• Janus - Meetecho - SIP - WebRTC

    Dial-out and cascaded mixing with the AudioBridge

    April 3, 2025 - By 

    A couple of years ago I wrote a blog post explaining how I used some existing features in the AudioBridge plugin to allow SIP endpoints to dial-in and join a conversation that would normally be limited to WebRTC participants alone. That worked nicely and, at the time, it also was an interesting opportunity to play a bit with Drachtio as the framework to mediate between the Janus API and SIP. That said, it also showed some limitations: in particular, it only allowed SIP endpoints to dial-in, but there was no way to have the conference bridge “dial-out” instead. As such,…

  • Events - Janus - Meetecho - SIP - WebRTC

    The return of JanusCon!

    May 10, 2024 - By 

    After five long years from the first edition, our Janus conference was finally back, once again in our beautiful Naples! More precisely, we met RTC developers and aficionados from all over the world on the 29th and 30th of April 2024, for two days of fun and interesting presentations. Great food was eaten, amazing weather was enjoyed, and I think we all had a lot of fun! If you want to check the recordings or the slides of the talks, you can find them in the schedule section of the JanusCon website. This is something I was really looking forward…

  • Janus - SIP - WebRTC

    Bridging AudioBridge and SIP with Drachtio

    October 5, 2022 - By 

    One question we often get is: how do I use the SIP and VideoRoom or AudioBridge plugin together? The short answer is that directly you can’t, because of how Janus is conceived. While Janus is indeed modular, and made of different plugins that all refer to the core, its plugins don’t really talk to each other. This means that it’s true that the SIP plugin allows Janus to act as a SIP gateway, but to facilitate WebRTC users to engage with a SIP infrastructure, not add SIP functionality to other Janus plugins. That said, there are indeed some more “creative”…

  • IETF - Janus - SIP - WebRTC

    Real-time text, SIP and WebRTC

    December 18, 2019 - By 

    Real-time text (RTT) has been around for quite some time. Many people actually confuse RTT with Instant Messaging (IM), while the two are quite different, not only in terms of user experience but also from a technological perspective. In fact, where IM envisages people exchanging more or less complete messages among each other only after a trigger of some sort (e.g., when the enter key or a send button is pressed), RTT has text exchanged while it’s being typed. This means that any party in the conversation can, at any time, see the text while the person is typing it,…