Inside the Asterisk

Category: WebRTC

Local Channel Multistream and Re-Negotiation Support

When stream support was initially added to Asterisk we did it in the most backwards compatible way possible to ensure that we did not have to modify the entirety of Asterisk. This has allowed us to gradually improve parts of Asterisk as we’ve expanded our stream and video support. To that end the next part

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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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Introducing Dana the Stream Gatekeeper

In this troubling time of dealing with COVID-19 around the world we’re seeing more and more need for tools to help in communicating with co-workers, friends and family. Asterisk has historically proven itself as one of the key puzzle pieces when it comes to enacting change and evolution in the communications space. The power of

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Issue Walkthrough: DTLS, and the Case of Bad Audio

A past post discusses some tips and tricks to employ when encountering a problem. Here we’ll walk through an issue applying some of those techniques. Recently an Asterisk issue came up involving occasional static and/or silence for audio. Broadly speaking the problem can be described as Alice calls Bob (using SIP), Bob answers, the call

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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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Support for large certificate sizes in DTLS now available!

When DTLS support in Asterisk was added the information about how to use DTLS support in OpenSSL was not as flushed out as it is today. To that end the implementation was written to use OpenSSL memory buffers. These are places for OpenSSL to place received data or for OpenSSL to consult when sending a

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transport-cc: Improving feedback for better video quality

I’ve previously written about REMB, or receiver estimated maximum bitrate, and its effect on video quality. While this provides periodic feedback from receivers to Asterisk and a mechanism to set the video bitrate of a sender it does not allow a sender to have any feedback about the packets it is sending to Asterisk. To

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Enrich Your Conference App with Asterisk Enhanced Messaging – Part 2

In “Enrich Your Conference App with Asterisk Enhanced Messaging – Part 1” I demonstrated how you could include chat or other messaging features in your conference app.  In Part 2, I’ll show you how to include information about the conference bridge itself and the participants. What data is available? If you’re familiar with the AMI

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Improving Video Quality In The Real World

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

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