GXV-3000 User Manual
Grandstream Networks, Inc.
User Manual
GXV-3000
Innovative IP Video Phone
For Firmware Version 1.0.0.33
Grandstream Networks, Inc.
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5.3 CONFIGURATION THROUGH CENTRAL PROVISIONING SERVER
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6 SOFTWARE UPGRADE & CUSTOMIZATION
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6.1 FIRMWARE UPGRADE THROUGH TFTP/HTTP
6.2 CONFIGURATION FILE DOWNLOAD
6.3 MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD
6.4 CUSTOMIZATION OWN SCREENSAVER IMAGES
6.5 CUSTOMIZATION OWN RING TONES
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7 AUXILIARY PORTS
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7.1 USB 2.0 PORT
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7.1.1
Capture pictures via USB port
7.2 RCA STYLE STEREO AUDIO & COMPOSITE VIDEO OUTPUT
7.3 HEADSET JACK
8 VIDEO SURVEILLANCE
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9 RESTORE FACTORY DEFAULT SETTING
10 GLOSSARY OF TERMS
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GXV-3000 User Manual
Grandstream Networks, Inc.
1
Welcome
Thank you for purchasing Grandstream award-winning GXV-3000 Video IP Phone. You made an ex-
cellent choice and we hope you will enjoy all its capabilities.
Grandstream GXV-3000 SIP IP Video Phone is a next generation advanced IP videophone based on
SIP and H.264 standard. Built upon Grandstream’s innovative technology, the GXV-3000 IP Video
telephone offers a rich set of functionality, ease of use, superb sound and video quality, stylish exterior
design and highly attractive price. They are fully compatible with SIP industry standard and can inter-
operate with many other SIP compliant devices and software on the market.
This document is subject to changes without notice. The latest electronic version of this user manual is
available for download from the following location:
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2
Installation
2.1 What is Included in the Package
The GXV-3000 phone package contains:
1) One GXV-3000 Main Case
2) One Handset
3) One Phone Cord
4) One Universal Power Adaptor
5) One Ethernet Cable
6) Two Wall Mount Sustaining Brackets
2.2 Connecting Your Phone
Following is a backside picture of GXV-3000, each connection port is labeled with the name in the fol-
lowing table:
Table 2-1: The connectors on the GXV-3000 phone:
USB Interface for external USB device like flash to store captured
image, ring tones and phone address books, etc.
USB
10/100 RJ-45 port for uplink to Ethernet; or WAN port when con-
figured in Router mode.
LAN
10/100 RJ-45 port for connecting PC in switch mode or LAN port
in router mode
PC
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Power
RCA Output
Headset
12V DC power port
RCA Output Interface to external display like TV set
2.5mm Headset port
2.3 Wall Mount
GXV-3000 can be wall mounted. The two wall mount sustaining brackets can be attached to the bot-
tom of the main body:
User can simply place the device against the wall with two holes to the fixed hanger and the two sus-
taining brackets balance the bottom to position the phone on the wall.
Handset
Rest
Tab
Tab with exten-
sion down
Tab with exten-
sion up
User will need to pull out the tab (extension downward) from handset cradle on the top of the handset
rest, and rotate the tab and plug it back into the slot with the extension up for handset holding.
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2.4 Safety Compliances
The GXV-3000 phone is compliant with various safety standards including FCC/CE. Its power adaptor
is compliant with UL standard. The phone should only be operated with the universal power adaptor
provided with the package. Damages to the phone caused by using other unsupported power adaptors
are not covered by the manufacturer’s warranty.
2.5 Warranty
Grandstream has a reseller agreement with our reseller customer. End user should contact the company
from whom you purchased the product for replacement, repair or refund.
If you purchased the product directly from Grandstream, contact your Grandstream Sales and Service
Representative for a RMA (Return Materials Authorization) number before you return the product.
Grandstream reserves the right to remedy warranty policy without prior notification.
Warning: Please do not attempt to use a different power adaptor. Using other power adaptor may
damage the GXV-3000 and will void the manufacturer warranty.
Caution: Changes or modifications to this product not expressly approved by Grandstream, or op-
eration of this product in any way other than as detailed by this User Manual, could void your
manufacturer warranty.
Information in this document is subject to change without notice. No part of this document may be repro-
duced or transmitted in any form or by any means,-el7ec-tronic or mechanical, for any purpose without the
express written permission of Grandstream Networks, Inc..
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3
Product Overview
GXV-3000 IP Video Phone is designed to be used in general household or office. The following photo
illustrates the appearance of a GXV-3000 IP Video phone.
Front View
Back View
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3.1 Key Features
Grandstream GXV-3000 IP Video Phone is a next generation advanced IP video telephone based on
industry’s open standard SIP (Session Initiation Protocol) and H.264. Built upon Grandstream’s inno-
vative technology, GXV-3000 IP Video Phone features superb audio and video quality, rich function-
alities, stylish exterior design and highly attractive price.
Software Feature:
Support SIP 2.0 ， TCP/UDP/IP ， RTP/RTCP ， HTTP/HTTPS ， ICMP ， ARP/RARP ，
DNS，DHCP (both client and server), NTP, PPPoE, TFTP, Telnet, TLS (pending), etc.
Support dual 10M/100M auto-sensing Ethernet ports configurable to operate under either
switch or router/NAT mode (pending).
Support Quick IP Call Mode (partially simulating PBX in a LAN environment w/o SIP server)
Powerful video DSP with advanced adaptive jitter control and packet loss concealment tech-
nology to ensure superb audio and video quality
Support advanced H.264 base line real-time video codec (at CIF or QVGA resolution and up to
30 frames/second) to ensure highest quality video delivery at 32kbps – 1Mbps bandwidth level
(bit rate/frame rate configurable to reach best audio/video quality in available bandwidth)
Support various audio codecs including G.711 A/U law (PCMA/PCMU)，G.723.1, G.729A/B,
GSM, G.726 (pending) and iLBC (pending) with dynamic negotiation of codec type and packet
time during call setup
Support popular voice features including 3 line indicators (each of which can be configured us-
ing independent SIP account), Caller ID/Name Display/Block, Hold, Call Waiting, Call Trans-
fer, Call Forward (no answer, busy, unconditional), Full Duplex Hands-Free Speaker Phone
with advanced Acoustic Echo Cancellation (AEC/G.167), 3-way Conference, Do-not-Disturb,
Incoming Call Rejection, various DTMF options (in audio, RFC2833, SIP INFO), download-
able music ring tone, etc.
Support advanced video features including a high quality 5.6-inch TFT color LCD (allowing 2-
dimensional view angle adjustment including 180 degrees vertical rotation and nearly 300 de-
grees horizontal rotation), advanced VGA resolution CMOS camera sensor (view angle adjust-
able), anti-flickering, auto focus and auto exposure, 2X analog and 4X digital Zoom, PIP (Pic-
ture-in-Picture), audio mute and camera block (for privacy), call log, phone/address book, con-
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figurable screen-saver pictures, still picture capture/store/send (VGA resolution), visual voice
message indicator, intuitive graphic user interface enabled by 5 navigation buttons, etc.
Support 2 USB (2.0) host ports, 1 audio and 1 video output jack (capable of outputting video to
an external TV simultaneously), headset jack (2.5mm)
Support silence suppression and VAD，AGC and acoustic echo cancellation (G.167)
Support standard encryption and authentication using DIGEST (MD5 and MD5-sess) and AES
Support secure signaling (SIP over TLS, pending) and secure voice/video communication
(SRTP) (pending)
Support layer-2 (802.1Q VLAN, 802.1p) and layer-3 (DiffServ, ToS) QoS
Support automated NAT traversal without manual manipulation of firewall/NAT
Support remote automated and secure provisioning and software upgrade through firewall/NAT
to enable “zero configuration” and “plug-and-dial” for end users
Support remote device monitoring and events reporting using Syslog
Support device configuration via LCD, Web browser or central secure configuration file
Support video surveillance in LAN environment, WAN possible if NAT configured right.
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3.2 Hardware Specification
The table below describes the hardware specification of GXV-3000:
Model
Ethernet Port
GXV-3000
Dual 10M/100M auto-sensing Ethernet ports
Switch or router/NAT mode configurable
LCD
5.6 inch TFT color LCD
Advanced CMOS sensor (VGA resolution),
330K Pixels
Camera
RCA style stereo audio & composite video jack,
1 headset jack, 2 USB 2.0 ports
Auxiliary Port
Black or Silver ABS plastic, 30 buttons (6 of
which are LED buttons)
Exterior
Headset Jack
2.5mm Headset Port
100-240V input，+12VDC/1.2A output
US/Euro/UK/Japan/Australian style available
6.5cm x 18cm x 16cm
Universal Power Supply
Dimension
Weight
1.2kg
Operating Temperature 0-40 ˚C
Humidity
10-95% non-condensing
Compliance
FCC/CE/C-Tick (pending)
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4
Using GXV-3000 IP Video Phone
4.1 Getting Familiar with LCD
GXV-3000 IP Video phone has a 5.6 inch TFT color LCD. Here is the sample LCD display showing
the phone is registered to VoIP service providers’ SIP server/proxy with one missed call icon.
The phone has a screen saver. When it is configured, the phone will show the screen saver or turn off
the LCD just like normal laptop computer screen.
Icon
LCD Icon Definitions
Network Status Icon:
RED and FLASH in the case of Ethernet link failure
RED if IP address or SIP server is not found
ON if IP address and SIP server are located
Phone Status Icon:
OFF when the handset is on-hook
ON when the handset is off-hook
Speaker Phone Status Icon:
FLASH when phone rings or a call is pending
OFF when the speakerphone is off
ON when the speakerphone is on
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Handset, Speakerphone and Ring Volume Icon:
0-7 scales to adjust handset / speakerphone / ring volume
Real-time Clock:
Synchronized to Internet time server
Time zone configurable via web browser
Time Icon:
AM
PM
AM for the morning
PM for the afternoon
4.2 Getting Familiar with Keypad
Here is the key assignment in the phone’s keypad:
Line 1 – 3 Keys
Message
Camera Block
Mute/Del
Menu Keys
Conference
Transfer
Address Book
Camera Local Loop-
back Display
Hold Speaker Send/Re-Dial
Standard Keypad
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GXV-3000 IP Video phone has 30 key buttons:
Key Button
Key Button Definitions
LINE1-LINE3
3 Line keys with LED, can be configured to different SIP profile
Next “Menu Item” when phone is in keypad configuration mode
Or increase handset/speakerphone volume when phone is ACTIVE
Or increase ring volume when phone is in IDLE mode
UP ↑
Previous “Menu Item” when phone is in keypad configure mode
Or decrease handset/speakerphone volume when phone is ACTIVE
Or decrease ring volume when is in IDLE mode
DOWN ↓
LEFT Å
Shift cursor to left
Shift cursor to right
RIGHT Æ
Enter Keypad Configuration “MENU” mode when phone is in
IDLE mode.
OK
It is also the ENTER key once entering Keypad Configuration.
TRNF
CONF
TRANSFER key: Transfer an ACTIVE call to another number
Bring Calling/Called party into conference
Enter to retrieve (video) voice mails or other messages
MESSAGE
Enable/Disable Block Camera for private, just doing normal
voice/audio call without sending video. When this turned on, the
phone is just function as a normal IP phone
CAMERA BLOCK
Enable/Disable to Mute an ACTIVE call; or function as Delete a
key entry, call log, etc; or shortcut to Activate/Deactivate DND
when idle; or Reject an incoming call when ringing
MUTE/DEL
Enable/Disable camera local loopback display
CAMERA LOCAL
LOOPBACK DISPLAY
Video Phone Address Book (Not fully ready yet, will implement
downloadable address book using XML in future f/w)
ADDRESS BOOK
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Enable/Disable hands-free speaker mode
Temporarily hold an ACTIVE call
SPEAKER
HOLD
Dial a new number or Redial the last number dialed. After entering
the phone number, pressing this key would force a call to go out
immediately before “no key entry timeout” value expires
SEND
12 standard Digit, * and # keys are usually used to make phone
calls
0 - 9, *, #
4.3 Making and Answering Phone Calls
4.3.1 Handset, Speakerphone and Headset Mode
Handset can be switched between either Speaker or Headset. However, whenever the Headset is
plugged in, Speaker will be switched to Headset.
To Switch between Handset and Speaker/Headset, simply press the Hook Flash in the Handset cradle
or press the SPEAKER button in the phone.
4.3.2 Multiple SIP Accounts and Lines
GXV-3000 can support up to 3 independent SIP accounts. Each account is capable of independent SIP
server, user and NAT settings among others. Each of the 3 LINE buttons (LINE1-LINE3) is “virtually”
mapped to each SIP account. In off hook state, when user chooses an idle line, the name of the ac-
count (as configured in the web interface) will be displayed in the LCD while a dial tone is being
played out. For example, if the 3 SIP accounts are named FWD (FreeWorldDialup), BroadVoice and
Asterisk PBX respectively and they are all active and registered. When LINE1 is pressed, user will
hear dial tone and see “FWD”. When LINE2 is pressed, user will hear dial tone and see “BroadVoice”.
When LINE3 is pressed, user will hear dial tone and see “Asterisk PBX”.
For outgoing calls, GXV-3000 will pick up the LINE pressed, which the LED will be lit up in solid red
color. User can switch the dialing account before dialing any digits by pressing the same LINE button
one or more times. If user continues to press one LINE, the selected account will circulate among the
registered accounts. For example, when LINE1 is pressed, LCD displays “FWD”. If LINE1 is pressed
twice, LCD displays “BroadVoice” and the subsequent call will be made through SIP account 2.
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For incoming calls, if an account is configured and registered, all incoming calls for that account will
attempt to use its corresponding LINE if it is not in use. When the “virtually” mapped line is in use,
GXV-3000 will flash the next available LINE (from Left to Right) in red color.
A LINE is defined as “ACTIVE” when it is making or receiving a call, and its corresponding LINE
LED is lit up in solid RED.
4.3.3 Making Calls
There are four ways to make phone calls:
1. Make Handset/SPEAKER/Headset off hook, or press the available LINE key to select a SIP
account, the corresponding LINE LED will light up in solid red. Enter the phone numbers and
press the SEND key.
2. Make Handset/SPEAKER/Headset off hook, or press the available LINE key, the correspond-
ing LINE LED will light up in solid red. Press the SEND button to redial the last number
called.
3. Press the OK to bring up the Main Menu, select Phone Book, browse phone book to the person
you want to dial, press OK to select and OK again when Dial is selected. The call will dial out
in SPEAKER mode. (Right now this only applies to primary a/c which is LINE 1)
4. Paging/Intercom: This is ONLY valid if the SERVER/PBX support this feature and both
phone and PBX (for example, Asterisk) are configured correctly:
Make Handset/SPEAKER/Headset off hook, select the LINE key to related feature available
account, then press OK key again to see LCD showing: LINEx: PAGE USING. Now input the
phone number you want to Page/Intercom and press SEND key.
NOTE:
• Once off hook and key pressed, the dialed number will be displayed on the LCD and the corre-
sponding DTMF tone will be played out.
• If the “SEND” button is not pressed after the phone number pressed, the phone (by default) will
wait for 4 seconds (no key entry timeout) before sending all digits out and initiating the call.
4.3.4 Making Calls using IP Address
Direct IP calling allows two phones to talk to each other in an ad hoc fashion without a SIP proxy.
VoIP calls can be made between two phones if
• Both phones have public IP addresses, or
• Both phones are on a same LAN/VPN using private or public IP addresses, or
• Both phones can be connected through a router using public or private IP addresses (with nec-
essary port forwarding or DMZ)
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To make a direct IP calling, press “OK” button to bring up MAIN MENU, the select “Direct IP Call”
in the MENU and press OK to select, then key in the 12-digit target IP address, press OK key again to
call out.
Examples:
If the target IP address is 192.168.1.60 and port is 5062, like 192.168.1.60:5062, the key input to the
Menu UI will be:
192*168*1*60#5062
followed by OK key to dial out, where “*” key represent “.” and “#” key represent “:”
• Quick IP Call Mode:
This model has the ability to dial an IP address under the same LAN/VPN segment by simply pressing
the last octet in the IP address. This will allow user partially simulate a PBX functions using
CMSA/CD without a SIP server. When doing this, controlled static IP usage is recommended.
In the “Advanced Settings” page there is an option "Use Quick IP-call mode", by default it is set to No.
When this option is set to YES, and #XXX is dialed, where X is 0-9 and XXX <=255, phone will make
direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP address REGARDLESS
of subnet mask.
#XX or #X are also valid so leading 0 is not required (but OK).
eg.
192.168.0.2 calling 192.168.0.3 just dial #3 follow by SEND or #
192.168.0.2 calling 192.168.0.23 just dial #23 follow by SEND or #
192.168.0.2 calling 192.168.0.123 just dial #123 follow by SEND or #
192.168.0.2 dial #3 and #03 and #003 has same effect --> call 192.168.0.3
NOTE:
• If you have a SIP Server configured, Direct IP-IP call will still work. However, if you are using
STUN, Direct IP-IP call will also use STUN.
• When doing Direct IP calling, make sure the “Use Random Port” is configured as “NO”. Otherwise
it will not work unless the phone is configured to listen to the special port you specified and also
that port need to be inputted in the dialing digits.
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4.3.5 Receiving Calls
There are three states when GXV-3000 receives a call:
1. When receiving an initial call. Besides ringing with selected Ring Tone, the corresponding ac-
count LINE will flash in red, taking Handset/SPEAKER/Headset off hook will enable user to
hear the calling party in Handset/SPEAKER/Headset.
2. When receiving second or more incoming calls, besides playing stutter Call Waiting tone,
GXV-3000 will pick up the corresponding account LINE or the next available LINE as de-
scribed in section 4.3.2.
3. When the related configuration is enabled for Paging/Intercom and the PBX (or Server) also
4. supports this feature, the phone will beep once and automatically establish the call via
SPEAKER.
4.3.6 Call Hold
While in conversation, pressing the “HOLD” button will put the other party on hold. User can resume
the conversation by pressing the corresponding blinking LINE. User will also automatically put the
current line on “HOLD” by pressing another available LINE for making or receiving other phone calls.
4.3.7 Call Waiting and Switch between Calls
GXV-3000 can switch to another line for making or answering calls and automatically put an ACTIVE
call on Hold.
When receiving second or more incoming calls, besides playing a stutter Call Waiting tone, GXV-3000
will pick up the corresponding account or the next available LINE as described in section 4.3.2.
4.3.8 Call Transfer
GXV-3000 supports both BLIND and ATTENDED (or SUPERVISED) Transfer:
1. Blind Transfer: When a LINE is “ACTIVE”, user will get a dial tone by pressing the “TRNF”
button, then dial the number and press the “SEND” button, this will transfer the other party in
the corresponding LINE to the dialed number.
2. Attended (or Supervised) Transfer: When in conversation with an “ACTIVE” LINE as defined
in section 4.3.2, user shall press “Linex” button to make a call and thus automatically puts the
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current ACTIVE LINE on HOLD. Once the call established, user just press “TRNF” key to
transfer the call then hand up.
NOTE:
• Transferring calls across SIP domains needs to be supported by SIP service providers.
• Blind Transfer will usually use the Primary account SIP profile
4.3.9 3-Way Conferencing
GXV-3000 supports 3-way conferencing. With one LINE ACTIVE and another LINE on HOLD, press
the CONF button then the LINE that is on HOLD (blinking), this will join the three parties together in
a conference.
Once the conference established, press the CONF key will toggle between each other two sides of the
videos.
If after pressing the “CONF” button, a user decides not to conference anyone, user can cancel it and
resume the conversation by pressing CONF again or the original LINE button.
If the conference holder wishes to end a conference, simply press HOLD, which breaks the conference
and places both parties on hold. The conference holder user can then talk to each individual party by
selecting the corresponding blinking LINE.
4.3.10 Checking Message and Message Waiting Indication
When GXV-3000 is on-hook, pressing the MSG button will trigger the phone to call the
VoiceMail Server (VMS) configured for the primary account. If a line/account is selected first, it dials
the VMS configured for that account.
The MWI (Message Waiting Indicator) LED will flash in red color in three quarters of a second when
voicemail server sends message waiting information to GXV-3000.
NOTE:
• This feature requires the phone to be configured correctly in the “VoiceMail User ID” field the
VM portal access number, also requires the Server to support this.
4.3.11 Mute / Delete
When in conversation with an ACTIVE LINE, pressing “MUTE/DEL” will mute the conversation,
which means you can hear the other party but the other party cannot hear you. Pressing the button
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again will resume the conversation. When pressed, the red icon (muted microphone) will blink in the
LCD to remind call is muted.
When dialing a number or during the Key Pad “MAIN MENU” configuration mode, press
“MUTE/DEL” will delete the last entered digit.
When phone is idle, press the Mute/Delete key will function as a shortcut to Enable/Disable the DND
(Do Not Disturb). When enabled, the red icon (Do Not Disturb) will blink in the LCD to remind user
the DND is activated.
When there is an incoming call, press the Mute/Delete key while phone is ringing will REJECT the
incoming call. The server will tell caller the user is busy and/or forward the call to Voice Mail if server
support this feature and configured properly.
4.3.12 CAMERA BLOCK
User can press CAMERA BLOCK key to Disable/Enable the video sending out when in video call.
If no video call wanted (privacy), user can press this button and make Audio Only call. When CAM-
ERA BLOCK key is pressed, a red blocked camera will blink in the LCD to remind that the call is not
in video mode and the camera is blocked.
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4.4 Call Features
GXV-3000 series phone supports a list of call features: Caller ID Block (or Anonymous Call), Dis-
able/Enable Call Waiting, Call Forward on Busy, Delay, or Unconditional, etc.
Following table shows the call features of GXV-3000 series phone.
Key
*30
*31
*67
*82
*70
*71
*72
Call Features
Block Caller ID (for all subsequent calls)
Send Caller ID (for all subsequent calls)
Block Caller ID (per call)
Send Caller ID (per call)
Disable Call Waiting. (Per Call)
Enable Call Waiting (Per Call)
Unconditional Call Forward
To use this feature, dial “*72” and get the dial tone. Dial the forward
number and “#” for a dial tone, then hang up.
Cancel Unconditional Call Forward
To cancel “Unconditional Call Forward”, dial “*73” and get the dial
tone, then hang up.
*73
*90
*91
*92
*93
Busy Call Forward
To use this feature, dial “*90” and get the dial tone. Dial the forward
number and “#” for a dial tone, then hang up.
Cancel Busy Call Forward
To cancel “Busy Call Forward”, dial “*91” and get the dial tone, then
hang up.
Delayed Call Forward
To use this feature, dial “*92” and get the dial tone. Dial the forward
number and “#” for a dial tone, then hang up.
Cancel Delayed Call Forward
To cancel this Forward, dial “*93” and get the dial tone, then hang up.
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5
Configuration Guide
5.1 Configuration with Keypad
When the phone is on hook, press the OK button to enter MENU mode. When the phone goes off hook
or a call comes in, the phone automatically exits the MENU state and prepares for the call. It also exits
the MENU state if left idle for 20 seconds.
Here are the Menu options supported:
Here is the flow chart of keypad Menu configuration. During operation of the keypad, OK key is func-
tion as select or confirm key (similar to Enter in PC), UP and DOWN arrow keys are functioning as
browsing (by highlight the item selected), LEFT and RIGHT arrow keys are functioning as input selec-
tion, MUTE/DEL key is functioning as delete if input is wrong.
Table 5-1: Flow Chart of Key Pad Configuration Menu:
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5.2 Configuration with Web Browser
GXV-3000 series IP phone has an embedded Web server that will respond to HTTP/HTTPS
GET/POST requests. It also has embedded HTML pages that allow a user to configure the IP phone
through a Web browser such as Microsoft’s IE or Mozilla Firefox.
5.2.1 Access the Web Configuration Menu
The IP Phone Web Configuration Menu can be accessed by the following URI:
where the Phone-IP-Address is the IP address of the phone, which showed in the LCD screen.
5.2.2 End User Configuration
Once this HTTP request is entered and sent from a Web browser, the GXV-3000 will respond with the
following login screen:
Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2005-2006
The password is case sensitive with maximum length of 25 characters and the factory default password
for End User is “123”, for Administrator is “admin”. Only administrator has the privilege to access the
Advanced Setting and Account information. User trying to access without privilege will get an error.
After a correct password is entered in the login screen, the embedded Web server inside the GXV-3000
will respond with the Configuration page, which is explained in details below. Following is screen shot
of BASIC SETTINGS page when user using End User privilege login will see:
Grandstream Device Configuration
STATUS BASIC SETTINGS
ADVANCED SETTINGS
ACCOUNT 1 ACCOUNT 2 ACCOUNT 3
Web Access:
HTTP
HTTPS
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80
Web Port:
(default for HTTP is 80 and HTTPS is 443)
(purposely not displayed for security protection)
via DHCP (default) or PPPoE
End User Pass-
word:
IP Address:
dynamically
assigned
(will attempt PPPoE first if PPPoE setting is non-blank)
DHCP hostname:
DHCP domain:
Grandstream GXV-3000
DHCP vendor class ID:
PPPoE account ID:
PPPoE password:
PPPoE service name:
0
0
0
0
Preferred DNS server:
.
.
.
statically configured as:
IP Address:
192 168
0
0
0
0
0
160
0
.
.
.
.
.
.
.
.
.
.
.
0
0
0
0
0
0
0
0
Subnet Mask:
Default Router:
DNS Server 1:
DNS Server 2:
.
.
.
.
0
0
0
Time Zone:
GMT-5:00 (US Eastern Time, New York)
Daylight Savings
Time:
No
Yes (if set to Yes, display time will be 1 hour ahead of normal time)
24 HOUR
Time Display
Format:
12 HOUR
Year-Month-Day
Month-Day-Year
Day-Month-Year
Date Display
Format:
Display Clock
instead of Date:
No
600
Yes
LCD Screen Saver
Interval:
(in seconds. 0 means screen saver is off)
(in seconds. 0 means LCD will be always on)
(from 0-255, default is 128)
(from 0-255, default is 128)
(from 0-255, default is 128)
LCD Auto Power
Off Interval:
0
128
128
128
LCD Brightness:
LCD Contrast:
LCD Chroma
Saturation:
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LCD Background
0
0
0
Red (0-255)
Green (0-255)
Blue (0-255)
Color:
OSD Text Color:
Purple
Camera zoom
mode:
1.0:1 (w ide)
Normal
Camera Expo-
sure:
Camera Color
Mode:
Color
Auto
Monochrome
Fixed
Camera White
Balance:
Camera Lens
Correction:
No
Yes
TV Output:
NTSC
Update
Cancel
Reboot
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Web Access
Web Port
Select HTTP or secure HTTPS protocol for Web Access
By default, HTTP uses port 80 and HTTPS uses port 443. This field is for
customizable web port.
End User
Password
This contains the password to access the Web Configuration Menu. This
field is case sensitive with a maximum length of 25 characters.
IP Address
There are two modes under which the GXV-3000 can operate:
• If DHCP mode is enabled, then all the field values for the Static IP
mode are not used (even though they are still saved in the Flash
memory.) The GXV-3000 will acquire its IP address from the first
DHCP server it discovers from the LAN it is connected. The DHCP
option is reserved for NAT router mode (not implemented yet)
• To use the PPPoE feature the PPPoE account settings need to be set.
The GXV-3000 will attempt to establish a PPPoE session if any of
the PPPoE fields is set.
• If Static IP mode is enabled, then the IP address, Subnet Mask, De-
fault Router IP address, DNS Server 1 (primary), DNS Server 2 (sec-
ondary) fields will need to be configured. These fields are set to zero
by default.
Time Zone
This parameter controls how the date/time is displayed according to the
specified time zone.
Daylight Savings Time This parameter controls whether the time will be displayed in daylight sav-
ings time or not. If set to “Yes”, then the displayed time will be 1 hour ahead
of normal time.
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Time Display Format LCD time display in 12 hour or 24 hour format
Date Display Format Allow user to choose among the following three formats:
Year-Month-Day
Month-Day-Year
Day-Month-Year
Display Clock instead LCD displays clock if set to “Yes”, it will show clock in the LCD. Default is
of Date
No.
LCD Screen Saver In- Default value is 60 seconds, thus after 60 seconds of idle time the LCD
terval
Screen Saver will be enabled. Currently the screen saver file limitation is 20
images or 320kb whichever limit is reached first. Screen saver picture can be
changed the same way as customized ring tone.
LCD Auto Power Off Default value is 300 seconds, thus after 300 seconds of idle time the LCD
Interval
will automatically Power Off.
LCD Brightness
LCD Contrast
Default value is 128
Default value is 128
LCD Chroma Satura- Default value is 128
tion
LCD Background
Color
Select desired LCD Background color from the drop down box. Initial all is
0, can be adjusted to the color designed
OSD Text Color
Select desired On Screen Display Text color from drop down box. Default is
blue. The color can distinguish OSD from video background
Camera Zoom Mode Select desired Camera zoom mode (Tele, optical, digital). Can adjust local
video zoom either in mirror mode or during the call on the fly by press the
Left or Right Arrow Key
Camera Exposure
Select desired Camera Exposure mode based on surroundings.
Select desired Camera Color Mode (Color or Monochromatic)
Select desired Camera White Balance to be automatic or Fixed.
Camera Color Mode
Camera White Bal-
ance
Camera Lens Correc- Correct Camera Lens noise. Default is No. Recommend to set to YES to re-
tion
duce camera lens noise and improve video quality.
TV Output
Select desired TV Output type (PAL or NTSC).
In addition to the Basic Settings configuration page, end user also has access to the device Status page.
The following is a screen shot of the device Status page. Details are explained next.
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Grandstream Device Configuration
ADVANCED SETTINGS ACCOUNT 1 ACCOUNT 2 ACCOUNT 3
STATUS BASIC SETTINGS
Main-- 1.1 Rev C Interface-- 1.1 Rev A
00.0B.82.09.B9.DA
Hardware Revision:
MAC Address:
IP Address:
192.168.1.108
GXV3000
Product Model:
Software Version:
System Up Time:
Registered:
Application-- 1.0.0.33 Loader-- 1.0.0.3 Boot-- 1.0.0.13
9 day(s) 8 hour(s) 56 minute(s)
Account 1: Yes
Account 2: Yes
Account 3: No
disabled
PPPoE Link Up:
NAT:
73.123.74.110
NAT Mapped IP:
1
Total Inbound Calls:
Total Outbound Calls:
Total Missed Calls:
11
0
131
129
21939
20690
3212
Total SIP Message Sent:
Total SIP Message Received:
Total RTP Packet Sent:
Total RTP Packet Received:
Total RTP Packet Loss:
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Hardware Revision Hardware version number: Main Board, Interface Board
MAC Address
The device ID, in HEX format. This is a very important ID for ISP trouble-
shooting.
IP Address
This field shows IP address of GXV-3000
Product Model
Software Version
This field contains the product model information.
• Program: This is the main software (firmware) release number, always
used for identify the whole software (firmware) system of the phone.
• Loader: Driver loader code version number.
• Boot: Booting code version number
System Up Time
Registered
This field shows system up time since the last reboot.
This field indicates whether the accounts are registered to the related SIP
server(s). GXV-3000 can support three different SIP profile.
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PPPoE Link Up
This field shows whether the PPPoE connection is up if connected to DSL
modem.
Detected NAT Type This field shows what kind NAT router the GXV-3000 is connected to. It is
based on the result of STUN protocol resolution.
5.2.3 Advanced User Configuration
To login to the Advanced User Configuration page, please follow the instructions in section 5.2.1 to
get to the following login page. The password is case sensitive with a maximum length of 25 charac-
ters and the factory default password for Advanced User is “admin”.
Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2004
Advanced User configuration includes not only the end user configuration, but also advanced configu-
ration such as SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous con-
figuration. Following is a screen shot of the advanced configuration page:
Grandstream Device Configuration
STATUS
Admin Password:
G723 rate:
BASIC SETTINGS
ADVANCED SETTINGS
ACCOUNT 1 ACCOUNT 2 ACCOUNT 3
(purposely not displayed for security protection)
5.3kbps encoding rate
6.3kbps encoding rate
No Yes
(up to 10/20/32/64 for G711/G726/G723/other codecs respectively)
Silence Suppression:
Voice Frames per TX:
Video frame rate:
2
15 frames/second
128 kbps
Video bit rate (kbps):
Video Packet Size:
1400
(from 100 to 1400, default is 1400)
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Video fit to screen:
Enable Video Surveillance:
RTSP port:
Scaling
No
Cropping
Yes
554
(default is 554)
48
Layer 3 QoS:
(Diff-Serv or Precedence value)
0
0
Layer 2 QoS:
802.1Q/VLAN Tag
802.1p priority value
(0-7)
4
No Key Entry Timeout:
Use # as Dial Key:
(in seconds, default is 4 seconds)
No
Yes (if set to Yes, "#" will function as the "(Re-)Dial" key)
5004
local RTP port:
(1024-65535, default 5004)
Use random port:
No
Yes
20
keep-alive interval:
Use NAT IP
(in seconds, default 20 seconds)
(if specified, this will be used in SIP/SDP message)
stun.fw dnet.net
STUN server:
(URI or IP:port)
fm.grandstream.com/gs
Firmware Upgrade:
Upgrade Server:
Via
TFTP
HTTP
Automatic Upgrade:
10080
minutes (default 7 days)
No
Yes, check for upgrade every
101
DTMF Payload Type:
Syslog Server:
Syslog Level:
DEBUG
ntp.nasa.gov
NTP Server:
(URI or IP address)
Custom ring tone 1, used if incoming caller ID is
Custom ring tone 2, used if incoming caller ID is
Custom ring tone 3, used if incoming caller ID is
Distinctive Ring Tone:
Disable Call-Waiting:
No
No
No
Yes
Yes
Use Quick IP-call mode:
Lock keypad update:
Yes (configuration update via keypad is disabled if set to Yes)
Update Cancel Reboot
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Admin Pass-
word
Administrator password. Only administrator can configure the “Advanced Settings”
page. Password field is purposely left blank for security reason after clicking update
and saved. The maximum password length is 25 characters.
Silence Sup-
pression
This controls the silence suppression/VAD feature of G723 and G729. If set to
“Yes”, when a silence is detected, small quantity of VAD packets (instead of audio
packets) will be sent during the period of no talking. If set to “No”, this feature is
disabled.
Voice Frames This field contains the number of voice frames to be transmitted in a single Ethernet
per TX
packet (be advised the max. size of Ethernet packet is 1500 byte (or 120k bit) so
user should be aware that there IS a limit there). When setting this value, the user
should be aware of the requested packet time (ptime, used in SDP message) as a re-
sult of configuring this parameter. This parameter is associated with the first codec
in the above codec Preference List or the actual used payload type negotiated be-
tween the 2 conversation parties at run time.
e.g., if the first codec is configured as G723 and the “Voice Frames per TX” is set to
be 2, then the “ptime” value in the SDP message of an INVITE request will be 60ms
because each G723 voice frame contains 30ms of audio. Similarly, if this field is set
to be 2 and if the first codec chosen is G729 or G711 or G726, then the “ptime”
value in the SDP message of an INVITE request will be 20ms.
If the configured voice frames per TX exceeds the maximum allowed value, the IP
phone will use and save the maximum allowed value for the corresponding first co-
dec choice. The maximum value for PCM is 10 (x10ms) frames; for G726, it is 20
(x10ms) frames; for G723, it is 32 (x30ms) frames; for G729/G728, 64 (x10ms) and
64 (x2.5ms) frames respectively.
Please be very careful when massage those parameters. By adjust this, user also get
jitter buffer changed accordingly. BT-100 phone has patent dynamic jitter buffer
handling algorithm. The jitter buffer range from 20 ~ 200 ms.
Incorrect setting will affect voice quality so do not touch the parameter if not under-
stand and most of the case the default value will work in GS products.
Please refer to the Codec FAQ in our website for more technical details:
Video Frame Default value is 15 frames/second. User adjustable based on network condition. In-
Rate
crease frame rate will increase data significantly.
Video bit rate Default value is 128 kbps. User adjustable based on network environment.
(kbps)
Video Packet Default value is 1400, range from 100 to 1400
Size
Video fit to
Screen
When the received image size is bigger than the LCD resolution (QVGA), the
GXV3000 can Scale the image to fit the LCD or Crop it.
Enable Video Default is No. If configured to YES, the video phone can be turned to an video sur-
Surveillance
veillance camera once the phone rebooted.
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RTSP port
Default is 554. The port for video surveillance camera data stream.
Layer 3 QoS
This field defines the layer 3 QoS parameter which can be the value used for IP
Precedence or Diff-Serv or MPLS. Default value is 48.
Layer 2 QoS
This contains the value used for layer 2 VLAN tag. Default setting is
blank.
No Key Entry Default is 4 seconds.
Timeout
Use # as
Dial Key
This parameter allows users to configure the “#” key to be used as the “Send” (or
“Dial”) key. If set to “Yes”, pressing this key will immediately trigger the sending of
dialed string collected so far. In this case, this key is essentially equivalent to the
“(Re)Dial” key. If set to “No”, this “#” key will then be included as part of the dial
string to be sent out.
Local RTP
port
This parameter defines the local RTP-RTCP port pair the GXV-3000 will listen and
transmit. It is the base RTP port for channel 0. When configured, channel 0 will use
this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use
port_value+2 for RTP and port_value+3 for its RTCP. The default value is 5004.
Use Random
Port
This parameter, when set to Yes, will force random generation of both the local SIP
and RTP ports. This is usually necessary when multiple GXV-3000s are behind the
same NAT. Default is No.
Keep-alive in- This parameter specifies how often the GXV-3000 sends a blank UDP packet to the
terval
SIP server in order to keep the “hole” on the NAT open. Default is 20 seconds.
Use NAT IP
NAT IP address used in SIP/SDP message. Default is blank.
STUN Server IP address or Domain name of the STUN server. STUN resolution result will display
in the STATUS page of the Web UI.
Firmware Up- This radio button will enable GXV-3000 to download firmware or configuration file
grade
through either HTTP or local TFTP server. This is mutual exclusive choice.
Via TFTP
Server
This is the IP address of the configured TFTP server. If selected and it is non-zero or
not blank, the GXV-3000 will attempt to retrieve new configuration file or new code
image from the specified TFTP server at boot time. It will make up to 3 attempts be-
fore timeout and then it will start the boot process using the existing code image in
the Flash memory. If a TFTP server is configured and a new code image is retrieved,
the new downloaded image will be verified and then saved into the Flash memory.
Note: It is strongly recommended that user upgrade firmware locally in LAN envi-
ronment if using TFTP to upgrade. Please do NOT interrupt the TFTP upgrade proc-
ess (especially the power supply) as this will damage the device.
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Via HTTP
Server
The URL for the HTTP server used for firmware upgrade and configuration via
HTTP. For example,
Here “:6688” is the specific TCP port that the HTTP server is listening at, it can be
omitted if using default port 80.
Note: If Auto Upgrade is set to No, GXV-3000 will only do HTTP download once
at boot up.
Automatic
Upgrade
Default is No. Choose Yes to enable automatic HTTP upgrade and provisioning.
In “Check for upgrade every” field, enter the number of minutes to enable the phone
to check the HTTP server for firmware upgrade or configuration changes in the de-
fined period of time.
When set to No, the phone will only do HTTP upgrade and configuration check once
at boot up. Used by ITSP. End user should NOT touch these parameters.
DTMF Pay-
load Type
This parameter sets the payload type for DTMF using RFC2833. Default is 101.
Syslog Server The IP address or URL of System log server. This feature is especially useful for
ITSP (Internet Telephone Service Provider)
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Syslog Level
Select the ATA to report the log level. Default is NONE. The level is one of DE-
BUG, INFO, WARNING or ERROR. Syslog messages are sent based on the follow-
ing events:
•
•
•
•
•
•
•
•
•
•
product model/version on boot up (INFO level)
NAT related info (INFO level)
sent or received SIP message (DEBUG level)
SIP message summary (INFO level)
inbound and outbound calls (INFO level)
registration status change (INFO level)
negotiated codec (INFO level)
Ethernet link up (INFO level)
SLIC chip exception (WARNING and ERROR levels)
memory exception (ERROR level)
The Syslog uses USER facility. In addition to standard Syslog payload, it contains
the following components:
GS_LOG: [device MAC address][error code] error message
Here is an example: May 19 02:40:38 192.168.1.14 GS_LOG:
[00:0b:82:00:a1:be][000] Ethernet link is up
NTP server
This parameter defines the URI or IP address of the NTP (Network Time Protocol)
server which is used by GXV-3000 to display the current date/time.
Distinctive
Ring Tone
Customer Ring Tone 1 to 3 with associate Caller ID: when selected, if Caller ID is
configured, then the device will ONLY sound this ring tone when the incoming call
is from the Caller ID, device will use System Ring Tone for all other calls.
When selected but no Caller ID is configured, the selected ring tone will be used for
all incoming calls.
Disable Call
Waiting
Default is No. If set to Yes, the call waiting will be disabled.
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Use Quick IP This model has the ability to dial an IP address under the same LAN/VPN segment
Call Mode
by simply pressing the last octet in the IP address, without SIP server required.
In the Advanced Settings page there is an option "Use Quick IP-call mode", by de-
fault it is set to No. When this option is set to YES, and #XXX is dialed, where X is
0-9 and XXX <=255, phone will make direct IP call to aaa.bbb.ccc.XXX where
aaa.bbb.ccc comes from the local IP address REGARDLESS of subnet mask.
#XX or #X are also valid so leading 0 is not required (but OK).
eg.
192.168.0.2 calling 192.168.0.3 just dial #3 follow by SEND or #
192.168.0.2 calling 192.168.0.23 just dial #23 follow by SEND or #
192.168.0.2 calling 192.168.0.123 just dial #123 follow by SEND or #
192.168.0.2 dial #3 and #03 and #003 has same effect --> call 192.168.0.3
Note:- If you have a SIP Server configured, Direct IP-IP call will still work. How-
ever, if you are using STUN, Direct IP-IP call will also use STUN.
Default is set to No.
Lock keypad
update
If this parameter is set to “Yes”, the configuration changes via keypad are disabled
therefore the phone is some kind of locked.
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5.2.4 Individual Account Settings
Three independent SIP accounts each has its own configuration page. Their configurations are identi-
cal. The following is a screen shot of SIP Account 2 settings.
Grandstream Device Configuration
STATUS BASIC SETTINGS
ADVANCED SETTINGS
ACCOUNT 1 ACCOUNT 2 ACCOUNT 3
Account Active:
Yes
No
ABC Company
Account Name:
SIP Server:
(e.g., MyCompany)
sip.abc.com
(e.g., sip.mycompany.com, or IP address)
(e.g., proxy.myprovider.com, or IP address, if any)
(the user part of an SIP address)
proxy.abc.com
Outbound Proxy:
SIP User ID:
20001014084505168
20001014084505168
Authenticate ID:
Authenticate Password:
Name:
(can be identical to or different from SIP User ID)
(purposely not displayed for security protection)
(optional, e.g., John Doe)
4084505168
Use DNS SRV:
No
Yes
Yes
User ID is phone number:
SIP Registration:
Unregister On Reboot:
Register Expiration:
No
Yes
Yes
No
No
2
(in minutes. default 1 hour, max 45 days)
5062
local SIP port:
(default 5062)
NAT Traversal (STUN):
No
No
Yes
Yes
SUBSCRIBE for MWI:
Proxy-Require:
Voice Mail UserID:
Send DTMF:
4084505050
(User ID/extension for 3rd party voice mail system)
in-audio
No
via RTP (RFC2833)
via SIP INFO
Early Dial:
Yes (use "Yes" only if proxy supports 484 response)
Dial Plan Prefix:
(this prefix string is added to each dialed number)
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Enable Call Features:
No
180
Yes (if Yes, Call Forwarding & Call-Waiting-Disable are supported locally)
Session Expiration:
Min-SE:
(in seconds. default 180 seconds)
90
(in seconds. default and minimum 90 seconds)
No (Request for timer when making outbound calls)
No (When caller supports timer but did not request one)
No (Use timer even when remote party does not support)
Caller Request Timer:
Callee Request Timer:
Force Timer:
Yes
Yes
Yes
UAC Specify Refresher:
UAS Specify Refresher:
Force INVITE:
UAC
UAC
Yes
UAS
Omit (Recommended)
UAS (When UAC did not specify refresher tag)
No (Always refresh with INVITE instead of UPDATE)
No
Enable 100rel:
Yes
system ring tone
custom ring tone 1
custom ring tone 2
custom ring tone 3
Account Ring Tone:
Send Anonymous:
Auto Answer:
No
No
Yes (caller ID will be blocked if set to Yes)
Yes
Intercom/Paging
GSM
GSM
choice 1:
choice 2:
choice 3:
choice 4:
choice 5:
choice 6:
choice 7:
choice 8:
G.723.1
G.729A/B
PCMU
PCMA
PCMU
PCMA
Preferred Vocoder:
(in listed order)
Preferred Video Coder:
(in listed order)
H.264
H.264
choice 1:
choice 2:
Jitter Delay:
Medium
Enable Video:
No
Yes
99
H.264 payload type:
Special Feature:
(between 96 and 127, default is 99)
Standard
Update
Cancel
Reboot
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Account Active
This field indicates whether the account is active or not. The default value for
the primary account Account 1 is Yes. The default values for the other three
accounts are No.
Account Name
SIP Server
A name to identify an account which will be displayed in LCD.
SIP Server’s IP address or Domain name provided by VoIP service provider.
Outbound Proxy
IP address or Domain name of Outbound Proxy, or Media Gateway, or Session
Border Controller. Used by GXV-3000 for firewall or NAT penetration in dif-
ferent network environment. If symmetric NAT is detected, STUN will not
work and ONLY outbound proxy can provide solution for it.
SIP User ID
User account information, provided by VoIP service provider (ITSP), usually
has the form of digit similar to phone number or actually a phone number.
Authenticate ID
SIP service subscriber’s Authenticate ID used for authentication. Can be iden-
tical to or different from SIP User ID.
Authenticate Pass-
word
SIP service subscriber’s account password for GXV-3000 to register to (SIP)
servers of ITSP.
Name
SIP service subscriber’s name which will be used for Caller ID display.
Default is No. If set to Yes the client will use DNS SRV to look up server.
Use DNS SRV:
User ID is Phone
Number
If the GXV-3000 has an assigned PSTN telephone number, this field should
be set to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” pa-
rameter will be attached to the “From” header in SIP request
SIP Registration
This parameter controls whether the GXV-3000 needs to send REGISTER
messages to the proxy server. The default setting is “Yes”.
Unregister on Reboot Default is No. If set to yes, the SIP user’s registration information will be
cleared on reboot.
Register Expiration
This parameter allows user to specify the time frequency (in minutes) that
GXV-3000 refreshes its registration with the specified registrar. The default
interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes
(about 45 days).
Local SIP port
This parameter defines the local SIP port the GXV-3000 will listen and trans-
mit. The default value for Account 1 is 5060. It is 5062, 5064, 5066 for Ac-
count 2, Account 3 and Account 4 respectively.
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NAT Traversal
This parameter defines whether the GXV-3000 NAT traversal mechanism will
be activated or not. If activated (by choosing “Yes”) and a STUN server is also
specified, then the GXV-3000 will behave according to the STUN client speci-
fication. Under this mode, the embedded STUN client inside the GXV-3000
will attempt to detect if and what type of firewall/NAT it is sitting behind
through communication with the specified STUN server. If the detected NAT
is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the GXV-3000 will
attempt to use its mapped public IP address and port in all of its SIP and SDP
messages. If the NAT Traversal field is set to “Yes” with no specified STUN
server, the GXV-3000 will periodically (every 20 seconds or so) send a blank
UDP packet (with no payload data) to the SIP server to keep the “hole” on the
NAT open.
Subscribe for MWI: Default is No. When set to “Yes” a SUBSCRIBE for Message Waiting Indica-
tion will be sent periodically.
Proxy-Require
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Voice Mail User ID
When configured, user will be able to dial voice mail server by pressing
“MSG” button. This ID is usually the VM portal access number.
Send DTMF
This parameter specifies the mechanism to transmit DTMF digit. There are 3
modes supported: in audio which means DTMF is combined in audio signal
(not very reliable with low-bit-rate codec), via RTP (RFC2833), or via SIP
INFO.
Early Dial
Default is No. Use only if proxy supports 484 response.
Sets the prefix added to each dialed number.
Dial Plan Prefix
Enable Call Features Default is No. If set to Yes, Call transfer, Call Forwarding & Do-Not-Disturb
are supported locally provided ITSP’s SIP server supporting those features.
Session Expiration
Grandstream implemented SIP Session Timer. The session timer extension en-
ables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE,
or re-INVITE. Once the session interval expires, if there is no refresh via a
UPDATE or re-INVITE message, the session will be terminated.
Session Expiration is the time (in seconds) at which the session is considered
timed out, if no successful session refresh transaction occurs beforehand. The
default value is 180 seconds.
Min-SE
The minimum session expiration (in seconds). The default value is 90 sec-
onds.
Caller Request Timer If selecting “Yes” the phone will use session timer when it makes outbound
calls if remote party supports session timer.
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Callee Request Timer If selecting “Yes” the phone will use session timer when it receives inbound
calls with session timer request.
Force Timer
If selecting “Yes” the phone will use session timer even if the remote party
does not support this feature. Selecting “No” will allow the phone to enable
session timer only when the remote party support this feature.
To turn off Session Timer, select “No” for Caller Request Timer, Callee Re-
quest Timer, and Force Timer.
UAC Specify Re-
fresher
As a Caller, select UAC to use the phone as the refresher, or UAS to use the
Callee or proxy server as the refresher.
UAS Specify Re-
fresher
As a Callee, select UAC to use caller or proxy server as the refresher, or UAS
to use the phone as the refresher.
Force INVITE
Session Timer can be refreshed using INVITE method or UPDATE method.
Select “Yes” to use INVITE method to refresh the session timer.
Enable 100rel
The use of the PRACK (Provisional Acknowledgment) method enables reli-
ability to be offered to SIP provisional responses (1xx series). This is very im-
portant if PSTN internetworking is to be supported. A user’s wish to use reli-
able provisional responses is invoked by the 100rel tag which is appended to
the value of the required header of initial signalling messages.
Account Ring Tone
There are 4 different ring tone that are defined:
• System Ring Tone: when selected, all calls will ring with system ring
tone.
• Customer Ring Tone 1 to 3: when selected, GXV-3000 will ONLY
play this ring tone for all the incoming calls for this account.
Send Anonymous
Auto Answer
If this parameter is set to “Yes”, the “From” header in outgoing INVITE mes-
sage will be set to anonymous, essentially blocking the Caller ID from display-
ing.
When set to “Yes”, GXV-3000 will automatically switch on speaker to answer
the incoming call. Set to Intercom/Paging mode, it will answer the call based
on the SIP info header from the server. Default is No.
Preferred Vocoder
The GXV-3000 supports up to 5 different Vocoder types including G.711 A-
/U-law (nickname as PCMU/PCMA), GSM, G.723.1, G.729A/B.
User can configure Vocoders in a preference list that will be included with the
same preference order in SDP message. The first Vocoder in this list can be
entered by choosing the appropriate option in “Choice 1”. Similarly, the last
Vocoder in this list can be entered by choosing the appropriate option in
“Choice 8”.
Preferred Video
Coder
Currently GXV-3000 supports only H.264 codec. H.263 is not supported now
but will be supported in the future via firmware upgrade.
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Jitter Delay
Select desired Jitter Buffer Delay. Poor network suggested using High. Default
is Medium.
Enable Video
When set to Yes, video is enabled on calls, otherwise disabled.
H.264 payload type
Enter a desired value (96-127) for dynamic RTP payload type for H.264 codec.
Default is 99.
Special Feature
Default is Standard. Choose the selection to meet some special requirements
from Soft Switch vendors like Nortel, Broadsoft, etc.
5.2.5 Saving the Configuration Changes
Once a change is made, the user should press the “Update” button in the Configuration Menu. The IP
phone will then display the following screen to confirm that the changes have been saved:
Grandstream Device Configuration
STATUS BASIC SETTINGS
ADVANCED SETTINGS
ACCOUNT 1 ACCOUNT 2 ACCOUNT 3
Your configuration changes have been saved.
They will take effect on next reboot.
Reboot
All Rights Reserved Grandstream Networks, Inc. 2005-2006
User is recommended to power cycle the IP phone after seeing the above message.
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5.2.6 Rebooting the Phone from Remote
The administrator of the phone can remotely reboot the phone by pressing the “Reboot” button at the
bottom of the configuration menu. Once done, the following screen will be displayed to indicate that
rebooting is underway.
Grandstream Device Configuration
The device is rebooting now...
You may relogin by clicking on the link below in 30 seconds.
Click to relogin
All Rights Reserved Grandstream Networks, Inc. 2004
At this point, user can relogin to the phone after waiting for about 30 seconds.
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5.3 Configuration through Central Provisioning Server
Grandstream GXV-3000 can be automatically configured from a central provisioning system.
When GXV-3000 boots up, it will send TFTP or HTTP request to download configuration file which is
“cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXV-3000.
The configuration file can be downloaded via TFTP or HTTP from the central server. A service pro-
vider or an enterprise with large deployment of GXV-3000 can easily manage the configuration and
service provisioned for individual device from a central server remotely.
Grandstream provides a licensed provisioning system called GAPS that can be used to support auto-
mated configuration of GXV-3000. GAPS (Grandstream Automated Provisioning System) uses en-
hanced (NAT friendly) TFTP or HTTP (thus no NAT issues) and other communication protocols to
communicate with each individual GXV-3000 for firmware upgrade, remote reboot, etc.
Grandstream provide GAPS (Grandstream Automated Provisioning System) service to VoIP service
providers. It could be either simple redirection or with certain special provisioning settings. Initially
upon booting up, Grandstream devices by default point to Grandstream provisioning server GAPS,
based on the unique MAC address of each device, GAPS provision the devices with redirection set-
tings so that they will be redirected to customer’s TFTP or HTTP server for further provisioning.
Grandstream also provides GAPSLite software package which contains our NAT friendly TFTP server
and a configuration tool to facilitate the task of generating device configuration files.
The GAPSLite configuration tool is now free to end users. The tool and configuration templates can be
downloaded from http://www.grandstream.com/DOWNLOAD/Configuration_Tool/.
For details on how GAPS works, please refer to the documentation of GAPS product or contact Grand-
stream Sales Department for more information.
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6
Software Upgrade & Customization
Software (or firmware) upgrade can be done via either TFTP or HTTP. The corresponding configura-
tion settings are in the ADVANCED SETTINGS configuration page.
6.1 Firmware Upgrade through TFTP/HTTP
To upgrade via TFTP or HTTP, user needs to select TFTP or HTTP upgrade method. “Upgrade
Server” needs to be set to a valid URL of a HTTP server. Server name can be in either FQDN or IP
address format. Here are examples of some valid URL.
e.g. firmware.mycompany.com:6688/Grandstream/1.0.0.18
e.g. 168.75.215.189
There are two ways to set up the Upgrade Server to upgrade firmware, namely through the Key Pad
Menu or via the GXV-3000’s Web configuration interface. To configure the Upgrade Server via Key
Pad Menu options, select “System Config” from the Main Menu, then select “Upgrade Preferences”.
Under this sub Menu, user can edit Upgrade Server in either an IP address format or FQDN format.
Choose “Save and use TFTP” or “Save and use HTTP” to select upgrade method. And select “Re-
boot” from the Main Menu to reboot the phone.
To configure the Upgrade Server via the Web configuration interface, open up your web browser and
enter the IP address of GXV-3000. Enter the admin password to get into the web configuration inter-
face. In the ADVANCD SETTINGS page, enter the Upgrade Server’s IP address or FQDN in the
“Upgrade Server” field. Select TFTP or HTTP upgrade method. Once these settings are entered, user
needs to update the change by clicking the “Update” button. Then “Reboot” or power cycle the phone,
the firmware files will be fetched upon booting up.
If the configured upgrade server is found and a new code image is available, the GXV-3000 phone will
attempt to retrieve the new image files by downloading them into the phone’s SRAM. During this
stage, the GXV-3000 phone’s LCD will show firmware file downloading process. Upon verification of
checksum, the new code image will be saved into the Flash. If firmware upgrade fails for any reason
(e.g., TFTP/HTTP server is not responding, there are no code image files available for upgrade, or
checksum test fails, etc), the GXV-3000 phone will stop the upgrading process and simply boot using
the existing code image in the flash.
Firmware upgrading may take as long as 45 minutes over Internet, or just 5 minutes on a controlled
LAN. It is strongly recommended to conduct firmware upgrade in a controlled LAN environment
whenever possible.
For users who do not have local TFTP server, Grandstream provides a NAT-friendly TFTP server on
the public Internet for users to download the latest firmware upgrade automatically. Please check the
Services section of Grandstream’s Web site to obtain this TFTP server IP address:
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Alternatively, user can download and install a free TFTP or HTTP server in his/her LAN to do firm-
ware upgrading.
A free Windows version TFTP server can be downloaded from:
Our latest official release can be downloaded from:
Unzip the file and put all of them under the root directory of the TFTP server. Put the PC running the
TFTP server and the GXV-3000 phone in the same LAN segment. Please go to File -> Configure ->
Security to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the
firmware upgrade. Start the TFTP server, in the phone’s web configuration page, configure
the Firmware Server Path with the IP address of the PC, update the change and reboot the unit.
crosoft IIS web server
NOTE:
• When GXV-3000 phone boots up, it will send TFTP or HTTP request to download configura-
tion file “cfg000b82xxxxxx”, where “000b82xxxxxx” is the MAC address of the GXV-3000
phone. This file is for initial automatically provisioning purpose only, for normal TFTP or
HTTP firmware upgrade, the following error messages in a TFTP or HTTP server log can be
ignored.
TFTP Error from [IP ADRESS] requesting cfg000b82023dd4 : File does not exist
Configuration File Download
NOTE:
• After flashing the firmware, a power cycle is required. Please power cycle the phone after
the downloading process finished and when the LCD becomes pale or white. For new
hardware version 1.1 or later, the phone will automatically reboot after the new firmware
get flashed.
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6.2 Configuration File Download
Grandstream SIP Device can be configured via Web Interface as well as via Configuration File
through TFTP or HTTP. “Upgrade Server” is the TFTP or HTTP server path for configuration file. It
needs to be set to a valid URL, either in FQDN or IP address format.
A configuration parameter is associated with each particular field in the web configuration page. A
parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric
numbers. i.e., P2 is associated with “Admin Password” in the ADVANCED SETTINGS page. For a
detailed parameter list, please refer to the corresponding configuration template of the firmware.
When Grandstream Device boots up or reboots, it will issue request for configuration file named
“cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the MAC address of the device, i.e.,
“cfg000b820102ab”. The configuration file name should be in lower cases.
6.3 Managing Firmware and Configuration File Download
When “Automatic Upgrade” is set to “Yes”, Service Provider can use P193 (Auto Check Interval, in
minutes, default and minimum is 60 minutes) to have the devices periodically check with either Up-
grade Server. This allows the device periodically check if there are any new changes need to be taken
on a scheduled time. By defining different intervals in P193 for different devices, Server Provider can
spread the Firmware or Configuration File download in minutes to reduce the Firmware or Provision-
ing Server load at any given time.
6.4 Customization own Screensaver Images
User can customize the screensaver images by using the free “screensaver picture generator tool”
downloadable from our website:
User can put own preferred JPEG images into the phone as screensaver. Here are the requirements of
the pictures which can be converted as screensaver:
• Baseline JPEG
• 320x240 in dimension
• RGB color space
• 8-bit data
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There is a limitation for the flash size to hold the screensaver image file (image.bin) also. Currently
the screensaver file (image.bin) limitation is 20 images or 320kb whichever limit is reached first.
Screensaver picture can be changed the same way as customized ring tone.
Users are recommended to resize the picture using popular photo processing software. Following
screen shots are captured from Adobe Photoshop to resize image to prepare the screensaver images.
• Picture MUST be preprocessed and resized to 320 x 240
• Picture must be saved in Baseline JPEG format.
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Here is the screen shot of the Screensaver Tool Utility:
Please keep in mind the limitation of the image.bin file. Once the file is generated by this utility tool,
please put the file into the same folders containing firmware files.
The screensaver image flashing follows the same way of firmware flashing, either TFTP or HTTP. But
if the firmware is already the latest one, user can just put the only one file “image.bin” there to allow
the phone to get it.
Be advised do NOT interrupt the power supply when doing ANY file flashing as that will risk damage
the phone. Users with high frequent power outage are recommended to have UPS hooked on before
doing any firmware or screensaver or ring tone flashing.
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6.5 Customization own Ring Tones
User can customize the ring tone using the free tool downloadable from our website:
Please follow the same rules as the Screensaver to generate the ring tone files and load them into the
phone.
Here are the requirements of the files which can be converted as ring tones:
• Must be either .wav or .mp3 format, recommended .wav format
• Converted file size must be less than 64KB
Here is the screen shot of the ring tone generating tools:
Since the phone only can hold 3 ring tones files in total with each less than 64KB in file size. If user
loading a ring tone file bigger than 64KB in size, it will occupy next available ring tone file slot and
the total ring tones WILL be reduced accordingly. If the big file is “ring3.bin”, the ring tone will get
dropped and can not be loaded into the phone.
User can not change the file name generated to other names except the ring1.bin, ring2.bin and
ring3.bin.
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7
Auxiliary Ports
7.1 USB 2.0 Port
When a USB device is plugged into one of the two USB 2.0 ports, an icon will show up in the lower
part of the LCD to illustrate the connected device. Following is the snap shot of the LCD when an USB
device is plugged into the interface:
Figure 7-1:
(Remark: This screen shot is NOT correct, just put here for space occupation temporarily)
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7.1.1 Capture pictures via USB port
When a USB flash drive is plugged into the USB port and icon showed up to illustrate the drive is
ready, user can snap shot or capture the picture showed on the LCD and save it into the flash disk.
Here are the procedures: When the USB flash disk device is ready, whether the video phone is in cam-
era local loopback mode or in a live video call, just press “OK” button once, the LCD will freeze 1 or 2
seconds, the captured picture will be saved in the disk using a format like: “gxv_xxxx.yuv”.
Only one picture can be saved at one time.
User needs special decoding software to see the captured pictures stored in the USB flash disk. A free
software called XnView can be downloaded from:
Once this software installed, please choose file type “YUV - YUV 4:2:2” to open and see the captured
pictures.
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7.2 RCA style stereo audio & composite video output
The picture below shows the connection of GXV-3000 to TV set which allows use to watch the video
and hear the audio on home TV set.
Figure 7-2:
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7.3 Headset Jack
The picture below shows the handset and headset connectors’ wiring schema of GXV-3000.
Figure 7-3:
As show in the schema, the left side is pin assignment of a RJ-22 handset plug; while the right side is
showing a normal 2.5mm headset plug. If use want to connect the phone to external Speaker or Inter-
com/Paging system, necessary adaptors are required from local electronic stores.
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8
Video Surveillance
GXV-3000 can be turned into a video surveillance camera. It works very well if the monitor device
and the GXV-3000 are in the same LAN. If the monitor device is out side the router in another LAN
through two routers, unless tweaks the routers correctly, it is very hard to make that working correctly
and reliably. This is mostly caused by the routers involved.
We suggest this kind of application in LAN environment or both sides have public IP.
Here are procedures of how to configure this feature working:
¾ Phone side:
In the ADVANCED SETTING page, find the following field and change from default setting NO
to YES, reboot the device.
¾ PC side (Monitor Device):
and support RFC 3984.
2. Launch VLC.
3. Go to Preferences->Input/Codecs->Demuxers->H264, check “Advanced options” in the bot-
tom. The option “Frames per Second” will show. Change that value to 5 and then save.
4. Go to Preferences->Input/Codecs->Access modules->Real RTSP, check “Advanced options” in
the bottom. The option “Caching value (ms) will show. Change that value to 1000 and then
save. You may change it to a smaller value to reduce the delay.
5. If the viewer is under NAT, go to Preferences->Demuxers->Access modules->RTP/RTSP,
check “Advanced options” in the bottom. The option “Use RTP over RTSP (TCP)” will show.
Check that option box. (We do NOT recommend this network environment as the router will
cause unreliable issues)
6. Close the Preferences window and go to File->Open Network Stream:
a) Select RTSP as the protocol
b) Enter the URL in the format of
rtsp://admin:ADMIN_PASSWORD@DEVICE_IP_ADDRESS:DEVICE_RTSP_PORT.
You need to change the text in red according to your configuration:
ADMIN_PASSWORD is the device’s web configuration password for admin.
DEVICE_IP_ADDRESS is the device IP.
DEVICE_RTSP_PORT is the RTSP port setting of the device.
If the port uses default value 554, the port portion can be omitted from the URL
c) Click OK and you should see the video starts
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There are several limitations:
1. Currently it supports only 2 concurrent views on GXV. The video stream is set at 5 frame per sec-
ond and 128kbps.
2. The video stream will stop if a call starts. The phone’s video stream server will not actively termi-
nate the session, but most clients will tear down the session after certain interval without video
stream.
Following is a sample screen shot of the PC client side running VLC as monitoring station:
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Restore Factory Default Setting
WARNING !!!
Restore the Factory Default Setting will DELETE all configuration information of the phone.
Please BACKUP or PRINT out all the settings before you approach to following steps. Grand-
stream will not take any responsibility if you lose all the parameters of setting and cannot connect to
your VoIP service provider.
Please disconnect network cable and power cycle the unit before trying to reset the unit to factory de-
fault. The steps are as follows:
Step 1:
Press “OK” key to bring up the key pad configuration UI menu, select “System Config”, press “OK” to
enter submenu, select “Factory Reset” (Please refer to Table 5-1 of keypad flow chart)
Step 2:
Key in the MAC address which printed on the bottom of the sticker. Please use the following mapping:
0-9: 0-9
A: 22
(when pressed 2 twice, the “A” letter will show on the LCD)
B: 222
C: 2222
D: 33
E: 333
F: 3333
For example, if the MAC address is 000b8200e395, it should be key in as
“0002228200333395”.
NOTE:
• If there is digits like “22” in the MAC, you need to type “2” then press “->” right arrow key to
move the cursor or wait for 4 seconds to continue to key in another “2”.
Step 3:
Press the “OK” key again to move the cursor to “OK” button then press “OK” key again to confirm. It
the MAC address you input is correct, the phone will reboot now. Otherwise it will exist to previous
keypad menu interface.
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10 Glossary of Terms
ADSL
AGC
Asymmetric Digital Subscriber Line: Modems attached to twisted pair copper wiring that
transmit from 1.5 Mbps to 9 Mbps downstream (to the subscriber) and from 16 kbps to 800
kbps upstream, depending on line distance.
Automatic Gain Control, is an electronic system found in many types of devices. Its purpose is
to control the gain of a system in order to maintain some measure of performance over a
changing range of real world conditions.
ARP
ATA
Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826], pacifi-
cally IPv4, to map IP network addresses to the hardware addresses used by a data link protocol.
The protocol operates below the network layer as a part of the interface between the OSI net-
work and OSI link layer. It is used when IPv4 is used over Ethernet
Analogue Telephone Adapter. Covert analogue telephone to be used in data network for VoIP,
like Grandstream HT series products.
CODEC
Abbreviation for Coder-Decoder. It's an analog-to-digital (A/D) and digital-to-analog (D/A)
converter for translating the signals from the outside world to digital, and back again.
CNG
Comfort Noise Generator, generate artificial background noise used in radio and wireless
communications to fill the silent time in a transmission resulting from voice activity detection.
DATAGRAM
A data packet carrying its own address information so it can be independently routed from its
source to the destination computer
DECIMATE
To discard portions of a signal in order to reduce the amount of information to be encoded or
compressed. Lossy compression algorithms ordinarily decimate while subsampling.
DECT
Digital Enhanced Cordless Telecommunications: A standard developed by the European Tele-
communication Standard Institute from 1988, governing pan-European digital mobile teleph-
ony. DECT covers wireless PBXs, telepoint, residential cordless telephones, wireless access to
the public switched telephone network, Closed User Groups (CUGs), Local Area Networks,
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and wireless local loop. The DECT Common Interface radio standard is a multicarrier time di-
vision multiple access, time division duplex (MC-TDMA-TDD) radio transmission technique
using ten radio frequency channels from 1880 to 1930 MHz, each divided into 24 time slots of
10ms, and twelve full-duplex accesses per carrier, for a total of 120 possible combinations. A
DECT base station (an RFP, Radio Fixed Part) can transmit all 12 possible accesses (time slots)
simultaneously by using different frequencies or using only one frequency. All signaling in-
formation is transmitted from the RFP within a multiframe (16 frames). Voice signals are digi-
tally encoded into a 32 Kbit/s signal using Adaptive Differential Pulse Code Modulation.
DNS
DID
Short for Domain Name System (or Service or Server), an Internet service that translates
domain names into IP addresses
Direct Inward Dialing
Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without go-
ing through an attendant or auto-attendant.
DSP
Digital Signal Processing. Using computers to process signals such as sound, video, and other
analog signals which have been converted to digital form.
Digital Signal Processor. A specialized CPU used for digital signal processing.
Grandstream products all have DSP chips built inside.
DTMF
Dual Tone Multi Frequency
The standard tone-pairs used on telephone terminals for dialing using in-band signaling. The
standards define 16 tone-pairs (0-9, #, * and A-F) although most terminals support only 12 of
them (0-9, * and #).
FQDN
FXO
Fully Qualified Domain Name
A FQDN consists of a host and domain name, including top-level domain. For example,
www.grandstream.com is a fully qualified domain name. www is the host, grandstream is the
second-level domain, and com is the top level domain.
Foreign eXchange Office
An FXO device can be an analog phone, answering machine, fax, or anything that handles a
call from the telephone company like AT&T. They should also operate the same way when
connected to an FXS interface.
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An FXO interface will accept calls from FXS or PSTN interfaces. All countries and regions
have their own standards.
FXO is complimentary to FXS (and the PSTN).
FXS
Foreign eXchange Station
An FXS device has hardware to generate the ring signal to the FXO extension (usually an ana-
log phone).
An FXS device will allow any FXO device to operate as if it were connected to the phone com-
pany. This makes your PBX the POTS+PSTN for the phone.
The FXS Interface connects to FXO devices (by an FXO interface, of course).
DHCP
The Dynamic Host Configuration Protocol (DHCP) is an Internet protocol for automating the
configuration of computers that use TCP/IP. DHCP can be used to automatically assign IP ad-
dresses, to deliver TCP/IP stack configuration parameters such as the subnet mask and default
router, and to provide other configuration information such as the addresses for printer, time
and news servers.
ECHO CANCELLATION
Echo Cancellation is used in telephony to describe the process of removing echo from a voice
communication in order to improve voice quality on a telephone call. In addition to improving
quality, this process improves bandwidth savings achieved through silence suppression by
preventing echo from traveling across a network.
There are two types of echo of relevance in telephony: acoustic echo and hybrid echo. Speech
compression techniques and digital processing delay often contribute to echo generation in
telephone networks.
H.323
H.264
A suite of standards for multimedia conferences on traditional packet-switched networks.
H.264, MPEG-4 Part 10, or AVC, for Advanced Video Coding, is a digital video codec
standard which is noted for achieving very high data compression. It was written by the ITU-T
Video Coding Experts Group (VCEG) together with the ISO/IEC Moving Picture Experts
Group (MPEG) as the product of a collective partnership effort known as the Joint Video Team
(JVT). The ITU-T H.264 standard and the ISO/IEC MPEG-4 Part 10 standard (formally,
ISO/IEC 14496-10) are technically identical. The final drafting work on the first version of the
standard was completed in May of 2003.
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The intent of H.264/AVC has been to create a standard that would be capable of providing
good video quality at bit rates that are substantially lower (e.g., half or less) than what previous
standards would need (e.g., relative to MPEG-2, H.263, or MPEG-4 Part 2).
HTTP
IP
Hyper Text Transfer Protocol; the World Wide Web protocol that performs the request and re-
trieve functions of a server
Internet Protocol. A packet-based protocol for delivering data across networks.
IP-PBX
IP-based Private Branch Exchange
IP Telephony
(Internet Protocol telephony, also known as Voice over IP Telephony) A general term for the
technologies that use the Internet Protocol's packet-switched connections to exchange voice,
fax, and other forms of information that have traditionally been carried over the dedicated cir-
cuit-switched connections of the public switched telephone network (PSTN). The basic steps
involved in originating an IP Telephony call are conversion of the analog voice signal to digital
format and compression/translation of the signal into Internet protocol (IP) packets for trans-
mission over the Internet or other packet-switched networks; the process is reversed at the re-
ceiving end. The terms IP Telephony and Internet Telephony are often used to mean the same;
however, they are not 100 per cent interchangeable, since Internet is only a subcase of packet-
switched networks. For users who have free or fixed-price Internet access, IP Telephony soft-
ware essentially provides free telephone calls anywhere in the world. However, the challenge
of IP Telephony is maintaining the quality of service expected by subscribers. Session border
controllers resolve this issue by providing quality assurance comparable to legacy telephone
systems.
IVR
IVR is a software application that accepts a combination of voice telephone input and touch-
tone keypad selection and provides appropriate responses in the form of voice, fax, callback, e-
mail and perhaps other media.
MTU
A Maximum Transmission Unit (MTU) is the largest size packet or frame, specified in octets
(eight-bit bytes), that can be sent in a packet- or frame-based network such as the Internet. The
maximum for Ethernet is 1500 byte.
NAT
NTP
Network Address Translation
Network Time Protocol, a protocol to exchange and synchronize time over networks
The port used is UDP 123
Grandstream products using NTP to get time from Internet
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OBP/SBC
Outbound Proxy or another name Session Border Controller. A device used in VoIP networks.
OBP/SBCs are put into the signaling and media path between calling and called party. The
OBP/SBC acts as if it was the called VoIP phone and places a second call to the called party.
The effect of this behavior is that not only the signaling traffic, but also the media traffic
(voice, video etc) crosses the OBP/SBC. Without an OBP/SBC, the media traffic travels
directly between the VoIP phones. Private OBP/SBCs are used along with firewalls to enable
VoIP calls to and from a protected enterprise network. Public VoIP service providers use
OBP/SBCs to allow the use of VoIP protocols from private networks with internet connections
using NAT.
PPPoE
PSTN
Point-to-Point Protocol over Ethernet, is a network protocol for encapsulating PPP frames in
Ethernet frames. It is used mainly with cable modem and DSL services.
Public Switched Telephone Network
i.e. the phone service we use for every ordinary phone call, or called POT (Plain Old Tele-
phone), or circuit switched network.
RTCP
RTP
Real-time Transport Control Protocol, defined in RFC 3550, a sister protocol of the Real-time
Transport Protocol (RTP), It partners RTP in the delivery and packaging of multimedia data,
but does not transport any data itself. It is used periodically to transmit control packets to
participants in a streaming multimedia session. The primary function of RTCP is to provide
feedback on the quality of service being provided by RTP.
Real-time Transport Protocol defines a standardized packet format for delivering audio and
video over the Internet. It was developed by the Audio-Video Transport Working Group of the
IETF and first published in 1996 as RFC 1889
SDP
SIP
Session Description Protocol, is a format for describing streaming media initialization
parameters. It has been published by the IETF as RFC 2327.
Session Initiation Protocol, An IP telephony signaling protocol developed by the IETF
(RFC3261). SIP is a text-based protocol suitable for integrated voice-data applications. SIP is
designed for voice transmission and uses fewer resources and is considerably less complex than
H.323.
All Grandstream products are SIP based
STUN
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Simple Traversal of UDP over NATs, is a network protocol allowing clients behind NAT (or
multiple NATs) to find out its public address, the type of NAT it is behind and the internet side
port associated by the NAT with a particular local port. This information is used to set up UDP
communication between two hosts that are both behind NAT routers. The protocol is defined in
RFC 3489. STUN will usually work well with non-symmetric NAT routers.
TCP
Transmission Control Protocol, is one of the core protocols of the Internet protocol suite. Using
TCP, applications on networked hosts can create connections to one another, over which they
can exchange data or packets. The protocol guarantees reliable and in-order delivery of sender
to receiver data.
TFTP
UDP
Trivial File Transfer Protocol, is a very simple file transfer protocol, with the functionality of a
very basic form of FTP; It uses UDP (port 69) as its transport protocol.
User Datagram Protocol (UDP) is one of the core protocols of the Internet protocol suite. Using
UDP, programs on networked computers can send short messages known as datagrams to one
another. UDP does not provide the reliability and ordering guarantees that TCP does;
datagrams may arrive out of order or go missing without notice. However, as a result, UDP is
faster and more efficient for many lightweight or time-sensitive purposes.
VAD
Voice Activity Detection or Voice Activity Detector is an algorithm used in speech processing
wherein, the presence or absence of human speech is detected from the audio samples.
VLAN
VoIP
A virtual LAN, known as a VLAN, is a logically-independent network. Several VLANs can co-
exist on a single physical switch. It is usually refer to the IEEE 802.1Q tagging protocol.
Voice over IP
VoIP encompasses many protocols. All the protocols do some form of signaling of call capa-
bilities and transport of voice data from one point to another. e.g.: SIP, H.323, etc.
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