🔴 LIVE • REAL-TIME

Real-Time Audio Streaming

Published: March 20, 2024
Updated: December 07, 2025
E
Written by
Equipo JRMStream
Ingenieros de Streaming WebRTC

Equipo especializado en streaming de ultra-baja latencia.

The only real-time streaming technology with 0.1 seconds latency

0.1s
Real Latency
Listeners
24/7
Availability

What is Real-Time Streaming?

Real-time audio streaming means audio reaches your listeners exactly at the same time you broadcast it. No delay, no buffering, no waiting.

✅ Real-Time (JRMStream)

0.1 seconds latency
No forced buffering
Natural conversations
Synchronized events

❌ False "Real-Time"

15-30 seconds delay
Mandatory buffering
Calls impossible
Desynchronized chat

LIVE Real-Time Demo

Hear the difference RIGHT NOW

🔴 REAL-TIME - JRMStream

WebRTC - No Delay

~50-100ms de latencia

Romántica 90 FM

Demo WebRTC🔴 JRMStream Player

Ready to play
00:00
Connection Quality:
--
State: Disconnected
Latency: -- ms
Jitter: -- ms
Packet Loss: 0%
Buffer: -- KB
Bitrate: -- kbps
Codec: Opus 48kHz
WebRTC Technology
Opus 48kHz Codec
No forced buffering
⏰ WITH DELAY - MP3

HTTP/MP3 - Traditional

~5-10 seconds latency

Romántica 90 FM

Demo Transcoder🔴 JRMStream Player - MP3

Ready to play
00:00
HTTP/TCP Protocol
MP3 Codec (Legacy)
5-10 seconds buffer

Difference: JRMStream is 200x faster

When You Need Real-Time

📻

Radio with Calls

Natural conversations with listeners. No annoying 20-second delay.

✅ Critical for interaction
🎙️

Interactive Podcasts

Live Q&A, instant polls, immediate responses.

✅ Maximum engagement

Sports Events

Synchronized narration with TV. Goals are heard instantly.

✅ Perfect synchronization
🎵

Live DJ Sets

Perfect mixes, immediate audience reaction, real energy.

✅ Artist-audience connection
🎓

Online Education

Interactive classes, instant questions, fluid learning.

✅ Educational interaction
🏢

Corporate Conferences

Dynamic presentations, real-time questions, networking.

✅ Effective communication

How Real-Time Works

WebRTC: The Key

Direct P2P Connection

No intermediate servers adding latency

UDP Protocol

No waiting for confirmations, transmits immediately

Codec Opus

Specifically designed for real-time

No Forced Buffering

Plays audio instantly

HTTP: The Problem

Multiple Servers

Each hop adds cumulative latency

TCP Protocol

Waits for confirmation before sending more data

Mandatory Buffer

Requires 10-30 seconds of audio in buffer

Additional Compression

MP3 codecs add more processing delay

Experience True Real-Time

No more delay. No more buffering. Just pure real-time audio streaming.

Risk-Free Trial

  • 7 days completely free
  • 0.1 seconds latency guaranteed
  • 5-minute setup
  • No credit card required
🚀 Try Real-Time Free

Join the +10,000 stations already streaming in real-time