Bandwidth SIP Trunking

This guide connects a Bandwidth (formerly Voxbone) SIP trunk to Live Hub. Complete the following steps:

Create a SIP connection in Live Hub.
Configure Bandwidth SIP Trunking.

Create a SIP connection in Live Hub

To create the SIP connection that represents your Bandwidth SIP trunking account:

Log in to Live Hub.
In the Navigation pane, select SIP connections, and then click Add new SIP connection.
Select the General tab:
1. Under the General group:
  - In the 'Name' field, enter a unique SIP connection name (for example, Bandwidth).
  - From the 'Provider type' drop-down list, select SIP Trunk.
  - From the 'Provider' drop-down list, select Bandwidth SIP Trunk.
2. Under the Security group:
  - From the 'Encryption' drop-down list, select Enabled.
  - Generate a username and password that authenticate the connection between Bandwidth SIP Trunking and Live Hub. Use a username that is at least 12 characters long and consists of the following:
    - Uppercase characters (A-Z)
    - Lowercase characters (a-z)
    - Digits (0-9)
    - Special characters (._-)
  - Use a strong password that is at least 12 characters long and contains at least three of the following character categories:
    - Uppercase characters (A-Z)
    - Lowercase characters (a-z)
    - Digits (0-9)
    - Special characters (~!@#$%^&*_-+=`|(){}[]:;"'<>,.?/)
  - Save the generated username and password. You need them when you provision the Bandwidth SIP Trunking account.
3. Under the Registration and authentication group:
  - Under Credentials, enter the generated username and password.
Select the Incoming tab:
1. Under the Authentication group, select the 'FQDN (Request-URI)' check box, and then copy the 'SIP connection FQDN' value that appears. You need this value when you configure SIP trunking in your Bandwidth account.
Select the Outgoing tab:
1. In the 'SIP server hostname' field, enter outbound.voxbone.com.
2. Under the Addresses group, click Add to create a new entry.
  - In the 'ADDRESS' field, enter outbound.voxbone.com.
  - In the 'PORT' field, enter 5061.
  - In the 'PROTOCOL' field, enter TLS.
Click Create.

Configure Bandwidth SIP Trunking

To configure SIP trunking in the Bandwidth account:

Log in to the Bandwidth SIP Trunking portal (https://login.voxbone.com) as a user with Admin privileges.
In the navigation panel, choose SETTINGS > VOICE URIS.
Click New to create a new Voice URI.
In the 'PROTOCOL' field, enter SIP.
In the 'URI' field, enter +{E.164}sip:{sipConnectionFqdn}:5061;transport=TLS, where {sipConnectionFqdn} is the Live Hub 'SIP connection FQDN' value shown on the Incoming tab of the SIP connection.
In the 'Description' field, enter a description, for example Live Hub.
Click SAVE.
In the navigation panel, choose Numbers > MY NUMBERS.
Select the number that you want to configure.
Under the Basic group, in the 'Voice URI' field, enter the Voice URI that you created earlier.
Under the Voice group, do the following:
1. In the 'DTMF' field, enter RFC 2833.
2. In the 'NARROW BAND CODECS' field, enter G.711a, G.711u, and G.729.
3. In the 'CALLER ID' field, enter E.164 CLI.
4. In the 'PREFIX' field, enter +.
5. Select the 'OUTBOUND INTERNATIONAL' check box.
6. In the 'SRTP' field, select Enabled.
  
  If you do not see the SRTP parameter, or cannot change it to Enabled, contact Bandwidth support and ask them to activate the SRTP service for your account.
7. Click SAVE.
In the navigation panel, choose SETTINGS > OUTBOUND.
Under the Generate Password for Outbound Service group, in the 'Username' and 'Password' fields, enter the same values as configured on the Live Hub SIP connection's General tab, under Credentials.
Click SAVE.