SIP Trunking Between SIP Border Elements

This article explains the configuration steps necessary to allow SIP calls in an IP.PBX(1) – Sonus SBC 1000/2000 – IP.PBX(2) topology as illustrated in the diagram below. For the purpose of this article, it is assumed that the IP.PBX - Border Elements - are configured to send/receive SIP calls:

Sonus SBC 1000/2000 Configuration

Configuration Overview

The following configuration steps must be accomplished in the order presented as shown below.

1. Tone Table — You may use the default Tone Table or create a custom table.
2. Media Profiles — In this configuration, G.711A/Mu-law profiles are configured and used.
3. Media List — You may use the default Media List or create a custom list.
4. SIP Server Tables — Add an entry for border elements (i.e., the IP_PBX's used in this example).
5. SIP Profile — You may use the default SIP Profile or create a custom profile.
6. Transformation Table — Create a transformation table to be used in the relevant Call Routing Table(s) for Called/Calling Number transformations.
7. Call Routing Table — Create two Call Routing Tables (one for ISDN and another for SIP) to be used in the relevant Signaling Groups.
8. Signaling Groups — Create two SIP Signaling Groups (one for each border element).
9. Call Routing Table Entries — Create two Call Routing Table Entries to route the calls.

Tone Table

1. In the WebUI, click the Settings tab.
2. In the left navigation pane, go to Tone Tables.

3. Add a Tone Profile Table (SIPtrunking: Tone Table) as shown below.

   Add Tone Profile Table

   

Media Profiles

1. In the WebUI, click the Settings tab.
2. In the left navigation pane, go to Media > Media Profiles.
3. Add two new Voice Codec Profiles, as shown below.
   1. Name them SIPtrunking: G.711 A-Law Voice and SIPtrunking: G.711 Mu-Law Voice.
   2. Select the G.711 A-Law Codec.

Add New Voice Codec Profiles

 

Media List

1. In the WebUI, click the Settings tab.
2. In the left navigation pane, go to Media > Media List.
3. Add a new Media List.
   1. Specify the Media Profiles created in the previous section in the Media Profiles List text box.

Note that in this example, the other variants in the Media List entry are left with their default settings (i.e: Crypto as NONE, Dead Call Detection as Disabled, Silence Suppression as Enabled,...etc.)

Add New Media List

SIP Server Tables

1. In the WebUI, click the Settings tab.
2. In the left navigation pane, go to SIP > SIP Server Tables.
3. Add two new SIP Server Table entries(one for each border element).
   1. Name them SIPtrunking: IP-PBX-1 and SIPtrunking: IP-PBX-2.

4. Configure the new entries from the previous step as shown below.

   Add New SIP Server Table Entries

   

SIP Profile

1. In the WebUI, click the Settings tab.
2. In the left navigation pane, go to SIP > SIP Profiles.
3. Add a new SIP Profile as shown below.

4. Name it SIPtrunking: SIP Profile.

   SIP Profile Table

   

Transformation Table

1. In the WebUI, click the Settings tab.
2. In the left navigation pane, go to Transformation.
3. Add two new entries and name them SIPtrunking: Translation for IP-PBX-1 ext. and SIPtrunking: Translation for IP-PBX-2 ext..

4. Configure the entries created in the previous step as shown below.

   Transformation Table

   

Call Routing Table

1. In the WebUI, click the Settings tab.
2. In the left navigation pane, go to Call Routing Table.
3. Add two new Call Routing Tables, and name them SIPtrunking: to IP-PBX-1 and SIPtrunking: to IP-PBX-2.

Signaling Groups

1. In the WebUI, click the Settings tab.
2. In the left navigation pane, go to Signaling Groups
3. Add two new SIP Signaling Groups, and configure them as show below.

Call Routing Table Entries

1. In the WebUI, click the Settings tab.
2. In the left navigation pane, go to Call Routing Table > (relevant entry).
3. Configure each of Call Routing entries that you created earlier in this article as shown below.

Call verification

Once the above configuration steps have been completed, a SIP call from Border Element 1 (IP-PBX-1) to Border Element 2 (IP-PBX-2) using Sonus SBC 1000/2000 SIP Trunking will have the following call flow:

Call Flow