Grandstream Networks, Inc.
Analog IP Gateway GXW-410x
4 or 8 FXO Ports w/Video Surveillance
GXW-410x User’s Manual
Firmware Version 1.0.0.53
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TABLE OF FIGURES
GXW-410X USER MANUAL
FIGURE 1: DIAGRAM OF GXW-410X BACK PANEL ............................................................................................6
FIGURE 2: DIAGRAM OF GXW-410X DISPLAY PANEL........................................................................................7
FIGURE 3: FUNCTIONAL DIAGRAM OF IP-PBX & GXW-410X ............................................................................8
FIGURE 4: SCREEN-SHOT OF GXW-410X LOG-IN SCREEN..............................................................................13
FIGURE 5: SCREEN-SHOT OF SAVE CONFIGURATION ......................................................................................26
FIGURE 6: SCREEN-SHOT OF REBOOTING ......................................................................................................26
FIGURE 7: SCREEN-SHOT OF VIDEO SURVEILLANCE*......................................................................................28
APPLICATION ONE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER........................................................32
APPLICATION TWO: GXW TO EXTEND A TRADITIONAL PBX SCENARIO ............................................................32
APPLICATION FOUR: USING A GXW FOR PURE IP- IP COMMUNICATION CONFIGURATION.................................33
TABLE OF TABLES
GXW-410X USER MANUAL
TABLE 1: DEFINITIONS OF THE GXW CONNECTORS.........................................................................................6
TABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL ......................................................................................7
TABLE 3: GXW-410X SOFTWARE FEATURES ................................................................................................11
TABLE 4: HARDWARE SPECIFICATION OF GXW-410X.....................................................................................10
TABLE 6: WEB LOG-IN DEFINITIONS...............................................................................................................13
TABLE 7: STATUS PAGE DEFINITIONS ............................................................................................................15
TABLE 8: ADVANCED CONFIGURATION PAGE DEFINITIONS ..............................................................................16
TABLE 9: FXO LINES CONFIGURATION DEFINITIONS .......................................................................................19
TABLE 10: CHANNELS PAGE DEFINITIONS ......................................................................................................21
TABLE 11: GRANDSTREAM CONFIGURATION PAGE DEFINITIONS......................................................................23
GUI INTERFACES
GXW-410X USER MANUAL
1. SCREENSHOT OF ADVANCED SETTINGS CONFIGURATION PAGE
2. SCREENSHOT OF BASIC SETTINGS CONFIGURATION PAGE
3. SCREENSHOT OF CHANNELS CONFIGURATION PAGE
4. SCREENSHOT OF FXO LINES CONFIGURATION PAGE
5. SCREENSHOT OF PROFILE 1 CONFIGURATION PAGE
6. SCREENSHOT OF STATUS CONFIGURATION PAGE
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WELCOME
Thank you for purchasing the Grandstream GXW–410x IP Analog FXO Gateway. The GXW–410x is a
cost effective, easy to use and easy to configure IP communications solution for any business. The
GXW–410x supports popular voice codecs and is designed for full SIP compatibility and interoperability
with 3rd party SIP providers, thus enabling you to fully leverage the benefits of VoIP technology, integrate
a traditional phone system into a VoIP network, and efficiently manage communication costs.
This manual will help you learn how to operate and manage your GXW FXO Analog IP Gateway and
make the best use of its many upgraded features including simple and quick installation, multi-party
conferencing, etc. This IP Analog Gateway is very easy to manage and scalable, specifically designed to
be an easy to use and affordable VoIP solution for the small – medium business or enterprise. Enable
the video surveillance port to give piece of mind while you are away from your business.
Gateway GXW-410x Overview
The GXW410x offers an easy to manage, feature rich IP communications solution for any small business
or businesses with virtual and/or branch locations who want to leverage their broadband network and/or
add new IP Technology to their current phone system.
The Grandstream Enterprise Analog VoIP
Gateway GXW410x series converts SIP/RTP IP calls to traditional PSTN calls and vice versa. There are
two models - the GXW-4104 and GXW-4108, which have either 4 or 8 FXO ports respectively. The
installation is the same for either model.
Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation
of this product in any way other than as detailed by this User Manual, could void your manufacturer
warranty.
•
•
This document is contains links to Grandstream GUI Interfaces. Please remember to download these
This document is subject to change without notice. The latest electronic version of this user manual
is available for download from the following location:
•
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print,
for any purpose without the express written permission of Grandstream Networks, Inc. is not
permitted.
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PACKAGING
Unpack and check all accessories. Equipment included in the package:
1) One GXW-410x Unit
2) One universal power adaptor
3) One Ethernet cable
SAFETY COMPLIANCES
The GXW-410x is compliant with various safety standards including FCC/CE. Its power adaptor is
compliant with UL standard. Warning: use only the power adapter included in the GXW-410x package.
Using an alternative power adapter may permanently damage the unit.
WARRANTY
Grandstream has a reseller agreement with our reseller customer. End users 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. Grandstream reserves the right to
remedy warranty policy without prior notification.
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CONNECTING THE GXW-410X
FIGURE 1: DIAGRAM OF GXW-410X BACK PANEL
GXW-410x
Power Supply
On/Off Switch
FXO Ports
LAN/WAN RJ-45
Ethernet Ports
VIDEO IN Jack
TABLE 1: DEFINITIONS OF THE GXW CONNECTORS
LAN (or PC)
Connect your PC to this port. It will then be assigned an IP address from
your Router/DHCP Server. The GXW-410x acts as a switch only.
WAN (or LAN) Connect to the internal LAN network or Public Internet.
VIDEO IN
RESET
Connection for Analog based Video Surveillance Camera (RCA)
Factory Reset button. Press for 7 seconds to reset factory default settings.
Power adapter connection
POWER IN
OFF/ON
Off/On switch
FXO1 - FXO8
FXO ports to be connected to physical PSTN lines from a traditional PSTN
PBX or PSTN Central Office.
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FIGURE 2: DIAGRAM OF GXW-410X DISPLAY PANEL
GXW- 410x
Display
FXO port
TABLE 2: DEFINITIONS OF THE GXW DISPLAY PANEL
Power LED
Indicates Power. Remains ON when Power is connected and unit is
turned ON.
Ready LED
LAN LED
PC LED
Remains ON after boot-up.
Indicates LAN (or WAN) port activity
Indicates PC (or LAN) port activity
Video LED
Remains solid green on boot-up. If Video IN terminal is connected,
indicates video activity.
LEDs 1 - 8
Indicate status of the respective FXO Ports on the back panel
Busy - ON
Available - OFF
NOTE: All LEDs display green when ON.
During a firmware upgrade or configuration download the following LED pattern will be observed:
Power, Ready, Video and WAN LEDs will be ON. The FXO port LED will keep flashing during download
and then stay OFF while the new files are written. The entire process may take between 5 to 15 minutes.
The firmware upgrade is complete when you can login into the web configuration pages.
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APPLICATION DESCRIPTION
A. IP PBX / SIP Server with GXW410x
A SIP proxy server such as Asterisk or a SIP registrar server can be deployed with the GXW-410x series.
In this environment, the SIP server handles SIP registration and call control and the GXW-410x
processes media conversion between IP and PSTN calls.
There are 2 ways to configure GXW410x when using with a SIP Server:-
1. With SIP accounts configured on Channels page. In this case, the GXW acts like an endpoint
requesting registration from the SIP Server. Under the Channels webpage you will need to fill in
the information like SIP User ID, Password, etc. Now, when you try to make calls from IP, the call
will be routed to the SIP Server which will forward it to one of the SIP accounts on the GXW410x,
which will then forward it to the PSTN line.
2. Without SIP accounts. In this case, you simply have to configure the SIP Server to perform
forwarding of the SIP INVITE message with the FXO destination number to the gateways IP
Address. The GXW410x will receive the digits and immediately forward them on the FXO lines to
the destination PSTN. Most of the configuration on the Gateway for this case will remain default,
except Stage Method needs to be set to 1, and SIP Server IP Address/DNS name has to be filled.
FUNCTIONAL DIAGRAM OF IP-PBX & GXW-410X
Anywhere in the world
IPPBX or
SIP Server
FXO Lines
4 or 8 Ports
GXW-410x
PSTN
Cloud
IP/LAN
IP/WAN
Grandstream IP Phones
PSTN Analog endpoints
For incoming calls from the PSTN analog endpoints to the GXW410x, the device will auto forward each
call to a configured IP extension. The SIP Server can then route the call based on its own configuration or
IVR system.
B. FXS Gateway with GXW410x [No SIP Server required]
Alternatively, the GXW410x can be used without a SIP Server. You can use it in conjunction with a FXS
Gateway (Ex. GXW400x) and still be able to originate and terminate calls from IP to PSTN and vice
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versa. All you need to make sure is that the 2 gateways are able to locate each other (they should be on
the same LAN or on Public IPs).
GXW–400X & GXW–410X SCENARIO/TOLL- FREE CALLING BETWEEN LOCATIONS
Branch A - Boston, MA
4 employees
Analog
Phones
FX0
Internet
PSTN Cloud
Cloud
GXW-400x
GXW-410x
Branch B – Denver, CO
4 employees
In this diagram, configure the SIP Server field to be the IP Address of the other gateway (i.e. configure IP
address of FXS gateway to be SIP Server of GXW410x and vice versa). Please be sure you set SIP
Registration to No.
EXPECTED CALL FLOW: Analog Phone (GXW400x) picks up and dials destination PSTN number. The call
gets routed to the GXW410x which dials out the digit string onto the FXO Lines, thus reaching the
destination PSTN endpoint. On the reverse, incoming calls from PSTN endpoints will be routed
automatically to the FXS Gateway through the GXW410x.
FXS AND FXO GATEWAY CONFIGURATION EXAMPLE
GXW-400x
GXW-410x
Profile 1
SIP Server - Set it to IP Address of
GXW410x
Advanced Settings
STUN Server - Blank
Use Random Port - No
SIP Registration - No
Outgoing Call without Registration - Yes
NAT traversal – No
Advanced Settings
FXO lines
STUN Server - Blank
Wait for Dial Tone - Y or N (whichever works for your
PSTN Service Provider)
Stage Method - 1
Offhook Auto Dial - 444 @ch1-8:p1; ch1-8:5060++;
Channels
1-8 5060 Profile 1
Local SIP Listen port (For VOIP to PSTN calls) - 5060++
Profile 1
SIP Server - Set it to IP Address of GXW400x
SIP Registration - No
NAT traversal - No
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For more information regarding this setup, email Grandstream technical support at
FEATURES
GXW–410x is a next generation IP voice and video gateway that features full interoperability with leading
IP-PBXs, SoftSwitches and SIP platforms. The Gateway series offers superb voice and video quality,
traditional telephony functionality, simple configuration, feature rich functionality and an additional video
port that enables the gateway to act like a video surveillance gateway.
SOFTWARE FEATURES OVERVIEW
•
•
•
•
•
•
•
•
•
•
•
•
4 and 8 FXO port media gateways
Video surveillance port
External power supply
Two RJ-45 ports (switched or routed)
TFTP and HTTP firmware upgrade support
Multiple SIP accounts, multiple SIP profiles (choice of 3 profiles per account)
Supports Audio Codecs: G711U/A, G723, G729A/B and GSM
Supports Video Codecs: H.264
G.168 – echo cancellation
Flexible DTMF transmission: In Audio, RFC2833, SIP Info or any combination of the 3
Selectable, multiple LBR coders per channel
T.38 compliant
HARDWARE SPECIFICATION
TABLE 4: HARDWARE SPECIFICATION OF GXW-410X
LAN interface
LED
2xRJ45 10/100Mbps
8 LEDs (GREEN)
Universal Switching
Power Adaptor
Input: 100-240V AC, 50/60Hz, 0.5A Max
Output: 12V DC, 1.25A
UL certified
Dimension
Weight
225mm (L) x 172mm (W) x 42mm (H)
0.29 lbs (3.5 oz)
Temperature
32~104°F
0~40°C
Humidity
10% - 90% (non-condensing)
FCC, CE
Compliance
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TABLE 3: GXW-410X SOFTWARE FEATURES
GXW– 410x FX0 Analog Gateway Series
IP settings
GXW-4104: 4 ports; 4 SIP accounts w/ choice of 3 SIP Server profiles
GXW-4108: 8 ports; 8 SIP accounts w/ choice of 3 SIP Server profiles
Round-robin port scheduling to ensure available lines to access PSTN
networks
Telephone Interface
Network Interface
LED Indicators
FXO, RJ11
Two (2) 10/100 Mbps, RJ45
Power, Video, and Line LEDs
Yes
On/Off Switch
Voice over Packet
Capabilities
G.168 compliant Echo Cancellation, Dynamic Jitter Buffer,
Modem detection & auto-switch to G.711
Voice Compression
Video Surveillance
DHCP Server/Client
Fax over IP
G.711U, G711A, G.723, G.729A/B, GSM
Real-time H.264 base CIF resolution
Switch Mode and PPPoE
T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711
for Fax Pass-through
QoS
Diffserve, TOS, 802.1 P/Q VLAN tagging
RTP/RTCP and RTSP
IP Transport
PSTN Signaling
DTMF Method
FXO Loop start, Current Disconnect.
Flexible DTMF transmission method,
User interface of In-audio, RFC2833, and SIP Info
IP Signaling
Provisioning
Media
SIP (RFC 3261)
TFTP and HTTP
SRTP
Control
TLS and SIPS (pending)
Management
Syslog support,
HTTPS and telnet (pending), remote management using Web browser
Short and long haul
Caller ID
REN3: Up to150 ft on 24 AWG line
Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID
Polarity Reversal /
Wink
Yes (Detection only). The PSTN lines will need to be subscribed to PR
service from the Service Provider.
EMC
GXW-410x: EN55022 Class B, CFR Part 15 Class B, EN55024;
GXW-4104: FCC, CE (in addition)
Safety
GXW-410x: EN60950-1 GXW-4108: UL60950-1 (in addition)
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CONFIGURATION GUIDE
CONFIGURATION WITH WEB BROWSER
The GXW-410x has an embedded Web server that will respond to HTTP GET/POST requests. It also has
embedded HTML pages that allow a user to configure the IP phone through any common web browser.
Examples of GUI interfaces can be downloaded @
ACCESSING THE WEB CONFIGURATION MENU
1. Connect the Power to the GXW-410x unit.
2. Connect an Ethernet cable between the WAN port on GXW-410x to your PC.
3. You will have to assign a dummy IP with the same subnet as the GXW IP Address, which is
192.168.0.160 by default. So, set an IP address like 192.168.0.x for your PC.
to the GXW-410x web server.
You may choose to use DHCP or PPPoE connection or another static IP address according to your local
network environment.
The Gateway Web Configuration Menu can be then accessed by the following URI: http://Gateway-IP-
NOTE: To access the configuration page, type the GXW IP address into the browser, stripping out the
leading “0” because the browser will parse in octet. e.g. if the IP address is: 192.168.001.014, please type
in: 192.168.1.14.
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END USER CONFIGURATION
Once this HTTP request is entered and sent from a Web browser, the GXW-410x will respond with a login
screen. There are two default passwords for the login page:
User Level:
Password: Webpages allowed:
End User Level
Administrator Level
123
Only Status and Basic Settings
All pages can be browsed.
admin
FIGURE 4: SCREEN-SHOT OF GXW-410X LOG-IN SCREEN
Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2005-2006
After login, the next configuration page is the Basic Configuration page, explained in detail in Table 6:
Web Log-in Definition.
TABLE 6: WEB LOG-IN DEFINITIONS (BASIC SETTINGS PAGE)
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 End User Web Configuration Menu
(Status and Basic Settings). This field is case sensitive with a maximum length
of 25 characters.
IP Address
There are two modes to operate the GXW-410x:
DHCP mode: all the field values for the Static IP mode are not used (even
though they are still saved in the Flash memory.) The GXW-410x acquires its IP
address from the first DHCP server it discovers from the LAN it is connected.
Using the PPPoE feature: set the PPPoE account settings. The GXW-410x will
establish a PPPoE session if any of the PPPoE fields is set.
Static IP mode: configure the IP address, Subnet Mask, Default Router IP
address, DNS Server 1 (primary), DNS Server 2 (secondary) fields.
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DHCP hostname
DHCP domain
This option specifies the name of the client. This field is optional but may be
required by some Internet Service Providers. Default is blank.
This option specifies the domain name that client should use when resolving
hostnames via the Domain Name System. Default is blank.
DHCP vendor class
ID
Used by clients and servers to exchange vendor-specific information. Default is
Grandstream GXW-410x.
PPPoE account ID
PPPoE username. Necessary if ISP requires you to use a PPPoE (Point to
Point Protocol over Ethernet) connection.
PPPoE password
PPPoE account password.
PPPoE Service
Name
This field is optional. If your ISP uses a service name for the PPPoE connection,
enter the service name here. Default is blank.
Time Zone
Controls how the date/time is displayed according to the specified time zone.
Daylight Savings
Time
This parameter controls whether the displayed time will be daylight savings time
or not. If set to “Yes” and the Optional Rule is empty, then the displayed time will
be 1 hour ahead of normal time. The “Automatic Daylight Saving Time Rule”
shall have the following syntax:
start-time;end-time;saving
Both start-time and end-time have the same syntax:
Month ,day ,weekday ,hour ,minute
month: 1,2,3,..,12 (for Jan, Feb, .., Dec)
day: [+|-]1,2,3,..,31
weekday: 1, 2, 3, .., 7 (for Mon, Tue, .., Sun), or 0 which means the daylight
saving rule is not based on week days but based on the day of the month.
hour: hour (0-23),
minute: minute (0-59)
If “weekday” is 0, it means the date to start or end daylight saving is at exactly
the given date. In that case, the “day” value must not be negative. If “weekday”
is not zero and “day” is positive, then the daylight saving starts on the first
“day”th iteration of the weekday (1st Sunday, 3rd Tuesday etc). If “weekday” us
not zero and “day” is negative, then the daylight saving starts on the last “day”th
iteration of the weekday (last Sunday, 3rd last Tuesday etc). The saving is in the
unit of minutes. The saving time may also be preceded by a negative (-) sign if
subtraction is desired instead of addition. The default value for “Automatic
Daylight Saving Time Rule” shall be set to
“03,11,0,02,00;11,04,0,02,00;60” which is the rule for US.
Examples
US where daylight saving time is applicable:
03,11,0,02,00;11,4,0,02,00;60
This means the daylight saving time starts from 11th March at 2AM and ends
November 4th at 2AM. The saving is 60 minutes (1hour).
You may also access the Device Status page which provides details of the GXW product. The Device
Status page terms are defined in Table 7: Status Page Definitions.
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TABLE 7: STATUS PAGE DEFINITIONS
Hardware Revision
MAC Address
Hardware version number: Main Board, Interface Board
The device ID in HEX format.
troubleshooting.
This is a very important ID for ISP
IP Address
This field shows WAN IP address of GXW-410x
Product Model
Software Version
This field contains the product model info (GXW4104 or GXW4108)
Program: This is the main software release. Boot and Loader are not changed
often.
System Up Time
Registered
This field shows system up time since the last reboot.
This field indicates whether the different SIP Accounts configured under
Channels page are successfully registered to the SIP server(s).
FXO Line Connected
This field will give the status of each physical FXO Line connected to the
Gateway. It will update the status regularly.
Yes - Connected and Idle
Busy - Connected and Busy
No - Not connected
Additionally it will also provide real time Caller ID information of Incoming as
well as Outgoing calls.
PPPoE Link Up
This field shows whether the PPPoE connection is running if connected to DSL
modem.
Detected NAT Type
This field shows what kind NAT the GXW-410x is connected to via its WAN
port. It is based on STUN protocol.
ADVANCED USER SETTINGS
ADVANCED USER CONFIGURATION
The end-user needs to login to the advanced user configuration page the same way as for the basic
configuration page.
FIGURE 5: SCREENSHOT OF ADVANCED USER CONFIGURATION
Grandstream Device Configuration
Password
Login
All Rights Reserved Grandstream Networks, Inc. 2005-2006
Advanced User configuration includes the end user configuration and advanced configurations including:
SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration.
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TABLE 8: ADVANCED CONFIGURATION PAGE DEFINITIONS
Admin
Administrator password. Only the administrator can configure the “Advanced Settings”
Password
page. Password field is purposely left blank for security reasons. The maximum
password length is 25 characters.
G723 Rate
G723 encoding rate (6.3kbps or 5.3kbps)
Voice Frames This field contains the number of voice frames to be transmitted in a single packet.
per Tx
When setting this value, the user should be aware of the requested packet time (used
in SDP message) as a result of configuring this parameter. This parameter is
associated with the first vocoder in the above vocoder Preference List or the actual
used payload type negotiated between the 2 conversation parties at run time.
e.g., if the first vocoder 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 vocoder 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
BudgeTone 200 will use and save the maximum allowed value for the corresponding
first vocoder 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.
Layer 3 QoS
Layer 2 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. Its range lies from 0 to 63.
This contains the value used for layer 2 VLAN tag.
802.1q / VLAN tag : Default value is 0. Range lies from 0 to 4095.
802.1p Priority value : Default value is 0. Range lies from 0 to 7.
*** The above 2 settings need to be supported on the network and then configured
accordingly on the GXW410x. Incorrect configuration will cause blocked access, which
will result in Factory Reset as the only option to renew access.
Local RTP port This parameter defines the local RTP-RTCP port pair the GXW-410x 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 and so on. 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 GXW-410xs are behind the same
NAT.
Keep-alive
interval
This parameter specifies how often the GXW-410x sends a blank UDP packet to the
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 (Simple Traversal of UDP through NATs)
server.
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Firmware
Upgrade &
Provisioning
This radio button will enable GXW-410x to download firmware or configuration file
through either TFTP or HTTP.
Via TFTP
Server
If selected, the GXW410x will attempt to retrieve new configuration file or new code
image from the specified TFTP server at boot time. It will make up to 5 attempts before
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: Please do NOT interrupt the TFTP upgrade process (especially the power supply)
as this will damage the device. Depending on the network environment this process
can take up to 15 or 20 minutes.
Via HTTP
Server
The URL for the HTTP server used for firmware upgrade and configuration via HTTP.
For example, ttp://provisioning.mycompany.com:6688/Grandstream/1.0.0.54
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, GXW-410x will only do HTTP download once at boot
up.
Firmware
IP address or domain name of firmware server.
Server Path
Config Server IP address or domain name of configuration server.
Path
Firmware File
Prefix
Default is blank. If configured, GXW400X will request firmware file with the prefix. This
setting is useful for ITSPs. End user should keep it blank.
Firmware File
Postfix
Default is blank. End user should keep it blank.
Config File
Prefix
Default is blank. End user should keep it blank.
Default is blank. End user should keep it blank.
Config File
Postfix
Automatic
Upgrade
Choose Yes to enable automatic upgrade and provisioning. In “Check for new firmware
every” field, enter the number of minutes to enable GXW-410x to check the server for
firmware upgrade or configuration. When set to No, GXW-410x will only do upgrade
once at boot up. Other options are:-
“ Always check for New Firmware.”
“ Check New Firmware only when F/W pre/suffix changes”
“ Always skip the Firmware check”
Authenticate
Conf File
If set to Yes, config file is authenticated before acceptance. This protects the
configuration from an unauthorized change.
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 DEBUG,
INFO, WARNING or ERROR. Syslog messages are sent based on the following
events:
1. product model/version on boot up (INFO level)
2. NAT related info (INFO level)
3. sent or received SIP message (DEBUG level)
4. SIP message summary (INFO level)
5. inbound and outbound calls (INFO level)
6. registration status change (INFO level)
7. negotiated codec (INFO level)
8. Ethernet link up (INFO level)
9. SLIC chip exception (WARNING and ERROR levels)
10. 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
URI or IP address of the NTP (Network Time Protocol) server, which will be used by the
phone to synchronize the date and time.
Enable Video
Surveillance
When set to Yes, GXW410x will start converting video feed received from analog
camera to IP packets. In order to view this video feed, please follow instructions given
under Video Surveillance chapter on page 23.
RTSP Port
By default it is 554.
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CONFIGURING THE FXO CHANNELS
Configuring the FXO channels on the GXW – 410x is an easy process. Follow the GUI interfaces. The
Device Status page terms are defined in Table 9: FXO Lines Configuration Definitions. An example
of the Channel Dialing Configuration is shown in Figure 6. Please note the default is always configured.
The user has the option to change the default settings as described in the Table 9.
TABLE 9: FXO LINES CONFIGURATION DEFINITIONS
Enable Current
Disconnect
When set to Y, Current Disconnect is enabled. Certain PSTN Cos require this to be
enabled, in order to realize disconnect signal from PSTN side. Default is Y.
If enabled use threshold : Default is 100ms. Range is 50ms to 800ms.
Certain PSTN service providers have a threshold time within which the line stabilizes
after off-hook. It is entirely dependent on provider, however if you experience PSTN
line detection issues, please modify this setting appropriately in 100ms increments.
Enable Tone
Disconnect
Default is No. If PSTN provider uses call progress tones then it should be set to Yes in
order to realize disconnect tone. Please configure accurate Call Progress Tones on
Channels webpage based on PSTN provider (or traditional PBX) settings.
Enable Polarity
Reversal
Default is No. This should be set to Yes only if the FXO lines are subscribed to PR
service from PSTN Service provider. It is merely a PR detect feature.
***Note:- If there is no PR service from provider on the FXO line, and this setting is
configured to Yes, calls will not be successful.
AC Termination
Impedance
Selects the impedance of the analog Line connected to the FXO port on the GXW-
410x.
Silence Timeout Terminate call after long silence detected. Default is 60 seconds, max 65536.
Syntax for Channel Specific Settings:
Default - ch1-8:X; {all channels 1 to 8 are set to value X}
Additional Examples:-
ch1,3-6:10;ch2,7-8:12 {channels 1,3,4,5 and 6 are set to value 10 while channels 2,7
and 8 are set to value 12.
DTMF Digit
Length
Default value is 100ms.
Default value is -11dB.
DTMF Digit
Volume
DTMF Dial Pause Default value is 100ms.
Wait for Dial-tone Default is Y (Yes). When set to Yes, the gateway will wait till it hears dialtone from
FXO line before dialing out the destination digits.
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Stage Method
Syntax - ch1-8:1; {all channels 1 to 8 are set to value 1 or 2}
Stage method can be set to either 1 or 2.
In simple words, if you need to make a direct PSTN call from a VOIP endpoint you
should set this to 1. When you set it to 2, you will first dial one of the VOIP channel
accounts from the VOIP endpoint, this will result in getting a PSTN line dial-tone to
then dial out the destination PSTN number.
Most implementations require this setting to be configured to 1.
Min. Delay before Defaut is 100ms.
Dial PSTN
This needs to be equal or slightly higher than the Current Disconnect threshold setting,
so that once the FXO line is stable after the threshold period, the gateway can dial out.
Again, it should be touched only if there are PSTN line detection issues.
Offhook Auto
Dial
This is an extremely important setting to make sure incoming PSTN calls are picked
up and forwarded to the correct VOIP destination.
User ID - This parameter allows users to configure a User ID or extension number to
be automatically dialed upon FXO line off-hook.
SIP Server - You also need to specify the Profile of the user id configured above (p1
stands for Profile 1, p2 stands for Profile 2 and so on).
SIP Destination Port - Along with the user-id and Profile, you also have the option to
choose the destination port where you would like to send the call. By default it should
be set to ch1-x:5060; (x can be 4 or 8 depending on number of ports).
Caller ID Scheme The GXW410x supports 5 different types of schemes:-
1. Bellcore (US standard)
2. ETSI_RING
3. ETSI_TAS
4. DTMF
5. NTT (Japanese standard)
Please check with your PSTN service provider (or traditional PBX specs) for which
caller ID scheme it supports.
Caller ID
Transport type
Default is “relay via From header”. You may also select :-
“relay via P_Asserted_Identity header”
“Disable” :- Caller ID feature will be disabled.
“Send anonymous” :- All calls forwarded to VOIP end will be sent as anonymous.
T.38 Setting
This setting allows you to make several options related to Facsimile.
You can select the method: T.38 or Pass through (G711)
You can select the fax transmission rates (2400/4800/7200/9600/12000/14400bps)
You can enable or disable ECM (Error Checking Mode)
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TABLE 10: CHANNELS PAGE DEFINITIONS
Note – The channels here are basically SIP endpoints that will act as
clients registering to the SIP Server configured under the appropriate
Profile page.
Channels
It should be set same as the channel number (i.e 1, 2..4 or 8 depending on
number of FXO ports).
It is NOT the same as SIP Account ID.
SIP User ID
This is the SIP account information. Enter the SIP User ID part of the
account.
Authentication ID
SIP service subscriber’s Authenticate ID used for authentication. It can be
identical to, or different from SIP User ID.
Authentication Password SIP account password needs to be entered here.
Note:- After entering the password, it will show up as blank but the
password still remains active.
Profile ID
Select the corresponding Profile ID (1/2/3). Profiles are SIP Server
configurations.
Call Progress Tones
Using these settings, user can configure tone frequencies according to user
preference. By default, the tones are set to North American frequencies.
Frequencies should be configured with known values to avoid
uncomfortable high pitch sounds. ON is the period of ringing (ON time in
ms) while OFF is the period of silence. In order to set a continuous ring,
OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms
and then repeat the pattern.
•
•
•
•
“Dial tone”
“Ringback tone”
“Busy/Re-order tone”
“Confirmation tone”
Please refer the document below to determine your local call progress
tones:-
Channel Voice Settings
Channel voice settings mentioned below.
Tx to PSTN Audio Gain
(dB)
Allows user to set a value in dB for transmission to PSTN Audio Gain.
Default is 1. Range is from -12 to 12dB.
Rx from PSTN Audio
Gain (dB)
Allows user to set a value in dB for receive from PSTN Audio Gain. Default
is 0. Range is from -12 to 12dB.
Silence Suppression
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.
Echo Cancellation
When set to Y, Echo cancellation is enabled.
Channel specific Setting Channel specific settings mentioned below.
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DTMF Method
This parameter specifies the mechanism to transmit DTMF digits. There
are7 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. Multiple DTMF transmission schemas can be selected.
1 – in-audio
2 – RFC2833
3 – in-audio and RFC2833
4 – SIP Info
5 – in-audio and RFC2833
6 – SIP Info and RFC2833
7 – in-audio, RFC2833, and SIP Info
No Key Entry Timeout
Local SIP Listen Port
Default is 4 seconds.
Default is ch1-8:5060++;. Here the ++ indicates increments by 2, so port 1
is set at 5060, port at 5062 and so on. This setting can be used with Round
Robin and/or Flexible setting below to configure different ports to be placed
under different Round Robin groups.
Round Robin and/or
Flexible
Default is rr:1-8;
The syntax is pretty straight-forward here. The rr stands for Round Robin
and the numbers stand for the ports that belong to that round robin group.
For example:-
rr:1-8; -> Round robin within the first 4 ports i.e. outgoing calls will be
forwarded to the next available port within the group of ports 1 to 4.
rr:1,3-6,8;rr:2,7; -> Round robin within port 1,3,4,5,6 and 8; Second round
robin group within ports 2 and 7 i.e. outgoing calls to ports 1,3,4,5,6 and 8
will be forwarded to the next available port within this group ONLY.
Outgoing calls to port 2 and 7 will be forwarded to the next available port
between ports 2 and 7 ONLY.
** In order to terminate a call on FXO port 2 or 7 you will need to change its
Local SIP Listen port accordingly.
Prefix to specify Port
Default is 99.
(1 stage dialing method) Syntax to USE this feature: prefix# (thats 99) + ch# (could be anything from
1 to 8) + dialing# will result in this call forwarded to FXO port (ch#)
immediately.
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DIAL PLAN
The Dial Plan feature implemented is applicable for VOIP to PSTN calls only. You may configure a dial
plan based on the following grammar:-
1. Accept Digits: 1,2,3,4,5,6,7,8,9,0,*,#
2. Grammar:
x - any digit from 0-9;
xx+ - at least 2 digit number;
^ - exclude;
[3-5] - any digit of 3, 4, or 5;
[147] - any digit 1, 4, or 7;
<2=011> - replace digit 2 with 011 when dialing
WARNING - illegal input will fall back to default
Example 1: {[369]11 | 1617xxxxxxx} - Allow 311, 611, 911, and any 10 digit numbers of leading
digits 1617.
Example 2: {^1900x+ | <=1617>xxxxxxx} - Block any number of leading digits 1900 and add
prefix 1617 for any dialed 7 digit numbers.
Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} - Allow any length of number with leading digit 2 and
10 digit-numbers of leading digit 1 and leading exchange number between 2 and 9;
If leading digit is 2, replace leading digit 2 with 011 before dialing
Default: PSTN Outgoing - {x+}
Note – If you do not plan to use this feature just let it be default {x+}
PROFILES
Profiles are basically IP PBX / SIP Server configuration templates. If you have more than one IP PBX
system or SIP Server that you would like to use with the GXW410x, then you can configure Profile 2 or 3.
Note – Make sure you select the correct profile for each channel under Channels WEBPAGE.
TABLE 11: PROFILE PAGE DEFINITIONS
Activate Profile
Profile Name
SIP Server
When set to Yes the SIP Profile is activated.
A name to identify a Profile.
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 GXW-410x for firewall or NAT penetration in different
network environments. If symmetric NAT is detected, STUN will not work and
ONLY outbound proxy can correct the problem.
Use DNS SRV:
Default is No. If set to Yes the client will use DNS SRV to look up server.
User ID is Phone If the GXW-410x has an assigned PSTN telephone number, this field should be set
Number
to “Yes”. Otherwise, set it to “No”. If “Yes” is set, a “user=phone” parameter will be
attached to the “From” header in SIP request.
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SIP Registration This parameter controls whether the GXW-410x needs to send REGISTER
messages to the SIP 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 the user to specify the time frequency (in minutes) for the
GXW-410x to refresh 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
NAT Traversal
This parameter defines the local SIP port the GXW-410x will listen and transmit.
The default value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2,
Account 3 and Account 4 respectively.
This parameter defines whether the GXW-410x NAT traversal mechanism will be
activated or not. If activated (by choosing “Yes”) and a STUN server is also
specified, then the GXW-410x will behave according to the STUN client
specification. Under this mode, the embedded STUN client inside the GXW-410x
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 GXW-410x 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 GXW-410x 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.
Proxy-Require
Early Dial
SIP Extension to notify SIP server that the unit is behind the NAT/Firewall.
Default is No. Use only if proxy supports 484 response.
Session
Expiration
Grandstream implemented SIP Session Timer. The session timer extension
enables 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 seconds.
Caller Request
Timer
If selecting “Yes” the phone will use session timer when it makes outbound calls if
remote party supports session timer.
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 Request Timer, and Force Timer.
UAC Specify
Refresher
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
Refresher
As a Callee, select UAC to use caller or proxy server as the refresher, or UAS to
use the phone as the refresher.
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Force INVITE
Enable 100rel
Session Timer can be refreshed using INVITE method or UPDATE method. Select
“Yes” to use INVITE method to refresh the session timer.
The use of the PRACK (Provisional Acknowledgment) method enables reliability to
be offered to SIP provisional responses (1xx series). This is very important if PSTN
inter-networking is to be supported. A user’s request to use reliable provisional
responses is invoked by the 100rel tag which is appended to the value of the
required header of initial signalling messages.
Refer-to uses
Target Contact
Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header uses
the transferred target’s Contact header information.
Preferred
Vocoder
The GXW-410x supports up to 5 different Vocoder types including G.711 A-/U-law,
GSM, G.723.1, G.729A/B. The 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”.
Special Feature
Default is Standard. Choose the selection to meet some special requirements from
Soft Switch vendors like Nortel, Broadsoft, etc.
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SAVING THE CONFIGURATION CHANGES
Once a change is made, press the “Update” button in the Configuration Menu. The GXW-410x will display
the following screen to confirm that the changes have been saved. To activate changes, reboot or power
cycle the GXW-410x after all changes are made.
FIGURE 5: SCREEN-SHOT OF SAVE CONFIGURATION
REBOOTING FROM REMOTE
The administrator can remotely reboot the unit by pressing the “Reboot” button at the bottom of the
configuration menu. The following screen will indicate that rebooting is underway.
FIGURE 6: SCREEN-SHOT OF REBOOTING
Grandstream Device Configuration
The device is rebooting now...
You may re-login by clicking on the link below in 30 seconds.
All Rights Reserved Grandstream Networks, Inc. 2005
The user can re-login to the unit after waiting for about 30 seconds.
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VIDEO SURVEILLANCE
The GXW-410x can be used with an Analog Surveillance CCD Camera to perform video surveillance
function. This application should be used in a LAN environment or when both sides have public IP
address.
VIDEO SURVEILLANCE PROCEDURES
¾
Gateway side:
1. In the ADVANCED SETTING page, find the following field and change from default setting
NO to YES, reboot the device.
2. Connect an analog based surveillance camera to the VIDEOIN connection at the back panel
of the unit.
¾
PC side (Monitor Device):
RFC 3984.
2. Launch VLC.
3. Go to Preferences->Input/Codecs->Demuxers->H264, check “Advanced options” in the
bottom. 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. (Grandstream does NOT recommend this network environment)
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.
Change the blue text 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 to start the video.
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FIRMWARE UPGRADE
Firmware (or software) upgrades can be done either via TFTP or HTTP. The corresponding configuration
settings are on the configuration page. End users should NOT touch the configuration settings that are
useful for ITSPs. To upgrade your unit firmware, follow these steps:
1. Under Advanced Settings webpage, enter your TFTP or HTTP Server IP address (or FQDN) next
to the “Firmware Upgrade: Upgrade Server” field.
2. Select via TFTP or HTTP accordingly.
3. If you plan to use Automatic Upgrade, set it to “Yes”, otherwise No (this will make it check for
upgrade every time you reboot).
UPGRADE THROUGH HTTP
To upgrade firmware via HTTP, the field “Firmware Upgrade and Provisioning: Upgrade Via” needs to be
set to HTTP. The “Firmware Server Path” should be set to where the firmware files are located.
For example, the user can use the following URL in the Firmware Server Path:
firmware.mycompany.com: 6688/Grandstream/1.0.0.53 where firmware.mycompany.com is the FQDN
of the HTTP server. It can also be in IP address format. “:6688” is the TCP port the HTTP server listening
to, default http server listens to port 80. “/Grandstream/1.0.0.53” is the RELATIVE directory to the root dir
on HTTP web server.
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UPGRADE THROUGH TFTP
To upgrade firmware via TFTP, set the field “Firmware Upgrade and Provisioning: Upgrade Via” to TFTP.
The TFTP server can be configured in either IP address format or FQDN.
To configure the TFTP server via the Web configuration interface, follow these five steps:
1. Open your browser to input the IP address of the GXW-410x.
2. Enter the admin password to enter the configuration screen.
3. Enter the TFTP server address or URL in the “Firmware Server Path” field near the
bottom of the configuration screen.
4. Once the “Firmware Server Path” is set, update the change by clicking the “Update”
button.
5. Reboot or power cycle the unit.
If the configured updating server is found and a new code image is available, the GXW-410x will retrieve
the new image files by downloading them into the GXW-410x’s SRAM. During this stage, the GXW-
410x’s LED will blink until the checking/downloading process is completed. Upon verification of
checksum, the new code image will be saved into the Flash. If TFTP fails for any reason (e.g., TFTP
server is not responding, there are no code image files available for upgrade, or checksum test fails, etc),
the GXW-410x will stop the TFTP process and simply boot using the existing code image in the flash.
Firmware upgrading may take as long as 20 minutes over the Internet, or just 20+ seconds if it is
performed on a LAN. Grandstream recommends conducting firmware upgrades in a controlled LAN
environment if possible.
DOWNLOAD TFTP SERVER
For users who do not have a 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. Alternatively, user
can download and install a free TFTP or HTTP server in his LAN for a firmware upgrade.
A free Windows version TFTP server can be downloaded from:
Directions For Downloading TFTP Server:
1. Unzip the file and put all of the files under the root directory of the TFTP server.
2. Put the PC running the TFTP server and the GXW–410x in the same LAN segment.
3. Go to File -> Configure -> Security to change the TFTP server's default setting from "Receive
Only" to "Transmit Only" for the firmware upgrade.
4. Start the TFTP server, in the phone’s web configuration page.
5. Configure the Firmware Server Path with the IP address of the PC.
6. Update the change and reboot the unit.
You can also download the free HTTP server from http://httpd.apache.org or just use Microsoft IIS web.
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RESTORE FACTORY DEFAULT SETTING
WARNING! Restoring 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. Grandstream will
not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP
service provider.
The ONLY way to restore default factory settings is as follows:
1. Unplug the Ethernet cable.
2. Locate the needle sized hole on the back panel of the gateway next to the Power connection.
3. Enter a thin object in this hole and keep it pressed for about 7 seconds.
4. You will see the Network port LEDs (green and orange) go off and on simultaneously; this
indicates the reset went through.
5. All settings have been erased and the gateway is back to factory settings.
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Examples of GXW-410x Configurations
APPLICATION ONE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER
Scenario: A business with a traditional phone system (with or without broadband access) and an IP
PBX or SIP Servers connecting to an Internet Telephone Service Provider (ITSP).
Anywhere in the world
IPPBX or
SIP Server
FXO Lines
4 or 8 Ports
PSTN
IP/LAN
IP/WAN
Cloud
GXW-410x
Grandstream IP Phones
PSTN Analog
APPLICATION TWO: GXW TO EXTEND A TRADITIONAL PBX SCENARIO
Scenario: a small business with traditional analog PBX lines and broadband access who want to extend
their traditional PBX to virtually anywhere in the world, using the internet. (Any SIP End point, such as
Grandstream BugeTone, HandyTone, GXP-2000 or GXV-3000 are needed in this scenario)
Company A - Boston, MA
6 employees
IPPBX or
SIP Server
Internet
Cloud
IP/LAN
IP/LAN
Optional
FXO Lines
GXW-410x
FXO Lines
Traditional
PBX
PSTN
Cloud
GXW-410x
FXS | IPPBX | SIP Platform
Grandstream IP Phones
Anywhere in the world
Any SIP endpoint
Any branch, anywhere
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APPLICATION THREE: GXW CONNECTED WITH AN IP-PBX OR SIP SERVER AND VIDEO SURVEILLANCE
Scenario: The GXW-410x offers an additional video surveillance port which can be configured for
surveillance. It is the only small business analog gateway that offers this security feature.
Branch A – Boston, MA
6 employees
IPPBX or
SIP Server
IPPBX or
SIP Server
Internet
Cloud
IP/LAN
IP/LAN
PSTN
Cloud
GXW 410x
GXW 410x
Grandstream IP
Grandstream IP
Branch B – Denver, CO
4 employees
Anywhere in the world
APPLICATION FOUR: USING A GXW FOR PURE IP- IP COMMUNICATION CONFIGURATION
Scenario Four: The GXW-410x offers an IP to IP pure IP Communications System configuration, where
all locations use IP phones.
Branch A - Boston, MA
6 employees
IPPBX or
SIP Server
IPPBX or
SIP Server
Internet
Cloud
IP/LAN
IP/LAN
PSTN
Cloud
GXW 410x
GXW 410x
Grandstream IP Phones
Grandstream IP Phones
Branch B – Denver, CO
4 employees
Anywhere in the world
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GLOSSARY OF TERMS
ADSL 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.
AGC 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 Address Resolution Protocol is a protocol used by the Internet Protocol (IP) [RFC826], specifically
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 network and OSI link layer. It
is used when IPv4 is used over Ethernet
ATA 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 sub-sampling.
DECT Digital Enhanced Cordless Telecommunications: A standard developed by the European
Telecommunication Standard Institute from 1988, governing pan-European digital mobile telephony.
DECT covers wireless PBXs, telepoint, residential cordless telephones, wireless access to the public
switched telephone network, Closed User Groups (CUGs), Local Area Networks, and wireless local loop.
The DECT Common Interface radio standard is a multi-carrier time division 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 information is transmitted from the RFP within a multi-frame (16 frames). Voice signals are
digitally encoded into a 32 Kbit/s signal using Adaptive Differential Pulse Code Modulation.
DNS Short for Domain Name System (or Service or Server), an Internet service that translates domain
names into IP addresses
DID Direct Inward Dialing. The ability for an outside caller to dial to a PBX extension without going
through an attendant or auto-attendant.
DSP 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 #).
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FQDN Fully Qualified Domain Name. A FQDN consists of a host and domain name, including top-level
Grandstream is the second-level domain, and and.com is the top level domain.
FXO 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.
•
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 analog phone).
•
An FXS device will allow any FXO device to operate as if it were connected to the phone
company. 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 addresses, 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 from a voice communication in order to improve voice quality on a telephone call. In addition to
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 A suite of standards for multimedia conferences on traditional packet-switched networks.
HTTP Hyper Text Transfer Protocol; the World Wide Web protocol that performs the request and retrieve
functions of a server
IP 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 circuit-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 transmission over the Internet or other packet-switched
networks; the process is reversed at the receiving 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 software 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.
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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 Network Address Translation
NTP 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
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 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.
PSTN Public Switched Telephone Network. The phone service we use for every ordinary phone call, or
called POT (Plain Old Telephone), or circuit switched network.
RTCP 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.
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 Session Description Protocol is a format for describing streaming media initialization parameters. It
SIP 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 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
TFTP Trivial File Transfer Protocol, is a very simple file transfer protocol, with the functionality of a very
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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.
VoIP Voice over the Internet. VoIP encompasses many protocols. All the protocols do some form of
signaling of call capabilities and transport of voice data from one point to another. e.g.: SIP, H.323, etc.
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