A few years ago the Converged Networks, also known as Next Generation Networks or NGN ( Next Generation Network ), has been widely publicized and came to be present in our lives. Operators, corporations, and ISPs already make targeted investments in equipment and technologies to serve the NGN networks. Convergence of data, video, and voice ( triple play ) into a single infrastructure has also become a reality in the telecommunications market, in companies and in our homes. But how did this happen and what is its relationship with Voice over IP (VoIP)?
In order for network convergence to reach current status, a long way was needed, from the creation of the TCP / IP standard in the late 1960s to Ethernet in the 1970s and its constant evolution to the large-scale use of networks fiber optics and VoIP technology. In parallel, there was also the contribution of the growing popularization of computers, fixed and mobile telephony, internet and providers in the last three decades.
VoIP: a low-cost solution
The growth of voice communication solutions through Voice Over IP ( VoIP ) technology, seeks to offer an integrated and low-cost solution that can effectively replace conventional telephony. For this, users will have available in a single network data services, telephony and multimedia content such as TV, radio, cinema, videoconference, digital texts, and distance learning.
The VoIP is not exactly new. It had its initial development in Israel in the 1990s by VocalTec Communications, but it only became popular in the 2000s, by increasing the speed of transmission links, codecs that could efficiently perform voice coding and compression, and also by the emergence of SIP, which until today is the most used protocol for VoIP.
What is SIP?
Session Initiation Protocol (SIP ) provides a low-cost, open, and secure solution for the privacy of the telephone conversation. It is an OSI (application layer in the TCP / IP model) session layer protocol that can establish, modify, and terminate multimedia sessions – where session is considered to be a data exchange between a UAs ( User Agent ) association – as per for example, a telephone call over the Internet (ROSENBERG, SCHULZRINNE and CAMARILLO et al, 2011).
SIP is an Internet Engineering Task Force (IETF) standard that uses the “request-response” model to initiate interactive communication sessions between users. Inspired by other text-based Internet protocols such as SMTP (e-mail) and HTTP (web pages), it was developed to establish, change and terminate calls from one or more users on a network in a completely content-independent manner of call data. Like HTTP, SIP takes application controls to the terminal, eliminating the need for a switchboard. Historically, RFC 2543 was released with some bugs and vague definitions in its text, but after some products have been fixed, the 9 ‘bis’ version has finally been released that clarifies and defines the new RFC 3261.
This simplified mode has led several manufacturers to develop standards-based devices and solutions, increasing their popularity by becoming the main parameter of the VoIP protocol for the market.
Although the protocol allows for a peer – to – peer (P2P) connection without the need for an intermediary, the common framework for service implementation involves terminals known as the UA – User Agent (IP phone, softphone, ATA, or an HGU such as Cianet CTS2742BW), other server elements.
Sip trunks explained SIP servers are applications that receive requests, perform some operation, and return a response to the requesting device. There are three types of SIP servers: proxy server, redirection server, and log server. Because they are logical entities, more than one SIP server can be hosted on the same physical device (YOSHIOKA, 2003).
The proxy server acts as a facilitator of SIP message exchange, being the element responsible for the location of the target UA of the request. When it receives a SIP request, the proxy analyzes for which domain the message is destined and obtains the address of the proxy server of the destination domain, performing the forwarding.
The proxy can maintain a timer and records for all the requests and responses it receives and passes. The proxy uses this information in future processing of messages belonging to the same dialog. It is this information that will generate the billing ( billing ) used for accounting, billing or judicial interception purposes.
Like a proxy server, a redirection server also assists in the process of locating the recipient of a SIP request but does not forward SIP requests. Instead, it only responds to the originating UA which is the contact address to reach the target UA, or a next hop address to a proxy server or other nearest redirection server, without actually engaging in communication (YOSHIOK A, 2003).
A log server, also called Registrar, will contain the base location of the UA. Upon receiving the REGISTER request, the registration server updates the database used by the proxy and redirection servers for user location and responds with a message that the registration has been successful.
Solutions that allow telephone communication over the Internet with the use of VoIP should also be concerned with security, availability, and service quality aspects since the telephone communication that travels in data packets is critical for delays and loss of packets. The voice traffic, due to its characteristics, must be transmitted continuously and without intervals of silence, so that the message is intelligible by the listener.