Configure SIP Trunking

SIP Trunk Options

Field

Description

Notes

Name

Enter a name for the SIP trunk.

Remote trunk endpoint address

Enter the IP address or DNS hostname of the remote endpoint. If the carrier has multiple IP addresses, enter a Fully Qualified Domain Name (FQDN) for the hostname.   

If a FQDN is entered as the hostname, the system will use DNS to resolve the name into a list of IP addresses and then accept traffic from any of them.

Remote trunk endpoint port

Enter port configuration for the remote endpoint OR accept the default.

Default is 5060.

Accept traffic from any port

Select to accept incoming requests from any port at the remote trunk endpoint IP address. Clear to accept requests only from the specified IP address and port.  

Note that outgoing requests are always sent to the specified IP address and port.  

Default is False.

DNS SRV Support

Select the checkbox to enable the DNS SRV feature.

• If this feature is enabled, MBG uses SRV Trunking mode for the Trunk Service Provider.

Note: the "Remote Trunk Endpoint Port" and "Remote Trunk Endpoint address" fields are automatically configured, and cannot be edited.

• If this feature is not enabled, MBG uses Fixed Trunking mode for the Trunk Service Provider.

Note: The "Remote Trunk Endpoint Port" and "Remote Trunk Endpoint address" fields must be manually configured.

Default is no selection.

DNS SRV query domain

Enter the domain of the SIP Trunk Provider for SRV query.

For example, abccompany.com

DNS SRV resiliency timeout

Set the time to lapse between an unresponsive Trunk Endpoint before attempting to use an alternate Trunk Endpoint.

Default is 5 seconds.

Options keepalives

This setting controls whether SIP Options messages are sent to SIP Trunks as keepalive mechanisms.

• Never to prevent keepalives from being sent.

• Always to force keepalives to be sent.

Note:

• For a resilient trunk configuration, set the Options keepalives to Always and Options interval to 20.

• This setting is automatically disabled when DNS SRV Support is enabled, because SRV does keepalives by doing resiliency when needed.

Default is Always.

Options interval

If "Options keepalives" is enabled, this is the interval in seconds at which keep-alive messages are sent.

Default is 60.

Rewrite host in PAI

Select to change the P-Asserted-Identity (PAI) header contained in outgoing ICP messages to the host IP of the MBG's public interface. Clear to leave the PAI in outgoing ICP messages unchanged.

Default is True.

Remote RTP framesize (ms)

This setting controls the RTP framesize on the set side of MBG. By default, the requested framesize is respected. If you select a non-default setting (e.g. 30 ms), the specified value will be used. For more information about RTP framesize, see the MBG Engineering Guidelines.

Notes:

• This setting should be changed only if a specific framesize is required on the set side of MBG.

• This setting does not affect the framesize used between MBG and the ICP (MiVoice Business or MiVoice Office 250).

Default is Auto.

Idle timeout (s)

This timer monitors "no traffic" conditions on the trunk. When there is no traffic on the trunk for the selected amount of time (default is one hour), the timer expires and all connections are closed until the next packet is received.

Default is 3600.

RTP address override

Select the interface the MBG server uses to access the SIP trunk provider. If no selection is made, the interface is set in accordance to the configured route to the SIP trunk provider.

Default is no selection.

After you update this setting, restart the system to have the change take effect. For details, see Stop MBG and Start MBG.

Local streaming between trunk calls

Select this option to allow the MBG to drop out of the media path on calls that involve only public trunk participants.

With this option disabled (the default), MBG will anchor the media streams and connect them internally. This uses more bandwidth. With this option enabled, MBG will set up the call such that the carrier's SBC connects the two streams directly, and MBG is only involved in call signaling. Some carriers require the former and some require the latter. To determine which setting is required by your SIP service provider, consult the Mitel SIP Center of Excellence Interop Guide.

Default is disabled.

Enable this option only if local streaming is required by your SIP service provider. In all other circumstances, leave this option disabled.

The wrong Local streaming setting for the SIP trunk can result in call failures or no audio.

PRACK support

This option controls whether the "Provisional Response ACKnowledgement" (PRACK) method is used.

• Use master setting to use the global PRACK option programmed on the Settings screen. This option controls whether MBG supports PRACK between itself and the peer, and is the default setting for all SIP trunks.

• Enabled to enable the trunk to use PRACK while the global PRACK option is disabled.

• Disabled to prevent the trunk from using PRACK while the global PRACK option is enabled.

For example, if the PRACK option is globally disabled on MBG and the peer but enabled on the trunk, then MBG will support PRACK only on calls between itself and the trunk. Alternatively, if the PRACK option is globally enabled on MBG but disabled on the trunk, then MBG will use PRACK only between itself and the peer but not with the trunk.

Default is Use master setting.

Most peers now support PRACK, which can be useful in interoperability scenarios with the PSTN (see RFC 3262). If the remote SIP trunk supports PRACK, this option should be enabled.

Log verbosity

At times, it may be necessary to increase log verbosity for specific troubleshooting purposes. Select Normal, Very Quiet, Quiet, Verbose, Very Verbose, or Use master setting to use the Log Verbosity setting programmed on the Settings screen.

Default is Use master setting.

Authentication username

Some SIP service providers require authentication of PRACK requests before allowing a trunk connection. Username and password information is provided by the SIP Service Provider and must match the authentication credentials in the SIP Peer Profile form in the MiVoice Business System Administration Tool.

If you cannot make a call and the log contains this string:
"SipUA: RX <<< PRACK (401) response",

then you must enter this information.

Authentication password

Set-side RTP security

This option controls whether the Secure Real-time Transport Protocol (SRTP) is required to provide encryption, message authentication and integrity for the media stream on the set side of MBG.

• Allow to use either RTP or SRTP.

• Require to use only SRTP; RTP is disallowed. If this option is selected, the remote SIP set connected to MBG must support SRTP. If the set lacks SRTP, MBG will cause its calls to fail.  

Default is Allow.

ICP-side RTP security

This option controls whether the Secure Real-time Transport Protocol (SRTP) should be used as the preferred protocol on the ICP side of MBG.

• Disable to use only RTP; SRTP is disallowed.  

• Prefer to use either SRTP (the default) or RTP.

Default is Disable (only RTP is allowed).

SIP adaptation receive pipeline

From the drop-down, select the pipeline to receive SIP messages.

The drop-down list contains the pipelines configured in SIP Adaptation page.

SIP adaptation send pipeline

From the drop-down, select the pipeline to send SIP messages.

Add additional routing rule

(Optional) Enter the DID routing rule mask for this SIP trunk to use. See Configuring DID Routing Rules for more information.

1. In the Routing rules section of the screen, locate the place where you wish to add the new rule. Use the following screen tools to sort through the list.

Note: If the trunk does not yet have any rules, a single unprogrammed rule will be listed by default.

2. Click Prepend to add the new rule ahead of an existing rule, or Append to add the new rule after an existing rule.

3. Specify the rule parameters:

4. Match: Select the match criteria: Request URI, From header URI, or To header URI.

5. Rule: Enter the rule, making sure to adhere to the format outlined below.  

6. Primary: Select the primary ICP to route the call.

7. Secondary: Select the secondary ICP to route the call (if applicable).

4. Click Save.

The rules are applied in descending order starting from the top of the list. Use the Raise and Lower fields to re-sort the list.

There is no limit to the number of routing rules you may add.

Re-Invite Conversion

If this option is selected, when MBG receives RE-INVITE messages from the ICP that lack SDP, it will perform  a conversion, adding SDP before forwarding the messages to the SIP trunk.

If this option is cleared, MBG will not perform the conversion.

• Default is no selection (MBG does not perform the conversion).

• RE-INVITE messages without SDP are also known as SLOWSTART RE-INVITE messages.

• Enable this option only if required by your SIP service provider. For more information, consult the Mitel SIP Center of Excellence Interop Guide.

• If this option is enabled and if your SIP service provider requires authentication for INVITE messages, you must configure the Authentication Username and Password for your trunk.