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VoIP Report

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  • 1. VoIP Report (Voice Transmission Over Data Network) Ibrahim Hakki Bulut 33673458908
  • 2. Outline of VoIP 1. Voip a. Definitions of VoIP b. Brief explanation of voice transmission c. Types of voice transmission i. TDM, FR and ATM ii. IP packets 1. IP to IP 2. IP to PSTN 3. PSTN to PSTN d. Types of VoIP i. IP to IP ii. IP to PSTN iii. PSTN to PSTN e. Circuit Switching and Packet Switching f. Comparision of PSTN and VoIP 2. Codecs 3. Soft Switches 4 . Protocols a. H.323 b. SIP 5. Factors that Affect Call Quality in VoIP a. Packet Loss b. Latency c. Jitter 6. RTP 7. VoIP Packet Headers 8. Advantages of VoIP 9. Disadvantages of VoIP
  • 3. VoIP (Voice Transmission Over Data Network) 1. VoIP When internet started to become popular in its first years, people tried to transmit data over existing voice networks. Dial-up connection over PSTN is one of the example of this. Huge increase of data traffic volume led to build independent data networks. Then people tried to make reverse that they made in first years: to transmit voice over data network. VoIP explains how this technology works. a. Definitions of VoIP VoIP is a method for taking analog audio signals and turning them into digital data that can be transmitted over the internet. A more comprehensive definition may be like this: VoIP is the the technology used to transmit voice conversations over a data network using IP. This technology is done by digitizing voice into discrete packets that are transferred independently over the network, instead of traditional circuit-committed protocols of the PSTN. b. Brief Explanation of Voice Transmission Audio is digitized by using a sampling circuit. Then digitized audio signal is compressed with various methods. Therefore, it uses less bandwidth. Lastly, it is given to data transmission channel. c. Types of Voice Transmission Voice can be transmitted over data network in 2 ways: 1) Voice is transmitted over TDM, FR or ATM data networks without using IP. These are called VoDSL, VoTDM, VoFR or VoATM. These applications have network dependency. This means that endpoints that voice data transfer occurs must be connected to same network. Therefore, these applications can’t be reached from everywhere. 2) Voice is transmitted by converting them into IP packets. IP is a protocol that can be transmitted on every type of data networks. Therefore, it has no network dependency. These applications are called VoIP.
  • 4. Warning: There is a confusing situation. VoIP over ATM network and VoATM are different. In VoATM applicaton, IP is not used. Voice is converted into ATM cells directly. On the other hand, in VoIP application, firstly voice is converted into IP packets. Then, IP packets are converted into ATM cells. This is also valid for VoFR, VoDSL etc. d. Types of VoIP Voice transmission can be carried out over IP networks in 3 ways: 1) IP to IP (Computer to computer) Typically, server-client based programs are used. Users are connected to same server by using their client programs on their PC. Therefore, people who connects to same server talks to each other. Typical examples are MSN and ICQ. 2) IP to PSTN (Computer to Phone) It is carried out by the help of client programs that uses such SIP and H.323 standart VoIP protocols. This client programs connects to commercial servers. Therefore, a call is started between computer and fixed or mobile phones. In this type of VoIP, a call that starts from Internet needs to finish in PSTN network. For this , it is necessary to use a media gateway. This media gateway makes circuit-switched and packet-switched environments compatible. 3) PSTN to PSTN (Phone to Phone) A call starts from PSTN is transmitted over internet and again it ends on PSTN. For end users, it is not different from traditional call. This type application may be used for long distance voice transmission to decrease bandwidth cost. e. Circuit Switching and Packet Switching In order to understand VoIP better, it may be useful to remind circuit-switching and packet- switching briefly. Because VoIP uses packet-swtiching. 1) Circuit Switching To transmit information over network, we need a path. In circuit-switching, this path is decided before the data transmission starts. The system decides on which route to follow and transmission goes according to the path. For the whole length of the communication session between the two communicating bodies, the route is dedicated and exclusive and released only when the session terminates.
  • 5. 2) Packet Switching Internet Protocol breaks data into packets. Each packet contains information about the IP address of the source and destination along with the data load. Once packets reach their destination, the packets are reassembled to make up the original data again. To transmit data in packets, it has to be digital data. In packet switching, the packets are sent towards the destination irrespective of each other. Each packet has to find its own route to the destination. There is no predetermined path. Each packet finds its way using the information it carries, such as the source and destination IP addresses. f. Comparision of PSTN and VoIP • Traditional PSTN phone system uses circuit switching while VoIP uses packet switching. PSTN is old and expensive. VoIP is more modern. • There is circuit dedication for PSTN and cost is not shared between speakers. There is no circuit dedication for VoIP and cost is shared. • PSTN is more reliable than VoIP. Because in VoIP, no circuit dedication. Therefore, circuit is also open for other services. There is a big possibility of congestion and this may result in delays and even packet loss. 2. Codecs Codec stands for coder-decoder. A codec converts an audio signal into a compressed digital form for transmission and then back into an uncompressed audio signal for replay. This is the essence of VoIP. Codecs accomplish the conversion by sampling the audio signal several thousand times per second. For instance, a G.711 codec samples the audio 64.000 times a second. It converts each tiny sample into digitized data and compresses it for transmission. To summarize, a codec has 4 main function: To sample audio signals, to sort each tiny sample, to compress each tiny sample for transmission and to packetize audio data. 3. Soft Switches The codec works with the algorithm to convert and sort audio signals, but none of that is any good without knowing where to send the data. In VoIP, this task is handled by soft switches. As it is declared before, VoIP uses IP-based networks. The challenge with VoIP is that IP- based networks don't read phone numbers. They look for IP addresses. IP addresses
  • 6. correspond to a particular device on the network. This device may be a computer, a router, a switch, a gateway. For VoIP applications, this device may be a telephone. Another problem that makes matter worse is that IP addresses are not always static. Generally, A different IP address is assigned with each new connection. Therefore, the challenge with VoIP is to figure out a way to translate phone numbers to IP addresses and then to find out the current IP address of the requested number. This is called mapping process and is handled by a central call processor running a soft switch. The central call processor is a piece of hardware that runs a specialized database/mapping program called soft switch. Think of the user and the phone or computer associated with that user as one package. That package is called the endpoint. The soft switch connects endpoints. Soft switches know: • Where the endpoint is on the network • What phone number is associated with that endpoint • The current IP address assigned to that endpoint Therefore, when a call is placed using VoIP, a request is sent to the soft switch asking which endpoint is associated with the dialed phone number and what that endpoint's current IP address is. The soft switch contains a database of users and phone numbers. If it doesn't have the information it needs, it hands off the request downstream to other soft switches until it finds one that can answer the request. Once it finds the user, it locates the current IP address of the device associated with that user in a similar series of requests. It sends back all the relevant information to the IP phone which allows the exchange of data between the two endpoints. 4. Protocols Up to now, lots of devices worked together on the network to make VoIP possible. For instance, a phone acting as a user interface, client software working with a codec to handle the digital-to-analog conversion and soft switches mapping the calls needs to communicate in the same way. To communicate efficiently between completely different pieces of software and hardware, we need protocols. There are several protocols currently used for VoIP. These protocols define ways in which devices like codecs connect to each other and to the network using VoIP. They also include specifications for audio codecs.
  • 7. There are 2 most widely used protocols: a) H.323 H.323 is a comprehensive and very complex protocol that was originally designed for video conferencing. It provides specifications for real-time, interactive videoconferencing, data sharing and audio applications such as VoIP. In fact, H.323 is a large collection of individual protocols and specifications. Therefore, it allows to be used for so many applications. The problem with H.323 is that it is not specifically tailored to VoIP. b) SIP (Session Initiation Protocol ) SIP is an alternative to H.323. It is smaller and more flexible and efficient than H.323. SIP is specifically developed for initiating, building and terminating multimedia applications. Therefore, it is more suitable for VoIP. A challenge for VoIP is that there is no universal specific protocol for VoIP. Therefore, different protocols may not be always compatible with each other. This may result in conflict in VoIP applications. 5. Factors that Affect Call Quality in VoIP Real-time applications like phone call and video-conferencing over Internet are more privileged and more sensitive than other applications. For a better call quality in VoIP, some factors must be taken into consideration. There are 3 main factors that affect call quality: a) Packet Loss: For real-time applications, retransmission of loss packet is impossible. Because waiting packet again results in very serious latency. Because of the fact that human ear is not too sensitive, limited packet loss is accepted. However, this limit should be in an acceptable value. b) Latency: To send packets from source to destination takes a period of time. This period of time is related to connection type between source and destination. Increase in latency means that talkings between end-users reach very late. This takes too much time. To minimize latency, routers must provide priority for VoIP packets. This means that when a VoIP packet comes to router, router should postpone other packets and send VoIP packet immediately. Codecs may also result in latency. However, to packet voice data, this latency is necessary.
  • 8. c) Jitter: Jitter means that variation in delay times of packets. Connection state and density of two endpoints vary continuously over Internet. Therefore, packet delay time between destination and source may change from one packet to another. This means that previous packet may reach later than following packet. This results in confusion. So, packet delays must remain at certain levels. Jitter is an important parameter that decreases call quality. To prevent jitter, RTP (Real-time Transport Protocol) should be used. 6. RTP (Real-time Transport Protocol) RTP is an important protocol for VoIP. RTP has 2 main functions that is necessary for VoIP: 1) To prevent jitter is one of the responsibilities of RTP. RTP calculates average delay time of packets. Then it postpones packets for a period of time that comes early. By doing this, RTP tries to equalize delay time of packets. Therefore, it prevents possible confusion. 2) To arrange packets in the right order is another responsibility of RTP. RTP adds its own header to packets. This header includes the order number of packets. Therefore, packets are put into order. 7. VoIP Packet Headers For real-time applications, data needs to reach destination immediately. Real time applications have no tolerance to error check and retransmission of loss packet mechanisms. Therefore, UDP is preferred in VoIP applications. Another reason to prefer UDP to TCP is that UDP header info is smaller than TCP header info. VoIP packets are smaller than like FTP or e-mail packets. For small packets, the size of packet header is important. Because header size affects the transmission efficiency. However, the use of UDP for voice transmission has also some disadvantages. Because UDP does not put order number to the packets. Therefore, destination can’t know how packets will be sorted. This results in voice confusion in destination part. To prevent this, RTP is used as it is explained above. Now, there are two header info, UDP and RTP. But again total size of UDP and RTP headers are smaller than TCP. Therefore, a VoIP packet includes IP Header (20 bytes), UDP Header (8 bytes), RTP Header (12 bytes) and digitized audio data (20 to 150 bytes)
  • 9. 8. Advantages of VoIP • VoIP uses less bandwidth when it is compared to PSTN. Therefore, it decreases the cost of operators. Moreover, the internet users can also open other applications while using VoIP because of the less bandwidth. • VoIP applications also offer low or no cost for customers. Only a headphone and microphone is enough for a basic VoIP application. • VoIP is more efficient than PSTN. Because average %30 of phone talks does not include audio signals. We know that if there is no audio signal then there is no data packet in VoIP. When there is no packet, there is no data transfer and network is closed. This means that %30 less bandwidth is used in VoIP compared to PSTN. • In VoIP, the cost is shared between endpoints. 9. Disadvantages of VoIP • VoIP is less reliable than PSTN. Because latency, jitter and loss packet factors that are seen in VoIP applictions make VoIP less reliable. • VoIP is dependent on wall power. This means that when electricity goes off, we can’t use VoIP. However, there is no such a problem on PSTN. • VoIP has no integration to other systems like digital video recorders, digital tv home security systems. • VoIP applications can be attacked by worms and viruses, nd can be exposed to hacking. • There is no universal and specific VoIP standards. This may result in conflict between systems using different standards.

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