sipgate basic Help

Problems with audio and speech quality

Audio quality will depend on a number of factors that together will determine the quality of your VoIP calls.

Combining a suitable internet connection, appropriate setup of your VoIP phone(s) and the correct configuration of your local network will ensure a reliable sipgate VoIP service.

Internet Connectivity Test:

1. Quick, easy online tests of your internet connection can be made using the following websites:

(Of course, for a fair test, don't choose the nearest test server location!)

Please test your internet connection at different times throughout the day to gain a clear picture of your internet connectivity.

After each test click the 'Direct Link' option and then click 'Copy'. Paste the copied links into a text file so that you can send us the results at the end of the day.

2. Ping and traceroute tests can be made to using your Windows, Mac or Linux machine's command line terminal.

3. Further testing can be done to using applications like WinMTR and Pingplotter.
The more information the better in setting up and ensuring a business standard VoIP service.


Available bandwidth alone will not indicate a connection's suitability for VoIP calling.

Latency (ping), Jitter and packet loss are also very important.

A VoIP call (using the G711 codec) will require circa 90 kbps both up and downstream.

As general rules of thumb, a VoIP call will tolerate latencies up to circa 80 ms. Voice quality will degrade noticeably with Jitter values above 15ms (with calls being entirely unfeasible with Jitter values above 25ms).

Codec Selection:

If you have limited bandwidth available, enabling only lower bandwidth codecs in your phone settings and creating router Quality of Service (QoS), or Bandwidth Management Rules for VoIP traffic may provide a solution.

Please click here for a list of the codecs supported by sipgate.

Network Setup:

­- Clipping of sound is often due to intermittent network congestion.

A heavy file or video download may be stealing bandwidth from your calls, increasing latencies and causing packets (speech!) to be dropped. A download taking an extra few seconds will rarely ever be noticed, but sounding like a Dalek on a conference call will be heard by all.

Router QoS rules prioritising VoIP traffic are recommendable for all customers, and not just where issues arise.


- If you experience lost or one way audio, as a first step, please check that SIP ALG is not enabled in your router settings. SIP ALG is often poorly implemented and is a common source of router issues.

Please also test with Stateful Packet Inspection, SPI, options disabled.

- Be sure to avoid local port conflicts between multiple VoIP Phones in use on the same local network. We recommend using the local port numbering scheme detailed in the following Help article:

Connecting More Than One VoIP Phone or Device to the Same Router


- There's a useful general Help Article about Problems Making & Receiving Calls at:

Problems Making And Receiving Calls

What next?

Unsure what your test results mean?

Not sure whether your connection is unsuitable, or if you just need to tweak a setting or two?

Please let our support Team know, including as much information as you can:
What to do or Check Before Emailing Support?


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