DEV Community

vmodal_ai
vmodal_ai

Posted on

Kotlin + WebRTC: Building Real-Time Audio/Video Applications

Kotlin + WebRTC: Building Real-Time Audio/Video Applications

WebRTC enables real-time peer-to-peer audio and video communication.

Typical use cases include:

  • video calls
  • voice calls
  • remote assistance
  • online classrooms
  • telemedicine
  • live collaboration

A simplified architecture is:

Android A                         Android B
   |                                  |
Camera / Mic                      Camera / Mic
   |                                  |
   +-------- WebRTC Peer Connection--+
                    |
              Media Transport
Enter fullscreen mode Exit fullscreen mode

WebRTC Is More Than a Camera API

A complete WebRTC application usually needs:

  1. media capture
  2. peer connections
  3. signaling
  4. ICE candidate exchange
  5. STUN/TURN infrastructure
  6. connection state handling
  7. permissions
  8. lifecycle management

Signaling

WebRTC itself does not define your application's signaling server.

Signaling is used to exchange:

  • SDP offers
  • SDP answers
  • ICE candidates

A typical sequence:

Caller                      Signaling Server                 Callee

  |------ SDP Offer ------------>|
  |                              |------ SDP Offer ---------->|
  |                              |<----- SDP Answer ----------|
  |<----- SDP Answer ------------|
  |                              |
  |------ ICE Candidate -------->|
  |<----- ICE Candidate ---------|
Enter fullscreen mode Exit fullscreen mode

The signaling server does not normally carry the actual media stream.

Android Permissions

For camera and microphone access, request the appropriate runtime permissions.

Your application should clearly explain why it needs them.

Conceptually:

CAMERA
RECORD_AUDIO
Enter fullscreen mode Exit fullscreen mode

Follow Android's current permission requirements and privacy guidance.

WebRTC Peer Connection

The exact Android API depends on the WebRTC distribution used by your project.

Conceptually:

val peerConnection = peerConnectionFactory.createPeerConnection(
    rtcConfig,
    observer
)
Enter fullscreen mode Exit fullscreen mode

The peer connection manages the media transport.

Capturing Camera and Microphone

The media pipeline is approximately:

Camera
  ↓
Video Source
  ↓
Video Track
  ↓
Peer Connection

Microphone
  ↓
Audio Source
  ↓
Audio Track
  ↓
Peer Connection
Enter fullscreen mode Exit fullscreen mode

Add tracks to the peer connection before negotiating.

SDP Offer / Answer

Caller:

Create PeerConnection
      ↓
Create Offer
      ↓
Set Local Description
      ↓
Send SDP through signaling
Enter fullscreen mode Exit fullscreen mode

Callee:

Receive Offer
      ↓
Set Remote Description
      ↓
Create Answer
      ↓
Set Local Description
      ↓
Send Answer
Enter fullscreen mode Exit fullscreen mode

ICE Candidates

ICE candidates describe possible network paths.

Exchange them through your signaling layer:

Peer A
  |
  | ICE candidate
  v
Signaling
  |
  | ICE candidate
  v
Peer B
Enter fullscreen mode Exit fullscreen mode

STUN and TURN

Direct peer-to-peer connectivity is not always possible.

STUN helps peers discover their public network address.

TURN relays media when a direct path cannot be established.

Production deployments should plan for TURN capacity because relay traffic can be expensive.

Connection States

Track states such as:

NEW
CONNECTING
CONNECTED
DISCONNECTED
FAILED
CLOSED
Enter fullscreen mode Exit fullscreen mode

The UI should respond appropriately.

For example:

Connecting...
Connected
Reconnecting...
Call ended
Enter fullscreen mode Exit fullscreen mode

Call Lifecycle

A robust Android implementation needs to handle:

  • app backgrounding
  • screen rotation
  • permission changes
  • incoming calls
  • network changes
  • Bluetooth/headset changes
  • call termination

Do not assume the Activity lifetime equals the call lifetime.

Architecture

A scalable application can use:

UI
 |
CallViewModel
 |
CallRepository
 |
WebRTC Engine
 |
Signaling Client
 |
Signaling Server
Enter fullscreen mode Exit fullscreen mode

Keep WebRTC-specific APIs out of UI components whenever possible.

Video Rendering

Remote video is normally rendered through the WebRTC video rendering API supplied by your WebRTC implementation.

Keep rendering separate from call control.

Security

Use:

  • authenticated signaling
  • secure signaling transport
  • authorization
  • encrypted WebRTC media
  • server-side access control
  • careful handling of call metadata

Never trust identifiers received from the client.

Testing

Test:

  • camera permission denied
  • microphone permission denied
  • Wi-Fi to mobile-data transition
  • poor connectivity
  • call rejection
  • call termination
  • background/foreground transitions
  • TURN fallback
  • multiple calls

Conclusion

Building a WebRTC application is a systems problem rather than simply adding a video widget.

The media engine, signaling layer, network traversal, Android lifecycle, permissions, and application state must work together.

A clean architecture keeps WebRTC complexity behind a dedicated call engine or repository so the rest of the Android application remains maintainable.

Useful Links

SDK Flutter: https://github.com/v-modal/vmodal_sdk_flutter

SDK Android: https://github.com/v-modal/vmodal_sdk_android

Discord: https://discord.gg/K72z28KUx

Top comments (0)