3. Outline
Aims
Objectives
Over View of VoIP Technology
VoIP Quality
VoIP Codecs
4. Time Table
April May June July
Project proposal
submission 10%
Collection and study
30%
information
Analysis 35%
0.5%
Simulation
Write Report 0.5%
85%
6. Objective
VoIP voice quality
Codecs
VoIP Equipment
Phone Frequencies
Bandwidth requirements
Setup simulation
7. VoIP (Voice over Internet Protocol).
Sometimes referred to as Internet telephony.
Sending voice information as digital form in discrete packets.
8. VoIP Protocol Description
H.323 ITU standard protocol for interactive conference.
MGCP (IETF) standard for PSTN gateway control.
SIP IETF standard protocol for interactive and no interactive
conference .
RTP IETF standard protocol for media streaming .
RTCP IETF standard that provides out-of-band control information
for an RTP flow .
ISDN (Integrated Services Network). RTP (Real Time Protocol).
IETF (Internet Engineering Task Force ). RTCP (Real Time Control Protocol).
MGCP(Media Gateway Control Protocol). ISDN (Integrated Services Digital
SIP(Session Initiation Protocol ) Network )
9. OSI Layer VoIP Protocols
Application Softphone/Call Manager/Human Speech
Presentation Codecs
Session H.323/SIP/MGCP
Transport RTP/UDP (Media) ; TCP/UDP (Signal)
Network Internet Protocol (IP)
Data link FR, ATM, MLPPP, PPP and HDLC
Physical ….
FR (Frame Relay). RTP (Real Time Protocol) .
PPP (Point to Point Protocol). ATM (Asynchronous Transfer Mode).
MLPPP(Multi Link PPP).
Codecs (Coding decoding).
HDLC (High-Level Data Link Control).
10. Low cost.
Use existing infrastructure.
Call forwarding.
Voice mail and fax applications.
Call waiting.
Caller ID.
Send documents and/or pictures while you
talk at the same time.
11. Sound quality and reliability .
Lack of continuous service during a power outage.
Emergency calls (911) (Problem of locating call).
Vulnerable to same attacks as IP data networks
Viruses, Worms and spams .
Packet loss.
12. Packet Loss.
Loss of packets severely degrades the voice application.
Network packets loss (as a result of congestion or rerouting in the IP network).
Late arrival loss (dropped at receiver).
Link failures or system errors.
End-to-end Delay.
Transmission and queuing delay.
VoIP Typically tolerates delay up to 150ms before the quality of the call degrades .
Codec processing delay .
Packetizing/depacketizing delay.
Jitter (delay variation).
Caused by queuing delay within the IP network.
Instantaneous buffer use causes delay variation in the same voice stream .
13. Sender Receiver
De- Jitter
Encoder Packetizer IP Network Decoder
packetizer buffer
coding distortion delay packet loss codec
delay buffer-delay impairment
codec delay network delay buffer-loss delay
jitter
Other impairments: echo, sidetone, background noise
MOS (Mean Opinion Score).
14. Codec is a process of digitizing the voice sample , or converting
digitized signal into an analog signal.
Each VoIP equipment must implement Codec in order to
implement VoIP.
15. Codec Bandwidth/kbps Comments
G.711 Delivers precise speech transmission. Very low processor
requirements. Needs at least 128 kbps for two-way.
G.722 Adapts to varying compressions and bandwidth is
conserved with network congestion.
G.723 5.3/6.3 High compression with high quality audio. Can use with
dial-up. Lot of processor power.
G.726 16/24/32/40 An improved version of G.723 .
G.729 8 Excellent bandwidth utilization. Error tolerant. License
required.
GSM 13 High compression ratio. Free and available in many
hardware and software platforms. Same encoding is used
in GSM cellphones (improved versions are often used
nowadays).
