VoIP Analog Telephone Adapter
VIP-156
VIP-157
User’s manual
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Revision
User’s Manual for PLANET VoIP Analog Telephone Adapter:
Model: ATA
Rev: 1.0 (2006, January)
Part No. EM-VIP_ATAV1
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Table of Contents
Chapter 1 Introduction ............................................................................................................6
Overview....................................................................................................................................................... 6
Chapter 2 Preparations & Installation.................................................................................13
Chapter 3 Network Service Configurations.........................................................................18
Configuring and monitoring your Phone Adapter from web browser.................................................. 18
Overview on the web interface of ATA .............................................................................................. 18
Manipulation of ATA via web browser............................................................................................... 18
Chapter 4 VoIP Telephone Adapter Configurations............................................................20
Speed Dial settings............................................................................................................................. 20
SNTP settings..................................................................................................................................... 21
[VIP-156] Volume Setting.................................................................................................................. 22
[VIP-157] Volume Setting.................................................................................................................. 22
Block Setting...................................................................................................................................... 23
Country Settings................................................................................................................................. 31
ICMP Setting...................................................................................................................................... 31
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Reset to Default.................................................................................................................................. 33
Appendix A Voice Communication Samples......................................................................................... 34
ATA to ATA connection via IP address............................................................................................... 34
Voice communication via SIP proxy server – SIP-50......................................................................... 36
Voice communication via IP PBX system – IPX-2000 (Auto-config) ............................................... 38
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1
Chapter 1 Introduction
Overview
Based on years of VoIP manufacturing experiences, PLANET Technology VoIP total solutions are
known for advanced implementation of standards based telephony with mass deployment capability.
Cost-effective, easy-to-install and simple-to-use, the PLANET VIP-156/VIP-157 VoIP Phone Adapter
(“ATA” in the following term) converts standard telephones to IP-based networks. The service
providers and enterprises offer users traditional and enhanced the telephony communication services
via the existing broadband connection to the Internet or corporation network.
With the ATA, home users and companies are able to save the installation cost and extend their past
investments in telephones, conference and speakerphones. The ATA can be the bridge between
traditional analog systems and IP network with an extremely affordable investment.
The ATA includes two alternatively Ethernet interface for Internet (PPPoE, DHCP or Fixed IP), or office
LAN connection. With adding the auto-provision feature of our IP PBX product - IPX-2000, the ATA can
be seamlessly integrated into the telephony network and be used in consumer and business IP
telephony service, no expertise required!
The ATA and our IP PBX system integration are the ideal combination for your office daily
communications.
Product Features
•
•
•
•
•
•
Feature-rich telephone service over home or office Internet/Intranet connection
Auto-config feature for flexible, ease-of-use IP PBX system integration
Cost effective, field proven compatibility, and stability
Web-based and telephone keypad machine configuration
Remote machine administration authentication
Voice prompt for machine configurations
VoIP Features
•
•
•
SIP 2.0 (RFC3261) compliant
Peer-to-Peer / SIP proxy calls
Voice codec support: G.711, G.723.1, G.729A/G.729B
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•
•
Voice processing: Voice Active Detection, DTMF detection/ generation, G.168 echo cancellation
(16mSec.), Comfort noise generation (CNG)
In band, out-of-band, and SIP-info DTMF support
Package Content
The contents of your product should contain the following items:
VoIP Telephone Adapter
Power adapter
Quick Installation Guide
User’s Manual CD
RJ-11 cable x 1
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Physical Details
The following figure illustrates the front/rear panel of ATA.
Respective model/descriptions are shown below:
VIP-156: SIP Analog Telephone Adapter
VIP-156PE: 802.3af PoE SIP Analog Telephone Adapter
VIP-157: 1 FXS/ 1 FXO SIP Analog Telephone Adapter
VIP-157S: 2-port FXS SIP Analog Telephone Adapter
Front Panel of VIP-156
Left / Right Panel of VIP-156
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Front Panel of VIP-156PE
Left / Right Panel of VIP-156PE
Front Panel of VIP-157
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Left / Right Panel of VIP-157
Front Panel of VIP-157S
Left / Right Panel of VIP-157S
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LED Display & Button
RJ-45 connector, to maintain the existing network structure,
1
2
PC
connected directly to the PC through straight CAT-5 cable
RJ-45 connector, for Internet access, connected directly to
Switch/Hub through straight CAT-5 cable.
LAN
The LAN interface also can be connected with 802.3af PoE
switch or converter for power supply (VIP-156PE)
12V DC Power input outlet
3
4
12V DC
Reset
Reset to the factory default setting
buttononrearpanelover5secondswillresettheVoIPPhone
Adapter to factory default value. (Except speed dial and
call forward settings)
ÍNote
LED display of VIP-156 / VIP-156PE
LED Indicators
Descriptions
Power is supplied to the device.
PWR
The Status LED will be flashing when the machine is operational
STATUS
LNK/ACT
OFF: the device is connected to LAN at 10Mb/s.
ON: the device is connected to LAN at 100Mb/s.
OFF: the phone is idle.
RING
ON: the phone is in use (offhook).
Blinking: the phone is ringing.
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LED display of VIP-157 / VIP-157S
LED Indicators
Descriptions
The Status LED will be flashing when the machine is operational
STATUS
LNK/ACT
OFF: the device is connected to LAN at 10Mb/s.
ON: the device is connected to LAN at 100Mb/s.
OFF: the phone is idle.
RING 1
RING 2
ON: the phone is in use (offhook).
Blinking: the phone is ringing.
OFF: the phone is idle.
ON: the phone is in use (offhook).
Blinking: the phone is ringing.
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Chapter 2
2
Preparations & Installation
Physical Installation Requirement
This chapter illustrates basic installation of ATA analog Phone Adapter ((“ATA” in the following term))
•
•
Network cables. Use standard 10/100BaseT network (UTP) cables with RJ45 connectors.
TCP/IP protocol must be installed on all PCs.
For Internet Access, an Internet Access account with an ISP, and either of a DSL or Cable modem
Administration Interface
PLANET ATA provides GUI (Web based, Graphical User Interface) for machine management and
administration.
Web configuration access
To start ATA web configuration, you must have one of these web browsers installed on computer for
management
•
Microsoft Internet Explorer 4.01 or higher with Java support
Default LAN interface IP address of ATA is 192.168.0.1. You may now open your web browser, and
insert 192.168.0.1 in the address bar of web browser to logon ATA web configuration page.
ATA will prompt for logon username/password, please enter: root / null (no password) to continue
machine administration.
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Please locate your PC in the same network segment
(192.168.0.x) of ATA. If you’re not familiar with TCP/IP,
please refer to related chapter on user’s manual CD or
consult your network administrator for proper network
configurations.
ÍNote
LAN IP address configuration via web configuration interface
Execute your web browser, and insert the IP address (default: 192.168.0.1) of VIP in the adddress bar.
After logging on machine with username/password (default: root / no passwrd), browse to “Network”
--> “Network settings” configuration menu:
Parameter Description
IP address
LAN IP address of ATA
Default: 192.168.0.1
LAN mask of ATA
Subnet Mask
Default Gateway
Default: 255.255.255.0
Gateway of ATA
Default: 192.168.0.254
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Network settings via Keypad commands
The ATA series phone adapters support telephone keypad configurations, please connect analog
telephone set and refer to the following table for machine network configurations.
IVR Menu Choice
#111#
Machine operation
Parameter(s)
None
Use the * (star) key
Notes:
ATA will change to DHCP
Client
Set DHCP client
DHCP will be disabled and
#112xxx*xxx*xxx*
xxx#
Setup Static IP Address
Set Network Mask
when entering a decimal system will change to the
point.
Static IP type.
Use the * (star) key
#113xxx*xxx*xxx*
xxx#
when entering a decimal Must set Static IP first.
point.
Use the * (star) key
#114xxx*xxx*xxx*
xxx#
Set Gateway IP Address
Set Primary DNS Server
when entering a decimal Must set Static IP first.
point.
Use the * (star) key
#115xxx*xxx*xxx*
xxx#
when entering a decimal Must set Static IP first.
point.
For PLANET IPX -2000 IP
Enable auto-config mode
Disable auto-config mode
Reboot
None
#136#
PBX System
For PLANET IPX -2000 IP
None
#137#
#195#
PBX System
The system will reboot
None
automatically.
The system will be reset to
#198#
Factory Reset
None
None
factory default value and
reboot automatically.
VIP-157 only
＊
To switch IP Phone mode
Following keypad commands can be used to display the network settings enabled on ATA via voice
prompt.
IVR Menu Choice
Machine operation
Notes:
IVR will announce the current IP address of
the ATA.
#120#
Check IP Address
IVR will announce if DHCP in enabled or
disabled.
#121#
#122#
Check network connection Type
Check the Phone Number
IVR will announce current enabled VoIP
number.
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IVR will announce the current network mask
of the ATA.
#123#
#124#
#125#
#128#
Check Network Mask
IVR will announce the current gateway IP
address of the ATA.
Check Gateway IP Address
Check Primary DNS Server Setting
Check Firmware Version
IVR will announce the current setting in the
Primary DNS field.
IVR will announce the version of the
firmware running on the ATA.
PleasecontactyourInternetserviceprovidertoobtainthe
Internetaccesstype,andselectthepropernetworksettings
in ATA to establish the network connections.
L Hint
After confirming the modification you’ve done, Please click on the Submit button to apply settings and
browse to “Save & Reboot” menu to reboot the machine to make the settings effective.
Connection Type
Data required.
In most circumstances, it is no need to configure the DHCP
Fixed IP
settings.
The ISP will assign IP Address, and related information.
The ISP will assign PPPoE username / password for Internet
access,
DHCP clinet
PPPoE
PleaseconsultyourISPpersonneltoobtainproperPPPoE/IP
address related information, and input carefully.
If Internet connection cannot be established, please check
the physical connection or contact the ISP service staff
for support information.
L Hint
Save Modification to Flash Memory
Most of the VoIP router parameters will take effective after modifications, but it is just temporary stored
on RAM only, it will disappear after your reboot or power off the VoIP Phone Adapter, to save the
parameters into Flash ROM and let it take effective forever, please remember to press the Save &
Reboot button after you modify the parameters.
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Chapter 3
3
Network Service Configurations
Configuring and monitoring your Phone Adapter from web browser
The ATA integrates a web-based graphical user interface that can cover most configurations and
machine status monitoring. Via standard web browser, you can configure and check machine status
from anywhere around the world.
Overview on the web interface of ATA
With web graphical user interface, you may have:
Š More comprehensive setting feels than traditional command line interface.
Š Provides user input data fields, check boxes, and for changing machine configuration settings
Š Displays machine running configuration
To start ATA web configuration, you must have one of these web browsers installed on computer for
management
Š Microsoft Internet Explorer 4.01 or higher with Java support
Manipulation of ATA via web browser
Log on ATA via web browser
After TCP/IP configurations on your PC, you may now open your web browser, and input
http://192.168.0.1 to logon Phone Adapter web configuration page.
Phone Adapter will prompt for logon username/password: root / null (no password)
ATA log in page
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When users login the web page, users can see the Phone Adapter system information like firmware
version, company…etc in this main page.
VoIP Phone Adatper main page
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Chapter 4
4
VoIP Telephone Adapter Configurations
Speed Dial settings
In Speed Dial setting function you can add/delete Speed Dial number. You can input maximum 10
entries speed dial list.
If you need to add a phone number into the Speed Dial list, you need to input the position, the name,
and the phone number (by URL type). When you finished a new phone list, just click the “Add Phone”
button.
If you want to delete a phone number, you can select the phone number you want to delete then click
“Delete Selected” button.
If you want to delete all phone numbers, you can click “Delete All” button.
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Call forward
Call Forward function: you can setup the phone number you want to forward in this page. There are
three type of Forward mode. You can choose All Forward, Busy Forward, and No Answer Forward by
click the icon.
All Forward: All incoming call will forward to the number you choosed. You can input the name and the
phone number in URL field. If you select this function, then all the incoming call will direct forward to the
speed dial number you choose.
Busy Forward: If you are on the phone, the new incoming call will forward to the number you choosed.
You can input the name and the phone number in URL field.
No Answer Forward: If you can not answer the phone, the incoming call will forward to the number you
choosed. You can input the name and the phone number in URL field. Also you have to set the Time
Out time for system to start to forward the call to the number you choosed.
When you finished the setting, please click the Submit button.
SNTP settings
This page defines the primary and second SNTP Server IP Address, to get the date/time information.
Also you can base on your location to set the Time Zone, and how long need to synchronize again.
When you finished the setting, please click the Submit button.
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[VIP-156] Volume Setting
This page defines the Handset Volume, Ringer Volume, and the Handset Gain. When you finished the
setting, please click the Submit button.
Handset Volume is to set the volume for you can hear from the handset.
Ringer Volume is to set the ringer volume for you can hear.
Handset Gain is to set the volume send out to the other side’s handset.
[VIP-157] Volume Setting
This page defines the Handset Volume, Ringer Volume, and the Handset Gain. When you finished the
setting, please click the Submit button.
Handset Volume is to set the volume for you can hear from the handset.
Ringer Volume is to set the ringer volume for you can hear.
PSTN-Out Volume is to set the PSTN volume for you can hear. [VIP-157 only]
Handset Gain is to set the volume send out to the other side’s handset.
PSTN-In Gain is to set the volume send out to the other side’s handset. [VIP-157 only]
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Volume Setting of VIP-157
Block Setting
This page defines the Block Setting to keep the phone slience. You can choose Always Block or Block a
period.
Always Block: All incoming call will be blocked until disable this feature.
Block Period: Set a time period and the phone will be blocked during the time period. If the “From”
time is large than the “To” time, the Block time will from Day 1 to Day 2.
When you finished the setting, please click the Submit button.
[VIP-157 only] Auto Answer
This page set the Auto Answer function to answer the incoming call by the phone. If the call is come
from the IP, then the VIP-157 can let user to redial the call to PSTN phone number. If the call is coming
from PSTN, then the VIP-157 can let user to redial to IP Phone number. Auto Answer Counter is to set
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after the ring count meet the number you set then the auto answer will enable.
Caller ID settings
This page defines the device to show Caller ID in your PSTN Phone or IP Phone. There are four
selection of Caller ID. You need to base on your environment to set the Caller ID function for FSK or
DTMF.
Dial Interval Settings
This function is when you input the phone number by the keypad but you don’t need to press “#”. After
time out the system will dial directly.
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Flash Time Setting
When you use the PSTN Phone and you need to press the Hook to do the Flash (Switch to the other
phone line or HOLD), this function is for you to set the time you press the Hook to represent the Flash
function.
Call waiting Settings
When you are talking with other people, You can choose If you want to hear the notice when there is a
new coming call. If the call waiting function is On, if there is a new incomeing call, you will hear the call
waiting notice in your current call. If you set the function to Off, then you will not hear any notice.
DDNS Settings
This page defines the DDNS setting in this page. You need to have the DDNS account and input the
informations properly. You can have a DDNS account with a public IP address then others can call you
via the DDNS account. But now most of the VoIP applications are work with a SIP Proxy Server. When
you finished the setting, please click the Submit button.
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Port Settings
This page defines the SIP and RTP port number in this page. Each ISP provider will have different
SIP/RTPport setting, please refer to the ISP to setup the port number correctly. When you finished the
setting, please click the Submit button.
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Service Domain Settings
In Service Domain Function you need to input the account and the related informations in this page,
please refer to your ISP provider. You can register three SIP account in the ATA. You can dial the VoIP
phone to your friends via first enable SIP account and receive the phone from these three SIP
accounts.
First you need click Active to enable the Service Domain, then you can input the following items:
Display Name: you can input the name you want to display.
Line number: you need to input the User Name get from your ISP.
Register Name: you need to input the Register Name get from your ISP.
Register Password: you need to input the Register Password get from your ISP.
Domain Server: you need to input the Domain Server get from your ISP.
Proxy Server: you need to input the Proxy Server get from your ISP.
Outbound Proxy: you need to input the Outbound Proxy get from your ISP. If your ISP does not
provide the information, then you can skip this item.
Register Period: you need to input the Register Period get from your ISP. This is count in minute.
You can see the Register Status in the Status item. If the item shows “Registered”, then your Phone
Adapter is registered to the ISP, you can make a phone call direcly.
If you have more than one SIP account, you can following the steps to register to the other ISP.
When you finished the setting, please click the Submit button.
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RTP Setting
This page defines the Out-Band DTMF and Send DTMF SIP Info Enable/Disable in this page. To
change this setting, please following your ISP information. When you finished the setting, please click
the Submit button.
DTMF Setting
This page defines the RFC2833 Out-Band DTMF, Inband DTMF and Send DTMF SIP Info in this page.
To change this setting, please following your ISP information. When you finished the setting, please
click the Submit button.
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RPort Settings
This page defines the RPort Enable/Disable in this page. To change this setting, please following your
ISP information. When you finished the setting, please click the Submit button.
Other Settings
This page defines the Hold by RFC, Voice/SIP QoS and SIP expire time in this page. To change these
settings please following your ISP information. When you finished the setting, please click the Submit
button. The QoS setting is to set the voice packets’ priority. If you set the value higher than 0, then the
voice packets will get the higher priority to the Internet. But the QoS function still need to cooperate with
the others Internet devices.
STUN settings
This page defines the STUN Enable/Disable and STUN Server IP address in this page. This function
can help your Phone Adapter working properly behind NAT. To change these settings please following
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your ISP information. When you finished the setting, please click the Submit button.
Auto Configuration
In Auto Configuration Setting you need to check with your service provider if they have provided this
function. Usually this function will be boundle with an IP PBX to use in the office.
Country Settings
In Country Settings is for you to set the Country, different country will have different settings in FXS inter
face.
ICMP Setting
This function can disable echo when someone ping this device, it can avoid haker try to attack the
device. When you finished the setting, please click the Submit button.
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System Authority
In System Authority you can change your login password.
Save & Reboot
In Save & Reboot you can save the changes you have done. If you want to use new setting in the
Phone Adapter, You have to click the Save button. After you click the Save button, the Phone Adapter
will automatically restart and the new setting will effect.
Firmware Upgrade
In Firmware Upgrade function you can update new firmware via HTTP in this page. You can ugrade the
firmware by the following steps:
Select the firmware code type, AP or DSP code.
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Click the “Browse” button in the right side of the File Location or you can type the correct path and the
filename in File Location blank.
Select the correct file you want to download to the Phone Adapter then click the Update button.
Reset to Default
In Default Setting you can restore the Phone Adapter to factory default in this page. You can just click
the Restore button, then the Phone Adapter will restore to default and automatically restart again.
Reboot without saving
Reboot function you can restart the Phone Adapter. If you want to restart the Phone Adapter, you can
just click the Reboot button, then the Phone Adapter will automatically.
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Appendix A Voice Communication Samples
There are several ways to make calls to desired destination in ATA. In this section, we’ll lead you step
by step to establish your first voice communication via keypad and web browsers operations.
ATA to ATA connection via IP address
Assume there are two ATAs in the network the IP address are 192.168.0.1, 192.168.0.2
Analog telephone sets are connected to the phone (RJ-11) port of ATAs respectively
192.168.0.1
192.168.0.2
1
9
2
*
1
6
8
*
0
*
2
#
Operation steps:
1. Pick up the telephone set on ATA A.
2. [VIP-157 only, for VIP-156, please skip this step] Press the “*” key to switch to IP phone mode, and
you should be able to hear the dial-tone
3. Press the keypad: 192*168*0*2# shall be able to connect to the ATA B.
4. Then the phone in 192.168.0.2 should ring. Please repeat the same dialing steps on ATA B to
establish the first voice communication from ATA A
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(Peer-to-Peer mode) VIP-157S Port 1 to Port 2 communications
Supposing one VIP-157S connects to two telephones, just pick up phone 1 and dial
‘192*168*0*1**5062’, phone 2 will ring.
Analog telephone sets are connected to the phone (RJ-11) ports of VIP-157S respectively
192.168.0.1
1001
1002
1
9
2
*
1
6
8
*
0
*
1
*
5
0
6
2
#
*
Operation steps:
a) Pick up the telephone set on VIP-157S port 1, and you should be able to hear the dial-tone
b) Press the keypad: 192*168*0*1**5062# shall be able to connect to the VIP-157S port 2
c) Then the telephone set in VIP-157S port 2 should ring. Please repeat the same dialing steps on port
2 to establish the first voice communication from VIP-157S
y In default machine operation, the VIP-157 is at PSTN
mode. If you want to make an IP call, press the “*” key
toswitchtoIPPhonemode.ForVIP-156,noneedtopress
the star “*” key in advance.
L Hint
y If the IP address of the remote calling party is known,
you may directly make calls via its IP address and end
with a “#”.
y If the ATAs are installed behind a NAT/firewall/IP
sharing device for Peer-to-Peer VoIP application,
please make sure the NAT device support SIP
applications, and suitable settings should be applied
to the NAT device to enable the SIP communications
before making calls
y
[VIP-157S] in PLANET ATA series products, to connect
to remote ATA, press the keypad in the following
sequence to connect to the remote VIP-157S port 2:
[remote ATA IP address]**5062, for example:
192*168*0*2**5062
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Voice communication via SIP proxy server – SIP-50
Registration /
Authentication
Registration /
Authentication
SIP-50 IP Address: 192.168.0.50
„
ATA A IP Address: 192.168.0.1
Line Number: 1001
„
ATA B IP Address: 192.168.0.2
Line Number: 2002
Device configurations on the ATA:
STEP 1:
Log in SIP-50 and create two testing accounts/password: 1001 / 123 (for ATA A), and 1002 /
123 (for ATA B) for the voice calls.
STEP 2:
Please log in ATA via web browser, browse to the SIP setting menu and select the Domain
Service config menu. In the setting page, please insert the account/password information
obtained from your service provider (in this sample, we’re using PLANET SIP-50 as the SIP
Proxy server for SIP account, call authentications), and then the sample configuration
screen is shown below:
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STEP 3:
Repeat the same configuration steps on ATA B, and check the machine registration status,
make sure the registrations are completed.
Test the scenario:
To verify the VoIP communication, please
Pick up the telephone
[VIP-157 only] Press the “*” key to switch to IP phone mode
Dial the destination number to make call between SIP clients. For example, ATA A (with
number 1001) with keypad number 1002 to ATA B, or reversely make calls from SIP client
(ATA B) to the number 1001 (ATAA)
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Voice communication via IP PBX system – IPX-2000 (Auto-config)
In the following sample, we’ll introduce how to integrate the ATA with our IP PBX system IPX-2000 via
Auto-config feature.
Registration /
Registration /
Authentication
Authentication
IPX-2000 LAN IP Address: 192.168.0.50
„
ATA B IP Address: 192.168.0.2
Line Number: 2002
„
ATA A IP Address: 192.168.0.1
Line Number: 1001
Device configurations on the IPX-2000:
STEP 1:
Log in IPX-2000 and browse to the DHCP menu and create new options list for the auto
configuration.
Code: please insert 151 as the DHCP server option.
STEP 2:
Please browse to the Device menu and create new device for the auto configuration.
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STEP 3:
Please press the Show extensions button to create the two extension accounts/password:
1001/123 (for ATA A), and 1002/123(for ATA B) for the voice calls.
STEP 4:
After setting up the parameters, please refer to the path to activate the settings: Service --->
IP PBX service ---> IP PBX configuration reload
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Device configurations on the ATA:
STEP 5:
Please log in ATA via web browser, browse to the SIP setting menu and select the Domain
Service config menu. In the setting page, please browse to the Auto-config page, and
enable the Auto Configuration features for IP PBX system. (Your may connect telephone set
to ATA, press #136 to enable the Auto configuration, or press #137 to disable the Auto
Configuration setting.)
STEP 6:
After enabling the auto-config feature, the ATA shall be able to obtain IP address and SIP
extension information from IP PBX system IPX-2000 information. To verify the auto-config
results, you may connect telephone set to ATA; press #120# to check if the IP address is
obtained from IPX-2000. And #122# can be used to verify the extension number assigned by
IPX-2000.
STEP 7:
Repeat the same configuration steps on ATA B, and check if the ATA B is successfully
registered with the IPX-2000 as one of the IP extensions.
Test the scenario:
To verify the VoIP communication, please:
1. Pick up the telephone
2. [VIP-157 only] Press the “*” key to switch to IP phone mode
3. Dial the destination number to make calls. For example, ATA A (with number 1001) dial the keypad
with number 1002 (ATA B) or reversely make calls from SIP client (ATA B) to the number 1001 (ATA
A)
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Appendix B VIP-156 / VIP-157 Specifications
Product
SIP Telephone Adapter
VIP-156 VIP-156PE
Model
VIP-157
VIP-157S
Hardware
LAN
1 x 10/100Mbps RJ-45 port (802.3af PoE for VIP-156PE)
1 x 10/100Mbps RJ-45 port
PC
FXS (for telephone set
connection)
FXO (PSTN connection)
Protocols and Standard
Standard
1 x RJ-11
2 x RJ-11
---
---
SIP 2.0 (RFC3261)
1 x RJ-11
Voice codec
Voice Standard
G.723.1 (6.3k/5.3k), G.729A, G.729B, G.711
Voice activity detection (VAD)
Comfort noise generation (CNG)
Acoustic echo canceller (AEC)
G.165: Line echo canceller (LEC)
Jitter Buffer
Protocols
SIP 2.0 (RFC-3261), TCP//IP, UDP/RTP/RTCP, HTTP, ICMP, ARP, DNS,
DHCP, NTP/SNTP, PPP, PPPoE
Network and Configuration
Access Mode
Static IP, PPPoE, DHCP
Web, keypad
Management
Dimension (W x D x H)
Operating Environment
Power Requirement
EMC/EMI
94 x 72 x 30 mm
0~40 degree C, 10~95% humidity
12V DC
CE, FCC Class B
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