WebRTC vs RTP (2025): Comprehensive Comparison of Real-Time Streaming Protocols

Introduction

What is RTP?

Core Features of RTP

• Packet-Based Transmission: RTP breaks media streams into packets, enabling timely and efficient delivery over UDP/IP.
• Timestamping and Sequence Numbers: Ensures in-order playback and synchronization of media streams.
• Payload Type Identification: Allows dynamic media type negotiation and switching.
• Extensibility: RTP can incorporate new codecs and features with minimal protocol changes.

Typical Use Cases for RTP

• VoIP (Voice over IP) Calls: Widely used in SIP and H.323 systems.
• IPTV and Broadcast Streaming: Reliable transport for one-way media delivery.
• Surveillance and Telepresence: Integrates with hardware and software endpoints.
• Conference Bridges and Media Servers: Centralized processing and routing of RTP streams.

What is WebRTC?

WebRTC Architecture and Key Features

• Peer-to-Peer Media Transmission: Directly connects endpoints for low-latency communication.
• Browser Support: Native APIs in all major browsers (Chrome, Firefox, Edge, Safari) as of 2025.
• Secure by Default: Encrypts media with Secure RTP (SRTP) and data with DTLS.
• APIs for Developers:
  • RTCPeerConnection: Handles signaling, negotiation, and media transport.
  • RTCDataChannel: Enables low-latency, peer-to-peer data exchange.
  • MediaStream: Provides access to local audio/video devices.
• Interactive Streaming: Ideal for video calls, chat, gaming, and collaborative tools. For those building interactive broadcast solutions, a

  Live Streaming API SDK

  can further enhance WebRTC-powered experiences.

Typical Use Cases for WebRTC

• Video Conferencing & Collaboration: Zoom, Google Meet, and other platforms leverage WebRTC for browser-based meetings.
• Customer Support Widgets: Real-time audio/video chat in web apps.
• Live Education & Telehealth: Secure, interactive communication between browsers and mobile devices.
• Peer-to-Peer File Sharing: Using data channels for decentralized transfer.

How Does RTP Work?

RTP Packetization & Transport

• Packetization: Audio and video frames are split into RTP packets, each labeled with sequence numbers and timestamps.
• Transport: Most commonly over UDP for low latency, but can also use other transports.
• Session Management: Sessions are identified with unique SSRC (synchronization source) identifiers.

The Role of RTCP

Example: Setting Up an RTP Stream

1import socket
2
3def send_rtp_packet(ip, port, payload):
4    s = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
5    rtp_header = b'\x80\xe0\x00\x01'  # RTP version, payload type, sequence number
6    packet = rtp_header + payload
7    s.sendto(packet, (ip, port))
8    s.close()
9
10# Example usage:
11send_rtp_packet('192.168.1.10', 5004, b'MY_AUDIO_FRAME')
12

RTP Packet Flow Diagram

How Does WebRTC Use RTP?

WebRTC Media Transport Stack

• SRTP (Secure RTP): All media is encrypted, protecting against eavesdropping.
• ICE (Interactive Connectivity Establishment): Handles candidate gathering and connectivity checks for NAT/firewall traversal.
• STUN/TURN: Assists in discovering public IPs or relaying media when direct connections fail.
• DTLS (Datagram Transport Layer Security): Secures key negotiation for SRTP.

Differences from Traditional RTP

• Encryption is Mandatory: WebRTC requires SRTP, while RTP alone does not.
• Dynamic Peer-to-Peer Connections: ICE and STUN/TURN are used for establishing connections even across restricted networks.
• Signaling Layer: WebRTC does not specify signaling; developers can use WebSocket, SIP, or custom protocols.

Example: Establishing a WebRTC Connection

1const pc = new RTCPeerConnection();
2navigator.mediaDevices.getUserMedia({ video: true, audio: true })
3  .then(stream => {
4    stream.getTracks().forEach(track => pc.addTrack(track, stream));
5    return pc.createOffer();
6  })
7  .then(offer => pc.setLocalDescription(offer));
8// Signaling logic and ICE candidate exchange required for full connection
9

WebRTC Signaling and Media Flow

WebRTC vs RTP: Feature-by-Feature Comparison

Low-Latency Performance

• WebRTC: Designed for ultra-low latency, with direct peer-to-peer transmission and built-in congestion control.
• RTP: Low latency when used over UDP, but can be affected by network conditions and lacks built-in NAT traversal.

Security

• WebRTC: All streams use SRTP and DTLS, making encryption non-optional.
• RTP: Security is optional; standard RTP streams are unencrypted unless SRTP is manually implemented.

NAT/Firewall Traversal

• WebRTC: Uses ICE, STUN, and TURN to traverse NATs and firewalls efficiently.
• RTP: Requires additional setup (e.g., SIP with ICE) for NAT traversal.

Scalability & Network Usage

• WebRTC: Peer-to-peer is efficient for small groups; SFUs/MCUs needed for larger conferences.
• RTP: Scales well with server-based architectures, but lacks browser integration.

Browser & Device Support

• WebRTC: Natively supported in all major browsers and mobile platforms in 2025.
• RTP: Requires third-party libraries or plugins for browser use.

Feature Comparison Table

When to Use WebRTC vs When to Use RTP

• Use WebRTC when: You need interactive, browser-based, or mobile peer-to-peer communication. Ideal for video conferencing, collaborative apps, and embedded widgets.
• Use RTP when: Integrating with legacy VoIP systems, hardware encoders, or broadcast workflows. Best for centralized, server-based streaming.

Decision Flowchart

Implementation Challenges and Considerations

• Interoperability: Integrating WebRTC with legacy RTP systems may require gateways or protocol adapters.
• Packet Loss, QoS, and Jitter: Both protocols must address network unreliability. WebRTC includes adaptive bitrate and FEC; RTP relies on RTCP feedback.
• Security & Privacy: Always use SRTP for RTP, and follow best practices for key management, authentication, and access control in WebRTC.

Conclusion