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As per the links you provided, you must change your status to "Offline" before closing the application. Until Skype changes this behavior, it seems like your only 1-step solution.


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As there are many settings that could influence the quality of VoIP, I recommend comparing all settings with those listed in the article : How to configure your router for blazing speeds. You should especially disable all unnecessary WiFi modes and protocols, for example "b+g WiFi mode" if you only have "n" devices, or 5GHz if you only have 2.4 GHz devices. ...


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A SIP trunk is a direct connection between your organization and an Internet telephony service provider. SIP trunk enables you to extend VoIP telephony beyond your organization’s firewall without the need for an IP-PSTN gateway. This simpler configuration is easier and less expensive to design, operate, maintain, and upgrade. Here is a suggested [blog]: ...


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There are two channels used per VoIP call. One is upload channel (talking) and the download channel (listening). Assuming, you and the person you are talking to take turns talking, then we can assume your telephone call is using approximately a maximum of 110 Kbps. There are various aspects you can check your bandwidth before installing the VoIP service. ...


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yes since media in webrtc is by default, peer to peer , and if the endpoints do not support the same codes , you definitely need a media server that acts like a trans-coder . you also need the rtpproxy module in kamilio to force the rtp to proxy through a media server .


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I tried to think of any particular feature that you might miss and.. here is my list: Very long calls (mobile phones battery is drained out quickly with wifi on and compression w/o dedicated hw chips) More codecs available on hw phone More NAT traversal options on hw phone Three-way Conference Calling with local mixing on the hw phone SPA3000 has also a ...



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