DEV Community: Amar Thodupunoori

WebRTC vs. MoQ — Two Protocols, One Platform Completely Built for Both

Amar Thodupunoori — Wed, 08 Apr 2026 10:55:56 +0000

Two powerful protocols. One streaming platform built for both. Lets focus on what happens today and what’s waiting for us in the future as Ant Media Server perspective.

The live streaming world is buzzing about Media over QUIC (MoQ) — a new IETF-standard protocol that promises to combine the scalability of CDN-based streaming with the sub-second latency as we used to associate with WebRTC only so far. At Ant Media Server, we’ve built our platform around WebRTC since day one as known globally.

So the question we get asked constantly is: Should you be worried? Is WebRTC dead?

The short answer: No. But MoQ is genuinely exciting — and understanding the difference between the two is critical to making smart infrastructure decisions now and also for future.

Two Protocols, Two Philosophies

WebRTC and MoQ weren’t designed for the same problem. They emerged from different eras, different constraints, and different visions of what the real-time web should look like.

WebRTC- Web Real-Time Communication

Born in 2011
Standardized by W3C & IETF, shipped in Chrome in 2012
~0.2–0.5s latency
True sub-second, ideal for interactive apps
SFU Architecture
Server-side Selective Forwarding Units for scalability
Universal browser support
Chrome, Safari, Firefox, Edge — no plugins needed
Transport: UDP / DTLS-SRTP
Built on RTP, approximately 20 referenced standards

MoQ- Media over QUIC

Emerging standard, ~2022–present
IETF working group, still in active development
Sub-second to near-real-time
Configurable latency from ultra-low to VOD-grade
Pub/Sub + CDN Relay Architecture
Relays fan out live media with structured tracks
Chrome & Edge only (2026)
Safari iOS WebTransport support is on the way
Transport: QUIC / WebTransport
Built on HTTP/3, no RTP dependency

WebRTC: Where Ant Media Focuses Today

Ant Media Server was built around WebRTC — and for good reason. WebRTC delivers sub-0.5 second latency across every major browser on the planet without requiring a plugin, app download, or special configuration from your end users. For use cases where responsiveness is existential — live auctions, telehealth consultations, remote drone monitoring, interactive sports betting — there is simply no better option available at production scale today.

Our SFU-based architecture means viewer connections are handled efficiently: the origin node accepts and transcodes incoming streams, while edge nodes play them out. This scales from a few person virtual classroom to a global live event with tens of thousands of concurrent viewers — and it does so on infrastructure that auto-scales on AWS, Azure, GCP, or your own on-premise cluster via Kubernetes.

Where WebRTC Wins

Telehealth & Remote Consultation
HIPAA-compliant, real-time patient-provider video with sub-500ms responsiveness. Latency matters when a doctor needs to notice a patient’s reaction.
Live Auctions & Bidding
Fairness depends on all bidders seeing the same moment simultaneously. Any latency asymmetry is a legal and commercial liability.
Interactive Gaming & Betting
Engagement and revenue in real-time gaming require immediate feedback loops. WebRTC delivers the interactivity that keeps users in the moment.
Surveillance & IoT Monitoring
Real-time CCTV and IP camera feeds benefit from WebRTC’s encrypted, browser-native delivery without buffering delays or plugins.

The Honest Limitations

WebRTC’s complexity is legendary. The protocol stack references approximately 20 standards, making it genuinely difficult to customize outside the bounds of what browser vendors choose to implement. ICE negotiation, STUN/TURN traversal, and SDP signaling are all layers of complexity that sit between you and “just streaming video.” At Ant Media, we abstract most of this — but it’s worth being honest that at true internet-scale one-to-many streaming, WebRTC’s architecture requires significant investment to remain cost-efficient.

WebRTC also has no native concept of CDN-friendly relay architectures. It scales through SFUs and clustering which means infrastructure costs grow with your viewer count in ways that pure CDN-used protocols avoid.

MoQ: The Architecture That Fixes the Middle Ground

Media over QUIC is the most thoughtful attempt yet to bridge the long-standing gap between two worlds: the cost efficiency and CDN-scalability of HLS, and the near-zero latency that WebRTC enables. MoQ is built on QUIC — the same transport layer behind HTTP/3 — which eliminates TCP’s head-of-line blocking, handles connection migration gracefully.

The key innovation in MoQ is its publish/subscribe model built around “tracks” — linear flows of media data (video, audio, captions, metadata) that relays can cache and fan out at the live edge. Unlike WebRTC, which requires a full SFU session per viewer, MoQ’s relay architecture lets CDN nodes participate natively. That’s why giant companies like YouTube are paying attention: MoQ lets existing CDN infrastructure be upgraded rather than replaced.

MoQ’s goal is to give you WebRTC-like interactivity and HLS-like scalability in a single protocol. Sub-second join times + internet-scale fan-out without maintaining thousands of individual real-time sessions.

Where MoQ Shines (When Ready)

Large-Scale Live Events
Concerts, sports broadcasts, and political events where you need sub-second latency for a million simultaneous viewers — a CDN relay model makes this economical.
Hybrid Live + VOD Platforms
A single protocol handling live streaming and on-demand playback means dramatically simpler architecture and unified infrastructure costs.
Next-Gen CDN Integration
MoQ’s HTTP/3 compatibility means CDNs can extend their existing networks rather than replace them wholesale.

The Honest Limitations

MoQ is genuinely exciting — but it is not production-ready today, and the numbers back that up. As of late 2025, WebTransport (which MoQ depends on in browsers) represents a fraction of a percent of web page loads, versus WebRTC’s stable ~0.35%. Chrome metrics show brief experimental spikes followed by drop-offs.

Safari on iOS was a significant blocker until it was recently (a week ago) announced that WebTransport is supported with Safari iOS 26.4 , may help removing fallback implementations that add complexity. Some networks still block UDP traffic. And the MoQ specification itself, while advancing rapidly through IETF, is still evolving — meaning production deployments today may carry interoperability risk.

Where Ant Media Is Positioned: Protocol-Agnostic Pragmatism

Ant Media Server has always been protocol-pragmatic. We started with RTMP and WebRTC, layered in SRT, RTSP, HLS, LL-HLS, CMAF, WHIP/WHEP — because the right protocol depends on the use case, not industry fashion cycles.

Our position on WebRTC vs MoQ mirrors what the most credible voices in the streaming space have concluded: these protocols are not competitors — they are complements. WebRTC is the definitive answer for interactive, browser-native, sub-500ms experiences. MoQ is the most architecturally elegant answer for the future of one-to-many streaming at internet scale with CDN economics.

The industry consensus forming around a hybrid workflow makes intuitive sense: WebRTC for browser-based contribution and ingest, with MoQ as the delivery layer when it matures. This is exactly the kind of architecture Ant Media Server is designed to support — accepting streams over any protocol and delivering them over whatever transport best fits the viewer context.

What This Means for Ant Media Users

Today: build on WebRTC with confidence. Our SFU-based clustering, adaptive bitrate engine, auto-scaling on major clouds or on premise, and ~0.5s latency guarantee are production-proven. When MoQ achieves production maturity, Ant Media’s multi-protocol architecture means you add it as a delivery option — not a platform migration.

How Ant Media Approach: Don’t Choose. Prepare for Both.

If you need to build something today — a telehealth platform, a live auction, a drone monitoring system, an interactive sports stream — build it on WebRTC. It’s proven, universally supported, and with Ant Media Server, it scales gracefully from prototype to production without infrastructure dependencies.

If you’re designing a platform for 2028 and beyond — especially one where CDN economics and massive concurrent audiences matter — keep a close eye on MoQ. The fundamentals are solid. The IETF momentum is real. The giant companies are all investing. When cross-browser support closes, MoQ will be ready for the architectures that WebRTC was never designed for.

At Ant Media, our strategy is simple: the future of streaming is to have multi-protocol capability in one platform, and your infrastructure should be too. We’re watching MoQ closely, supporting WHIP/WHEP as the bridge between today and tomorrow, and building the platform that lets you change your delivery layer without changing your application.

To demonstrate our commitment to our users, we’ve decided to showcase how MoQ works and performs compared to other protocols at NAB Show, starting April 19, 2026, at Booth 3318 in Las Vegas. We invite you to join us and experience it firsthand.

Pick the right tool for the right job — and build on infrastructure flexible enough to evolve with it.

Ant Media at NAB Show 2026 on April 19-22

Amar Thodupunoori — Wed, 25 Mar 2026 10:47:56 +0000

The countdown to NAB 2026 has begun, and we’re excited to announce that Ant Media will be part of this year’s premier gathering for media, entertainment, and technology innovators.

At Ant Media, our mission has always been clear: to empower businesses and developers to take steps towards their dreams by offering ultra-low latency streaming, scalable infrastructure, and cutting-edge real-time communication solutions. NAB 2026 provides the perfect stage to showcase how far streaming technology has come—and where it’s headed next.

Get your free Exhibits Pass by using our FREE code NS4424 when registering at here

Join us at NAB 2026

Please join us at West Hall, Booth W3317 and find out what awaits you at our booth:

Live Demos

Experience our fully auto-scalable and self-managed live streaming service, designed to run seamlessly on any cloud network with just one click. See Media Over QUIC (MoQ) in action and how it compares to WebRTC and other delivery protocols. You’ll also get a closer look at how Ant Media Server supports AI integration within your streaming workflows, whether for video or audio.

Meet Our Partners

Discover Ant Media’s trusted partners — SyncWords, Mobiotics, Raskenlund, 1000Volt, Spaceport — and explore how these collaborations are driving innovation. From video processing and free viewpoint video capture to AI-powered captioning, Server-Guided Ad Insertion (SGAI), Server-Side Ad Insertion (SSAI), and automatic subtitling through Speech-to-Text AI, and more.
SyncWords   Mobiotics   Raskenlund  1000Volt    Spaceport

Expert Guidance

Connect with our team of experts, who are ready to share insights, answer your questions, and help tailor solutions to fit your specific streaming needs.

At Ant Media, we are passionate about pioneering the future of live streaming, and we can’t wait to share this thrilling journey with you at NAB 2026!

We look forward to welcoming you to NAB 2026 and sharing our passion for innovation and excellence in live video streaming.

Harness Powers of DeepAR and Custom Overlay with Ant Media Android SDK

Amar Thodupunoori — Thu, 05 Mar 2026 10:29:03 +0000

Live streaming has evolved beyond simple camera-to-viewer broadcasts. Today’s audiences expect interactive, engaging content with visual effects, branding elements, and augmented reality features.

In this blog post, we’ll explore two powerful approaches to enhance your Android live streams using Ant Media Server: DeepAR integration for stunning AR effects and Custom Canvas overlays for logo overlay or adding some custom overlays to the video on the client side. One can apply these effects and start streaming with the Ant Media Server SDK.

Both approaches leverage Ant Media Server’s WebRTC capabilities to deliver low-latency, high-quality streams with custom visual enhancements applied in real-time

GitHub Repository: https://github.com/USAMAWIZARD/AntMedia-DeepAR-And-Overlay

Getting Started
Section 1: DeepAR Activity – Augmented Reality for Live Streaming
What is DeepAR?
The DeepAR Streaming Experience
How It Works (High Level)
Use Cases
Code Overview
Section 2: Custom Canvas Activity – Branded Overlays for Professional Streams
Why Custom Overlays?
The Custom Overlay Experience
Built-in Overlay Features
How It Works (High Level)
Use Cases
Code Overview
The Ant Media Server Advantage
Conclusion

Getting Started

There are two activity samples, DeepARActivity.java and CustomCanvasActivity.java.

You may need to create a licence key for Deep AR, sign up, and create a licence by creating a new project, then select Android, and set the licence key in code.

git clone https://github.com/USAMAWIZARD/AntMedia-DeepAR-And-Overlay
Build and run on your Android device
Start streaming with effects or overlays!

Play the stream with WebRTC

Section 1: DeepAR Activity – Augmented Reality for Live Streaming

View Source Code: DeepARActivity.java

What is DeepAR?

DeepAR is an augmented reality SDK that enables real-time face tracking, filters, and effects on mobile devices. When combined with Ant Media Server, you can stream AR-enhanced video to your audience, creating engaging and fun live experiences.

The DeepAR Streaming Experience

The DeepARActivity brings the power of augmented reality to your live streams. Imagine going live with a Viking helmet, neon devil horns, or even an elephant trunk – all rendered in real-time and streamed to your viewers.

How It Works (High Level)

The camera captures your video feed
Each frame passes through the DeepAR SDK, which applies face tracking and the selected AR effect
The processed frame is sent to Ant Media Server via WebRTC
Your viewers see the AR-enhanced stream in real-time with ultra-low latency

DeepAR Viking Helmet Effect: The Viking Helmet, neon horn effects in action – one of many AR filters available.

Use Cases

Entertainment Streaming: Add fun filters to engage your audience during live shows
Gaming Streams: React to gameplay with expressive emotion effects
Virtual Events: Create memorable virtual appearances with unique AR effects
Social Streaming: Stand out with creative filters on your live broadcasts

Code Overview

Let’s take a brief look at how the DeepARActivity is structured:

Initialization

The activity initializes DeepAR with a license key and sets up the camera and streaming components:
`deepAR = new DeepAR(this);
deepAR.setLicenseKey("your-license-key");
deepAR.initialize(this, this);

initializeEffects();
setupCamera();
setupStreamingAndPreview();`

WebRTC Client Setup

The Ant Media WebRTC client is configured to use a custom video source, which allows us to feed AR-processed frames:
webRTCClient = IWebRTCClient.builder() .setServerUrl("wss://test.antmedia.io/LiveApp/websocket") .setActivity(this) .setVideoSource(IWebRTCClient.StreamSource.CUSTOM) .setWebRTCListener(createWebRTCListener()) .setInitiateBeforeStream(true) .build();

Camera Frame Processing

Each camera frame is captured via CameraX’s ImageAnalysis and fed to DeepAR for AR processing:
imageAnalysis.setAnalyzer(ContextCompat.getMainExecutor(this), image -> { try { feedDeepAR(image); // Send frame to DeepAR SDK } finally { image.close(); } });

Effect Switching

Users can cycle through effects using simple navigation methods:

`public void nextEffect(View v) {
currentEffect = (currentEffect + 1) % effects.size();
deepAR.switchEffect("effect", getFilterPath(effects.get(currentEffect)));
}

public void previousEffect(View v) {
currentEffect = (currentEffect - 1 + effects.size()) % effects.size();
deepAR.switchEffect("effect", getFilterPath(effects.get(currentEffect)));
}`

Stream Control

Starting and stopping the stream is handled with a simple toggle:

public void startStopStream(View v, String streamId) { if (!webRTCClient.isStreaming(streamId)) { ((Button) v).setText("Stop"); webRTCClient.publish(streamId); } else { ((Button) v).setText("Start"); webRTCClient.stop(streamId); } }

The DeepARRenderer handles the OpenGL rendering and sends processed frames to the WebRTC client for streaming.

Section 2: Custom Canvas Activity – Branded Overlays for Professional Streams

View Source Code: CustomCanvasActivity.java

Why Custom Overlays?

While AR effects are fun, sometimes you need professional branding elements on your stream – your logo, text announcements, watermarks, or promotional graphics. The CustomCanvasActivity provides exactly this capability.

The Custom Overlay Experience

This activity demonstrates how to add static visual elements on top of your camera feed before streaming. Think of it as having your own broadcast graphics system built right into your app.

Built-in Overlay Features

The sample implementation includes two types of overlays:

Image Overlay

Displays a custom image (logo or branding graphic)
- set custom image positionx
- set custom image size
- Perfect for watermarks and brand logos
Text Overlay
- Renders custom text directly on the video feed
- Customizable font size (64pt in the sample)
- Custom colors (red text in the sample)
- Set custom position.
- Great for titles, announcements, or hashtags

Custom Overlay Demo Camera feed with logo overlay and text overlay applied.

How It Works (High Level)

CameraX captures your camera feed
An OpenGL ES renderer processes each frame
Custom overlays (images and text) are composited onto the video
The final composited frame streams to Ant Media Server via WebRTC
Viewers receive your branded stream in real-time

Use Cases

Corporate Streaming: Add company logos and branding to internal broadcasts
Educational Content: Display titles, chapter names, or key points
News & Media: Show channel branding and lower-thirds
Product Launches: Overlay promotional text and graphics
Influencer Streams: Watermark your content with your brand

Code Overview

Let’s explore how the CustomCanvasActivity brings overlays to your stream:

WebRTC Client Setup

Similar to DeepAR, we configure the WebRTC client to use a custom video source:

webRTCClient = IWebRTCClient.builder() .setServerUrl("wss://test.antmedia.io/LiveApp/websocket") .setActivity(this) .setVideoSource(IWebRTCClient.StreamSource.CUSTOM) .setWebRTCListener(createWebRTCListener()) .setInitiateBeforeStream(true) .build();

Creating Overlays

Overlays are created when the OpenGL surface is initialized. You can add image overlays and text overlays with custom positioning:

`imageProxyRenderer = new ImageProxyRenderer(webRTCClient, this, surfaceView, new CanvasListener() {
@override
public void onSurfaceInitialized() {
// Image overlay: positioned at 80% X, 80% Y with 20% size
logoOverlay = new Overlay(getApplicationContext(), R.drawable.test, 0.8f, 0.8f);
logoOverlay.setSize(0.2f);

    // Text overlay: "Hello" in red, 64pt font, positioned at center X, -30% Y
    textOverlay = new Overlay(getApplicationContext(), "Hello", 64, Color.RED, 0f, -0.3f);
    textOverlay.setSize(0.12f);
}

});`

Camera Frame Processing

Each camera frame is submitted to the renderer for overlay compositing:

imageAnalysis.setAnalyzer(ContextCompat.getMainExecutor(this), new ImageAnalysis.Analyzer() { @Override public void analyze(@NonNull ImageProxy image) { imageProxyRenderer.submitImage(image); // Send frame to renderer if (surfaceView != null) { surfaceView.requestRender(); // Trigger OpenGL render } image.close(); } });

Camera Switching

Switching between front and back cameras is handled by the CameraProviderHelper:

switchCam.setOnClickListener(new View.OnClickListener() { @Override public void onClick(View v) { cameraProviderHelper.switchCamera(imageAnalysis); } });

The ImageProxyRenderer handles the OpenGL compositing of overlays onto the camera frames before sending them to the WebRTC client.

The Ant Media Server Advantage

Both activities connect to Ant Media Server using WebRTC, which provides:

Ultra-Low Latency: Sub-second delay for real-time interaction
Scalability: Handle thousands of concurrent viewers
Cross-Platform Playback: Viewers can watch on any device or browser
Adaptive Bitrate: Automatic quality adjustment based on network conditions

Conclusion

Whether you’re looking to add fun AR effects to engage your audience or professional branding to your streams, Ant Media Server provides the foundation for high-quality, low-latency broadcasts. The DeepARActivity and CustomCanvasActivity demonstrate just how easy it is to elevate your live streaming experience on Android.

The best part? Both approaches can be customized and extended to match your specific needs. Add your own AR effects, design custom overlays, or combine both techniques for the ultimate streaming experience.

Ready to take your live streams to the next level? Clone the project and start experimenting today!

What’s New in Ant Media Server v2.17.0?

Amar Thodupunoori — Wed, 25 Feb 2026 08:49:07 +0000

Ant Media Server v2.17.0 reflects our ongoing goal to make Ant Media Server extensible and to build a strong AMS ecosystem. This allows both AMS users and our internal team to develop new plugins and add new features.

We are continuously adding new plugins into the AMS ecosystem. At the same time, we continuously improve Ant Media Server to simplify its usage and enhance the overall user experience.

We release new versions on a quarterly basis, and v2.17.0 is the Q4 2025 release.

In this post, I’ll walk through the key features and improvements introduced in v2.17.0.

Table of Contents

Professional Ad Insertion Without the Hassle
A Simpler, More Stable Web SDK
Low Latency HLS at Scale
Conlusion

Professional Ad Insertion Without the Hassle

SSAI with SCTE-35

The first new feature in v2.17.0 is Server Side Ad Insertion with SCTE-35. Monetizing live streams is important feature for the content providers and event broadcasters. According to the demads from Ant Media Server user, we have added SCTE-35 support in AMS.

It shortly works like this:

The server automatically detects SCTE-35 ad markers coming from SRT stream.

These markers are converted into standard HLS cue tags in real time.

Ads can be inserted using platforms like Google Ad Manager or AWS MediaTailor, directly into the stream.

Because ads are stitched on the server side, viewers don’t see buffering or player reloads. Ads play smoothly, just like the content itself—and they’re much harder to skip or block.

If you want to learn more about this feature and understand how it works on AMS, please check this detailed blog post on SCTE-35 ad insertion

A Simpler, More Stable Web SDK

Web SDK v2

We rebuilt our JavaScript SDK with one clear goal: make it easier to use and easier to trust.

The SDK now fully supports async/await so your code is cleaner and easier to follow.

Connection and reconnection handling is more reliable, even on unstable networks.

The SDK is modular, so you only include what you actually need.

This means fewer bugs, smaller bundles, and faster development.

You may reach the new Web SDK in v2 folder of the web applications deployed with AMS installation.

We will continue to support the old Web SDK till the users adapt the new version.

Low Latency HLS at Scale

Among the small improvements and fixes in the new release, one of the most important one is improvements in the Low Latency HLS.

Firstly, Low Latency HLS is no longer limited to a single server. You can now deliver low-latency streams across a cluster while keeping delays around 2–5 seconds—even for very large audiences. Streams are distributed across multiple nodes in the cluster.

Also we had improvements on Adaptive Bitrate (ABR) with LL-HLS playback. You feel smooth switches among the different resolutions while playing your stream with LL-HLS.

Conlusion

I have shared the recent key features in the lastest release v2.17.0 above. You can check the other improvements and fixes list here.

We are planning to release a new version each quarter. The next big release will be at the end of the March. See you then.

Run Ant Media Server on Azure — Free Credits Available Through Azure Sponsorship 2026

Amar Thodupunoori — Wed, 18 Feb 2026 10:34:35 +0000

Like every year, Ant Media Server is proud to participate in the Microsoft Azure Sponsorship Program in 2026, offering free infrastructure credits to qualified users.

Azure Sponsorship Program will grant qualified users access to complimentary infrastructure for thorough evaluations of Ant Media Server Enterprise Edition on Azure Marketplace. This program offers evaluation subscriptions to Azure for up to 90 days for qualified proofs of concept (PoC).

Azure Sponsorship

As part of the Microsoft Azure Sponsorship program and in coordination with Microsoft, Ant Media is offering up to $1,000 of free Azure credits of infrastructure for assessments, proof of concepts, and deployments for each applicant. This program could also help for the clients who plans to migrate from Azure Media Services to Ant Media Server in Azure Marketplace.

Available to eligible applicants, who require to have PAYG or EA Subscription on Azure, these funds can help you to experience the live streaming and VoD services with Ant Media Server, and also integration capability with your backend to deliver your use case.

Learn more about Ant Media Server Enterprise Edition Cluster in Azure
To learn more about running Ant Media Enterprise Edition Cluster on Azure, please refer to read this guide.

Learn more about the Microsoft Azure Sponsorship Program
Interested in taking advantage of the Azure Sponsorship Program? Send us a note at contact@antmedia.io to be connected with an Ant Media Solution specialist who can help you get started.

Learn more about how to migrate from Azure Media Services to Ant Media Server
To learn more about switching from Azure Media Services to Ant Media Server Enterprise Edition in Azure Marketplace, please refer to read this guide.

Please just note that Azure Sponsorship can work only on pay-as-you-go (PAYG) or Enterprise Agreement (EA) Azure subscriptions. We cannot add free Azure on any other Azure subscription type (e.g., MCA, through CSP) or to accounts that already have free Azure added to them.

How to Enable SSL Certificate for Your Ant Media Server with 1 Command?

Amar Thodupunoori — Wed, 11 Feb 2026 10:36:59 +0000

Enabling SSL on Ant Media Server is essential for securing communications and supporting modern browser requirements. While SSL is not mandatory for all streaming use cases, it is required for accessing microphones and cameras, and for running WebRTC and WebSocket-based applications in browsers such as Google Chrome.

Ant Media Server provides multiple ways to enable SSL, including automatic certificate generation with Let’s Encrypt, free subdomain support for Enterprise users, and the ability to import custom certificates. Depending on your setup, SSL can be enabled either through the Web Panel or via command-line tools.

Port 80 Must Be Available
Domain Must Point to the Server
Option 1: Enabling SSL from the Web Panel. Steps to Enable SSL from the Web Panel.
Option 2: Get a Free Subdomain and Install SSL with Let’s Encrypt. Steps to Enable Free Subdomain SSL.
Option 3: Create a Let’s Encrypt Certificate with Your Domain
Prerequisites
Installation Steps.
Option 4: Use Your Own SSL Certificates.
Required Certificate Files.
Installation Command.

Prerequisites for SSL Configuration

Ant Media Server uses Let’s Encrypt to generate free SSL certificates. Before enabling SSL, ensure that the following requirements are met.

1. Port 80 Must Be Available

Let’s Encrypt uses port 80 to verify domain ownership. If another service is running on this port, SSL certificate generation will fail.

If Apache is running, stop it temporarily:

sudo service apache2 stop
Important: Make sure that your domain points to your server’s public IP address in the DNS records before running enable_ssl.sh script.

2. Domain Must Point to the Server

If you are using your own domain, make sure that the domain’s DNS A record points to your server’s public IP address before running the SSL script. DNS changes must be fully propagated before proceeding.

SSL Configuration Methods

Option 1: Enabling SSL from the Web Panel
Starting with Ant Media Server version 2.6.2, SSL can be enabled directly from the Web Panel without using the command line. This is the recommended method for most users.

Steps to Enable SSL from the Web Panel

Open the Ant Media Server Web Panel in your browser.

Navigate to Settings → SSL.
In the Type drop-down menu, select the SSL option that matches your setup:
Use your own domain with Let’s Encrypt
Get a free subdomain (*.antmedia.cloud)
Import your own SSL certificate
Click Activate to enable SSL.
Once activated, SSL is applied automatically. No server restart is required. Simply refresh your browser and log in again using HTTPS.

Enabling SSL from the Web Panel
SSL Configuration in Web Panel
Gets a free subdomain and install SSL

Option 2: Get a Free Subdomain and Install SSL with Let’s Encrypt

Ant Media Server Enterprise provides the ability to obtain a free subdomain and automatically install a Let’s Encrypt SSL certificate using a single command. This option is useful if you do not already own a domain name.

This feature is available for Enterprise Edition users starting from version 2.5.2 and later.

When enabled, Ant Media Server assigns a subdomain in the following format:

ams-<id>.antmedia.cloud

Steps to Enable Free Subdomain SSL
Navigate to the Ant Media Server installation directory:

cd /usr/local/antmedia
Run the SSL enablement script without any parameters:

sudo ./enable_ssl.sh
The script will:

Request a free subdomain
Generate a Let’s Encrypt SSL certificate
Configure Ant Media Server to use HTTPS and WSS

Note: The free subdomain option is not accessible for Ant Media Server marketplace images on AWS, Azure, or Alibaba as it requires a license key.

Option 3: Create a Let’s Encrypt Certificate with Your Domain

Use this option if you already own a domain name and want to secure Ant Media Server with a Let’s Encrypt SSL certificate for that domain.

Prerequisites
Your domain’s DNS A record points to the server’s public IP address
Port 80 is available and not used by another service

Installation Steps
Navigate to the Ant Media Server installation directory:

cd /usr/local/antmedia
Run the SSL enablement script with your domain name:

sudo ./enable_ssl.sh -d example.com
After successful completion, access Ant Media Server securely using HTTPS on port 5443:

https://example.com:5443

Option 4: Use Your Own SSL Certificates

Use this option if you already have an SSL certificate issued by a third-party certificate authority and want to configure Ant Media Server with your own certificate files.

Required Certificate Files
Make sure you have the following files available in PEM format:

fullchain.pem – Full certificate chain
privkey.pem – Private key
chain.pem – Certificate chain

Installation Command
Navigate to the Ant Media Server installation directory:

cd /usr/local/antmedia
Run the SSL enablement script with your certificate files:

sudo ./enable_ssl.sh -f {FULL_CHAIN_FILE} -p {PRIVATE_KEY_FILE} -c {CHAIN_FILE} -d {DOMAIN_NAME}
Example usage:

sudo ./enable_ssl.sh -f yourdomain.crt -p yourdomain.key -c yourdomainchain.crt -d yourdomain.com

Important Considerations

Port 80 must be available during certificate generation
DNS records must be properly configured and propagated
Free subdomain option requires a valid Enterprise license
Marketplace images on AWS, Azure, or Alibaba Cloud do not support free subdomains
After SSL enablement, HTTPS traffic is served on port 5443 by default

Frequently Asked Questions

Is SSL mandatory for Ant Media Server?
SSL is not mandatory for all use cases. However, HTTPS and WSS are required for accessing the microphone and camera in modern browsers, and for running WebRTC and WebSocket-based applications, especially in Google Chrome.
Which Ant Media Server editions support free SSL and subdomains?
The free subdomain and automatic SSL installation feature is available for Enterprise Edition users starting from version 2.5.2 and later. This feature is not available for marketplace images on AWS, Azure, or Alibaba Cloud.
Why must port 80 be open when enabling SSL?
Let’s Encrypt uses port 80 to verify domain ownership during certificate generation. If another service is using or forwarding port 80, SSL certificate creation will fail. The service can be restarted after the SSL setup is completed.
Which HTTPS port does Ant Media Server use after SSL is enabled?
After SSL is enabled, Ant Media Server serves HTTPS traffic on port 5443 by default. You can access the server securely using a URL such as https://your-domain.com:5443.
Can I use my own SSL certificate instead of Let’s Encrypt?
Yes. Ant Media Server allows you to use your own SSL certificates by providing the full chain certificate, private key, and certificate chain files in PEM format using the enable_ssl.sh script.

Conclusion

Enabling SSL on Ant Media Server is a critical step for securing communications and meeting modern browser requirements, especially for WebRTC and WebSocket-based applications. Depending on your setup, SSL can be enabled through the Web Panel, automatically with a free subdomain, by generating a Let’s Encrypt certificate for your own domain, or by importing custom SSL certificates.

Ant Media Server provides flexible SSL configuration options to accommodate different deployment scenarios, from quick setups to enterprise-grade environments. By selecting the appropriate SSL method and ensuring required prerequisites such as port availability and DNS configuration, SSL can be enabled reliably and with minimal effort.

Proper SSL configuration not only improves security but also ensures compatibility with modern browsers and real-time streaming technologies, providing a solid foundation for production-ready streaming deployments. Please let us know if you have a question or need help with this issue or any other one.

Bitrate vs. Resolution: 4 Key Differences and Their Role in Video Streaming.

Amar Thodupunoori — Wed, 04 Feb 2026 10:02:45 +0000

Video bitrate vs resolution is one of the most common — and misunderstood — concepts in video streaming, directly impacting video quality, bandwidth usage, and viewer experience.

Whether you’re streaming a live sports event, producing video content for YouTube, or hosting a business webinar, your audience expects sharp visuals and smooth, lag-free playback. Achieving this goes beyond investing in the latest camera or microphone—the key lies in understanding and balancing video bitrate and resolution.

These two terms, video bitrate vs resolution, are often confused, and while both influence video quality, they are not the same. Think of them as two sides of the same coin: resolution determines how much visual detail your video contains, while bitrate controls how efficiently that detail is transmitted over streaming protocols such as WebRTC and HLS. Add frames per second (FPS) into the equation, and video quality becomes even more dynamic.

This guide breaks down bitrate, resolution, and FPS — how they differ, how they work together, and how you can choose the right settings for your content.

Table of Contents
Video Bitrate vs Resolution: What Is Video Bitrate?
Video Bitrate vs Resolution: What Is Video Resolution?
What is FPS (Frames Per Second)?
Video Bitrate vs Resolution: 4 Key Differences Explained

Units of Measurement
Quality
Compression
Settings Encoding Methods: CBR vs VBR What are Codecs? Potential Streaming Problems (and Fixes) Platform-Specific Bitrate Settings Frequently Asked Questions What’s the optimal bitrate for 1080p streaming? Why does my stream look pixelated despite the high bitrate? Can I stream 4K on a 10 Mbps upload connection? How does WebRTC achieve such low latency? Should I prioritize resolution or frame rate for sports content? Conclusion: Beyond Basic Streaming Video Bitrate vs Resolution: What Is Video Bitrate? Video Bitrate vs Resolution Video bitrate is the amount of data transmitted per second in a video stream. It is typically measured in megabits per second (Mbps) for video and kilobits per second (kbps) for audio. In simple terms, bitrate represents how much information your video carries every second.

Higher bitrate → More data per second, larger file size, and higher potential quality.
Lower bitrate → Less data, smaller file size, but a greater risk of blurry or pixelated playback.
For ultra-low latency streaming scenarios—such as WebRTC—bitrate optimization becomes even more critical, since every millisecond of delay directly affects the viewing experience.

Example: If you live-stream a webinar at 5 Mbps, that means five million bits of video data are sent to your viewers every second.

Why it matters: Bitrate has a direct impact on how smooth and clear your video looks and sounds. If the bitrate is too low, the result is grainy visuals and poor audio. If the bitrate is too high for a viewer’s internet connection, they’ll experience buffering. The key is striking the right balance between video quality and network capacity.

Video Bitrate vs Resolution: What Is Video Resolution?
Video Resolution
Video resolution refers to the number of pixels that make up your video’s width and height. More pixels mean more detail and sharper visuals.

Common resolutions include:

720p (HD): 1280 × 720 pixels
1080p (Full HD): 1920 × 1080 pixels
4K (Ultra HD): 3840 × 2160 pixels
Example: A 1920 × 1080 (1080p) video contains over 2 million pixels per frame, which is why it looks sharper than a 720p video.

For advanced applications, 4K at 60 FPS streaming can deliver exceptional clarity. However, with protocols like WebRTC, maintaining this quality without latency issues requires careful optimization of bitrate, network conditions, and hardware performance.

Why it matters: Resolution defines visual clarity, but resolution alone doesn’t guarantee quality. A 1080p stream at too low a bitrate can look worse than a properly encoded 720p stream. This is why resolution and bitrate must always be balanced together.

What is FPS (Frames Per Second)?
Frames Per Second
FPS refers to the number of frames (individual images) displayed per second in a video. A higher FPS results in smoother motion, while a lower FPS creates a more cinematic or stylized look.

Typical standards include:

24 FPS: Traditional film and cinematic style.
30 FPS: Common for video calls, YouTube uploads, and most online streaming.
60 FPS: Extra smooth motion, ideal for gaming, sports, and fast-paced content. In real-time sports streaming with WebRTC, 60 FPS helps ensure viewers don’t miss critical moments.
120+ FPS: Used in advanced slow-motion capture or professional cinematography.
Example: A 60 FPS stream of a football match feels fluid and lifelike, while 24 FPS may appear slightly choppy in fast motion scenes.

Why it matters: FPS impacts how natural and engaging your video feels. However, higher FPS requires more bitrate to maintain quality. For example, a 1080p 30 FPS video may stream smoothly at around 4 Mbps, while a 1080p 60 FPS video may require 6 Mbps or more to avoid quality loss.

Video Bitrate vs Resolution: 4 Key Differences Explained

Units of Measurement The unit of measurement for video bitrate is megabits per second (Mbps) for video and kilobits per second (kbps) for audio. For example, if a video has a bitrate of 5 Mbps, it means you are transmitting five million bits of video data every second. Audio, being less data-intensive, is usually measured in kbps, with common streaming values ranging from 96 kbps to 320 kbps depending on quality.

Video resolution, in contrast, is measured in pixels (p). Resolution is represented as width × height. For instance, a video with a resolution of 1920 × 1080 has 1920 pixels horizontally and 1080 pixels vertically. In common usage, people often identify videos by the height only—so 1920 × 1080 becomes simply 1080p. Higher resolutions like 4K (3840 × 2160) and 8K (7680 × 4320) continue this same pattern.

Quality A higher bitrate usually results in better video quality, as more data per second allows for richer detail, smoother gradients, and fewer compression artifacts. Conversely, a lower bitrate reduces quality, potentially introducing pixelation, blurriness, or lag.

You might assume that higher resolution automatically means better quality, but that’s not entirely true. Resolution only defines the number of pixels, not how much detail those pixels carry. Network speed and available bandwidth play just as critical a role. For instance, if your Wi-Fi connection is weak, a 1080p video may stutter and buffer, while the same video delivered in 720p at a stable bitrate might play far more smoothly.

It’s also important to consider platform limitations. For example:

YouTube supports resolutions from 240p all the way up to 8K, with adaptive streaming.
Facebook Live limits streams to 720p.
WebRTC (commonly used for real-time video calls) often defaults to lower resolutions like 480p or 720p, unless the network and hardware allow scaling up.
Another tradeoff: higher bitrate = larger file size and longer encoding time. This is critical when exporting or archiving video content.

Compression Both bitrate and resolution can be reduced, but the methods and reasons differ.

Reducing bitrate is essentially a form of compression. The encoder removes or simplifies data to fit the target bitrate. For instance, if you set your encoder to 4 Mbps, each second of video is compressed so that no more than 4 megabits of data are transmitted. Ideally, this is done without noticeable quality loss. However, excessive compression can introduce artifacts such as blockiness, ghosting, or banding.

The impact is more noticeable in high-motion content (e.g., sports, gaming), where the difference between 1.5 Mbps and 5 Mbps is very clear. For static content (like a lecture or news broadcast), the human eye may not detect much difference at lower bitrates.
Reducing resolution, on the other hand, lowers the pixel count being transmitted. This doesn’t just reduce the size of each frame but also lowers the overall data needed per second. For viewers with slower internet connections, dropping from 1080p to 720p (or even 480p) can dramatically improve playback smoothness.
Modern video delivery uses Adaptive Bitrate Streaming (ABR) to adjust both bitrate and resolution automatically based on each viewer’s connection. This ensures that someone on a fast fiber line can watch in 1080p at a high bitrate, while someone on a weaker mobile connection may seamlessly fall back to 480p at a lower bitrate.

Settings Resolution settings are often handled automatically by the video player or platform. For example, YouTube adjusts playback resolution to match the viewer’s device screen and available bandwidth. A smartphone might default to 720p, while a 4K TV will display higher resolutions if the network can handle it.

Bitrate settings, however, are usually configured by the content creator, encoder, or streaming server. You define how much data per second the encoder should output. Setting the optimum bitrate is critical:

Too high, and you waste bandwidth, stress hardware, and risk buffering for viewers on slower networks.
Too low, and you sacrifice detail, causing a poor viewing experience.
As a guideline:

Full HD (1080p) video usually needs 3,500–5,000 kbps for standard quality.
High-motion 1080p may require 4,500–6,000 kbps.
4K streaming can demand 15,000–25,000 kbps depending on codec efficiency.
A correctly tuned bitrate also improves audio quality since streaming platforms often allocate separate bandwidth for audio streams. For businesses, this matters greatly—clear voice and minimal latency in video calls mean smoother collaboration and stronger client relationships.

With Adaptive Bitrate Streaming (ABR), many modern platforms allow you to configure multiple renditions (e.g., 480p at 1 Mbps, 720p at 2.5 Mbps, 1080p at 5 Mbps), and the streaming software automatically serves the best version for each user. On platforms like Ant Media Server, you can configure ABR settings directly through the web panel to automate these optimizations, ensuring every viewer gets the best possible experience without manual tweaking.

Encoding Methods: CBR vs VBR
Encoding is the process of converting raw video files captured by your camera into compressed digital video files that can be transmitted efficiently over the internet. Without encoding, the raw video data would be far too large to stream in real time.

Understanding the difference between encoding and transcoding is critical when optimizing your streaming setup:

Encoding: The initial conversion of raw video into a digital format (e.g., H.264, H.265, VP9, AV1).
Transcoding: Re-encoding an already encoded video—often into multiple versions with different resolutions and bitrates for adaptive delivery. For more details, check this Article on Transcoding.
When configuring your encoder, one of the most important choices is how to handle bitrate control. There are two main types of bitrate encoding: Constant Bitrate (CBR) and Variable Bitrate (VBR).

Constant Bitrate (CBR) CBR encoding maintains a fixed bitrate throughout the video, regardless of scene complexity.

Pros:
Creates consistent file sizes, which makes bandwidth usage predictable.
Faster and more efficient to encode than VBR.
Ensures steady audio quality, which is particularly useful for real-time multimedia streaming (e.g., webinars, video calls, live events).
Cons:
Offers less flexibility—it uses the same amount of data for simple static scenes as it does for complex high-motion scenes.
Requires viewers to have a strong, stable internet connection; buffering may occur.
CBR is best suited when latency and stability matter more than efficiency, such as in live streaming with WebRTC, RTMP ingest, or video conferencing.

Variable Bitrate (VBR) VBR encoding adjusts the bitrate dynamically, allocating more bits to complex scenes and fewer to static ones. The goal is to keep video quality consistent while saving bandwidth.

Pros:
Optimized quality: Complex scenes (like sports or gaming) get more bits, while static scenes (like interviews) use fewer.
Produces smaller file sizes overall compared to CBR at the same quality.
Better suited for on-demand video where encoding speed isn’t critical.
Cons:
Encoding takes longer than CBR, making it less ideal for real-time use cases.
Less predictable file sizes and bandwidth needs.
Not as widely supported in some older streaming systems compared to CBR.
Many platforms also use Capped VBR, where the bitrate is allowed to fluctuate but within a set maximum limit, offering a compromise between efficiency and predictability.

Multi-Bitrate Streaming If you’re serving a diverse audience across different devices and networks, using a single fixed bitrate isn’t practical. For example, imagine you’ve created a video explaining “What is a .ai domain?” Viewers may watch it on:

A mobile device with 4G,
A laptop on public Wi-Fi,
Or a smart TV with a high-speed fiber connection.
Each viewer has different bandwidth capabilities. Multi-bitrate streaming solves this problem by generating several versions of the same video at different bitrates and resolutions (e.g., 480p at 1 Mbps, 720p at 2.5 Mbps, 1080p at 5 Mbps).

Adaptive Bitrate Streaming (ABR) Multi-bitrate really shines when paired with adaptive video players. Adaptive Bitrate Streaming automatically detects each viewer’s available bandwidth and device performance, then delivers the most appropriate stream in real time.

If the viewer’s connection improves, the player may seamlessly switch them from 720p to 1080p.
If the connection weakens, it can drop to 480p to prevent buffering.
Modern delivery protocols such as CMAF (Common Media Application Format) and LL-HLS (Low-Latency HLS) make ABR streaming even more efficient by reducing overhead and improving latency, which is critical for real-time interactivity.

What are Codecs?
‘Codec’ is a portmanteau word combining ‘coder’ and ‘decoder’. It’s a device that turns raw recorded video data into a manageable and viewable format by processing and encoding it for storage, so it can be decoded for watching or editing. Without a codec, raw camera footage would be far too bulky to handle efficiently.

You’ve probably come across MPEG (.mp3) and MPEG-4 (H.264, AAC) for audio and video. These create small files without losing quality and can be played on almost any platform and media player. The new MPEG codec “HEVC” (or h.265) for smartphones allows even better compression. For detailed technical implementation, explore H.264, VP8, and H.265 codec support in modern streaming servers.

Potential Streaming Problems (and Fixes)
Low bitrate can be caused by network congestion, an unstable connection to your ISP, or streaming over wi-fi. When your connection is unable to keep up, it will drop frames to improve the stability of your stream and minimize latency.

Low internet speed and bandwidth can also slow the transfer of video data. The minimum upload speed required for streaming 1080p video is 2.75 Mbps, but higher is better. For video calls, you can improve the sound quality by using Bluetooth headsets with a mic.

If your stream looks pixelated or choppy, comprehensive WebRTC troubleshooting and optimization guides can help identify and resolve technical issues quickly.

Platform-Specific Bitrate Settings
Platform Max Resolution Recommended Bitrate Notes
YouTube 4K @ 60 FPS 3,000 – 20,000 kbps Supports HDR, adaptive streaming
YouTube 4K @ 60 FPS 3,000 – 20,000 kbps Supports HDR, adaptive streaming
Twitch 1080p @ 60 FPS 4,500 – 6,000 kbps Streamers limited by partner status
Facebook 720p @ 30 FPS 2,500 – 4,000 kbps Lower the cap to reduce bandwidth
Zoom/Meet 720p / 1080p Auto-adjusts Prioritizes audio stability
Netflix 4K UHD 15–25 Mbps Uses adaptive ABR streaming
While traditional platforms impose these limitations, WebRTC’s sub-second latency advantage enables real-time interaction that transforms viewer engagement beyond what conventional streaming protocols can achieve.

Frequently Asked Questions
What’s the optimal bitrate for 1080p streaming?
For live streaming, 3,500-5,000 kbps delivers professional quality. For VOD, 8,000+ kbps maximizes quality. Ant Media Server automatically optimizes based on content complexity.

Why does my stream look pixelated despite the high bitrate?
Three common causes: keyframe interval too long (should be 1-2 seconds), codec mismatch, or encoding preset too fast. Ant Media Server auto-configures optimal encoding parameters.

Can I stream 4K on a 10 Mbps upload connection?
Technically possible at 8-9 Mbps, but leaves no headroom for network fluctuations. We recommend 25 Mbps minimum for reliable 4K streaming. Ant Media Server can dynamically downgrade to 1080p if network conditions deteriorate.

How does WebRTC achieve such low latency?
WebRTC eliminates traditional streaming’s segmentation delay by transmitting data peer-to-peer when possible. You can learn more about WebRTC’s peer-to-peer architecture and its technical advantages. Ant Media Server’s WebRTC implementation achieves 200-500ms latency consistently.

Should I prioritize resolution or frame rate for sports content?
Frame rate typically matters more for sports. 720p60 often provides a better viewing experience than 1080p30 for fast action. Ant Media Server can deliver both simultaneously, letting viewers choose.

For practical implementation, you can embed WebRTC streaming into websites with just a few lines of code, making high-quality streaming accessible to any application.

Conclusion: Beyond Basic Streaming
Success in modern streaming isn’t about cranking every setting to the max — it’s about smart optimization. While many platforms still wrestle with legacy RTMP and painful 15-second delays, Ant Media Server turns technical choices into real business advantages:

Sub-second latency with WebRTC for real-time interaction.
Adaptive bitrate so every viewer gets the best experience their network allows.
Efficient compression that saves bandwidth costs without compromising quality.
Whether you’re streaming to a hundred viewers or scaling to millions, Ant Media Server grows with your ambitions. From ultra-low latency WebRTC to scalable HLS delivery, it’s the platform professionals trust when “good enough” isn’t good enough.

Ready to see the difference? Start your free trial today and experience how optimized bitrate, resolution, and codecs can transform your streaming.

SCTE-35 Ad Insertion: Easiest Way to Professional Ads

Amar Thodupunoori — Wed, 21 Jan 2026 09:04:52 +0000

SCTE-35 is the industry standard that enables TV networks and professional broadcasters to seamlessly cut away from a live event to a commercial break at exactly the right moment.

It isn’t done manually by someone hitting a switch. It’s fully automated using SCTE-35 markers embedded in the video stream.

If you are building a streaming platform, you likely want to monetize your content. However, Ant Media Server (AMS) does not insert the ads itself. Instead, it acts as the crucial bridge in the ecosystem. It preserves the ad markers from your source stream and passes them downstream so that specialized Server-Side Ad Insertion (SSAI) services (like AWS Elemental MediaTailor) know exactly when to swap your live feed for an ad.

We are excited to announce a new plugin for Ant Media Server that enables full SCTE-35 support, converting SRT stream markers into HLS cues for professional ad workflows.

What is SCTE-35 and Why Does It Matter?
SCTE-35 is the “digital cue card” of the video industry. It signals downstream systems that an event, like an ad break, is about to happen. Without these signals, ad insertion servers are blind; they don’t know when to trigger an ad. The diagram below illustrates how SCTE-35 markers flow through the system:

scte scte-35 hls manifest ad insertion

This plugin solves a specific interoperability challenge:

Ingest: It takes an SRT stream containing SCTE-35 data.
Process: It parses the MPEG-TS payload to find splice commands (Table ID 0xFC).
Output: It injects standard HLS ad markers (#EXT-X-CUE-OUT / #EXT-X-CUE-IN) into the manifest (.m3u8).
Importantly, Ant Media Server does not remove your original video segments. The plugin simply “wraps” your existing content with SCTE markers, so the stream remains playable even without an ad insertion server. Your original segments stay in the manifest as slate or placeholder content.

When an Ad Insertion Server like AWS MediaTailor reads the manifest, it uses these markers to seamlessly stitch ads into the stream, replacing your slate during CUE-OUT and switching back to live content at CUE-IN.

Installation
First of course, you need ant media server installed. Follow the docs here.

The plugin source code is available on GitHub. It can be compiled by using build.sh script available in the repository. Or download compiled .jar here.

Getting the plugin running requires adding the JAR file and registering a filter in your application configuration.

Deploy the Plugin
Copy the plugin JAR file to your application’s plugin directory:
cp SCTE35Plugin.jar /usr/local/ant-media-server/plugins/

Configure the Filter
You must register the SCTE35ManifestModifierFilter in your application’s web.xml. This filter intercepts the HLS manifest generation to inject the tags.
Open /usr/local/ant-media-server/webapps/AppName/WEB-INF/web.xml and add the following entry

SCTE35ManifestModifierFilter
io.antmedia.scte35.SCTE35ManifestModifierFilter
true

SCTE35ManifestModifierFilter
/streams/*

Important: Place this after the HlsManifestModifierFilter entry to ensure the execution order is correct.
Restart and Verify
Restart the server to load the new config:
sudo systemctl restart antmedia

Check the logs (/var/log/antmedia/ant-media-server.log) to confirm successful initialization. You should see:
SCTE-35 Plugin initialized successfully
SCTE35ManifestModifierFilter is properly registered
Preparing Your Source Stream
For production, you simply need to push an SRT stream that already contains SCTE-35 data in its MPEG-TS payload to Ant Media Server. The plugin handles the rest automatically.

However, testing SCTE-35 can be tricky. If you want to run a quick test to verify your pipeline is working, we strongly recommend using a known valid source rather than generating one from scratch.

Why not FFMPEG? During our testing, we found that while FFMPEG handles video well, it often fails to transmit SCTE-35 packets correctly over SRT to the server side.

The Recommended Test Approach: We have provided a pre-baked test file that contains SCTE-35 ad triggers every 2 to 5 minutes. You can use the srt-live-transmit tool to stream this file to AMS reliably.

Download the test stream from this link
Stream it using srt-live-transmit:
cat scte35_spliceInsert_2hour_demo.ts | pv -L 19K | srt-live-transmit file://con "srt://your-server:4200?streamid=WebRTCAppEE/your_stream"
Stream the file with a bitrate limit to match the content
Integration: Testing with an Ad Insertion Server
Once your stream is running in AMS, the HLS manifest will start populating with SCTE tags. Any SSAI (Server-Side Ad Insertion) platform that supports HLS with SCTE-35 markers will work—such as AWS MediaTailor, Google Ad Manager, Broadpeak, or Yospace. In this example, we’ll use AWS MediaTailor to demonstrate the integration. When SCTE cue get hit in media timeline, you should see something like this in the .m3u8 file:

EXTINF:6.000,segment001.ts

EXT-X-DISCONTINUITY

EXT-X-CUE-OUT:30.000

EXTINF:6.000,segment002.ts

To see the ads in action:

Create a Configuration in MediaTailor

Video Content Source: http:///YourAppName/
Important: MediaTailor requires port 80 (HTTP). Ensure your AMS is accessible via standard HTTP.
Set Ad Decision Server URL: You can use a standard VAST/VMAP test URL (like TheoPlayer’s demo VAST for verification.

Playback with ads:

MediaTailor replaces the AMS base URL with its own. Append your stream path to the MediaTailor URL:

Original: http:///YourAppName/streams/stream1.m3u8
MediaTailor: https:///streams/stream1.m3u8
Summary
This plugin opens the door for broadcast-grade monetization on Ant Media Server. By bridging the gap between SRT ingest and HLS-based ad insertion, you can now integrate seamlessly with the industry’s leading SSAI tools.

In this blog post, we explored how to use the SCTE-35 plugin to enable server-side ad insertion with Ant Media Server. We hope this guide helps you get started with broadcast-grade monetization for your streams. If you have any questions, please feel free to contact us via contact@antmedia.io.

DRM Plugin for Ant Media Server: Secure Streaming with Widevine, FairPlay & PlayReady

Amar Thodupunoori — Wed, 10 Dec 2025 09:06:21 +0000

DRM Plugin for Ant Media Server

The Digital Rights Management (DRM) Plugin for Ant Media Server enables secure streaming by integrating with the CPIX (Content Protection Information Exchange) API. It ensures that only authorized users can access your content through encryption and multi-DRM support (Widevine, FairPlay, and PlayReady).

Key Features

Integration with CPIX API for content key management
Support for both DASH and HLS outputs
Multi-DRM support: Widevine, FairPlay, PlayReady

Installation

Prerequisites

Ensure the Ant Media Server is already installed and running on your machine or server instance.

1. Purchase and Install the DRM Plugin

Install the plugin JAR file into your Ant Media plugins directory:


bash
sudo cp DRM-Plugin-bundle.jar /usr/local/antmedia/plugins
Restart Ant Media Server:

bash
Copy code
sudo service antmedia restart
2. Install Shaka Packager
Download Shaka Packager:

bash
Copy code
wget https://github.com/shaka-project/shaka-packager/releases/download/v3.4.1/packager-linux-x64 -O shakapackager
Move it to /usr/local/bin and make it executable:

bash
Copy code
sudo cp shakapackager /usr/local/bin/
sudo chmod +x /usr/local/bin/shakapackager
Configuration
DRM plugin configurations are managed under customSettings in your Ant Media Server application settings.

1. Navigate to Custom Settings
Open the Ant Media Server web panel

Select your application (e.g., live)

Go to Settings → Advanced

Locate customSettings

2. Add DRM Settings
Minimal Configuration
json
Copy code
{
  "customSettings": {
    "plugin.drm-plugin": {
      "enabledDRMSystems": ["Widevine"],
      "keyManagementServerURL": "{KMS_URL}"
    }
  }
}
Enable Multiple DRM Systems
json
Copy code
{
  "enabledDRMSystems": [
    "Widevine",
    "PlayReady",
    "FairPlay"
  ]
}
Available Configuration Fields
Field   Description
keyManagementServerURL  URL for CPIX key retrieval from DRM provider
enabledDRMSystems   List of DRM systems such as Widevine, FairPlay, PlayReady
encryptionScheme    "cbcs" (default) or "cenc"
hlsPlayListType LIVE (default), VOD, or EVENT
segmentDurationSecs Segment duration in seconds (default: 2)
timeShiftBufferDepthSecs    Live buffer depth (default: 60)
segmentsOutsideLiveWindow   Extra segments outside buffer (default: 5)

Multi-DRM Integration (Widevine via DoveRunner)
1. Obtain KMS Token from DoveRunner
In the DoveRunner dashboard, go to DRM Settings and copy your KMS token.

Construct the Key Management Server URL:

bash
Copy code
https://kms.pallycon.com/v2/cpix/pallycon/getKey/{YOUR_KMS_TOKEN}
Update your customSettings to include this URL.

2. Add HTML5 Player (Video.js Example)
Clone the DRM-enabled HTML5 player samples:

bash
Copy code
git clone https://github.com/doverunner/html5-player-drm-samples
sudo cp -r html5-player-drm-samples /usr/local/antmedia/webapps/live/
Copy required .html, .js, and .css files into your application web folder.

3. Publish a WebRTC Stream
Publish a WebRTC stream using Ant Media’s built-in WebRTC publish page.

Your output directory should contain:

master.mpd

master.m3u8

Playback URLs
DASH:

ruby
Copy code
https://{YOUR_ANTMEDIA_SERVER}:5443/live/streams/drm/stream007/master.mpd
HLS:

ruby
Copy code
https://{YOUR_ANTMEDIA_SERVER}:5443/live/streams/drm/stream007/master.m3u8
Generate a Widevine token using DoveRunner’s Token Generator.

Update your player's JS config with the token and stream URLs.

Open the player page — your DRM-protected stream should play successfully.

To verify DRM protection, try taking a screenshot. The player should block screen capture.

🎉 Congratulations
You have successfully:

Installed and configured the DRM Plugin

Integrated Widevine DRM using DoveRunner

Published a DRM-protected live stream

Tested playback using a DRM-enabled HTML5 player

One Stream, Every Platform: Multi-streaming Made Easy

Amar Thodupunoori — Wed, 03 Dec 2025 10:41:02 +0000

Simulcasting lets you publish a single live stream to multiple platforms—YouTube, Facebook, Twitch, and more—all at once.
This way, you expand your audience without increasing your broadcasting workload.

Ant Media Server makes this process simple and efficient. Let’s walk through how to set it up.

🎯 What You Need

Ant Media Server (Community or Enterprise Edition)
A live stream publishing tool (OBS, Larix, WebRTC, etc.)
Stream keys / RTMP endpoints from the platforms you want to broadcast to

🔧 Step 1: Start a Live Stream

Start streaming to your Ant Media Server using:

WebRTC Publish
RTMP Encoder (e.g., OBS, vMix)
SRT Publish

Once your stream is live, you can attach external RTMP endpoints for simulcasting.

➕ Step 2: Add Simulcast Targets

You can do this in two ways:

Option A — Using the Web Dashboard (Enterprise Edition)

Go to Ant Media Server Dashboard → Applications → LiveApp → Live Streams
Select your stream
Click on the "Add RTMP Endpoint" option
Add the endpoint and save.

Enter:

RTMP URL (e.g., rtmp://a.rtmp.youtube.com/live2)
Stream Key

Save the target

Ant Media Server will automatically push your stream to each target.

Option B — Using the REST API (Community & Enterprise)

Add a Simulcast endpoint:
POST /v2/broadcasts/{id}/rtmp-endpoint Content-Type: application/json { "rtmpUrl": "rtmp://a.rtmp.youtube.com/live2/your-stream-key" }

Delete a Target:
DELETE /v2/broadcasts/{id}/rtmp-endpoint/{endpointId}

List Targets:
GET /v2/broadcasts/{id}/rtmp-endpoint

▶️** Step 3: Start Simulcasting**
Once the stream is running, Ant Media Server automatically pushes it to all active RTMP endpoints you’ve added.

You can monitor each target’s status from the dashboard or via the API:

BROADCASTING
FAILED
NOT_STARTED

✔️ Tips for a Smooth Simulcast

Use a stable upload bandwidth that can comfortably handle your highest bitrate
Make sure your streaming platform keys are valid
Some platforms (like Facebook) expire keys after one use—generate a fresh one if needed
If pushing to many platforms, consider using transcoding (Enterprise) to send optimized bitrates

🎉 That’s It!

Simulcasting with Ant Media Server is straightforward and powerful. Whether you're building a social streaming platform, hosting events, or expanding your reach, this feature helps you broadcast everywhere at once.

What is Transcoding and How Does It Work? Why is It Important for Streaming?

Amar Thodupunoori — Wed, 05 Nov 2025 10:54:30 +0000

Table of Contents
What Is Encoding?
What Is Transcoding?
What Is Transmuxing?
How Does Transcoding Work?
Why Is Transcoding Important For Streaming?
Transcoding Example
How Can Ant Media Help You With Transcoding?
What Is Encoding?
what is encoding?
Encoding is the process of converting raw audio and video data from your input devices—such as a webcam, microphone, capture card, or streaming software—into a digital format that can be transmitted to streaming platforms. In a typical streaming setup, encoding serves as the bridge between your input sources and the final output, making it a fundamental step in getting your content online.

This process relies on a codec (short for coder-decoder), which is software that determines how your raw data is compressed and formatted. One of the most widely used codecs for live streaming is H.264, known for its ability to efficiently compress video without sacrificing too much quality. It supports a range of resolutions, even up to 8K, making it versatile for various streaming needs.

During encoding, several parameters are set that directly impact the final video’s quality and size. These include compression level, resolution, and bitrate. The encoded data is then packaged into a container (such as MP4 or MKV), which holds both the video and audio streams along with metadata like duration, codec information, and more. The result of this process is a fully-formed digital video file—ready to be streamed, stored, or further processed.

Some of the common video encoding standards

H.264
VP9
AV1
HEVC
VP8
Let’s dive into the question of what is transcoding.

What Is Transcoding?
what is transcoding - Ant Media
Now that we’ve covered encoding, we’re ready to answer the next big question: What is transcoding?

At its core, transcoding is the process of converting an audio or video file from one encoding format to another. This step is crucial for increasing compatibility across a wide range of devices, platforms, and network conditions. By transcoding, you’re ensuring that your content can be played smoothly whether it’s being viewed on a high-end desktop with a fast internet connection or a mobile device on a limited data plan.

The term transcoding actually includes two key sub-processes: transrating and transsizing.

Transrating involves changing the bitrate of a video stream. For example, a high-quality 2K stream at 16 Mbps might be converted into lower bitrate versions—like 720p at 5 Mbps or 480p at 1.5 Mbps. These versions, often referred to as renditions, allow streaming platforms to deliver video suited to the viewer’s available bandwidth and device capability.
Transsizing, on the other hand, refers to adjusting the resolution of the video. A stream originally captured at 2560×1440 (2K UHD) might be resized to more common resolutions like 1920×1080 (1080p), 1280×720 (720p), or even 720×480 (SD). This allows the content to be accessible to a wider audience while reducing the processing and bandwidth load.
Together, these processes make up the foundation of adaptive streaming—a method that dynamically delivers the best possible version of a stream based on the viewer’s real-time conditions.

What Is Transmuxing?
An important point! The what is transmuxing question is as critical as the question of what is transcoding. Transcoding is not the same as transmuxing. Transmuxing is also called recoding, repacking, packaging, or repacking. Transmuxing is a process that you take compressed audio and video and pack/repack it in different formats with keeping audio or video content original.

For example, you might have H.264 / AAC content and change the container in which it is packaged so you can use it HTTP Live Streaming (HLS), Smooth Streaming, HTTP Dynamic Streaming (HDS), or Dynamic Adaptive Streaming over HTTP (DASH). The computational overhead for transmuxing is much smaller than transcoding.

How Does Transcoding Work?
When a video is recorded—whether live or on-demand—it’s typically saved in a format specific to the capturing device or software. These raw formats are often not optimized for playback across different devices or platforms.

That’s where transcoding comes in.

Transcoding is a two-step process:

Decoding – The original compressed video is first decoded into an uncompressed format.
Re-encoding – That uncompressed video is then encoded into a new format, resolution, or bitrate that’s compatible with the viewer’s device or network conditions.
This process ensures smooth playback and broad compatibility, especially important for adaptive streaming and delivering content across varying devices and bandwidths.

Why Is Transcoding Important For Streaming?
why is transcoding important for video streaming

The most important benefit of Video Transcoding is that it enables live streams to be watched by a much wider audience regardless of connection or device.

For example, you want to live stream using a camera and encoder. Suppose you’re compressing your content with an RTMP encoder and selecting the H.264 video codec at 1080p.

You prepared the live broadcast content, you worked hard. Of course, you don’t want this effort to be wasted. But if you try to stream your perfect live stream directly, you’ll likely run into a few issues. Quick information: The world’s average fixed broadband download speed has increased by 38% in just two years and is currently around 64Mbsp. However, actual speeds vary greatly between different countries, even between different regions in the same country and different types of connections.

So First issue, viewers who do not have enough bandwidth cannot watch the stream. It will buffer the players continuously while waiting for the packets of the 1080p video to arrive. For example, It will not be possible for you to make an audience living in America and a viewer living in Nigeria happy with the same bitrate.

Second, the RTMP protocol has now lost Adobe’s support. Therefore, it will not be possible to reach large audiences with RTMP playback. Apple’s HLS is much more widely used. You exclude almost anyone with slow data speed, tablets, mobile phones, and smart TV devices without transcoding and transmuxing video.

With video transcoding software, you can stream video files that have different bitrates and frame sizes, while converting the codecs and protocols to reach a wider audience. These devices and status compatible streams can be packaged into several streaming formats (such as HLS, WebRTC, or CMAF). Transcoding allows us to play videos on almost any screen.

multi device streaming with ant media
Another important area of use is IP camera streaming such as surveillance for traffic cameras. These streams often need to be delivered to many viewers simultaneously, sometimes over limited bandwidth connections. Adaptive bitrate streaming enabled by transcoding, ensures that these broadcasts can be watched smoothly by a wide audience without buffering or interruptions.

We gave the answer to the question of what is transcoding and we touched on the details. So, how does YouTube which we all visit a lot during the day use transcoding?

Transcoding Example
YouTube, one of the world’s largest video platforms, receives hundreds of hours of uploads every minute. To ensure smooth playback across devices and connection speeds, YouTube transcodes each video into multiple versions—often dozens—at different resolutions and formats.

This process starts immediately after upload, which is why new videos usually appear first in low resolution. Higher-quality versions, including 4K and beyond, become available after more intensive transcoding is complete.

YouTube also uses advanced codecs like VP9 and AV1 to optimize quality and bandwidth, delivering the best viewing experience for every user.

How Can Ant Media Help You With Transcoding?
If you’re live streaming to a small, consistent audience, maintaining a single video quality might be enough. But if you want to reach a broader audience and deliver a truly successful broadcast, you essentially have two choices.

You could either settle for low video quality to accommodate everyone—or choose a smarter approach. With Ant Media, you can deliver the highest quality stream to each viewer, no matter their connection speed, location, or device. Ant Media offers scalable, ultra-low latency, and adaptive WebRTC streaming, enabling live broadcasts that are not only smooth and reliable but also interactive and engaging. Simply put, Ant Media helps you create live streams your audience will love.

Streaming Made Simple: RTMP with OBS in 5 Minutes

Amar Thodupunoori — Wed, 03 Sep 2025 08:53:58 +0000

If you’ve ever wanted to broadcast your video feed to a streaming server or platform, chances are you’ll come across RTMP (Real-Time Messaging Protocol). It’s one of the most common ways to send live video from an encoder like OBS Studio to a media server.

Here’s a quick guide to setting it up.

Step 1: Install OBS Studio

Download and install OBS Studio — it’s free, open-source, and works on Windows, macOS, and Linux.

Step 2: Grab Your RTMP Server URL & Stream Key

You’ll need an RTMP endpoint to stream to. This could be:

A live streaming platform (YouTube, Twitch, Facebook, etc.)
Your own media server (e.g., Ant Media Server, Wowza, Nginx-RTMP).

Usually, you’ll get something like:
rtmp://your-server-ip:1935/live

and a Stream Key, which uniquely identifies your broadcast.

Step 3: Configure OBS for RTMP

In OBS:

Go to Settings → Stream
Set the Service to Custom
Enter your RTMP URL in Server
Paste your Stream Key

Step 4: Start Streaming

Click Start Streaming in OBS. Your feed will now be sent over RTMP to your target server/platform.

💡 Pro tip: If you’re streaming to your own RTMP server, make sure port 1935 is open in your firewall/security group.

That’s it — you’ve just set up RTMP streaming with OBS!