Bandwidth Optimization Guide: Running 4K Video Meetings Without Slowing Down Your Network

There’s a particular kind of dread that sets in about ninety seconds into an important call. The client’s face freezes mid-sentence. Their voices keep going, detached from their mouths, like a badly dubbed film. Someone says “you’re frozen” into the void, and for the next four minutes the meeting limps along on audio only while everyone pretends this is fine. Nobody in that meeting was short on internet speed. Most of them were paying for connections that would have seemed like science fiction a decade ago. The problem was never about how much bandwidth was available. It was about how that bandwidth was being used, shared, prioritized, and adapted to in real time. And nowhere does that distinction matter more than with 4K video, which asks more of a network than almost any other everyday application, streaming included. This guide exists because “just get faster internet” is the advice equivalent of telling someone with a leaky roof to buy a bigger bucket. It might help. It won’t fix the actual problem. What follows is a practical breakdown of what 4K video meetings cost in bandwidth, why that cost is more complicated than a single number, and the specific things you can change, at the network level, the platform level, and the architecture level, to make 4K meetings something your infrastructure handles rather than survives.

Why 4K video meetings are a different animal from 4K streaming

It’s worth pausing on why this is even hard, because on paper, 4K video conferencing should be easier than 4K streaming. A single video call is one stream, not the millions Netflix pushes out simultaneously. Compare the two properly, though, and video conferencing is the harder engineering problem, for three reasons that have nothing to do with resolution. It must happen in both directions, in real time. Streaming a 4K film is a one-way problem: encode once, cache aggressively, deliver whenever the viewer’s buffer needs more. A video call is two ways, live, and has essentially no buffer to hide behind. Netflix can spend three seconds building a buffer before you notice anything. A video call has a budget measured in milliseconds before delay becomes obviously awkward. That means every device in a 4K call is simultaneously encoding its own camera feed and decoding everyone else’s feed, with no do-overs. The network conditions are unknown and constantly changing. A streaming service can pre-encode a video into a dozen different bitrate versions and let your player pick the best one for your current connection. A live call can’t pre-encode anything. It must guess, continuously, what the network will tolerate right now, for every single participant, most of whom are on wildly different networks with wildly different problems. Everyone is also a sender. In a 4K meeting with eight participants, you’re not receiving one 4K stream, you’re potentially receiving several, while also uploading your own. Upload bandwidth, historically the neglected stepchild of home and office internet plans, suddenly matters just as much as download. This is the detail most bandwidth guides skip, and it’s usually the actual bottleneck. Put those three things together and you get why “4K video call” is such a demanding workload: it’s live, bidirectional, unpredictable, and multiplies with every participant. Streaming solved its problem with buffering and pre-computation. Video conferencing must solve the same class of problems in real time, which is exactly why the technologies discussed later in this guide, simulcast, SVC, adaptive bitrate, exist at all. They’re the real time equivalent of Netflix’s pre-encoded bitrate ladder, built to run live instead of ahead of time.

What does 4K video cost in bandwidth

Ask five different video conferencing vendors how much bandwidth 4K requires and you’ll get five different answers, all technically correct and all somewhat useless in isolation, because bandwidth requirements for video aren’t a fixed number. They’re a function of resolution, frame rate, codec efficiency, and how much motion is in the picture. Here’s the honest range. A single 4K (2160p) video stream at 30 frames per second typically needs somewhere between 8 and 20 Mbps using a modern, efficient codec, and can climb past 25 Mbps with an older, less efficient one or with a lot of motion in frame, think screen sharing a video, or someone gesturing energetically in a well-lit room with a busy background. A talking head against a plain wall compresses beautifully because most of the frame doesn’t change from one moment to the next. The same resolution pointed at a whiteboard covered in scribbles, or panned across a factory floor, generates far more data because the codec must describe far more change. Compare that to 1080p, which typically runs 3 to 6 Mbps, and 720p, which comfortably sits under 2.5 Mbps for most content. The jump from 1080p to 4K isn’t linear, it roughly triples to quadruples the pixel count, and bandwidth requirements scale with it, though good codecs soften that curve somewhat. Now multiply that by direction and participant count, because this is where most bandwidth planning quietly falls apart. If you’re hosting a 4K call and want to see three other participants at high quality, you’re not budgeting for one 4K stream, you’re budgeting for one 4K upload (your own camera) plus, depending on the platform’s architecture, either one combined downstream or several individual incoming streams. On platforms using a selective forwarding unit (more on this shortly), your download requirement can be the sum of every other high resolution participant’s stream, which is how a “4K call” between six people quietly becomes a 60 to 80 Mbps combined bandwidth event on a single connection, even though no individual video ever exceeds 4K. This is the number that catches offices off guard. A conference room with a good 4K camera, connected to an internet plan that comfortably handles video streaming for the whole building, can still choke during a large 4K meeting, not because the plan is too slow in general, but because that specific moment asks for more simultaneous upload and download than almost anything else the office does. Streaming a movie pulls maybe 15 to 25 Mbps download and essentially nothing upload. A six-person 4K meeting can ask for that much in both directions at once, sustained, for the length of the call. The practical takeaway: don’t budget for “a 4K meeting.” Budget for “the number of simultaneous 4K equivalent streams your busiest realistic meeting will generate, in both directions, at the same time as everything else on that network.” That number is almost always higher than people expect, and it’s the actual figure worth testing against, not the marketing page bandwidth requirement that assumes one perfect stream in a lab with no other traffic.

The real enemy: jitter and packet loss, not raw speed

Here’s the part that surprises people who’ve spent money solving the wrong problem. A connection with 500 Mbps of raw throughput can produce a worse video call than one with 15 Mbps, if the faster connection is inconsistent. Video, unlike a file download, doesn’t care primarily about total capacity. It cares about consistency. Every video codec used in real time communication is built around the assumption that packets will arrive close enough together, in close enough order, that the decoder can reconstruct a smooth picture. Break that assumption and things degrade fast, regardless of how much theoretical bandwidth is sitting unused. Jitter is the variation in how long it takes packets to arrive. Average latency of 40 milliseconds sounds fine. Average latency of 40 milliseconds that swings wildly between 10 and 150 milliseconds, packet to packet, is not fine at all, even though the average looks identical on paper. The decoder must buffer against the worst-case jitter it’s seeing, which either adds delays to smooth things out, or, if it doesn’t buffer enough, produces the stuttering, freezing effect everyone recognizes instantly. Jitter tends to spike when a network is shared aggressively, a video call competing with a large file upload, a backup job, or another 4K call in the next room over the same router. Packet loss is exactly what it sounds like: some of the data simply doesn’t arrive. Video codecs are somewhat forgiving of small amounts of loss, they can conceal a missing frame or two, extrapolate briefly, recover on the next keyframe. But losses above roughly 1 to 2 percent start to visibly degrade picture quality, and losses above 5 percent make a call difficult to sit through, full of blocky artifacts and audio dropouts, even if the connection’s advertised speed is enormous. Wi-Fi is particularly prone to this in busy environments, since it’s a shared, contested medium where interference from other networks, microwaves, and simple distance from the access point all quietly eat into reliability without ever showing up as a headline “speed” number. The reason this matters practically: if your team’s 4K calls are struggling, the first diagnostic question shouldn’t be “how fast is our internet.” It should be “how consistent is our internet, right now, during the call.” Most operating systems and routers can show packet loss and jitter figures, and most video platforms, Ollasync included, surface a basic connection quality indicator during a call for exactly this reason. If that indicator is flashing amber while your speed test shows 300 Mbps, the fix isn’t a faster plan. It’s addressing whatever is making delivery inconsistent, which is almost always a local network issue, not an internet service provider issue.

Codecs: the quiet workhorse doing most of the heavy lifting

If bandwidth is the budget, the codec is the thing deciding how far that budget stretches. This is the single most underappreciated lever in the entire bandwidth conversation, because it happens invisibly, in software, and most people never think about which codec their video platform is using.

H.264 (AVC) has been the workhorse of video conferencing for over a decade. It’s universally supported, hardware accelerated on essentially every device made in the last ten years, and reasonably efficient. It’s also, by modern standards, no longer the most efficient option available, which matters a great deal at 4K, where every percentage point of compression efficiency translates directly into real megabits saved.
VP9, developed by Google, typically achieves 25 to 50 percent better compression than H.264 at equivalent visual quality, meaning a 4K stream that needed 16 Mbps under H.264 might need closer to 10 to 12 Mbps under VP9 for a comparable result. The catch is hardware support, older devices sometimes fall back to software decoding, which costs battery life and CPU rather than bandwidth, a fair trade in most conferencing scenarios.
AV1 is the newer entrant, and the one worth watching closely. It offers meaningfully better compression than VP9 again, sometimes another 20 to 30 percent, which at 4K resolutions is the difference between a stream comfortably fitting on a congested Wi-Fi connection and one that doesn’t. AV1’s tradeoff has historically been encoding complexity, it’s computationally expensive to encode in real time, though hardware encoder support has expanded rapidly and is now common in recent laptops, phones, and dedicated conferencing hardware.

The practical implication is straightforward: two platforms delivering visually identical 4K video can have meaningfully different bandwidth footprints purely based on codec choice, before any other optimization is applied. When evaluating or configuring a video meetings platform for bandwidth constrained environments, it’s worth asking directly which codec it supports and whether it negotiates the most efficient one available for each participant’s hardware, rather than defaulting to the lowest common denominator for universal compatibility. The best systems do both: negotiate up to a modern, efficient codec when both ends support it, and fall back gracefully to something universally compatible when they don’t. Simulcast and SVC: the technologies that save a meeting this is the part of the bandwidth story that does more practical work than almost anything else discussed here, and it’s rarely explained clearly, so it’s worth taking the time.

Imagine a video call with six participants, and one of them, on a congested hotel Wi-Fi network, can reliably handle only about 2 Mbps of incoming video. In a naive system, that one person’s poor connection would force everyone else to either drop to a lower shared resolution, or that participant would simply have a broken experience while everyone else continues at full quality, unaware anything is wrong on their end. Simulcast solves this by having each sender encoding and sending multiple versions of their own video simultaneously, typically a high-resolution version (say, 4K or 1080p), a medium version (perhaps 540p), and a low version (perhaps 180p), all at once, from the same camera feed. The receiving infrastructure, specifically a selective forwarding unit, then decides which version to forward to each individual participant based on that participant’s actual measured network conditions. The hotel Wi-Fi participant gets the low-resolution version of everyone else’s video, seamlessly, without anyone else’s stream being affected, because the sender was already producing that lower resolution version regardless. The cost of simulcast is that senders spend more of their own upload bandwidth and CPU producing multiple versions of their stream, generally 15 to 40 percent more upload than sending a single stream would require. That’s a worthwhile trade in almost every real meeting scenario, because it means one struggling participant no longer degrades the experience for everyone.

Scalable Video Coding (SVC) achieves something similar through a different, more elegant mechanism. Rather than encoding entirely separate streams, SVC encodes a single stream in layers, a base layer that alone produces a lower quality picture, plus enhancement layers that, when added to the base, progressively improve resolution and quality. The forwarding infrastructure can then simply drop enhancement layers for participants who can’t handle them, rather than switching between entirely separate encoded streams. This is generally more bandwidth efficient than full simulcast on the sending side, though it demands more from decoders and hasn’t achieved quite the universal hardware support that simpler simulcast approaches enjoy. Why does any of this matter for your 4K meetings specifically? Because without simulcast or SVC, a 4K call is only as strong as its weakest connection, full stop. Everyone gets dragged down to whatever the worst participant’s network can handle, which defeats the entire point of running the meeting at 4K in the first place. With simulcast or SVC properly implemented, the people with strong connections get 4K, the people on weaker connections get a version that works for their situation, and nobody’s experience is held hostage by someone else’s hotel Wi-Fi. This is, quietly, one of the most important questions to ask any platform you’re evaluating for high resolution meetings: does it support simulcast or SVC or does everyone in the call get capped to the network conditions of the least capable participant.

Adaptive bitrate and congestion control: the system’s real time judgment calls

Underneath simulcast and codec choice sits a layer that’s arguably more important than either, because it’s the part deciding, moment to moment, how much data to send. This is congestion control, and in the WebRTC world, the dominant approach is something called Google Congestion Control (GCC), alongside newer approaches building on similar principles. Here’s roughly how it works, without drowning in protocol detail. The sending device continuously monitors feedback from the receiver about how packets arrive, are they delayed, are they arriving out of order, is anything being lost. Based on that feedback, arriving many times per second, the sender adjusts its outgoing bitrate up or down, trying to find the maximum quality the current network path can sustain without overwhelming it. Push too hard and the network queue backs up, causing exactly the jitter and loss discussed earlier. Being too conservative and quality suffers unnecessarily on a connection that could have been handled more. This is difficult to get right, and it’s a major differentiator between video platforms that people describe as “solid, even on bad Wi-Fi” and platforms that people describe as “constantly freezing.” A well-tuned congestion control system reacts to a sudden network hiccup, someone else in the house starting a large download, a Wi-Fi interference burst, within a second or two, smoothly stepping down resolution or frame rate temporarily, then stepping back up once conditions improve. A poorly tuned one either reacts too slowly, letting quality degrade badly before adjusting, or overreacts, oscillating between quality levels in a way that’s more distracting than a steady, slightly lower quality stream would have been. For anyone trying to optimize their own 4K meeting experience, the practical implication is this: you generally cannot out-configure bad congestion control from the network side. If a platform’s adaptive bitrate logic is poorly implemented, no amount of router tuning on your end will fully compensate, because the sender simply isn’t responding well to the conditions your network is reporting. This is one of the genuine reasons platform choice matters for bandwidth sensitive use cases, not as marketing, but as an actual technical variable worth testing directly, ideally by running the same meeting, at the same resolution, over the same network, across a couple of different platforms, and watching how each one handles an intentional bandwidth squeeze, like starting a large download mid call.

Fixing it at the network level: what moves the needle

With the platform side covered, here’s where most of the actionable, immediate fixes live, because these are things you control directly, today, without waiting on any vendor.

Get off Wi-Fi when it matters. This is the single highest impact change available to almost anyone, and it’s consistently underused because wired connections feel like a step backward in a wireless world. Wi-Fi is a shared, contested medium, every other device on the network, every neighboring network on the same channel, every microwave and Bluetooth device in range, is competing for the same airwaves. A wired ethernet connection is essential for all that contention. For any 4K meeting that matters, plugging in, even just for that call, is worth more than almost any other single change on this list.
If Wi-Fi is unavoidable, use 5GHz or 6GHz, not 2.4GHz. The 2.4GHz band is more crowded, has fewer non overlapping channels, and is used by far more household devices, cordless phones, baby monitors, older smart home gear. The 5GHz and newer 6GHz bands (available on Wi-Fi 6E and Wi-Fi 7 hardware) offer more channels, less interference, and meaningfully better throughput consistency for exactly the kind of sustained, latency sensitive traffic that video calls generate.
Prioritize meeting traffic with QoS. Most modern routers, and certainly most business-grade ones, support Quality of Service configuration, which lets you tell the router to prioritize certain kinds of traffic over others. Video conferencing traffic is a textbook use case for this, because it’s latency sensitive in a way that, say, a background software update simply isn’t. Properly configured QoS, often using DSCP tagging that many conferencing platforms already apply to their outgoing packets, ensures that when the network is under pressure, video call packets get sent first, ahead of less time sensitive traffic. This is a one-time router configuration change that pays off every single call afterward.
Segment meeting traffic onto its own VLAN in office environments. For businesses running multiple simultaneous meetings, particularly hybrid offices with several conference rooms active at once, putting video conferencing traffic on a dedicated VLAN, separate from general office traffic, printers, guest Wi-Fi, and background business applications, gives IT teams a clean way to apply prioritization and monitor usage specifically for meeting traffic, without it being tangled up with everything else happening on the network.
Watch out for VPN backhaul. This one catches a surprising number of organizations. If your corporate VPN routes all traffic, including video call traffic, through a central office before it reaches the internet, you’ve just added a detour, and every hop adds latency and, more importantly, a new point where jitter and loss can be introduced, especially if that central location is geographically distant from wherever the meeting participants are. Split tunnel VPN configurations, which route only traffic that needs the VPN through it while letting real time media traffic go directly to its destination, solve this cleanly and are worth specifically requesting from IT if your organization hasn’t already implemented one.
Check for buffer bloat. This is a less commonly known issue, but a real one: many consumer and even some business routers will queue up outgoing traffic aggressively during periods of heavy use, trying to avoid dropping packets, which sounds good in theory but introduces significant delay, the packets aren’t lost, they’re just late, arriving in a burst rather than steadily. This shows up as your internet speed test looks fine while video calls still stutter under load. Modern router firmware with active queue management, sometimes labeled “Smart Queue Management” or similar, addresses this directly and is worth checking for specifically if QoS alone doesn’t resolve stuttering under load.

Fixing it at the organizational level: the decisions that scale

Individual network tweaks help individual calls. For organizations running many 4K meetings simultaneously, tutoring companies, training academies, distributed teams doing daily standups on camera, the bigger wins come from architectural and policy decisions. Understand the difference between mesh, MCU, and SFU architectures, because this single choice determines how bandwidth scales with participant count. A peer-to-peer mesh, where every participant sends video directly to every other participant, scales terribly, each additional person multiplies the upload burden on everyone already in the call, which is why mesh architectures are essentially unusable past three or four participants at any meaningful resolution. A Multipoint Control Unit (MCU) solves this by having a central server decode everyone’s video, composite it into a single combined stream, and send that one stream back out, which is bandwidth efficient for participants but computationally expensive for the server and introduces additional latency from all that decoding and re-encoding. A Selective Forwarding Unit (SFU), the architecture most modern platforms including Ollasync are built around, sits in between: it receives each participant’s stream once and forwards copies to whoever needs them, without decoding and re-encoding the actual video content, which keeps latency low while still solving the mesh scaling problem. Understanding which architecture your platform uses explains a lot about why bandwidth behaves the way it does as your meetings get larger. Consider where that server infrastructure physically lives. This matters more than most bandwidth guides acknowledge. A media server sitting in a data center on a different continent from most of your participants adds latency and, often, additional network hops where jitter can creep in, purely from the physical distance the data must travel. This is part of why architecture questions and bandwidth questions overlap so heavily with the kind of infrastructure decisions covered in our piece on how WebRTC powers low latency video conferencing in private cloud architectures, and why organizations running frequent high resolution meetings across a distributed team increasingly look at self-hosted deployments specifically so media routing can live closer to where their people are, rather than being routed through a single distant region by default. Stagger meetings deliberately. This sounds almost too simple to mention, but it’s one of the highest leverage fixes available to any organization, and it costs nothing. If your office has four conference rooms and all of them book 4K calls at the top of every hour, you’ve created a predictable bandwidth spike that a network sized for average usage will struggle with, even if it comfortably handles the same total load spread across the hour. Staggering start times by even ten- or fifteen-minutes smooth that demand curve meaningfully. Recognize that not every meeting needs to be 4K. This one is more about policy than technology, but it’s worth saying plainly: a two person one on one check-in benefits far less from 4K than a training session where a trainer is demonstrating fine detail work, or a legal deposition where document clarity matters. Reserving 4K for the meetings where resolution adds value, and defaulting to 1080p or even 720p elsewhere, is a legitimate bandwidth strategy, not a compromise. It’s the same logic behind why local recording, keeping the recorded file on device rather than continuously uploading it to a cloud service throughout the call, exists as a deliberate bandwidth and privacy choice rather than an afterthought: not every piece of a meeting’s data needs to travel over the network in real time just because it technically could.

A practical checklist before your next important 4K meeting

For anyone who wants a shorter, more actionable version of everything above, here’s what matters, roughly in order of impact.

Plug in via ethernet if the meeting matters and a wired connection is available at all. This single change resolves more real world 4K call problems than any other item on this list.
Close or pause any other bandwidth heavy activity running on the same network during the call, large downloads, cloud backups, other simultaneous video calls in the building.
Confirm your platform supports simulcast or SVC, so a struggling participant on the call doesn’t drag everyone else’s experience down with them.
Check whether your router has QoS enabled and configured to prioritize video conferencing traffic specifically, not just generic “gaming” or “streaming” presets that may not recognize meeting traffic correctly.
If you’re on a corporate network, confirm whether VPN backhaul is routing meeting traffic through an unnecessary detour, and ask IT about split tunneling if so.
Test actual jitter and packet loss, not just download speed, using your router’s diagnostics or a dedicated network testing tool, particularly if calls are degrading despite a fast advertised internet plan.
For recurring high-stakes 4K meetings, consider whether meeting infrastructure located closer to your participants, rather than a distant default region, would meaningfully reduce latency and jitter.
Reserve genuine 4K for meetings where resolution adds real value, and don’t hesitate to default to 1080p for routine calls, freeing up bandwidth headroom for when it matters.

Common myths worth retiring

A few persistent misunderstandings are worth addressing directly, because they lead people to spend money and effort in the wrong places.

We just need a faster internet plan. Sometimes true, often not. If your existing plan comfortably handles the raw throughput math discussed earlier, and your calls are still struggling, the problem is very likely jitter, packet loss, or contention on your local network, none of which a faster plan from your internet provider fixes. Diagnose before upgrading.
More bandwidth always means better video quality. Only up to a point, and that point arrives sooner than people expect. Beyond what the sender’s camera, codec, and the receiver’s display can make use of additional bandwidth does nothing for visible quality. A 4K stream doesn’t improve by throwing 100 Mbps at it instead of 20 Mbps, the codec and resolution have already defined the ceiling.
Everyone on the call needs the same bandwidth. This misunderstanding is exactly what simulcasts and SVC exist to correct. In a properly built system, participants with strong connections and weak ones can coexist in the same 4K meeting, each receiving a quality level appropriate to their own network, without either group’s experience depending on the other.
Wi-Fi 6 alone solves bandwidth problems. Newer Wi-Fi standards help, more channels, better handling of multiple simultaneous devices, improved efficiency, but they don’t eliminate the fundamental physics of a shared, contested wireless medium. A wired connection still outperforms even the best Wi-Fi 6E setup for latency sensitive, sustained traffic like a 4K video call, particularly in environments with a lot of other wireless devices nearby.

How Ollasync approaches bandwidth for high resolution meetings

Everything discussed in this guide, codec efficiency, simulcast, congestion control, network architecture, is why bandwidth optimization isn’t really a single feature you turn on. It’s a set of decisions baked into how a video platform is built, from the codec negotiation happening in the first few hundred milliseconds of joining a call, to where the underlying media infrastructure physically lives.

Ollasync’s live classes and video meetings are built on modern WebRTC foundations with adaptive bitrate and simulcast support specifically so a session with one participant on hotel Wi-Fi and another on gigabit fiber doesn’t force either of them into the other’s constraints. For organizations that need media infrastructure closer to their actual users, particularly relevant for the latency and jitter considerations covered above, self-hosting puts that routing decision directly in your hands rather than defaulting to whatever region a vendor happened to pick.
Bandwidth and security decisions overlap more than people expect, too. Sessions running end-to-end encryption trade certain server-side optimizations, like the kind of processing that requires a server to read the media, for a guarantee that nothing in the middle can see the call content at all. That’s a deliberate tradeoff, not a limitation nobody thought about, and it’s worth understanding as part of your own architecture decisions if compliance or confidentiality requirements sit alongside your bandwidth ones.

Testing your own network before it matters

The single most useful thing anyone can do after reading this guide isn’t buy anything, it’s testing. Before your next important 4K meeting, run an actual test call, at the resolution you plan to use, on the exact network and exact device you’ll be using, ideally with the same number of participants you expect. Watch the connection quality indicator most platforms display during a call. Try deliberately stressing the network mid call, start a large download, have someone else on the network begin a video call of their own, and watch how gracefully or badly things degrade. That single test tells you more than any bandwidth calculator or vendor specification sheet, because it shows you your actual conditions, not a lab’s best-case scenario. If the call holds up cleanly under that kind of pressure, you’re in good shape. If it doesn’t, you now know exactly where to look first, wired versus wireless, QoS configuration, VPN routing, or simply too much else happening on the same connection at the same time, rather than guessing. 4K video meetings are entirely achievable on ordinary office and home networks. They just require understanding that the fight isn’t against a single bandwidth number, it’s against inconsistency, contention, and architecture choices that either adapt gracefully to real world conditions or don’t. Get those right, and 4K stops being something your network barely survives, and starts being something it simply handles.

Wrapping Up

4K video meetings aren’t hard because of bandwidth scarcity, they’re hard because most networks and platforms weren’t built to handle live, bidirectional, high-resolution traffic gracefully under real world conditions. The fixes that work aren’t exotic: use a codec that compresses efficiently, choose a platform that supports simulcast or SVC so one weak connection doesn’t sink the whole call, prioritize meeting traffic at the router level, and get wired whenever a meeting matters. Bandwidth is the budget. Consistency is what spends it well. Test your setup before it counts, fix what’s inconsistent rather than what’s merely slow, and 4K meetings go from something your network tolerates to something it simply runs. If you’re evaluating infrastructure for regular high-resolution meetings, that’s exactly what Ollasync’s video meetings are built around, and it’s worth pairing with a look at self-hosted deployment if latency and data location matter as much as raw quality for your team.

Global Infrastructure — What Is Multi-Region Video Routing and How Does It Prevent Lag in Global Meetings?: How distributed regional relays eliminate latency across continents.
On-Premises Architecture — How to Deploy an On-Premises Video Conferencing Server in Under 30 Minutes: Step-by-step architecture for running private media infrastructure.
Self-Hosting Options — Top 7 Open Source & Self Hosted Meeting Software Solutions for 2026: Evaluating Jitsi, BigBlueButton, Nextcloud Talk, Matrix, LiveKit, OpenMeetings, and Jami.