rtcp-mux in WebRTC
Mark Michelson
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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?
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?
This week, we’re pleased to say that we’ve updated the Asterisk 13, 14 and master branches’ bundled version of pjproject to 2.6.
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
Busy Asterisk systems can be affected by the SIP timers T1 and B timeout values configured. Consideration of their values impacts how quickly a transaction
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
In this post, we’re going to take a deep dive into a bug that was reported against the 13.9.0 version of Asterisk shortly after its
It seems just like yesterday that Asterisk 13.8.0 was released but here we are with an Asterisk 13.9.0 (and subsequently 13.9.1). This post shines the
Been wanting to try the new PJSIP stack but finding the configuration a little daunting? Then this blog post is for you! While the basic
