Inside the Asterisk

Keyword: WebRTC

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

Read More

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

Read More

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

Read More

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

Read More

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

Read More

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

Read More

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

Read More

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

Read More

Asterisk-16.0.0-rc1 Released!

Hello Everybody! Can you believe how quickly a year has passed? Time is drawing near to many exciting things – Astricon in October and with that, preparations for the first release of Asterisk 16. The first step in creating a 16.0.0 release of Asterisk is the cutting of the first pre-release version, Asterisk-16.0.0-rc1, from the

Read More

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

Read More
Scroll to Top