ECE 599 VoIP Project

Project Proposal : Voice over IP

Voice Over IP is the new fancy development in the telecom industry. It promises to deliver cost-savings to users and service providers and is driving the convergence of network and telecom. It offers improvements in quality, interoperability and applications in the near future.

I. Content

I.1. Voice Encoding

In order to transmit speech/audio, we need to talk about different transmission techniques also known as codecs. For different applications there are different codecs, e.g. for ISDN the PCM standard is used. The ITU-T specifies these codecs in their G-series recommendations. We would like to focus on G.711 PCM, G.726 ADPCM, G.728 LD-CELP, G.729 CS-ACELP, G.729a, G.723.1 MP-MLQ, and G.723.1 ACELP. In addition to these codecs, we also want to consider MPEG codecs.

I.2. Transportation

Multimedia networking has become a main issue in the development of the Internet today. Usually, multimedia-networking applications are acutely delay-sensitive but loss tolerant. The Internet's network-layer protocol IP provides a best-effort service, which means that each datagram is delivered as quickly as possible from the source to destination. Thus, we first consider the limitations of such a best-effort service, which leads to packet loss, excessive end-to-end delay and delay jitter.
UDP segments generated by an Internet phone application are encapsulated in IP datagrams. These datagrams are delivered to destination through buffers in the routers. If some buffers are full and are unable to accept the IP datagram, the datagram is discarded. Since retransmission mechanisms are often unacceptable for interactive real-time audio applications such as Internet phone, because of the increased end-to-end delay, applications that run over UDP don't retransmit lost packets. The packet loss rates depend on the encoding and transmission of voice and on how the loss is concealed at the receiver.
We also have to deal with jitter which is the variation from packet to packet of the time a packet takes to get to destination and end-to-end delay which is the accumulation of transmission processing and queuing delays in routers, propagation delays, etc.

After this detailed survey, we discuss the mechanisms like client buffers, packet sequence numbers and timestamps that reduce delay and jitter. This discussion is lead in the context of Internet phone application.

We also mention a new IP protocol, IPv6 that was developed to solve the upcoming problem of the limitation of the 32-bit IP address space. Developing IPv6 gave the opportunity to improve several aspects of IPv4.

I.3. Devices and signaling protocols

• H323
H323 is a specification from the International Telecom Union (ITU) describing transmission of audio, video and data across an IP network.
Equipment compliant with H323 can communicate and interoperate with each other.
H323 specifications are based on definition of devices performing determined tasks and also a suite of different protocols leading the process of an IP multimedia communication.

• SIP
SIP is a text-based protocol defined by the Internet Engineering Task Force (IETF).
Its main goal is to define how to establish, maintain and terminate multimedia sessions.

II. Objectives

In our project, we intend to cover the following topics:
- A detailed survey and explanation of the mentioned audio codecs in respect to their prerequisites, the bit rate, the quality and complexity. With these results we like to point out the different applications of these codecs.
- Presentation of the limitation of the best-effort service provided by the IP protocol: packet loss, end-to-end delay, jitter.
- Discussion about jitter removal and packet-loss recovery, QoS
- IPv6 overview
- Definition of an IP Phone, Gateway, Proxy and Gatekeeper.
- Presentation of the tasks allocated to devices in H323 architecture and the different protocols used in the process.
- Advanced discussion about SIP architecture.
- Partial simulation of VoIP

III. Schedule

IV. References

[1] Computer Networking: A Top-Down Approach Featuring the Internet,
James F. Kurose & Keith W. Ross., First Edition, Addison Wesley Longman Editions,
ISBN 0-201-47711-4

[2] Internetworking Multimedia,
Jon Crowcroft, Mark Handley, Ian Wakeman, Morgan Kaufmann ISBN 1-55860-584-3

[3] Voice Over IP Fundamentals,
J Davidson & J Peters, Cisco Press ISBN 1-57870-168-6

[4] ITU-T G-Series

[5] Multimedia: Computing communications and applications, R. Steinmetz & K. Nahrstedt, Prentice Hall,1995

[6] CISCO website: http://www.cisco.com