How to verify the Internet and network configuration for calls


The webphone allows Web browsers and mobile devices to send and receive voice calls via a two-way ongoing audio channel.

The system works without any problem on Internet box accesses. These accesses are based on a classical NAT system which accept outgoing connections to Internet and only accept incoming answers to already established requests.

If you own a BOX type classical Internet connection

You have nothing to change

Advanced settings

If you have a restricted access or must indicate to your IT department accesses to go through. Please refer to the indications below.

If you wish to set up a specific router, the ideal is to allow our addresses in "box" mode, that is to say allowing all outgoing connections to Internet and only "STATE-ETABLISHED" type incoming connections, that is to say the feedback after an outgoing connection.

Addresses to allow

Here are the telephony servers adresses

Warning. This server adresses are likely to be changed depending on our architecture. We recommend to check this page if any dysfunction appears.

Ports and protocols to allow

Si vous souhaitez restreindre les ports, les ports utilisés par le webphone sont :


 Port destination à autoriser


WSS (Signaling)   



HTTPS (Signaling)                



SIP (Signaling) 



RTP (audio)

16384 à 32768







Codec and Bandwidth required

The webphone uses the G711 codec. You must plan 100 Kb/s per user


Testing my configuration

The webphone is based on the WebRTC protocol, available on most of recent browsers.

We recommend to use an updated version of Chrome or Firefox.

If you want to test the proper functioning of your computer or configuration, you can use one of the following diagnostic pages.


Warning. These testing tools don't depend on Web2Contact and we cannot guarantee  their proper functioning.

To test your available bandwidth :



For any questions, contact the Support.