Internet Telephony
Abstract Page
Internet telephony uses the internet as the connection medium for transmitting voice calls and other telephony services used in communication like SMS, fax, and other voice messaging applications. It is an economical and powerful communication option in which telephone networks and data networks are combined. As the Internet protocol network can carry traditional telephone traffic, it offers both opportunities and challenges to all the telephone service companies. Since the traditional circuit-switched network works as a complex web of a large number of interrelated technologies that have evolved over more than 15 years, just the transmission technology replacement is required in order to replace it with Internet telephony.
The digital telecommunication services offered by Internet telephony are based on Voice over Internet protocol. Other multimedia and signaling protocols used are Session Initiation protocol (SIP), the Media Gateway Control Protocol (MGCP), and the H.323 protocol. The present study aims to understand the concept of Internet telephony and various telephony technologies that are available.
Internet telephony refers to using computer networks for making telephone calls whereas the Internet protocol standard is used for the transmission of the data. In the internet telephony, the analog voice signals are converted into digital format and then the translation and compression of the signal into the Internet protocol packets takes place so that they can be transmitted over the Internet and the same process is reversed at the reversing end. Session Initiation protocol is widely used communication protocol for internet telephony which is used for providing voice and video calls, instant messaging etc. It is used for signaling and controlling multimedia sessions.
Internet Telephony Hardware
Appropriate hardware is needed to use Internet telephony. There are four different alternatives available. Standard computers can be used with microphones, and loudspeaker or headphones by the participants involved in the conversation. Special software can also be installed on the PC. Secondly, specific VoIP end devices can be used like SIP or IP telephones which differ only in terms of technology from the traditional telephones. Thirdly, traditional telephones with the special adapter connected to them can be used; the adapter helps in conversion of analog telephone signals into digital data. Lastly, it is also possible to use any mobile phone by using an FMC client to connect through the telephone system. In last three options, there is an advantage, that the devices can be used in the same way as the traditional telephone and even when the PC is switched off, the user is still attainable.
The most common protocol used in Internet telephony these days is SIP (Session Initiation Protocol). Though, MGCP and H.323 are also used. H.323 is one of the earliest VoIP protocols, but its use is declining these days. Following is the detail of various protocols used for VoIP calls or internet telephony:
SIP (Session Initiation Protocol): SIP protocol runs on the User Datagram protocol (UDP), Transmission control protocol (TCP) or the Stream control transmission protocol (SCTP). It can be used for both multi-party (multicast) or two-party (unicast) sessions. The SIP protocol makes use of the request//response transaction model where client request invokes any particular function or method on the server and the corresponding response is generated. Each and every resource in the SIP network has a URI i.e. Uniform Resource Identifier. It is basically a client-server protocol, where the client initiates the session and the server function is performed by the call recipient. The video streaming and voice communication in SIP protocol are carried over RTP (Real-time Transport Protocol). SIP packet body contains the Session Description protocol containing the information of the parameters (like codec, port numbers, transmission protocol etc) for the defined media streams.
MGCP (Media Gateway Control protocol): It is the call control and signaling communication protocol that is used in Voice over Internet protocol systems (VoIP). The media gateways on Internet protocol network that is connected to Public switched telephone network (PSTN) is controlled by implementing the architecture used in Media Gateway Control protocol. It is basically a Master/slave protocol in which a Call Agent takes control of the call and port on the Media Gateway. The system consists of a Call Agent, one signaling gateway which is connected to the PSTN and the Media gateway that performs media conversion between packet and circuit switched Network.
In MGCP protocol, the call agent tells the MGW what all events should be reported to the call agent, the interconnection of the end points and the what all signals to be activated. The Media Gateway, on the other hand, reports the events such as dialed digits, telephone off-hook, on-hook etc. to the call Agent. The SIGTRAN protocol is used for signaling between the Call Agent and the SGW i.e. signaling gateway.
H.323 protocol: It is used for audio-visual communication sessions on the packet network. It is best suited for transmitting calls over networks where a mixture of PSTN, ISDN, and IP network exists. The IP telephony can be introduced into the existing network of ISDN- PBX based systems when transitions to IP-based PBXs takes place. With the help of H.323 protocol, an IP-based PBX acts as any call control element or a gatekeeper that can provide basic services to videophones or IP telephones. It also helps in providing supplementary services like call holding, call forwarding, call transferring etc.
PBX (Private Branch Exchange): Within an enterprise, a telephone system can be established that helps in switching calls between enterprise users on local lines while a certain number of external phone lines can be shared between all the users. It originally used Analog technology. Now, PBX can use Digital technology as well where digital signals are converted to Analog signals for outside calls making use of the Plain Old Telephone Service (POTS). The main constituents of the PBX are:
A computer that manages the calls switching within the PBX and in and out of it.
Trunk lines from telephones that are terminated on PBX.
A switchboard for human operations which is optional
The lines network within the PBX.
Architecture of IP Telephony
IP telephony network has to interwork with older technologies also PSTN, ISDN. Thus, there can be three classes of IP telephony operation, which depends on the number of IP and traditional telephones. Following can be the architecture of the IP telephony:
How VoIP works: In the first architecture, the direct IP-based connection is established end-to-end between the callee and the caller. The analog signals are converted into digital signals and divided into data packages which after labeling with headers are sent over the internet connection. Session Initiation Protocol address is used for connection establishment. The request is logged on to the SIP server which then passes it to the end device and a telephone conversation is established. Since the user is connected to the internet by the end device, so calls can be attained anywhere in the world.
In the second architecture, both the callee and the caller uses circuit-switched telephone networks. The VoIP calls takes place through Gateways, the caller dials into the gateway and then through the public/private internet, the connection is established to the gateway near to the caller through SIP protocol.
Another architecture is where one end user has an IP-based phone and another user has PSTN phone. In this architecture, an IP-based user uses SIP protocol to place/receive call whereas PSTN phone user uses Gateways to connect to the internet world and to the end user subsequently. IP PBX has closely related architecture where within an enterprise the phones are connected to a gateway for providing PSTN dial tone.
The architecture of the Internet telephony can be represented as shown in figure 1 below.
Figure 1 End-to-End IP Telephony
Real Life Example for SIP call/session
Peter is a die-hard cricket fan, he managed to get tickets to watch the match at the stadium. While he was watching the match, he thought of making his friend envious. Through his mobile phone, he places a call to his friend John. John is at his desk in the office and receives a pop-up on his PC screen, indicating that Peter is calling. He picks up the call and starts talking. While talking, Peter also started the video-sharing application from the field. John receives the incoming video request. The PC client starts showing the game and John watches the scores of both the teams. After the match, John ends the video-stream. Thus, all the required communication took place through IP connectivity. Thus, IP telephony offers the best-suited communication media, media can be changed during the conversation, and the SIP-capable communication devices are used for IP access.
Telephone Number Mapping
In order to communicate via Internet telephony, the telephone numbers should be mapped to SIP and other URIs. The VoIP gateway may have to reach a terminal identified by the telephone number, so mapping between telephone numbers and URIs is essential. This is provided through ENUM services. A new DNS record type called NAPTR helps in the conversion of the telephone numbers to the DNS name.
Conclusion
Internet telephony is one of the most powerful and economical options for communication which is gaining so much popularity these days with the advancement in technology. The transition from circuit-switched to packet switched telephony is going on slowly, but after third-generation mobile networks, the majority of the voice calls is becoming packet-based. This transition allows the user to address many of the limitations of the conventional telephone system, thus, empowering end user to customize their own services.
References
Beasley, J. S., & Nilkaew, P. (2012). Networking Essentials. New York: Pearson Education.
Boucadair, M., Borges, I., Neves, P. M., & Einarsson, O. P. (2012). IP Telephony Interconnection Reference: Challenges, Models, and Engineering. Boca Raton: CRC Press.
Flanagan, W. A. (2012). VoIP and Unified Communications: Internet Telephony and the Future Voice Network. New Jersey: John Wiley & Sons.
Schulzrinne, H. (2004). Internet Telephony. Practical Handbook of Internet Computing, pp. 1-32.
