Distributed or grid computing in general entails parallel computing, which is based on conventional computers (with standard processors, storage devices, power supplies, etc.), connected to the network (LAN or WAN) using conventional protocols such as Ethernet. While conventional supercomputer comprises a plurality of processors connected to the local high-speed bus.
The main advantage of distributed computing is that a single cell of a computer system can be purchased as an ordinary non-specialized computer. Thus it is possible to obtain substantially the same computational power as in conventional supercomputers, but with much lower cost.
What are the main protocols used in VoIP? The main protocols used for establishing voice over internet protocol (VoIP) connections include H.323, IAX (Asterisk), Jingle, based on instant messaging protocol Jabber open standard, MGCP, SCCP (Cisco proprietary systems), Network Configuration Management, SIP and more.
Packet loss
When saturation occurs, the buffers need to release some of the bandwidth, thus neglecting some packages. However, the VoIP traffic is transmitted over the UDP layer, which implies that no mechanism of flow control or re-transmission of lost packets is provided by the transport layer. This means giving high importance to RTP and RTCP (Real-Time Transport (Control) Protocol) which will be used to calculate the rate of packet loss and react accordingly at the application layer.
Latency
Latency is the delay between the time between sending a packet and its reception by the addressee. More latency, the greater the transfer is long and will be shifted as part of Network Configuration Management. For optimal communication, control of transmission delay is an important point to reduce the echo effect or feeling metallic voice. Time packet transmission in IP-based network depends on many factors, such as the number of active devices in the network crossed, the flow of transit available and propagation delay information.
Delay variation (jitter)
Jitter is the variation of transmission time from beginning to end between packets belonging to the same data stream. It is due to the variance of the transmission time of the packet (ie: voice samples) in the telecommunication network and may result if it is too high, a deterioration of voice quality. The cause of this problem may be due to the difference taken by the packets in the network paths to a point network congestion or for encapsulating IP packets concern.
This is the term used to refer to the log stream. It is a unit often mistaken for a flow unit, but it actually defines the frequency range and throughput depends, hence the confusion. Several solutions exist to make Voice over IP.
