# VoIP Troubleshooting Guide

## Things to consider <a href="#voiptroubleshootingguide-somethingstoconsider" id="voiptroubleshootingguide-somethingstoconsider"></a>

{% hint style="warning" %}
Hosted Network support may ask you to use this guide in order to gather support information which will aide with isolating and resolving the issue.
{% endhint %}

You may want to consider a few different things before making changes to the VoIP system or touching the configuration on phones.

* Confirm that there are no issues with the speed of the customer/end-user/s Internet connection. Quite a lot of VoIP issues are caused by high latency or packet loss on the Internet connection
* Is there any devices that the phone connects through to get to the internet other than the modem/router itself? (i.e. any switches) Check to see if the issue lies with one of these devices.

## Common Issues/Problems <a href="#voiptroubleshootingguide-commonissues-problems" id="voiptroubleshootingguide-commonissues-problems"></a>

Below is  a list of some common issues & problems that you may encounter, as well as a few steps you can try to do in order to resolve them before calling support.

### Crackling / static on phone calls

Things to test/check when encountering "Crackling / Static on phone call/s"

Run an **Isolation test** if at all possible (Only have one of the phones connected and try to replicate the issue)

* *This is primarily to check if the issue occurs only when there are more than one SIP device within the network, or if the issue is to do with a specific phone*

Change the phone from using **"UDP"** to using **"TCP"**

* *You can find these settings by logging into the phones **"WebGUI"** and navigating to **"Accounts > Account X > SIP Settings"***
* *The reason for this is that the TCP protocol allows the packets to be tracked end-to-end which can reduce packet loss but tends to require a little more bandwidth*

Change the VoIP codec to **"G729a"** temporarily to see if the crackling/static issue improves

* The **G729a** codec uses higher compression and less bandwidth
* **NOTE:** this codec will cause a decrease in the audio quality of phone calls and as such should not be used as a permanent solution

### Call drops or dead air

This is usually caused by NAT issues (The device getting in/out of the network to the hosted PBX)

Check that **"SIP ALG"** is disabled on the router

* If **"SIP ALG"** is enabled you will need to disable it then reboot the router, once the router comes back online you will then need to reboot all of the phones.
* Rebooting the Router and Phones is important after changing the **ALG** setting as the devices will retain the old settings and **NAT** tables until they are rebooted.
* * ***NOTE:** Some routers don’t allow you to disable **SIP ALG**, in this case it's recommended you replace the router with one where you can.*

Set the local SIP port to be something other than **"5060"**

* You can find this setting within the **"WebGUI"** of the phone under **"Account > Account X > SIP Settings > Local SIP Port"**
* Sometimes this helps with the **"UPnP"** settings in routers as some don’t like having multiple devices using the same **"UPnP"** port

### Phones stop receiving calls after sitting idle

This issue is usually caused by the UPnP ports with the modem/router closing at an interval that is lower than what the phone is set to renew it at.

Change the registration interval from **60** minutes down to between **1-5** minutes

* This setting is located in the phone by going to **"Account > Account X > SIP Settings > Basic Settings"**, the field that needs to be changed is **“Register Expiration”**
* Some routers will remove the **"UPnP"** port quicker than **60** minutes which can cause the phones to be unable to communicate with the VoIP server but still think they are registered.
* Setting the registration interval to a lower setting makes the phone re-register with the server more often which keeps the **"UPnP"** port alive

### SIP registration failure

There are multiple things that may cause SIP registration to **fail**, including being unable to access the VoIP server at all from the internet connection (i.e. routing issues)

Confirm that you can ping and browse to [**sip01.mhn.net.au**](http://sip01.mhn.net.au/)

* This will confirm whether there is a routing issue on the connection or not

Reset the SIP Password and re-enter it into the phone.

* This will eliminate any possibility of an incorrect password.

Reboot the phone or PBX **(if it is a SIP Trunk)** to force re-registration

{% hint style="info" %}
&#x20;If none of the above items fix the issues that the customer/end-user are experiencing please go to the [VoIP Fault](/support/services/voice-over-ip-voip/troubleshooting/voip-fault-guide.md) page
{% endhint %}


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