Session Initiation Procol in VoIP

SIP is a signaling protocol, slowly supporting H.323 and its main function consists in initiating sessions between clients. Basically, it provides various functions enabling operations such as dialing or the engaged signal. Additionally, it supports advanced possibilities of operating a connection, used by signaling protocol (used for signaling in conventional telephony). However, SIP is considerably different from the centralized SS7, even with the fact that it is a protocol of peer-to-peer type, which means that it requires only a simple skeletal network, without extensive equipment support. The whole intelligence of SIP focuses in the end knots of SIP. SIP is responsible for signaling and, by extension, it cooperates with other protocols. Session Description Protocol (SDP) describes the way of transport of multimedia in a network. SDP messages, just like SIP messages are of text character and they specify what possibilities the network ends must have. Session Announcement Protocol (SAP) is used for informing a greater number of users on the session that is being open at the moment, for instance in case of a conference or broadcast services, such as radio or television.

Basically SIP is quite similar to HTTP protocol in that, among other, it uses casual text and simple mechanisms of the request - answer to the request type or code messages such as error 404. To identify terminals, SIP uses addresses similar to e-mail addresses user@domain:port, where the default port is port 5060. A SIP-based system consists of two elements:

- Terminals functionally similar to H.323 terminals. At least two terminals are necessary to begin a session and they can communicate without engaging infrastructure of SIP network. A computer with proper software, special telephone connected to the network outlet or a gateway plugged into a network outlet, to which we plug a casual telephone, can become terminals.

- SIP servers have functionality similar to a gatekeeper in H.323. However, they do not play as significant role, for the notifications do not have to go through the server. The servers are used mainly for routing and redirecting notifications and sometimes for simple authorization. Basically, there are two work modes of a server, namely proxy and redirect. It is advisable to let each server work in both modes. Proxy server's task is to transit connection request to the right domain. It analyzes the INVITE order of SIP and on basis of the address it contains, it directs it to a different knot in the network. Such a way of making connections makes it possible to skip problems connected with address translation (NAT). SIP proxy also enables connections between VoIP terminals and PSTN network. A redirect server informs the client so that he contacts directly another server and, unlike proxy server, it does not have to monitor notifications.

Quality in VoIP

What influences quality of the service is first and foremost the quality of our Internet connection, namely:

a) CAPCITY

Basically, VoIP does not require too big symmetrical link capacity so that it can work properly. It all depends on the codec that is used for compression of the speech signal. Below you will find a list of the most popular VoIP codecs with values of symmetrical capacities necessary for their correct speech transmission. Unfortunately, however, nothing comes for free and very often, when the capacity required by a given codec decreases, the delay it introduces increases. The time the processors on the sending and receiving side need to convert the sound is prolonged. Choice of a codec is then a compromise between capacity, delay and required sound quality.

b) DELAY

Too long delay in transmission on our link may make the conversation lose interactiveness. In other words, we will have to wait long to hear our interlocutor's answer and the whole dialogue may be made very difficult. Below there are conditions our connection has to fulfill in relation to delays, to make VoIP operate properly. Obviously, delays are measured one-way, between us and our interlocutor.

c) JITTER

Jitter is created when packets reach the other point of the network with delay. There simply is a time difference between subsequent delays. This phenomenon is extremely onerous and over a certain value, it degrades quality of the conversation. This results in fragments of the conversation being cut out and the quality is generally decreased. Below, you will find a table with acceptable jitter values, with which the conversation is still possible.

d) LOST PACKETS

Number of the lost packets also plays an important role for the more packets fail to reach our interlocutor, the more broken our voice will be. Generally, in order to maintain good conversation quality, number of the lost packets should not exceed 5%.

Basic VoIP devices

There are many devices that are part of VoIP network, however, the two most popular ones are a VoIP gateway and VoIP telephone. Gateways are devices which make using VoIP telephony via casual telephone possible. The make it possible for users with no connection to the Internet, to use VoIP network. The gateways have two main ports, namely:

- Port with RJ-11 outlet, to which a casual, telephone is connected
. - Port WAN, to which the Internet is connected, usually with a cable with RJ-45 junction.

The main task of a gateway is change of signals coming from a telephone (speech and signaling signals) to VoIP signals. The most often used signaling protocol is SIP. Moreover, a gateway implements a few speech codecs. Developed gateways have many other ports:

- additional ports making connection of a greater number of telephones possible
- port ISDN
- ports LAN making connection of computers possible
- USB ports
- others...

Apart from its basic function, the gateway can perform a few others:

- router, NAT, DNS, movement shaping, QoS, many others...

These devices are becoming more and more popular due to their simplicity, functionality and decreasing VoIP connections prices. A VoIP telephone is the type of telephone which connects itself directly to the Internet, usually via RJ-45 junction. Just like a gateway, it makes carrying out conversations without a computer possible (it may be needed just during configuration). Telephones, just like gateways, usually use the SIP protocol.

See also: VoIP - Voice Over IP - Internet Telephony