Top 8 WebRTC Companies in 2026

WebRTC has become a central framework responsible for high-speed video, audio, and data transfer on the internet.

Though developers can build apps using only the raw WebRTC protocol, many will likely run into problems like dropped calls, laggy videos, and users who can't connect due to office firewalls. WebRTC companies handle many of the technicalities associated with the technology, so developers can focus their efforts where it really matters.

Here are the top WebRTC development companies in 2026 and how to choose the best one for your organization.

What Is WebRTC?

WebRTC (Web Real-Time Communication) is an open-source framework for peer-to-peer video, audio, and data sharing directly between browsers, no plugins required.

A few things worth knowing before diving in:

• WebRTC handles media capture, encoding, and transmission natively.

• An initial server handshake (signaling) is required before peers connect directly.

• WebRTC has no standard signaling protocol; providers are responsible for building this themselves.

• Quality, reliability, and security vary significantly by implementation.

That last point is why choosing the right WebRTC company matters.

Let's look at the best ones.

Top 8 WebRTC Development Companies

The providers below represent the strongest options available today across infrastructure quality, developer experience, and pricing.

Here's a closer look at what each one offers.

Stream

Stream offers a WebRTC-powered video and audio API built for teams that need production-ready infrastructure fast, without sacrificing scalability. 

It's particularly strong for developer experience. Prebuilt UI components, cross-platform SDKs, and a managed backend mean most teams can go from zero to a working integration in a matter of days.

Key Features

• Dynascale: Dynamically adjusts video resolutions up to 1080p and switches codecs like AV1, VP9, and H.264 based on each participant's network and device conditions.This means every viewer gets the best quality their connection can support, automatically.

• SFU cascading: For very large calls, Stream cascades multiple SFUs together, distributing participants across servers while maintaining real-time communication between them. This is what allows Stream to scale to massive participant counts without architectural changes.

• Automatic subscription management: Stream's SDKs automatically handle subscribing to the right video streams. If a participant isn't visible on screen, their video isn't downloaded. This significantly reduces bandwidth consumption in large calls.

• Opus RED and DTX: Redundant audio encoding handles packet loss, while DTX skips silence to save bandwidth. This keeps audio crisp even on unreliable connections.

• Prebuilt UI components: Ready-made, customizable components for web, iOS, Android, Flutter, and React Native, so teams don't have to build call UI from scratch.

• Multi-layer redundancy: Multi-datacenter failover and auto-scaling handle spikes to 100,000+ participants at 225 Gbps peak with zero API failures and 30 FPS stability.

Pricing

Note: Stream also offers a free tier for development and testing, with no upfront costs. Pricing scales with usage, and enterprise plans are available for high-volume deployments.

Bottom Line

Stream is one of the strongest options for teams that want serious infrastructure without the serious overhead. The combination of Dynascale, SFU cascading, and genuinely developer-friendly SDKs makes it suited for everything from 1:1 video calls to large-scale live events. It's also a natural fit for product teams already using Stream's Chat or Feed APIs, since the video layer integrates directly with the same platform.

Agora

Agora is one of the most established WebRTC companies. Its core differentiator is the proprietary global network it has built on top of WebRTC, making it especially strong for cross-border, high-concurrency deployments.

Key Features

• Software-Defined Real-Time Network (SDRTN®): Agora's algorithms monitor network conditions in real time and automatically select the most efficient routing path, targeting latency of 400ms or less. This is what sets Agora apart from most competitors; raw WebRTC struggles internationally, and SDRTN is purpose-built to solve that.
• Broad SDK support: SDKs available for web, iOS, Android, React Native, Flutter, Unity, Unreal Engine, and more, making it one of the widest platform footprints in the market.
• Cloud recording and transcription: Save and transcribe sessions without managing your own storage or pipeline.
• AI-powered audio/video enhancement: Machine learning is applied across the full pipeline, from capture to playback, to maintain performance quality even in poor network conditions.
• Real-time messaging: Data channels and chat run alongside audio/video within the same session, no separate integration required.

Pricing

Note: Agora charges for most features individually (analytics, whiteboards, and chat all carry separate fees), which can make costs difficult to predict as usage scales.

Bottom Line

Agora is a strong choice for teams building products that need to perform consistently across international markets. Its infrastructure advantage is real and well-documented. The tradeoffs are cost complexity and a steeper developer experience. If your use case is domestic or mid-scale, there are simpler and cheaper options on this list.

LiveKit

LiveKit is an open-source WebRTC infrastructure platform. What sets it apart from most providers on this list is its deployment model: the server is fully open source under the Apache 2.0 license, meaning teams can self-host at no cost or use LiveKit Cloud for a managed experience, with identical APIs either way.

Key Features

• Open-source SFU server: LiveKit's core is an SFU that routes audio and video streams without decoding or re-encoding them, keeping latency low and CPU usage minimal. And because it's open source, teams can inspect, fork, and customize the entire stack.
• Flexible deployment: Self-host for full infrastructure control, or use LiveKit Cloud for faster setup, reduced maintenance, and automatic scaling. Switching between the two requires no code changes.
• Broad SDK support: Client SDKs for JavaScript, Swift, Android, Flutter, React Native, Unity, Rust, Python, Go, and more, plus server SDKs for backend integrations.
• AI-native Agents framework: LiveKit Agents lets developers add Python or Node.js programs as full real-time participants in any room, processing audio, video, and data streams with integrations for major LLM, STT, and TTS providers.
• End-to-end encryption: Connections use 256-bit TLS encryption, media streams are encrypted with AES-128, and all data at rest is encrypted with AES-256.
• Horizontal scaling: Built on Redis and Kubernetes, LiveKit scales across nodes to support up to 100,000 concurrent participants per session.

Pricing

LiveKit is free to self-host. For LiveKit Cloud, three managed plans are available:

Note: Beyond plan allotments, additional usage is billed at $0.0005 per connection minute and $0.12 per GB of downstream bandwidth, with volume discounts available. Enterprise plans are available for larger deployments.

Bottom Line

LiveKit is the best choice for teams that want infrastructure-level control or are building AI-native products. The open-source model means no vendor lock-in and no surprise costs at scale if you're willing to manage your own servers. For teams that aren't, LiveKit Cloud offers the same flexibility with a managed layer on top. The tradeoff versus more turnkey options like Stream or Daily is that LiveKit requires more engineering investment to get production-ready; there are no prebuilt UI components.

Daily

Daily has been providing WebRTC infrastructure since 2016, and its team includes authors of WebRTC standards, video technology pioneers, and engineers with experience in codecs, real-time networking, and infrastructure. That pedigree shows. Daily is one of the most technically deep options on this list, with a particularly strong focus on fast integration and AI-native voice applications.

Key Features

• Prebuilt UI and flexible SDKs: Daily's PreBuilt option allows developers to integrate simple video chat into a webpage with only a few lines of JavaScript, ideal for rapid prototyping, while the full Client SDK enables more customized builds for web, iOS, and Android.
• Managed SFU infrastructure: Global, fully managed SFU network with support for up to 100,000 participants, ensuring low latency and high-quality media delivery worldwide.
• Recording, screen sharing, and live transcription: Core collaboration features are available out of the box, with straightforward per-minute pricing for add-ons.
• Participant-minute pricing model: Daily calculates participant-minutes rather than subscriber-minutes, which is a factorial calculation used by some competitors. This means Daily counts roughly half the minutes in multi-party calls.
• Pipecat — open-source AI framework: Daily created and maintains Pipecat, a vendor-neutral, open-source framework for building voice and multimodal AI agents, allowing developers to chain together services for transcription, language model inference, and voice generation from providers.
• HIPAA, SOC-2, and GDPR compliance: Enterprise-grade compliance available out of the box, making it a practical choice for healthcare and regulated industries.

Pricing

Daily offers a free tier of 10,000 participant minutes per month. Beyond that:

Bottom Line

Daily is a good choice for teams that want fast time-to-launch without sacrificing infrastructure quality. The prebuilt UI lowers the barrier to a working prototype significantly. And for teams building voice AI products, Pipecat is a compelling reason to choose Daily. The main limitation is less infrastructure control compared to LiveKit, and fewer enterprise customization options than Vonage.

Vonage

Vonage Video API is one of the longest-standing WebRTC platforms on the market. It's built on open WebRTC standards with support for a broad set of client SDKs, including iOS, Android, Windows, Mac, Linux, React Native, and Web.

Today, it sits within the broader Vonage communications ecosystem, which gives it an advantage for organizations that need video to work alongside voice, messaging, and telephony under one platform.

Features

• SIP interoperability: One of Vonage's most unique capabilities. It enables WebRTC sessions to connect with traditional telephony systems via SIP, meaning participants on a desk phone or in a legacy call center environment can join the same session as WebRTC users. Few competitors offer this out of the box.
• Scale: Vonage Video API supports up to 15,000 participants in real-time, plus live streaming to an unlimited number of viewers via HLS, with delivery to social platforms including Facebook Live, Twitch, and YouTube Live.
• Post-call transcription and AI summaries: Post-call transcription is now generally available, using AI to generate transcripts of recorded video calls with channel-specific audio, word-level timing, and both raw and formatted transcript segments, alongside AI-generated summaries of key takeaways.
• Scalable Video Coding (SVC) with VP9: VP9 with SVC is now generally available for routed sessions, allowing scalable layers within a single stream for more efficient bandwidth usage across varying network conditions.
• Compliance and security: HIPAA, SOC2, and GDPR compliant, with always-on encryption, making it a practical fit for healthcare, finance, and other regulated industries.
• Integrated developer tooling: Session insights, quality metrics, a developer playground, and a consolidated reporting API for usage and billing across all Vonage products in one place.

Pricing

Note: New customers receive 100,000 free minutes to start.

Bottom Line

Vonage is a good choice for enterprises that need WebRTC to interoperate with existing telephony infrastructure, or for regulated industries where compliance and security are non-negotiable. The depth of features reflects years of enterprise-focused development. It's not the fastest path to a working integration, but for organizations with complex communication environments, it may be the most complete one.

Twilio

Twilio is best known as the company that made SMS and voice APIs accessible to developers, but its video offering has a complicated recent history. In 2024, Twilio announced it would sunset Programmable Video, then reversed course entirely, confirming the product would remain and receive continued investment. The uncertainty drove many developers to migrate away, and that's worth factoring in when evaluating it for the long term.

That said, the case for Twilio Video today is less about video in isolation and more about the broader ecosystem it sits within.

Key Features

• Omnichannel ecosystem: Twilio video lives alongside voice, SMS, WhatsApp, email, and RCS on a single platform. Its platform is built to incorporate omnichannel communications alongside authentication, intelligent automation, and predictive insights. For organizations already using Twilio for other channels, adding video is a natural extension.
• SFU-based group rooms: Twilio uses an SFU architecture, receiving media streams from all participants and selectively forwarding them to others without mixing or processing.
• Real-time transcription: The JavaScript SDK now supports live in-call transcription via a simple event listener, making it easier to build accessibility features or AI-assisted call summaries.
• Noise cancellation: Third-party noise cancellation (via Krisp) is supported natively in the SDK for cleaner audio in noisy environments.
• Diagnostics and network quality tools: Twilio provides an open-source diagnostics app, a Preflight API for testing connectivity, and an RTC Diagnostics SDK for testing participant devices, microphones, speakers, and cameras before joining a call.
• Flex integration: For contact center use cases, Twilio Video integrates with Twilio Flex, enabling video calls as part of a broader customer service workflow alongside voice and messaging channels.

Pricing

Note: Standard Group Rooms support up to 50 participants, well below competitors like Vonage (15,000) or Stream (100,000).

Bottom Line

Twilio Video makes the most sense for teams already deeply embedded in the Twilio ecosystem. As a standalone WebRTC provider, it's harder to recommend: the 50-participant room limit is a significant ceiling, it lacks prebuilt UI components, and the product's near-sunset experience has understandably shaken developer confidence.

100ms

100ms is a live video infrastructure platform built by engineers who scaled video at Meta and Disney+ Hotstar. And that heritage is evident in the product's design philosophy. Rather than offering the most features, 100ms focuses on making the complex parts of WebRTC feel simple: a single SDK handles both video conferencing and HLS live streaming.

Key Features

• Single SDK for conferencing and streaming: 100ms combines video conferencing built over WebRTC, live streaming built over HLS, and real-time databases in a single SDK. This removes the need to stitch together multiple services for different use cases.
• Rich out-of-the-box functionality: Features available without additional configuration include screen sharing, recording, whiteboard and PDF annotation, polls and quizzes, emoji reactions, post-call transcription, AI-generated summaries, and RTMP in and out.
• Prebuilt UI components: A low-code "Prebuilt" UI option lets teams embed a working video experience with minimal code, while the full SDK allows custom UI builds across Android, iOS, Flutter, React Native, and React.
• Role-based room management: Granular participant roles (host, viewer, moderator) can be configured on the dashboard and adjusted in real time without code changes, making it suited for structured formats like classrooms and webinars.
• Scale: 100ms supports up to 10,000 attendees, with the ability to add up to 1,000 hosts to a video conference without compromising audio-video quality.
• Compliance: HIPAA and SOC-2 compliant, with enterprise-grade analytics included.

Pricing

Note: 100ms offers 10,000 free conferencing minutes per month.

Bottom Line

100ms works well for teams building structured, interactive video experiences, like edtech platforms, telehealth tools, virtual events, and fitness apps. The single-SDK approach and built-in feature set reduce integration complexity compared to providers where features must be pieced together from separate services. The main limitation to be aware of is that 100ms does not operate its own global network of media nodes, which can affect latency consistency at scale across geographically distributed audiences.

Dyte

Dyte is a developer-first video SDK built around a single core idea: get from zero to a production-ready video integration as fast as possible. Its prebuilt UI templates let developers add a working video call in fewer than ten lines of code, while still offering complete control over layout and permissions for teams that need more customization.

Key Features

• Prebuilt UI kit with deep customization: Dyte offers pre-configured UI templates and customizable brand kits alongside configurable permission settings, meaning teams can match the video experience to their product's look and feel without building from scratch.
• All-in-one SDK: The SDK includes out-of-the-box support for chat, polls, screen sharing, and plugins across PC, mobile, tablet, and web, covering the collaboration layer without requiring separate integrations.
• Recording and transcription: Recording supports customizable layouts, multi-track recording, automatic transcription, and secure cloud storage integration.
• Live streaming: Dyte supports broadcasting to large audiences with global infrastructure, multi-host streams, and re-streaming to platforms like Twitch and YouTube.
• Broad SDK support: Compatible with React, Angular, Flutter, Kotlin, Swift, React Native, Android, and iOS — covering web, mobile, and desktop with consistent APIs.
• Compliance: SOC-2, GDPR, and HIPAA compliant, with 99.99% uptime SLA.

Pricing

Note: Dyte offers 10,000 free participant minutes each month on the full platform.

Bottom Line

Dyte is the right choice for product teams that want a polished, branded video experience without a lengthy build cycle. The UI kit is genuinely well-designed, the pricing is transparent, and the all-in-one SDK removes a lot of the integration work that other providers leave to the developer. It's a less obvious fit for teams with very high scale requirements or those building highly custom infrastructure.

Features vs. Infrastructure: How to Choose a WebRTC Company

For most businesses, feature parity matters less than infrastructure philosophy. What WebRTC companies ultimately provide is the underlying system for real-time data transfer, and most features your product needs can be built on top of that foundation.

Infrastructure, on the other hand, is much harder to retrofit once you've scaled.

There is no universally "best" provider, only the right fit for your product's technical and operational constraints. Teams that evaluate pricing models, scaling behavior, and long-term ownership costs early tend to avoid expensive migrations down the road.

Frequently Asked Questions

1. Which companies use WebRTC?

WebRTC powers real-time communication across a huge range of products. Google Meet, Microsoft Teams, and WhatsApp all run on WebRTC, as do Discord, Facebook Messenger calls, and many telehealth and edtech platforms. Effectively, most real-time video and audio experiences you encounter on the web today are built on WebRTC in some form.

2. Is WebRTC owned by Google?

Not exactly. Google open-sourced the foundational technology in 2011 after acquiring Global IP Solutions in 2010, and then engaged with standards bodies at the IETF and W3C to ensure industry consensus. WebRTC specifications are now published by the W3C and IETF, and are supported by Apple, Google, Microsoft, Mozilla, and Opera. Google maintains the open-source project, but no single company owns WebRTC.

3. Is WebRTC faster than RTMP?

Yes, meaningfully so. WebRTC is designed for sub-second, interactive communication. Typical latency is under 500ms. RTMP (Real-Time Messaging Protocol) is a streaming protocol optimized for broadcast delivery, with latency typically ranging from 3–30 seconds. If you need two-way, real-time interaction, WebRTC is the right choice. If you're broadcasting one-to-many with no interactivity required, RTMP may be more practical.

4. Are WebSockets better than WebRTC?

They serve different purposes. WebRTC is purpose-built for real-time media; it handles audio/video encoding, adaptive bitrate, echo cancellation, and peer-to-peer connections natively. WebSockets are a general-purpose, bidirectional messaging protocol better suited to signaling, chat, or data sync. For audio/video applications, WebRTC and WebSockets are often used together, with WebSockets handling the signaling layer and WebRTC carrying the media.

5. Is Zoom based on WebRTC?

Mostly no, though the answer has become more nuanced. Zoom's proprietary video stack uses its own implementation of the H.264 codec rather than standard WebRTC, which historically gave it performance advantages in native apps. With the release of Video SDK v2, Zoom added WebRTC support to its web stack, but its core platform remains largely proprietary. Most WebRTC providers covered in this article are fully WebRTC-native.