W4_JTO_Ph2_Datacom_IT-part-12.pdf

JTO Phase-II Data Network & IT VOIP 12 VOIP 12.1 Learning Objective This chapter gives you an introduction on Voice Over IP or VOIP in short. The document covers a short introduction to...

JTO Phase-II Data Network & IT VOIP 12 VOIP 12.1 Learning Objective This chapter gives you an introduction on Voice Over IP or VOIP in short. The document covers a short introduction to what is VOIP services and what are the basic components in the VOIP network and how it works. 12.2 Introduction Voice over IP is a collection of technologies and protocols that define the basic rules for communication through the Internet Protocol. VoIP works by encoding voice information into a digital format, which can be carried across IP networks in discrete packets. The process of VoIP communication basically consists of voice digitization, connection establishment, data transmission and voice decoding at the remote end. These processes use a lot of hardware and software equipments. The process itself is basically similar to the traditional telephone communication but has some differences from it too. 12.3 VOIP Architecture Figure below shows the basic architecture for VoIP technology. The parts of the VoIP system will be introduced in the following sections. Figure 62: VOIP Architecture In the ordinary telephone network the connection between two telephones was established by a switchboard. In the VoIP model this role is dedicated to the soft switch. It is a software solution that controls connection between the end-points where there is no end to end VoIP connection is not available. The soft switch can be divided into two parts, the media gateway and the call agent. The call agent is mainly responsible for call JTO Phase II (DNIT) Version 1.0 Sep 2021 Page 160 of 174 For Restricted Circulation JTO Phase-II Data Network & IT VOIP routing, network signaling and billing. The media gateway is used to connect different physical networks in order to provide end-to-end connectivity. It functions very similarly to a typical network switch in that it can create a heterogeneous link between endpoints, regardless of the network media in between. In addition, the Media Gateway can also connect a VoIP circuit to a PSTN circuit, allowing the use of VoIP even when only one of the endpoints is VoIP enabled. The VoIP telephone sets can be divided into two classes, softphones and hard phones. Hard phones are the traditional telephone devices that are capable for VoIP communication. If the soft switch is the switchboard for VoIP the softphone is the telephone solution originally dedicated to VoIP communication. It is a software with all the telephone functions that can run on an ordinary personal computer and uses the main hardware devices (sound card, microphone, speaker) for realizing the communication. 12.3.1 Analog Terminal Adapters (ATA) Analog Terminal Adapters (ATA) perform analog and digital conversions between a traditional analog phone and the broadband modem, allowing users to use the VoIP service with an existing phone. The ATA functions much as a Media Gateway does, translating data between analog and digital communication. ATAs typically have two ports: a telephone jack (FXS port) and a LAN port. This enables a user to plug into the FXS port an analog telephone, and then attach the ATA to the network with an UTP cable. This way the analog telephones will be available for VoIP calls too. The voice digitization, compression or conversion back to analog data need the implementation of certain coder-decoder solutions called CODECs. These codecs are defined in standards that have to be supported by the VoIP communication entities like softphones, hardphones, ATAs in order to be able to communicate. 12.3.2 Core ip network The core IP network is a segment of the Internet (or is special cases the Internet itself) that guarantees a quality of service when transporting a stream of IP packet between voice applications. The core IP network basically is the MPLS core which enables the routing of packets on to the network. 12.3.3 signaling gateway The signaling gateway is responsible for notifying a VoIP end point that another end point requests for communication. This process is called signaling. The main goal of the signaling gateway is to implement the difference between the VoIP signaling and the PSTN notifying system in order to provide the communication between VoIP and non- VoIP networks. 12.3.4 VoIP trunk gateway A VoIP trunk gateway is an interface that facilitates the use of plain old telephone service (POTS) equipment, such as conventional phone sets and fax machines, with a voice over IP network. The VoIP trunk gateway is basically used for connecting the Private Branch Exchange (PBX) to the PSTN. The VoIP solutions are connected to a PBX that is a software or hardware that establishes the connection between the end points using a SIP account. JTO Phase II (DNIT) Version 1.0 Sep 2021 Page 161 of 174 For Restricted Circulation JTO Phase-II Data Network & IT VOIP 12.4 sip protocol The Session Initiation Protocol (SIP) was originally designed as a means of notifying or inviting users to Internet multicast and broadcast sessions. It provides control over multimedia sessions. Implemented at the application layer, it is capable of establishing, modifying, and terminating sessions. In the context of IP telephony, these sessions are the VoIP ―calls‖ themselves, and SIP is used to place the calls, modify them in-session (for example, inviting other users for features such as three-way calling), and to hang up. Also integrated into SIP‘s design is mobile capability, because SIP handles name mapping and redirection servers. This allows users to use IP telephony without regard to their physical or network location. 12.4.1 Session Initiation Protocol The primary SIP functions are the following: User location and name translation - this ensures that the caller reaches the called end point User availability - the presence information about the user, indicates if the user is willing to engage in communication Users capabilities - This allows the group on the call to agree on the different features supported. If a certain CODEC rate is not supported by SIP, there is room for negotiation. Session setup - it is used for communication session establishment by using session parameters Session management - it is used for modifying session parameters and invoking services The SIP communication is made via SIP requests and responses. The following figure contain these with some basic explanation about them Figure 63: SIP request The SIP requests are sent to the PBX from the VoIP clients and the PBX sends back some SIP responses in return or vice versa. The connection between two clients is established by these SIP request-response collections and after the connection is established the PBX gets out of the line and the two clients can communicate directly. JTO Phase II (DNIT) Version 1.0 Sep 2021 Page 162 of 174 For Restricted Circulation JTO Phase-II Data Network & IT VOIP The following table contains the SIP response categories that can be sent between a VoIP client and the PBX. Figure 64: SIP Response 12.5 SIP Communication flow Figure below shows an average communication flow between two VoIP clients from the registration to the PBX, through the connection establishment and the actual communication till the end of the communication Figure 65: SIP Call Flow The figure shows the whole VoIP communication process and the related SIP messages. The first step of the communication must be the registration to the PBX. This is shown by the red arrows in the figure. The registration is mainly a JTO Phase II (DNIT) Version 1.0 Sep 2021 Page 163 of 174 For Restricted Circulation JTO Phase-II Data Network & IT VOIP REGISTER request and an OK response pair. In some cases the registration can consist of more messages when the clients have to possess a SIP account for authentication. In those cases the PBX sends back the SIP registration data and the client has to send another REGISTER request with the provided SIP account. A successful registration is always completed by the PBX sending an OK 200 message to the client. When both clients are registered to the PBX they can request for communication line establishment. This is shown in the figure by the green arrows. The communication start request is an INVITE message that is sent from a client to the PBX. The PBX sends back a synchronous TRYING message to the client and sends the INVITE message to the other client. The called client sends back a TRYING response and, if the connection can be established, it also sends back a RING message that is transferred from the PBX to the caller client. When the called client accepts the call an OK message is sent to the PBX, and it also sends an OK response to the caller client. At this point the connection between the two clients is established and the PBX gets out the communication. The communication itself is made directly between the two VoIP clients and it is shown in the figure by the black arrow. When one of the clients want to end the call a BYE SIP request is sent from it to the PBX and it transfers it to the other end point. The other client sends back an ACK message that is sent to the first client too. This ACK notifies the client that the remote party acknowledged the end call request. After this the communication line is closed and both clients stop the software tools and hardware devices that were in use during the communication. The SIP messages for ending the communication is shown by the magenta arrows in the figure above. 12.6 Conclusion With the adoption of IP based networks and migration of legacy networks to an all IP network, VOIP is increasingly becoming a popular service. All the voice services in today‘s network are VOIP based service in PSTN or PLMN network. JTO Phase II (DNIT) Version 1.0 Sep 2021 Page 164 of 174 For Restricted Circulation