The IP networks have become the standard for the connection of networks. As a result of its popularity, IP has become the most used flexible protocol. The IP networks are implemented on all the operating systems, they are highly scalable and they are ideal for the transmission of data in WAN connections.
Voice Over IP (VoIP), as its name indicates, deals with the sending of voice through an IP network. The IP networks are based on the packet switching technologies and each package transports a portion of the information. Each packet is semi-autonomous and it has its own header. Moreover, it is sent independently. This makes the infrastructure work more flexibly than a traditional telephoning network, but, at the same time, it has important issues to be taken into account such as the delay and the jitter in the traffic flows.
There are many technologies that can be implemented when a VoIP solution is built. Each manufacturer has a series of devices and models, depending on the client’s needs.
Normally, a VoIP network has the following components:
- Call Server: Server to which the phones are connected. It deals with the operations related to the register and admission control and the establishment of calls.
- Gateway: It is used to interconnect an internal network with other networks of the same type or not, translating protocols when needed.
- VoIP protocols: There are two of protocol families : The transport protocols and signaling protocols. The signaling protocols deal with the functions derived from the architecture of the telephonic systems. The establishment of calls and the control of them. The most widely-spread protocols nowadays are the family H. 323 and Session Initiation Protocol (SIP). The transport protocols are used to encapsulate and transport. The protocol used with this purpose is known as Real Time Transport Protocol (RTP). The voice packets are created using a codec, which is defined between both endpoints to be encapsulated in RTP afterwards.
- Codec: It is used to convert the analogic signal of a source into a series of digital samples and vice versa. The most common codecs are the series ITU-T G ((G.711, G.722, G.723, G.726, G.729.1).
- IP telephones (Softphone or deskphone): This is a device based on hardware or software that contains the elements to create the analogic-digital conversion and vice versa. This conversion is needed for the voice flow both ways.
- IP network components: the functions and own equipment of the IP network are necessary for the functioning of the system, like the DHCP, DNS, TFTP and NTP services.
The most important reason to migrateto VoIP architectures is the savings that it represents for companies. Instead of investing in a series of telephonic circuits, the clients only need to have a data connection, since the VoIP traffic travels in IP packets and the connection can be shared. Additionally, the IP packets can switch to any point in the internet, and the costs of transport are reduced drastically. Taking this into account, the benefits of the investment needed for the migration to VoIP systems can be seen in short-term.
Another benefit that has an impact on the economic level is the abstention of a convergent network, where the data services are shared with the telephoning services. This way, many things are accomplished: an only support operation, an only convergent network fulfilling both functions (an interconnection network is deployed), the maintenance of the network deploying an only operation and training plans for operators with an only transmission technology. Additionally, the changes in the network are significantly simplified (add, move or change terminals and users). It is important to highlight that having these costs into account, the investment in specialized equipment for VoIP is enormously mitigated.
More information at: http://tools.ietf.org/html/rfc2543 y http://www.itu.int/rec/T-REC-H.323/en