP technology such as H.323 has enabled integrated communication with other communication networks such as PSTN.

1. Delay

In designing a VoIP network, delay is a problem that must be taken into account because the quality of sound depends on the delay time. The maximum delay recommended by ITU for voice applications is 150 ms, while the maximum delay with sound quality that is still acceptable to users is 250 ms. End-to-end delay is the sum of the analog-digital voice conversion delay, the packetization time delay or can also be called the packet length delay and the network delay at time t (time). Several delays that can interfere with sound quality in designing a VoIP network can be grouped into:

  • Propagation delay (delay that occurs due to transmission over distance between sender and receiver).
  • Serialization delay (delay during the process of placing bits into the circuit).
  • Processing delay (delay that occurs during the coding, compression, decompression and decoding processes).
  • Packetization delay (delay that occurs during the digital voice sample packetization process.
  • Queuing delay (delay due to waiting time for a package to be served).
  • Jitter buffer (delay due to the presence of a buffer to overcome jitter).

In addition, other parameters that affect are Quality of Service (QoS), so that the sound results are the same as using a traditional telephone (PSTN). Some parameters that affect QoS include: 

  • Fulfillment of bandwidth needs
  • Data delay (latency)
  • Packet loss and desequencing
  • Types of data compression
  • Equipment interoperability (different vendors)
  • The type of multimedia standard used (H.323/SIP/MGCP).

To communicate using VoIP technology that must be real time is jitter, echo and packet loss. Jitter is a variation of delay that occurs due to the difference in time or interval between packet arrivals at the receiver. To overcome jitter, incoming data packets are first collected in the jitter buffer for a predetermined time until the packet can be received on the receiver side in the correct order. Echo is caused by the difference in impedance of the network using four-wire with two-wire. The echo effect is an effect experienced by hearing one's own voice while having a conversation.

2. Bandwidth

It has been explained above that bandwidth is the maximum speed that can be used to transmit data between computers on an IP network or the internet. In VoIP design, bandwidth is something that must be taken into account in order to meet customer needs that can be used as a parameter to calculate the amount of equipment needed in a network. This calculation is also very necessary in network efficiency and costs and as a reference for meeting the needs for future development. Packet loss (loss of data packets in the transmission process) and desequencing are problems related to bandwidth requirements, but are more influenced by the stability of the route passed by data on the network, efficient queuing methods, router settings, and the use of congestion control (excess data load) on the network. Packet loss occurs when there is a buildup of data on the path passed and causes a buffer overflow on the router.

2.1 VoIP Network Supporting Protocols

a. TCP/IP protocol

TCP/IP (Transfer Control Protocol/Internet Protocol) is a protocol used on the Internet network. This protocol consists of two major parts, namely TCP and IP. Illustration of data processing to be sent using the TCP/IP protocol is given in the image below.

Figure 5.33. TCP/IP Protocol Mechanism
Figure 5.33. TCP/IP Protocol Mechanism

b. Application layer

The main function of this layer is file transfer. File transfer from one system to another different system requires a control system to overcome the incompatibility of different file systems. This protocol is related to the application. One example of a well-known application is HTTP (Hypertext Transfer Protocol) for the web, FTP (File Transfer Protocol) for file transfer, and TELNET for remote virtual terminals.

c. Transmission Control Protocol (TCP)

In transmitting data on the Transport layer, there are two protocols that play a role, namely TCP and UDP. TCP is a connection-oriented protocol, which means it maintains the reliability of end-to-end communication relationships. The basic concept of how TCP works is to send and receive segments of information with varying data lengths on an internet datagram. TCP ensures the reliability of communication relationships because it repairs damaged, lost or mis-sent data. This is done by assigning a sequence number to each octet sent and requiring a positive response signal from the recipient in the form of an ACK (acknowledgement) signal. If this ACK signal is not received at a certain time interval, the data will be resent. On the recipient side, the sequence number is useful for preventing data sequence errors and data duplication. TCP also has a flow control mechanism by including information in the ACK signal regarding the limit on the number of data octets that may still be transmitted on each segment that is successfully received.

In VoIP connection, TCP is used during signaling, TCP is used to ensure the setup of a call in a signaling session. TCP is not used in sending voice data on VoIP because in a VoIP data communication handling data that experiences delays is more important than handling lost packets.

d. User Datagram Protocol (UDP)

UDP, which is one of the main protocols above IP, is a simpler transport protocol compared to TCP. UDP is used for situations that do not prioritize reliability mechanisms. The UDP header only contains four fields, namely source port, destination port, length and UDP checksum, where its function is almost the same as TCP, but the checksum facility in UDP is optional. UDP in VoIP is used to send audio streams that are sent continuously. UDP is used in VoIP because in sending audio streaming that takes place continuously, it is more important for the speed of data delivery to arrive at the destination without paying attention to lost packets even though it reaches 50% of the number of packets sent. (VoIP) fundamental, Davidson Peters, Cisco System, 163) because UDP is able to send streaming data quickly, then in VoIP technology UDP is one of the important protocols used as a header in sending data besides RTP and IP. To reduce the number of packets lost during data transmission (because there is no retransmission mechanism), in VoIP technology, data transmission is often done on a private network.

e. Internet Protocol (IP)

Internet Protocol is designed for interconnection of computer communication systems on packet-switched networks. On a TCP/IP network, a computer is identified by an IP address. Each computer has a unique IP address, each different from the others. This is done to prevent errors in data transfer. Finally, the data access protocol is directly related to the physical media. In general, this protocol is tasked with handling error detection during data transfer. For data communication, the Internet Protocol implements two basic functions, namely addressing and fragmentation. One of the important things in IP in sending information is the sender and recipient addressing method. Currently there is an addressing standard that has been used, namely IPv4 with an address consisting of 32 bits. The number of addresses created with IPv4 is estimated to be insufficient to meet the needs of IP addressing so that in the next few years a new addressing system will be implemented, namely IPv6 which uses a 128-bit addressing system.

3. VoIP Applications 

One of the available VoIP applications is Skype. Skype is an IP-based voice communication application software via the internet between fellow Skype users. When using Skype, Skype users who are online will search for other Skype users and then start building a network to find other users. Skype has a variety of features that can make it easier for its users. Skype is also equipped with SkypeOut and SkypeIn which allow Skype users to connect with conventional telephone users and mobile phones.

Skype uses the HTTP protocol to communicate with the Skype server for username/password authentication and registration with the Skype directory server. A modified version of the HTTP protocol is used to communicate with other Skype clients. The advantage of this application is the availability of security services in the transmission of data in the form of voice.

4. Advantages of VoIP

  • Lower costs for long distance calls. The main emphasis of VoIP is cost. With two locations connected to the internet the cost of a call becomes very low.
  • Leverage existing data network infrastructure for voice. Useful if the company already has a network. If possible, the existing network can be built into a VoIP network easily. No additional monthly fees are required for adding voice communications.
  • Smaller bandwidth usage than regular telephone. With the advancement of technology, bandwidth usage for voice is now very small. Data compression techniques allow voice to only require about 8 kbps of bandwidth.
  • Allows to be combined with existing local telephone networks. With the gateway, the VoIP network can be connected to the PABX in the office. Communication between offices can use regular telephones.
  • Various forms of VoIP networks can be combined into a large network. An example in Indonesia is VoIP Merdeka.
  • Variations in the use of existing equipment, for example from a PC connected to a regular telephone, IP phone handset.

5. Disadvantages of VoIP

  • The sound quality is not as clear as Telkom. It is the effect of sound compression with small bandwidth, so there will be a decrease in sound quality compared to conventional PSTN networks.
  • There is a delay in communication. The process of converting data to voice, network delays, create a delay in communication using VoIP. Unless using a Broadband connection (see above).
  • If you are not connected to the internet for 24 hours, you need to make an appointment to get in touch.
  • If you use the internet and the computer is behind NAT (Network Address Translation), then a special configuration is needed to make VoIP work.
  • There is never a guarantee of quality when VoIP goes over the internet.
  • Equipment is relatively expensive. VoIP equipment that connects VoIP to PABX (IP telephony gateway) is relatively expensive. It is expected that with the increasing popularity of VoIP, the price of this equipment will also start to decrease.
  • Potential to cause network congestion/stuck. If VoIP usage increases, then there is a potential for the existing data network to become full if not managed properly. Bandwidth management is necessary so that the network in the company does not become saturated due to VoIP usage.
  • Uncoordinated network merging will cause chaos in the numbering system.

What is Video Conferencing?

Video conferencing is the use of audio and video equipment to hold conferences with people in different locations. This service system is currently still used only to a limited extent. Current users are business and industrial sectors such as financial institutions. Multimedia satellite systems are a very suitable infrastructure for video conferencing compared to other networks because of their flexibility and ease of installation anywhere.

Telecommunications Video conferencing uses video and audio to bring people at different locations together for a meeting. This can be as simple as a conversation between two people in private offices (point-to-point) or involve several locations (multi-point) with more than one person in a large room at different locations. In addition to audio and visual transmission, video conferencing can be used to share documents, computer-displayed information, and whiteboards.

Simple analog videoconferencing dates back to the invention of television. Such a videoconferencing system consisted of two closed-circuit television systems connected via cable. During the first space flights, NASA used two radiofrequency (UHF or VHF) bands, one in all directions. TV channels used this type of videoconferencing, for example, reporting from remote locations. Later mobile communications to satellites using special trucks became necessary.

Figure 5.30. First Video Conferencing in 1968
Figure 5.30. First Video Conferencing in 1968

This technique was very expensive, however, and could not be used for more mundane applications, such as telemedicine, distance education, business meetings, and so on, especially in long-distance applications. Attempts to use normal telephony networks to transmit slow-scan video, such as the first system developed by AT&T, failed mostly due to poor picture quality and the lack of efficient video compression techniques. The larger 1 MHz bandwidth and 6 Mbit/s bit rate Picturephone of the 1970s also did not produce good service.

The core technology used in a videoteleconference (VTC) system is the compression of digital video and audio streams in real time. The hardware or software that performs the compression is called a codec (coder/decoder). Compression rates up to 1:500 can be achieved. The resulting digital stream of 0's and 1's is subdivided into labeled packets, which are then transmitted over a digital network (usually ISDN or IP). The use of audio modems in the transmission path precludes the use of POTS, or simply System Old Telephone, in some low-speed applications, such as videotelephony, because they convert digital to/from analog waves in the audio spectrum.

Figure 5.31. Modern Dual Plasma Video Conferencing System
Figure 5.31. Modern Dual Plasma Video Conferencing System

Other components required for a VTC system include: 

  1. Video input: video camera or webcam
  2. Video output: computer monitor, projector or television
  3. Audio input : microphone
  4. Audio output: generally the speaker is connected to a telephone or other display.
  5. Data transfer: digital or analog telephone network, LAN or Internet.

The influences on video teleconferencing include:

1. In the general public

High-speed Internet connectivity has become more widely available at a reasonable cost and the cost of video capture and display technology has decreased. Consequently, personal video teleconferencing systems based on a webcam, personal computer system, streaming software and broadband Internet connectivity have become affordable to the masses. Also, the hardware used for this technology has continued to improve in quality, and prices have dropped dramatically. The availability of freeware often as part of a chat program has made software-based videoconferencing accessible to many.

2. In education

Videoconferencing provides students with the opportunity to learn by taking part in a 2-way communication platform. Moreover, teachers and lecturers from all over the world can be brought into the classroom in remote or otherwise isolated locations. Students from different communities and backgrounds can come together to learn about each other. Students can investigate, communicate, research and share ideas and information with each other. Through video conferencing students can visit other parts of the world to talk to others, visit a zoo, a museum and so on, to learn. Here are some examples of how video conferencing can benefit people around campus, faculty members stay in touch with the class while being swept up one week at a conference guest lecturers are brought into a classroom from another institution researchers collaborate with colleagues at other institutions on a regular basis without losing time due to travel.

3. In medicine and health

Videoconferencing is a very useful technology for telemedicine and telenursing applications, such as diagnosis results, consultation, transmission of medical images, etc., in real time. Using VTC, patients can contact doctors and nurses in routine or emergency situations, doctors and other paramedical professionals can discuss cases across long distances.

Specialized devices such as microscopes fitted with digital cameras, videoendoscopes, medical ultrasound imaging devices, ear probes, etc., can be used in conjunction with VTC equipment to transmit data about a patient.

4. In business

Videoconferencing can allow individuals in remote locations to have meetings at short notice. The money and time that would be spent traveling can be used to have short meetings. Technologies such as VOIP can be used in conjunction with desktop videoconferencing to allow face-to-face business meetings without leaving the desktop, especially for businesses with wide-spread offices. The technology is also being used for telecommuting, where employees work from home. Videoconferencing is now being introduced to online networking websites, in order to help businesses form profitable relationships quickly and efficiently without leaving their workplace.


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