Protocol Selection Guide

WAVE supports five streaming protocols, each optimized for different production scenarios. This guide helps you choose the right protocol based on your latency requirements, source equipment, network conditions, and production goals.

{% callout type=“info” title=“Protocol-agnostic architecture” %} WAVE’s unified streaming architecture means every Live Input supports all protocols simultaneously. You receive ingest URLs for SRT, RTMP, and WebRTC when you create a stream — choose whichever fits your encoder and workflow. Viewers always receive the stream via HLS or WebRTC (WHEP) playback regardless of the ingest protocol. {% /callout %}

Protocol comparison

Protocol	Transport	Typical latency	Video quality	Firewall friendliness	Encryption	Best for
WebRTC	UDP/TCP (ICE)	200-500ms	Good to high	Excellent (TURN fallback on 443)	DTLS/SRTP	Browser ingest, interactive streaming, real-time collaboration
SRT	UDP (ARQ)	200ms-2s	Excellent	Good (single UDP port)	AES-128/256	Remote contribution, professional encoders, unreliable networks
RTMP	TCP	2-5s	Good	Excellent (port 1935 or 443 via RTMPS)	TLS (RTMPS)	Legacy encoders, widest compatibility, simple setup
NDI	TCP/UDP (LAN)	1-3 frames (16-50ms at 60fps)	Lossless to mezzanine	LAN only (not routable over internet)	Network-level	Studio production, multi-camera switching, local networks
OMT	TCP (zero-copy)	Sub-frame (<16ms)	Broadcast-grade (4:2:2 + alpha)	LAN/WAN with configuration	AES	Professional broadcast, ultra-low latency production, multi-protocol bridging

Decision tree

Use these questions to narrow down the right protocol for your workflow.

{% stepper %} {% step title=“What is your latency requirement?” %}

• Sub-second (<500ms): You need WebRTC, NDI, or OMT. Continue to the next question.
• Low latency (500ms-2s): SRT is your best option. It handles packet loss gracefully and works well over the public internet.
• Standard latency (2-5s is acceptable): RTMP gives you the widest encoder compatibility with the simplest setup. {% /step %}

{% step title=“Where is your source relative to WAVE?” %}

• Same LAN or facility: NDI or OMT deliver the lowest latency with the highest quality. NDI is ideal for standard studio equipment; OMT adds broadcast-grade features like alpha channel and 32-channel audio.
• Over the public internet: WebRTC or SRT. WebRTC works directly in the browser; SRT is better for dedicated encoder software like OBS or vMix.
• Unreliable network (cellular, satellite, Wi-Fi): SRT’s ARQ error correction handles packet loss far better than any other protocol. Increase the latency buffer to match network conditions. {% /step %}

{% step title=“Do you need browser-native ingest?” %}

• Yes — no encoder software allowed: WebRTC is your only option. It runs natively in Chrome, Firefox, Edge, and Safari with no plugins.
• No — encoder software is available: SRT provides better quality and reliability than WebRTC for the same latency class. Use SRT with OBS, vMix, Wirecast, or FFmpeg. {% /step %}

{% step title=“What is your quality priority?” %}

• Broadcast-grade (lossless or near-lossless): NDI (LAN) or OMT (LAN/WAN) deliver the highest quality. OMT supports 4:2:2 color with alpha channel.
• Professional (high bitrate, reliable delivery): SRT with AES encryption provides professional-grade transport over the internet.
• Good enough for interactive use: WebRTC adapts quality automatically based on available bandwidth. Enable Simulcast for the best viewer experience. {% /step %} {% /stepper %}

Quick recommendation by use case

{% tabs %} {% tab label=“Interactive streaming” %}

Live Q&A, auctions, sports betting, gaming

Recommended: WebRTC

Interactive scenarios demand the lowest possible round-trip latency so that audience participation feels immediate. WebRTC delivers 200-500ms latency directly from the browser.

Setting	Value
Protocol	WebRTC (WHIP ingest)
Simulcast	Enabled (3 layers)
Latency target	<500ms
Fallback	SRT (if using external encoder)

When to choose SRT instead: If your encoder does not support WebRTC (most hardware encoders), use SRT with a 200ms latency buffer for near-real-time performance. {% /tab %}

{% tab label=“Remote contribution” %}

News field units, sports venues, mobile backpack encoders

Recommended: SRT

Remote contribution often traverses unpredictable networks — cellular bonding, satellite uplinks, or congested public Wi-Fi. SRT’s ARQ error correction recovers lost packets without retransmission delays that would plague TCP-based protocols.

Setting	Value
Protocol	SRT (caller mode)
Latency buffer	500-2000ms (match to network quality)
Encryption	AES-256
Fallback	RTMP (if encoder lacks SRT support)

Tip: Start with a 500ms latency buffer and increase it if the WAVE dashboard shows packet recovery exceeding 5%. {% /tab %}

{% tab label=“Studio production” %}

Multi-camera switching, graphics overlay, local production

Recommended: NDI (LAN) or OMT (LAN/WAN)

Studio productions need frame-accurate timing for clean camera cuts and graphics transitions. NDI and OMT both deliver sub-frame latency within a local network.

Scenario	Protocol	Latency
Single facility, standard equipment	NDI	1-3 frames
Multi-facility or WAN-connected studios	OMT	<16ms
Mixed equipment (some NDI, some SDI)	OMT with NDI bridge	<30ms

NDI is the simpler choice for facilities already running NDI-compatible cameras, switchers, and graphics systems (TriCaster, vMix, Blackmagic ATEM).

OMT adds broadcast-grade features: 4:2:2 video with alpha channel, up to 32 audio channels, and SMPTE/EBU compliance. Choose OMT for professional broadcast workflows that require WAN connectivity or advanced quality controls. {% /tab %}

{% tab label=“Traditional broadcast” %}

24/7 channels, scheduled programming, simulcast to platforms

Recommended: RTMP or SRT

For traditional one-to-many broadcasting where a few seconds of latency is acceptable, RTMP offers the widest encoder compatibility. If your encoder supports SRT, prefer it for better quality and lower latency.

Setting	Value
Protocol	RTMP (or RTMPS for encryption)
Keyframe interval	2 seconds
Audio codec	AAC-LC at 48 kHz
Fallback	SRT (if encoder supports it)

Migration note: If you are currently using RTMP and want to reduce latency, see the RTMP to SRT migration guide below. {% /tab %}

{% tab label=“Multi-protocol production” %}

Mixed sources feeding a single broadcast

Recommended: Mix protocols per source

WAVE’s unified architecture lets you mix protocols within a single production. Each camera or source can use the protocol that best matches its equipment and network path.

Source	Protocol	Reason
Studio cameras (LAN)	NDI	Frame-accurate, zero-config
Field reporter (cellular)	SRT	Error recovery for lossy networks
Guest via browser	WebRTC	No software install required
Legacy encoder	RTMP	Backward compatibility

All sources feed into WAVE’s production switcher and are delivered to viewers via a single HLS or WebRTC (WHEP) playback stream. {% /tab %} {% /tabs %}

Protocol details

WebRTC

WebRTC (Web Real-Time Communication) is the only protocol that runs natively in the browser. WAVE uses the WHIP standard for ingest and WHEP for playback.

Strengths:

• No encoder software required — works in Chrome, Firefox, Edge, Safari
• Sub-second latency (200-500ms typical)
• Adaptive bitrate via Simulcast
• Excellent firewall traversal (TURN fallback over TCP 443)

Limitations:

• Lower maximum bitrate than SRT or NDI (browser-constrained)
• Quality depends on the user’s device and browser
• Not suitable for broadcast-grade production workflows

Configuration: See WebRTC Setup for STUN/TURN configuration and Simulcast settings.

SRT

Secure Reliable Transport (SRT) is an open-source UDP protocol with built-in error correction (ARQ) and AES encryption. It is the standard for professional remote contribution.

Strengths:

• Excellent error recovery over lossy networks
• AES-128/256 encryption built in
• Supported by OBS, vMix, Wirecast, FFmpeg, and most hardware encoders
• Configurable latency buffer to match network conditions

Limitations:

• Requires encoder software (not browser-native)
• Single UDP port must be reachable (firewall configuration needed)
• Higher latency than WebRTC on clean networks (due to retransmission buffer)

Configuration: See SRT Configuration for encryption, latency tuning, and caller/listener modes.

RTMP

Real-Time Messaging Protocol (RTMP) is the legacy standard for live streaming. Nearly every encoder and streaming platform supports it.

Strengths:

• Universal encoder compatibility
• Simple setup (URL + stream key)
• TLS encryption via RTMPS
• Mature ecosystem and documentation

Limitations:

• TCP-based — no error recovery, susceptible to buffering on lossy networks
• Higher latency (2-5 seconds typical)
• No built-in encryption (must use RTMPS)
• Flash-era protocol with limited modern development

Best for: Legacy encoders, simple setups where latency is not critical, and as a fallback when other protocols are unavailable.

NDI

Network Device Interface (NDI) is a royalty-free standard from Vizrt for high-quality video over local area networks.

Strengths:

• Frame-accurate timing (1-3 frames of latency at 60fps)
• Lossless to mezzanine quality
• Automatic device discovery via mDNS — zero configuration
• Professional tally, timecode, and metadata support
• Native support in TriCaster, vMix, ATEM, CasparCG, and more

Limitations:

• Designed for LAN only — not routable over the public internet without tunneling
• High bandwidth (up to 150 Mbps per HD source)
• Requires a dedicated, high-capacity network infrastructure (10 Gbps recommended)

Best for: Studio production, multi-camera switching, local facilities with professional broadcast equipment.

OMT

Open Media Transport (OMT) is an open-source, MIT-licensed protocol designed for ultra-low latency professional broadcast over LAN and WAN.

Strengths:

• Sub-frame latency (<16ms glass-to-glass)
• 4:2:2 video with alpha channel support
• Up to 32 channels of uncompressed 32-bit float audio
• SMPTE ST-2110 and EBU R128 compliance
• Zero-copy architecture with GPU/FPGA hardware acceleration
• Native multi-protocol bridging to WebRTC, SRT, RTMP, NDI

Limitations:

• Higher complexity to configure than SRT or WebRTC
• Requires WAVE Enterprise tier for full feature access
• Less widely supported by consumer-grade encoders

{% if-tier minTier=“enterprise” %} Enterprise features: AI-driven quality optimization, predictive scaling, global edge distribution, and multi-tenant isolation are available on the Enterprise tier. {% /if-tier %}

Best for: Professional broadcast workflows requiring the lowest possible latency with broadcast-grade quality, especially when bridging between multiple protocols in a single production.

Bandwidth requirements

Protocol	Resolution	Frame rate	Typical bandwidth
WebRTC	1080p	30 fps	2.5-6 Mbps
SRT	1080p	30 fps	4-8 Mbps
RTMP	1080p	30 fps	4-6 Mbps
NDI	1080p	60 fps	~150 Mbps (mezzanine)
OMT	1080p	60 fps	6-50 Mbps (adaptive)

{% callout type=“tip” title=“Upload headroom” %} Your available upload bandwidth should be at least 1.5 times your target stream bitrate to account for network fluctuations and protocol overhead. SRT adds 2-20% overhead depending on packet loss; WebRTC Simulcast sends multiple quality layers simultaneously. {% /callout %}

Migration guides

Migrating from RTMP to SRT

SRT provides 10x lower latency, better error recovery, and built-in encryption compared to RTMP. Most modern encoders support both protocols.

{% stepper %} {% step title=“Verify encoder compatibility” %} Check that your encoder supports SRT. OBS Studio 27+, vMix 24+, Wirecast 14+, and FFmpeg 4.0+ all include SRT support. Hardware encoders from Haivision, Teradek, and Matrox also support SRT natively. {% /step %}

{% step title=“Test SRT alongside RTMP” %} Create a WAVE stream and send the same content via both RTMP and SRT simultaneously (WAVE’s Live Input accepts both). Compare latency and quality in the WAVE dashboard under Stream > Health. {% /step %}

{% step title=“Configure SRT encryption” %} Enable AES-256 encryption in your WAVE stream settings and configure the same passphrase in your encoder. See SRT Configuration for details. {% /step %}

{% step title=“Switch primary to SRT” %} Once you have verified quality and latency, switch your primary ingest to SRT. Keep the RTMP URL as a fallback in case of encoder issues. {% /step %} {% /stepper %}

Migrating from SRT to WebRTC

WebRTC eliminates the need for encoder software entirely. This is ideal for workflows where contributors stream directly from a browser.

{% stepper %} {% step title=“Evaluate browser requirements” %} WebRTC requires Chrome 90+, Firefox 85+, Edge 90+, or Safari 14.1+. Ensure all contributors have compatible browsers and have granted camera/microphone permissions. {% /step %}

{% step title=“Test browser streaming” %} Open your WAVE stream’s browser ingest page and start a test stream. Compare the latency and quality against your SRT encoder output using the WAVE dashboard. {% /step %}

{% step title=“Enable Simulcast” %} For streams with more than 50 viewers, enable Simulcast to send multiple quality layers. This allows WAVE to serve each viewer the optimal resolution. See WebRTC Setup for configuration. {% /step %}

{% step title=“Deploy for contributors” %} Share the browser ingest URL with contributors. No software installation is needed — they open the link, grant permissions, and start streaming. {% /step %} {% /stepper %}

Adding NDI or OMT to an existing production

If you are already streaming with SRT or WebRTC and want to add local studio sources via NDI or OMT:

1. Connect your NDI or OMT sources to the WAVE production switcher via the dashboard under Production > Sources > Add Source.
2. Select the discovered NDI source from the dropdown, or enter the OMT session configuration.
3. WAVE bridges the local protocol source into your production alongside internet-based sources.
4. All sources are mixed in the WAVE switcher and delivered to viewers via a single HLS or WebRTC playback stream.

{% callout type=“info” title=“Multi-protocol mixing” %} There is no penalty for mixing protocols within a single production. WAVE’s unified streaming architecture handles protocol bridging transparently. Each source uses the protocol that best matches its equipment and network path. {% /callout %}

Frequently asked questions

{% accordion %} {% accordion title=“Can I change protocols without creating a new stream?” %} Yes. Every WAVE Live Input provides ingest URLs for SRT, RTMP, and WebRTC simultaneously. You can switch between them at any time by pointing your encoder to a different URL. NDI and OMT sources are added as production sources through the dashboard. {% /accordion %}

{% accordion title=“What happens if my primary protocol fails?” %} WAVE does not automatically fail over between ingest protocols — the encoder controls which protocol it sends. However, you can configure your encoder with a fallback URL (for example, RTMP as a backup for SRT). For viewer-side resilience, WAVE automatically switches viewers between HLS and WebRTC (WHEP) playback based on their network conditions. {% /accordion %}

{% accordion title=“Which protocol has the best quality?” %} For local production, NDI in mezzanine mode or OMT in lossless mode provide the highest quality. Over the internet, SRT at high bitrate (8-15 Mbps for 1080p60) delivers excellent quality with error recovery. WebRTC quality is constrained by the browser’s encoder and the user’s upload bandwidth. {% /accordion %}

{% accordion title=“Do I need to open firewall ports?” %}

• WebRTC: No. WAVE’s TURN servers relay traffic over TCP 443 if UDP is blocked.
• SRT: Yes. A single UDP port (default 9000) must be open for outbound traffic in caller mode.
• RTMP: Port 1935 (or 443 for RTMPS) must be open for outbound TCP traffic.
• NDI: mDNS (UDP 5353) and TCP ports 5960-5969 must be open within your LAN.
• OMT: Depends on your configuration. Contact WAVE support for WAN deployment guidance. {% /accordion %}

{% accordion title=“Which tier do I need for each protocol?” %} WebRTC, SRT, and RTMP are available on all WAVE tiers including Free. NDI integration requires the Pro tier or higher. OMT with enterprise features (AI optimization, global distribution) requires the Enterprise tier. {% /accordion %} {% /accordion %}

{% contact-support category=“streaming” /%}

{% related-articles /%}