In order to perform estimations of a traffic model for multimedia flows, it is necessary for the researcher to have precise control of the test environment, minimizing errors and interference that may occur, leaving a test scenario that allows a smooth development to the application to be modeled. As an example, we have selected a VoIP traffic which is quite easy to interpret and it provides some of the things that must be done.
Scenario for testing
Figure # 1 shows the diagram used for testing in order to obtain the pattern of traffic on the transmission of voice over IP is shown. In this case, we use a voice over IP gateway Linksys SPA 3102, with the aim of connecting a conventional telephone system with an IP network. A sniffer can capture traffic using the network for later analysis.
Figure # 1: Scenario used to determine the voice traffic.
The development of VoIP (Voice over Internet Protocol) technologies have been widely accepted by companies seeking lower costs for voice communications mainly in SMEs (Small and Medium Enterprises) environments. The term VoIP should not be confused with ToIP (Telephony over IP), VoIP refers to the technology required for voice communication over IP, while ToIP is a telephone service for users, which uses VoIP technology to give such service.
VoIP allows voice transmission via an IP network, including those connected to the Internet. It involves digitizing voice signals, via a codec. The bandwidth consumption of such communication is directly related to the codec as shown in Table # 1.
|G.711||56 or 64 Kbps|
|G.722||48, 56 or 64 Kbps|
|G.723||bit-rate 5,3 or 6,4 Kbps|
|G.729||8 or 13 Kbps|
Tabla # 1: Bandwidth consumption for different codec
Furthermore, VoIP uses various types of techniques for call signaling, not having a defined protocol in this field. SIP (Session Initiation Protocol) is one of the protocols used for this purpose, also deployments with H.323 (a recommendation from the ITU Telecommunication Standardization Sector, ITU-T) or IAX (Inter-Asterisk eXchange protocol, native to the Asterisk private branch exchange, PBX) can be found.
SIP is one of the most widespread protocol in the implementation of ToIP. This protocol is responsible for the end-to-end communication signaling, call establishment procedures, communication modification and its termination.For transmitting real-time data, VoIP uses the RTP (Real time Transport Protocol) protocol, which is responsible for transmission control in the multimedia sessions and uses UDP (User Datagram Protocol) as the transport protocol. In Figure # 1, the header distribution for each VoIP packet is observed.
Figure # 1: VoIP Packet transmitted by the stations.