[dropshadowbox align=”none” effect=”lifted-both” width=”auto” height=”” background_color=”#ffffff” border_width=”1″ border_color=”#dddddd” ]tl;dr: We fixed the vulnerabilities. If you’d like to read the conclusion of this admittedly long saga,
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
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?
In the previous post, Josh introduced the forthcoming addition of streams to Asterisk. I’m going to piggyback on that to introduce a unified SDP API
A new feature that was initially implemented during a recent visit to SIPit has now been merged into the 13, 14, and master Asterisk branches.
When the PJSIP work for Asterisk began one of the primary concerns kept in mind was that it be extensible. One of the APIs derived
Asterisk 14 now has the ability to publish extension state using PJSIP PUBLISH requests to another entity acting as an event state compositor. For larger
Asterisk 13.8.0 will come with a new option for enabling PJSIP functionality. This functionality is called bundling and comes courtesy of a community member, George
Debugging SIP Messages the Traditional Way Since its release, the PJSIP stack has provided logging of SIP message traffic via the pjsip set logger CLI
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