---
title: Troubleshooting SIP Connection / SIP Trunks
description: Help Center
---

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1. [Help Center](https://help.voipsupply.com/migration/en/knowledge?hsLang=en)

# Troubleshooting SIP Connection / SIP Trunks

For SIP trunks only:  
\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*  
Ensure you are port forwarding SIP and RTP traffic in the router to your PBX's IP address.

SIP is 5060 UDP and RTP is a range of 10000 - 20000 UDP.

Check your NAT settings in the PBX \> SIP Settings \> NAT

Make sure you define your static outside IP and internal IP/24 or your matching CIDR IP range.

No Inbound calls:  
\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*

Firewall – Make sure the firewall is not blocking inbound connection to the IP/PBX. 

As a general rule, make sure that UDP ports 5060, 10000-20000 are open for SIP/RTP Streams, also in some configurations you may need to extend the range of ports to 31000. 

You might also need to check other Audio RTP Ports based on the phone/Equipment you are using.

Broadband – Make sure that your Broadband provider is not blocking ports or the SIP service to and from your VoIP provider.

One way audio:  
\*\*\*\*\*\*\*\*\*\*\*\*\*\*\*  
SIP Transformations /ALG-"SIP transformations" on some routers may be enabled by default: Please disable. You can run "sip-alg-detector".exe to ensure your network is set as a FULL CONE NAT and there are no PAT issues occurring.

Firewall Ports – Make sure the firewall is not blocking ports, as mentioned above.

Codec Settings – Make sure you are using the correct Codec settings. A lot of providers prefer you to use G.711Alaw/G.729. Using the wrong codec on any port of the call can stop audio from working in a direction if the VoIP provider is not Transcoding between the codecs.  
CPU Spike!

Bad Audio/Garbled-Choppy Sounding?

Check LAN – Make sure you have no internal LAN problems like Loops or programs that send out high volumes of multi-cast packets.

Check Bandwidth – Make sure that on both your LAN and WAN/ISP you have enough bandwidth to suit the codec’s you are using for SIP/VoIP. 

As a rule, G.711Alaw uses 87.2kbps(8.7KBps). 

Check Broadband –  Check the Bandwidth you have available from your ISP to make sure you can make calls with a guaranteed bandwidth amount. 

As a rule you want a minimum of 2Mb down, and 2Mb up.

Check For Packet Loss: Wireshark Much?! – run a packet loss trace using either MTR (Linux) or WinMTR (Windows) to both your ISP's DNS Servers AND the VoIP provider's servers you are connecting to, if you see any lost packets,   
then you know that your calls are going to have words and bits missing and you will be either asking to repeat stuff or be asked to repeat stuff.

Choppy or Garbled? Please ensure you have QoS established on the network for packet prioritization. No fatty packets are allowed! 

That's it!

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