Maintaining and updating WebRTC applications is complex, involving frequent updates to libraries and infrastructure components to keep them secure and efficient. Canary releases provide a strategic way to handle these updates with minimal risk. By deploying changes to a small group of users initially, developers can monitor the new version's performance and address any issues before a full rollout.  In this post, we describe how canary releases are particularly beneficial for updating WebRTC applications, why

If there’s a single theme on the WebRTC.ventures blog, it’s that WebRTC is about much more than just video. While most applications we develop include a video component, the audio and data channels of WebRTC can be used independently of the video channels. In this post, I explore a few of the WebRTC audio use cases that have come up across our content over the last six months. These conversations with industry experts in WebRTC,

On the June 18, 2024 episode of WebRTC Live, Arin welcomed Pion founder Sean DuBois to talk about about broadcasting with WebRTC and OBS. Sean is very active in the WebRTC community. He is the creator of Pion (a WebRTC implementation 100% in Go), a developer at Twitch, Field CTO for LiveKit, and author of WebRTC for the Curious. Bonus Content Updates from CommCon Our regular monthly industry chat with Tsahi Levent-Levi. Key Insights and Episode Highlights below. Watch Episode 92! https://www.youtube.com/embed/9vtuxGR4Lro

After a promising minimum viable product (MVP) using Bubble, our client approached us to help plan the production-ready iteration of their native voice chat app. The project extended beyond conventional mobile app development, encompassing features like capturing and streaming local content, connecting users within the social application, and implementing audio recording and transcription capabilities for moderation purposes. Type of Application: Native iOS Social Audio App Assessment Client: Under NDA until Launch Deliverables: Audio streaming architecture

Imagine turning hours of call transcripts into concise summaries with just a few lines of code, all without provisioning any servers. In this "WebRTC Tips" video, our WebRTC Developer Advocate, Hector Zelaya, demonstrates how to leverage AWS Lambda to build a serverless process that reads call transcript files from Amazon S3 and uses Meta Llama 3 on Amazon Bedrock for seamless summarization. https://www.youtube.com/watch?v=vwyj0RF37U8 At WebRTC.ventures, we specialize in integrating AI features into real-time apps. Visit

It's crucial to ensure your application runs smoothly not only in a variety of network conditions, but also on all platforms. My exploration of network performance testing began in a previous post titled, Testing macOS Network Conditions Using Apple’s Network Link Conditioner. Today, I will turn from macOS to iOS and iPadOS devices.  As defined in that first post, network throttling is a technique that helps us understand how applications perform under various network conditions.

Welcome to a new WebRTC.ventures series where I ask members of our expert WebRTC.ventures team how they work their magic. Ensuring a flawless experience for live event streaming applications means preparing for the inevitable—unstable network conditions when broadcasting. To tackle this, developers and testers must simulate various network issues to see how their applications will perform in the real world. But how? With this question in mind, I reached out to our team of skilled QA

The integration of Generative AI (GenAI) and WebRTC has the potential to revolutionize the way businesses operate, particularly in contact centers. GenAI’s ability to generate content and perform actions based on given instructions is opening new frontiers across various industries. When combined with the real-time communication capabilities of WebRTC, organizations can build seamless, efficient, and highly personalized customer experiences. In this post, we explore the transformative potential of these technologies in enhancing customer service and

Each month, we ask our dynamic team of WebRTC developers, mobile developers, designers, testers, DevOps experts, and project leads to share some of the impactful work they did in the past month. Here’s a short list of some that might interest you — at least the ones we can talk about! Explored designs and architectures for: Adding production-grade monitoring to a telehealth platform using the Amazon Chime SDK Building a highly scalable communications application  Building

Real-time communication has become indispensable for both processes and products. Video meetings support internal communications for remote teams and facilitate sales and customer service activities. They are also increasingly integrated as a standard feature in software products and platforms sold by businesses. When developing video call capabilities, companies face a crucial decision: should they use a starter kit or build an application from scratch?  Here are key factors to consider: 5 Reasons to Use a

WebRTC enables developers to build powerful web-based real-time communication applications. While typically implemented in web browsers, you can also integrate this functionality into native applications by embedding the WebRTC standard into the application’s codebase. This is usually done using libWebRTC, an open-source project maintained by Google that provides a comprehensive implementation of the WebRTC protocols and APIs.  In this post, we provide an overview of the process of building a native video chat application using

On the May 15, 2024 episode of WebRTC Live, Arin welcomed industry guests Jim Rand and James Meier of Synervoz to focus on unique audio use cases for WebRTC. Synervoz is an innovation and software development studio focused on audio, entertainment and online collaboration. They have worked on sound technology with many well known companies such as Bose, Meta, Amazon, and Unity. They discussed their Switchboard SDK for audio processing and explored audio processing use cases

Transcription, a crucial component of modern contact centers activities, is largely facilitated by Automatic Speech Recognition (ASR) systems. These tools, however, can fall short in accuracy and reliability. Consequently, assessing transcript quality becomes imperative and has traditionally involved costly manual processes. Enter Large Language Models (LLMs), presenting a promising and efficient solution to evaluate and improve the quality of automated transcriptions.  In this post, we explore how LLMs can be used not only to evaluate

As Managed Services Manager, I've seen firsthand how critical effective communication is in support of a client’s application infrastructure and end-user systems. That's why I'm excited to share our experience with using Slack as a support channel. Slack has revolutionized the way we handle urgent support requests, providing a real-time communication platform that is integral to the rapid response and resolution of critical issues.  In this post, I'll dive into the guidelines and best practices

Are you looking for efficient ways to transcribe the call recordings of your Amazon Connect instance or custom contact center solution in AWS? In this video, our WebRTC Developer Advocate Hector Zelaya explores an approach that you can implement right away and doesn't require any big changes in your infrastructure: leveraging Amazon S3, Amazon Transcribe, and AWS Lambda to build a serverless transcription mechanism. https://www.youtube.com/watch?v=Lx3ttnbHLrg Our team at WebRTC.ventures are experts integrating AI features into

In today's fast-paced world, effective communication is the cornerstone of success. Whether you're integrating video into a user application or enhancing your own team’s communication efforts, ensuring you have a reliable and scalable platform is paramount. That's where WebRTC.ventures comes in, offering a cutting-edge solution designed to jumpstart your communication journey: the Video Call Starter Kit, powered by the Amazon Chime SDK. Streamlined for Modern Communication Imagine a virtual meeting platform tailored to your specific

Each month, we ask our dynamic team of WebRTC developers, mobile developers, designers, testers, DevOps experts, and project leads to share some of the impactful work they did in the past month. Here’s a short list of some that might interest you — at least the ones we can talk about! Architected a solution to provide SIP video capabilities to the Twilio Flex digital engagement center Attended the JanusCon 2024 conference in Italy and presented

The bustling streets of Naples and plentiful sunshine set the scene for an exciting start to the JanusCon 2024 conference, held at Centro Congressi Federico II. Amidst the chaotic morning traffic, I eagerly made my way to the venue, ready to delve into the world of RTC technology using Janus. Napoli, Photo by Francesco Baerhard Day 1: Morning Highlights The first day kicked off with a captivating overview of WebRTC fundamentals by Simon Pietro, setting the

In a prior post, I spoke about the importance of client-side monitoring your real-time application and introduced you to some of the more important elements, like events and browser metrics such as packet loss and bitrate. If your application is built with a CPaaS, most providers offer dashboards where customers monitor the performance of their WebRTC applications. Daily, a WebRTC.ventures CPaaS partner, stands out for its simple, yet powerful, approach for providing real-time communication features

Client-side monitoring is a crucial task in the development of any real-time communication application. In this post, we emphasize the importance of monitoring for WebRTC applications, highlighting its role in identifying and troubleshooting user-side issues.  What is Monitoring? Sometimes overlooked in the development of web applications, monitoring is a vital task alongside testing, especially in WebRTC applications. It aids in identifying and troubleshooting user-side issues that can disrupt meetings and cause poor media quality. When