Inside the Asterisk

Keyword: SFU

Dana and Asterisk, part 2

A couple of weeks ago, Dan Jenkins kindly wrote a guest blog post about Dana — an up-and-coming open source project which helps to highlight some of the great video-conferencing capabilities in Asterisk. In this blog post, I’d like to expand on that, and show you how to get a simple video-conferencing solution up and

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Adding and Removing Media Streams

When stream support was added to Asterisk it was initially done with the focus being for SFU with a single video stream from each participant with the call starting out with video. This is a use case which is useful for a lot of people and has worked well. Coming soon, however, is the ability

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An Asterisk Video Update

Over the past few years we’ve been working to improve the video support in Asterisk. We initially started with adding stream support[1] in a backwards compatible fashion so we could individually address streams and add/remove them. Next we added support for REMB[2] to be able to control the video bitrate with supported clients. We continued

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Enrich Your Conference App with Asterisk Enhanced Messaging – Part 1

At last year’s AstriDevcon, we showed a video conference demonstration application called CyberMegaPhone.  It was a very simple app but it showed how a web developer could create a video conference app of their own using Asterisk’s new WebRTC capabilities.   While we made some significant enhancements to Asterisk’s video capabilities over the past year, we also

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Enhanced Messaging in Asterisk 13 and 15

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 MESSAGE method in both chan_sip and chan_pjsip for some time, with this enhancement, if a conference bridge participant (connected via chan_pjsip) sends an in-dialog MESSAGE to a conference bridge, the

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