Creative Broadcast Agency

Venue testing

Packet loss test for live streaming: why streams drop when speed looks fine.

A packet loss test measures the share of packets that never arrive. Live video has no time to wait for a re-send, so loss of 1% can break a stream on a line that speed-tests perfectly. This is why it happens, the thresholds StreamCertified uses, and how we test a venue for loss before we go live.

Reading time
4 min read
Updated
Updated
Run the free packet loss test
Loss for Ready
< 0.5%
Not ready from here
1%
Probes per test
40
Port we test on
Wired
The short answer

Why run a packet loss test before a live stream?

A packet loss test shows the failure a speed test cannot see. A live stream drops when packets go missing, even on a fast line. Speed is how much a connection can carry; packet loss is how much of it never arrives. A venue can show 100 Mbps upload and still lose enough packets to freeze a stream.

Loss is the most common reason a stream that "tested fine" fails on the day. It is also the one a normal speed test does not report. We test for it on every venue, with the StreamCertified packet loss test.

Run the free packet loss test on StreamCertified

What loss is

What packet loss is

Every video stream is cut into thousands of small packets. Packet loss is the percentage that never reach the other end. They are dropped by a congested link, an overloaded switch, weak Wi-Fi, or a firewall that quietly discards them.

The receiver cannot rebuild a picture from packets it never got. So the viewer sees blocks, frozen frames, audio glitches, or a stream that stops and reconnects.

Why speed tests miss it

Why speed looks fine but the stream still breaks

Live protocols have no time to wait. The show is happening now.

  • SRT re-sends lost packets inside a small latency window. Light loss is recovered and nobody sees it. Past that window, the missing packets are gone and the picture breaks. See RTMP vs SRT.
  • RTMP runs over TCP, which re-sends everything. On a lossy line it slows down, the encoder's buffer fills, and the stream stalls or drops.
  • WebRTC favours speed over completeness. Loss shows up straight away as a soft or broken picture.

There is a second trap. Ping and most speed tests use TCP and ICMP. SRT and WebRTC use UDP. Plenty of venue firewalls pass TCP and ping perfectly while dropping or throttling UDP. The speed test looks great. The encoder cannot hold a connection. That is why we check the protocol path as well as the line.

Thresholds

How much packet loss is too much

Technical specification

Broadcast tolerances are far tighter than web browsing. These are the thresholds StreamCertified uses. The verdict is strict: under 0.5% is ready, and 1% or more fails the room for a live stream.

Packet lossVerdictWhat it means for the show
Under 0.5%ReadyBroadcast-grade. Multi-camera, 4K, news-grade live.
0.5% to under 1%CheckComfortable for 1080p, with no margin left.
1% to 2.5%Not readyOccasional artefacts, especially under load.
Above 2.5%Not readyVisible stutter, freezes, dropped streams.

Loss also compounds with jitter and latency. A room that is borderline on all three is worse than any one number suggests.

What the test measures

What the packet loss test measures, and what it does not

The tool sends 40 requests to its test server, each with a 2-second timeout. The loss figure is the share that do not come back in time.

We are clear about its limit, and so is the tool. It measures loss over HTTP, not UDP loss directly, and it says so on every result. A clean reading tells us the line itself is not dropping traffic. It does not prove the firewall passes UDP. So for SRT or WebRTC we pair it with the NAT type check and, where something at the room listens for an incoming feed, the inbound port check. The final proof is a test stream on the real protocol.

How we do it

How we test a venue for packet loss

  1. Wired first. We test on the port the encoder will use. If the venue insists on Wi-Fi, we test that too and show them the difference.
  2. Test under load. Loss appears when the network is busy. We run the test during registration, breaks, or another event in the building.
  3. Repeat it. One clean run can be luck. We run it several times across the visit, and name the room so the readings stay together.
  4. Push a real stream. Before show day we send a test stream on the real protocol to our receiver and read SRT's own statistics: loss, retransmissions and round-trip time.

A site visit is the right time for this. Read our site survey glossary entry for what else we check in the room.

The second path

What we do when a venue loses packets

We ask venue IT for a dedicated line or VLAN, and for the UDP ports our protocol needs. We give SRT a larger latency window so it has time to recover what is lost. And we add a second path: bonded 5G, running alongside the venue line. If the venue network starts dropping packets, the stream carries on over cellular, so a failure never reaches your audience.

This is how we run live streaming in Dubai, from hotel ballrooms to exhibition halls.

Run the free packet loss test on StreamCertified

If the result worries you, tell us which venue. We will test it before you commit.

FAQ

Questions we get asked.

What is an acceptable packet loss for live streaming?

Why does my stream freeze when my internet speed is fine?

Does SRT fix packet loss?

Can I test a venue for packet loss myself?

Tell us the date, the venue and who is watching.

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