ASTPP IP and SIP Based Authenticaiton

ASTPP IP and SIP Based Authenticaiton

ASTPP uses two main types of authentication to verify VoIP traffic:

  1. IP-Based Authentication

  2. SIP Credential-Based Authentication

These methods are used to authenticate incoming traffic from VoIP carriers (origination) or outbound calls from end-users (termination).

1. IP-Based Authentication

Description:

IP-based authentication is commonly used when Customers or carriers send traffic to your ASTPP system. It doesn’t require any username or password. Instead, ASTPP checks the source IP address of the SIP INVITE request and allows or denies access accordingly.

How it Works:

  1. Customer/Carrier sends a SIP INVITE to ASTPP.

  2. ASTPP inspects the source IP.

  3. If the IP matches an allowed IP configured in the ASTPP system in IP settings under any customer or carrier account, the call proceeds.

  4. If not, the call is rejected (typically with 403 Forbidden).

Example:
Customer Name: PBX1
IP Address: 203.0.113.10
Prefix: 91
This configuration allows traffic from IP 203.0.113.10 to ASTPP for all numbers starting with the prefix 91.

Customer Name: PBX12
IP Address: 203.0.113.10
Prefix: 92
This configuration allows traffic from IP 203.0.113.10 to ASTPP for all numbers starting with the prefix 92.

Configuration Steps in ASTPP:

  1. Login to ASTPP Admin Panel

  2. Go to Accounts > Customer > IP settings

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  1. Click Add IP Settings

  2. Fill in customer/carrier name, IP, And prefix

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2. SIP Credential-Based Authentication

Description:

SIP credential-based authentication is a method used primarily for end-users, retail customers who connect to your ASTPP VoIP system using individual SIP accounts. This method requires the user to authenticate using a username, password and domain/IP, similar to how traditional SIP phones or softphones register with a PBX system.

In ASTPP, these credentials are managed under customer accounts, where you can create and assign individual SIP devices. Each SIP device acts like a virtual phone line or extension.

How it Works:

Registering ASTPP SIP Devices

Once you’ve created a SIP device (account) in ASTPP with a username and password, you can use that device to register with any SIP-compatible client—this includes:

  • Softphones (Zoiper, Linphone, X-Lite, etc.)

  • Physical IP phones (Yealink, Cisco, Grandstream, etc.)

  • Third-party PBX systems (like Asterisk, FreeSWITCH, FusionPBX, etc.)

When you register an ASTPP SIP device, it acts like a SIP trunk or extension that can send and receive calls via ASTPP, depending on the configuration.

Registering on a Softphone or SIP Device

To make direct calls from a softphone or IP phone:

  1. Open your softphone app (e.g., Zoiper).

  2. Add a new SIP account using:

    • Username: SIP account username created in ASTPP (e.g., 1001)

    • Password: The password set during account creation

    • Domain / SIP Server: The IP address or domain name of your ASTPP server (e.g., voip.myastpp.com)

    • SIP Port: Usually 5060 (unless otherwise configured)

Registering ASTPP SIP Devices in an External PBX (e.g., any Asterisk or FreeSWITCH based PBX)

You can also register an ASTPP SIP device as a trunk in your own PBX system to send calls from that PBX to ASTPP for routing and billing.

Why do this?

  • You may already have a PBX setup but want to use ASTPP for billing, routing, or LCR (least-cost routing).

  • ASTPP acts as a SIP provider, receiving calls from your PBX and routing them onward.

Important Points for SIP Registration from a PBX to ASTPP

When registering ASTPP SIP devices on another PBX (like Asterisk), make sure to:

1. Configure SIP Trunk Settings Properly

Your PBX should treat ASTPP as an external SIP trunk. In your PBX configuration (e.g., sip.conf for Asterisk), provide:

[astpp-trunk] type=peer host=voip.myastpp.com username=1001 secret=yourpassword fromuser=1001 fromdomain=voip.myastpp.com context=from-internal qualify=yes insecure=invite,port

Once registered, the user can start making outbound calls using ASTPP’s routing and billing engine.

2. Use the Correct Registration String

Most PBX systems require a registration string to authenticate with ASTPP.

register => 1001:yourpassword@voip.myastpp.com/1001

3. Include fromuser and fromdomain

These fields ensure that the SIP INVITE packets sent from your PBX include the correct From and Contact headers. Without this, ASTPP may reject the registration or treat it as unauthorized.

After successful registration, the user can make and receive calls based on their account permissions and available balance.After successful registration, the user can make and receive calls based on their account permissions and available balance.