Improved PJSIP Qualify Support Performance
Joshua C. Colp
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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
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
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?
This week, we’re pleased to say that we’ve updated the Asterisk 13, 15 and master branches’ bundled version of pjproject to 2.7.1. This release contains
Security releases of Asterisk were recently created. In this post, we’d like to go into the depths of two of the security issues and how
Do you use WebRTC with Asterisk? Did you notice calls stop working after updating Google Chrome to version 57? Are you curious why that happened?