Troubleshooting PBX Router Issues: Fixing RTP Call Failures

Yaswanth RYaswanth R
2 min read

Setting up a PBX system in an office environment can be challenging, especially when dealing with network configurations that impact VoIP calls. Recently, I encountered an issue where incoming calls were not being received over the RTP (Real-time Transport Protocol) in our office PBX setup. After extensive troubleshooting, I identified and resolved the problem by adjusting router settings. Here’s a breakdown of the issue and how I fixed it.

The Problem

After configuring the PBX system and setting up SIP trunking, I realized that while outbound calls worked fine, inbound calls were failing. The phone would ring, but there was no audio, or the call would drop immediately.

Upon further inspection, I identified that RTP packets were not passing through, causing one-way or no audio issues. Since RTP handles media transmission in VoIP calls, blocking these packets prevented proper communication.

The Fix

After diagnosing the problem, I made the following changes to my router’s settings to ensure RTP traffic could pass through without interference:

1. Allowing RTP Ports (10000-20000)

RTP uses a wide range of ports (typically 10000-20000) for media transmission. By default, many routers block these ports for security reasons. I manually configured the firewall rules to allow incoming and outgoing traffic on these ports:

  • Opened ports 10000-20000 for both UDP and TCP

  • Enabled port forwarding for these RTP ports to direct traffic to the PBX server

2. Disabling SPI (Stateful Packet Inspection) Firewall

*SPI firewalls analyze each packet’s state, which sometimes interferes with VoIP traffic, leading to RTP issues. To resolve this, I disabled SPI on the router’s firewall settings.

  • 3. Disabling SIP ALG (Application Layer Gateway)

*Some routers have SIP ALG enabled by default to assist in VoIP traffic routing. However, in many cases, SIP ALG modifies SIP packets incorrectly, causing disruptions in call processing. Disabling SIP ALG helped establish proper SIP signaling and RTP media flow.

The Outcome

After applying these changes:

  • Incoming calls started working correctly.

  • RTP packets were successfully transmitted, resolving the no-audio issue.

  • The PBX system operated smoothly without any further disruptions.

Final Thoughts

Networking issues with PBX setups can be frustrating, especially when dealing with RTP and SIP configurations. If you’re experiencing similar problems:

  • Check your router’s firewall rules.

  • Ensure RTP ports (10000-20000) are open.

  • Disable SPI firewall and SIP ALG if needed.

  • Verify your PBX SIP settings.

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Written by

Yaswanth R
Yaswanth R

As a final-year Computer Science and Engineering student at Sri Eshwar College of Engineering, I am deeply passionate about technology and innovation. My academic journey has been marked by consistent performance, achieving a CGPA of 8.1, and by actively participating in hackathons and expos, including securing 1st place in the MiniProject Expo Hackathon. Beyond academics, I have a strong interest in cloud computing, DevOps methodologies, and automation tools. I have honed my skills through internships and real-world projects, including my role as a DevOps Intern at Mahat Labs Pvt. Ltd. During this experience, I automated CI/CD pipelines for .NET and Flutter applications using GitLab CI/CD, containerized microservices with Docker, and deployed them to Kubernetes clusters built with kubeadm. I also integrated SonarQube into CI/CD workflows for code quality analysis and developed custom monitoring scripts to enhance infrastructure reliability. My hands-on experience extends to tools like Jenkins, Proxmox, and AWS, where I have successfully modernized deployment workflows and streamlined software delivery processes.