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NAT Management
How to configure QuesCom architecture through NAT?

QuesCom Product System Release

: QuesCom 300 / 400 / 420 : 4.20B410 and above

Document version 02.00, August 08, 6 pages

Summary
This document defined ports used by QuesCom devices. It also explains how to configure NAT.

1. 2. 3. 4. 5. 6.

Purpose of NAT service Different kinds of NAT What kind of NAT shall be used? NAT configuration on Master gateway NAT configuration for Relay gateways Summary of used ports

Please check the QuesCom documentation for additional information or contact the QuesCom Support hotline at: +33 (0) 820 20 38 46 or by email to support@quescom.com This document and information it contains are property of QuesCom. They shall not be reproduced nor disclosed to any person except to those having a need to know them without prior written consent of QuesCom.

Techniicall Documentatiion Techn ca Documentat on 1. Purpose of NAT service


When a Quescom gateway is behind a NAT router, it must send the public IP to operate with the remote Media gateway, in order to establish RTP flows. Quescom NAT feature allows you to configure this public IP on the QuesCom gateway. Also, while todays routers are able to dynamically open and close multiple ports as required by VOIP signaling protocols, such as SIP and H323, they remain ineffective at supporting unsolicited incoming media flows. Then, Quescom gateways use fixes UDP ports ranges for RTP/RTCP, allowing the network administrator to configure the front end router, to forward RTP traffics towards Natted gateways. In SIM Server architecture, it also allows the SIM Server to know the public IP of gateways in order to correctly assign SIM cards.

2.

Different kinds of NAT


QuesCom devices can be configured in 3 different ways regarding NAT: 1) No NAT configured 2) Static NAT 3) QStun

2.1

No NAT configured

By default, no NAT is configured on the QuesCom. It is supposed that the device is using a public IP address or that the Media gateway can directly access the QuesCom.

2.2

Static NAT

When you are using a static public IP for the QuesCom to the outer world, you can configure Static NAT on the device.

2.3

QStun (Dynamic)

STUN is a protocol which allows a device to know its own address. QStun can only be used within SIM Server architecture: QuesCom device will connect to the SIM Server asking for its IP address. Then, the SIM Server will return the incoming IP address (which is the public IP address).

3.

What kind of NAT shall be used?


Depending of your usage, you have to choose between on NAT mode (No NAT, Static, Dynamic).

NAT Management
How to configure QuesCom architecture through NAT?

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Techniicall Documentatiion Techn ca Documentat on 4. NAT configuration on Master gateway


First of all, NAT management has to be configured on the Master device for VoIP Signaling. Using NAT information, Master device will be able to correctly discuss with the Media gateway.

4.1

Configuration on an ITSP Series QuesCom using version 4.50 or above

For ITSP Series QuesCom, a simplified configuration wizard has been created. In order to configure NAT, connect to Telnet on the Master device and type command config /nat:
Do you want to configure NAT for this Gateway [2]: 0 No 1 Static Mode 2 Dynamic Mode (STUN) >

Then, just follow indicated steps in order to configure NAT using your needed mode.

Warning Once NAT is configured, the device will reboot in order to take changes in account.

4.2

Configuration on other devices

As the wizard is only available on ITSP Series, all other devices must be configured manually (Enterprise, HBS, MCS, ).
4.2.1 Case of a static NAT configuration

If the public IP address is a fix IP address, you can configure a static NAT. To configure it, connect to Telnet on the Master device and type following commands:
X:\> gwconfig [GWconfig_Gateway] root callserver [GWconfig_CallServer] add REG_SZ NATAddress=XXX.XXX.XXX.XXX [GWconfig_CallServer] add REG_SZ NetMask=YYY.YYY.YYY.YYY [GWconfig_CallServer] exit

NATAddress value (XXX.XXX.XXX.XXX) represents the public IP address. If DNS is configured on the device, it can be a DNS name. NetMask value (YYY.YYY.YYY.YYY) represents the network mask of the public IP. This value is needed in order to differentiate calls initiate from the LAN or from Internet (if QuesCom needs to send the public IP or not).

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4.2.2 Case of a dynamic NAT configuration (QStun)

If the public IP address is a dynamic IP address and if the QuesCom device is part of a SIM Server solution, you can configure Dynamic NAT (QStun). In this case, the device will connect to the SIM Server and ask for its public IP address.

To configure it, connect to Telnet on the Master device and type following commands:
X:\> gwconfig [GWconfig_Gateway] root callserver [GWconfig_CallServer] set StunMgr=1 [GWconfig_CallServer] root system [GWconfig_system] cd services\qstun\parameters [GWconfig_system\services\qstun\parameters] set QSTUNServerIP=XXX.XXX.XXX.XXX [GWconfig_system\services\qstun\parameters] exit

QSTUNServerIP value (XXX.XXX.XXX.XXX) represents the SIM Server IP address.

5.

NAT configuration for Relay gateways


VoIP calls to QuesCom Master/Relay group are made to the Master gateway. However, the voice flow (RTP) has to be able to transit to the Relay gateway. There is no problem for relay gateways to use a dynamic IP regarding the Master. When IP address of relay gateway change, this will be updated in Master database.

5.1

Managing RTP flow

By default, all gateways use UDP ports 11000 to 11999 for RTP (voice). This can be problematic when all gateways are reached using the same public IP (using NAT). In this case it is mandatory to reconfigure used ports for relay gateways using following commands in Telnet:
X:\> gwconfig [GWconfig_Gateway] cd SessMgr [GWconfig_Gateway\SessMgr] set BaseUDPPort=XXXXX [GWconfig_Gateway\SessMgr] exit

BaseUDPPort represent the lowest port that will be used for RTP flow for this gateway. The gateway can use up to 1000 ports for RTP flow (ie: 12000 to 12999). Usually, following UDP ports are used depending of the relay: Relay number Relay 1 Relay 2 Relay 3 Relay 4 BaseUDPPort 12000 13000 14000 15000

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5.2 NAT between Master and Relays

Relay gateways need to access TCP port 1163 to Master gateway in order to register.

In order to place calls on relays, Master gateway need to contact relays on TCP port 1163. If you want to use several relay gateways on a same site (using the same IP regarding the Master), please contact QuesCom support in order to configure it.

6.
6.1

Summary of used ports


Used ports on SIM Server

Following ports are used to access to the SIM Server. PORT 21 23 1170 1174 1179 3478 8000 9628 27051 PROTOCOL TCP TCP TCP TCP TCP UDP TCP TCP TCP USAGE FTP access (administration) Telnet access (administration) QWAlerter (administration) Automatic updates (administration) QWAdmin (administration) QStun QPortal (web administration) ODBC Gateways registration

6.2

Used ports on SIM Array

Following ports are used to access to the SIM Array. PORT 21 23 51503 PROTOCOL TCP TCP TCP USAGE FTP access (administration) Telnet access (administration) SIM Server and gateways connection

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6.3 Used ports on gateways

Following ports are used to access to gateways. PORT 21 23 80 1163 1168 1169 1170 1174 1200 to 1210 8000 9628 20000 to 20005 27031 27032 27051 PROTOCOL TCP TCP TCP TCP TCP TCP TCP TCP TCP TCP TCP TCP TCP TCP TCP USAGE FTP access (administration) Telnet access (administration) QPuser (user web interface) Master Relay connection QWAdmin (administration) QWAdmin (administration) QWAlerter (administration) Automatic updates (administration) CTI Applications QPortal (web administration) ODBC / HBS Reserved Georoute Geosoft console (GeovoxIVR / TopUp) Gateways registration

Following table show used ports for VoIP. For more information, please check previous paragraphs: PORT 1719 and 1720 5060 11000 to 11999 12000 to 12999 13000 to 13999 14000 to 14999 15000 to 15999 PROTOCOL TCP UDP / TCP UDP UDP UDP UDP UDP APPLICABLE ON Master Master Master Relay 1 Relay 2 Relay 3 Relay 4 USAGE H323 signaling SIP signaling RTP (voice flow) RTP (voice flow) RTP (voice flow) RTP (voice flow) RTP (voice flow)

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