# Studying the Impact of Different TCP DoS Attacks on the Parameters of VoIP Streams
## 4 Results and Discussion
### Results for Only Voice Streams  
Figure 2 represents the number of different TCP packets sent during normal operation. The x-axis represents time, and the y-axis represents the number of TCP packets. As can be seen from the graphs in normal operation mode, when the system was not under attack, there were only a few TCP packets; the presented result is for when the Asterisk settings were accessed through a browser and the page was loaded. There was a successful TCP session. Prove of this statement is the presence of TCP FIN packets, which were sent when the TCP session was closed from one of the two sides.  
Figure 3 provides the summarized results of the voice traffic exchanged in both directions between VM_4 (10.10.10.2) and VM_3 (10.10.10.6). As can be seen from the results, the traffic was passed through and processed by Asterisk (10.10.10.5). For all subsequent similar figures, the designations are: **SSRC**--indicates the synchronizing source. **Max Delta** is the difference between the arrival of one packet and the arrival of the previous packet; **Max jitter** is the maximum measured jitter for the entire duration of the call; **Mean jitter** is the average value of the jitter for the entire duration of the call; **Max skew** is the maximum distortion; **RTP packets** is the number of RTP packets; **Lost** is the percentage of lost packets out of the total number of packets; **Seq error** indicates errors in TCP/UDP sessions; **Start at** indicates at which moment the corresponding studied voice/video stream starts; **Duration** is the duration of the studied voice/video stream; **Clock drift** shows what the deviation from the clock frequency is; **Freq Drift** shows what the deviation from the sample rate is. In this article, only **max jitter, mean jitter and packet lost** will be monitored. Figure 3a shows the summarized results for the voice stream exchanged between VM_4 and Asterisk, and Figure 3b shows the summarized results for the voice stream exchanged between VM_3 and Asterisk. In terms of the mean jitter, both streams were within the norm, where the maximum allowed value is 30 ms [55,56]. The maximum jitter values were also within the norm. No packet losses were observed.  
Figure 2: Number of different TCP packets sent during normal operation.  
Figure 4a shows the variation in the jitter for the voice stream between VM_4 and Asterisk in both directions during the whole conversation time. Figure 4b shows the variation in the jitter for the voice stream between VM_3 and Asterisk in both directions. The x-axis (Arrival time) represents time, and the y-axis (value ms) represents the value of the jitter in ms. As can be seen from the graph, the values were very far from the maximum allowable levels. Parenthetically, it should be noted that in this type of simulation study, when real-time audio/video streams are studied, the computing capabilities of the workstation used for network modeling are of great importance. Computing capabilities affect the values of the jitter and the delay. If the computing capabilities are low, the jitter and the delay values increase. This finding is a result of many years of working with GNS3 and the use of different kinds of workstations for network modeling.