SIP

SIP Integration Guide

What is SIP?

SIP, or Session Initiation Protocol, is a modern protocol for initiating, managing, and terminating voice and video sessions over the Internet. It offers flexibility, scalability, and the capability to transmit multimedia content. Parloa uses SIP to efficiently connect with customer telephony infrastructures, providing enhanced communication capabilities.

What is PSTN?

The Public Switched Telephone Network (PSTN) is the traditional network for voice communication, consisting of telephone lines, cellular networks, and international cables. Although reliable, PSTN lacks the flexibility and multimedia support that SIP technology provides.

SIP vs. PSTN: The Key Differences

  • SIP is preferred for its flexibility, multimedia support, and integration capabilities, making it ideal for modern businesses.

  • PSTN is reliable for voice communications but is less scalable and lacks the multimedia capabilities of SIP.

Parloa favors SIP for its efficiency, cost-effectiveness, and advanced features like direct call routing and data linkage capabilities.

SIP Integration Methods

Integrating with Your Phone System

  • Internet using SIP Connection – This is the preferred method for its direct, encrypted communication, reducing both costs and complexity.

  • Public Switched Telephony Network (PSTN) – Used when SIP is not feasible, involving third-party services for call routing.

SIP Methods in Use

The following describes the SIP methods commonly used with Parloa: SIP INVITE and SIP REFER.

  • Purpose – To initiate or modify SIP sessions.

  • Functionality – Starts a voice interaction or adds participants to a call.

  • Call Flow –

    1. Your PBX initiates the process by sending an INVITE to Parloa.

    2. Parloa responds with provisional responses, leading to a final response (such as 200 OK) to accept the call.

    3. The PBX sends an ACK request to Parloa, acknowledging the final response, at which point the bot begins speaking.

SIP Integration Requirements

Before initiating SIP integration, ensure your system:

SIP Integration Steps

Step 1 – Establish SIP Trunk

1 – Register Your Telephony System with Parloa

Parloa will provide you with a specific URI (Uniform Resource Identifier), such as customer.voip.parloa.com, to connect your system with Parloa's SIP server.

2 – Configure DNS for FQDN Resolution

Set up your telephony system to resolve the FQDN (Fully Qualified Domain Name) provided by Parloa. This step is essential for your system to locate Parloa's SIP server over the Internet. In this case, customer.voip.parloa.com is the FQDN.

3 – Configure Port Numbers

For unencrypted communication, use UDP port 5060.

Additionally, it's essential to ensure that you have whitelisted our IPs on your network and opened the necessary ports. For more details on our public IPs, please refer to our Public IPs page.

4 – Whitelist Parloa's IPs

You must whitelist Parloa's IPs on your network and open the necessary ports.

To whitelist IPs –

  1. In the Parloa app, click Deployments -> Releases -> Release Settings:

5 – Optional – Configure Authentication

Step 2 – Forward Calls to Parloa

1 – Identify the Call Forwarding Feature

Locate the call forwarding settings in your telephony system or PBX (Private Branch Exchange).

2 – Set Up Forwarding Rules

Specify conditions under which calls should be forwarded to Parloa, such as all incoming calls or calls during non-business hours.

3 – Test Call Forwarding

Perform a test call to verify that calls are being correctly routed to Parloa.

Step 3 – Retrieve Calls from Parloa

1 – Configure Your System for REFER Requests

Enable it to accept and process SIP REFER requests. This allows for effective call redirection as per the REFER method's workflow.

2 – Prepare for SIP REFER Method

Configure your system to receive and recognize SIP URIs from Parloa. This step is essential for call redirection and sets the foundation for handling REFER requests.

3 – Test Call Reception and Routing
  1. Work with Parloa to perform test calls that include transferring calls back to your system using the REFER method.

  2. During testing, ensure that calls are being correctly routed to the intended destination within your system.

4 – Optional – Integrate Custom SIP Headers

If needed, configure your system to recognize and use custom SIP headers, which Parloa can add for advanced routing decisions and data exchange.

Additional Considerations

PSTN Integration

Parloa offers PSTN integration by providing a dedicated phone number for call forwarding. Configure your PBX (Private Branch Exchange) or telephony system to forward and receive calls. Ensure to provide a return number for callbacks.

For further information or assistance, please contact your Parloa representative.

Encryption and Security

Parloa offers the option to uphold the highest security standards by enabling encryption of all SIP data using TLS (Transport Layer Security) and securing voice streams with sRTP (Secure Real-Time Transport Protocol) upon request. To utilize these security features and ensure compliance with European data protection laws, please consult with your Parloa representative or reach out to our support team for the necessary certificates.

Security and Privacy Standards

Parloa adheres to internationally recognized security and privacy standards, ensuring that our SIP solutions meet the stringent requirements of various regulatory bodies.

Backend Integration

Parloa offers a generic HTTPS web service interface for seamless integration with external data sources and APIs. This integration is facilitated using the Service block.

Regional Access and Compliance

Parloa's services are currently available in Germany and Austria, with ongoing plans for further expansion. Understanding local telecommunications regulations is essential for compliance.

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