Connecting Bandwidth SIP Trunking to Live Hub

This section describes how to connect Bandwidth (formerly Voxbone) SIP Trunking to Live Hub.

To connect Bandwidth SIP Trunking to Live Hub, complete the following:

  1. Create SIP Connection in Live Hub.

  2. Configure SIP Trunking in Bandwidth account.

Create SIP Connections in Live Hub

The following procedure describes how to create SIP connections in Live Hub, that represents your Bandwidth SIP Trunking account.

  1. Log in to your Live Hub.

  2. From the navigation menu, choose SIP Connections, and then click the Add new SIP connection button to add a new SIP connection.

  3. Click the GENERAL tab:

    1. Under the GENERAL group:

      • In the 'Name' field, enter the unique SIP Connection name (e.g., 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 to be used to authenticate the connection between Bandwidth SIP Trunking and AudioCodes 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-character categories among the following:

        • Uppercase characters (A-Z)

        • Lowercase characters (a-z)

        • Digits (0-9)

        • Special characters (~!@#$%^&*_-+=`|\(){}[]:;"'<>,.?/)

      • Save the generated username and password. You will need them during the Bandwidth SIP Trunking account provisioning.

    3. Under the REGISTRATION AND AUTHENTICATION group:

      • Under Credentials, enter the generated username and password.

  4. Click the INCOMING tab:

    1. Under the AUTHENTICATION group, select the FQDN (Request-URI) check box. Write down (e.g., copy/paste to some text document on your laptop/PC) the displayed SIP Connection FQDN value. You will need this value when configuring the Elastic SIP Trunking in your Bandwidth account.

  5. Click 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”.

  6. Click Create to create a new SIP Connection.

Configure Bandwidth SIP Trunking

The following procedure describes how to configure SIP Trunking in the Bandwidth account.

  1. Log in to the Bandwidth SIP Trunking portal (https://login.voxbone.com) as a user with Admin privileges.

  2. In the navigation panel, choose SETTINGS > VOICE URIS.

  3. Click New to create a new Voice URI.

  4. In the 'PROTOCOL' field, enter “SIP”.

  5. In the 'URI' field, enter the following: "+{E.164}sip:<sip-connection-fqdn>:5061;transport=TLS", where <sip-connection-fqdn> is the Live Hub platform’s SIP Connection FQDN value, as displayed in the SIP Connection > INCOMING tab.

  6. In the 'Description' field, enter a description, e.g., “Live Hub”.

  7. Click SAVE.

  8. In the navigation panel, choose NUMBERS > MY NUMBERS.

  9. Select a number that you want to configure.

  10. Under the Basic group, in the 'Voice URI' field, enter the Voice URI configured above.

  11. 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. Enable the 'OUTBOUND INTERNATIONAL' check box.

    6. In the 'SRTP' field, select Enabled.

      Note: If you do not see the SRTP parameter or cannot change it to Enabled, contact your Bandwidth support and request to activate the SRTP Service for your account.

    7. Click SAVE.

  12. In the navigation panel, choose SETTINGS > OUTBOUND.

  13. Under the Generate Password for Outbound Service group, in the 'Username' and 'Password' fields, enter the same values as configured in the AudioCodes Live Hub platform’s SIP Connection > GENERAL tab > Credentials.

  14. Click SAVE.