# Editing the SIP provider

These settings provide the interface between the SIP provider and the communication server. You can enter the coordinates of your SIP providers here and specify the communication parameters to the SIP provider.

## Routing path

- An incoming call follows the routing path: SIP exchange -> SIP provider settings -> SIP account settings -> direct dialling plan -> ...

- An outgoing call via an SIP exchange follows the routing path: ... -> route - SIP trunk group -> SIP provider settings -> direct dialling plan -> SIP account settings -> SIP exchange.

- Each SIP provider has its own SIP trunk group.

## Resources and licences

- Up to 30 SIP voice channels are available for each SIP provider.

- For each SIP voice channel you need an SIP Access Channels licence.

Table 1. General settings
Parameter | Explanation
SIP provider | System-internal reference number of the SIP provider
Name | System-internal designation of the SIP provider
Trunk group | Trunk group allocation
Maximum incoming calls | No further outgoing calls are routed via this trunk group once the set limit is reached. This is signalled to the caller by means of the congestion tone. This allows you to ensure that lines remain free for incoming calls. You can also adjust this setting in the trunk group configuration.
Provider authentication | With several accounts: Each DDI number is linked with one SIP account. You can create and manage the SIP account at the end of this overlay view. With one account: Only one SIP account is needed for several DDI numbers. You can create and manage the SIP account at the end of this overlay view. Without accounts: The SIP access is direct and not via an SIP account. This setting is only relevant if the SIP exchange is not a SIP provider, but a different communication system (see also "Notes on the setting Provider authentication = Without accounts"). Already created SIP accounts are automatically erased.
Bandwidth control area | Predefined broadband range used for this SIP provider.
Gateway | If the default gateway is not selected, the default route is used.

For more information on multiple-gateway, see Multi-Gateway for SIP Trunks section in the System Functions and Features document.

## Notes on the setting Provider authentication = Without accounts

Note:

This setting is only relevant if the SIP exchange is not an SIP provider, but a different communication system. Before activating, you need to remove any SIP accounts already assigned.

Tip:

It is preferable to configure a private SIP network using the settings in Private SIP networking ( =uy).

Table 2. Registrar settings
Parameter | Explanation
Registrar address | Enter the host name or the registrar IP address and the port here. Note: To ensure the host name is resolved, make sure a valid DNS server address is configured in the IP network / IP addressing view. If you do not enter the port, the standard port 5060 is used.
Preferred registration interval | Once this period of time has elapsed, the communication server registers with the SIP registrar on a regular basis in order to maintain a faultless connection.
Realm name | Configurable second route header (used in special configurations, e.g. with an Mitel Border Gateway).
Registration procedure | Select the registration procedure.

Table 3. Proxy settings
Parameter | Explanation
Use DNS-SRV (RFC 3263) | The proxy addresses of the SIP provider are evaluated on the basis of the registrar name in accordance with RFC 3263, (locating SIP server). Note: This function is deactivated the instant you enter the proxy addresses under Primary proxy and Secondary proxy.
Primary proxy | Lets you enter the IP address or the host name of the SIP provider's primary proxy server. Syntax: <IP address>:<Port> or <Host name>:<Port>. If you do not enter the port, the standard port 5060 is used.
Secondary proxy | Lets you enter the IP address or the host name of the SIP provider's secondary proxy server. Syntax: <IP address>:<Port> or <Host name>:<Port>. If you do not enter the port, the standard port 5060 is used.

Table 4. SIP signalling
Parameter | Explanation
Use ‘+’ for the international prefix | The call number is also included, in canonical format.
Try to make external calls: Timeout | Once that time has elapsed, the communication server tries to set up the call via the next trunk group defined in the route (default value: 32 seconds).
’From’ field for CLIR | If call identification is suppressed at the calling user, the following sender is provided, depending on the selection (display name and address): Anonymous with privacy/critical (RFC 3261): Display name: anonymous@anonymous.invalid; Address: anonymous@anonymous.invalid As defined in SIP account (RFC 3323): Display name and address as defined in the SIP account. Displayed name is ’Anonymous’: Display name: anonymous@anonymous.invalid; address remains unchanged. Anonymous without privacy header (RFC 3261): Display name: anonymous@anonymous.invalid; Address: anonymous@anonymous.invalid
Send session refresh (RFC 4026) | The communication server attempts to negotiate a period for regular "Session Refresh Messages" with the SIP provider. For this, the SIP provider must support RFC4028.
Use destination URL from | The destination URL can be formed from the ’To’-Field or from the request line. The choice depends on the SIP provider.
Music on hold | Music on hold is played, provided it is activated throughout the system.
Music on hold: Signalling | The type of signalling for music on hold depends on what the SIP provider supports: Automatic: The communication server itself tries to recognise which of the two RFCs the SIP provider supports. According to RFC 3264 The SIP provider supports signalling according to the RFC An Offer/Answer Model with the Session Description Protocol, (SDP), June 2002. According to RFC 2543 The SIP provider supports signalling according to the RFC SIP: Session Initiation Protocol As active media update: The communication server keeps the 2-way media connection. This allows to play music on hold into the call channel from the communication server instead from the SIP provider. Signal connection update: The SIP provider is informed about the change of the media port by a separate SDP message. No signalling (no media update): the SIP provider does not deliver any signalling and the communication server plays music on hold into the call channel.
Send redirecting information | Redirecting information allows the called party to see whether the call was redirected and, if so, by whom. Two different methods are defined for this purpose for SIP. The parameter can be configured for each SIP provider as well as for each SIP terminal. No: No redirection/redirecting information is displayed. Yes, using 'Diversion header (recursing)': Redirection/redirecting information is displayed in the case of incoming calls only. The call is forwarded in the communication server. Yes, using 'Diversion header (non recursing)': The call forwarding for outgoing calls is indirect, with the communication server sending back ‘Response 302’ (Moved Temporarily) with the necessary redirection information to the SIP phone. The SIP phone itself then makes the call to the forwarding destination and shows the redirection/redirecting information on its own display. Note: Call forwarding with Response 302 is not possible in every case.
Codec | Choose the preferred codec here: G.711a: Uncompressed codec with high audio quality. Suitable for links with large bandwidths. The bit rate is 64 kbit/s. Uses the German tone signalling process. G.711u: Uncompressed codec with higher audio quality. Suitable for links with large bandwidths. The bit rate is 64 kbit/s. Uses the American tone signalling process. G.729: Compressed codec with medium audio quality. Suitable for links with limited bandwidth. The bit rate is 8 kbit/s.
Call transfer mode | You can select here whether the REFER or Re-INVITE method should be used during an external call transfer. Note: The REFER method is only used when both of the users to be transferred are found at the same SIP provider.
Relay RTP data via communication server for trunk-trunk connections (indirect switching) |
Identity (RFC 3325) | The identity method according to RFC 3325 is supported.
Identify (RFC 3325) | Lets you select the header type which your SIP provider evaluates for identification information. PPI P-Preferred-Identity or PAI P-Asserted-Identity
PPI/PAI header content | Lets you specify the information to be communicated to the SIP provider in the PPI/PAI header: System CLIP: Depending on the user's CLIP settings, either the direct dialling number or the CLIP of the user is communicated (default setting). Suppressed: No PPI header communicated. SIP ID: The SIP ID of the registered SIP account is communicated.
Ignore ‘Display name’ | None: Incoming calls: Outgoing calls: Both:
Use originator URL from | PAI header: ‘From’ field:
PRACK support (RFC 3262) | The PRACK method according to RFC 3262 is supported.
Use SAVP for SRTP | This is a provider-dependent setting. SAVP is used together with TLS/SRTP. AVP is used together with TLS/SRTP (default setting).
Passive·support·of the 'Precondition' mechanism (RFC3312) | This is a provider-dependent setting. Enable for compatibility with 1TR114/1TR118, Deutsche Telekom.
Include 'Digest' in each SIP request | This is a provider-dependent setting. Enable for compatibility with 1TR114/1TR118, Deutsche Telekom.
Event Package for Registrations (RFC 3680) | This is a provider-dependent setting. Enable for compatibility with 1TR114/1TR118, Deutsche Telekom.
Status send when no free channel available | This is a provider-dependent setting. It determines the SIP message to the provider, when there are no free channels available at the moment. 503 ‘Service unavailable’ (default value for most countries) 486 ‘Busy here’ (default value for Sweden, Denmark and Norway)
URI used for SIP signalling | This is a provider-dependent setting (for example, used for Telia Sweden) URI provider: Default value for Sweden URI symmetric: Default value for all other countries

Table 5. NAT settings
Parameter | Explanation
TCP keep alive | Display the status of whether the system remains reachable for incoming SIP call due to keep alive on the TCP layer (if this is supported by the SIP provider).
SIP keep alive | The communication server periodically updates the NAT table on its own firewall using notification messages to the proxy server. This means that the system remains reachable for incoming SIP calls.
ALG support | Supports the connection to SIP providers (depends on the provider). The IP packets that contain SIP signalling information are opened by the ALG (Application Layer Gateway) and the private IP address is replaced by the public IP address. (The public IP address in the communication server must be configured.)
Relay RTP data via communication server | Indirect switching: While setting up connection to another IP endpoint, the voice data is forwarded via the communication server and not directly. This helps reduce NAT and firewall problems. Direct switching: During connection set-up to another IP endpoint, the voice data is forwarded directly. Note: Indirect switching requires two additional VoIP channels on the communication server.

Table 6. Transport protocol and SIP access
Parameter | Explanation
Transport protocol | Allows you to select the required transport protocol, or the transport protocol you want.
Support Security Mechanism (RFC 3329) | This field is enabled when Persistent TLS is selected from Transport Protocol drop-down list.
No Path MTU Discovery | Enables fragmentation of UDP/TCP messages sent to the SIP provider if the packet size exceeds the configured MTU size. For this configuration, Transport protocol must be set to UDP or TCP.

## Working with SIP provider profiles

You have the provision for storing the main SIP provider settings in XML files, managing and modifying these files, and calling up the SIP providers:

- To create a SIP provider profile, configure the settings for a SIP provider, and then click the Export SIP provider profile button. The parameters are exported to an XML file and stored in a temporary directory on your communication system. Depending on your browser settings, the file opens in the browser or you are prompted to save it.

- To create a new SIP provider with the help of a SIP provider profile, right-click the box of the corresponding network interface in the routing overview and select the entry Import SIP provider profile in the context menu.

## Managing a SIP account

To manage the SIP account, proceed as follows:

- To open a new SIP account, click the New button and edit the settings on the Add SIP account overlay view.

- To delete a SIP account, click the recycle bin icon to the left of the SIP account and follow the user prompts.

- To edit a SIP account, click the SIP account reference number and edit the settings on the SIP account overlay view.

Parent topic: Incoming call routing
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