🛰️ Supported Protocols (RTMP, SRT, WHIP)

🎥 Overview

InterScribe supports several industry-standard streaming protocols that allow you to ingest high-quality audio (and optional video) from professional production tools like OBS, vMix, Wirecast, or hardware encoders. These protocols are ideal when you're already producing a livestream and want to generate real-time captions and translations for remote audiences.

⚠️ Important: These protocols are not recommended for in-person or interpreter-facing events due to the inherent delay. For low-latency use cases, see Desktop Agent Setup.

📡 Protocol Comparison

🔁 Protocol Details

🟥 RTMP – Real-Time Messaging Protocol

Overview: RTMP is the most common protocol for streaming video and audio. It’s supported by virtually all livestreaming tools and platforms, including OBS, Wirecast, vMix, and many hardware encoders.

✅ Pros:

• Extremely widespread support
• Easy to set up with a URL + stream key
• Great for platforms like YouTube, Facebook, etc.

⚠️ Cons:

• High latency (typically 5–10 seconds)
• Poor for live interpretation or in-room playback
• No encryption unless tunneled via TLS

💡 Best For: Pre-recorded streams, keynote broadcasts, or remote-only events where delay is acceptable.

🟦 SRT – Secure Reliable Transport

Overview: SRT is a secure, open-source protocol built for low-latency and resilient transmission over unstable or unpredictable networks.

✅ Pros:

• Better performance over poor networks (e.g. packet loss, jitter)
• Lower latency than RTMP (2–5 seconds)
• Encryption built-in

⚠️ Cons:

• Requires encoder support (e.g. OBS with SRT plugin, vMix, LiveU)
• Setup is more complex (SRT URL includes port, stream ID, passphrase)

💡 Best For:

• Online events with unstable networks or higher quality requirements.*

🟨 WHIP – WebRTC-HTTP Ingest Protocol

Overview: WHIP is a new WebRTC-based protocol designed to enable ultra-low-latency streaming to media servers over a simplified HTTP interface.

✅ Pros:

• Latency typically under 1 second
• Based on modern WebRTC technology
• Ideal for real-time use cases

⚠️ Cons:

• Still emerging; not supported by most traditional encoders
• May require custom tools or experimental plugins (e.g. GStreamer, Elemental Live, Janus)

💡 Best For: In-Person sessions

🌐 InterScribe's Native Agents Use WebRTC

While this page focuses on external protocols like RTMP, SRT, and WHIP, it's important to note that InterScribe’s own audio input tools — including the Streamer Dashboard, Desktop Agent, and Web Agent — use WebRTC-based streaming behind the scenes.

Benefits of InterScribe’s built-in WebRTC streaming:

• Ultra-low latency (often under 1 second)
• No encoder setup required
• Tightly integrated with AV Channels and event logic
• Best suited for live, in-person, or interpreter-driven events

✅ Recommended: Use these agents for sessions where real-time responsiveness is critical — such as interpretation booths, hybrid meetings, or local audiences.

If you're using external encoders (like OBS or vMix), then RTMP, SRT, or WHIP are your available protocols — but expect added latency due to the nature of those formats.

🛠️ Best Practices

• 🟢 Use InterScribe Native Agents (WebRTC) — including the Streamer Dashboard, Desktop Agent, and Web Agent — for the lowest latency and best experience in real-time and in-person events.

• 🟡 Use RTMP for maximum compatibility with tools like OBS, vMix, or hardware encoders — but only for online-only events, as it introduces noticeable delay.

• 🟠 Choose SRT when you need more resilience over unstable networks. It offers better performance than RTMP, but is also best suited for online events due to moderate delay.

• 🔴 Try WHIP only if you're working with a compatible encoder or integration and can’t use InterScribe's native agents. WHIP is low-latency but requires advanced configuration and is still emerging.

• ✅ Always test your full setup — including the audio source, ingress configuration, and session playback — before going live.

• 🎯 Make sure you've created the required Ingress under A/V Inputs → Ingresses and assigned it to the correct AV Channel before streaming.

❓ FAQs

Can I use multiple protocols at once?

You can create multiple Ingresses for different AV Channels, each using a different protocol. However, each session can only receive audio from one assigned AV Channel at a time.

Do these protocols support video too?

Yes — all protocols support video, but InterScribe only uses the audio for captioning and interpretation. You can embed the video (e.g. YouTube/Vimeo) separately in the session.

Can I use WebRTC with OBS or vMix?**

No. OBS/vMix typically use RTMP or SRT. WebRTC is used internally by InterScribe's own agents like the Streamer Dashboard or Desktop Agent.

Which protocol is best for interpretation booths?**

WebRTC via the Desktop Agent or Streamer Dashboard is ideal due to ultra-low latency and ease of configuration.

Will these protocols work with the Desktop or Web Agent?

No — these protocols are for external encoders only. The Desktop Agent and Web Agent use direct browser or system-level audio streaming instead.

🔗 Related Articles

• Connecting OBS, vMix & Hardware Encoders
• Managing AV Channels & Inputs
• Latency Management & Best Practices
• Desktop Agent Setup

Would you like a visual diagram comparing all input methods and protocols across use cases? I can generate one.