Course notes by Gregor v. Bochmann, University of Ottawa

Multimedia application protocols

(A) Communication requirements of multimedia applications

MM applications are classified into

presentational applications: one or several users obtain a MM presentation which is stored at some server(s)
interactive applications: two or more users communicate with one another in real time

Communication requirements for the media streams:

high bandwidth transmission (for high quality video)
reasonable loss rate (typically less than 1%)
small delay and delay jitter (of the order of 0.1 seconds; only for interactive applications)

Note: for presentational applications, one can use a larger play-out buffer (which may introduce an additional delay of, say, a second) in order to eliminate the delay jitter

if many users are involved in a presentational application or a teleconference, multicast transmission would be very useful to reduce the bandwidth requirement at the interface between the source and the network.

Note: In the Internet context, multicasting is available in the LAN context. The so-called MBone is an experimental multicasting service for long distances over the Internet, but it will probably never become a standard Internet service.

Communication requirements for controlling the applications: reliable transmission of control messages

(B) Transport level protocols

IP: provides packet transmission between different end-systems (no reliability: possible losses, possible out-of-order delivery; however, if received the packet is received without transmission error)
TCP: provides reliability for streams, but uncontrollable delay. It is suitable for control traffic, but not for MM streams. However, for applications involving multicasting, one may also use UDP for the transmission of control information (since UDP can run on top of IP multicasting, which TCP cannot do).
UDP: provides source and destination port numbers which are useful to define logical flows of data. This concept is used to identify streams of MM information. Otherwise the transmission service is identical to what is provided by IP.
RTP and RTCP (for an overview see paper on SIP, here are slides on RTP by H. Schulzrinne. Note: there are many pointers to interesting information about Internet telephony and related IETF protocols on Schulzrinne's personal Web page)

(C) IP telephony and teleconferencing

The ITU standard on IP telephony is H.323 (see paper on H.323). Within the IETF, the Session Initiation Protocol (SIP) has been proposed (see paper on SIP). Within the last years, some of the differences between these two standards have disappeared through successive revisions. << A comparison between these two standards is given here >> The provision of additional services in the context of SIP is discussed here. SIP uses the so-called Session Description Protocol (SDP) to describe the proposed MM quality of service parameters for the communication session (see extract from SDP document). << Different approaches to providing telephony services using Internet protocols are described here, including telephony over the Internet and using Internet protocols over private networks. The issue of integration of voice and data applications is also discussed. >>

(D) Mobility issues

One can distinguish the following kinds of "mobility":

device mobility: the device can move while it is involved in a session, or betwen sessions
user mobility: the user can move between sessions and use any available device for a new session, wherever s/he may be. This issues is discussed here in conjunction
session mobility: an ongoing session using a given device may continue without interruption on a different device (e.g. go from a hand-held device to a high-quality video screen)

Last updated: October 8, 2002