Introduction

If you have a telephone exchange in your company and need to route inbound calls to the wolkvox Contact Center platform, it is necessary to perform a technical integration using a SIP trunk.

This integration allows calls arriving at your telephone infrastructure to be managed by wolkvox, where they can be distributed to agents, automated via IVR, or integrated with other service flows.

To achieve a successful integration, it is important to understand the inbound connection topology, the required ports and protocols, and follow a configuration procedure that ensures stable communication between the client's infrastructure and wolkvox's SIP servers.

Configuration

1) Understand the Inbound Integration Topology

The integration between your telephone exchange and wolkvox follows a SIP-based connection structure involving the following components:

Wolkvox Side

The following elements are involved in the wolkvox infrastructure:

• SIP Servers: Responsible for managing call signaling and processing.
• Virtual Private Cloud (VPC): Private environment that isolates and protects the infrastructure.
• Cloud Firewall Rules: Control incoming and outgoing traffic to ensure security.
• Cloud Router: Enables connectivity between the wolkvox infrastructure and the public network.

Internet Connection

All communication between the client's infrastructure and wolkvox is carried out over the internet, using the defined protocols and ports for signaling and audio transmission.

Client Side

The following are involved in the client's infrastructure:

• Firewall: Responsible for controlling access to the corporate network.
• Server or PBX (telephone exchange): From where calls are originated or routed to wolkvox.

The agent finally attends the call from the wolkvox interface.

2) Verify the Technical Integration Requirements

Before starting the configuration, it is important to validate the following technical requirements.

Communication Protocol

Integration with wolkvox is done exclusively via:

• SIP (Session Initiation Protocol)

Required Ports

For communication to work correctly, the following ports must be enabled on the firewall.

TCP

• 80 (HTTP)
• 443 (HTTPS)

UDP

• 5060 – 5061 (SIP signaling)
• 10000 – 20000 (RTP voice traffic)

Supported Codecs

wolkvox supports the following audio codecs:

• GSM
• G711
• G729
• G723

It is recommended to use GSM for better performance and stability.

3) Enable Ports on the Firewall

Configure your infrastructure's firewall to allow communication with wolkvox servers. You must enable:

TCP

• 80 (HTTP)
• 443 (HTTPS)

UDP

• 5060 – 5061 (SIP signaling)
• 10000 – 20000 (RTP traffic)

This configuration is essential to allow both call signaling and audio transmission.

4) Provide the Public IP of the Infrastructure

You must provide wolkvox with the public IP from which call traffic will be generated.

Once this information is received:

• wolkvox will configure access from its servers.
• wolkvox will provide you with the public IP of the SIP server to be configured in your telephone exchange or firewall.

It is important that the previously defined ports are enabled for this IP.

5) Configure the SIP Trunk Credentials

The wolkvox implementation team will provide the SIP trunk authentication credentials needed to complete the integration. This configuration should be validated with the support of the technical team.

For initial testing, it is recommended to use a softphone, such as:

• Zoiper
• X-Lite

Preferably, the test should be performed from a PC to correctly validate the connection.

6) Perform Connectivity and Audio Tests

Before performing the production configuration, tests must be executed to validate the correct operation of the trunk. During these tests, you should:

• Verify that calls are established correctly.
• Validate two-way RTP audio.
• Keep a call active for at least 10 minutes to check stability.

These tests allow confirming that the configuration is correct before starting real operation.

7) Validate the Available Bandwidth

Each simultaneous call requires approximately: 150 kbps of symmetrical bandwidth.

Therefore, you must ensure that the network infrastructure has sufficient capacity to support the expected call volume.

8) Apply Best Configuration Practices

To ensure service stability, it is recommended to apply the following best practices:

• Avoid channel balancing on the configured trunk.
• Segment the telephony network to avoid interference with other services.
• Configure QoS (Quality of Service) to prioritize voice traffic.
• Disable SIP-ALG or SIP-Helper on the firewall, as they can cause problems such as:
  • SIP registration failures.
  • NAT issues.
  • One-way audio.
  • Call drops.

Specific Case for Fortinet

For Fortinet equipment, it is recommended to:

• Change SIP-ALG to kernel mode
• Disable SIP-Helper

More information by [clicking here].