This guide will help you to troubleshoot any call quality issues you may encounter.
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, press the Start button, and in the Search box, type in cmd and hit enter. (In Windows XP, press Start then Run and type in cmd and hit Run).
A black command prompt should appear.
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 [220.127.116.11] with 32 bytes of data: Reply from 18.104.22.168: bytes=32 time<26ms TTL=54 Reply from 22.214.171.124: bytes=32 time<28ms TTL=54 Reply from 126.96.36.199: bytes=32 time<27ms TTL=54 Reply from 188.8.131.52: bytes=32 time<27ms TTL=54 Ping statistics for 184.108.40.206: 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. If your download speed is 14 Meg or more, some download bandwidth can be sacrificed for an increase in upload speed.
If you find 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 is typically Auto + SIP INFO if available. If that does not work, try Inband + SIP INFO if available.
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.
By default, SIP Transport is typically UDP as it is faster. Setting it to TCP can make it more reliable, at the expense of being slightly slower, but nothing noticable. This has no effect on the quality of audio.
See Using Command Line Tools on Windows and GNU/Linux for more advanced troubleshooting techniques, such as network captures.
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