WebRTC and QUIC: how hard can it be? @ RTC.ON 2024Lorenzo Miniero
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狠狠撸s for my "WebRTC and QUIC: how hard can it be?" presentation at the RTC.ON 2024 event.
They describe my efforts studying and prototyping QUIC, to then move to WebTransport and ways to do real-time media over QUIC. It focuses specifically on RTP Over QUIC (RoQ) and Media Over QUIC (MoQ), documenting my attempts to get them to "talk" to WebRTC with the help of Janus.
WebRTC and SIP not just audio and video @ OpenSIPS 2024Lorenzo Miniero
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狠狠撸s for my "WebRTC-to-SIP and back: it's not all about audio and video" presentation at the OpenSIPS Summit 2024.
They describe my prototype efforts to add gatewaying support for a few SIP application protocols (T.140 for real-time text and MSRP) to Janus via data channels, with the related implementation challenges and the interesting opportunities they open.
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WebRTC and QUIC: how hard can it be? @ RTC.ON 2024Lorenzo Miniero
?
狠狠撸s for my "WebRTC and QUIC: how hard can it be?" presentation at the RTC.ON 2024 event.
They describe my efforts studying and prototyping QUIC, to then move to WebTransport and ways to do real-time media over QUIC. It focuses specifically on RTP Over QUIC (RoQ) and Media Over QUIC (MoQ), documenting my attempts to get them to "talk" to WebRTC with the help of Janus.
WebRTC and SIP not just audio and video @ OpenSIPS 2024Lorenzo Miniero
?
狠狠撸s for my "WebRTC-to-SIP and back: it's not all about audio and video" presentation at the OpenSIPS Summit 2024.
They describe my prototype efforts to add gatewaying support for a few SIP application protocols (T.140 for real-time text and MSRP) to Janus via data channels, with the related implementation challenges and the interesting opportunities they open.
狠狠撸s for my "Am I sober or am I trunk? A Janus story" presentation at Kamailio World 2024.
They describe my prototype efforts to add an option to create a trunk between a Janus instance and a SIP server, with the related implementation challenges and the interesting opportunities it opens.
Getting AV1/SVC to work in the Janus WebRTC ServerLorenzo Miniero
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狠狠撸s for the "Getting AV1/SVC to work in the Janus WebRTC Server" presentation I made at the Real-Time Communications devroom of FOSDEM 2024 in Brussels. It describes in detail how AV1 is used in real-time communications (e.g., RTP packetization rules) and how the Dependency Descriptor extensions allows for SVC to be used in a server, by sharing my experience integrating it in the Janus WebRTC Server.
狠狠撸s for the "WebRTC broadcasting: standardization, challenges and opportunities" presentation I made at TADSummit 2023 in Paris. It presents the problems traditional broadcasting has with new scenarios that would benefit from a much lower latency solution, and how WebRTC can help. It also introduces the standard WHIP and WHEP protocols for ingestion and egress, with a few details on how a WebRTC stream could be scaled to a very wide audience using something like SOLEIL (Streaming Of Large scale Events over Internet cLouds).
This document provides an overview of bandwidth estimation in the Janus WebRTC server. It discusses:
- The importance of bandwidth estimation and congestion control for real-time media like WebRTC.
- Challenges in applying existing bandwidth estimation algorithms designed for endpoints (like GCC) to servers that don't generate their own media.
- An approach taken in Janus to develop a simpler, ad-hoc bandwidth estimation technique for servers based on acknowledged rate, losses, and delays - without relying on existing complex standards-track algorithms.
The challenges of hybrid meetings @ CommCon 2023Lorenzo Miniero
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The document discusses the challenges of hybrid meetings that involve both in-person and remote attendees. It describes how a company called Meetecho has adapted their virtual event platform to address these challenges. Some of the key points discussed are integrating local and remote audio through a Janus AudioBridge, adding multiple cameras and dynamic video feeds, sharing all presentation content through the platform regardless of source, putting all attendees in a single queue, and using an on-site Meetecho tool to give local attendees remote-like functions.
Real-Time Text and WebRTC @ Kamailio World 2023Lorenzo Miniero
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狠狠撸s for my "Bringing real-time text to WebRTC for NG Emergency Services" presentation at Kamailio World 2023.
They describe my prototype efforts to get SIP-based T.140 Real-Time Text to work with WebRTC endpoints via data channels, thanks to Janus acting as a gateway for the purpose.
狠狠撸s I presented in the Open Media devroom at FOSDEM 2023, where I gave an intro on how to capture, record and produce music using just open source software on Linux. It's a very high level overview on available software to do different things, and how they can be used together using JACK and/or Pipewire.
These are the slides for the presentation I shared at the virtual edition of IIT-RTC 2022. I talked about how cascading/scalability worked with Janus 0.x, and what steps we've taken to do the same for 1.x (multistream) as well. In particular, the focus is on the new integrated cascading support in the VideoRoom plugin.
SIP transfer with Janus/WebRTC @ OpenSIPS 2022Lorenzo Miniero
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1. The document discusses adding call transfer and multiple call support to the Janus SIP plugin. It describes implementing generic SUBSCRIBE/NOTIFY support, a mechanism for handling multiple lines and calls simultaneously, and actual call transfer signaling orchestration.
2. A demo showed transferring a call between Alice, Bob, and Carol using the Janus API and new SIP plugin features. Code snippets demonstrated the API messages for initiating, transferring, and receiving transferred calls.
3. The development was sponsored by a Colombian company and involved merging several pull requests to add the new functionality to Janus and improve call transfer support when using the SIP plugin.
狠狠撸s for the talk I made at the virtual edition of FOSDEM 2022, on how to use WHIP for WebRTC broadcasting ingestion, and how the distribution process could be done via WebRTC as well, e.g., via Janus (and the SOLEIL architecture).
狠狠撸s for the presentation I made at ClueCon 21 on the experimental RED support in WebRTC, and how we've started tinkering with it in Janus. The presentation also addresses a more generic overview on audio features in WebRTC.
This document discusses WebRTC ingestion for broadcasting (WHIP), a new protocol being standardized by the IETF. It provides an overview of WHIP, including how it can be implemented using existing WebRTC servers like Janus. The presenter describes creating a simple WHIP server and client as proof-of-concept implementations. Interoperability between different WHIP server and client implementations is also discussed.
These slides cover a workshop called "Having fun with Janus and WebRTC" at the virtual edition of OpenSIPS 2021. The workshop guided viewers to how they could use different features in Janus to build a WebRTC Social TV application, including how to write a new plugin in JavaScript to build a virtual remote.
狠狠撸s for the presentation I did remotely at Open Source World, to talk about audio-only WebRTC applications, and what we've done in Janus to improve and cover the requirements so far.
Just a few slides to talk about the first efforts on JamRTC, a native application based on GStreamer to do live jam sessions using WebRTC and Janus as an SFU. Mostly an overview of the initial architecture, with questions at the end to figure out if the approach is right or not, how to minimize latency, etc.
Scaling WebRTC deployments with multicast @ IETF 110 MBONEDLorenzo Miniero
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An overview of how multicast can be used to scale WebRTC deployments, with focus on the Virtual Event Platform used to provide remote participation support to IETF meetings, given during the MBONED WG session at IETF 110.
狠狠撸s for the 60 minutes "part 2" Janus workshop I presented at the virtual edition of ClueCon 2021. This time the slides covered Janus ability to bridge WebRTC and non-WebRTC applications to do interesting things, especially with the help of plain RTP and RTP forwarders. Check the conference recordings to see the actual demos in action.