VoIP optimization options

Description:

In certain situations, it can be necessary to enable special functions, e.g. to allow connections to the SIP provider or to make sure that the correct telephone number is displayed.

This article describes functions for optimizing VoIP.

(Re)registration and registration interval:

1) (Re)registration:

(Re)registration controls whether registration with the SIP provider takes place. Some lines do not require registration.

1.1) Open the configuration for the router in LANconfig and navigate to the menu item Voice Call Manager -> Lines -> SIP lines.

Screenshot of a technical configuration interface for a communication system, displaying options for SIP provider settings, call routing, ISDN lines, and other network services.

1.2) Edit the desired SIP line and enable or disable (Re-)Registration as needed.

Screenshot of a telecommunications settings interface featuring various configuration options such as security settings, provider details, and login information, with fields for entering passwords and account modes.

Important:
If (Re-)Registration is disabled, the SIP proxy port assigned by the provider must be specified for the SIP line on the Advanced tab.

Screenshot of a technical configuration interface for SIP (Session Initiation Protocol) services, showing options such as routing tags, codec settings, and DTMF signaling configuration.

2) Registration interval:

The Registration interval is the interval proposed to the provider by the Voice Call Manager for the registration. By default, the interval is 480 seconds. The Registration interval can only be changed in WEBconfig or via the command line.

2.1) Use any web browser to connect to the web interface of the router and navigate to the menu LCOS Menu Tree -> Setup -> Voice-Call-Manager -> Lines -> SIP-Provider -> Line -> <Name of the SIP line>.

Info:
The command-line path is Setup/Voice-Call-Manager/Lines/SIP-Provider/Line/<Name of the SIP Line>.

Screenshot of a network configuration menu displaying options for setup wizards, system information, file management, HTTP session setup, and SIP provider configurations for voice calls.

2.2) Change the Registration interval according to your provider’s specifications.

Image depicts a technical configuration menu with various settings including user IDs, domain details, display names, privacy methods, and registration options.

Trunk-Inc-Cld-In-ToHeader:

The function Trunk-Inc-Cld-In-ToHeader must be enabled if the SIP provider sends the number of an incoming call in the To field instead of the Request Line field.

This function only affects incoming calls. The configuration can only be changed in WEBconfig or via the command line.

1) Use any web browser to connect to the web interface of the router and navigate to the menu LCOS Menu Tree -> Setup -> Voice-Call-Manager -> Lines -> SIP-Provider -> Line -> <Name of the SIP line>.

Info:
The command-line path is Setup/Voice-Call-Manager/Lines/SIP-Provider/Line/<Name of the SIP line>.

Screenshot of a software's configuration interface displaying menu options such as System Information, LCOS Menu Tree, File Management, HTTP Session Setup, and Voice Call Manager, with fields for SIP provider configuration including line name, mode, domain, port, user ID, authentication name, display name, and secret.

2) Enable or disable the function Trunk-Inc-Cld-In-ToHeader as required.

An image displaying a configuration menu for a technical interface, featuring various settings such as registration status, privacy method, transport type, and line control options.

Using diversion headers:

Diversion headers are used for call diversion. Options for this are as follows:

1) SIP 302:

SIP 302 is used to signal call diversion to the provider. That way, no additional voice channel is required to establish a call to the second party.

SIP 302 can only be used with a SIP trunk.

If an ISDN PBX is used, the Partial rerouting / Call deflection feature must be enabled on the PBX.

1.1) Open the configuration for the router in LANconfig and switch to the menu item Voice Call Manager -> Lines -> SIP lines.

Screenshot of a technical configuration interface for networking equipment, displaying sections for SIP router settings, outgoing calls management, SIP lines configuration, and various networking protocols.

1.2 Edit the SIP line, open the Advanced tab, and change the Call forwarding using SIP 302 settings as required.

Screenshot of a technical interface with various configuration options and labels such as 'SIPlinesEditEntry', 'Crimes', 'Temenos', and 'Dialing'.

2) CLIP no screening:

This feature establishes a separate call to the diversion destination, which occupies a second voice channel. At the destination, the number of the diverting line is displayed, not the original number.

CLIP no screening must be supported by the provider. Single accounts are usually not supported.

3) Call transfer at the provider

As with SIP 302, no separate voice channel is required. Call transfer takes place at the provider.

Call transfer at the provider usually only works with single accounts.

Transmission of number fields in outgoing calls (FROM, PPI, PAI):

The SIP ID can be transmitted either in the FROM field, via the P-Preferred Identity (PPI), or via the P-Asserted Identity (PAI).

Depending on the provider, it may be necessary to transmit the SIP ID via a different field, as otherwise the call might be rejected by the provider.

The transmission of the SIP ID is handled differently, depending on whether a SIP trunk or a single account is used.

1) SIP trunk:

By default, the SIP ID is transmitted via the PPI / PAI. The actual telephone number is transmitted via the FROM field. If required, the configuration can be changed so that the SIP ID is transmitted via the FROM field and the actual number via the PPI / PAI.

1.1) Navigate to the menu Voice Call Manager -> Lines-> SIP lines.

Screenshot of a technical configuration interface displaying options for managing SIP lines, call routing, ISDN settings, and other telecommunications features.

1.2) Edit the SIP line, open the Advanced tab, and select FROM from the drop-down menu for SIP-ID Transmission.

An image of a complex technical configuration interface displaying options for DTMF signaling, telephone events, and call forwarding, with scattered, partially visible text labels and settings.

1.3) If the option Trusted Area activated is checked and RFC3325 is selected from the drop-down menu for Transmission method, the PPI is turned into the PAI if the provider is considered a “trusted area”.

Note:
With the setting None, the SIP ID is not transmitted at all, and with the setting IETF-Draft-Sip-Privacy-04, the SIP ID is transmitted as Remote Party ID (RPID).

Image showing a technical configuration menu related to SIP (Session Initiation Protocol) with options for DTMF signaling, telephone event settings, and call forwarding.

2) Single account:

The SIP ID is always transmitted in the FROM field.

Transmission of number fields in incoming calls (PPI/ PAI):

In LCOS versions up to 10.30 RU1, only the source telephone number of the SIP client or SIP PBX in the contact header is processed.

With LCOS 10.32 Rel or later, the PPI / PAI is evaluated in addition to the contact header.