Broadcasting, Live Streaming Hardware, Compare BoxCast, Live Streaming Software
Not all live video streaming protocols are built the same. This blog breaks down how three of the most popular and widely adopted protocols handle real-world network challenges when they occur — like network fluctuations, internet disconnection, packet loss, and more. Which one of these protocols will be able to deliver a crystal clear stream, even when the pressure is on?
| RTMP | SRT | BoxCast Flow | |
|
Healthy Network |
✅ | ✅ | ✅ |
| Network Drop Will this protocol perform with a network that experiences unexpected drops in upload speed? |
❌ | ✅ | ✅ |
| Auto Adjustment to Network Can this protocol dynamically and smoothly adjust its bitrate when network speed fluctuates up and down? |
❌ | ❌ | ✅ |
| Disconnected Network Does this protocol weather a total loss of network for a brief period of time? |
❌ | ❌ | ✅ |
| Missing Data Retransmission Will this protocol retransmit missing video and audio data if it doesn’t arrive as expected? |
✅ | ✅ | ✅ |
| IP Protocol Diversity What transport layer protocol(s) does this protocol use when data doesn’t arrive as expected? |
❌ TCP Only |
❌ UDP Only |
✅ TCP + UDP |
| Packet Loss Can this protocol perform well when its network experiences a prolonged period of high packet loss? |
❌ | ✅ | ✅ |
Live Streaming in Ideal Conditions
Network Speed Suddenly Drops
Internet Cuts Out Completely
Missing Data Mid-Stream
Packet Loss Lasts For Minutes (or the Entire Stream)
Frequently Asked Questions
Final Thoughts + Further Reading
Everything is perfect. Your upload speed is high and consistently more than twice the bitrate of your live stream. Packets are reaching your servers smoothly with no loss, and you’ve fine-tuned your latency to just a few seconds — giving your live event a truly real-time feel.
When your network is running flawlessly, you can focus on delivering a top-tier production and providing a seamless viewing experience for your audience, no matter where they’re watching from.
But if you’ve been live streaming for a while, you know that perfect conditions don’t always last. Even with carefully optimized networks — trouble shows up exactly when you don’t have time for it. BoxCast broadcasters find that over 10,000 of their weekly streams encounter network-related problems.
So, to ensure your stream stays smooth and professional, even when your network isn’t at its best, you’re going to need a streaming protocol built for real-world resilience. With this in mind, let’s roll up our sleeves and dive into the four most common network challenges live streamers face—and find out which protocol handles them best.
Imagine a broadcaster has 8 Mbps of upload speed available and has conservatively set their video bitrate to 4 Mbps. What happens to their stream if their available network speed suddenly drops to 2 Mbps?
This could happen for a variety of reasons—maybe their ISP throttles their speed during peak usage hours, their venue's shared network fluctuates, or another device on their private network starts using high upload bandwidth unexpectedly. Sudden drops in network speed are common and often unavoidable.
Whatever the cause, your streaming protocol needs to be able to gracefully handle significant network drops in order to keep your broadcast running smoothly. Here’s how different protocols respond to a sudden reduction in network speed:
RTMP relies on a constant bitrate and doesn’t handle fluctuations well. When bandwidth drops, it buffers heavily, leading to the dreaded spinning wheel as the stream tries to catch up. If the drop is too severe or lasts too long, the stream can disconnect entirely, forcing the broadcaster to restart.
As a more modern protocol, SRT improves on RTP and RTMP by using selective retransmission and error correction. When network speed drops, SRT increases latency by a few seconds to allow time for lost packets to be recovered. However, if bandwidth is severely limited, buffering will still occur as the protocol prioritizes reliability over real-time playback.
BoxCast Flow was designed to solve many networking issues including adjusting to less bandwidth through a Link Quality Adjustment, by dynamically adapting to both improvements and drops in network speed. When upload speed dips significantly, it automatically lowers the video bitrate, optimizing quality in real time to match available bandwidth—much like a surfer adjusting to the shape of a wave. For viewers, this means the stream remains stable with minimal buffering, significantly reducing interruptions and maintaining a smooth experience.
You need the internet to stream, right? Well… kind of. Power failures, unplugged cables, and unexpected ISP maintenance can cause temporary—or even prolonged—internet outages. Because these disruptions are a reality in live streaming, some protocols are designed to withstand brief internet loss and keep broadcasts running smoothly. But which ones actually deliver?
RTMP will attempt to buffer, but when the network disconnects, the stream completely shuts down. Viewers are left staring at a black screen until three things happen in perfect sequence:
With so many manual steps, the odds of recovering both the stream and the audience are slim.
SRT queues lost data for selective retransmission, but an outage longer than 8 seconds is too much for a smooth recovery. The stream will first buffer heavily, then eventually stop playback until the connection returns. Some video players may even require a manual refresh before playback can resume.
"Help me, BoxCast Flow. You’re my only hope."
BoxCast Flow includes a feature called Flow Control, which allows latency to be adjusted from 2 to 90 seconds. This gives poor or unreliable networks extra time to recover, ensuring that all streaming data is delivered on time, even if a temporary outage occurs. If the disruption lasts around 40 seconds or less, viewers won’t even notice a thing—the stream continues seamlessly, without requiring the internet for a portion of the broadcast.
One of the most complex yet essential aspects of a live video streaming protocol is how it handles lost data at the transport layer. Streaming smooth, high-resolution video requires massive amounts of data to be uploaded to a server, transcoded, distributed, and played back by viewers. If network disruptions occur, stream quality will suffer—unless the streaming protocol efficiently retransmits lost data and ensures it arrives in the correct order.
SRT uses User Datagram Protocol (UDP), which prioritizes keeping the stream going, even if some data is lost along the way. Because UDP does not request retransmissions, this protocol can operate with lower latency, making them ideal for real-time applications.
However, the downside is that lost packets stay lost—which can cause buffering, visual artifacts, or even stream disconnection if network conditions deteriorate.
P.S. As mentioned previously, though, SRT does use selective retransmission, since UDP is not ideal for retransmission.
RTMP, on the other hand, uses Transmission Control Protocol (TCP), which ensures that every packet arrives correctly and in order. TCP prioritizes accuracy over speed, meaning that if packets are lost, they are requested again and retransmitted.
This improves stream integrity but introduces higher latency and can cause buffering or freezing if the network struggles to keep up. In RTMP streams, even minor network disruptions can result in delays or interruptions, frustrating viewers.
BoxCast Flow combines the best of both UDP and TCP, then takes it a step further:
This hybrid approach results in a seamless, visually stunning stream that stays resilient even in challenging network conditions—without the interruptions, buffering, or delays of other protocols.
Packet loss can happen in many ways, and we've already covered different types of data loss in previous sections. But there's one more scenario worth diving into—prolonged packet loss that persists for minutes or even for an entire broadcast.
ISPs can experience last-mile delivery issues, where something goes wrong between their infrastructure and your building. Bad cables, faulty network equipment, fluctuating Wi-Fi, or network congestion from a sudden influx of devices can all create what Apple’s Network Link Conditioner refers to as a "Very Bad Network", where 10% of data is lost continually throughout a broadcast.
RTMP’s reliance on TCP means every lost packet is retransmitted, leading to constant buffering and interruptions while the stream tries to catch up during prolonged packet loss. Furthermore, RTMP puts limits on how many packets can be queued up between original transmission and retransmission before eventually giving up on both types of packets. So, even if the stream may technically stay online, viewers will struggle through repeated buffering cycles at best and buffering alongside heavy artifacting at worst, making the experience frustrating either way.
SRT's latency adapts in the face of ongoing packet loss to allow time for some selective retransmission, which can temporarily pause a stream. As 10% loss persists, latency will eventually max out (typically 8 seconds) and begin buffering or degrade stream quality. Though the stream will likely remain online, playback stability declines. In our testing, we even found that removing packet loss results in significant buffering or artifacts for a number of seconds.
BoxCast Flow combines Link Quality Adjustment, Flow Control, Forward Error Correction, and Smart Retransmission to handle sustained packet loss better than any other protocol. Instead of constant buffering or total failure, Flow adapts in real time, retransmitting lost packets while keeping playback as smooth as possible. Even in very bad network conditions, viewers will experience minimal buffering and the best possible stream quality.
No. Each protocol handles encoding, packet loss, transcoding, and distribution differently, meaning performance can vary widely.
Yes, but some protocols can handle brief internet outages without disrupting the stream. Features like forward error correction and packet retransmission help keep streams stable during short dropouts.
Most don’t, but some, like BoxCast Flow, use link quality adjustment to dynamically adapt bitrate based on real-time network conditions.
It depends on the protocol. UDP-based protocols typically move forward without retransmitting lost packets, which can lower video quality. TCP-based protocols attempt retransmission, but this often causes buffering and delays.
It depends on your bitrate, encoder, and protocol. A good rule of thumb is to have at least double your stream's bitrate in upload speed for a stable experience.
If you stream live video long enough, network issues aren’t a possibility—they’re inevitable. Whether it’s a bandwidth drop, packet loss, or a failing connection, every broadcaster will face a moment when their stream is put to the test.
Different streaming protocols handle disruptions in different ways. RTMP sacrifices smooth playback for reliability. SRT adapts but still struggles under severe network stress.
BoxCast Flow is built differently. Instead of buffering, freezing, or dropping out, it makes real-time adjustments to keep streams stable and viewers engaged. Every week, it saves over 10,000 broadcasts from major network failures—ensuring streams don’t just stay online, but remain high-quality and professional.
The real question isn’t if your network will fail—it’s whether your stream is ready when it does. With BoxCast Flow, you’re built for the real world.
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