Low Latency in Real-Time Video Calling: What It Means & How to Achieve It

Video calling is something that is keeping the world connected. From chit-chatting with friends to business-related discussions, low latency is highly needed for effective real-time video calling.

As demand increases for flawless face-to-face interaction on different devices, even the slightest delay has become considerably important for both developers and enterprises.

With video communications continuing to grow, huge industries like telehealth, remote work, and live commerce are controlling the world. There, latency optimization has become more relevant than ever.

This blog explores what low latency means in real-time video calling, how it works, and the best practices to achieve it. Keep reading to learn more!

What Does Low Latency Mean and How Does It Work?

To define what is low latency: it refers to the minimal delay between sending and receiving data in real time. Even a few milliseconds of lag can make live calls feel out of sync. Understanding this concept helps developers create faster, smoother, and more reliable video experiences.

What Is Latency in Real-Time Video Calling

In the context of real-time delivery of media from one user to another, latency determines how quickly your voice and video are delivered to the other user.

For example, with a natural face-to-face conversation, the human brain can identify delays of greater than 150-200 milliseconds. If it is anything beyond that, it doesn’t feel like a real-time interaction.

Video call latency involves multiple components:

• One-Way Latency: The time it takes for a single packet of data to travel from the sender to the receiver.
• Round Trip Time (RTT): A two-way measurement that shows how long it takes for data to go from the sender to the receiver and then return again.
• Audio vs. Video Sync: The human ear is more sensitive to delay than the eye, so keeping speech aligned with lip movement is essential for a natural, real-time conversation.

Other related metrics include jitter, which refers to variations in how packets arrive, and packet loss, which occurs when data never reaches its destination. Either issue can cause choppy audio, frozen frames, or echoing voices. In short, when these numbers are high, real-time communication suffers.

Together, these metrics explain the naturalness and responsiveness of a live call.

How Is Latency Measured in Video Calls?

Latency is determined through the use of tools that log the time in milliseconds (ms) for the transfer of data packets between the endpoints. The following are the two prominent techniques:

• One-Way Delay: This method requires the sender and receiver clocks to be synchronized. It provides more precise measurements but is harder to implement.
• Round-Trip Time (RTT): A simpler approach that measures how long it takes for data to travel to the destination and back.

Engineers also compare the average against tail latency, such as p90 and p99, in order to understand worst-case performance. Latency differs among the mobile, Wi-Fi, and wired networks, and often it is higher on mobile because of handovers and signal interference.

For quality evaluation, metrics include MOS, which stands for Mean Opinion Score, jitter, packet loss, and frame drops, showing the real call experience.

What Is “Good” Latency for Video Calling?

In video chats, good latency refers to delays that are brief enough to allow for undisturbed and natural talks.

For most people, video calls feel natural when latency stays within the 150–200 ms range. Once it goes beyond that, delays become noticeable, leading to pauses, interruptions, and people talking over each other.

Also, because audio is more sensitive to delay, maintaining tight audio-video sync is essential. In use cases like gaming, telemedicine, or financial trading, keeping latency below 100 milliseconds helps ensure smooth performance and gives users a greater sense of confidence and security.

Why Low Latency Matters in Video Calling

Low latency isn’t just about having a fast connection — it’s about creating a smooth, natural video calling experience. When lag is minimal, conversations flow easily, and people on both sides feel heard and understood.

So, low latency plays a major role in determining whether a conversation feels smooth and natural — or awkward and frustrating due to delays.

• Real-Time Experience and User Satisfaction: Smooth, natural conversation enhances the overall user experience, keeping people engaged and satisfied.
• Business Conversion and Retention: In live commerce or customer support, fast responses matter. Low latency improves trust, attention, and conversion rates.
• Accuracy and Reaction Time in Telehealth: Healthcare professionals rely on instant, clear communication to make accurate decisions and reassure patients.
• Multi-User Collaboration Flow: Teams collaborate more effectively when ideas can be exchanged instantly, maintaining momentum and creativity.
• Trust and Emotional Presence: Lag-free interaction strengthens confidence, empathy, and the feeling of genuine human connection — a major UX advantage.

Causes of Latency in Video Calling Apps

Latency in video calls can arise from several technical and environmental factors:

• Network Propagation Delay: Longer distances between users and media servers increase delay. Even light traveling through fiber takes measurable time, especially across continents.
• Encoding & Decoding Delay: Device processing speed and codec complexity determine how quickly audio and video can be compressed and rendered.
• Transport & Protocol Overhead: Protocols like WebRTC over UDP help reduce delay, while TURN traversal (used to bypass firewalls) can introduce extra milliseconds compared to STUN.
• Architecture-Based Latency: P2P works well for small calls, SFU scales more efficiently for larger groups, and MCU adds more processing—resulting in higher latency.
• Mobile Device & OS Constraints: CPU load, battery-saving modes, and network transitions (e.g., switching from 4G to 5G) can cause temporary latency spikes.

Video Streaming vs Two-Way Video Calling Latency

Though both deliver live video, streaming and real-time video calling differ sharply in latency, direction, and interaction needs.

When​‍​‌‍​‍‌​‍​‌‍​‍‌ it comes to streaming, a few seconds of delay do not cause any problem because it is a one-way flow of media, similar to a live broadcast. On the other hand, real-time video calling is a two-way, interactive communication that has to remain under 150 to 200 milliseconds to be able to have a conversation that sounds natural and human-like.

Buffering is another major difference. Streaming relies on buffering to keep playback smooth—no one wants a video that freezes every few seconds. But in video calling, buffering disrupts the conversation and breaks the natural flow of real-time interaction.

Streaming normally goes for HLS, DASH, or other similar segmented delivery methods that divide the data into chunks and then send them. Video calls take a different route. They use WebRTC over UDP, a setup built for real-time communication. It keeps delays low and the conversation feeling immediate.

In terms of network tolerance, streaming can handle fluctuations through preloading, whereas video calls demand a stable, low-jitter connection to maintain instant feedback.

Lastly, the audience watching a streaming event is okay with a few seconds of delay. However, on a live call, people want responses right now, perfectly in sync. Anything less feels off.

Note: Low-latency streaming should not be confused with real-time interaction. While it reduces delay, streaming protocols prioritize smooth playback—not instant exchange. This makes them unsuitable for live, two-way communication such as real-time video calls.

How to Achieve Ultra-Low Latency in Video Calling

Reaching ultra-low latency requires optimizing every layer of the communication stack from network routing to codec efficiency and performance monitoring.

• Making Use of RTC Protocols for Real-Time: While it's possible to rely solely on UDP, WebRTC offers a stronger foundation for fast, stable real-time calling. Choosing efficient codecs such as AV1, H.264, VP8, or VP9 can further improve performance.
• Global Edge/PoP Distribution: Deploy media servers closer to users by establishing global Points of Presence. This reduces propagation delay and helps calls feel faster and more responsive.
• Bandwidth Control and Adaptive Bitrate: When network conditions fluctuate, adaptive bitrate automatically adjusts video quality to maintain smooth, interruption-free playback.
• Network Adaptation: Devices should intelligently switch between 4G, 5G, and Wi-Fi based on the strongest available connection. If a network drops, STUN fallback can keep the session alive with minimal user disruption.
• Jitter Buffer Tuning: Prioritizing audio stability ensures better lip-sync accuracy and improves the overall natural flow of the conversation.
• Performance Monitoring: Continuously track key metrics—such as MOS, latency spikes, jitter, and packet loss—to identify and resolve issues before they affect call quality.

Latency vs Bandwidth vs Throughput

These three terms are frequently mixed up with each other. However, they represent different sides of network ​‍​‌‍​‍‌​‍​‌‍​‍‌performance.

• Latency: The delay that shows how long data takes to travel from the sender to the receiver.
• Bandwidth: The capacity that determines how much data can be transmitted per second.
• Throughput: The actual speed achieved after accounting for network overhead and congestion.

Why High Bandwidth ≠ Low Latency:

No​‍​‌‍​‍‌​‍​‌‍​‍‌ matter how big the data pipe is, it does not always guarantee a fast delivery. Sometimes, even those connections that have a very high bandwidth can have high latency if there are routing, congestion, or processing delays.

Throughput affects frame re-synchronization, especially during video recovery after packet loss, a common area where teams misdiagnose performance issues.

Video Calling Latency Thresholds by Application

Different real-time applications demand different latency ranges for optimal performance. What feels “acceptable” in one use case can completely disrupt another.

• Live Sports and Event Broadcasting: A delay of 1–2 seconds is generally acceptable for audience viewing.
• Interactive Gaming and eSports: Response times should stay under 100 ms for precise, real-time reactions.
• Video Conferencing and Collaboration: A delay below 150–200 ms helps maintain a natural conversational flow.
• Telemedicine and Remote Healthcare: Optimal performance typically falls within the 100–150 ms range.
• Online Education and Virtual Classrooms: Delays up to 250 ms can still support smooth interaction.
• Financial Trading and Market Analysis: Latency must stay under 50 ms to support instant decision-making.
• IoT and Robotics Control: Responses need to be under 30–50 ms for immediate, real-time control.
• Live Shopping, Auctions, and Social Streaming: Engagement feels natural when latency stays below 200 ms.
• XR/AR/VR Experiences: Latency must remain under 20 ms to maintain natural motion and user immersion.

How to Choose a Video Calling SDK/API Based on Latency

When you’re looking at video calling SDKs or APIs, focus on what really affects real-time performance and keeps latency steady.

• Global PoP (Points of Presence) Coverage: Choose providers with servers or edge nodes across multiple regions. The closer users are to media servers, the lower the propagation delay.
• Advanced Jitter Buffer Technology: Look for SDKs that automatically adjust to network fluctuations to keep audio and video synchronized, even under unstable conditions.
• High Participant Handling Without MCU: Platforms that use SFU (Selective Forwarding Unit) instead of MCU reduce processing overhead and help maintain lower latency in multi-party calls.
• Efficient TURN Fallback Performance: An optimized fallback strategy ensures stable performance even under restrictive network environments or poor connectivity.
• Strong Service Level Agreement (SLA): Prioritize SDKs that guarantee low latency, high uptime (99.9% or more), and strict jitter limits for dependable performance.
• Analytics and QoE Monitoring: Real-time dashboards for tracking latency spikes, MOS, packet loss, and other quality metrics help identify and resolve issues quickly.
• Compliance and Security: In sensitive sectors like telehealth or finance, ensure the SDK meets HIPAA or other industry standards while still delivering low latency.