Technologies
IP Streaming
Professional video and audio transmission over IP networks
IP streaming carries video and audio signals over local networks, fibre connections and the public internet.
Different transport protocols are available for different applications.
Some methods are designed for minimal delay, while others offer advantages over unstable networks or when distributing content to many receivers.
SRT – Reliable transport over public networks
SRT stands for Secure Reliable Transport.
The protocol was developed for reliable transport of time-sensitive data over IP networks and is widely used in professional broadcasting.
SRT is based on UDP and adds mechanisms to detect lost packets and request their retransmission.
Configurable latency allows the system to accommodate short-term network fluctuations and packet loss within certain limits.
SRT also supports optional AES-based encryption.
Why SRT rather than RTMP or plain UDP?
Compared with RTMP
RTMP is traditionally used to deliver live streams to streaming servers.
Classic RTMP uses TCP. Packet loss can cause delays when missing data needs to be retransmitted.
SRT instead provides selective retransmission of lost UDP packets within a defined time window.
This makes SRT particularly relevant to professional point-to-point links over public internet connections.
Compared with plain UDP
UDP has low protocol overhead and is suitable for time-sensitive transmission.
However, UDP itself has no mechanism for recovering lost packets.
SRT adds retransmission-based loss recovery, receive buffering and other transport functions to UDP.
This can often improve stability over variable internet connections.
SRT is not inherently faster or lower-latency than UDP. Additional reliability requires buffering time and, where necessary, additional bandwidth.
Overview of streaming methods
| Method | Main characteristics | Typical application |
|---|---|---|
| SRT | UDP-based, packet retransmission, configurable latency, optional encryption | Remote production, contribution |
| RTMP | Classic TCP-based streaming, broad ingest support | Delivery to streaming platforms |
| UDP / MPEG-TS | Low overhead, no built-in packet retransmission | Local broadcast networks |
| RTP | Timestamps and sequence numbers for real-time media | Professional audio/video networks |
| RIST | Reliable IP media transport with selective packet retransmission (ARQ) | Broadcast contribution and distribution |
| WebRTC | Low-delay real-time communication | Interactive video applications |
| HLS | Segmented HTTP streaming, broad device support | Delivery to end users |
| MPEG-DASH | Adaptive HTTP streaming | OTT platforms and video distribution |
Actual delay depends on the network, codec, buffering and system configuration for every method.
SMPTE ST 2110 – Professional media transport over IP
SMPTE ST 2110 is a family of standards for transporting professional media over IP networks.
Unlike many traditional streaming methods, video, audio and ancillary data are carried as separate streams that can be synchronised.
ST 2110 is used especially in modern broadcast production environments where IP infrastructure complements or replaces conventional SDI connections.
Key parts include:
ST 2110-20: Uncompressed active video.
ST 2110-30: PCM audio.
ST 2110-40: Ancillary data, including certain metadata.
ST 2110-21: Requirements for video stream traffic timing and corresponding receiver buffering.
Synchronisation typically uses Precision Time Protocol (PTP) according to SMPTE ST 2059.
ST 2110 is not a direct replacement for SRT. SRT is often used for reliable transport over public or variable IP networks, whereas ST 2110 primarily serves professional media distribution within appropriately designed production networks.
Elbstone supports the integration of IP-based transmission systems, from SRT links and streaming gateways to professional broadcast IP infrastructure.