Video has been a continued theme of Asterisk for some years now. We put into place the foundation to allow us to do video better, and have over time taken advantage of this and advanced things further. I thought I would take a little bit of time to reflect back on what has been done.
Inside the Asterisk
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
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
Over the past few years we’ve been working to improve the video support in Asterisk. We initially started with adding stream support in a backwards compatible fashion so we could individually address streams and add/remove them. Next we added support for REMB to be able to control the video bitrate with supported clients. We continued
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
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