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Traditional protocols like HTTP, designed for web browsing, struggle to meet the stringent requirements of real-time streaming due to factors like latency and packet loss. Thankfully more advanced protocols are here to save the day like HLS, WebRTC, RTMP and SRT. In this blog, let’s take RTMP and SRT and compare both in terms of latency, reliability, security and compatibility.
Live streaming protocols are used for transmitting live video and audio over the internet in real time ensuring a low latency and secure experiences. The most common protocols include RTMP, known for its low latency and wide support next one HLS, which offers scalability and compatibility, especially on iOS devices; and DASH, an open standard that supports adaptive bitrate streaming and various codecs. SRT provides secure, low-latency streaming over unpredictable networks, making it ideal for remote contributions and live sports. WebRTC, with its very low latency and peer-to-peer capabilities, is perfect for video conferencing and interactive broadcasts. Each protocol has unique strengths, allowing content creators to choose the best option for their specific streaming needs.
RTMP or real-time messaging protocols is a one of the most used live streaming protocols. RTMP streaming was developed by Macromedia for their flash player when Macromedia was acquired by Adobe it became Adobe Flash Player. RTMP is based on Transmission Control Protocol (TCP) technology.
TCP is a standard protocol that helps with transmission between data and the device over the internet. It is known for being the more reliable compared to other protocols as TCP requires an already established connection before transmitting any data, but it also can make the transmission slower than others. RTMP was developed to transmit between a hosting server and a video player, but its purpose now has taken a different role, in today’s streaming world RTMP’s main job is to send content from an encoder to an online video host, this process called "ingestion.
The reason RTMP is the universal choice for live streaming is because of its capability of low latency and ensuring that live content reaches viewers with minimal delay or latency. RTMP also supports high-quality video and audio and it’s widely compatible with many encoder, streaming platforms and CDNs making it a vast choice for streaming. Besides providing a low latency RTMP allow real time interactivity features like live chat.
Secure Reliable Transport (SRT) is an open-source video transport protocol designed by Haivision to deliver high-quality, low-latency video over unreliable networks. SRT is built on UDP (User Datagram Protocol) but it has features from TCP like flow control and congestion control.
SRT can provide even lower low latency compare to RTMP, making it the best choice real time application where a single second timing matters.
SRT uses strong encryption measures to secure video content from unauthorized access during transmission, enhancing privacy and data security. The only drawback of SRT is, like RTMP this protocol is not widely compatible.
Now that we understand the basic difference of both the protocols. Let’s see the technical difference in more details.
In terms of RTMP it relies on a buffering mechanism to ensure smooth playback, which can cause latency in case of network issues or packet loss. However, RTMP is operating over TCP, thus it prioritizes reliable data delivery over speed, with TCP’s the feature like error correction and retransmission mechanisms add up to the latency. Due to this RTMP typically incurs a fixed latency ranging from 3 to 30 seconds, depending on network conditions and server configurations.
SRT is specially designed to minimize latency using a UDP-based transport layer, which avoids the delays caused by TCP’s error correction. SRT uses a selective retransmission method called Automatic Repeat reQuest (ARQ), which only resends lost packets, cutting down on delays. Additionally, SRT lets users adjust the latency based on network conditions, often achieving very low latencies of less than one seconds. This makes SRT perfect for live events where keeping delays to a minimum is essential.
The first graph shows that SRT has lower latency than RTMP. SRT starts at 50ms in stable conditions and only goes up to 120ms with high packet loss. In comparison, RTMP starts at 60ms and jumps to 250ms with high packet loss. This makes SRT better for real-time streaming.
The second graph shows that as packet loss increases, SRT maintains higher video quality compared to RTMP. This highlights SRT's better resilience and error correction, making it more reliable for streaming in unstable network conditions.
RTMP is based on TCP which is best feature to provide reliable data as it requires an already established connection before transmitting any data delivery. It ensures that packets are delivered in the correct order and retransmitting lost packets. This reliability is useful in stable network conditions where ensuring every packet arrives intact is needed. However, in unreliable or congested networks, RTMP can struggle to maintain a stable connection, leading to potential latency or interruptions in the stream.
SRT on the other hand is not really designed for reliable network. It uses UDP which sacrifices ordered, guaranteed delivery in return of lower latency and better performance over bad networks conditions. SRT includes mechanisms such as packet loss recovery, retransmission, and dynamic bit rate adjustment to enhance reliability even in challenging network conditions.
RTMP uses SSL/TLS encryption to secure data during transmission. This method encrypts the entire stream, ensuring data stays private while it's being sent. RTMP also supports authentication like usernames, passwords, or tokens to verify who's accessing the data. It's widely used but can be less efficient in high-volume scenarios due to the continuous encryption process.
The best feature for SRT is it security, it offers strong security designed for reliable data transfer over unreliable networks. It uses AES-128/256 encryption with flexible key management, encrypting only the necessary data to conserve resources. SRT includes better authentication options such as passwords or X.509 certificates to ensure secure connections. It also has built-in error recovery features like retransmission, which protect against data interception or tampering. SRT is the best option for applications needing high security.
RTMP is one of the most used protocols for live streaming, as it is supported by wide range of players, browser and platforms. RTMP is highly compatible with older streaming platforms, software, and devices. However, as streaming technology evolves, RTMP may require additional configuration to work with newer platforms and devices, which might not natively support it.
SRT (Secure Reliable Transport), while designed for modern streaming needs and offering superior performance in terms of reliability and security, is not yet as widely supported across all platforms and devices. It is gaining traction and being increasingly adopted by newer streaming solutions, but it might not be natively compatible with many older systems. This means that while SRT offers advanced features, its use may be limited to platforms and devices that have adopted this newer technology, and additional integration efforts might be required for widespread compatibility.
While deciding between RTMP vs SRT, both protocols bring unique strengths to the table. RTMP remains a constant choice for its simplicity and widespread support, especially for traditional live streaming setups. On the other hand, SRT stands out with its security, lower latency, and resilience to network issues, making it ideal for modern, demanding streaming environments. So, in the question of which is better, the answer remains in your requirement and needs.
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