WebRTC Platforms in 2026: 9 Picks for Real-Time Video, Voice and Screen Sharing

  • WebRTC is a transport standard, not a product. The platforms in this list differ dramatically on SFU architecture, TURN fleet density, SDK maturity, and observability , those gaps determine whether your call works at 500 concurrent participants or falls apart at 50.
  • Managed cloud APIs (Daily, Twilio, Agora, Vonage) trade control for reliability and speed to production. Open infrastructure (LiveKit, mediasoup) trades simplicity for full ownership of cost, topology, and data residency.
  • For AI voice agents specifically, server-side participant support and sub-200ms end-to-end latency matter more than any other spec. Only a few platforms in this list are actually built for that use case.
  • Recording, SIP/PSTN bridging, and moderation are not commodity features , the implementations differ enough to change your architecture decisions before you write a line of code.
  • Pricing models vary widely: per-participant-minute, per-room, flat monthly, or self-hosted compute. Running the right model for your traffic pattern can cut your real-time communications bill by 60% or more.

The best WebRTC platform for most production applications is LiveKit or Daily, depending on whether you want infrastructure control or a fully managed API. LiveKit’s open-source SFU with optional cloud hosting suits teams that need custom topologies, AI agent integration, or strict data residency. Daily suits teams that want enterprise-grade reliability and global routing without managing media servers. Agora, Vonage, and Twilio are strong choices when SIP/PSTN bridging or carrier-grade compliance is non-negotiable.


What Are You Actually Buying When You Pick a WebRTC Platform?

The browser’s WebRTC API handles peer-to-peer media negotiation. What it does not handle is everything that makes real-time video work at scale: routing media through firewalls, mixing or forwarding streams across dozens of participants, selecting the lowest-latency path across a global network, recording sessions, and recovering gracefully when a participant’s connection degrades. That is what you are buying from a WebRTC platform.

Every platform in this list runs a Selective Forwarding Unit, or SFU. An SFU receives each participant’s stream and forwards it selectively to other participants without decoding and re-encoding , this keeps latency low and scales better than older MCU architectures. The quality of the SFU implementation, the density of TURN servers (the relay nodes that punch through symmetric NAT and corporate firewalls), and the routing logic between them determine whether you get 80ms median latency or 400ms in practice.

The second decision is build vs. buy for the media server itself. Platforms like LiveKit and mediasoup publish open-source SFU code you can self-host. Platforms like Daily, Agora, and Twilio run proprietary infrastructure and charge per participant minute. Neither model is universally better. Your traffic volume, compliance requirements, and engineering capacity determine which makes sense.


How to Evaluate a WebRTC Platform Before You Commit: The Found On AI Infrastructure Stack Test

Most WebRTC comparisons look at feature checklists. That surfaces the wrong signal. The meaningful gaps between platforms live in six dimensions we call the Found On AI Infrastructure Stack Test:

  1. SFU topology: Does the platform run cascaded SFUs (better for large rooms), regional SFUs with intelligent routing, or a flat architecture? Cascaded SFUs matter when a single room exceeds 50 participants.
  2. TURN fleet density: How many global TURN regions does the vendor operate? Fewer than 10 regions means users in Southeast Asia, South America, or Africa will see significantly higher latency.
  3. SDK surface area: Are the web, iOS, Android, and React Native SDKs first-party and actively maintained, or are some community-maintained with months of lag? A stale mobile SDK means your iOS build breaks on OS updates.
  4. Server-side participant support: Can a server process join a room as a participant to inject audio, process streams, or run an AI model? This is the architectural requirement for voice agents and real-time transcription pipelines.
  5. Observability: Does the platform expose per-session latency, packet loss, jitter, and room-level analytics through a dashboard or API? Without this, production debugging is guesswork.
  6. Egress cost model: Is recording billed by the minute, by storage, or bundled? SIP/PSTN is it per-minute outbound or a flat connector fee? The pricing model matters as much as the per-participant rate.

Run every vendor through these six dimensions before you look at the pricing page. A platform that scores well on price but poorly on TURN density will cost you more in engineering time and user churn than you saved on API fees.


Which WebRTC Platforms Are Worth Evaluating in 2026?

1. LiveKit

LiveKit

LiveKit is an open-source SFU written in Go, purpose-built for the modern AI communications stack. Its distinguishing feature is the Agents framework: a server-side participant model that lets LLM-powered voice agents join rooms directly, process audio streams in real time, and inject synthesized speech without any client-side orchestration. This architecture has made LiveKit the default choice for teams building AI voice assistants and real-time transcription pipelines.

You can self-host LiveKit on your own infrastructure or use LiveKit Cloud, which handles global TURN, SFU orchestration, and recording. The open-source repo is actively maintained, with first-party SDKs for web (JavaScript/TypeScript), iOS (Swift), Android (Kotlin), React Native, Flutter, Unity, and server-side Python and Node. Screen sharing works natively across all SDK targets.

LiveKit Cloud pricing is consumption-based, billed per participant-minute with additional charges for recording egress. The company publishes its pricing at livekit.io/pricing. Self-hosting eliminates the per-minute cost but requires you to operate TURN servers, which is nontrivial at global scale. If you are building for AI voice agents, LiveKit is the strongest option available right now because its server-side participant model was designed for that use case from day one, not retrofitted.

2. Daily

daily

Daily has operated managed WebRTC infrastructure since 2016 and has positioned itself explicitly for global scale and AI integration. Its SFU network spans multiple regions, and it publishes latency and uptime data on a public status page. The REST API and Prebuilt UI components let teams embed video calling in hours rather than days.

Daily’s Pipecat open-source framework, released in 2024, makes it a credible competitor to LiveKit for AI voice agent pipelines. Pipecat handles the transport layer while you plug in your LLM, TTS, and STT of choice. Server-side bot support is first-class. Recording is available as a composited or individual track option, billed separately.

The platform supports SIP via partner integrations. It does not offer native PSTN bridging at the same depth as Twilio or Vonage. Pricing is published per participant-minute at daily.co/pricing, with a generous free tier that covers prototype and low-volume production use. For teams that want a fully managed WebRTC API with genuine AI-agent support and no infrastructure to operate, Daily is the clearest alternative to LiveKit Cloud.

3. Agora

agora

Agora operates one of the largest private real-time communications networks in the world, with points of presence spanning over 200 countries. That physical infrastructure is its primary competitive advantage over platforms that lease from cloud providers. In regions where AWS or GCP edge coverage is sparse , particularly Southeast Asia and parts of Africa , Agora consistently delivers lower median latency than competitors relying on public cloud backbones.

The Agora Video SDK supports web, iOS, Android, Electron, React Native, Flutter, and Unity. Agora also offers extensions for noise suppression, virtual backgrounds, face detection, and AI-powered video enhancement as purchasable add-ons. Its Signaling SDK provides data channels and presence independently of the video layer, which is useful when you need low-latency messaging alongside media. SIP connectivity is available through Agora’s SD-RTN (Software Defined Real-time Network) gateway.

Agora publishes per-minute pricing by resolution tier at agora.io/en/pricing. HD video costs more than audio-only or SD video, so the actual cost per participant-minute depends heavily on your resolution profile. Teams building global consumer apps with heterogeneous network conditions in emerging markets will find Agora’s network coverage difficult to match elsewhere.

4. Twilio Video (and Programmable Voice)

twilio video

Twilio Video brings carrier-grade PSTN integration and a mature compliance posture that few WebRTC platforms match. If your application needs to bridge real-time video with inbound or outbound phone calls, or if HIPAA and SOC 2 certifications are procurement requirements, Twilio’s existing compliance infrastructure shortens that process significantly. Twilio also offers Programmable Voice separately, which handles PSTN calls at scale through a different SDK path.

Twilio Video’s SFU (called Group Rooms) supports up to 50 participants per room. It does not support the very large rooms that Agora or LiveKit handle through cascaded SFU topologies. SDK coverage is solid across web, iOS, and Android. Twilio is not a strong choice for AI voice agent pipelines or for applications that need server-side stream processing , the architecture was built for human-to-human calling, not bot participants.

Pricing is per participant-minute, published at twilio.com/en-us/video/pricing. For telehealth platforms, compliance-heavy enterprise deployments, and any application that genuinely needs SIP/PSTN bridging baked into the same vendor relationship, Twilio remains a defensible choice despite being more expensive per minute than newer entrants.

5. Vonage Video API (formerly TokBox)

vonage

Vonage Video API (originally OpenTok from TokBox, acquired by Vonage in 2019, now part of Ericsson) has one of the longest production histories in the WebRTC space. The OpenTok SDK remains widely deployed in telehealth, telemedicine, and enterprise collaboration products built before 2020. Its session-based recording model and archive storage are battle-tested.

The platform’s Session Monitoring and Quality Stats APIs give operators reasonable observability into per-session conditions. SIP interconnect is available. The SDK surface covers web, iOS, Android, Windows, Linux, and macOS. Where Vonage underperforms relative to LiveKit or Agora is in server-side agent support and the pace of new feature development since the Ericsson acquisition. Teams with existing OpenTok deployments have a legitimate reason to stay; new teams evaluating from scratch have faster, more developer-friendly options.

6. Stream Video SDK

stream 1

Stream entered the WebRTC market from a messaging and activity feeds background, and that lineage shows in its developer experience. The Stream Video SDK is tightly integrated with Stream Chat, which means if your application needs real-time chat alongside video calling, you can manage both through a single SDK and a single API key. The combined video-plus-chat integration is genuinely faster to build than wiring separate SDKs together.

Stream Video supports web, iOS, Android, React Native, and Flutter with first-party SDKs. It offers composited recording, RTMP broadcast, and live streaming capabilities, though this article focuses on its multi-party calling use case. Screen sharing works across platforms. Server-side calling features and SIP bridging are more limited than Twilio or Vonage. Stream’s pricing model is published at getstream.io/video/pricing and includes a free tier with monthly active user-based pricing above the threshold.

7. Liveswitch

liveswitch compressed

Liveswitch (formerly Frozen Mountain) is the platform most developers have not evaluated but probably should if they have enterprise on-premises requirements. Liveswitch sells both a cloud-hosted version and a self-hosted server that runs on your own infrastructure, with a commercial license and professional support. The server supports Windows, Linux, and cloud-native Kubernetes deployments.

SDK coverage is unusually broad: web, iOS, Android, Windows, macOS, Linux, and even Xamarin. Data channels, screen sharing, and SIP integration are all supported. Liveswitch targets enterprise teams in financial services, healthcare, and government where data sovereignty requirements make it impossible to route media through a third-party cloud. Pricing is not published publicly , the company sells through custom enterprise contracts. If self-hosting with vendor support is your requirement, Liveswitch and LiveKit are the two most mature options.

8. Whereby Embedded

whereby

Whereby Embedded takes the opposite approach from every other platform in this list. Rather than exposing a low-level WebRTC SDK, Whereby gives you a fully built video room you embed via iframe or API into your product. Configuration happens through URL parameters and a REST API, not SDK calls.

This model has a specific fit: product teams that need video calling in their application but have no dedicated engineering resources to build and maintain a custom video UI. Telehealth platforms, tutoring marketplaces, and consultation tools with standardized call flows benefit from this approach. Teams that need custom call controls, server-side processing, or anything beyond a standard room layout will hit the limits of the embedded model quickly. Pricing is room-based rather than per participant-minute, published at whereby.com/pricing.

9. mediasoup

mediasoup compressed scaled

mediasoup is a Node.js library that provides SFU primitives , not a platform, not a managed service, not an SDK with UI components. You write the signaling layer, the room management logic, the TURN integration, and the media routing yourself. mediasoup gives you the lowest-level control of any option in this list, and that cuts both ways.

Teams that choose mediasoup typically have a specific technical requirement , unusual codec support, custom media processing pipelines, or extreme cost sensitivity at high volume , that no managed platform accommodates well. The operational burden is real: you need engineers with WebRTC and media server expertise to deploy and maintain a mediasoup-based system. For most product teams, LiveKit provides 90% of mediasoup’s flexibility with a fraction of the operational overhead. mediasoup is the right choice when LiveKit’s abstractions are genuinely in your way.


Side-by-Side: How the 9 Platforms Score on Key Dimensions

PlatformSFU ArchitectureWeb/Mobile SDKsScreen ShareComposited RecordingSIP/PSTNServer-Side AgentsSelf-Host OptionPricing Model
LiveKitCascaded, open-sourceFull (8 targets)YesYes (cloud)Via pluginNative (Agents framework)Yes (OSS)Per participant-minute or self-host compute
DailyManaged, regionalFull (6 targets)YesYesPartner integrationsYes (Pipecat)NoPer participant-minute
AgoraProprietary SD-RTNFull (7 targets)YesYesYes (SD-RTN gateway)LimitedNoPer minute by resolution tier
Twilio VideoManaged, group roomsWeb, iOS, AndroidYesYesYes (Programmable Voice)LimitedNoPer participant-minute
Vonage Video APIManaged (OpenTok)Full (6 targets)YesYes (archive)YesLimitedNoPer minute
Stream VideoManagedFull (5 targets)YesYesLimitedLimitedNoMonthly active users
LiveswitchSFU, MCU hybridFull (7 targets)YesYesYesLimitedYes (commercial)Enterprise contract
Whereby EmbeddedManagediframe/API onlyYesYesNoNoNoPer room/month
mediasoupDIY SFU primitivesNone (build yourself)DIYDIYDIYDIYYes (OSS)Self-host compute only

Which WebRTC Platform Is Best for Telehealth, Collaboration, AI Voice Agents, and Developer Control?

Generic rankings obscure the real signal. A platform that is right for a telehealth startup building one-to-one consults is wrong for a team embedding an AI sales agent that needs to join 200 concurrent calls. The matrix below maps each major use case to the platforms that are genuinely built for it , not just capable of it.

Use CaseBest ChoiceSecond ChoiceWhy
Telehealth (HIPAA, 1:1 or small group)Twilio Video or VonageDailyTwilio and Vonage carry existing BAA agreements and compliance documentation that accelerate enterprise procurement. Daily is faster to build on but requires more compliance legwork.
Team collaboration / conferencing (10-50 participants)Daily or AgoraLiveKit CloudDaily’s SFU handles mid-size rooms with excellent adaptive bitrate and noise suppression. Agora wins in regions where its private network outperforms public cloud routing.
AI voice agents (real-time LLM conversation)LiveKitDaily (Pipecat)LiveKit’s Agents framework was designed from scratch for server-side AI participants. Pipecat on Daily is a strong alternative but involves a separate open-source dependency.
Consumer apps in emerging markets (global latency)AgoraLiveKit CloudAgora’s private SD-RTN consistently delivers lower latency in Southeast Asia, South Asia, and Sub-Saharan Africa than competitors relying on AWS or GCP edge alone.
Rapid prototyping / no engineering resourcesWhereby EmbeddedDaily PrebuiltWhereby requires near-zero WebRTC knowledge to ship. Daily Prebuilt is slightly more flexible but still far faster than a custom SDK integration.
Full infrastructure control / data sovereigntyLiveKit (self-hosted)LiveswitchLiveKit’s open-source SFU runs on your own Kubernetes cluster with no media touching a third-party network. Liveswitch adds vendor support and a commercial license if you need SLAs.
SIP/PSTN bridging at scaleTwilioVonageTwilio’s Programmable Voice network handles billions of PSTN minutes annually. No WebRTC-first platform comes close to that carrier footprint.
Video + chat in a single SDKStream VideoDailyStream’s activity feed and messaging SDKs share the same authentication and channel model as Stream Video, which eliminates a significant integration surface area.

Self-Hosted vs. Managed WebRTC: What Does the Cost Difference Actually Look Like?

Consider a mid-size SaaS team running 10,000 participant-minutes per day , roughly 500 thirty-minute calls with two participants each. At a managed platform rate of $0.004 per participant-minute (a common starting rate for HD video), that is $40 per day or about $1,460 per month. At scale , say, 10 times that volume , you are looking at $14,600 per month before recording and TURN egress charges.

Self-hosting LiveKit on three regional cloud instances (one in North America, one in Europe, one in Asia-Pacific) runs substantially less in raw compute, but you need to factor in engineering time to maintain the deployment, update the SFU software, monitor TURN relay capacity, and handle incident response. A realistic estimate for a team without prior media server experience is 0.5 to 1 full-time engineer equivalent of ongoing operational burden. At senior engineer compensation rates, self-hosting only wins financially at sustained volumes above roughly 5 million participant-minutes per month for most teams , though that break-even point shifts based on your team’s existing infrastructure expertise.

This scenario is illustrative, but the structure of the decision is consistent: self-hosting is a cost play that only pays off at high volume and with engineering capacity to absorb it. Teams evaluating AI voice agent costs should also check our analysis of AI voice agents for customer support, where the compute cost model follows similar logic.


What Do Data Channels and Moderation Features Add to a WebRTC Platform?

Data channels in WebRTC let you send arbitrary binary or text data between participants over the same peer connection that carries audio and video. Platforms expose these with varying levels of abstraction. LiveKit’s Data Channel API lets you broadcast or send targeted messages to specific participants with millisecond latency, which is how collaborative whiteboarding, cursor sharing, and synchronized state typically work in a custom implementation. Daily exposes similar primitives. Agora separates data messaging into its Signaling SDK rather than bundling it with media.

Moderation , muting participants, removing users from rooms, locking rooms, and applying content filters , matters more as room sizes grow. LiveKit supports server-side room management through its REST API and Go/Python server SDKs, meaning you can mute a participant or eject a user from your backend without any client action. Twilio Video’s REST API does the same. Whereby handles moderation through a simplified host/guest model that works for most small-group use cases but cannot replicate the programmatic control a custom SDK integration offers.


How Much Does a WebRTC API Cost Per Participant Minute?

Pricing across these platforms is not directly comparable because the billing models differ. The table below maps publicly listed starting rates as of their respective pricing pages. Actual costs depend on resolution tier, recording, TURN egress, and enterprise negotiation.

PlatformPricing ModelPublished Starting RateFree Tier
LiveKit CloudPer participant-minuteSee livekit.io/pricingYes
DailyPer participant-minuteSee daily.co/pricingYes (2,000 min/month)
AgoraPer minute, by resolutionSee agora.io/en/pricingYes (10,000 min/month)
Twilio VideoPer participant-minuteSee twilio.com/en-us/video/pricingNo (pay-as-you-go from first minute)
Vonage Video APIPer minuteContact salesLimited developer credits
Stream VideoMonthly active usersSee getstream.io/video/pricingYes
LiveswitchEnterprise contractNot publishedTrial available
Whereby EmbeddedPer room/monthSee whereby.com/pricingYes (1 room)
mediasoupSelf-host compute cost only$0 licensingN/A

Agora is notable for offering 10,000 free participant-minutes per month, which is enough to run a meaningful beta. Twilio’s absence of a free tier means even test traffic generates a bill, though the amounts are small at low volume. For teams building AI voice products, the cost of the WebRTC transport layer is often secondary to LLM inference and TTS costs , context worth keeping in mind when you see AI voice generator pricing compared to transport pricing.


Does Google Meet Use WebRTC, and Does That Matter for Your Decision?

Yes, Google Meet runs on WebRTC, as does Zoom’s browser client, Microsoft Teams’ web client, and most modern video conferencing tools. This is sometimes cited as evidence that “WebRTC just works” , which misreads the situation. Google operates one of the largest private media routing networks in the world to make Meet reliable. The browser standard is the same one your application uses. The infrastructure behind it is not.

The relevant lesson for platform selection is that sub-100ms median latency in production requires either a dense private TURN and relay network (Agora’s model) or intelligent routing across cloud regions (LiveKit Cloud and Daily’s model). Browser WebRTC with a single STUN server and no SFU will give you peer-to-peer calls that break the moment either participant is behind a symmetric NAT or a restrictive corporate firewall. TURN servers handle relay in those cases, and their geographic distribution directly determines your worst-case latency.


How Does LiveKit’s Open-Source Infrastructure Differ From Fully Managed WebRTC APIs?

The architectural difference is meaningful and worth understanding before you commit. A fully managed API like Daily or Agora runs proprietary media servers that you access through SDK calls. You never see the SFU code, you cannot customize the media processing pipeline, and your data traverses the vendor’s network. In return, you get a service with SLAs, global TURN management, and a support team.

LiveKit’s open-source SFU (Apache 2.0 licensed) means you can read, fork, and modify the media routing code itself. You can run it in a VPC with no media leaving your network. You can add custom media processing steps , noise filtering, packet inspection, custom codec handling , at the SFU layer. The Agents framework lets you write server-side participants in Python or TypeScript that process real audio streams, not just post-hoc recordings. This is why teams building serious AI communications products , voice agents that interrupt, respond, and adapt to real-time audio , tend to converge on LiveKit.

If you’re comparing real-time video API options more broadly beyond WebRTC-native platforms, the best video API platforms for developers covers adjacent options including Mux and Cloudflare Stream that handle one-to-many delivery rather than multi-party real-time calls.


Frequently Asked Questions About WebRTC Platforms

What is the difference between a WebRTC SDK and a WebRTC API?

A WebRTC API is the HTTP-based interface you call from your server to manage rooms, participants, tokens, and recordings. A WebRTC SDK is the client-side library , JavaScript, Swift, Kotlin , that handles media capture, peer connection setup, SFU negotiation, and stream rendering inside your application. Most platforms provide both. When developers say “WebRTC API,” they often mean the full platform including both the server API and client SDK. The browser also exposes its own native WebRTC API for raw peer connection control, which is what MDN documents at developer.mozilla.org.

Is WebRTC free to use?

The WebRTC browser standard is free and open-source. Running a production WebRTC application is not free: you need STUN and TURN servers to handle NAT traversal, an SFU to handle multi-party rooms, and global infrastructure to achieve acceptable latency. Managed platforms like Daily and Agora offer free tiers sufficient for development and low-volume production. Self-hosting LiveKit or mediasoup eliminates licensing costs but introduces infrastructure and engineering overhead. The “free” framing breaks down the moment you need more than two participants or users in constrained network environments.

Which WebRTC provider handles global scale best?

Agora has the most geographically distributed private relay network, with coverage in markets where AWS and GCP edge capacity is limited. For teams with a globally uniform user base, LiveKit Cloud and Daily both deliver strong results through intelligent routing across cloud regions. The meaningful variable is whether your users are concentrated in well-served markets (North America, Western Europe, Japan, Australia) or include significant traffic from Southeast Asia, India, Sub-Saharan Africa, or Latin America , in the latter case, Agora’s private network typically outperforms cloud-routed alternatives on median latency.

What WebRTC platform should I use for building AI voice agents?

LiveKit is the strongest choice for AI voice agents because its Agents framework provides native server-side participant support , your AI model joins a room as a first-class participant, receives audio in real time, and injects synthesized speech without client-side orchestration. Daily’s Pipecat framework offers similar capabilities on top of Daily’s managed infrastructure and is a viable alternative. Platforms like Twilio, Agora, and Vonage were not architected for server-side AI participants and require significantly more custom integration work to achieve the same result. For background on the broader AI voice agent space, see our coverage of AI voice agents for customer support.

Does a WebRTC platform need to be HIPAA compliant for telehealth?

For US telehealth applications, the platform must be willing to sign a Business Associate Agreement (BAA), and your implementation must handle PHI appropriately , no recording to unsecured storage, access controls on room tokens, audit logging. Twilio and Vonage have established BAA processes. Daily offers HIPAA-eligible configurations. Agora’s HIPAA posture requires direct inquiry with their enterprise team. Self-hosting LiveKit gives you full control over data handling but does not come with a BAA , you are the covered entity and the business associate simultaneously. For HIPAA-compliant video storage considerations, see our guide on HIPAA-compliant video hosting platforms.

What is the difference between an SFU and an MCU in WebRTC?

An SFU (Selective Forwarding Unit) receives each participant’s media stream and forwards them individually to other participants without decoding. This keeps CPU load low and latency minimal, but each recipient’s client must decode multiple incoming streams. An MCU (Multipoint Control Unit) decodes all streams, mixes them into a single composite stream, and sends one stream to each participant , simpler for the client but computationally expensive and higher latency on the server. All modern WebRTC platforms use SFU architectures or SFU/MCU hybrids for recording and layout composition. Pure MCU deployments are rare in 2026 outside legacy enterprise video conferencing hardware.

How do I choose between self-hosted and managed WebRTC infrastructure?

Self-hosting (LiveKit OSS, mediasoup, Liveswitch) makes financial sense when you exceed roughly 5 million participant-minutes per month and have engineering capacity to operate media servers. Below that threshold, the per-minute cost of managed platforms is almost always less than the engineering overhead of running and scaling your own SFU fleet. Data residency requirements , where regulations mandate that media not leave a specific jurisdiction , are the other primary reason to self-host, regardless of volume. If neither cost at scale nor data residency is a constraint, a managed platform is almost always the right starting point.


The Decision That Actually Matters Before You Write Code

Most teams pick a WebRTC platform based on which SDK example compiles first. That is the wrong input. The decision that shapes your architecture for years is whether you want a managed network you operate through API calls, or media infrastructure you control at the code level. Every other dimension , price, SDK ergonomics, recording quality , is secondary to that one.

If you are building an AI-native communications product , voice agents, real-time transcription, LLM-in-the-loop call processing , start with LiveKit. The Agents framework is two to three years ahead of what any managed platform has shipped for server-side AI participant support. If you need to ship a reliable video calling feature in weeks and your team has no media server expertise, Daily or Agora will serve you better than any amount of open-source infrastructure.

The WebRTC commodity thesis , that managed platforms sell identical transport , is wrong in practice. SFU code quality, TURN fleet density, global routing intelligence, and observability tooling create measurable differences in call quality at production traffic levels. Evaluate on those dimensions. The platform that wins your proof of concept will probably run your production traffic too, so get the evaluation right the first time. For teams who also need to evaluate adjacent real-time communications tooling, our comparison of AI meeting assistants and notetakers covers the layer that sits above the WebRTC transport.

Jason C
Jason C