Procedure

First, we must configure the gateway, this can be done with relative ease by an IVR (Interative Voice Response) menu or through any Internet browser. The most relevant aspects of the configuration are the use of SIP (Session Initiation Protocol) as the signaling protocol, assigning IP addresses to each gateway and disabling NAT (Network Address Translation). For audio settings: the G.729 codec is used, a packetization of two samples per packet and no silence suppression is performed.

After the equipment configuration, a sniffer is launched in order to make the traffic captures, subsequently a call between terminals is made. Packet capture is performed by sniffer, while the call is in progress and the data is stored for later analysis.

VoIP model

If we analyze the captured trace, with any of the recommended tools for the capture and the interpretation of network data, we can reconstruct the headers distribution of each captured packet and thus deduce the Figure # 2.

Figure # 2: Header description for each packet for voice traffic.

As we can be seen, VoIP uses RTP (Real-Time Transport Protocol) for transmitting data in real time and this is transported by UDP, Figure # 2 allows checking the use of those protocols. We also highlight in the analysis of packet capture, that the transmission of each packet has a constant distribution (no packet bursts). In addition, we confirm that each terminal sends packets every 20 ms and the data content in packets corresponds to the type of used codec (G.729) and the number of defined samples in the configuration for each gateway (two samples per packet).

An important aspect to consider for a traffic model is time, or variation thereof, in which each packet is forwarded. This requires to extract from trace file, the time between the beginning (or arrivals) of each packet. Figure # 3 shows the variation in the duration of each packet, for the sent and received voice traffic, based on the transmission time. For samples taken in this case, an average inter-packet time of 20 ms with a standard deviation of 0.62 ms was obtained. Each voice connection has a bandwidth consumption, on IP level of BW = (60x8) / 20X10^-3, which corresponds to 24 Kbps.

Figure # 3: Time detail between the beginnings of each packet for voice traffic.

Conclusion

The voice transmission model is relatively simple and stable in terms of time and size because no burst behavior is detected, or it can be consider as only one. Generating this type of IP flow, consists on the sending of 60-byte packets (including headers) every 20 ms with a standard deviation of 0.62 ms, this if the G.729 codec and a packetization of two samples are used. In the case, where the change of codec or the number of samples per packet is required, the changes are simple to make. For this, we must take into account the sampling frequency defined by the new used codec and the amount of samples that we want to package which would be included after the RTP header.