Call routing rules

This page explains how to manage and configure call routing rules for Pexip within Mividas Core. It includes options to create, edit, or test rules, prioritize them, and define specific conditions for routing calls, such as protocol, location, and media settings. It also covers rule synchronization with Pexip, fallback configurations, and bandwidth limitations. A “Test Rule” feature allows validation of rule behavior for troubleshooting.

Page actions
ADD	Opens a form to add a new rule.
DOWNLOAD RULES	Downloads the latest set of rules from Pexip.
Actions	1. Page refresh ( ) allows you to refresh the view manually 2. Access relevant documentation ( )

Overview

Search and filters	Description
	Search for a rule.
Show for	Choose to display rules based on a specific cluster. This option only shows if there is multiple Pexip Infinity clusters.
TEST RULE	Opens the Test Rule window. More details can be found under Test Rule.

• Active – Displays whether the rule is active and if it matches on incoming and/or outgoing calls:
  • ( ) – The rule is enabled.
  • ( ) – The rule is disabled.
  • ( ) – Matches on incoming calls.
  • ( ) – Does not match on incoming calls.
  • ( ) – Matches on outgoing calls.
  • ( ) – Does not match on outgoing calls.
• Prio – Shows the priority of the rule. The lower the value, the earlier it triggers.
• Name – Displays the rule’s name; clicking on the name allows you to edit the rule. It also displays some information about the rule:
  • Sync – Shows that the rule is synchronized.
  • Type: registration – Shows the call target of the rule. Can be either: external, registered, mssip_conference_id, mssip_server, gms_conference or teams_conference.
  • Protocol: SIP – Displays the outgoing protocol for this rule. Can be either: SIP, H.323, MS-SIP, RTMP, Google meet or Microsoft Teams.
• Match incoming – Displays what protocols of incoming calls it should match on. X is enabled and X is disabled.
  • R – Matches only on registered devices.
  • W – Matches on WebRTC calls.
  • S – Matches on SIP calls.
  • L – Matches on Lync/Skype calls.
  • H – Matches on H.323 calls.
• Local in / out – Displays selected incoming and outgoing locations for the rule.
  • LocationA – The rule is matching on “LocationA” for incoming calls.
  • No – No specific incoming location has been chosen for the rule match.
  • LocationB – The rule is matching on “LocationB” for outgoing calls.
  • No – No specific outgoing location has been chosen for the rule match.
• Hits – Displays the total amount of times this rule has been triggered since its creation.
• 6m – Displays the total amount of times this rule has been triggered during the last 6 months.

Option	Description
EDIT	Allows you to edit the rule.
CREATE COPY	Creates a copy of the rule and opens it to be edited.
REMOVE	Remove the rule.

Add rule

General

Option	Description
Enable this rule	Enable or disable this rule. Any disabled rules are still displayed in the rules list but are ignored. This can be used to test configuration changes or temporarily disable a specific rule, e.g., during troubleshooting.
Is fallback	Select if this should be a fallback rule. Fallback rules are always placed last.
Sync back to Pexip	Select if this rule should be synced back to Pexip.
Name	Type the name of the rule.
Service tag	Type unique identifier that can be used to track usage of this rule.
Description	Type a description of the rule.
Priority	The priority of this rule. Rules are checked in ascending priority until the first matching rule is found. Range: 1-200
	Click NEXT to go to the next page, or click CANCEL to cancel.

Match

Option	Description
Match incoming gateway calls	Choose to apply this rule to incoming calls that have not been routed to a conference service within Pexip, and should be routed via Pexip’s distributed gateway service.
Match outgoing calls from a conference	Choose to apply this rule to outgoing calls made from a conference service (e.g. adding a participant to a virtual meeting room).
Match against full alias URI	This is for advanced use cases and will not normally be required. By default, Pexop matches against a parsed version of the destination alias, i.e., it ignores the URI scheme, other parameters, and host IP addresses. So, if the original alias is sip:[email protected];transport=tls by default the rule will match against [email protected]. Select this option to match against the full, unparsed alias instead.
Destination alias regex match	The regular expression that the destination alias(the dialed alias) is checked against to see if the rules apply to this call.
Destination alias regexs replace string	The regular expression string used to transform the originally dialed alias. Leave blank to leave the original dialed alias unchanged.
	Click NEXT to go to the next page, or click CANCEL to cancel.

Media settings

Option	Description
Call capabillity	Choose the media capabilities of the call, between; Audio only, Main video + presentation, Main video only, Same as incoming call. The participant will not be able to escalate beyond the selected capability.
Maximum inbound call bandwidth	This field allows you to limit the bandwidth Pexip receives from each participant dialed via this rule. Range: 128-8192.
Maximum outbound call bandwidth	This field allows you to limit the bandwidth Pexip sends to each participant dialed via this rule. Range: 128-8192.
Maximum call quality	This rule sets the maximum call quality for each participant dialing via this rule.
Media encryption	Controls the media encryption requirements for participants connecting via this rule. Default – Use the global media encryption setting from Pexip. Best effort – Each participant will use media encryption if their device supports it, otherwise the connection will be unencrypted. On – All participants (including RTMP participants) must use media encryption. Off – All H.323, SIP and MS-SIP participants must use unencrypted media. (RTMP participants will use encryption if their device supports it, otherwise the connection will be unencrypted.)
Theme	Select what theme to use for this rule. If no theme is selected here, the theme selected as the default theme will be used.
	Click NEXT to go to the next page, or click CANCEL to cancel.

Outgoing

Option	Description
Call target	Choose how the call should be routed: Registered devices or external systems – Routes the call to a matching registered device if available; otherwise, call via an external system, chosen under Protocol. Registered devices only – Routes the call to a registered device only. Lync / Skype for business meeting direct (conference ID in dialed alias) -Routes the call via a Lync / Skype for Business server to a Lync / Skype for business meeting. Note that the destination alias must be transformed into just a Lync / Skype for Business conference ID. Lync / Skype for Business clients, or meetings via a Virtual Reception – Routes the call via a Lync / Skype for Business server either to a Lync / Skype for Business client or, for calls that have come via a Virtual Reception, to a Lync / Skype for Business meeting. For Lync / Skype for Business meeting via Virtual Reception, ensure that “match against full alias URI” is selected and that the “Destination alias regex match” ends with .*. Google Meet meeting – Routes the call to a Google Meet meeting. Microsoft Teams meeting – Routes the call to a Microsoft Teams meeting.
Outgoing location	Choose a location to use for the outgoing call. Select automatic to allow Pexip to select which conferencing node to use automatically.
Protocol	Select which protocol to use for the outgoing call: h323, mssip, sip, rtmp, gms, teams.

The last few steps differ slightly depending on the protocol you choose. To see more information, select the protocol you chose.

h323

Option	Description
H.323 gatekeeper	Select one of your configured H.323 gatekeepers to send your matching calls to.

mssip

Option	Description
Lync / Skype for Business server	Select one of your configured Lync / Skype for Business server to send your matching calls to.
STUN server	Select one of your existing STUN servers to use.
TURN server	Select one of your existing TURN servers to use.

sip

Option	Description
SIP proxy	Select one of your configured Sip Proxies to send your matching calls to.

rtmp

No more information is needed.

gms

Option	Description
Access token	Select one of your configured Access tokens to use for your matching calls.
STUN server	Select one of your existing STUN servers to use.
TURN server	Select one of your existing TURN servers to use.

teams

Option	Description
Teams Connector	Select one of your configured Teams Connector to use for your matching calls.
External participant avatar lookup	Determines whether or not avatars from external participants will be retrieved using the method appropriate for the external meeting type. You can use it to override the global configuration setting.

Last steps

Treat as trusted	This indicates that the target of this routing rule will treat the caller as part of the target organization for trust purposes.
	Click SAVE to save and create the rule, or click CANCEL to cancel.

Test rules

Here, you can test your rules to see if they behave as you expect, troubleshoot, and more.

New test

Option	Description
Destination alias	Enter the alias you wish to reach. E.g. [email protected].
Source alias address	Enter the source alias to use for the test. E.g. [email protected].
Incoming or Outgoing	Choose if the test is an incoming or outgoing call. If outgoing, no more fields need to be filled in.
Call being handled in location	Choose at which location this test should come in to.
Registered device	Select if it is a registered device.
Source location type	Choose the call type for the source. SIP, WebRTC/RTMP, MS-SIP or H.323.
	Click SEND TEST to run the test or click CANCEL to cancel.

Results

Option	Description
Prio	If matching any rules, this will display the Prio of the matched rule.
Name	If you match any rules, this will display the Name of the matched rule; clicking the name allows you to edit it.