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Chapter 3 Installing IP Telephones 33
IP Telephony Configuration Guide
6 From the Codec menu, select a default Codec, or leave the Default Codec at Auto. This is the
Codec that is used if an IP telephone has not been configured with a preferred codec. For
information on this, see “Choosing a codec” on page 33.
7 From the Jitter Buffer menu, select a Jitter Buffer level. For information on choosing a Jitter
Buffer, see “Choosing a Jitter Buffer” on page 33.
Choosing a codec
The default codec is used when an IP client has not been configured to use a preferred Codec (see
the next section for individual IP client Codec settings). If the default Codec is set to AUTO, the
Business Communications Manager will choose the appropriate CODEC when an IP client goes
on a call. For example, if both endpoints of the call are I20XX telephones on the same subnet, the
Business Communications Manager chooses G.711 for maximum voice quality. If the telephones
are on different subnets, the Business Communications Manager will choose G.729 to minimize
network bandwidth consumption by voice data packets.
For IP telephones, the Business Communications Manager supports both A and MU law variants
of the G.711 CODEC, as well as the G.729 and G.723 CODECS.
• The G.711 CODEC samples the voice stream at a rate of 64Kbps (Kilo bits per second), and is
the CODEC to use for maximum voice quality.
• The G.729 CODEC samples the voice stream at 8Kbps. The voice quality is slightly lower
using a G.729 but it reduces network traffic by approximately 80%.
• The G.723 CODEC should be used only with third party devices that do not support G.729 or
G.711.
Choosing a Jitter Buffer
A jitter buffer is used to prevent the jitter associated with arriving (Rx) voice packets at the IP
telephones. The jitter is caused by packets arriving out of order due to having used different
network paths, and varying arrival rates of consecutive voice packets.The greater the size of the
jitter buffer, the "better sounding" the received voice is. However, voice latency (delay) also
increases. Latency is very problematic for telephone calls, as it increases the time between when
one user speaks and the user at the other end hears the voice. The administrator can adjust the
default jitter buffer size to the following values:
• NONE: Minimal latency, best for short-haul networks with good bandwidth.
• AUTO: Business Communications Manager will dynamically adjust the size.
• SMALL: Business Communications Manager will adjust the buffer size, depending on
CODEC type and number of frames per packet to introduce a 60 millisecond
delay.
• MEDIUM: 120 millisecond delay
• LARGE: 180 millisecond delay
- Business Communications 1
- Manager 2.5 1
- IP Telephony Configuration 1
- Trademarks 2
- Contents 3
- Chapter 4 4
- Chapter 5 5
- Appendix A 5
- Before you begin 13
- Symbols used in this guide 14
- Text conventions 14
- Acronyms 15
- UDP User Datagram Protocol 16
- WAN Wide Area Network 16
- Related publications 17
- How to get help 18
- Chapter 1 19
- Introduction 19
- The IP telephony network 20
- Chapter 1 Introduction 21 21
- Figure 1 Network diagram 21
- Telephones 22
- IP Network 22
- Key IP Telephony Concepts 23
- Jitter Buffer 24
- QoS routing 24
- Chapter 1 Introduction 25 25
- 26 Chapter 1 Introduction 26
- Chapter 2 27
- Prerequisites checklist 27
- Network assessment 28
- Resource Assessment 28
- Keycodes 29
- IP Telephones 30
- Chapter 3 31
- Installing IP Telephones 31
- Choosing a codec 33
- Choosing a Jitter Buffer 33
- Before Installing 34
- Manual Cfg DHCP(0 no, 1 yes) 35
- Configuring DHCP 38
- Figure 4 DHCP summary 39
- Figure 5 DHCP range 40
- Figure 6 IP Terminal status 41
- Figure 7 Configuration menu 42
- Figure 9 IP Terminal Status 43
- Download Pending 44
- Deregistering offline DNs 45
- Moving IP telephones 46
- H.323 devices 47
- 4 Click the Save button 49
- VoIP Trunk Configuration 51
- Configuring media parameters 53
- Configuring codecs 54
- Setting Silence Compression 54
- Setting Jitter Buffers 55
- Outgoing call configuration 56
- Figure 21 Trunk/Line data 57
- Figure 22 Line pool 58
- Configuring a remote gateway 59
- Configuring PSTN fallback 60
- Setting up a VoIP schedule 61
- Vo I P 62
- Configuring routes 63
- Figure 28 Route list 64
- Figure 29 Add route dialog 64
- Creating a destination code 65
- Figure 32 Normal schedule 66
- Configuring digits 67
- Activating the VoIP schedule 67
- Turning on QoS monitor 68
- Incoming call configuration 70
- Example configuration 71
- Making calls 74
- Connecting an i200X telephone 75
- Quality of Service Monitor 77
- Port settings 77
- Using firewalls 78
- Using a gatekeeper 80
- Gatekeeper call scenarios 82
- Typical applications 85
- MCDN functionality 86
- Networking 91
- Network Engineering 92
- Networking 93 93
- (64 kb/s) 93
- (5.3 kb/s) 93
- LAN Engineering Examples 95
- WAN engineering 96
- Setting Non-linear processing 96
- Networking 97 97
- Traceroute 97
- Enough link capacity 98
- Not enough link capacity 98
- Implementing the network 99
- Further network analysis 99
- Reduce link delay 100
- Reducing hop count 100
- Adjust the jitter buffer size 101
- Reduce packet errors 101
- Routing issues 102
- Silence compression 103
- Appendix B 103
- 104 Silence compression 104
- P0937663 02.0 104
- Silence compression 105 105
- 106 Silence compression 106
- Silence compression 107 107
- Comfort Noise 108
- Appendix C 109
- Network Performance Utilities 109
- Appendix D 111
- Interoperability 111
- Speech Path Setup Methods 112
- Media Path Redirection 112
- Gatekeeper 113
- Symbol Telephones 114
- Appendix E 115
- Quality of Service 115
- Measuring Intranet QoS 116
- Recording routes 117
- Adjustment for processing 118
- Late packets 118
- Quality of Service 119 119
- Traffic mix 120
- Network Quality of Service 121
- Packet loss 122
- Packet delay 122
- Delay variation (jitter) 123
- 124 Quality of Service 124
- Glossary 125
- 126 Glossary 126
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