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Yealink CP860 [304/417] Packetization time
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Administrator’s Guide for CP860 IP conference phones
292
Codec
Algorithm
Reference
Bit Rate
Sample
Rate
Packetization
Time
15.2 Kbps
Packetization Time
Ptime (Packetization Time) is a measurement of the duration (in milliseconds) of the
audio data in each RTP packet sent to the destination, and defines how much network
bandwidth is used for the RTP stream transfer. Before establishing a conversation, codec
and ptime are negotiated through SIP signaling. The valid values of ptime range from
10 to 60, in increments of 10 milliseconds. The default ptime is 20ms. You can also
disable the ptime negotiation.
The attribute “rtpmap” is used to define a mapping from RTP payload codes to a codec,
clock rate and other encoding parameters.
The corresponding attributes of the codec are listed as follows:
Codec
Configuration Methods
Priority
RTPmap
G722
Configuration Files
Web User Interface
1
9
PCMU
Configuration Files
Web User Interface
2
0
PCMA
Configuration Files
Web User Interface
3
8
G729
Configuration Files
Web User Interface
4
18
G723_53
Configuration Files
Web User Interface
0
4
G723_63
Configuration Files
Web User Interface
0
4
G726-16
Configuration Files
Web User Interface
0
103
G726-24
Configuration Files
Web User Interface
0
104
G726-32
Configuration Files
Web User Interface
0
102
G726-40
Configuration Files
Web User Interface
0
105
Содержание
- We are striving to improve our documentation quality and we appreciate your feedback email 3
- Your opinions and comments to docsfeedback yealink com 3
- About this guide 5
- Documentations 5
- In this guide 5
- About this guide v 7
- Getting started 7 7
- Product overview 1 7
- Table of contents 7
- Table of contents vii 7
- Configuring basic features 39 8
- Configuring advanced features 79 9
- Configuring audio features 91 9
- Configuring security features 05 10
- Resource files 23 10
- Troubleshooting 35 10
- Appendix 49 11
- Index 03 11
- Product overview 13
- Voip principle 13
- Sip components 14
- User agent client uac 14
- Introduction of cp860 ip conference phones 15
- User agent server uas 15
- Cp860 ip conference phone 16
- Key features of cp860 ip conference phones 16
- Phone features 16
- Physical features 16
- Physical features of cp860 ip conference phones 16
- Codecs and voice features 17
- Management 17
- Network features 17
- Security 17
- Connecting the ip phones 19
- Connecting the network and power 19
- Getting started 19
- Connecting the optional extension microphones 21
- Connecting the optional pc or mobile device 22
- Connecting the optional usb flash drive 22
- Initialization process overview 23
- Reading icons 25
- Verifying startup 25
- Configuration methods 26
- Central provisioning 27
- Common cfg file 27
- Configuration files 27
- Mac oriented cfg file 27
- Phone user interface 27
- Web user interface 27
- Provisioning server 28
- Supported provisioning protocols 28
- Deploying phones from the provisioning server 29
- Setting up the provisioning server 29
- Configuring basic network parameters 30
- Dhcp option 31
- Procedure 32
- Details of configuration parameters 33
- Configuring network parameters manually 36
- Details of configuration parameters 37
- Procedure 37
- Auto negotiation 42
- Configuring transmission methods of the internet port 42
- Half duplex 42
- Details of configuration parameters 43
- Full duplex 43
- Procedure 43
- Upgrading firmware 44
- Upgrade firmware from the provisioning server 45
- Upgrade via web user interface 45
- Details of configuration parameters 46
- Procedure 46
- Administrator password 51
- Anonymous call 51
- Anonymous call rejection 51
- Auto answer 51
- Auto redial 51
- Backlight 51
- Basic features 51
- Busy tone delay 51
- Call log 51
- Call waiting 51
- Configuring basic features 51
- Contrast 51
- Dial plan 51
- Directory 51
- Do not disturb 51
- Early media 51
- Hotline 51
- Key as send 51
- Language 51
- Live dialpad 51
- Local directory 51
- Logo customization 51
- Missed call log 51
- Phone lock 51
- Return code when refuse 51
- Ring workaround 51
- Search source list in dialing 51
- Softkey layout 51
- This chapter provides information for making configuration changes for the following 51
- Time and date 51
- User password 51
- Web server type 51
- Contrast 52
- Procedure 52
- Details of configuration parameters 53
- Backlight 54
- Details of configuration parameters 54
- Procedure 54
- Procedure 55
- Web server type 55
- Details of configuration parameters 56
- User password 58
- Details of the configuration parameter 59
- Procedure 59
- Administrator password 60
- Procedure 60
- Details of the configuration parameter 61
- Phone lock 62
- Procedure 62
- Details of configuration parameters 63
- Daylight saving time 66
- Time and date 66
- Time zone 66
- Procedure 67
- Details of configuration parameters 68
- Language 78
- Loading language packs 78
- Details of the configuration parameter 79
- Procedure 79
- Specifying the language to use 79
- Details of configuration parameters 80
- Procedure 80
- Details of configuration parameters 82
- Logo customization 82
- Procedure 82
- Softkey layout 84
- Procedure 86
- Details of configuration parameters 87
- Key as send 89
- Details of configuration parameters 90
- Procedure 90
- Dial plan 93
- Details of configuration parameters 94
- Procedure 94
- Replace rule 94
- Delay time for dial now rule 96
- Dial now 96
- Procedure 96
- Details of configuration parameters 97
- Area code 99
- Procedure 99
- Details of configuration parameters 100
- Block out 101
- Details of configuration parameters 101
- Procedure 101
- Hotline 102
- Details of configuration parameters 103
- Procedure 103
- Details of the configuration parameter 105
- Directory 105
- Procedure 105
- Procedure 106
- Search source list in dialing 106
- Details of the configuration parameter 107
- Call log 108
- Details of the configuration parameter 108
- Procedure 108
- Missed call log 109
- Procedure 109
- Details of the configuration parameter 110
- Details of the configuration parameter 111
- Local directory 111
- Procedure 111
- Details of configuration parameters 113
- Live dialpad 113
- Procedure 113
- Call waiting 115
- Procedure 115
- Details of configuration parameters 116
- Auto redial 119
- Details of configuration parameters 119
- Procedure 119
- Auto answer 121
- Procedure 121
- Details of configuration parameters 122
- Anonymous call 124
- Details of configuration parameters 125
- Procedure 125
- Anonymous call rejection 128
- Details of configuration parameters 128
- Procedure 128
- Do not disturb 130
- Procedure 131
- Return message when dnd 131
- Details of configuration parameters 132
- Busy tone delay 135
- Procedure 135
- Details of the configuration parameter 136
- Return code when refuse 136
- Details of the configuration parameter 137
- Procedure 137
- Early media 138
- Procedure 138
- Ring workaround 138
- Details of the configuration parameter 139
- Details of the configuration parameter 140
- Procedure 140
- Use outbound proxy in dialog 140
- Procedure 141
- Sip session timer 141
- Details of configuration parameters 142
- Call hold 143
- Procedure 144
- Details of configuration parameters 145
- Procedure 148
- Session timer 148
- Details of configuration parameters 149
- Call forward 151
- Forward international 151
- Procedure 151
- Details of configuration parameters 152
- Call transfer 159
- Details of configuration parameters 160
- Procedure 160
- Details of configuration parameters 162
- Network conference 162
- Procedure 162
- Procedure 164
- Transfer on conference hang up 164
- Details of the configuration parameter 165
- Directed call pickup 165
- Procedure 166
- Details of configuration parameters 167
- Group call pickup 169
- Procedure 169
- Details of configuration parameters 170
- Call return 172
- Details of configuration parameters 173
- Procedure 173
- Calling line identification presentation 174
- Procedure 174
- Details of the configuration parameter 175
- Connected line identification presentation 176
- Details of the configuration parameter 176
- Procedure 176
- Rfc 2833 177
- Inband 178
- Procedure 178
- Sip info 178
- Details of configuration parameters 179
- Suppress dtmf display 181
- Details of configuration parameters 182
- Procedure 182
- Details of configuration parameters 184
- Procedure 184
- Transfer via dtmf 184
- Details of configuration parameters 186
- Intercom 186
- Outgoing intercom calls 186
- Procedure 186
- Incoming intercom calls 187
- Details of configuration parameters 188
- Procedure 188
- Action uri 191
- Action url 191
- Advanced features 191
- Configuring advanced features 191
- Distinctive ring tones 191
- Ipv6 support 191
- Message waiting indicator 191
- Multicast paging 191
- Network address translation 191
- Quality of service 191
- Remote phone book 191
- Server redundancy 191
- Static dns cache 191
- This chapter provides information for making configuration changes for the following 191
- Tr 069 device management 191
- X authentication 191
- Distinctive ring tones 192
- Auto answer 194
- Procedure 194
- Details of configuration parameters 195
- Details of configuration parameters 200
- Procedure 200
- Remote phone book 204
- Procedure 205
- Details of configuration parameters 206
- Ldap attributes 209
- Procedure 209
- Details of configuration parameters 210
- Message waiting indicator 216
- Details of configuration parameters 217
- Procedure 217
- Multicast paging 220
- Procedure 220
- Sending rtp stream 220
- Details of the configuration parameter 221
- Receiving rtp stream 223
- Details of configuration parameters 224
- Procedure 224
- Action url 227
- Procedure 229
- Details of configuration parameters 230
- Action uri 242
- Details of the configuration parameter 243
- Procedure 243
- Server redundancy 244
- Phone configuration for redundancy implementation 245
- Phone registration 246
- Procedure 246
- Details of configuration parameters 247
- Sip server domain name resolution 251
- Outgoing call when the working server connection fails 253
- Details of configuration parameters 254
- Procedure 254
- Static dns cache 255
- Details of configuration parameters 256
- Procedure 256
- Lldp med media endpoint discovery 263
- Details of configuration parameters 265
- Procedure 265
- Procedure 267
- Vlan discovery via dhcp 267
- Details of configuration parameters 268
- Procedure 272
- Details of configuration parameters 273
- Quality of service 274
- Details of configuration parameters 276
- Procedure 276
- Sip qos 276
- Voice qos 276
- Nat traversal 278
- Network address translation 278
- Procedure 278
- Stun simple traversal of udp over nats 278
- Details of configuration parameters 279
- Procedure 281
- Details of configuration parameters 282
- Procedure 284
- X authentication 284
- Details of configuration parameters 285
- Tr 069 device management 290
- Procedure 291
- Details of configuration parameters 292
- Ipv6 address assignment method 295
- Ipv6 support 295
- Details of configuration parameters 296
- Procedure 296
- Audio codecs 303
- Configuring audio features 303
- Packetization time 304
- Details of the configuration parameter 305
- Procedure 305
- Acoustic clarity technology 310
- Acoustic echo cancellation 310
- Details of the configuration parameter 310
- Procedure 310
- Automatic gain control 311
- Background noise suppression 311
- Procedure 311
- Voice activity detection 311
- Details of the configuration parameter 312
- Comfort noise generation 313
- Details of the configuration parameter 313
- Procedure 313
- Jitter buffer 314
- Procedure 314
- Details of configuration parameters 315
- Configuring security features 317
- Transport layer security 317
- Administrator s guide for cp860 ip conference phones 318
- Aes128 sha 318
- Aes256 sha 318
- Des cbc sha 318
- Des cbc3 sha 318
- Dhe dss aes128 sha 318
- Dhe dss rc4 sha 318
- Dhe rsa aes128 sha 318
- Edh dss des cbc sha 318
- Edh dss des cbc3 sha 318
- Edh rsa des cbc sha 318
- Edh rsa des cbc3 sha 318
- Exp des cbc sha 318
- Exp edh dss des cbc sha 318
- Exp edh rsa des cbc sha 318
- Exp rc4 md5 318
- Exp1024 des cbc sha 318
- Exp1024 dhe dss des cbc sha 318
- Exp1024 dhe dss rc4 sha 318
- Exp1024 rc4 md5 318
- Exp1024 rc4 sha 318
- Idea cbc sha 318
- Public key information in server key exchange message and concludes its part of the 318
- Rc4 md5 318
- Rc4 sha 318
- Step1 ip phone sends client hello message proposing ssl options 318
- Step2 server responds with server hello message selecting the ssl options sends its 318
- The following figure illustrates the tls messages exchanged between the ip phone and 318
- Tls server to establish an encrypted communication channel 318
- Certificates 319
- Procedure 320
- Details of configuration parameters 321
- Secure real time transport protocol 326
- Details of the configuration parameter 327
- Procedure 327
- Encrypting configuration files 328
- Procedure to encrypt configuration files 329
- Details of configuration parameters 331
- Procedure 331
- Resource files 335
- Procedure 336
- Replace rule template 336
- Dial now template 337
- Procedure 337
- Procedure 338
- Softkey layout template 338
- Directory template 339
- Procedure 340
- Super search template 340
- Procedure 341
- Local contact file 342
- Procedure 342
- Remote xml phone book 343
- Procedure 344
- Troubleshooting 347
- Troubleshooting methods 347
- Viewing log files 347
- Details of configuration parameters 348
- Procedure 348
- Capturing packets 352
- Enabling the watch dog feature 353
- Procedure 353
- Details of the configuration parameter 354
- Getting information from status indicators 354
- Analyzing configuration files 355
- Troubleshooting solutions 355
- Why is the lcd screen blank 355
- How do i find the basic information of the ip phone 356
- Why doesn t the ip phone display time and date correctly 356
- Why doesn t the ip phone get an ip address 356
- Why doesn t the ip phone upgrade firmware successfully 356
- How to reboot the ip phone remotely 357
- What is the difference between a remote phone book and a local phone book 357
- What is the difference between user name register name and display name 357
- Why do i get poor sound quality during a call 357
- What is auto provisioning 358
- What is pnp 358
- What will happen if i connect both poe cable and power adapter which has the higher priority 358
- Why does the ip phone use dob format logo file instead of popular bmp jpg and so on 358
- How to reset your phone to factory configurations 359
- How to solve the ip conflict problem 359
- What do on code and off code mean 359
- Why doesn t the ip phone update the configuration 359
- How to restore the administrator password 360
- Appendix 361
- Appendix a glossary 361
- Appendix b time zones 363
- Appendix c configuring programable key 365
- Appendix d sip session initiation protocol 368
- Appendix 369
- Rfc 1321 the md5 message digest algorithm 369
- Rfc 1889 rtp media control 369
- Rfc 2112 multipart mime 369
- Rfc 2246 the tls protocol version 1 369
- Rfc 2327 sdp session description protocol 369
- Rfc 2543 sip session initiation protocol 369
- Rfc 2616 hypertext transfer protocol http 1 369
- Rfc 2617 http authentication basic and digest access authentication 369
- Rfc 2782 a dns rr for specifying the location of services dns srv 369
- Rfc 2806 urls for telephone calls 369
- Rfc 2833 rtp payload for dtmf digits telephony tones and telephony signals 369
- Rfc 3087 control of service context using sip request uri 369
- Rfc 3261 sip session initiation protocol replacement for rfc 2543 369
- Rfc 3262 reliability of provisional responses in the session initiation protocol 369
- Rfc 3263 session initiation protocol sip locating sip servers 369
- Rfc 3264 an offer answer model with the session description protocol sdp 369
- Rfc 3265 session initiation protocol sip specific event notification 369
- Rfc 3266 support for ipv6 in session description protocol sdp 369
- Rfc 3310 http digest authentication using authentication and key agreement 369
- Rfc 3311 the session initiation protocol sip update method 369
- Rfc 3312 integration of resource management and sip 369
- Rfc 3313 private sip extensions for media authorization 369
- Rfc 3323 a privacy mechanism for the session initiation protocol sip 369
- Rfc 3324 requirements for network asserted identity 369
- Rfc 3325 sip asserted identity 369
- Rfc 3326 the reason header field for the session initiation protocol sip 369
- Rfc 3361 dhcp for ipv4 option for sip servers 369
- Rfc 3372 sip for telephones sip t context and architectures 369
- Rfc 3420 internet media type message sipfrag 369
- Rfc and internet draft support 369
- Rfc2915 the naming authority pointer naptr dns resource record 369
- The following rfc s and internet drafts are supported 369
- Administrator s guide for cp860 ip conference phones 370
- Bandwidth 370
- For the session initiation protocol sip 370
- Identifier uri parameter registry for the session initiation protocol sip 370
- Initiation protocol sip 370
- Network address translators nats 370
- Parameter registry for the session initiation protocol sip 370
- Protocol sip 370
- Registration 370
- Rfc 3428 session initiation protocol sip extension for instant messaging 370
- Rfc 3455 private header p header extensions to the sip for the 3gpp 370
- Rfc 3486 compressing the session initiation protocol sip 370
- Rfc 3489 stun simple traversal of user datagram protocol udp through 370
- Rfc 3515 the session initiation protocol sip refer method 370
- Rfc 3550 rtp rtcp ietf rfc 3550 370
- Rfc 3556 session description protocol sdp bandwidth modifiers for rtcp 370
- Rfc 3581 an extension to the sip for symmetric response routing 370
- Rfc 3608 sip extension header field for service route discovery during 370
- Rfc 3665 session initiation protocol sip basic call flow examples 370
- Rfc 3666 sip public switched telephone network pstn call flows 370
- Rfc 3680 sip event package for registrations 370
- Rfc 3702 authentication authorization and accounting requirements for the sip 370
- Rfc 3711 the secure real time transport protocol srtp 370
- Rfc 3725 best current practices for third party call control 3pcc in the session 370
- Rfc 3842 a message summary and message waiting indication event package 370
- Rfc 3856 a presence event package for session initiation protocol sip 370
- Rfc 3890 a transport independent bandwidth modifier for the sdp 370
- Rfc 3891 the session initiation protocol sip replaces header 370
- Rfc 3892 the session initiation protocol sip referred by mechanism 370
- Rfc 3959 the early session disposition type for sip 370
- Rfc 3960 early media and ringing tone generation in sip 370
- Rfc 3968 the internet assigned number authority iana header field 370
- Rfc 3969 the internet assigned number authority iana uniform resource 370
- Rfc 4028 session timers in the session initiation protocol sip 370
- Rfc 4235 an invite initiated dialog event package for the session initiation 370
- Rfc 4244 an extension to the sip for request history information 370
- Rfc3966 the tel uri for telephone number 370
- Sip request 371
- Sip header 372
- Sip responses 373
- Xx response information responses 373
- Xx response redirection responses 374
- Xx response request failure responses 374
- Xx response successful responses 374
- Xx response server failure responses 375
- Appendix e sip call flows 376
- Sip session description protocol sdp usage 376
- Xx response global responses 376
- F1 invite b 377
- F10 ack 377
- F11 bye 377
- F12 bye 377
- F13 200 ok 377
- F14 200 ok 377
- F2 invite b 377
- F3 100 trying 377
- F4 100 trying 377
- F5 180 ringing 377
- F6 180 ringing 377
- F7 200 ok 377
- F8 200 ok 377
- F9 ack 377
- Successful call setup and disconnect 377
- User a proxy server user b 377
- Way rtp channel established 377
- Unsuccessful call setup called user is busy 379
- F1 invite b 380
- F2 invite b 380
- F3 100 trying 380
- F4 100 trying 380
- F5 486 busy here 380
- F6 486 busy here 380
- F7 ack 380
- F8 ack 380
- User a proxy server user b 380
- F1 invite b 382
- F2 invite b 382
- F3 180 ringing 382
- F4 180 ringing 382
- F5 cancel 382
- F6 cancel 382
- F7 200 ok 382
- F8 200 ok 382
- Unsuccessful call setup called user does not answer 382
- User a proxy server user b 382
- Successful call setup and call hold 384
- Successful call setup and call waiting 386
- Call transfer without consultation 391
- Administrator s guide for cp860 ip conference phones 392
- Call is established between user a and user c 392
- User c answers the call 392
- Call transfer with consultation 395
- Administrator s guide for cp860 ip conference phones 396
- Call is established between user b and user c 396
- User a transfers the call to user c 396
- Always call forward 400
- Appendix 401
- Call is established between user a and user c 401
- User c answers the call 401
- Busy call forward 404
- No answer call forward 407
- Call conference 410
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