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Warning, these tips may not be shocking, but you already know that and you clicked anyway! Shame! Many of us know the frustrating feeling of submitting a
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
Media streams are something that we all use every day. From watching videos on Youtube to placing calls media streams are right there with us
The recently announced Opus codec for Asterisk exposes a few configuration options that allow you to manipulate the encoder for your particular setup. These options
Jitter buffer functionality has been in Asterisk for quite some time now. However, knowing what jitter is in a voice over IP (VoIP) application and
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.
Slight interlude from your regularly scheduled programming. For any interested, Matthew Fredrickson, manager of the Asterisk project, will be giving a webinar today about Asterisk
