This guide will help you to troubleshoot your handset. Please refer to the section below that relates to the issue you are experiencing.
If you are hearing stuttering, bubbling, or choppy audio, then your internet connection may be over-utilised.
This can be due to a slow connection speed, large files being downloaded or uploaded, or capping by your ISP.
On Windows Vista/7, press the Start button, and in the Search box, type in cmd and hit enter. (On Windows XP, press Start then Run and type in cmd and hit Run).
A black command prompt should appear.
Type in
ping www.google.co.uk
And hit enter.
You should get back 3 or 4 responses, giving you the time it took for them to come back. If it is 40ms and under, then that is very good. But if they are over 350ms, then that can cause stuttering, and can be due to large amount of downloading.
If you get above 0% packet loss, try again. If you are getting packet loss quite often, then that can indicate a fault on your line, or excessive downloading.
ping www.google.com
Pinging www.l.google.com [74.125.230.115] with 32 bytes of data:
Reply from 74.125.230.115: bytes=32 time<26ms TTL=54
Reply from 74.125.230.115: bytes=32 time<28ms TTL=54
Reply from 74.125.230.115: bytes=32 time<27ms TTL=54
Reply from 74.125.230.115: bytes=32 time<27ms TTL=54
Ping statistics for 74.125.230.115:
Packets: Sent = 4, Received = 4, Lost = 0 (0% loss),
Approximate round trip times in milli-seconds:
Minimum = 26ms, Maximum = 28ms, Average =27ms
Please note that VoIP phone calls rely on available upload bandwidth as well as download. Most “Up to 24 Meg” can give very fast download speeds, however the upload speed is typically not more than 1 Meg.
It might be worth asking your ISP if an “Annex M” option is available. This sacrifices some download speed for a gain in upload speed. Contact our Sales Team for more information on the broadband and Ethernet connectivity we can provide.
If you have issues with DTMF (eg when asked to press 1 for Sales, 2 for Support etc), try modifying your DTMF method usually found under Advanced Account settings on your phone.
The setting DTMF Type should be RFC2833. SureVoIP does not support other DTMF methods.
If you find your phone needs to “wake up” before you can make calls, or transferring calls seem to time out, try changing your SIP Transport mode.
Also, ensure the NAT keep-alives are enabled.
You may also increase the keep-alive frequency to 30 seconds. Do not set your keep-alive frequency to a low figure - please adjust your NAT session timeout instead.
By default, SIP Transport is typically UDP. This should be used where possible.
If you have an internet connection prone to packet loss or high amount of jitter, setting to TCP may improve success making calls or transferring calls. However, this will not help with voice quality if your internet connection is poor.
If you are using a security appliance, you may need to adjust the UDP NAT time-outs to not close the translation for your phones prematurely. Increasing keep-alive frequency may help with this.
If you are using Powerline adapters then try plugging your handset into your broadband router directly. If your phone now works then you have an issue with your Powerline adapter pair that you will need to resolve.
For the Hosted VoIP service, do not forward any ports to your handsets.
Ensure SIP ALG has been disabled on your router. See this page for guidance.
See Using Command Line Tools on Windows and GNU/Linux for more advanced troubleshooting techniques, such as network captures.
See this guide on how to obtain SIP traces from your phone system.