Avaya IP Telephony Konfigurationsanleitung Seite 42
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42 Chapter 3 Installing IP telephones
P0993474 03
4 Go to “Installing i-series telephones” on page 43.
Choosing a codec
The default codec is used when an IP client has not been configured to use a preferred Codec.
Refer to 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 makes a call. For example, if both endpoints of the call are IP 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-law 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. Choose the G.711 CODEC with the
companding law (alaw or ulaw) that matches your system requirements.
• 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.
• Codecs with VAD (Voice Activity Detection) make VAD active on the system, which
performs the same function as having silence suppression active.
Jitter Buffer None
Auto
Small
Medium
Large
Choose one of these settings to change the default jitter buffer size:
• 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
For information about choosing a Jitter Buffer, refer to “Choosing a
Jitter Buffer” on page 43.
Note: If the IP telephones are using VoIP trunks for the call, the codec set for the trunks
overrides the telephone settings.
Table 9 IP terminals general record fields (Continued)
Field Value Description
- Business Communications 1
- Manager 3.0 1
- IP Telephony Configuration 1
- Trademarks 2
- Table of Contents 3
- Chapter 4 5
- Chapter 5 5
- 12 Tables 12
- Before you begin 13
- Symbols used in this guide 13
- Text conventions 14
- Acronyms 14
- Preface 15 15
- Related publications 16
- How to get help 16
- APAC (Asia Pacific) 17
- Technical Support - CTAS 17
- 18 Preface 18
- Chapter 1 19
- Introduction 19
- IP telephones and VoIP trunks 20
- Figure 1 Network diagram 22
- Telephones 23
- Gatekeepers on the network 24
- IP network 24
- Key IP telephony concepts 25
- Jitter Buffer 26
- QoS routing 27
- Chapter 2 29
- Prerequisites checklist 29
- Network devices 30
- Network assessment 30
- Resource assessment 31
- Keycodes 33
- Defining published IP address 35
- NetVision wireless telephones 37
- IP telephones 37
- Chapter 3 39
- Installing IP telephones 39
- 2 Select the General tab 41
- Choosing a codec 42
- Choosing a Jitter Buffer 43
- Before installing 43
- Using a 3-port switch 43
- to move forward 46
- Other troubleshooting tips 48
- Configuring DHCP 49
- IP telephony DHCP notes 50
- Checking IP server status 52
- Figure 7 IP Terminal status 53
- Using the Hot Desking feature 57
- Notes about Hot Desking 58
- Customizing feature labels 59
- Moving IP telephones 63
- Figure 15 i2050 Switch type 65
- Configuring NetVision records 68
- <http:/ 69
- /symbol.com> 69
- Testing the handset functions 71
- Changing a handset Name 72
- Configuring VoIP trunks 73
- Configuring media parameters 74
- Field Value Description 75
- Outgoing call configuration 76
- Figure 18 Trunk/Line data 77
- Configuring a remote gateway 78
- Configuring PSTN fallback 80
- Enabling PSTN fallback 81
- Setting up the VoIP schedule 82
- Configuring routes 82
- Figure 23 Add route dialog 83
- Figure 24 VoIP schedule 85
- Activating the VoIP schedule 87
- Turning on QoS monitor 87
- PSTN fallback metrics 88
- Incoming call configuration 89
- Making calls 95
- Connecting an i200X telephone 96
- Quality of Service Monitor 100
- Port settings 101
- P0993474 03 102
- Figure 33 Port Ranges 102
- Using a gatekeeper 103
- Notes about NPI-TON aliases 107
- Gatekeeper call scenarios 113
- Chapter 6 115
- WarehouseHead Office 116
- Appendix A 121
- Efficient Networking 121
- Network engineering 122
- LAN engineering examples 126
- WAN engineering 126
- (8 kb/s) 127
- Setting Non-linear processing 128
- Enough link capacity 130
- Not enough link capacity 130
- Further network analysis 131
- Reduce link delay 132
- Reducing hop count 132
- Adjust the jitter buffer size 133
- Reduce packet errors 133
- Routing issues 134
- Appendix B 135
- Silence compression 135
- Time -> 137
- Bandwidth -> 137
- Channel/Link Max 137
- Comfort noise 140
- Appendix C 141
- Network performance utilities 141
- Appendix D 143
- Interoperability 143
- Speech path setup methods 144
- Media path redirection 144
- Gatekeeper 145
- No feedback busy station 146
- Interoperability 147 147
- Using VLAN on the network 148
- Interoperability 149 149
- Symbol NetVision telephones 150
- Interoperability 151 151
- 152 Interoperability 152
- Appendix E 153
- Quality of Service 153
- 154 Quality of Service 154
- Measuring Intranet QoS 155
- Recording routes 156
- Adjusting Ping measurements 157
- Measurement procedure 158
- Quality of Service 159 159
- Traffic mix 160
- TCP traffic behavior 160
- Network Quality of Service 161
- Network monitoring 162
- Quality of Service parameters 162
- Fallback to PSTN 163
- 164 Quality of Service 164
- Glossary 165
- 166 Glossary 166
- Glossary 167 167
- 168 Glossary 168
- Glossary 169 169
- 170 Glossary 170
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