SRT is usually the better starting point for sending live video across an unpredictable network because it is designed to recover from packet loss, adapt to changing network conditions, and support encryption. RTMP is still the right choice when the destination or an established workflow requires RTMP. In practice, endpoint compatibility comes first: both the sender and receiver must support the protocol you choose.
What SRT and RTMP do
SRT (Secure Reliable Transport) and RTMP (Real-Time Messaging Protocol) carry live video between parts of a streaming workflow. They are transport choices, not guarantees about the quality or delay viewers will experience. The sender, receiver, network, encoding, buffering, and playback path all matter.
SRT
SRT is an open-source transport technology designed for video over unpredictable IP networks. Its documented capabilities include adapting to jitter and bandwidth fluctuations, recovering lost packets through retransmission, and AES encryption. See the Haivision SRT documentation and the SRT Alliance FAQ for the respective capability descriptions.
RTMP
RTMP is a mature protocol that remains in live-streaming workflows. Haivision’s comparison, published February 8, 2022, describes RTMP as TCP-based, with retransmission and adjustable buffers. Its detailed comparisons—including codec statements—reflect the article’s publication context, not necessarily every current implementation. Check the current requirements of your sender and destination rather than treating that comparison as a universal support matrix: Haivision’s RTMP and SRT comparison.
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Which protocol should you choose?
| Your situation | Better starting point | What to verify |
|---|---|---|
| Sending contribution video over a variable public network | SRT | Confirm both endpoints support SRT, choose a recovery buffer that fits your latency target, and check firewall and network requirements. |
| The platform or workflow specifies RTMP ingest | RTMP | Follow the destination’s current ingest, codec, and stream configuration requirements. A protocol’s theoretical benefits cannot override an endpoint requirement. |
| Protected contribution transport is a priority | SRT is a documented option | SRT supports AES encryption; verify that encryption is configured and supported at both endpoints. |
| Low end-to-end delay is the priority | Test the complete workflow | Measure encoding, transport buffering, receiving, processing, and playback together. A protocol name alone does not establish the delay viewers will see. |
| You need a particular codec, especially HEVC | Verify current endpoint support | Do not rely on a general or older protocol comparison. Confirm the exact codec and protocol combination accepted by both sender and receiver. |
The deciding question is often not which protocol sounds better in isolation, but which one both ends of your actual workflow support under acceptable network and latency conditions.
How SRT’s recovery and latency trade off
SRT’s recovery mechanisms need time to retransmit missing data and deliver it in order. That recovery buffer can help when packets are lost, but the time allowed for recovery is part of the latency budget. A tighter target leaves less room to recover; a more forgiving buffer may improve resilience while adding delay.
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Test with the network conditions you expect to encounter, not just a clean local connection. Include the chosen encoder settings, transport buffer, receiver, and playback path. There is no universal SRT-versus-RTMP latency figure established here, and a low transport delay does not by itself guarantee low viewer-facing delay.
Encryption and endpoint setup
SRT’s AES capability makes it a viable option when protected contribution transport matters, but capability is not the same as enabled protection. Check that the sender and receiver both support and correctly configure encryption. Do not assume every RTMP deployment is unencrypted; the security of a specific workflow depends on its implementation and configuration.
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Before changing protocols, confirm the destination’s ingest instructions, required codec, connection direction, port, firewall rules, and any authentication or encryption settings. If the destination only accepts RTMP, sending SRT to it will not work unless an intermediate system receives SRT and forwards a compatible stream.
Can you use SRT without dedicated hardware?
Yes. Dedicated SRT hardware is not inherently required. OBS Studio documents an SRT URL workflow using an address such as srt://IP:port, and documents MPEG-TS protocols supported through FFmpeg. This shows that software can be part of an SRT workflow; it does not mean every streaming destination accepts SRT. Consult the OBS Project SRT guide and verify the receiving endpoint’s support.
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What to check before switching
- Compatibility: Confirm the protocol is supported at both ends, including any relay or gateway in between.
- Network path: Check ports, firewall rules, connection direction, and whether the network permits the required traffic.
- Latency target: Account for encoding, transport recovery buffer, receiving, processing, and playback rather than setting a target for transport alone.
- Stream configuration: Confirm the destination’s current codec and other ingest requirements.
- Security: If using SRT encryption, verify that it is configured on both sides rather than assuming it is active.
- Realistic testing: Test under the network conditions and workflow the stream will actually face, then adjust buffer and latency settings together.
Common problems and fixes
The receiver never connects
First confirm that the receiver accepts the selected protocol and that the SRT address, port, and connection direction match its instructions. Then check firewall rules and whether an intermediate relay is required. A valid SRT sender cannot connect directly to an RTMP-only ingest endpoint.
The stream connects but stutters or drops packets
Check network stability and bandwidth variation, then review SRT recovery and latency-buffer settings at both endpoints. More time for recovery may help with loss but can increase delay; test changes against the full workflow.
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The stream is stable but too delayed
Review the complete delay path, including encoder, transport buffer, receiver processing, and playback. Reducing the recovery buffer may reduce transport delay while making recovery less tolerant of network loss, so validate the trade-off under representative conditions.
Video or audio is rejected or missing
Verify that the destination supports the codec and stream configuration you are sending, as well as the chosen transport. Do not infer current HEVC support from an older general comparison; check both endpoint specifications.
You expected encryption but cannot confirm it
Check the SRT configuration and encryption support on the sender and receiver. The protocol’s AES capability does not establish that a particular connection has encryption enabled.
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