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Mike Bradeen
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If you have ever tried to use direct media with Asterisk, you my have received this message before. It can happen when the User Agent
If you have ever tried to use direct media with Asterisk, you my have received this message before. It can happen when the User Agent
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,
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
One of the most common issues I see when people deploy SIP is calls hanging up after approximately 30 seconds or traffic not going to
Hey Everybody, It’s about a month out from AstriDevCon 2018 and I wanted to write a little bit to summarize what we discussed this year.
In the past, we’ve had a few blog posts talking about specific parts of new WebRTC work that has been done in Asterisk; but, with
Introduction Packet loss can be an annoying problem when dealing with real time communication, especially when dealing with video. It’s very noticeable when the screen
[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,
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?
The Sangoma Blog