Yealink CP860 — настройка джиттер-буфера для IP конференц-телефонов [314/417]
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Administrator’s Guide for CP860 IP conference phones
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2. Select the desired value from the pull-down list of CNG.
3. Click Confirm to accept the change.
Jitter buffer is a shared data area where voice packets can be collected, stored, and
sent to the voice processor in even intervals. Jitter is a term indicating variations in
packet arrival time, which can occur because of network congestion, timing drift or
route changes. The jitter buffer, located at the receiving end of the voice connection,
intentionally delays the arriving packets so that the end user experiences a clear
connection with very little sound distortion. IP phones support two types of jitter buffers:
fixed and adaptive. A fixed jitter buffer adds the fixed delay to voice packets. You can
configure the delay time for the static jitter buffer on IP phones. An adaptive jitter buffer
is capable of adapting the changes in the network's delay. The range of the delay time
for the dynamic jitter buffer added to packets can be also configured on IP phones.
Procedure
Jitter buffer can be configured using the configuration files or locally.
Configuration File
y000000000037.cfg
Configure the mode of jitter
buffer and the delay time for
jitter buffer.
Parameters:
voice.jib.adaptive
voice.jib.min
voice.jib.max
voice.jib.normal
Local
Web User Interface
Configure the mode of jitter
buffer and the delay time for
Содержание
- Your opinions and comments to docsfeedback yealink com p.3
- We are striving to improve our documentation quality and we appreciate your feedback email p.3
- In this guide p.5
- Documentations p.5
- About this guide p.5
- Getting started 7 p.7
- Table of contents vii p.7
- Table of contents p.7
- Product overview 1 p.7
- About this guide v p.7
- Configuring basic features 39 p.8
- Configuring advanced features 79 p.9
- Configuring audio features 91 p.9
- Troubleshooting 35 p.10
- Resource files 23 p.10
- Configuring security features 05 p.10
- Index 03 p.11
- Appendix 49 p.11
- Voip principle p.13
- Product overview p.13
- User agent client uac p.14
- Sip components p.14
- User agent server uas p.15
- Introduction of cp860 ip conference phones p.15
- Physical features p.16
- Phone features p.16
- Key features of cp860 ip conference phones p.16
- Cp860 ip conference phone p.16
- Physical features of cp860 ip conference phones p.16
- Security p.17
- Network features p.17
- Management p.17
- Codecs and voice features p.17
- Getting started p.19
- Connecting the network and power p.19
- Connecting the ip phones p.19
- Connecting the optional extension microphones p.21
- Connecting the optional usb flash drive p.22
- Connecting the optional pc or mobile device p.22
- Initialization process overview p.23
- Verifying startup p.25
- Reading icons p.25
- Configuration methods p.26
- Central provisioning p.27
- Web user interface p.27
- Phone user interface p.27
- Mac oriented cfg file p.27
- Configuration files p.27
- Common cfg file p.27
- Supported provisioning protocols p.28
- Provisioning server p.28
- Setting up the provisioning server p.29
- Deploying phones from the provisioning server p.29
- Configuring basic network parameters p.30
- Dhcp option p.31
- Procedure p.32
- Details of configuration parameters p.33
- Configuring network parameters manually p.36
- Procedure p.37
- Details of configuration parameters p.37
- Half duplex p.42
- Configuring transmission methods of the internet port p.42
- Auto negotiation p.42
- Procedure p.43
- Full duplex p.43
- Details of configuration parameters p.43
- Upgrading firmware p.44
- Upgrade via web user interface p.45
- Upgrade firmware from the provisioning server p.45
- Procedure p.46
- Details of configuration parameters p.46
- Local directory p.51
- Auto redial p.51
- Live dialpad p.51
- Auto answer p.51
- Language p.51
- Anonymous call rejection p.51
- Key as send p.51
- Anonymous call p.51
- Hotline p.51
- Administrator password p.51
- Web server type p.51
- Early media p.51
- User password p.51
- Do not disturb p.51
- Time and date p.51
- Directory p.51
- This chapter provides information for making configuration changes for the following p.51
- Dial plan p.51
- Softkey layout p.51
- Contrast p.51
- Search source list in dialing p.51
- Configuring basic features p.51
- Ring workaround p.51
- Call waiting p.51
- Return code when refuse p.51
- Call log p.51
- Phone lock p.51
- Busy tone delay p.51
- Missed call log p.51
- Basic features p.51
- Logo customization p.51
- Backlight p.51
- Procedure p.52
- Contrast p.52
- Details of configuration parameters p.53
- Details of configuration parameters p.54
- Backlight p.54
- Procedure p.54
- Web server type p.55
- Procedure p.55
- Details of configuration parameters p.56
- User password p.58
- Procedure p.59
- Details of the configuration parameter p.59
- Procedure p.60
- Administrator password p.60
- Details of the configuration parameter p.61
- Procedure p.62
- Phone lock p.62
- Details of configuration parameters p.63
- Time zone p.66
- Time and date p.66
- Daylight saving time p.66
- Procedure p.67
- Details of configuration parameters p.68
- Loading language packs p.78
- Language p.78
- Specifying the language to use p.79
- Procedure p.79
- Details of the configuration parameter p.79
- Procedure p.80
- Details of configuration parameters p.80
- Procedure p.82
- Logo customization p.82
- Details of configuration parameters p.82
- Softkey layout p.84
- Procedure p.86
- Details of configuration parameters p.87
- Key as send p.89
- Procedure p.90
- Details of configuration parameters p.90
- Dial plan p.93
- Replace rule p.94
- Procedure p.94
- Details of configuration parameters p.94
- Procedure p.96
- Dial now p.96
- Delay time for dial now rule p.96
- Details of configuration parameters p.97
- Procedure p.99
- Area code p.99
- Details of configuration parameters p.100
- Procedure p.101
- Details of configuration parameters p.101
- Block out p.101
- Hotline p.102
- Procedure p.103
- Details of configuration parameters p.103
- Procedure p.105
- Directory p.105
- Details of the configuration parameter p.105
- Search source list in dialing p.106
- Procedure p.106
- Details of the configuration parameter p.107
- Procedure p.108
- Details of the configuration parameter p.108
- Call log p.108
- Procedure p.109
- Missed call log p.109
- Details of the configuration parameter p.110
- Details of the configuration parameter p.111
- Procedure p.111
- Local directory p.111
- Procedure p.113
- Live dialpad p.113
- Details of configuration parameters p.113
- Procedure p.115
- Call waiting p.115
- Details of configuration parameters p.116
- Procedure p.119
- Details of configuration parameters p.119
- Auto redial p.119
- Procedure p.121
- Auto answer p.121
- Details of configuration parameters p.122
- Anonymous call p.124
- Details of configuration parameters p.125
- Procedure p.125
- Procedure p.128
- Details of configuration parameters p.128
- Anonymous call rejection p.128
- Do not disturb p.130
- Return message when dnd p.131
- Procedure p.131
- Details of configuration parameters p.132
- Procedure p.135
- Busy tone delay p.135
- Return code when refuse p.136
- Details of the configuration parameter p.136
- Procedure p.137
- Details of the configuration parameter p.137
- Procedure p.138
- Early media p.138
- Ring workaround p.138
- Details of the configuration parameter p.139
- Use outbound proxy in dialog p.140
- Procedure p.140
- Details of the configuration parameter p.140
- Sip session timer p.141
- Procedure p.141
- Details of configuration parameters p.142
- Call hold p.143
- Procedure p.144
- Details of configuration parameters p.145
- Session timer p.148
- Procedure p.148
- Details of configuration parameters p.149
- Forward international p.151
- Call forward p.151
- Procedure p.151
- Details of configuration parameters p.152
- Call transfer p.159
- Procedure p.160
- Details of configuration parameters p.160
- Procedure p.162
- Network conference p.162
- Details of configuration parameters p.162
- Transfer on conference hang up p.164
- Procedure p.164
- Directed call pickup p.165
- Details of the configuration parameter p.165
- Procedure p.166
- Details of configuration parameters p.167
- Procedure p.169
- Group call pickup p.169
- Details of configuration parameters p.170
- Call return p.172
- Procedure p.173
- Details of configuration parameters p.173
- Procedure p.174
- Calling line identification presentation p.174
- Details of the configuration parameter p.175
- Procedure p.176
- Details of the configuration parameter p.176
- Connected line identification presentation p.176
- Rfc 2833 p.177
- Sip info p.178
- Procedure p.178
- Inband p.178
- Details of configuration parameters p.179
- Suppress dtmf display p.181
- Procedure p.182
- Details of configuration parameters p.182
- Transfer via dtmf p.184
- Procedure p.184
- Details of configuration parameters p.184
- Procedure p.186
- Outgoing intercom calls p.186
- Intercom p.186
- Details of configuration parameters p.186
- Incoming intercom calls p.187
- Procedure p.188
- Details of configuration parameters p.188
- Configuring advanced features p.191
- Advanced features p.191
- Action url p.191
- Action uri p.191
- X authentication p.191
- Tr 069 device management p.191
- This chapter provides information for making configuration changes for the following p.191
- Static dns cache p.191
- Server redundancy p.191
- Remote phone book p.191
- Quality of service p.191
- Network address translation p.191
- Multicast paging p.191
- Message waiting indicator p.191
- Ipv6 support p.191
- Distinctive ring tones p.191
- Distinctive ring tones p.192
- Procedure p.194
- Auto answer p.194
- Details of configuration parameters p.195
- Procedure p.200
- Details of configuration parameters p.200
- Remote phone book p.204
- Procedure p.205
- Details of configuration parameters p.206
- Procedure p.209
- Ldap attributes p.209
- Details of configuration parameters p.210
- Message waiting indicator p.216
- Procedure p.217
- Details of configuration parameters p.217
- Sending rtp stream p.220
- Procedure p.220
- Multicast paging p.220
- Details of the configuration parameter p.221
- Receiving rtp stream p.223
- Procedure p.224
- Details of configuration parameters p.224
- Action url p.227
- Procedure p.229
- Details of configuration parameters p.230
- Action uri p.242
- Procedure p.243
- Details of the configuration parameter p.243
- Server redundancy p.244
- Phone configuration for redundancy implementation p.245
- Procedure p.246
- Phone registration p.246
- Details of configuration parameters p.247
- Sip server domain name resolution p.251
- Outgoing call when the working server connection fails p.253
- Details of configuration parameters p.254
- Procedure p.254
- Static dns cache p.255
- Procedure p.256
- Details of configuration parameters p.256
- Lldp med media endpoint discovery p.263
- Procedure p.265
- Details of configuration parameters p.265
- Vlan discovery via dhcp p.267
- Procedure p.267
- Details of configuration parameters p.268
- Procedure p.272
- Details of configuration parameters p.273
- Quality of service p.274
- Sip qos p.276
- Procedure p.276
- Details of configuration parameters p.276
- Voice qos p.276
- Stun simple traversal of udp over nats p.278
- Procedure p.278
- Network address translation p.278
- Nat traversal p.278
- Details of configuration parameters p.279
- Procedure p.281
- Details of configuration parameters p.282
- X authentication p.284
- Procedure p.284
- Details of configuration parameters p.285
- Tr 069 device management p.290
- Procedure p.291
- Details of configuration parameters p.292
- Ipv6 support p.295
- Ipv6 address assignment method p.295
- Procedure p.296
- Details of configuration parameters p.296
- Configuring audio features p.303
- Audio codecs p.303
- Packetization time p.304
- Procedure p.305
- Details of the configuration parameter p.305
- Procedure p.310
- Details of the configuration parameter p.310
- Acoustic echo cancellation p.310
- Acoustic clarity technology p.310
- Voice activity detection p.311
- Procedure p.311
- Background noise suppression p.311
- Automatic gain control p.311
- Details of the configuration parameter p.312
- Procedure p.313
- Details of the configuration parameter p.313
- Comfort noise generation p.313
- Procedure p.314
- Jitter buffer p.314
- Details of configuration parameters p.315
- Transport layer security p.317
- Configuring security features p.317
- Rc4 md5 p.318
- Dhe rsa aes128 sha p.318
- Public key information in server key exchange message and concludes its part of the p.318
- Dhe dss rc4 sha p.318
- Idea cbc sha p.318
- Dhe dss aes128 sha p.318
- Exp1024 rc4 sha p.318
- Des cbc3 sha p.318
- Exp1024 rc4 md5 p.318
- Des cbc sha p.318
- Exp1024 dhe dss rc4 sha p.318
- Aes256 sha p.318
- Exp1024 dhe dss des cbc sha p.318
- Aes128 sha p.318
- Exp1024 des cbc sha p.318
- Administrator s guide for cp860 ip conference phones p.318
- Exp rc4 md5 p.318
- Exp edh rsa des cbc sha p.318
- Exp edh dss des cbc sha p.318
- Tls server to establish an encrypted communication channel p.318
- Exp des cbc sha p.318
- The following figure illustrates the tls messages exchanged between the ip phone and p.318
- Edh rsa des cbc3 sha p.318
- Step2 server responds with server hello message selecting the ssl options sends its p.318
- Edh rsa des cbc sha p.318
- Step1 ip phone sends client hello message proposing ssl options p.318
- Edh dss des cbc3 sha p.318
- Rc4 sha p.318
- Edh dss des cbc sha p.318
- Certificates p.319
- Procedure p.320
- Details of configuration parameters p.321
- Secure real time transport protocol p.326
- Procedure p.327
- Details of the configuration parameter p.327
- Encrypting configuration files p.328
- Procedure to encrypt configuration files p.329
- Procedure p.331
- Details of configuration parameters p.331
- Resource files p.335
- Replace rule template p.336
- Procedure p.336
- Procedure p.337
- Dial now template p.337
- Softkey layout template p.338
- Procedure p.338
- Directory template p.339
- Super search template p.340
- Procedure p.340
- Procedure p.341
- Procedure p.342
- Local contact file p.342
- Remote xml phone book p.343
- Procedure p.344
- Troubleshooting methods p.347
- Troubleshooting p.347
- Viewing log files p.347
- Procedure p.348
- Details of configuration parameters p.348
- Capturing packets p.352
- Procedure p.353
- Enabling the watch dog feature p.353
- Getting information from status indicators p.354
- Details of the configuration parameter p.354
- Why is the lcd screen blank p.355
- Troubleshooting solutions p.355
- Analyzing configuration files p.355
- Why doesn t the ip phone upgrade firmware successfully p.356
- Why doesn t the ip phone get an ip address p.356
- Why doesn t the ip phone display time and date correctly p.356
- How do i find the basic information of the ip phone p.356
- How to reboot the ip phone remotely p.357
- Why do i get poor sound quality during a call p.357
- What is the difference between user name register name and display name p.357
- What is the difference between a remote phone book and a local phone book p.357
- Why does the ip phone use dob format logo file instead of popular bmp jpg and so on p.358
- What will happen if i connect both poe cable and power adapter which has the higher priority p.358
- What is pnp p.358
- What is auto provisioning p.358
- Why doesn t the ip phone update the configuration p.359
- What do on code and off code mean p.359
- How to solve the ip conflict problem p.359
- How to reset your phone to factory configurations p.359
- How to restore the administrator password p.360
- Appendix a glossary p.361
- Appendix p.361
- Appendix b time zones p.363
- Appendix c configuring programable key p.365
- Appendix d sip session initiation protocol p.368
- Rfc 3262 reliability of provisional responses in the session initiation protocol p.369
- Rfc and internet draft support p.369
- Rfc 3261 sip session initiation protocol replacement for rfc 2543 p.369
- Rfc 3420 internet media type message sipfrag p.369
- Rfc 3087 control of service context using sip request uri p.369
- Rfc 3372 sip for telephones sip t context and architectures p.369
- Rfc 3361 dhcp for ipv4 option for sip servers p.369
- Rfc 2833 rtp payload for dtmf digits telephony tones and telephony signals p.369
- Rfc 3326 the reason header field for the session initiation protocol sip p.369
- Rfc 2806 urls for telephone calls p.369
- Rfc 3325 sip asserted identity p.369
- Rfc 2782 a dns rr for specifying the location of services dns srv p.369
- Rfc 3324 requirements for network asserted identity p.369
- Rfc 2617 http authentication basic and digest access authentication p.369
- Rfc 3323 a privacy mechanism for the session initiation protocol sip p.369
- Rfc 2616 hypertext transfer protocol http 1 p.369
- Rfc 3313 private sip extensions for media authorization p.369
- Rfc 2543 sip session initiation protocol p.369
- Rfc 3312 integration of resource management and sip p.369
- Rfc 2327 sdp session description protocol p.369
- Rfc 3311 the session initiation protocol sip update method p.369
- Rfc 2246 the tls protocol version 1 p.369
- Rfc 3310 http digest authentication using authentication and key agreement p.369
- Rfc 2112 multipart mime p.369
- Rfc 3266 support for ipv6 in session description protocol sdp p.369
- Rfc 1889 rtp media control p.369
- Rfc 3265 session initiation protocol sip specific event notification p.369
- Rfc 1321 the md5 message digest algorithm p.369
- Rfc 3264 an offer answer model with the session description protocol sdp p.369
- Appendix p.369
- The following rfc s and internet drafts are supported p.369
- Rfc 3263 session initiation protocol sip locating sip servers p.369
- Rfc2915 the naming authority pointer naptr dns resource record p.369
- Rfc 3959 the early session disposition type for sip p.370
- Rfc 3515 the session initiation protocol sip refer method p.370
- Rfc 3892 the session initiation protocol sip referred by mechanism p.370
- Rfc 3489 stun simple traversal of user datagram protocol udp through p.370
- Rfc 3891 the session initiation protocol sip replaces header p.370
- Rfc 3486 compressing the session initiation protocol sip p.370
- Rfc 3890 a transport independent bandwidth modifier for the sdp p.370
- Rfc 3455 private header p header extensions to the sip for the 3gpp p.370
- Rfc 3856 a presence event package for session initiation protocol sip p.370
- Rfc 3428 session initiation protocol sip extension for instant messaging p.370
- Rfc 3842 a message summary and message waiting indication event package p.370
- Registration p.370
- Rfc 3725 best current practices for third party call control 3pcc in the session p.370
- Protocol sip p.370
- Rfc 3711 the secure real time transport protocol srtp p.370
- Parameter registry for the session initiation protocol sip p.370
- Network address translators nats p.370
- Rfc 3702 authentication authorization and accounting requirements for the sip p.370
- Initiation protocol sip p.370
- Rfc3966 the tel uri for telephone number p.370
- Rfc 3680 sip event package for registrations p.370
- Identifier uri parameter registry for the session initiation protocol sip p.370
- Rfc 4244 an extension to the sip for request history information p.370
- Rfc 3666 sip public switched telephone network pstn call flows p.370
- For the session initiation protocol sip p.370
- Rfc 4235 an invite initiated dialog event package for the session initiation p.370
- Rfc 3665 session initiation protocol sip basic call flow examples p.370
- Bandwidth p.370
- Rfc 4028 session timers in the session initiation protocol sip p.370
- Rfc 3608 sip extension header field for service route discovery during p.370
- Administrator s guide for cp860 ip conference phones p.370
- Rfc 3969 the internet assigned number authority iana uniform resource p.370
- Rfc 3581 an extension to the sip for symmetric response routing p.370
- Rfc 3968 the internet assigned number authority iana header field p.370
- Rfc 3556 session description protocol sdp bandwidth modifiers for rtcp p.370
- Rfc 3960 early media and ringing tone generation in sip p.370
- Rfc 3550 rtp rtcp ietf rfc 3550 p.370
- Sip request p.371
- Sip header p.372
- Xx response information responses p.373
- Sip responses p.373
- Xx response successful responses p.374
- Xx response request failure responses p.374
- Xx response redirection responses p.374
- Xx response server failure responses p.375
- Appendix e sip call flows p.376
- Xx response global responses p.376
- Sip session description protocol sdp usage p.376
- F9 ack p.377
- F8 200 ok p.377
- F7 200 ok p.377
- F6 180 ringing p.377
- F5 180 ringing p.377
- F4 100 trying p.377
- F3 100 trying p.377
- F2 invite b p.377
- F14 200 ok p.377
- F13 200 ok p.377
- F12 bye p.377
- F11 bye p.377
- F10 ack p.377
- F1 invite b p.377
- Way rtp channel established p.377
- User a proxy server user b p.377
- Successful call setup and disconnect p.377
- Unsuccessful call setup called user is busy p.379
- User a proxy server user b p.380
- F8 ack p.380
- F7 ack p.380
- F6 486 busy here p.380
- F5 486 busy here p.380
- F4 100 trying p.380
- F3 100 trying p.380
- F2 invite b p.380
- F1 invite b p.380
- F3 180 ringing p.382
- F2 invite b p.382
- F1 invite b p.382
- User a proxy server user b p.382
- Unsuccessful call setup called user does not answer p.382
- F8 200 ok p.382
- F7 200 ok p.382
- F6 cancel p.382
- F5 cancel p.382
- F4 180 ringing p.382
- Successful call setup and call hold p.384
- Successful call setup and call waiting p.386
- Call transfer without consultation p.391
- User c answers the call p.392
- Call is established between user a and user c p.392
- Administrator s guide for cp860 ip conference phones p.392
- Call transfer with consultation p.395
- Call is established between user b and user c p.396
- Administrator s guide for cp860 ip conference phones p.396
- User a transfers the call to user c p.396
- Always call forward p.400
- User c answers the call p.401
- Call is established between user a and user c p.401
- Appendix p.401
- Busy call forward p.404
- No answer call forward p.407
- Call conference p.410
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