How to Build a Video Conferencing App Like Zoom: What Product Owners Need to Know
- Riya ThambirajBuild & ShipLast updated on
Summary
To build a video conferencing app like Zoom, use WebRTC for peer-to-peer video (up to 4-6 participants) or a media server like mediasoup or Janus for larger groups. An MVP with 1-on-1 and group video calls, screen sharing, and recording takes 14-22 weeks and costs $100K-$220K. For most vertical SaaS products, embedding a third-party video SDK (Daily.co, Agora, Twilio Video) is faster and cheaper than building native WebRTC infrastructure.
Key Takeaways
WebRTC is the right technology for video calls, but it is not a drop-in solution. Browser support, NAT traversal (STUN/TURN servers), and codec negotiation require careful implementation.
For groups larger than 4-6 participants, peer-to-peer WebRTC breaks down. You need a media server (SFU architecture) to route video streams -- which adds significant infrastructure cost.
Recording is not a free feature. Server-side recording requires dedicated recording infrastructure and significant storage. Client-side recording is simpler but limited.
For most product teams, embedding a third-party video SDK (Daily.co, Agora, Twilio) and focusing on your product differentiation is better than building WebRTC infrastructure from scratch.
The real product opportunity in video conferencing is vertical context: telehealth platforms with EHR integration, tutoring platforms with whiteboard tools, legal platforms with session recording and e-signature.
Building a video conferencing app is one of those projects that looks straightforward until you are 8 weeks into it and your calls drop when users are behind corporate firewalls.
The good news: the technology is mature, well-documented, and available as both open-source infrastructure and commercial SDKs. The question is not whether you can build it. The question is whether you should build it from scratch, or focus your engineering resources on what makes your product different.
The decision: Build vs. embed
This is the first fork in the road.
Build on WebRTC infrastructure if:
Video conferencing is the core product (you are building a competitor to Zoom in a specific vertical)
You need complete control over media routing, data sovereignty, or specific compliance requirements
You will handle very high call volumes where SDK per-minute pricing becomes prohibitive
Embed a third-party video SDK if:
Video is one feature of a larger product (telehealth platform, tutoring app, legal SaaS)
You need to ship in weeks, not months
Your differentiation is in the surrounding product, not the video layer itself
Most product teams building vertical SaaS should embed a video SDK. Agora, Daily.co, Twilio Video, and 100ms all provide reliable video infrastructure with well-documented SDKs. You pay per-minute usage but save 3-6 months of engineering time.
This guide covers both paths.
How video conferencing actually works
WebRTC basics
WebRTC is the open standard that powers browser-based and mobile video. It handles:
Media capture: Accessing camera and microphone
Codec negotiation: Agreeing on video/audio formats with the other party
NAT traversal: Getting through firewalls and routers to establish a connection
Data transfer: Sending encoded media between peers
For 1-on-1 calls and small groups (2-4 participants), WebRTC establishes a peer-to-peer connection directly between browsers. This is bandwidth-efficient and has low latency.
The group call problem
When you add more participants, peer-to-peer breaks down. With 10 participants in a peer-to-peer mesh, each participant needs to upload their video stream to 9 other people simultaneously -- 9 outbound streams from every user. This is unsustainable.
The solution is a Selective Forwarding Unit (SFU): a server that receives one stream from each participant and selectively forwards streams to the others. Participants upload once; the SFU handles distribution.
Running an SFU is real infrastructure: servers, bandwidth, redundancy. mediasoup, Janus, Livekit, and Jitsi are popular open-source SFU options. This is where video infrastructure gets expensive to operate.
STUN and TURN servers
STUN servers help users discover their public IP address so WebRTC can attempt direct connections. Most connections succeed this way.
When they don't (corporate firewalls, symmetric NAT), TURN servers relay traffic. TURN servers are the expensive part: they handle full bidirectional media relay, generating significant bandwidth costs. At scale, TURN relay costs can exceed hosting costs.
For a small-scale app, Twilio STUN/TURN or coturn self-hosted works fine. At scale, this becomes a significant cost center.
Core features for a video conferencing MVP
Meeting creation and scheduling
Create a meeting room (generate a unique URL or room code), schedule for a specific time, invite participants by email or link. Calendar integration (Google Calendar, Outlook) is v2 -- for v1, email invitations with the meeting link work.
Audio and video controls
Mute/unmute microphone, enable/disable camera, speaker volume control, device selection (which camera, which microphone). These controls must be reliable and fast. Laggy audio controls frustrate users immediately.
Screen sharing
Share the entire screen, a specific window, or a browser tab. WebRTC supports this natively via getDisplayMedia(). Screen sharing needs quality management -- sharing a full 4K display to 10 participants is bandwidth-heavy. Apply resolution limits (1080p or lower) for group calls.
Chat
In-call text chat for links, questions, and notes. This is much simpler than a full messaging implementation -- just session-scoped messages with no persistence required after the call.
Participant management
See who is in the call, mute participants (host controls), remove participants, admit participants from a waiting room. Waiting rooms are a v1 feature for any platform where host control matters (telehealth, education, legal consultations).
Recording
Recording is harder than it sounds. Client-side recording captures what one participant sees -- simple but limited. Server-side recording captures the full session, mixes audio/video from all participants, and stores it reliably. Server-side requires dedicated recording infrastructure and generates significant storage costs. Plan your recording storage and retention policy before launch.
The AI layer in 2026
The video conferencing features that differentiate vertical platforms today are AI-powered:
Live transcription: Real-time speech-to-text using Whisper or Deepgram. Session transcripts available after the call.
Meeting summaries: LLM-generated post-call summaries highlighting decisions, action items, and key discussion points.
Speaker identification: Attribute transcript lines to specific participants by voice.
Noise cancellation: AI background noise suppression at the media layer. Krisp and NVIDIA RTX Voice APIs handle this well.
Background blur and virtual backgrounds: Segmentation models that separate the speaker from their background. Available via MediaPipe and similar libraries.
These are compelling differentiators for vertical platforms. A telehealth app with automatic consultation notes, or an educational platform with session transcripts and highlights, provides value that Zoom does not.
Tech stack
| Layer | Choice |
|---|---|
| Video SDK (embedded approach) | Daily.co, Agora, or 100ms |
| Video infrastructure (built approach) | mediasoup or Livekit SFU |
| STUN/TURN | coturn (self-hosted) or Twilio STUN/TURN |
| Backend | Node.js |
| Real-time signaling | WebSockets |
| Database | PostgreSQL |
| Recording storage | AWS S3 |
| Transcription | Deepgram or Whisper API |
| Mobile | React Native or Flutter |
| Frontend | Next.js |
Cost to build
| Scope | Timeline | Cost |
|---|---|---|
| SDK-embedded MVP | 8-14 weeks | $60K-$120K |
| Native WebRTC MVP | 14-22 weeks | $120K-$220K |
| With recording + transcription | Add 6-8 weeks | Add $40K-$70K |
Monthly costs scale with usage. For SDK-based: $0.002-$0.004 per participant-minute is typical. At 100,000 participant-minutes/month (small scale), that is $200-$400/month. At enterprise scale, per-minute costs drive the decision to build vs. buy.
What RaftLabs builds
We build video conferencing as a feature within vertical platforms, not as a standalone product competing with Zoom.
Telehealth platforms with appointment scheduling, EHR notes, and consultation recording. Virtual classroom tools with attendance tracking, whiteboard integration, and session archives. Legal platforms with secure call recording, automated transcripts, and compliance retention.
The video layer uses the appropriate SDK or infrastructure for the scale and compliance requirements. The differentiation is in the surrounding product -- the scheduling, the integrations, the AI layer, and the workflow that makes the video call part of a complete business process.
If you are building a vertical platform where video is a core feature, tell us about the use case.
Frequently Asked Questions
- Building on top of a video SDK: 8-14 weeks for an MVP with call management, scheduling, and recording. Building native WebRTC infrastructure for a Zoom-equivalent: 6-12 months minimum. Most product teams building video as a feature of a larger platform should use a third-party SDK and focus development resources on their vertical differentiators.
- SDK-based approach: $60K-$120K for MVP integration with call management and basic features. Native WebRTC infrastructure: $150K-$300K+ for equivalent functionality. Monthly video infrastructure costs depend on usage: third-party SDK pricing is typically per-minute per-participant ($0.002-$0.004 is typical), which becomes significant at scale.
- WebRTC (Web Real-Time Communication) is a browser API that enables real-time audio and video in browsers and mobile apps without plugins. It handles media capture, codec negotiation, and peer-to-peer data transfer. All major browsers support it. The challenge is NAT traversal -- getting video to flow between users on different networks -- which requires STUN and TURN server infrastructure.
- STUN servers help clients discover their public IP address and negotiate direct peer-to-peer connections. TURN servers relay traffic when direct connection is impossible (corporate firewalls, symmetric NAT). About 15-20% of WebRTC calls require TURN relay. TURN servers generate significant bandwidth costs at scale -- this is why video infrastructure is expensive.
- Yes. AI meeting transcription (Whisper or Deepgram), real-time caption overlay, post-meeting summaries, and AI noise cancellation are all buildable today. Background blur (segmentation models) and speaker identification are also mature. These are compelling differentiators for vertical platforms -- a telehealth app with automatic consultation notes, or a tutoring platform with session transcripts, adds real value above the raw video.
