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
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 where it should. This can be hard for users to grasp and is primarily due to the fact that SIP embeds routing information (IP addresses and ports) within the signaling
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. For those who are not familiar with AstriDevCon, it is an opportunity for Asterisk C-level developers, Asterisk ARI/AMI/AGI developers, and Asterisk integrators to get together to learn about what has
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 the release of Asterisk 16, we need to talk about the real-life impact of this work under poorly-performing networks and the resulting video experience. Before we start, let’s dive into
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 freezes for multiple seconds, then the footage resumes with everything in a completely different position than it was originally. We’ve all seen this before. Packet loss is inevitable, but it
[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, scroll down to the Conclusion at the end.[/dropshadowbox] Overview This month, the Asterisk project performed two security releases to address an unauthorized RTP data disclosure vulnerability in its real-time transport
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 started with is not the one we currently use. It is in thanks to the community that has contributed both issues and fixes that our WebRTC has continued to improve.
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 answer is the rtcp-mux feature. What is rtcp-mux? The majority of VoIP protocols make use of the Realtime Transmission Protocol (RTP) for transmitting and receiving media. In addition to
