Delivery Of Voice Transmissions Over A Packet example essay topic

Concept of Voice over Internet Protocol (VoIP) Voice over Internet Protocol, commonly known as VoIP, is a technology used for the transmission of voice information using the Internet Protocol (IP). VoIP allows the sending of voice information in a digital form using packets as opposed to the traditional circuit-committed protocols of the Public Switched Telephone network (PSTN). Traditionally when one wishes to make a phone call, a call is placed using a telephone that is connected to a circuit-switched network. In a circuit switched network, network resources are static; the network resources are allocated from the sender to the receiver before the start of the voice transmission.

Two points are connected in both directions thus creating a "circuit". The resources remain dedicated to the circuit during the entire transfer and the entire voice transmission follows the same path, this is the foundation of the PSTN. Up until circa 1960, every call had a dedicated wiring stretching from one end of the call to the other for the duration of the call. An example of this essential physical connection is a call from New York to Los Angeles required all the switches to connect pieces of copper wire between the two points for the entire duration of the call. Naturally this was expensive due to the resources in use. Telephone conversations today are somewhat more efficient and cost a lot less.

The voice transmission is digitized, and along with thousands of other phone calls can be combined onto a single fiber optic cable for much of the journey. Calls are transmitted at 64 Kbps in each direction, totaling 128 Kbps. There are 8 kilobits in a kilobyte; this translates to a 16 KB transmission every second and a ten minute phone call consuming nearly 10 MB. (Tyson and Valdes, 2000) Upon inspection of a typical phone conversation, much of the transmitted data is wasted. Only one party can talk at one time, which means that only half of the connection is in use at any given time. A significant amount of the time in most conversations is dead air where neither party is talking, even if this just occurs for seconds this is still a significant waste.

If these silent intervals could be removed and the transmission size be cut in half due to inactivity of one party at a time, the file size could be reduced to less than half. Furthermore, if only the packets that contained data (voice transmission) was sent only as opposed to a continuous stream of bytes being transmitted as with a standard circuit switched network, we have then formed the basis of a packet switched phone network. In packet-switched phone networks, the voice transmission is broken into packets, each of which can take a different route to the destination where the packets are recompiled into the original message. Data networks simply send and receive data as its needed, and, instead of the transmission occurring over a dedicated line, the data packets flow through thousands of possible paths.

This makes the network more efficient for a multitude of reasons; the network can balance the load across various pieces of equipment on a millisecond-by-millisecond basis. If there is a problem with one piece of equipment in the network while a message is being transferred, packets can then be routed around the problem, ensuring the delivery of the entire message. To better understand the delivery of voice transmissions over a packet-switched network, we must examine the protocols used in the process. A protocol is a special set of rules that end points in a telecommunication connection used when they communicate. Internet Protocol (IP) is a protocol used to send and receive data from one node to another at the Internet address level.

IP is part of a suite of protocols called TCP / IP (Transmission Control Protocol / Internet Protocol). Each node on the Internet has at least one IP address that uniquely identifies it from all other computers on the internet. When data is sent or received the data gets broken down into little chunks called packets. Each of these packets contains both the senders IP address and the receivers IP address.

This is what allows for each packet if necessary to be sent via a different route over the Internet. IP is only responsible for the delivery of packets; it is dependant upon TCP to re-order the packets correctly. In general, each IP packet contains a header, payload, and a trailer. The header contains instructions about the data carried by the packet.

This includes the origination and destination address, the packet number, and the length of the packet. The payload is the actual data that the packet is delivering to the destination. The trailer contains a couple of bits that tell the receiving node that it has reached the end of the packet. Other protocols involved in VoIP are User Datagram Protocol (UDP) and Real Time Transport Protocol. UDP much like IP is a stateless protocol, meaning that there is no continuing connection between the end points that are communicating.

Each packet is treated as an independent unit of data without any relation to any other unit of data. This is why IP relies on TCP to re-order the packets. The Real Time Transport protocol (RTP) is a protocol standard that that specifies how to manage real-time transmissions of data. RTP makes it possible to monitor data delivery and allows the receiver to detect if there is any packet loss and to compensate for any delay jitter. Information in the RTP header tells the receiver how to reconstruct the data and describes how the code bit streams are packet ized.

(Fogarty, 2002)..