How To | Understand network ports and firewalls to make a SIP call using Q-SYS

Gain an understanding of the network ports required to make a SIP call through a Q-SYS firewall.

Updated at June 8th, 2023

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Table of Contents

Procedure


There are two distinct elements to a VoIP call:

  • The Session Initiation Protocol (SIP), which is responsible for call negotiation, setup and teardown
  • The Real Time Protocol (RTP), which transports call audio between parties.

SIP

Most phone systems will use port 5060 for non-encrypted signaling traffic, whether UDP or TCP. There may be exceptions, so check the PBX user documentation. Port 5061 is typically used for TLS encrypted traffic.

Additional information: Older versions of the Q-SYS softphone only supported UDP but current versions support UDP as well as TCP and TLS (Transport Layer Security, a protocol that runs over TCP and provides end-to-end security for SIP signaling by encrypting SIP messages that are exchanged between the Q-SYS SIP softphones and far end SIP endpoints or PBXs.

 
 

RTP

While the SIP ports used by a VoIP system are typically fixed, The RTP ports are typically dynamic.  The typical RTP port range of a given system are UDP ports 10,000-30,000.

 
 

Summary

To make VoIP calls through a firewall, open:
UDP/TCP ports 5060-5061
UDP ports 10,000-30,000

This assumes the firewall is not making use if Port Address Translation (PAT).  This may complicate matters with proper VoIP call transversal.