SRT is usually the better starting point for sending live video across an unpredictable network because it is designed to recover from packet loss and adapt to changing network conditions, and it supports AES encryption. RTMP is still the right choice when your destination or established workflow requires RTMP. Neither protocol wins in every setup: confirm that both ends support your choice, then weigh network conditions, latency, security, and codec requirements.
What SRT and RTMP do
SRT: transport designed for variable networks
Secure Reliable Transport (SRT) is an open-source video transport technology for moving live video over IP networks. The SRT Alliance and Haivision describe features including adaptive retransmission, packet-loss recovery, and AES encryption. Those capabilities can help on a public or otherwise unpredictable network, but they depend on compatible, correctly configured sending and receiving endpoints. See the SRT Alliance FAQ and Guide Helped? Tip Us
RTMP: a mature option when the workflow calls for it
Real-Time Messaging Protocol (RTMP) remains in live-streaming use, particularly where a receiving service or existing workflow specifies RTMP ingest. Haivision’s comparison, published February 8, 2022, describes RTMP as TCP-based, with retransmission and adjustable buffers. Its implementation and codec comparisons are time-specific; check current documentation for the actual endpoints in your workflow rather than treating that article as a universal current compatibility rule. Haivision’s RTMP and SRT comparison.
Choose based on your workflow
Your situation
Better starting point
Verify before choosing
Sending contribution video over a variable public network
SRT
Both endpoints support SRT; the latency buffer allows time for recovery; and the firewall and network permit the connection.
The destination specifies RTMP ingest
RTMP
The destination’s current ingest specification, required codec, and stream settings. A sender cannot make an incompatible destination accept another protocol.
You prioritize protected contribution transport
SRT is a documented option
Confirm that encryption is enabled and configured at both ends. SRT’s AES capability does not establish that every deployment uses it, and it does not prove that every RTMP deployment is unencrypted.
You need a low-latency end-to-end workflow
Test the complete setup
Measure the delay you actually need to manage, including encoding, transport buffering, receiver processing, and playback. Protocol names alone do not establish end-to-end latency.
You need a particular codec, especially HEVC
Check current endpoint support
Confirm what the sender and receiver implement. A 2022 comparison’s codec description should not be treated as a universal rule for current protocol extensions or products.
The choice is constrained first by compatibility: the sending and receiving ends must both support the selected protocol. OBS documents an SRT URL workflow, which shows that dedicated SRT hardware is not inherently necessary; it does not mean every streaming destination accepts SRT. OBS’s SRT and RIST guide.
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Understand the latency and recovery trade-off
SRT’s recovery mechanisms need time to retransmit missing data. A buffer that allows more recovery time can help delivery under packet loss, but it also contributes to delay. Choose and test settings against the conditions and latency target that matter to your workflow; there is no universal SRT-versus-RTMP latency figure established here.
End-to-end delay is a system outcome, not just a protocol setting. The encoder, network, transport buffer, receiving application, processing, and playback all affect it. Test the full path under realistic network conditions rather than assuming that changing protocol alone will meet a latency target.
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Check compatibility and configure the connection
Read the receiver’s current ingest requirements. Identify the accepted protocol, address and port, codec, and any required stream configuration. If the receiver specifies RTMP, use RTMP unless it documents another supported option.
Confirm support at the sending end. Check your encoder or software’s current protocol options and version. For OBS, the project documents an SRT URL workflow using an address in the form srt://IP:port and MPEG-TS protocols supported through FFmpeg. Follow the receiver’s exact address, port, and connection instructions; the example form is not a destination address.
Coordinate network access. Confirm that the firewall and network allow the required connection and port. For SRT, verify that the chosen latency and recovery settings suit the link and receiver.
Configure security at both ends when using SRT encryption. SRT supports AES encryption, but the endpoints must be configured compatibly. Haivision’s official project notes that encryption may depend on build configuration; verify the implementation you are using rather than assuming the feature is enabled. Haivision SRT project.
Run a complete test. Check that the receiver gets a stable picture and sound, that the actual delay fits your use, and that the connection behaves acceptably under the network conditions you expect. Recheck after changing endpoint versions, codec settings, or network configuration.
Common problems and what to check
The receiver never connects: Recheck protocol compatibility, the receiver-provided address and port, firewall rules, and whether both ends are configured for the same connection workflow.
SRT video breaks up on a lossy link: Check network stability and whether the configured buffer allows enough time for recovery. More recovery time can increase delay, so retest against your latency requirement.
The video arrives but latency is too high: Inspect the entire chain, including encoder, transport buffer, receiver processing, and playback. Do not assume the protocol alone explains the delay.
Encryption does not work: Verify compatible encryption settings at both endpoints and confirm the software build supports the required configuration.
The codec is rejected: Check the current sender and receiver specifications. A general statement about one protocol’s codec support does not guarantee that a particular implementation accepts it.
SRT and RTMP are ways to transport a live feed between compatible endpoints. If your goal instead is to keep uploaded, pre-recorded videos running as a 24/7 YouTube channel, StreamNeo is a separate cloud service: upload a recording or build a playlist, add your YouTube stream key once, and go live. It loops uploaded videos; it does not go live from a camera and streams to YouTube only.
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