Table of Contents

 

 

 



Disclaimers and Restrictions
This Customer Configuration Guide (CCG) is offered as a convenience to AT&T's customers for informational purposes only. The specifications and information regarding the product in this CCG are subject to change without notice. While reasonable efforts have been made in the preparation of this publication, Sonus Networks and its suppliers assume no liability resulting from technical or editorial errors or omissions, or for any damages resulting from the use of this information. All statements, information, and recommendations in this CCG are presented without warranty of any kind, express or implied, and are provided as is.

In no event shall Sonus be liable for any indirect, special, consequential, or incidental damages. These damages include, without limitation, lost profits or loss or damage arising out of the use or inability to use this CCG, even if the user was advised of the possibility of such damage. Unless specifically included in a written agreement with Sonus Networks, Sonus Networks has no obligation to develop or deliver any future release, upgrade, feature, enhancement, or function.

Introduction

This document provides a configuration guide for the Sonus SBC 2000 Series (Session Border Controller) when connecting to AT&T IP Toll Free service using AVPN or MIS/PNT transport.

The Sonus SBC 2000 is a Session Border Controller that connects disparate SIP trunks, SIP PBXs, and communication applications within an enterprise. The SBC 2000 can also be used as a SIP routing and integration engine. The SBC is the point of connection between the ShoreTel Connect and AT&T IP Toll Free service, and not only secures the SIP trunk, but also makes adjustments to carrier and enterprise signaling for interoperability.

Note

This guide supports the SBC 2000 Series configurations for releases V06.X or higher.

 

Special Notes

Mid-Call Re-INVITEs

The Sonus SBC 2000 does not send out the shuffling Re-INVITES and Hold-Resume Re-INVITES from the ShoreTel Connect to the Network unless there are changes in the media information.
The Re-INVITES received from the ShoreTel Connect that have changes only in the media source information are not relayed to the Network. However, changes in the media codecs, p-time, or media attributes are relayed to the Network.

Alternate Destination Routing on Ring-No-Answer

Alternate Destination Routing on Ring-No-Answer (ADR/RNA) could not be tested in a lab environment, but will be supported in field deployments.

ACK for Initial INVITE's 200-OK

The Sonus SBC 2000 handles acknowledgement for the initial INVITE's 200-OK on a hop-by-hop basis. If no ACK is received from the next hop, the 200-OK is retransmitted according to the SIP timers defined in RFC 3261. The communication legs also terminate gracefully after the retransmissions timer expires using BYE.

ShoreTel Support for Ptime

The ShoreTel system initiates all calls with a 20ms payload. The ShoreTel system does not initiate calls with a 30ms payload. This configuration has been tested with a 20ms payload.

IP Transfer Connect

The configuration was not certified to support IP Transfer Connect and the customer should not attempt to configure the IP endpoint for this capability.

Highest Priority Codec Negotiation

The SBC 2000 does not support negotiating the highest priority in the codec negotiations.

Fax Testing

Fax test cases are currently BLOCKED since there is no test support at this time.

Network Topology

The preceding figure illustrates the equipment used for the IP Toll Free certification. See Hardware/Software Configuration for the hardware and software version details used for the setup in the preceding figure.

Hardware/Software Configuration

Equipment

Software

Sonus SBC 2000 Series

Software Version 6.1.1

 

Build Number 463

Third-party Equipments

 

ShoreTel

Connect


SBC 2000 Series Configuration

This section describes how to use the GUI to configure and manage the SBC 2000 Series. This guide lists the configuration.

Configuration Diagram


Naming Conventions

Entity

Customer Side

ATT Side

Call Routing Table

Shoretel_RL

ATT_RL

Signalling Group

(SIP)Shoretel_SG

(SIP)ATT_SG

Sip Server Table

Shoretel

ATT

Sip Profile

Default

Default

Media Profile

Shoretel

ATT

Media List

SHTL_ML

ATT_Media list

 

SBC Initial Task Configuration

Use the SBC's integrated web server to configure the SBC 2000. This guide assumes that the operator has already done the initial configuration to position the SBC on the IP network. To start the configuration process, use a standard web browser to connect to the IP or FQDN address of the SBC. Supply the username and password to complete the login process.


The Initial Task configuration is a process that completes the steps to position the SBC between AT&T's SIP Trunk and ShoreTel. This task creates SIP components and call routing basics.

Customer Side Configuration


Create profiles with a specific set of characteristics that correspond to the customer IP-PBX, which includes configuring the following entities on the customer side.

  1. Media Profile
  2. Media List
  3. SIP Profile
  4. SIP Server Table
  5. Signaling Group
  6. Call Routing Table

Media Profile

Create a Media Profile towards the Interactive Intelligence side with G729 as the first codec, G711ulaw as the second codec, and Fax as the third codec.

To create or modify a Media Profile:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select Media > Media Profile.








Media List

The Media List allows you to specify a set of codecs and fax profiles that are allowed on a given SIP Signaling Group. They contain one or more Media Profiles, which you must first define in a Media List. These lists allow you to accommodate specific transmission requirements and SIP devices that only implement a subset of the available voice codecs.

To create or modify a Media List:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select Media > Media List.


SIP Profile

To create or modify an existing SIP Profile:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select SIP > SIP Profiles.



SIP Server Table

SIP Server Tables contain information about the SIP devices connected to the SBC 2000. The entries in the tables provide information about the IP Addresses, ports, and transport protocols used to communicate with each server. The Table Entries also contain links to counters that are useful for troubleshooting.

To create or modify an existing SIP Server Table:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select SIP > SIP Server Tables.


Signaling Group

Signaling groups allow telephony channels to be grouped together for the purposes of routing and shared configuration. Signaling groups are the entity to which calls are routed, as well as the location from which Call Routes are selected. They are also the location from which Tone Tables and Action Sets are selected. In the case of SIP, signaling groups specify the protocol settings and link to the server, media, and mapping tables.

To create or modify an existing Signaling Group:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, click Signaling Groups.
  3. From the Create Signaling Group drop-down box, select SIP Signaling Group.







IP Address/FQDN: Specifies the IP Address or Fully Qualified Domain Name of a server from which the SBC 1000/2000 will accept SIP messages. The IP Address/FQDN should have the IP of the Interactive Intelligence PBX.

Call Routing Table

Call Routing allows calls to be carried between signaling groups, which allows calls to be carried between ports, and between protocols (like ISDN to SIP). Routes are defined by Call Routing Tables, which allow for flexible configuration of which calls are carried and how they are translated. These tables are one of the central connection points of the system, linking Transformation Tables, Message Translations, Cause Code Reroute Tables, Media Lists, and the three types of Signaling Groups (ISDN, SIP, and CAS).

To create or modify an existing Call Routing Table:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, click Call Routing Table.



Destination Signaling Group: Specifies the Signaling Groups used as the destination of calls, so this field should point toward the ATT Signaling Group.
Media Mode: Media Mode should be RTP DSP.
Media Transcoding: Enable this field so that the SBC 2000 can transcode.
Media List: Specifies the Media List used for this call route.

ATT Side Configuration

Create profiles with a specific set of characteristics that correspond to the ATT network, which includes configuring the following entities on the ATT side.

  1. Media Profile
  2. Media List
  3. SIP Profile
  4. SIP Server Table

  5. SIP Signaling Group

Media Profile

Create a Media Profile towards the ATT side with G729 as the first codec, G711ulaw as the second codec, and Fax as the third codec.

To create or modify a Media Profile:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select Media > Media Profile.





Media List

To create or modify a Media List for ATT:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select Media > Media Lists.






Media Profiles List: Select the media profiles created for ATT in the Media profile list.

SIP Profile

To create or modify an existing SIP Profile for ATT:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, select SIP > SIP Profiles.


SIP Server Table

Add the ATT Toll Free IP adress in the SIP Server Table.

SIP Signaling Group

To create or modify an existing Signaling Group:

  1. In the WebUI, click the Settings tab.
  2. In the left navigation pane, click Signaling Groups.
  3. From the Create Signaling Group drop-down box, select SIP Signaling Group.



ShoreTel Configuration

The ShoreTel Connect configuration is used with the Sonus SBC 2000 Series according to the details provided in the ShoreTel Admin Guide: AT&T IPFR-EF Service on MIS, MPLS PNT or AT&T VPN with ShoreTel Connect. This guide is available online.

The following are differences in the configuration when compared to the ShoreTel guide available online.

  • Set the DTMF Payload type to 100 to match with AT&T. The default on ShoreTel is 102.
    Administration > Features > Call Control >Options
  • Set the following configuration for File based Music on Hold
    Administration > Features >Music on Hold > Files - Select the file to play during a call hold.


    Administration > CallControl > Music on Hold ->System Defaults – Enable File based MOH

    Enable the file based Music on Hold on the main switch.
    Administration -> Application Server ->Platform Equipment -> Softswitch


Troubleshooting

Call traces can be gathered on an individual call by call basis. This is enabled by creating a Call Trace Filter on the SBC 2000 Series. When the filter is defined, a calling and called number can be entered to capture a single call. This data is viewed in the EMS with the Call Trace screen in the Tools major screen.

For troubleshooting help, call the ShoreTel's Technical Assistance Center at +1 (800) 742-2348 (Toll Free) or +1 (408) 331-3313 (International). This Customer Configuration Guide (CCG) is offered as a convenience to AT&T's customers. The specifications and information regarding the product in this CCG are subject to change without notice. All statements, information, and recommendations in this CCG are believed to be accurate but are presented without warranty of any kind, express or implied, and are provided as is. Users must take full responsibility for the application of the specifications and information in this CCG.

In no event shall AT&T or its suppliers be liable for any indirect, special, consequential, or incidental damages. These damages include, without limitation, lost profits or loss or damage arising out of the use or inability to use this CCG, even if AT&T or its suppliers were advised of the possibility of such damage.