Connections - Outbound Connections

An outbound connection is a collection of one or more trunks (IP addresses) used to authorize calls from a customer IP PBX or SBC for PSTN termination.  All configuration settings apply to all trunks in the connection.

Key Terminology

Trunks - A trunk is a logical connection between the Peeredge Orchestrator and the customer's IP PBX or SBC. A trunk is uniquely defined by the combination of the tech prefix (if present), the protocol (UDP/TCP/TLS), the customer's IP address and source port (typically 5060 for UDP or 1 to represent any random ephemeral port used by TCP/TLS), the Orchestrator's IP address and destination port (5060 for UDP/TCP and 5061 for TLS, not shown above).

Service Plan - The outbound service plan associates the connection with one or more rate decks (US & Canada Jurisdictional, US & Canada Peering, US & Canada Toll Free, International EEA, US & Canada Local, International Dialed) used to determine the rate (i.e. cost) for each established call.  It also associates a termination route plan to the connection.  The termination route plan contains the routing rules and list of potential vendors for each rule used to route calls to the PSTN.  All connections must be associated with an outbound service plan.  The configuration details of the service plan and the termination route plan are not viewable from the Enterprise Orchestrator portal. 

Ani Normalization - The two supported formats are E.164 or None.

   E.164 -  When ANI Normalization is set to E.164, If a 10 digit number is the user portion of the PAI (if present), RPID (if present) or From headers, then the Peeredge Orchestrator will assume the ANI is a North American Numbering Plan (NANP) number and convert it to an E.164 format (i.e. add the country code 1 to the number).  This will prevent the Orchestrator incorrectly applying an indeterminant jurisdiction to the call, when the DNIS is also a NANP number.  Indeterminant calls are often billed at a higher rate. 

   None - The customer IP PBX or SBC is expected to send the ANI formatted as an E.164 number in the PAI (if present), RPID (if present) and From headers. (with or without the leading + symbol). 

Tech Prefix - Used to guarantee uniqueness between trunk groups.

Maximum Call Length - When configured with a number in seconds, this setting determines how long an established call can exist before the Orchestrator terminates the call.

No ring back timeout -  This timer measures the amount of time that passes before a final response is received for a SIP Invite. The timer can be configured between 1 and 30 seconds.  If the timer is not configured a value of 30 seconds will always be used. the purpose of this timer is to route advance the call to the next available vendor when the vendor does not provide a final response (2XX - 6XX) before the timer expires.  

Concurrent call limit (PORTS) - When configured with a number, this setting determines how many calls (established and calls in progress) can exist at the same time. When configured, this setting takes precedence over any associated capacity group's port limit setting.  Note:  Although not shown in the Enterprise Orchestrator portal, connections are often associated with capacity groups to set port and CPS limits that can be shared across all the customer's connections.

Calls per second limit (CPS) - When configured with a number, this setting determines how many

  simultaneous call attempts can be processed each second. When configured, this setting takes precedence

  over any associated CPS setting in an associated capacity group.  

Nat traversal -  When enabled, if the source IP address of SIP messages is different then the IP address in the Request URI of the SIP message then the Orchestrator will use the source IP address for all SIP responses.  This situation can occur when the customer IP PBX or SBC is behind a NAT/Firewall that does not properly support SIP Inspection. When disabled, IP address/FQDNs for the SIP messages will be used.

Enable diversion rating - When present in a SIP Invite, the diversion header typically represents the original called party's number before the call was redirected or forwarded.  When enabled, the number in the diversion header will be as the ANI for call rating purposes.  

Secure RTP - This option is typically used with TLS to encrypt the RTP media between the Peeredge Orchestrator and the customer's media endpoint.

Dynamic transcoding -  When enabled, the Peeredge Orchestrator will dynamically transcode the RTP media between difference codecs when there is no mutually acceptable codec in the SDP offer and answer. This configuration is rarely used as all PSTN vendors support G.711u (PCMU) and G.711a (PCMA) and nearly all customer IP PBXs and SBCs support these codecs as well.

Codec limit - When enabled and dynamic transcoding is disabled, the Peeredge Orchestrator will filter (only include) codecs in SIP/SDP messages to the customer's IP PBX or SBC that are 1) selected in the Codec limit configuration and 2) listed in the SIP/SDP messages from the PSTN vendor. When enabled and dynamic transcoding is enabled, all codecs selected in the Codec limit configuration are included in the SIP/SDP messages to the customer's IP PBX or SBC regardless of the codecs listed in the SIP/SDP messages from the PSTN vendor

Performance Notifications - See Triggers.

Expert Settings - This feature allow the Orchestrator to apply number manipulations to calling party numbers (ANIs), called party numbers (DNISs) or change SIP response codes.  For example, if a customer's IP PBX only sends 10 digit numbers in the From field, a manipulation could be added to converted the ANI to an 11 digit number.

Route Advance Logic -  If a SIP response (failure response codes only) to a SIP Invite from the PSTN vendor is included the Route Advance Logic's regular expression, then the Orchestrator will route advance the call by sending a SIP Invite to the next available trunk in the inbound connection.  If no more trunks are available, then the call attempt will be terminated.