OpenSIPS is one of the most used Open Source SIP Servers in the world. It's routing millions of calls across the globe each day. It has been 100% Open Source and has been backed by a robust community for over 10 years.
This year will have to forget of the beautiful city of Amsterdam and move into the DISTRIBUTED conferencing clouds. The ONLINE OpenSIPS Summit aims to be a very dynamic, free event, with live presentations and a high level of interaction between participants. The event is built around a team of moderators, a group of remarkable people from the OpenSIPS, SIP and RTC communities. They will be the beating heart of the event, challenging the presenter in front of the audience with ad-hoc questions, ideas or debates around the presented topic. A kind of round table with the solely purpose of getting the best out the presentations.
Join us online to engage and learn about all the newest developments in OpenSIPS. You will surely walk away having captured important knowledge and insight into how OpenSIPS can be, and is being, used in some of the top companies in UCaaS and RTC. Experiencing first hand presentations on complex end-user deployments, high throughput infrastructure components, and the latest improvements to OpenSIPS is undoubtedly a "can't miss" experience for employees of carriers, telcos or ITSPs.
As the popularity of OpenSIPS grows and it becomes a de facto part of deployments, you will find conference attendees are drawn from many areas of business both technical and non-technical. They include CTOs, Lead Engineers and Technical decision makers from small, medium and large enterprises, corporations and organizations worldwide.Add to My Calendar
Around 15 slots are available for speakers from all over the world, developers, managers/leaders or C-level. They will cover the 30 minutes slots with presentation over technical aspects, integration challenges or solution overviews. Their topics are mainly OpenSIPS centric, but also tackling other softwares, projects or technologies. After all, each session is a massive knowledge transfer, in terms of news, experience and solutions, from speakers to audience.
The presentations are not only slides, but serveral will be realtime demonstrations for a real-life topics. The most important aspect of these demos is to explain from A to Z all the needed details and steps - editing config file, running in consoles, capturing live traffic, scenario testing and others. Under the eyes of the audience, the guru starts from scratch in order to build and run his demos, interacting with people by questions, answers and debates.
A 4-5 hours of OpenSIPS Training with the core developers team of the project. With a structure of 50% theoretical and 50% practical/labs sessions, we will teach you how to use the new calling API with OpenSIPS 3.1 and how to inject or capture RTP by using the media exchanger.
Use our back channels to meet and discuss with representative people from various Open Source project or from Industry companies. The coffee breaks and the Networking Lounge offer to all our attendees excellent oportunities to extend and strength their relationships with other key people from the VoIP & RTC world. Looking for latest news? Or for ingenious solutions? Or for people for work with? This is the right place to be!
Key people from various areas and various companies & projects are present to the OpenSIPS Summit. Pick up the right people and do a brainstorming session to talk over your ideas and challenges. Valuable input as experience or roadmaps may be provided from both sides, from the developers side and also from the users/integrators side.
Hop in and participate to our raffle for OpenSIPS T-shirt and other cool prizes and gadgets
Online, DistributedGoogle Meet, YouTube and more
7-11 September, 202015:00 PM – 18:00 PM GMT
+15 International SpeakersMore details coming soon!
Free prizesDon’t miss it
An online session of 4 to 5 hours of training. With 50% theoretical part and 50% of hands-on labs, the training covers two new topics specific to OpenSIPS 3.1.
In the first part of the training, the students will learn how to deploy and use the OpenSIPS Calling API - how to create, terminate, put on-hold or transfer calls running via OpenSIPS by using the WebSockets Call API.
The second part is allocated for mastering the media exchange between calls. The students will learn how to inject media/RTP into ongoing calls (from other SIP calls) or how to duplicate and stream the media of the ongoing calls to some third parties SIP entities (like a media server, for recording purposes).
The training will be online, each student having allocated an in-cloud public server for running the labs. All that's required is to have internet access, to have a lptop or PC for ssh-ing into the training server and to run 2 SIP soft clients (laptop/PC and mobile).