Reference11r1:Gateway/GK/GW

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Below is an overview of all the gateway's configurable VoIP interfaces:

  • Interface: The name of the interface. Clicking this name opens a popup page, on which all settings can be made. The settings are described in more detail in the following chapter "Administration/Gateway/VOIP/Interface (VoIP interfaces)".
  • CGPN In, CDPN In, CGPN Out, CDPN Out: Precise details on CGPN In, CDPN In, CGPN Out and CDPN Out mappings are contained in the chapter entitled "Administration/Gateway/Interfaces/CGPN-CDPN Mappings" further up in the text.
  • Registration: If a terminal has successfully registered with a gateway, then this is indicated in this column through specification of the IP address <Name of the VoIP interface:Call number:IP address>.


Contents

Interface (VoIP Interfaces)

Clicking the relevant VoIP interface (GW1-12 <Name of the VoIP interface>) in the Interface column opens a popup page, on which the VoIP interfaces can be individually configured. Like the PBX objects, this popup page also contains standard entry fields that occur, more or less, in all VoIP interfaces.

These standard fields are:

  • Name: The descriptive name of the VoIP interface.
  • Disable: A checked check box disables the relevant VoIP interface.
  • Protocol: The protocol to be used, that is, H.323 or SIP. Depending on which protocol is used, the set-up of the entry fields changes.
  • Mode: Describes the mode of registration. Possible registration modes are:
  1. Gateway without Registration - Logs the VoIP interface (gateway) on to the configured gatekeeper without a registration.
  2. Register as Endpoint - Registers a VoIP terminal with the configured gatekeeper.
  3. Register as Gateway - Registers a VoIP gateway with the configured gatekeeper.
  4. Gatekeeper/Registrar - Is required for managing all gatekeeper registrations on a gateway.
  5. ENUM - Is used to register an ENUM connection with the relevant interface.
  • Gatekeeper Address (primary): The primary Gatekeeper IP address at which the terminal or gateway is to register via the relevant interface. Only necessary for modes 2 and 3.
  • Gatekeeper Address (secondary): The alternative gatekeeper IP address at which the terminal or gateway is to register via the relevant interface, if registration with the primary gatekeeper fails. Only necessary for modes 2 and 3 .
  • Mask: By specifying a network mask, incoming calls can be filtered. Specification of the network mask 255.255.0.0 therefore allows incoming calls on the relevant interface for terminals from the IP address range 192.168.0.0 - 192.168.255.255 .
  • Gatekeeper Identifier: It is also sufficient to specify only the gatekeeper ID. Every gatekeeper in a network can be identified by means of its own gatekeeper ID, so that several gatekeepers can be operated in a network, with each terminal nevertheless identifying the correct gatekeeper by means of Gatekeeper Discovery (uses the multicast address 224.0.1.41).
  • STUN SERVER: The STUN server name or IP address must be configured if this device has no public IP address while the SIP server is accessible under a public IP address. The value is given by the SIP provider or administrator (for example, stun.xten.com or 64.69.76.23). You can choose any STUN server; it does not necessarily have to correspond to the one of the SIP provider.
  • Local Signalling Port: Signalling Port used by the Interface.
  • SIP - Register as Gateway/Endpoint:
    • Remote Domain (SIP Only): SIP Domain of the remote SIP Server to register (often equal to the SIP Proxy).
    • Proxy (SIP Only): DNS name or IP address of the SIP proxy where SIP messages (REGISTER,INVITE,etc) are to be sent to. Proxy can be omitted if domain part of AOR can be used as remote signaling destination and it's set at Remote Domain.
  • SIP - Gateway without Registration:
    • Remote Domain (SIP Only): Check the Domain/IP Address on the SIP URI "FROM" field and removes it if match.
    • Local Domain (SIP Only): Check the Domain/IP Address on the SIP URI "TO" field and removes it if match.
    • Proxy (SIP Only): DNS name or IP address of the SIP proxy where SIP messages (INVITE,etc) are to be sent to, this field it's mandatory.
    • Filter Incoming Calls (SIP Only): Accept Incoming Call on this interface only if the domain part of the "To:" field SIP URI matches the Local Domain configured. If not accepted, the call may still be accepted by another interface

Authorization section

In the Authorization section, you can store a password for the VoIP interface.

  • Password / Retype: The security of the registration can be raised by specifying a password (Password). The password must be confirmed (Retype).

Alias List section

In the Alias List section, you specify the call name (H.323) and the call number (E.164) of the relevant registration. For VoIP end points, you should define the assigned direct dialling number or MSN as the E.164 address, and the name as the H.323 name. For VoIP gateways it is sufficient to define the name.

  • Name: The H.323 name.
  • Number: The E.164 call number.

Coder Preferences section

The standard entry fields in the Coder Preferences section were already described in chapter "Administration/Gateway/Interfaces/Interface (physical and virtual interfaces)".

Media Properties

The standard entry fields in the Media Properties section were already described in chapter "Gateway/Interfaces/Physical (physical and virtual interfaces)".

H.323 Interop Tweaks section

In addition to the standard fields, several advanced settings are available in the H.323 Interop Tweaks section. They are normally not necessary and are merely used to solve compatibility problems with some PBXs:

  • No Faststart: The H.245 faststart procedure is enabled as standard. Outgoing calls are made with faststart, incoming calls with faststart are answered with faststart.
    A checked check box disables the H.245 faststart procedure. Outgoing calls are made without faststart, incoming calls with and without faststart are answered without faststart.
  • No H.245 Tunneling: The H.245 tunneling procedure is enabled as standard. The voice data connection is negotiated in the TCP signalling connection*) already available. This can be advantageous in connection with NAT and firewalls.

A checked check box disables the H.245 tunneling procedure, meaning that a separate TCP connection is set up for this negotiation. This applies to the signalling connection leading out of the gatekeeper.

  • Suppress HLC: A checked check box disables the transmission of HLC (High Layer Compatibility) information elements.
  • Suppress FTY: A checked check box disables the transmission of FTY (Facility) information elements.
  • Suppress Subaddress: A checked check box disables the transmission of Subaddress information elements.
*) From a technical viewpoint, the H.245 protocol does not establish its own TCP connection, 
but shares the H.225 TCP connection.


SIP Interop Tweaks section

Miscellaneous interoperability options for SIP.

Proposed Registration Interval Set in seconds, default is 120 seconds
Accept INVITE's from Anywhere If disabled, registered interfaces will reject INVITE's not coming from the SIP server with "305 Use Proxy".
Enforce Sending Complete Affects handling of "484 Address Incomplete" responses. If enabled and "484 Address Incomplete" is received, the call is cleared. If not enabled and "484 Address Incomplete" is received, the call is retained and re-initiated in case of new dialing digits.
No Video Removes Video Capabilities from outgoing media offer.
No Early Media Ignore any SDP answer received before final connect response.
No Inband Information on Error Controls interworking of Q.931 DISC message. If this option is set, DISC message is always interworked into BYE request.
No Inband Disconnect TBD.
No Remote Hold Signaling Disables interworking of "inactive" into RemoteHold (affects connected SIP calls only).
Take Refer-To URI as Remote Target URI TBD.
To Header when Sending INVITE Affects only outgoing diverted calls . Called Party: If set we write CDPN into To header of outgoing INVITE (and DGPN into History-Info header). Original Called Party: If set we write the DGPN into To header of outgoing INVITE (and CDPN into Request-URI).
From Header when Sending INVITE Controls the local URI (From header) of outgoing calls. Applys to registered interfaces only. Fixed AOR: Fixed AOR is used as From-URI regardless of the actual calling party number. AOR with CGPN as Display: Fixed AOR is used as From-URI and calling party number is added as display string. CGPN in user part of URI: Variable From-URI with actual calling party number.
Identity Header when Sending INVITE Controls the identity header (P-Preferred-Identity, P-Asserted-Identity and Remote-Party-Id). CGPN in user part of URI: Variable From-URI with actual calling party number. Fixed AOR: Fixed AOR is used as Identity-URI regardless of the actual calling party number.
Reliability of Provisional Responses Controls the support of PRACK (RFC-3262). Supported: Supported as optional extension. Required: Required as mandatory extension. Disabled: Hide support for PRACK extension.
Microsoft Presence Format Enables special Microsoft Presence format/scheme, only used for SIP Federation with Lync 2013.

CGPN/CDPN Mappings

A detailed description may be found in the chapter entitled "Administration/Gateway/Interface/CGPN-CDPN Mappings".

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