The Ninth IASTED International Conference on
Parallel and Distributed Computing and Networks
PDCN 2010

February 16 – 18, 2010
Innsbruck, Austria


Overview of VoIP, and Multimedia over IP Networks

Prof. Nader F. Mir
San Jose State University, USA


We are witnessing the multimedia over IP network technology is destined to play an increasingly important role in communication systems. With the demand for multimedia applications, there will be a growing interest in identifying suitable network architectures and transmitting facilities for this technology. Communication industry has spent considerable effort in designing an IP-based media transport mechanism, voice over IP (VoIP), and multimedia networks that can deliver voice-band telephony with the quality of the telephone networks. The Internet offers phone services less expensive and with numerous additional features such as video conferencing, online directory services, and the Web incorporation.
In this tutorial, we present the fundamentals of VoIP and also Multimedia over IP networks schemes. We explain the transportation of real-time signals along with the signaling protocols used in voice over IP (VoIP) telephony and multimedia networking. The tutorial covers the signaling protocols as H.323 series of protocols, and Session Initiation Protocol (SIP) which are responsible for session signaling. The H.323 protocols interact to provide ideal telephone communication, providing phone numbers to IP address mapping, handling digitized audio streaming in IP telephony, and providing signaling functions for call setup and call management. The H.323 series support simultaneous voice and data transmission and can transmit binary messages that are encoded using basic encoding rules. We also review the Session Initiation Protocol (SIP) as one of the most important VoIP signaling protocols operating in the application layer of TCP/IP model. SIP can perform both unicast and multicast sessions and supports user mobility and handles signals and identifies user location, call setup, call termination, and busy signals. SIP can use multicast to support conference calls and uses the Session Description Protocol (SDP) to negotiate parameters.
The tutorial further covers Compression of multimedia components such as Digital Voice and Video, focusing on data-compression techniques for voice and video to prepare digital voice and video for multimedia networking will be [resented. The topic starts with the analysis of information-source fundamentals, source coding, and limits of data compression and explains all the steps of the conversion from raw voice to compressed binary form, such as sampling, quantization, and encoding. The discussion also summarizes the limits of compression and explains typical processes of still-image and video-compression techniques, such as JPEG, MPEG, and MP3.
The tutorial will then present real-time transport protocols, such as Real-Time Transport protocol (RTP) and the Real-Time Control Protocol (RTCP). The next topic is streaming video in a single server, using content distribution networks (CDNs). Also discussed is the Stream Control Transmission Protocol (SCTP)), which provides a general-purpose transport protocol for transporting stream traffic. The tutorial describes detailed streaming source modeling and analysis. In real-time applications, a stream of data is sent at a constant rate. This data must be delivered to the appropriate application on the destination system, using real-time protocols. The most widely applied protocol for real-time transmission is the Real-Time Transport Protocol (RTP), including its companion version: Real-Time Control Protocol (RTCP). UDP cannot provide any timing information. RTP is built on top of the existing UDP stack. Real-time applications may use multicasting for data delivery.
We also cover video streaming that presents a significant challenge to network designers. A video in a single server can be streamed from a video server to a client at the client request. The high bit-rate video streaming must sometimes pass through many Internet service providers, leading to the likelihood of significant delay and loss on video. One practical solution to this challenge is to use content distribution networks (CDNs) for distributing stored multimedia content. Video streaming, e-mail, and image packets in the best-effort Internet are mixed in the output queue of the main exit router of a domain. We explain how a burst of packets, primarily from the image file source, could cause IP video streaming packets to be excessively delayed or lost at the router. One solution in this case is to mark each packet as to which class of traffic it belongs to. This can be done by using the type of-service (ToS) field in IPv4 packets. transmitted packets are first classified in terms of their priorities and are queued in a first in, first out (FIFO) order. The priority of an image file can be equal to or less than the one for video streaming, owing to the arrangement of purchased services. We explain a similar solution for IPv6 technology.
The Tutorial will then focus on the Stream Control Transmission Protocol (SCTP) providing a general-purpose transport protocol for message-oriented applications. SCTP is a reliable transport protocol for transporting stream traffic, can operate on top of unreliable connectionless networks, and offers acknowledged and non-duplicated transmission data on connectionless networks. The tutorial will describe the SCTP's unique features such as being error free, having ordered and unordered delivery modes, having effective methods to avoid flooding congestion and masquerade attacks, multipoint, and the fact that it allows several streams within a connection. The Tutorial will ultimately present multimedia over general Wireless and WiMAX networks.


Overview of IP Telephony (1 hour)

Overview of Digital Voice and Compression (1 hour)

Video Streaming and Real-Time Media Transport Protocols (1 hour)

Voice and Video Streaming over Wireless Networks (1 hour)

Background Knowledge Expected of the Participants

Audience can be from academia or industry. Any individual with basic knowledge of computer networking can benefit from this tutorial.

Qualifications of the Instructor(s)

Tutorial Chair Portrait

Nader F. Mir received a B.Sc. degree with honors in electrical and computer engineering in 1985 and MSc and PhD degrees, both in electrical engineering, from Washington University in St. Louis, in 1990 and 1994, respectively.
He is currently a Professor and Department Associate Chairman of Electrical Engineering at San Jose State University, California. He is also the Director of the MSE Program in Optical Sensors Networks for Lockheed Martin Space Systems.
His research interests are analysis of computer communication networks, design and analysis of switching systems, network design for wireless ad hoc, internet and sensor systems, information systems and applications of digital integrated circuits in computer communications.


Nader F. Mir, "Computer and Communication Networks," By Prentice-Hall Publishing Co., ISBN: 0131747991, 1st edition Nov. 2006
[2]  "Carrier Grade Voice Over IP," 2nd Ed., 2003, Daniel Collins, ISBN 0-07-140634-4, McGraw-Hill Networking Professional.