Introduction Packet loss can be an annoying problem when dealing with real time communication, especially when dealing with video. It’s very noticeable when the screen
With the changing weather, some of us are thinking about summer with excitement, others about winter with trepidation. Regardless of which hemisphere you live in though, it’s time to start thinking about Asterisk 16!
Hey Everybody! For those of you that don’t know me, my name is Matthew Fredrickson and I’m the Asterisk Open Source Project Lead. Does that
Like many things WebRTC is a complex stack of technology within Asterisk and also within the browser. The result of this is that to the
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?
