voip

Voice over IP in network virtual environments

http://research.edm.uhasselt.be/jori/thesis/onlinethesis/contents.html

The speech signal is transmitted by digitising tiny pieces of it at regular intervals and sending these to the destination where an analogue signal is reconstructed. For good quality communication, the overall delay should be below 200 ms. Delay variance or jitter should be eliminated through buffering.

When the digitised speech signal is left uncompressed, a bandwidth of 64 kbps is needed for telephone quality communication.  Various compression techniques can reduce this amount.

To transmit the speech data, TCP is not a good choice: it has a lot of features which are unnecessary for VoIP, but which increase the overall delay. UDP itself is too simple, but we can extend it: this is the RTP is used in the TCP/IP architecture. The Real-time Transport Protocol (RTP) provides information for synchronisation, flow and congestion control and identification. To provide some quality of service (QoS) guarantees, resources can be reserved by using RSVP, the Resource Reservation Protocol.