Improved PJSIP Qualify Support Performance
One of the most difficult things in PJSIP is ensuring that the experience is the best it can be for not just people who configure
One of the most difficult things in PJSIP is ensuring that the experience is the best it can be for not just people who configure
For the last few months I, along with Ben Ford, have been working on improving the user experience side of the WebRTC support in Asterisk.
Asterisk allows users to manipulate call party identification information through mechanisms like configuration options and dialplan functions (for instance CALLERID and CONNECTEDLINE to name a
The Story of Asterisk and Keep-Alives The vast majority of VoIP communications is done via UDP datagrams. It’s a no-overhead protocol which makes it fast
In my previous post I talked about what WebRTC support is like in Asterisk 14. Since Asterisk 15 is going to be released soon let’s
WebRTC has been in Asterisk since Asterisk 11 and over time has evolved just as the WebRTC specification itself has evolved. The WebRTC implementation we
So you’ve heard there is now an Opus codec for Asterisk that’s been released. However, you are having problems with poor audio quality due to
There are several handler routines available to allow you to customize behavior for the different states of a call. Handler routines execute outside of the
The recently announced Opus codec for Asterisk exposes a few configuration options that allow you to manipulate the encoder for your particular setup. These options
The Digium Phone Module for Asterisk and the g729a, silk, siren7 and siren14 codec binary modules hosted at downloads.digium.com can now be automatically downloaded and installed