What is low latency for voice calls?

Low latency means very small delay between speaking and the other party hearing you. Typical targets are under 150 ms round trip for local calls and slightly higher for international calls.

Can I call the PSTN from my browser with low latency?

Yes. Using WebRTC with regional PSTN gateways and optimized routing lets you reach mobile and landline phones with minimal additional delay. StartACall handles the bridging and edge routing for you.

Do I need to install anything?

No installation is required. StartACall runs in modern browsers and connects to the phone network without plugins.

What causes high latency in browser voice calls?

High latency usually comes from network congestion, long routing paths between peers, reliance on TURN relays, large jitter buffers, or inefficient codecs. Device CPU and browser audio processing can also add delay.

How can I test latency before deploying?

Measure round trip time using WebRTC stats, run synthetic calls from target regions, and monitor packet loss and jitter. StartACall includes real-time analytics to monitor these metrics during test calls.

Do I need any plugins or apps to get low latency calls in my browser?

No. Low latency voice calls can be made directly from modern browsers using WebRTC. StartACall connects WebRTC to the PSTN without requiring downloads.

Performance

Low Latency Voice Calls
from Your Browser

Make near real-time phone calls with minimal delay using WebRTC to PSTN bridging. StartACall connects your browser to the global phone network with tuned routing and built-in monitoring so conversations feel instant and natural.

Read the Guide

Browser CallLive

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Low Latency Voice Calls Guide

Practical steps to reduce delay and keep conversations fluid when calling from a browser.

Use WebRTC and Opus

WebRTC provides peer-to-peer media transport. Opus is a low-delay codec that adapts to network conditions and preserves voice quality at small frame sizes.

Prefer Direct Paths

Enable ICE with STUN to establish direct connections. Avoid TURN relays when possible because relays add extra hops and delay. Use region-aware routing so media stays close to users.

Optimize Buffers

Tune jitter buffers and playout delay to the minimum safe values. Adaptive jitter buffering reduces audible gaps while keeping latency low.

Additional Best Practices

Monitor packet loss and jitter in real time and adjust bitrate or codec settings when needed.
Prioritize UDP transport for media where possible. Use transport layer encryption like SRTP for secure, low-latency media.
Use server and gateway locations close to your users and the PSTN endpoint to avoid long routing paths.

Technology that keeps delay low

How StartACall builds for speed and reliability.

WebRTC to PSTN bridging

We bridge in-browser WebRTC sessions to the Public Switched Telephone Network using regional gateways. That keeps media paths short and reduces latency compared to long carrier chains.

Global number reach with local media edge points

AI Assistance without delay

Real-time transcription and AI Copilot features run with streaming models and low-latency speech recognition so you get instant prompts and summaries during calls.

Secure, low-latency AI processing integrated with live audio

Real-time analytics

Track RTT, jitter, packet loss, and codec performance during each call. Use analytics to spot regions with elevated delay and route around issues quickly.

Live metrics and call health monitoring

Virtual numbers and routing

Assign virtual numbers in the countries you call so caller responses stay local and media stays short. Regional routing cuts latency for international conversations.

Numbers in 190+ countries with edge-aware routing

Frequently Asked Questions

What latency should I expect for browser calls?

Good browser to browser calls often land under 150 ms round trip time when peers are nearby. Calls crossing continents will be higher. StartACall optimizes routing to minimize delay to the PSTN endpoint.

How does StartACall reduce delay compared to traditional calling?

StartACall uses WebRTC and regional media gateways to avoid long carrier chains and unnecessary relays. That shortens media paths and lowers end to end delay.

Will network conditions still impact latency?

Yes. Packet loss, jitter, and congestion will affect perceived delay. Our real-time analytics and adaptive media settings help you detect and mitigate these issues quickly.

Can I run test calls from my region?

Yes. Sign up and run test calls from your browser. Real-time metrics display latency, jitter, and packet loss so you can validate performance before rolling out to users.

Low Latency Voice Calls from Your Browser