Saturday |
|
VLC 4.0 4.0++ |
Open Media |
10:30 |
10:55 |
|
The final release of Kodi v18 A two year development story |
Open Media |
11:00 |
11:25 |
|
GStreamer 1.16 and beyond |
Open Media |
11:30 |
11:55 |
|
Fearless Multimedia Programming using GStreamer & Rust |
Open Media |
12:00 |
12:25 |
|
GStreamer embedded state of the union 2019 |
Open Media |
12:30 |
12:55 |
|
Inject the Web into your GStreamer pipeline with WPE using a GStreamer/WebKit source element |
Open Media |
13:00 |
13:10 |
|
Voice Controlled Radio Enabling broadcast reception for Smart Speakers |
Open Media |
13:15 |
13:25 |
|
EBUs - Live IP Software Toolkit Open Source Software in professional media |
Open Media |
13:30 |
13:55 |
|
RIST - an evolutionary video transport protocol |
Open Media |
14:00 |
14:25 |
|
Video Analysis using CUDA and OpenCV Detecting scene changes in videos using CUDA and OpenCV |
Open Media |
14:30 |
14:55 |
|
Futatabi: Multi-camera instant replay with slow motion |
Open Media |
15:00 |
15:25 |
|
The SReview review system |
Open Media |
15:30 |
15:55 |
|
How libre can you go? Reaching as many viewers as possible using only libre video technologies. |
Open Media |
16:00 |
16:25 |
|
Migrating from Adobe Connect - the Victory of FOSS Over Proprietary Software |
Open Media |
16:30 |
17:25 |
|
ossia ecosystem workshop Combining media of all kinds with libossia and ossia score |
Open Media |
17:30 |
18:25 |
Sunday |
|
Asterisk WebRTC frontier: make client SIP Phone with sipML5 and Janus Gateway Analyzing a real project on production (www.nethvoice.it) we will look at two different implementations of a SIP Phone WebRTC of NethCTI Web App. We will see great code examples, WebRTC technologies and a real demo of an audio/video call |
Real Time Communications (RTC) |
09:00 |
09:20 |
|
XMPP Beyond Instant Messaging How we use XMPP to do many neat features |
Real Time Communications (RTC) |
09:25 |
09:45 |
|
Kamailio VoIP development update |
Real Time Communications (RTC) |
09:50 |
10:10 |
|
Beyond the webrtc.org monoculture Alternative WebRTC implementations in C and Python |
Real Time Communications (RTC) |
10:15 |
10:35 |
|
Breaking the 100 bits per second barrier with Matrix An entirely new transport for Matrix for really terrible networks. |
Real Time Communications (RTC) |
10:40 |
11:00 |
|
Break the Messaging Silos with COI Get to know the Chat Over IMAP initiative |
Real Time Communications (RTC) |
11:05 |
11:25 |
|
Building Immersive Experiences with the Web The power of the Web with WebXR |
Real Time Communications (RTC) |
11:30 |
11:50 |
|
Introduction to reSIProcate A quickstart for C++ SIP application development |
Real Time Communications (RTC) |
11:55 |
12:15 |
|
Asterisk 16: What's new in the world of Asterisk |
Real Time Communications (RTC) |
12:20 |
12:40 |
|
Building a Multi-Node SIP Platform Using OpenSIPS Cluster multiple OpenSIPS nodes to create a highly available, multi-node SIP platform |
Real Time Communications (RTC) |
12:45 |
13:05 |
|
Going mobile with React Native and WebRTC How Jitsi Meet went from web to mobile, while sharing most of its code |
Real Time Communications (RTC) |
13:10 |
13:30 |
|
Artificial Intelligence, Fuzzing and WebRTC using Janus Having fun with Janus, libFuzzer, OpenCV and Tensorflow |
Real Time Communications (RTC) |
13:35 |
13:55 |
|
Converse: Open, federated teamchat with XMPP |
Real Time Communications (RTC) |
14:00 |
14:20 |
|
Unified Communications with Pàdé Making the X in XMPP work with software |
Real Time Communications (RTC) |
14:25 |
14:45 |
|
HOMER RTC Stats Timeseries for Fun and Profit |
Real Time Communications (RTC) |
14:50 |
15:10 |
|
VoIP Troubleshooting and Monitoring with SIP3 Twenty Thousand MPS under the SIP: VoIP network troubleshooting and monitoring simplified |
Real Time Communications (RTC) |
15:15 |
15:35 |
|
Fraud mitigation using traffic pattern monitoring with CGRateS |
Real Time Communications (RTC) |
15:40 |
16:00 |
|
Make XMPP Sprint Again |
Real Time Communications (RTC) |
16:05 |
16:25 |