SIP PSTN Trunk Lines

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Solution Integration > PBXs > BT Call Server > Example Configurations

Introduction

 

BTTrade provides an Audio Codes SBC to act as a security boundary between SIP PW IP carriers and the customer network where customers do not do this themselves.

 

In recent times as customers move to new enterprise voice platforms like Teams and Zoom  BT is providing the BT Call Server to take the place of the customer PBX that is no longer present in the architecture so that BTTrade turrets can still function. The BT Call Server sends a SIP trunk to a customer SBC that connects to PSTN SIP trunks from carriers. As with the SIP PW we are now finding a number of customers that want BT to provide the SBC layer for both SIP PW and PSTN as they lack the skills to set these up and maintain them.

 

This help topic aims to define the setup needed to allow the Audio Codes SBC to terminate both PW SIP carrier and PSTN SIP carriers on the same platform and to route the SIP PW to the TPO while sending the PSTN traffic to the BT Call Server.

 

Overview

SIPPSTNLinesOverview

Assumptions

•The reader knows how to set up DDI SIP Trunks on the BT Call Server if not see BT Call Server Trunk Definition

•The reader knows how to set up AudioCodes SBC for use with SIP PW if not see AudioCodes SBC Setup and Configuration

•The PSTN SIP Carrier and the SIP PW Carrier are presented in the same DMZ or both are routable to the outside interface of the SBC.

If this is not the case an extra interface may be needed on the SBC. While this is possible it is not covered in this example.

 

BT Call Server SIP Trunk Details

The BT Call Server is flexible but by default the SIP trunk will be as follows: -

 

Trunk Parameter

Value

SIP Signalling

Yes

SIP Port

TCP 5060

SIP Registration Required

No

SIP Digest Authentication Required

No

SIP Early Offer

Yes

SIP Messages Supported

INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

SIP OPTIONS

Sends every 60sec

SIP Session Timers

30 mins

RTP port range

10000-20000

RTP Codec

G711alaw + ulaw at 20ms, Opus, ILBC, G722

RTP DTMF

telephone-events 101

RTP Timeout

Not set in Asterisk but 5 seconds normally on TPO

 

The information in this table can be used to configure the inside interface of the SBC that will face the BT Call Server

PSTN Carrier SIP Trunk Details

The PSTN Carrier should issue a similar table outlining the configuration they require.

 

Trunk Parameter

Value

SIP Signalling

 

SIP Port

 

SIP Registration Required

If Yes needs testing

SIP Digest Authentication Required

If Yes needs testing

SIP Early Offer

 

SIP Messages Supported

 

SIP OPTIONS

 

SIP Session Timers

 

RTP port range

 

RTP Codec

 

RTP DTMF

 

RTP Timeout

BT timers should be longer than carrier

 

This information can be used to configure the outside interface of the SBC that will face the carrier.