Use SRT when both ends of your workflow support it and you need a live video contribution link that can recover from packet loss and jitter on an unpredictable network. Use RTMP when the platform, encoder, or established workflow requires RTMP. Neither protocol is the universal winner: compatibility comes first, and SRT’s recovery buffer trades some delay for a better chance of recovering late or lost packets.
What is the difference between SRT and RTMP?
SRT (Secure Reliable Transport) is an open-source transport technology designed for live-video contribution over unpredictable IP networks. Haivision describes it as compensating for jitter and bandwidth fluctuations and recovering from typical Internet packet loss. It runs over UDP and adds mechanisms including acknowledgments and retransmission (ARQ), latency management, and encryption; the project documentation also covers FEC, connection bonding, and Stream ID.
RTMP remains useful when a receiving platform or an existing production workflow expects it. The available official comparison resource is a Haivision white-paper landing page, not an independently verified head-to-head benchmark. The technical material reviewed does not establish comparable RTMP figures for latency, loss recovery, or encryption, so a blanket claim that one protocol is faster or more reliable would go beyond the evidence.
| Decision point | SRT | RTMP |
|---|---|---|
| Network conditions | Designed to handle unpredictable networks using jitter compensation and packet-loss recovery. | Choose it when the destination or workflow requires RTMP; comparable verified technical figures are not established in the sources cited here. |
| Latency and buffering | Configurable recovery buffer; the Haivision SRT Protocol 1.5.4 documentation gives a range of 20–8000 ms. | A comparable verified numeric value is not established in the sources cited here. |
| Encryption | Encryption is documented, including AES; whether it is enabled depends on the actual implementation and configuration. | Security variants and implementation details are not established by the sources cited here. Check the selected endpoints and configuration. |
| Compatibility | Both sending and receiving endpoints must support SRT. Stream ID can help route multiple SRT ingest streams to one address and port in supported workflows. | Use it when an endpoint or established workflow expects RTMP. |
How to choose for your workflow
Choose SRT for a variable contribution link
If you are sending a live feed over an Internet path where packet loss, jitter, or changing bandwidth is a concern, SRT is a strong candidate—provided the source and destination both support it. Its retransmission and buffering mechanisms are intended to help with those conditions. They do not guarantee uninterrupted delivery, and their effectiveness depends on the actual link and configuration.
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Choose RTMP when the destination expects it
If the streaming platform’s ingest service, encoder, decoder, or production workflow calls for RTMP, using SRT instead will not help unless both ends can accept and pass SRT. Confirm the exact ingest option rather than assuming a platform supports every protocol its hardware or software ecosystem may support.
Do not decide on an unverified speed claim
There is no neutral, controlled SRT-versus-RTMP benchmark in the sources cited here that establishes a universal winner for latency, bandwidth use, or packet loss. Test with the actual endpoints and network if those results matter to your production. Haivision’s comparison resource is vendor-authored; its landing page does not provide a neutral benchmark.
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Set SRT latency for the link, not by guesswork
SRT latency is a buffer window that gives the receiver time to reorder or retransmit packets. It is not the total glass-to-glass delay from the camera or encoder to the viewer. A smaller buffer may reduce this part of the delay, but if packets arrive too late for recovery, lowering the setting can undermine the recovery you chose SRT for.
Haivision’s SRT Protocol 1.5.4 documentation, last updated February 12, 2026, describes a fixed latency buffer setting from 20 to 8000 ms and says it should reflect the link’s characteristics. For a “fairly good network” with 0.1 to 0.2% loss and no significant burst loss, its rule of thumb is SRT Latency = RTT Multiplier * RTT, using four times the round-trip time (RTT). The documentation says the higher configured source or destination value is used. This is a version-specific vendor guideline under the stated conditions, not a universal guarantee or a measure of typical Internet loss.
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Measure RTT and observe whether loss is sporadic or bursty on the contribution path. Start with settings appropriate to those conditions, then test end-to-end. Account separately for encoder processing, other transport and platform buffers, and playback delay; the SRT buffer alone does not tell you when a viewer will see the video.
Check endpoint support and configuration before switching
- Verify both ends. Check the exact sender model or software and the destination’s current ingest service for SRT support. If either endpoint only accepts RTMP, use RTMP for that connection.
- Confirm the SRT mode and settings. Implementations may expose different features. The SRT project documents ARQ, FEC, bonding, Stream ID, and AES encryption, but that does not mean every product supports or enables each feature.
- Configure security deliberately. SRT encryption is a documented capability, not proof that a particular stream is encrypted. Confirm that the chosen implementation supports the required settings and that both endpoints are configured consistently.
- Tune and test latency on the real path. Use measured RTT and observed loss, including bursts, to choose the recovery buffer. Check the actual delivery delay as well as whether the receiver recovers packets.
- Test the complete route. Validate the source, network, receiver, and onward distribution with the intended video before relying on the connection for a live production.
Standards and version context
Haivision says SRT was released as open source in April 2017 in partnership with Wowza Media Systems and is distributed under the MPL-2.0 license. The IETF Datatracker record for “The SRT Protocol” is an individual Internet-Draft authored by Haivision and SK Telecom personnel, dated September 7, 2021, and expired on March 11, 2022. The Datatracker marks it archived and says it is not endorsed by the IETF and has no formal standing in the IETF standards process; it should not be described as an IETF standard.
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Protocol documentation and product support can change. The latency guidance cited above is for Haivision SRT Protocol 1.5.4; its introduction is in the 1.5.3 documentation. Check the documentation and exact product versions you will deploy.
Independent reader supportYour contribution helps us test, update, and keep practical guides available for everyone.Sources
- Haivision SRT Protocol introduction
- SRT project documentation and repository
- Haivision SRT Protocol 1.5.4 latency guidance
- IETF Datatracker: The SRT Protocol Internet-Draft
- Haivision SRT vs. RTMP comparison resource
- Haivision product catalog
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If your goal is to keep a YouTube channel live 24/7 from uploaded videos, StreamNeo is a different kind of solution: it handles the always-on playback in the cloud rather than asking you to operate a contribution link from a live encoder. Upload a recording or build a playlist, add your YouTube stream key, and go live. StreamNeo loops uploaded videos to YouTube; it does not stream from a camera or to other platforms.
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Quick Recap
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