WebTrit at Kamailio World 2023 – Recap: WebTrit in 5 slides and 5 minutes!, dangerous demos, and great networking.
Berlin in the summer, what could be better? This week our co-founder and advisor Andriy Zhylenko traveled to Kamailio World in Berlin to join an elite group of open source telephony engineers and business leaders interested in the Kamilio open source sip router technology.
Andriy had the opportunity to share an overview of the WebTrit service and open source features and how it addresses the problem of developing your own Softphone and WebRTC services.
TL;DW: WebTrit leverages the best of Open Source and CPaaS capabilities to make life easier for developers creating voice and video applications for iOS, Android, and web applications, drastically reducing the time and effort required to bring new services to market. Service providers and developers using Kamailio and other SIP infrastructure will be able to interconnect easily with WebTrit, and publish their applications under their own name using the WebTrit open source code. Join the WebTrit community, check out https://github.com/webtrit or visit our website to get started.
It’s a tradition at Open Source telecommunications conferences that the legendary James Body invites brave developers to do a “Dangerous Demo”. This year Andriy joined in as a first timer to show how easy it is to connect WebTrit in demo mode to a new VoIP system. Time is limited for the dangerous demos, it’s always a high wire act for the presenter, but Andriy showed WebTrit connecting to the Sipgate network on the fly, in under 3 minutes. Not enough time on the clock to show the entire process, but as James says “That’s what Dangerous Demos are all about.”
Contact WebTrit for your own (very safe) demo and we’ll be able to go into more detail!
Thanks to the Kamailio team and event organizers, see you again next year in Berlin!
Read the edited transcript of the video:
Simplifying WebRTC Application Development for Developers and Enhancing User Experience.
A Journey towards Easier Development and Happier Customers
In the world of software development, creating WebRTC applications can be a complex task for developers. It involves dealing with media transport, designing signaling mechanisms, and understanding the intricacies of VoIP (Voice over Internet Protocol). However, there is a solution that aims to simplify the process and provide an enhanced experience for end users. In this transcript, discover how Andriy Zhylenko addresses these challenges and introduces a game-changing SDK that brings a smile to WebRTC application development.
The Challenge of WebRTC Application Development
Andriy Zhylenko begins his talk by acknowledging the difficulties faced by developers in the audience. Many of them have attempted to create mobile dialer web applications or native voice and video calling applications for Windows or macOS. However, completing these projects successfully and achieving a perfect solution is no easy feat. The complexity of VoIP and the need to develop applications for multiple platforms often make it an overwhelming task for many companies.
The Dislike for Complex VoIP Implementations
Frontend developers, when presented with complex VoIP implementations, tend to dislike the process. The traditional methods of learning all the intricacies of VoIP, such as using CJs or other frameworks, are inefficient and time-consuming. Frontend developers prefer a simpler approach that allows them to focus on creating a great user interface and experience.
The Need for Multi-Platform Applications
The demand for multi-platform applications adds another layer of complexity to the development process. To cater to mobile, web, and desktop users, developers often have to triple their development efforts, making it unfeasible for many companies. Finding a way to streamline the development process and reduce the size of development teams becomes crucial.
Introducing the Solution: An SDK for Simplified WebRTC Application Development
Andriy Zhylenko reveals the solution his team has developed to address these challenges. The key is an SDK that provides frontend developers with an easy-to-use API, eliminating the need to delve into the complexities of voice. Instead of starting from scratch, the SDK offers a fully functional application with all the necessary features. This open-source solution can be obtained from GitHub, allowing developers to customize it according to their specific requirements.
Benefits of the SDK
The SDK is not only easy to use but also platform-agnostic. It can seamlessly connect to various VoIP systems, such as Kameo, FreePBX, or 3CX, without being tied to a specific vendor. This flexibility ensures that developers can adapt their applications to changes in the infrastructure or integrate additional VoIP systems without the need for extensive modifications. By utilizing Flutter and Dart, the SDK enables the use of a single codebase for Android, iOS, and native web applications, significantly reducing the size of the development team.
WebTrade: Connecting the Backend and Simplifying Signaling
To bridge the gap between the frontend and backend, the SDK includes WebTrade, which handles the conversion from WebRTC and provides a simple signaling API for web and mobile applications. It seamlessly connects to the VoIP system through SIP, ensuring compatibility with different vendors. Additionally, WebTrade offers essential features like push notifications and contact synchronization, allowing developers to deploy production-ready services quickly.
Empowering Developers and Building a Community
The SDK’s open-source nature encourages developers to contribute to its growth and improvement. Developers can customize the user interface, colors, and icons to build their unique applications, which they can then publish under their own names on popular app stores. Creating a community of users and contributors fosters collaboration and paves the way for easier deployments and future advancements.