Tests have been carried out by means of a prototype of a distributed multimedia DBMS, both on local and wide area network (Internet) and with respect to different network loads. Such a prototype runs on top of a distributed environment and is based on a client-server architecture.
Multimedia documents (text, audio, still images, videos) are stored in a compressed form on different servers in a way which is transparent to client applications. The compression is due, as usual, to the need of both saving disk space, and of limiting the bandwidth requirements.
The database is queried by standard relational calculus formulas through a suitable graphic interface. A set of icons representing the query result provides information on both the type and the content of documents, each of which can then be selected by ``pointing-and-clicking'' on the corresponding icon. As a consequence, the document of interest is retrieved on the server and is transmitted back to the client, which either returns the document to the user once it has been wholly transferred, or decodes the document in real time (as in the case of audio and video conferences).
Our main goal is evaluating the applicability of TCP, UDP and RTP  in order to satisfy real time playout requirements with respect to the transmission of MPEG compressed video streams. In addition, the achievable performances are compared with those guaranteed by some widely used videoconferencing tools .
As it may be expected, we did not detect any kind of problem during the tests conducted on local area networks with respect to TCP: no substantial delivery delay has been detected and the network bandwidth resulted widely sufficient. Poor frame rates have been revealed only with respect to low computing power or displaying wide image sizes.
On the other side, on wide area networks, performances are greatly influenced by the available bandwidth. Our tests revealed that, for particularly limited bandwidth, the network capacity is quickly saturated by video streams with large image size.
With respect to the particular application considered the major drawback of TCP is represented by packet retrasmission, which results in significant delay and overhead in bandwidth occupation, and in a consistent delay jitter in delivering image frames. Such a drawback substantially neutralizes the positive effect due to MPEG higher compression rates.
Transport protocols which do not guarantee packet delivery, such as UDP and RTP, could be the solution. A serious difficulty in this direction is represented by the poor capability of the MPEG decoder in recovering from packet loss situations. Up to now, our tests on UDP have been performed neglecting this aspect, revealing good improvements with respect to TCP performances. Anyway, it is our aim, in the next few months, to face this problem, perhaps by filtering the information processed by the MPEG decoder.
Concluding, in this paper we show under which conditions TCP could be a reasonable solution to real time playout requirements, testing at the same time the applicability of different transmission protocols, such as UDP and RTP.
For what concerns activities to be performed in the near future, we plan to apply protocols specifically introduced for multimedia broadcasting, such as ST II , to the real time transmission of MPEG documents and to compare its performances with the ones provided by TCP, UDP and RTP.