Streaming Latency Calculator
Compare real latency between different audio streaming technologies. Discover why WebRTC offers the lowest latency on the market.
JRMStream's latency calculator compares streaming technologies: WebRTC (<100ms), HLS (6-30s), DASH (3-10s), Shoutcast (10-30s) and Icecast (10-30s). WebRTC is up to 300x faster than traditional solutions.
WebRTC: <100ms
Next-generation technology used by JRMStream. Real-time communication with imperceptible latency.
HLS: 6-30 seconds
Apple's HTTP Live Streaming. Popular but with high latency due to segment-based design.
DASH: 3-10 seconds
Dynamic Adaptive Streaming. Better than HLS but still several seconds of delay.
Shoutcast: 10-30s
Legacy technology from the 2000s. HTTP streaming with high buffer needed.
Icecast: 10-30s
Open source alternative similar to Shoutcast. Same HTTP latency limitation.
RTMP: 2-5 seconds
Adobe's live streaming protocol. Good latency but requires Flash (deprecated).
Latency Comparison by Technology
| Technology | JRMStream (WebRTC) | Traditional |
|---|---|---|
| Latency | < 100ms | 10-30 seconds |
| Protocol | WebRTC (UDP) | HTTP (TCP) |
| Audio Codec | Opus (adaptive) | MP3/AAC (fixed) |
| Buffering | No buffer | 3-15 sec buffer |
| Network Adaptation | Automatic | Manual/limited |
| Real-Time | Yes | No |
Preguntas Frecuentes
What is streaming latency?
Latency is the time from when the broadcaster emits audio to when the listener hears it. Traditional streaming can be 10-30 seconds, while WebRTC is under 100 milliseconds.
Why does WebRTC have lower latency?
WebRTC uses UDP instead of TCP, eliminates traditional buffering and uses adaptive codecs. This enables practically real-time transmission.
Is low latency important for my radio?
Yes, especially for live content like sports, events, DJ sets and interactive shows where audience synchronization is critical.
Can I test the latency difference?
Yes, with JRMStream's free 7-day trial you can experience the latency difference on your own station.