⚑ No App or Sign-Up Needed πŸ”’ 1-on-1 True E2EE πŸ›‘οΈ Zero Audio Retention

No App, No Sign-Up.
Simple, Verifiably Secure Calls

Connect effortlessly even if your peer doesn't have the app. Simply share a disposable link to start an encrypted voice call directly in your browser with one tap.
Audio streams are directly encrypted end-to-end between devices with zero audio stored on servers. Seamlessly bridges with office phone systems and internal SIP extensions.

Early Release Ahead of the full edition, check out our voice-dedicated iOS/Android app: Tsuwa Mini!
iOS & Android Native App Coming Soon
HOW IT WORKS

Start Calling in 3 Simple Steps β€” Even Without the App

No sign-up forms, no password resets, no app store friction. Anyone can join an encrypted voice call immediately.

STEP 1
πŸ”—

Generate & Share Link

Create a temporary, cryptographically secure call URL in one click. Share it via Slack, WhatsApp, email, or SMS.

STEP 2
🌐

Open in Any Browser

The other person simply opens the link. No app download prompts or mandatory account creation screens.

STEP 3
πŸŽ™οΈ

Allow Mic & Start Talking

Grant browser microphone permission and you are instantly connected via peer-to-peer encryption. Clear, lag-free audio begins.

πŸ’‘
Test it right now in your browser Try our live interactive Web Call demo without registration or entering personal details.
Try Web Call Demo β†’
SECURITY & PRIVACY

Security First. Built into Architecture, Not Just Marketing.

We never settle for vague claims like "it's encrypted." From casual chats to sensitive corporate meetings, every transport path and data boundary is rigorously designed.

πŸ”’

1-on-1 Direct True End-to-End Encryption

One-on-one calls use WebRTC-standard DTLS-SRTP. Encryption keys are exchanged directly between devices; neither network providers nor intermediate servers can decrypt audio. When relaying through TURN in restrictive firewalls, encrypted SRTP packets pass untouched without server decryption.

DTLS 1.3 SRTP (AES-GCM) 1-on-1 P2P E2EE TURN Blind Relay
πŸ›‘οΈ

Zero Audio Retention Architecture

Tsuwalo's infrastructure has no audio recording or disk persistence mechanisms. Ephemeral session state is purged from memory instantly when a call ends. Only minimal operational telemetry (IP addresses, connection timestamps, jitter/packet loss) is stored for connection stability and is automatically deleted after a maximum of 30 days.

Zero Audio Recording Instant Memory Purge 30-Day Diagnostic Log Auto-Purge
πŸ”‘

Unpredictable One-Time Disposable URLs

Call room links are generated with cryptographically secure, high-entropy tokens. This prevents brute-force guessing, unauthorized drop-ins, and URL crawling.

Crypto-Random Tokens Anti-Intrusion Protection Automatic Expiration
🏒

Office SIP & PBX Integration with TLS/SRTP

Connections with corporate PBXs and telecom phone lines protect signaling with TLS and voice media with SRTP. Rigorous authentication and rate limiting shield against unauthorized dialing and spoofing.

SIP over TLS SRTP Toll Fraud Defense
TRANSPARENT SPECIFICATION

Encryption Protocols & Media Protection by Communication Path

Tsuwalo avoids ambiguous claims by transparently disclosing the encryption termination point and data protection scope for each connection mode.

Path / Mode Protocol Encryption End-to-End Encryption (E2EE) Voice Media Protection & Retention
1-on-1 Call
Direct P2P
WebRTC (UDP/ICE) DTLS 1.3 + SRTP
AES-GCM-128 / 256
True E2EE
Keys negotiated device-to-device; servers cannot decrypt
Bypasses central servers; no recording; zero retention
1-on-1 Call
NAT Traversal via TURN
WebRTC over TURN
UDP / TCP / TLS
DTLS 1.3 + SRTP
Blind packet relay
True E2EE
TURN relay forwards opaque SRTP packets without decryption keys
Relay servers forward encrypted payloads without inspection or storage
Group Voice / Rooms
Cloudflare Calls SFU
WebRTC SFU
Edge media routing
DTLS 1.3 + SRTP
Edge-terminated transport encryption
Transport Encrypted*
Terminated at edge memory for fan-out (SFrame planned)
Replicated purely in edge memory. Never recorded or written to disk
Office Extension
SIP / PBX Integration
SIP over TLS
SRTP
TLS 1.3 (Signaling)
SRTP (Audio media)
Hop-by-Hop Encrypted
Terminated at corporate PBX appliance
Governed by customer internal PBX policies. Zero storage on Tsuwalo side
Landline / Mobile Calls
PSTN / VoIP Gateway
VoIP GW / PSTN VoIP segment TLS/SRTP
Telecom carrier standards on PSTN
Segment Encrypted (Non-E2EE)
Transcoded to PCM at telecom gateway
Converted as required for public telephone network. No voice retention
* Technical Note on Group Call (SFU) E2EE:
In multi-party group calls routed through Cloudflare Calls, standard browser WebRTC terminates DTLS-SRTP sessions at the distributed edge node before replicating streams to participants. As such, this is "robust transport encryption" rather than application-layer E2EE (such as WebRTC Insertable Streams / SFrame standards). While Tsuwalo is actively preparing SFrame integration, our zero audio retention architecture strictly prevents audio recording, logging, or disk storage at the SFU edge.
USE CASES

Connect Seamlessly with Anyone, Anywhere, Any Device

Tsuwalo automatically selects the optimal network route based on participant environment and equipment.

For Customers & Friends
🌐

Guest Calls & Instant Web Desks

Zero app-install friction for callers.

Perfect for customer inquiries, rapid consultations, and urgent check-ins where you cannot ask someone to download an app. Send a link, and voice begins instantly.

For Teams & Communities
πŸ‘₯

Crystal-Clear Voice Rooms & Group Calls

Ultra-low latency audio for natural overlapping banter.

Ideal for agile team standups and casual co-working lounges. Direct P2P and edge-optimized routing save mobile bandwidth and extend battery life during long calls.

For Enterprises & Offices
🏒

Office Phone & SIP PBX Integration

Bridge corporate phone numbers with modern smartphones.

Connect smoothly with existing on-premise PBXs and internal phone lines. Make and receive calls from your company number on mobile devices even while working remotely.

MOBILE APP & PBX

Manage Office Extensions & Corporate Lines on Your Smartphone

With the Tsuwalo app, registering your internal PBX account (Asterisk, FreePBX, or cloud PBX providers) turns your smartphone into a full-featured office extension. Answer calls directed to your desk number or dial outbound using your corporate caller ID securely from anywhere.

Direct Extension Routing SIP over TLS SRTP Media Encryption CallKit Background Wake-up

* Call history, SIP credentials, and extension configs are encrypted on-device and never transmitted to external servers.

Tsuwalo Mobile App PBX & Extension Dial Screen
PERFORMANCE & ARCHITECTURE

Measurable Quality Supported by Transparent Architecture

Beyond buzzwords like "high fidelity," Tsuwalo is built upon verified telecommunication engineering and open standards.

Just 2Steps
Effort to Start Calling

Open the link and allow your microphone. Zero account registration, zero password logins, zero app downloads.

< 50ms *1
Average P2P Latency

Direct peer-to-peer transport cuts out unnecessary server hops, delivering conversational audio that feels like sitting in the same room.

OpusWideband *2
Next-Gen Audio Codec

Voice-optimized dynamic bitrate scaling (6kbps to 510kbps). Minimizes audio dropouts even on congested mobile connections.

99.9% *3
Fallback Connection Target

Seamlessly transitions to encrypted TURN relays if restrictive corporate firewalls or symmetric NATs block direct P2P connections.

*1 Measurement Conditions: Typical round-trip time (RTT) observed in 1-on-1 direct P2P sessions over standard fiber broadband and 5G connections. Actual latency varies based on carrier routing, terminal hardware, and signal conditions.

*2 Codec Specification: Full Opus audio standard compliant with IETF RFC 6716. Dynamically scales sampling rate and bitrate from 6kbps to 510kbps based on network bandwidth and packet loss (typically 24–64kbps for HD voice).

*3 Connectivity Architecture: Target reliability based on STUN-first direct session negotiation with automatic, transparent fallback to TLS/DTLS TURN relays when strict firewalls or symmetric NAT topologies are detected.

πŸ’‘ 4 Hybrid Communication Pillars

Tsuwalo automatically determines the best protocol and route based on participant count, device type, and network constraints.

1-on-1 Direct

P2P (Peer-to-Peer Mesh)

Direct DTLS-SRTP encrypted link between devices. Bypasses central servers to deliver sub-50ms latency and true End-to-End encryption.

Firewall Traversal

TURN (Secure Packet Relay)

Activates when direct P2P is blocked by restrictive enterprise firewalls. Relays encrypted SRTP packets transparently without server-side decryption.

Multi-Party Calls

SFU (Cloudflare Calls Edge Routing)

For calls with 3+ participants, streams are routed via Cloudflare Calls global edge network, maintaining a single upstream link per device to protect battery and bandwidth.

Legacy Telecom & PBX

SIP / VoIP Gateway

Bridges modern WebRTC with traditional telephone networks using TLS signaling and SRTP audio. Make and receive calls from office lines on any smartphone.

FREE & SECURE

No App Installation Required.
Test Voice Clarity in Your Browser in Seconds.

No credit cards, no account creation, no spam. Connect instantly from any modern desktop or smartphone browser.