SIP Advanced

Last Updated : Apr 22, 2022 |
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Resource Id: sipAdvanced:9521

Additional configuration information

For additional information regarding the Media Connection Preservation setting, see Media Connection Preservation.

Configuration settings

These settings are mergeable, with the exception of the Media Connection Preservation setting.
  • Changing the Media Connection Preservation setting requires a “merge with service disruption”. When the configuration file is sent to the system, the SIP trunk is restarted and all calls on the line are dropped.

Association Method

When the IP Office receives an incoming SIP call, it needs to match the call to one of its SIP line.
  • Lines are checked for a match in Line Number order until a match occurs.
  • The method used to check for a match on a line uses the line's Association Method.
  • If no match occurs on any line, the request is ignored.
This process enables support of multiple SIP lines with the same address settings. For example, for scenarios that require support of multiple SIP lines from the same ITSP. That can occur when the same ITSP supports different call plans on separate lines, or where all outgoing SIP lines are routed from the system through an additional on-site system.
Field
Description
By Source IP Address
Uses the source IP address and port of the incoming request for association. The match is against the configured remote end of the SIP line, using either an IP address/port or resolution of an FQDN. For UDP calls, the local Listen Port is also used for the match.
  • For TCP/TLS connections, the IP Office establishes a connection to the remote address and port specified on the SIP line.
  • For UDP, non-call dialogs and call starting dialogs must use the remote address and port specified on the SIP line.
It is recommended that the remote end does not change these value as that may prevent NAT traversal.
"From" header hostpart against ITSP domain
Uses the host part of the From header in the incoming SIP request for association.
  • The match is against the LineSIP LineITSP Domain Name.
R-URI hostpart against ITSP domain
Uses the host part of the Request-URI header in the incoming SIP request for association.
  • The match is against the LineSIP LineITSP Domain Name.
"To" header hostpart against ITSP domain
Uses the host part of the To header in the incoming SIP request for association.
  • The match is against the LineSIP LineITSP Domain Name.
"From" header hostpart against DNS-resolved ITSP domain
Uses the host part of the From header in the incoming SIP request for association.
  • The match is found by comparing the From header against the IP address resolution of the LineSIP LineITSP Domain Name or, if set, the LineSIP TransportITSP Proxy Address setting.
"Via" header hostpart against DNS-resolved ITSP domain
Uses the host part of the VIA header in the incoming SIP request for association.
  • The match is found by comparing the VIA header against the IP address resolution of the LineSIP LineITSP Domain Name or, if set, the LineSIP TransportITSP Proxy Address setting.
"From" header hostpart against ITSP proxy
Uses the host part of the From header in the incoming SIP request for association.
  • The match is against the LineSIP TransportITSP Proxy Address setting.
"To" header hostpart against ITSP proxy
Uses the host part of the From header in the incoming SIP request for association.
  • The match is against the LineSIP TransportITSP Proxy Address setting.
R-URI hostpart against ITSP proxy
Uses the host part of the Request-URI in the incoming SIP request for association.
  • The match is against the LineSIP TransportITSP Proxy Address setting.

Addressing

Field
Description
Call Routing Method
Default = Request URI.
This field selects which incoming SIP information is used for incoming number matching by the IP Office to route incoming calls. The options are to match the Request URI or the To Header element provided with the incoming call.
Use P-Called-Party
Default = Off.
When enabled, IP Office reads the P-Called-Party ID header if present in the SIP message and routes the incoming SIP calls based on it. The feature can be enabled on public SIP trunk interfaces.
If enabled and the header is not present in the SIP message, the IP Office uses the header configured in the Call Routing Method for incoming call routing.
Suppress DNS SRV Lookups
Default = Off.
Controls whether to send SRV queries for this endpoint, or just NAPTR and A record queries.

Identity

Field
Description
Use Phone Context
Default = Off.
When enabled, signals SIP enabled PBXs that the call routing identifier is a telephone number.
Add user=phone
Default = Off.
This setting is available when Use Phone Context is enabled.
When enabled, this setting adds the SIP parameter User with value Phone to the From and To SIP headers of outgoing calls.
Use + for International
Default = Off.
When enabled, outgoing international calls use E.164/International format with + followed by the country code and then the telephone number.
Use PAI for Privacy
Default = Off.
When enabled, if the caller ID is withheld:
  • The SIP message From header is made anonymous
  • The caller identity is inserted into the P-Asserted-Identity header.
This should only be used in a trusted network and must be stripped out of the SIP message before it is forwarded outside the trusted domain.
Use Domain for PAI
Default = Off.
  • When disabled, the DNS resolved IP address of the ITSP Proxy is used for the host part in the P-Asserted-Identity header.
  • When enabled, the Domain is used.
Caller ID FROM Header
Default = Off.
Incoming calls can include caller ID information in both the From field and in the PAI fields. When this option is enabled, the caller ID information in the From field is used rather than that in the PAI fields.
Send From In Clear
Default = Off.
When enabled, the user ID of the caller is included in the From field. This applies even if the caller has selected to be or is configured to be anonymous. However, their anonymous state is still honored in other fields used to display the caller identity.
Cache Auth Credentials
Default = On.
When enabled, credentials challenge and response information from a registration transaction is cached by the IP Office and automatically inserted into later SIP messages without waiting for a subsequent challenge. This speeds up connections but must be supported by the other end of the connection.
Add UUI header
Default = Off.
When enabled, the User-to-User Information (UUI) is passed in SIP headers to applications.
Add UUI header to redirected calls
Default = Off.
When enabled, the UUI is passed in SIP headers for calls that are redirected. For example, on forwarded and twinned calls.
This field can be enabled if Add UUI header is enabled.
User-Agent and Server Headers
Default = Blank (Use system type and software level).
The value set in this field is used as the User-Agent and Server value included in SIP request headers made the line.
  • If blank, the type of IP Office system and its software level are used.
  • Setting a unique value can be useful in call diagnostics when the IP Office has multiple SIP trunks.
Send Location Info
Default = Never.
This option is useable with SIP ISPs that support RFC 4119/RFC 5139. When enabled, emergency calls send the address information associated with the dialing extension's location. See Configuration for Emergency Calls.
The options are:
  • Never: Do not send location information.
  • Emergency Calls: For Dial Emergency calls, send the address information configured for the dialing extension's location.

Calling Number Verification

These settings configure the SIP trunks use of STIR protocols for calling number verification.
For more details, see SIP Calling Number Verification (STIR/SHAKEN).
Field
Description
Calling Number Verification
Default = Off
Sets whether the line uses calling number verification.
Incoming Calls Handling
Default = Allow Not Failed
Set which calls are accepted by the system based on the attestation level of the call.
  • System - Use the default system setting (System VoIPVoIP SecurityCallng Number Verification).
  • Allow All - Allow all calls regardless of calling number verification.
  • Allow Validated - Only accept verified calls with full or partial attestation.
  • Allow Not Failed - Accept all calls expect those that specifically failed verification. Note this can include calls with no reported verification result.

Media

Field
Description
Allow Empty INVITE
Default = Off.
When set to On, allows 3pcc devices to initiate calls to IP Office by sending an INVITE without SDP.
Send Empty re-INVITE
Default = Off.
This option is only available if Line | SIP Line | VoIP | Re-Invite Supported is selected.
If set to On, when connecting a call between two endpoints, IP Office sends an INVITE without SDP in order to solicit the full media capabilities of both parties.
Allow To Tag Change
Default = Off.
When set to On, allows the IP Office to change media parameters when connecting a call to a different party than that which was advertised in the media parameters of provisional responses, such as 183 Session Progress.
P-Early-Media Support
Default = None.
The options are:
  • None: IP Office will not advertise support of this SIP header and will always take incoming early media into account regardless of presence of this header
  • Receive: IP Office will advertise support of this SIP header and will discard incoming early media unless this header is present in the SIP message.
  • All: IP Office will advertise support of this SIP header, will discard incoming early media unless this header is present in the SIP message and will include this SIP header when providing early media.
Send SilenceSupp=off
Default = Off.
Used for the G711 codec. When checked, the silence suppression off attribute is sent in SDP on this trunk.
Force Early Direct Media
Default = Off.
When set to On, allows the direct connection of early media streams to IP endpoints rather than anchoring it at the IP Office.
Media Connection Preservation
Default = Disabled.
When enabled, allows established calls to continue despite brief network failures. Call handling features are no longer available when a call is in a preserved state. Preservation on public SIP trunks is not supported until tested with a specific service provider.
Indicate HOLD
Default = Off.
When enabled, the system sends a HOLD INVITE to the SIP trunk endpoint.
Media Security
Default = Off
When enabled, the IP Office advertises support of this SIP header, to indicate that audio is configured to be secure and is enforced to use SRTP only. This supports the SIP security header defined by RFC3329.
This option is available only when:
  • TLS is being used.
  • Line | SIP Line | VoIPMedia Security is selected and set to Enforced.
  • Line | SIP Line | VoIPFax Transport Support is not set to T38 or T38 Fallback.
When the configuration file is sent to the system, the SIP trunk is restarted and all calls on the line are dropped.

Call Control

Field
Description
Call Initiation Timeout (s)
Default = 4 seconds. Range = 1 to 99 seconds.
Sets how long the IP Office system should wait for a response to an attempt to initiate a call before following the alternate routes set in an ARS form.
Call Queuing Timeout (m)
Default = 5 minutes.
  • For incoming calls, this sets how many minutes the IP Office waits before dropping a call that is waiting for VCM resources or has remained in the unanswered state.
  • For outgoing calls, this sets how many minutes the IP Office waits for a call to be answered after receiving a provisional response.
Service Busy Response
Default = 486 - Busy Here (503 - Service Unavailable for the France2 locale).
For calls that result in a busy response from IP Office, this setting determines the response code. The options are:
  • 486 - Busy Here
  • 503 - Service Unavailable
on No User Responding Send
Default = 408-Request Timeout.
Specifies the cause to be used when releasing incoming calls from SIP trunks, when the cause of releasing is that user did not respond. The options are 408-Request Timeout or 480 Temporarily Unavailable.
Action on CAC Location Limit
Default = Allow Voicemail
When set to Allow Voicemail, the call is allowed to go to a user's voicemail when the user's location call limit has been reached. When set to Reject Call, the call is rejected with the failure response code configured in the Service Busy Response field.
Suppress Q.850 Reason Header
Default = Off.
When SIP calls are released by sending BYE and CANCEL, a release reason header is added to the message. When set to On, the Q.850 reason header is not included.
Emulate NOTIFY for REFER
Default = Off.
Use for SIP providers that do not send NOTIFY messages. When set to On, after IP Office issues a REFER, and the provider responds with 202 ACCEPTED, IP Office will assume the transfer is complete and issue a BYE.
No REFER if using Diversion
Default = Off.
When enabled, REFER is not sent on the trunk if the forwarding was done with 'Send Caller ID = Diversion Header'. Applies to Forwards and Twinning.
Related links
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