Managing SIP Peers (locations)



SIP Peer Management enables customers to configure and manage IP addresses and other attributes for their network connections to Bandwidth. Per section 9, the SIP Peer(s) are set up on the Site to manage the numbers assigned to the account. In the User Interface, the SIP Peer is referenced as a Location instead of a SIP Peer just as a Site is referenced as a Sub Account.

A SIP Peer is a gateway that sends or receives voice and data traffic to or from Bandwidth’s network. Customer must configure a default SIP Peer on the Site within an Account. Before you can create a SIP Peer, a Site (location) must be created. You can have multiple SIP Peers on a Site. Only one default SIP Peer is allowed per Site.

Aside from being an administration point for IP addresses for network ingress and egress, SIP Peers are also used to associate Telephone Numbers with a Site/Location and Account. For a Site to order any Telephone Numbers, at least one SIP Peer must be created for the Site, and one of the SIP Peers needs to be designated as the “Default” SIP Peer with which to associate numbers..

The API endpoint /accounts /{accountId} /sites /{siteId} /sippeers can be used to create SIP Peers, and /accounts /{accountId} /sites /{siteId} /sippeers /{sippeerId} can be used to update and delete SIP Peers.

The API documentation along with sample request parameters can be found here

Voice/Data Traffic

There are three different types of customer calls and messages to and from Bandwidth:

  1. Origination - Voice calls from PSTN phone numbers that are delivered to customer-owned telephone numbers.
  2. Termination - Voice calls from customer-owned telephone numbers that are delivered to PSTN or other on-net telephone numbers.
  3. SMS - Inbound text messages that are delivered to customer-owned telephone numbers and outbound text messages that are delivered to PSTN or other on-net telephone numbers.

These services can be used independently. The following sections outline how SIP Peers should be configured when different combinations of these services are used.

SIP Peer configuration for use of Origination service

Each SIP Peer can be configured with multiple IP addresses that will be used in managing traffic presentation to the Customer.

The traffic can be presented to the customer IP endpoints using one of two models – either as a Random distribution across the available voice host IP endpoints, or using a sequential failover model where traffic is preferentially presented to the first IP address until an endpoint failure is detected.

The VoiceHostGroups field is used to define randomly selected IP addresses, and the VoiceHosts field is used to define sequential IP addresses. Examples of each are shown below:





SIP Peer configuration for use of Origination and SMS services

This section provides information on defining a SIP Peer for receiving inbound calls (from Bandwidth to customer) and text messages. If there is a different host for origination and SMS, that fact can be identified in the tags. Multiple VoiceHosts are allowed as required. The TerminationHosts field is identified with the DataAllowed element to permit SMS Data to flow in to the Bandwidth network. The optional Port element is permitted if required.

The API endpoint /accounts /{accountId} /sites /{siteId} /sippeers /{sippeerId} /products /messaging/features/sms can be used to set SMS properties.

SIP Peer configuration for use of Termination service

To define a SIP Peer for sending outbound calls (from customer to Bandwidth), the configuration includes parameters called TerminationHosts, specifiying the IP address from which IP voice packets will be presented to the Bandwidth network.

Note: In spite of the TerminationHosts data, this configuration can be blocked from Termination traffic with account level constraints.

An example of TerminationHosts is shown below:


SIP Peer configuration for use of Origination, Termination and SMS services

A SIP Peer can also be configured for all services, namely the ability to send and receive both voice and text messages.

SIP Peer settings for Calling Name and other SIP Peer wide features

A number of features can be established at the SIP Peer level, including Calling Name Display and Final Destination URI.

Calling Name display

Enabling Calling Name display (CNAM) for all TNs subtending a SIP Peer is done by configuring the CallingName feature for the SIP Peer. This involves setting the CNAM Display default behavior on or off (true or false) and then declaring whether that default behavior needs to be enforced or can be overridden at the TN level.

Final Destination URI

The network configuration of a SIP Peer includes an optional attribute called finaldestinationuri. This is used when network connectivity to all previously provided hosts/IPs fails. Request format for the configuration of the SIP Peer, using this attribute, is:

Deleting a SIP Peer

A SIP peer can be deleted using the DELETE method on /accounts /{accountId} /sites /{siteId} /sippeers /{sippeerId}.

Note: A SIP peer can be deleted only if there are no TNs.

Moving TNs between SIP Peers

A POST to the SIP Peer resource can be used to move Telephone Numbers between SIP Peers.

The API endpoint for moving Telephone Numbers is/accounts /{accountId} /sites /{siteId} /sippeers /{sippeerId} /movetns.

Phone numbers are moved from their current SIP Peer to the SIP Peer in the URL.

Sample XML Request


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