In order to better tune the jitter correction, the best practice is to note packets that arrive late and, with that base data, calculate a ratio of those packets and packets that are successfully transferred. They will try to remedy packet delay if required and if possible. VoIP packets in networks have very changeable packet inter-arrival intervals because they are usually smaller than normal data packets, and are therefore more numerous, with a bigger chance to get some delay. One of the packets encounters some delay on its way and it is received a little later than it was expected. In the image above, you can notice that the time it takes for packets to be sent is not the same as the time in which they will arrive at the receiver side. We can look at it as the anomaly in tempo, with which packet is expected to come and the time he was late to really get there. It is not usual, but the packets could take different routes or get load-balanced through two similar paths where one of those is congested in that moment. When someone is sending VoIP communication at a normal interval, (let’s say one frame every 10 ms), those packets could have stuck somewhere in-between the network and not arrive at expected regular pace to the destined station. ![]() ![]() On the other hand, when we speak about Voice traffic and VoIP network environment this can be an issue. Actually, TCP/IP is responsible for dealing with the jitter impact on communication. As a time-shift phenomenon, it usually does not cause any communication problems. It is a specific phenomenon that normally exists in bigger packet-switched networks. ![]() In other words, jitter is measuring the time difference in packet inter-arrival time. If you know what delay is, jitter is simply the difference in packet delay.
0 Comments
Leave a Reply. |
AuthorWrite something about yourself. No need to be fancy, just an overview. ArchivesCategories |