1. Introduction
of Voice over IP |
|
Voice
over Internet Protocol transfers voice through IP
packets through the Internet. VoIP
changes analog (human voice) to digital use an ADC (analog to digital
converter). Then, it digitalizes voice in data packets and transmits
it. Finally, it reconverts them in voice at destination... |
2.
Pulse Code Modulation And ADC Convertor |
|
Pulse
Code Modulation (PCM) is a digital scheme for transmitting analog
data. It is the process of changing/converting signals from one form
(analog) to the other (digital)... |
| For more details please see
PulseCodeModulation.doc and
ADCConvertor.doc. |
|
There are many protocols
support Voice over IP, for overview of these protocols please see
VOIPProtocols.ppt. For more details
please see the documents below. |
| 3.1
Session Initiation Protocol
(SIP) |
| The Session Initiation
Protocol (SIP) is a signaling protocol used for establishing
sessions in an IP network. The main purpose of SIP is to help
session originators deliver invitations to potential session
participants, set up and tear down connections... |
| For more details please see
SessionInitiationProtocol(SIP).doc. |
|
RTP is the Internet-standard protocol
for the transport of real-time data, including audio and video.
Applications typically run RTP on top of UDP to make use of its
multiplexing and checksum services... |
| For more details please see
RealTimeProtocol(RTP).doc. |
|
This is the ITU-T’s
(International Telecommunications Union) standard that provides the
technical requirements for voice communication over LANs. The
standard encompasses both point to point communications and
multipoint conferences... |
| For more details please see
H323Standard.doc. |
3.4 Media
Gateway Control Protocol |
|
MGCP is a
control protocol (just like SIP), allowing a central operator
(generally, a server) to monitor and analyze events in IP phones and
gateways and instruct them to send media to specific addresses... |
| For more details please see
MediaGatewayControlProtocol(MGCP).doc
. |
| 3.5 Real Time Streaming
Protocol |
|
RTSP, the Real Time Streaming
Protocol, is a client-server protocol that provides control over the
delivery of real-time media streams. It provides "VCR-style" remote
control functionality for audio and video streams, like pause, fast
forward, reverse, and absolute positioning... |
| For more details please see
Real Time
Streaming Protocol(RTSP).doc. |
3.6
Resource Reservation Protocol |
|
The network delay and Quality of
Service are the most hindering factors in the voice-data
convergence. The most promising solution has been developed by IETF
viz., RSVP... |
| For more details please see
ResourceReservationProtocol(RSVP).doc. |
| 3.7 Session Announcement
protocol |
3.8
Session Description Protocol |
|
SDP is intended for describing
multimedia sessions for the purpose of session announcement, session
invitation etc... |
| For more details please see
SessionDescriptio
Protocol(SDP).doc. |
|
Hardware is the
fundamental element to build the VoIP network. This section shows
the basic hardware contained in the VoIP network. |
| For brief overview of these
hardware components please see
VOIPHardware.ppt. Fore more details please see
VoIP hardware.doc. |
|
When a voice conversation is passed
across a network using packet switching, the voice “data” becomes
the payload of a packet or frame, which is subsequently
packet-switched across the network according to the technology or
protocol used... |
| For brief overview of VoIP
network please see VOIPNetwork.ppt.
Fore more details please see
VOIPNetwork.doc. |
| 6. VoIP
Quality of
Service |
|
Since VoIP is a real-time service and is sensitive to delay and
packet loss, Quality of Service (Qos) becomes an important issue to
VoIP. |
| For brief overview of VoIP
Qos and VoIP security please see
VOIPQosandSecurity.ppt. Fore more details of VoIP Qos please see
VOIPQualityofService.doc. |
|
There are numerous threats to a VoIP
network. This leads VoIP another important issue, security issue. |
| For brief overview of VoIP
Qos and VoIP security please see
VOIPQosandSecurity.ppt. Fore more details of VoIP security
please see VoIPSecurity.doc . |
| 8.
End of Semester
Report and Presentation |
|
This end of semester report and
presentation gives the basic techniques needed to implement VoIP. |
| Please see
VOIPTermPresentation.ppt for
the end semester presentation and for end of semester report. please
see VoIPTermReport.doc. |
|
PLANET is on of VoIP product vendors.
This report search the whole network solutions for office use
include VoIP. |
| For brief overview please see
VoIPOfficeSolution.ppt. Fore
more details please see
PLANETsolutions.doc. |
|
The user brief is based on the
user side of view. When the users willing to have new services (such
as install new system or upgrade their system), they first provide a
list (User Brief) to the supplier to tell them what they want. Then
the supplier should try their best to get the users requirements.
This is the first step of doing a project. |
| The VoIP user brief shows
here VoIPUserBrief.doc. |
| 11.
VoIP Project
Brief Preparation |
| 13.
VoIP Product
Recommendation |
| 14. VoIP Vendors: Cisco,
3Com and Planet |
| 15. Project
Implementation Plan |
|
|
Welcome!
|