Measuring performance in Asterisk, like most software systems for that matter, can be a complicated task. When testing performance it is important to define goals,
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
The next releases of Asterisk 13 and 15 extend MESSAGE support in chan_pjsip and add it to conference bridges. While Asterisk has supported the SIP
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
TLS certificates and their management are something we take for granted every day when we visit a website. If you sit down and try to
A basic concept with chan_pjsip/res_pjsip is the endpoint. When a new SIP request comes in, res_pjsip needs to identify which endpoint the request is for.
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
Recently there’s been discussion on chan_sip going away in the future which led to many comparisons between it and chan_pjsip. What does chan_pjsip do better?
Setting the “max_contacts” configuration option too low can cause loss of registrations. How that can happen and some recommendations are described.
The Sangoma Blog