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Free Research Paper on VOIP

Free research paper example on VoIP:

Simply put, voice over IP (VoIP) technology, or IP telephony, as it is often called, is a system for transmitting telephone calls over data networks, such as the ones that make up the Internet. While VoIP technology is set to revolutionize communications, and is already being used by a number of traditional telephone companies to connect their regional offices, on a smaller scale it can also be a useful solution for businesses looking to trim their telephone expenses. The advantages of using VoIP technology are simple: its use can result in huge savings on the amount of physical and resources required to communicate by voice over long distances. It does so by working around circuit switching architecture, one of the fundamental drawbacks of traditional telephone networks. Traditional circuit switching-based telephone networks operate by opening a circuit between two points, identified by their telephone numbers. This circuit remains open, and transferring at its full capacity for the duration of the call, until somebody disconnects it by hanging up. Much of this capacity is wasted during a normal telephone conversation, because while the line is working at full capacity, not all of each user’s time is spent transferring data, or talking. Normal telephone users, of course, spend much of their time listening, or receiving data. Furthermore, during the course of a normal phone call, there is often dead air. All of these things are wasted capacity. Data networks operate in an entirely different way. They communicate through packet switching, a much more efficient scheme for exchanging data. Instead of keeping a circuit open constantly, they send and receive data only as needed, a bit at a time, in data packets. By doing so, packet switching-based data networks free up network resources, as well as the resources of the computers sending and receiving information. VoIP technology uses packet switching to minimize the amount of resources used in a telephone connection by exchanging the information in packets over a data network.

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This allows several phone calls to use the space that just one call would have occupied in a circuit-switched network. In the case of an office, telephones might be connected to a private branch exchange (PBX), a device designed to connect a number of phones or extensions to an outside line. Using a gateway, a device used to translate the standard circuit-switched signal generated by the telephones into digital information that can be sent over the data network. This signal is usually an IP signal, the standard protocol used by most data networks. One of the advantages of using packet-switched networks to carry telephone communication is that the infrastructure is already largely in place in the form of the many data networks that make up the Internet, and that infrastructure is already understands the technology. There are two major protocols used by VoIP technology to allow telephones, computers and other devices on the data network to communicate with each other: H.323 and SIP. The H.323 standard, a suite of protocols created by the International Telecommunications Union is a very wide-ranging and very complicated protocol, providing specifications for a range of communication including video conferencing, data sharing and VoIP, and it can be complicated to set up. The Session Initiation Protocol (SIP) emerged after H.323 as an alternative, guided by the Internet Engineering Task Force. SIP is a much simpler, more streamlined protocol developed specifically for VoIP use, and designed to employ other protocols in handling parts of the communication process. Using either gateway devices, or software applications on computers, VoIP technology can allow users to communicate by voice from computer to computer, computer to telephone, telephone to computer or telephone to telephone. Businesses can take advantage of the technology by using it to route voice data over their existing data networks, or by purchasing VoIP services from IP service providers. VoIP technology is growing in acceptance, and it seems inevitable that the cheaper, more efficient technology will play an important role in the world’s telephone communications. But it can also mean immediate cost savings and improvement in efficiency for businesses that chose to implement it now.

VoIP protocols
Two fundamental sets of protocols are required to transport voice over an IP data network namely signaling and transport protocols. Let’s look at signaling protocols first.

Signalling protocols
Signalling is required to manage call setup and routing as well as call supervision and tear down. The ITU H.323 set of recommendations is the most mature set of signalling standards for packet-based multimedia networks. Although H.323 is primarily associated with VoIP, it is independent of the network and transport layer protocols over which it runs. The Session Initiation Protocol (SIP) provides similar functionality to H.323, but it is still relatively new and less well developed. Consequently it has yet to be deployed to the same level as H.323 and for that reason I will confine the current discussion to a description of H.323. SIP however is a promising protocol that is under intense development and I will be writing a future article discussing its operation and what advantages it can offer over H.323.

H.323 network elements
There are four different functional entities on a H323 network:

– Terminal – Gatekeeper – Gateway – Multipoint Control Unit (MCU)

The H.323 Terminal
The H.323 terminal may entail an ethernet phone, or a PC running an H.323-based application such as Microsoft NetMeeting. It must support at least one audio CODEC (G.711 for uncompressed 64k voice) and optionally one or more compressed voice CODECs. The function of the CODEC is to digitize the audio signal from the microphone to prepare it for transmission and reconvert or decode it at the receiving end. CODEC standards that are commonly used for compression include G.729 (8kbps) and G.723.1 (5.3-6.3kbps). Developments in compression algorithms have enabled voice to be transmitted at toll quality at bandwidths significantly less than 64k, allowing for more efficient bandwidth consumption on packet networks. Optionally one or more video CODECs such as H.261 or H.263 may be supported by the H.323 terminal. The H.323 terminal must also support at least a scaled down implementation of the following H.323 subset protocols:

– H.225 Registration, Admission and Status (RAS). A RAS channel is opened between H.323 endpoints (such as terminals) and the gatekeeper prior to the opening of any other channels. The RAS function enables the terminal to register with the gatekeeper. When a call is terminated the H.225 RAS function also manages the final disengagement messages.

– H.225 call control signaling for call setup and teardown. When the gatekeeper has registered the terminal’s location and given it permission to make a call, the next step is the exchange of call signaling in order to connect to the destination terminal. H.225 messages are themselves transported in TCP and are very similar to the Q.931 messages used for call setup in ISDN.

– H.245 media control messages are then exchanged between the H.323 endpoints relating to the following: – Capabilities exchange between the terminals. This negotiates parameters such as deciding what UDP port to use for the real-time traffic, and what CODEC to use. – Opening and closing of logical channels to carry media streams – Flow control

– Real Time Transport Protocol (RTP). This protocol will be described in the next section.

– T.120 for data conferencing between H.323 terminals is optional.


A gatekeeper can be thought of the coordinator of a H.323 network and is the focal point for all calls within the network. The H.323 protocol scales by dividing internetworks into logical zones. A H.323 zone may span one or indeed several routed IP subnets, however a single gatekeeper services each zone.
The gatekeeper is not a mandatory feature of the H.323 implementation, however if it is employed then it must be utilized for the following functions:

– Address translation between E.164 telephone numbers and IP addresses.
– Admission control specifying what devices can call what numbers. This is performed as part of the H.225 RAS function. This may be a null-function all-accepting policy or it may be used to implement a specific call admission policy.
– Bandwidth control: Endstations as well as sending admission request messages also send bandwidth requests which may or may not be confirmed by the gatekeeper. This feature can be used to control the allocation of network bandwidth for specific users, groups or regions of the network. Like admission control it can also have an all-accepting policy.

A number of other optional functions may also be supported including:
– Endpoint authentication and authorization along with call accounting services – Call-signaling routing: Endpoints can signal directly to each other or alternatively via the gatekeeper. The later technique is called gatekeeper routed signaling and offers the advantages of call monitoring by the gatekeeper, as well as gatekeeper-assisted path determination and load balancing. – Providing SNMP management information.

The gateway interfaces between a H.323 network and a non-H.323 network such as the traditional PSTN. It is responsible for translating the signaling (e.g. between H.323 and Q.931 for ISDN) as well as compressing and de-compressing the voice. Several gateways may exist in a H.323 zone but they each must register with an active gatekeeper and in doing so indicate their gateway function. A gateway device might, for example, have an Ethernet connection to a LAN where H.323 telephony is being implemented along with one or more interfaces to a non-H.323 network such as ISDN or the PSTN. A terminal must communicate with a gateway using H.225 and H.245 when attempting to communicate with a device across a non-H.323 network.

Multipoint Control Unit (MCU)
MCUs provide conference capability for three or more H.323 terminals. All terminals engaging in the conference must establish connectivity with the MCU. The MCU itself is divided into two functional components:

Multipoint Controller (MC)
The MC provides control of the media channels such as negotiating codecs and deciding whether the conference can use multicasting instead of unicasts. Each device that joins the conference must establish a H.245 session with the MC, which is dynamically elected amongst conference participating devices.

Multipoint Processor (MP)
The MP sends and receives media streams to and from participating devices. It may also convert codec formats and mix media streams from multiple sources. The MP function may be implemented on the same physical device as the MC.

The final point that I made about the MP touches on a broader issue regarding H.323. Namely, all H.323 entities are defined as logical elements and are not affiliated with any physical device. While the gatekeeper function is often implemented on a standalone PC, it may for example be implemented on a device that is also acting as a gateway such as an IP router.

Another issue worth emphasizing is that H.323 does not guarantee quality of service (QOS) since the first version of the protocol suite was developed primarily for a LAN environment where bandwidth is plentiful. Later in this series I will talk about transporting voice over IP, frame relay and ATM. In these sections I will discuss QOS strategies relating to each of these technologies.

Transport protocols
Voice over IP employs UDP at the transport layer since the retransmission feature of TCP would be useless for a real-time traffic. The Real time Transport Protocol (RTP) provides the additional functionality necessary for real time applications. RTP provides packet sequencing and timestamping to facilitate efficient packet Reassembly at the receiving end. The Real time Control Protocol (RTCP) provides performance monitoring of the RTP stream. Packet loss, Delay and Jitter calculations can be performed from the information contained in the RTP header. It is important to note that while RTCP provides performance monitoring, it does not provide a QOS guarantee. So a voice packet is encapsulated with an RTP header (12 bytes), which then has a UDP header (8 bytes) attached before being wrapped up in an IP packet (20-byte header). Bearing in mind that voice packets are small in order to minimise delay, do you notice any issue here? Exactly, overhead! A voice packet’s payload is typically engineering so as to correspond to a delay of the order of 20-30ms. This works out at a payload of 20 bytes for a G.729 codec operating at 8kbps. Hence the header would be twice the payload. For this reason compressed RTP was developed, which can reduce the RTP/UDP/IP header from 40 bytes to 2 or 4 bytes.

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