Đề tài Voice over Internet Protocol

Ha Noi open university Center For International training Co-operation Thesis: Teacher : Nguyễn Thái Nguyên Group 3 : Đồng Xuân Thắng -Cap Lê Trọng Nghĩa Nguyễn Xuân Tư Mai Trọng Dũng Bùi Thanh Nhàn Ngô Thị Nhàn Hà Nội ngày 15/1/2003 Glossary ATM : Asynchronous Trasfer mode ACELP : Algebraic Code Excited Linear Predictive ARQ : Automatic Rrepeat Request ACF : Admission Confirm DES : Data Encryption Stadard PSTN : Public Switched Telephone Network PC : Personal Computer PCM : Pulse Code Modulation IP : Internet Protocol ITU : International telecommunication Union IETF : Internet Engineering Task Force ISUP : ISDN User Part INAP : Intelligent Network Application Part ITSP : Internet Telephony Service Provider MAP : Mobile Application Part MGCP : Multimedia Gateway Control Protocol MTP : Message Trasfer Part MP : Multi point MCU : Media Control Unit OLC : Open Logical Channel QoS : Quality of Service RC : Report Court RSVP : Resource Reservation Protocol RTCM : Real Time Control Mode RTP: Real Time Post SIP : Session Initiation Protocol SS7 : Signal No.7 SCCP : Signaling Connection Control Part STP : Signaling Transfer Point TCP : Transmission Control Protocol TCAP: Transaction Capabilities Application Part UDP : User Data Package VAD : Voice Activity Detector VoIP : Voice over Internet Protocol General of the thesis VoIP -Voice over Internet protocol VoIP ( Voice over IP- that is, vioce delivered using the Internet Protocol) is a term used in IP telephony for a set of faccilities for managimg the delivery of voice information using the Internet Protocol(IP). In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit – committed protocols of the public switched telephone network (PSTN). A major advantage of VoIP and Internet telephony is that it avoids the tolls charged by ordinary telephone service. VoIP, now used somewhat generally, derives from the VoIP Forum, an effort by major equipment providers, including Cisco, Vocltec, 3 Com, and Netspeak to promotethe use of ITU-T H.323, the standard for sending voice (audio) and video using IP on the public Internet and within anintranet. The Forum also promotes the user of directory service standard so that user can locate other users and the use of touch-tone signals for automatic call distribution and voice mail. In addition to IP, VoIP uses the real-time protocol (RTP) to help ensure that packets get delivered in a timely way. Using public networks,it is currently difficult to guarantee Quality of Service (QoS). Better service is possible with private network managed by an enterprise or by an Internet telephony service provider (ITSP). A technique used by at least one equipment manufacturer, Netspeak, to help ensure faster packet delivery is to Packet Internet or Inter- Network Groper (Ping) all possible network gateway computeres that have access to the public network and choose the fastest path before establishing a Transmission Control Protocol (TCP) sockets connection with the other end. Using VoIP, an enterprise positions a “VoIP device” (such as Cisco’s AS5300 access server with the VoIP feature) at a gateway. The gateway receiver packetixed voice tranmissions from users within the company and then routes them to othe parts of its intranet (local area or wide area netnork) or using a T- carrier system or E-carrier interface, sends them over the public switched telephone network (PSTN) Chapter1:Voice over IP (VoIP) Technology 1. Fundamental features of channel switching network and Internet: 1.1. Fundamental features of channel switching network: The channel switching network is designed for rapid connect and eliminating the ineffectiveness of time-consume on connecting. In the channel shifting network, the user is provided a conductive channel to exchange information together. When the exchange completed, the conductive channel is released. This could lead to loss because of limits of conductive channel. The utility is low but ensures the calling quality because a two-way 64 kbps channel is set aside for caller and receiver. The channel shifting network is designed optimum for real transmission time with high service quality. In the channel switching network, all terminal equipment and switch board are inserted a fixed number so no need to enter address for information exchanging process. The switching system in channel switching network will base on the address of called subscriber to define the conductive line. Because the band width is ensured not be changed during calling, calling fee of channel switching network is based on distance and calling time. 1.2. Fundamental features of Internet: Internet is the package switching network suitable with applications that are not exchanged according to the real time; Package delay doesn’t effect strongly on service quality like email and file transmission. Package switching networks don’t set aside a fixed line between two users, so, not ensure the service quality. All information on the network are divided into packages, these packages contain the destination address and its order. Channel fixer and host on the network will send these packages to the targeted address. On Internet, all packages are treated the same with out distinguishing their contents. When packages to the destination address, they will be arranged according to the initial number. By form of package information transmission, the utility is maximum. However, real time applications will be greatly effected on service quality. The fee is not calculated on distance or time but on used band width. On Internet, on address of package is marked by IP address, the IP address will be named for the host and terminal stations. Channel fixing will be controlled by the IP destination address. To create a understandable, convenient address type for the IP address by name like service of regional name or email address. Because the limit of IP address, the users are temporarily inserted IP while dialing. The IP address is only for one terminal equipment while connecting Internet and deleted while not connecting. The deleted IP address will be used for another connecting on the network. 1.3. Advantages of VOIP against PSTN: The users will pay for used time of PSTN if more time for call establishment, more increased fee to be paid. At one time, they can contact to one person. But with VoIP, the time for call establishment is independent to subscriber’s fee. One subscriber could have calls to different ones and exchange data, dialogue, pictures, paintings and video with other subscribers. Figure 1: The basic structure of telephone network by IP 1.4. Outlook of VoIP technology: + Some technical features of IP telephone: By analysis of fundamental features of channel switching network and Internet, we see that it is typical to accumulate real time signal into the package switching network and IP telephone. Firstly, we should classify IP telephones. All IP telephones change according to 3 characters: type of terminal equipment, position of gateway, between IP and PSTN networks and main transmission equipment. a. Terminal equipment and gateway: There are 03 main types of IP. They are PC to PC, PC to Phone, Phone to Phone. + PC to PC is the first model of IP telephone. Users at two ends of PC to PC should have 1 PC that is equipped audio, a software and connected to Internet. This service no need gateway and PCTN because PCTN never switch these calls, the main transmission tool is public Internet. Due to sound quality and complexity of use, the PC to PC has a litter affect on traditional telephone service. + PC to Phone expands the number of users but for exploiters, the call of PC to Phone is more complex than that of PC to PC. + Phone to Phone is very important market including mainly commercial services, because, people prefer to communicate by phones. However, the 3rd model of IP requires more investment capital because it needs input gateway to PSTN near places providing service. Services of Phone to Phone are nearly similar to that of traditional telephones. b. Transmission equipment: The classification between IP and VoIP telephone is based on the nature of main transmission equipment. IP telephone is for voice transmission, fax and services relating to package switching networks on IP. Internet phone and VoIP are basic types of IP. Internet phone is IP in which the main transmission network is public Internet (global super-network). Voice over IP is IP in which the main transmission network is private-used one basing on IP. Besides, being the replacing tools for distance and international phone, the IP technology creates a plenty of other services that can transmit every service by IP. This part only mentions the technology of VoIP and interests in the terminal equipment that is telephone on the channel switching network (Phone to Phone). Figure 2: IP call: Phone to Phone + Special features of VoIP: a. Adjustable quality: The quality of VoIP depends on each part (coding and low speed re-coding for each part). Internet is not specific service network, the exchanging methods are entirely selected by terminal systems. Thus, the terminal systems can control the compressed volume on the network bandwidth or content for transmission. b. Security: Using SIP to order a password and confirm messages indicating the terminal. RIP make and the password to be the password of transmission method. Therefore, all program is coded to secure transmission. c. Users interface: Terminal systems of VoIP have plentiful indications and can give out instructions and various graphic interface. d. Connecting telephone and computer: Available to solve these complex connections. 1.5. Conclusion: The VoIP technology has potential for future development, ability to replace the existing PSTN network. Due to differences in features of channel switching network and Internet, to apply VoIP for users of channel switching network (Phone to Phone), these differences should be solved. Concretely, there should be address changes, indication of two networks and proper inter-code for application of time transfer on network. 2. Problems relating to VoIP technology and talk quality on VoIP: Using the traditional channel switching telephone network will cost much when at distance, to reduce expenses for distant calls, use public data network or private data network for communication. The package switching network that applies IP is example. Using the package switching network by IP to transmit the talking signals. Voice over IP-VoIP is good basis to design global multi-instrument transmission system that can replace the infrastructure of existing network. Accumulating Audio, Video, data, fax... into a single common network on IP technology. It is possible to apply the Frame relay or unsynchronous transmission technology ATM to replace IP technology. The VoIP is more economic for distant call, because the fee is calculated by the width of bandwidth, not by distance. In IP, it uses talk compressing technology to save band width leading to cost reduction but the IP’s quality not as good as that of PSTN. The biggest difference when applying into the multi-instrument network is actual time service non-actual one. With actual time service and like Audio, Video... not allow over-delay on the network; in non-time network like email, file transmission, the delay is not worthy worrying. So, to carry out VoIP, special compressing and coding methods should be used to reduce the speed of talk signals that can’t be use 64 tps like channel switching. 2.1. Coding techniques and talk signal compression: In talk transmission, voice is usually numberidized and coded PCM by Rule A or U with speed of 64 Kps recorving sound rather actual. For some specific applications such as transmitting talk signals on TP network, sounds are transmitted with lower speed, so, there should have coding techniques and talk signal compression to lower speed according to standard of ITU and ETSI like G723.1; G729; G729A; GSM. + Standard G7213. According to the standard of ITU, the coding has 5.3Kbps and 6.3Kbps. The compression technicque uses MP-MLQ for high bit speed; for coding with low bit speed using ACELP. Delaying against algorithm is 67.5ms. + Standard G.729. According to the ITU standard, this coding has speed of 8 Kbps. This compression techniques uses algorithm predicting coded linear linked structure algebra excitation. Delaying against algorithm is 25ms. + Standard GSM06.10. According to ETSI, this code has 13Kbps. This compression technique is regular pulse excitation and long-term predictor. Delaying against algorithm is 40ms. 2.2. Voice Activity Detector (VAD): VAD is carried out by numeric signal processor to reduce the talk intensity that is transmitted by automatically detecting the dead space on the talk and stopping transmitting at that time. There are space approx. 50-60% of almost talks. This always occurs because when one speaking, the other must listen to. VAD allows band width for dead space saved for reserving other data. VAD actives by controlling power of talk signals; power change is change of talk signal frequency. The difficult of VAD is to define the exact time of talk ending and of talk signal. The double VAD is nearly 200ms after recognizing talk signals and stop and detect package processing. This top prevent VAD from missing the end talk or in the middle of small interrupt in talks. 2.3. Number and address: Due to cooperation between IP and SCN networks, there will be 2 types of address: address in CSN and in IP. a. Numbering on SCN network: On the channel switching network, all terminal and switchboard are fixed a number. Number E164 is telephone numbers subject to the structure and numbering program that were described on the proposal E164 by International Telecommunication Union. The line fixing process on the channel switching network is controlled by the address system of E164. Before dialing, the users of channel switching network have to dial E164 and callee’s number. + Local number: Code of Access Caller + National Post + National Destination Code + + Subscriber number. + For international numbers, we can use 03 following structures: Code of Access Caller + International Post + Country Code + Identification Code + Subscriber number. Code of Access Caller + International Post + Country Code + Destination Code + Subscriber. Code of Access Caller + International Post + Country Code + Global Subscriber’s Number. b. Numbering on IP: + Prefix is an identifier including one or more numbers allowing the used numerical types, network and service and can be used to select service provider, type of service in a nation. + Selecting service provider including numbers that allow to select service by IP network or SCN and there of to select appropriate switching. + Selecting service provider can be done by ways: pre-select by user or dialing, password. Incase, the Gateway connects to SCN where there are a lot of service providers, both Gateway and Gatekeeper should be able to identify and process the selected code of service provider. Incase, a lot of service providers on IP network, Gatekeeper is able to identify and process the selected code of service provider. To get the most common address types on Internet, it can use name address like email address: user@domain, user@host, user@IP-address, phone-number@gateway. 2.4. Fee: To ensure the effectiveness of network, the fee calculating will be done by a separate host system. The fee-calculator host will be responsible for collecting and reserving all detail of call from gateway or MGC. These data are used to make invoices for customers. Customers ca access into the host for their fee details on the website. The fee will be calculated by the used time. The fee calculating system should be able to calculate on 2 types of service: pre-paid and post-paid. This software must be able to carry out some following function. - Accepting call. - Informing the amount of account. - Fee calculating based on pre-fixed level for different directions. - Informing the maximum time of call. - Updating account’s amount after calling. 2.5. Signal cooperation: The standard of signal communication of IP Phone to PSTN is suggested to be signal No. 7 (SS7). The SS7 is used to transmit following information: - Information o call establishment. - Information about call control. - Property and application. The signal communication between 2 IP networks and signal network 7 of PSTN is carried out by signal Gateway. The signal gateway connects to STP on the SS7 as a SP and transfer signals fully. The signal Gateway should support signal news ISUP and SCCP/TCAP. Using the signal communication No. 7, IP telephone network will bring benefits as follows: - Fully connecting to PSTN. - Supplying additional services. - Improving call control. - Improving maintaining property for trunk. - Speeding up call establishment. Although new signaling, ssuch as H.323 ans SIP, exist for VoIP net works the standard in traditional telephony and in mobile networks is SS7. Therfore, if a VoIP based network is to communicate with any traditions network, not only must it network at the media level through media gateways, it must also interwork with SS7. To support this, the IETF has developed a set of protocols known as Sigtran. In order to understand Sigtran, it is worth considering the type of inter working that needed to occur. Imagine, for example, an MGC that control one or more media gatways. The MGC is a call control entity in the network and, such as uses call control signaling to and from other call control entities. If other call control entities use SS7 then the MGC must use SS7 at least to the extent that the other call control entities can communicate freely with it. This means that the MGC does not necessarily need to support the whole SS7- just the necessary application protocols. Consider figure 3 which shows the SS7 stack. The bottom three layer are called the Message Transfer Part (MTP). This is set of protocols responsible for getting a particular SS7 message from the source signaling point to the destination signaling point. Above the MTP we find either the Signaling Connection Control Part (SCCP) or the ISDN User Part (ISUP). ISUP is generally used for the establishment of regular phone calls. SCCP can also be used in the establishment of regular phone calls but it is more often used for the transport of higher layer applications, such as the GMS Mobile Application Part (MAP) or the Intelligent Network Application Part (INAP). In fact most such application use the services of the Transaction Capabilities Application Part (TCAP) which in turn uses the services of SCCP. Application Part ISDN User Part (ISUP) Transaction Capabilities Application Part (TCAP) Signaling Connection Control Part (SCCP) MTP Level 3 MTP Level 2 MTP Level 1 Figure 3: SS7 Stack SCCP provides an enhanced addressing mechanism to enable signaling between entities even when those entities do not know each other’s signaling addresses (known as point codes). This addressing is known as global title addressing. Basically it is a means wherby some other address, such as a telephone number, can be mapped to a point code, either at the node that initiated the message or some other node between the originator and destination of the message Figure 3 provides some examples of communication between different SS7 entities. Consider scenario A. In this case, the two entities, represented by point code 1 and point code, communicate at layer 1. At each layer, a peer to peer relationship exists between the two entities. Scenario B has a