Transmission errors and Transport Protocols

Data send in real-time hardly ever uses TCP more often UDP but the actual transmission is usually done with RTP protocol. The reason for that is used in TCP error control which might introduce an unacceptably long gap in the music. RTP is an internet transport protocol which allows the combining of several streams of data (plus timestamps and sequence numbers) into UDP packets – which are then delivered via IP unicast or multicast. According to their functionalities, the protocols can be classified into three categories: (Wang, Ostermann, Zhang, 2002)
-  Network-layer protocol  - provides basic network service support.
-  Transport protocol - provides end-to-end network transport functions for streaming applications.
-  Session control protocol - defines the messages and procedures to control the delivery of multimedia data during an established session.

Fig.2 Protocol stacks for media streaming
Figure 2 illustrates protocol stacks for media streaming. And so IP belongs to network-layer protocol. UDP and TCP are lower-layer transport protocols, whereas RTP and RTCP are upper-layer transport protocols. And RSTP is a session control protocol.

As RTP is layered on top of UDP packets may be lost during transmission. But the media players can handle this easily estimating the missing value and interpolate using the previous and succeeding samples (Tanenbaum, 2002).