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Voice over IP (VoIP) Glossary

Connection versus packet oriented
Networks are generally categorized into packet or connection oriented. In a connection oriented network data is transferred by switching, the network elements together to provide a virtual connection or to use a fixed connection (e.g. a telephone line). During the transfer the connection is reserved for this purpose only whether useful data is transmitted or not. Thus a lot of potential bandwidth is wasted. 
In a packet-oriented network, data is sent only when it needs to be and grouped for efficiency into packets or frames. The destination address is attached to the beginning of the data and the network elements then route the packets. While useful data is only sent when needed, each packet needs to include at least the address information and depending on the network the order of the packets is not always guaranteed or even whether they arrive. Thus, traditionally voice was transmitted on connection and data on packet-oriented networks. 
H.323
H.323 is an ITU standard for the usage of multimedia communication via packet-oriented networks that guarantees interoperability between different equipment vendors. The largest packet-oriented network is the Internet but also WAN, ISDN or dialup connections on which data is transported in packets (e.g. PPP) belong into this group. H.323 describes the general infrastructure and the utilization of different speech coders and protocol signaling stacks. The speech coders are defined in their respective sub standards, e.g. G.711 (Alaw and ulaw used in ISDN), G.722, G.723.1 and G.729.A for speech encoding. H.323 is definitely the most widely deployed and mature standard, but it is also criticized for being complicated to implement by vendors and uses a lot of resources, which are not abundant (especially in terminals). 
H.450
The H.450 Supplementary Services is a series of standards that define extended functionality and distribution thereof in a H.323 infrastructure. These services are called supplementary since they extend the basic services of H.323, which essentially boil down to being able to establish and release a connection. Examples of such Supplementary Services are Hold (local and remote), Call Waiting (an indication that a person is trying to reach someone who busy talking to someone else), Call Diverting (call is transferred when busy), Call Redirect (a call is transferred to a mobile after working hours), Pickup, Parking, etc - features that a classical PBX and ISDN offer. In addition, mechanisms are provided that enable vendors to tunnel proprietary supplementary services if need be - of course this is not intended to become a standard but is a workaround until these features are interesting enough to be integrated. 
Jitter
Jitter is the variance of latency (i.e. delay) in a connection. The problem is that audio devices or connection-oriented systems (e.g. ISDN or PSTN) need a continuous stream of data. In order to compensate for this, VoIP terminals and gateways implement a jitter buffer that collect the packets before relaying them onto their audio devices or connection-oriented lines (e.g. ISDN), respectively. An increase in the jitter buffer size decreases the likelihood of data being missed but also has the drawback that it increases latency of a connection. 
Latency
The delay or time span between the voice being digitalized at the senders Location and then output at the receivers end is the latency of a connection. Latency is influenced by the distance the data has to travel, the packet size, the number and delay time of network elements between the terminals and of course the latency generated by the terminals themselves when sending, receiving, encoding, decoding and compensating jitter. 
LPCP
LPCP (Lightweight Phone Control Protocol) is a standard that is used to control telephones in a pragmatic and simple way. The memory and resources needed for VoIP telephones and terminals can be reduced to the minimum, which is more cost effective. The call signaling (SIP / H.323) is done for the phone on a server that has enough memory resources. 
Megaco / H.248
Megaco / H.248 is also a media gateway control protocol such as LPCP but more complex and general. 
QoS
Quality of Service pertains to the quality of a connection and this is especially important for connections relaying voice since the user feels the impact immediately. A retransmission cannot make up for the lost data. The Internet protocol was devised as a "best effort" data network and thus it does consider jitter, latency or even data loss a problem. Ergo, it does not handle voice well per se. To make the transmission of voice possible it must be given the necessary priority and bandwidth. There are mechanisms for reserving bandwidth (see RSVP) but they add network equipment with an additional burden of handling this functionality and slow down establishing connections. The other pragmatic approach to this problem is to acknowledge that normally the access point (interconnection between LAN and WAN) is the most critical section. By prioritizing packets (see ToS) and ensuring the access point is not overloaded - QoS can be achieved. The data traffic load in the backbone is about 10 times that of voice (thanks to WWW) of carriers so this should not be the problem. 
RTP
The RTP (real-time transport protocol) labels all information transferred by a sender with a timestamp. By examining the timestamps the receiver is able to sort the packets in the original order and synchronize real time streams and/or compensate jitter in audio data. 
RTCP
The RTCP (real-time transport control protocol) was devised to give Applications a status on the quality of a network. With this information parameters affecting the transmission of data, e.g. the jitter buffer size, can be optimized. 
RSVP
The RSVP (resource reservation protocol) makes it possible to reserve bandwidth in non-terminal network elements such as routers. This and prioritization (see TOS) is done to practically eliminate latency, jitter and loss of packet problems for real-time application such as VoIP. 
SIP
SIP (Session Initiation Protocol) is a highly pragmatic, ASCII-based protocol and competing standard to H.323. Its main advantages are that it is easy to implement, debug and to integrate applications. It is newer than H.323 and but does not standardizes many of the supplementary services, yet. At the moment, it is not pushing H.323 out of the scene. 
TOS
In order to specify the priority of a packet the Internet protocol has a ToS (type of service) field. 
VoIP
VoIP (Voice over Internet Protocol) is a term used for voice being transported via the Internet, intranet or data links to the Internet regardless whether H.323, SIP or a proprietary standard is used. 

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