Reference8:Administration/Relay/VOIP/GW

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Contents

Basic

Name

Descriptive name of this gateway interface.

Disable

Disables an interface without deleting the interface configuration.

Protocol

Select the Voice over IP signaling protocol here. You can choose between:

  • H.323
  • SIP (over UDP)
  • TSIP (over TCP)
  • SIPS (over TLS)

Mode

Select the interface type here.

  • Gateway without Registration

This interface type does not register at any gatekeeper/registrar.

  • Register as Endpoint / Register as Gateway

This interface type acts as a registration client and registers itself at the configured gatekeeper/registrar.

  • ENUM

Like the 'Gateway without Registration' this type does not register itself. It additionally performs ENUM lookups for the called party number in order to find a appropriate URI (h323 or sip). If no useful URI can be retrieved, the call is re-routed to find another outgoing interface.

  • Gatekeeper / Registrar

This interface type acts as a registration server (gatekeeper/registrar). It does not register itself. It accepts incoming registrations for the configured aliases (name and/or number).

Gatekeeper Address (primary and secondary) and Mask

The meaning depends on the interface type:

  • Gateway without Registration

Here the remote side signalling IP address(es) is to be configured.
The secondary address is optional. A backup server can be configured here.
Together with Mask this also acts as a filter for incoming calls.

  • Register as Endpoint / Register as Gateway

Here the IP address(es) of the gatekeeper is to be configured.
The secondary address is optional. A backup server can be configured here.

  • ENUM

Not available.

  • Gatekeeper / Registrar

For H.323, clients are only allowed if their IP address match the Address/Mask settings (with no address specified matching all IP addresses).

Gatekeeper Identifier (H.323 only)

Used for Register as Endpoint / Register as Gateway only.
Used to select a dedicated gatekeeper running on the remote IP address.

Domain (SIP only)

Used for Register as Endpoint / Register as Gateway only.
Used to build the AOR (Address-of-Record).

Local Port

Not used for SIP.

Authorization

Alias List

Media Properties

General Coder Preference The coder preference (Coder,Framesize, Silence Compression) to be used if a non-local media address is detected. If the preference is marked as exclusive no other coder is offered.
Local Network Coder The coder preference if a local media address is detected.
Enable T.38 Switches on Fax detection and switchover to T.38
Enable SRTP Offer media encryption. If remote party also offers media encryption, media will be encrypted and save against wiretapping.
No DTMF Detection Special mode to switch off DTMF detection during a call. If checked DTMF is transfered inband during a call. Use G711 coder and a link with no packet loss to transfer DTMF inband. This option can be used to transfer continous DTMF tones for some special applications. Out of band DTMF tones have a fixed length of approx 150ms.
Enable PCM Enable the media to be connected using the local timeslot switch if the call is between physical interfaces of the same gateway.
Media-Relay If this checkbox is set, media stream is routed thru this box. Helpful to keep private RTP adresses from being passed to external endpoints.
MOH Mode Special mode to use a physical interface as Music on Hold Source. The media is one way only and multiple calls are connected to the same channel on the physical interface.
Record to (URL) HTTP url where the recording file is to be stored. HTTP server must allow write access (PUT) at this location. One PCAP file is written for every call via this interface containing both RTP streams. Audio streams can be played using Wireshark. PCAP file can be converted into WAV file using pcap2wav tool.

Note: Neither MAC addresses nor IP addresses are real in the capture files. Not even From, To or Call-ID of the SIP signaling are real. The file name is the key. It contains the real call identifier (e.g. 6327c082e909d3118f920090330608df.pcap). CDRs must be used to get the call information releated to this call identifier.

H.323 Interop Tweaks

SIP Interop Tweaks

Proposed Registration Interval Set in seconds, default it 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 Inband Information on Error Regards ISUP-to-SIP signaling. On receipt of DISCONNECT with progress indicator on an early call (not yet connected) a "183 Session Progress" is going to be sent in SIP in order to make the caller hear the inband announcement (tone pattern) provided by the PSTN. The caller will terminate the call then. If this option is enabled, a final error response will be sent to the caller and no inband announcement will be heard at the callers side.
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.

Notes (SIP specific)

domain/realm

Normally, it is sufficient to specify Domain/Realm. The DNS is queried then with this name to obtain the DNS SRV entry _sip._tcp or _sip._udp, respectively. This may result in a list of service addresses, ordered by priority and weight. If no SRV entry is found, then the configured domain is queried as a host name.

If there is a Server Address configured, the Domain/Realm is used only for SIP URIs and Server Address is used as remote peer (including DNS resolving as described above).

If there is no Domain/Realm, the Server Address is used for building SIP URIs.

The two Server Address fields will be DNS-resolved.

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