Frequently Asked Questions & Technical Reference
Everything you need to know about Dialxa's sub-50ms WebRTC media engine, OPUS HD audio streaming, telephony SIP trunks, browser voice integration, and carrier security.
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Contact Engineering SupportTraditional VoIP services route voice packets through central monolithic media servers with deep jitter buffers (often adding 250ms–400ms of lag). Dialxa utilizes a distributed edge architecture:
- Direct European Routing: User media connects directly to an optimized edge node over low-latency Tier-1 carrier transit, eliminating unnecessary routing hops.
- Zero-Transcoding P2P & SFU: When both parties support OPUS 48 kHz, packets bypass intermediate transcoders completely.
- Adaptive NetEQ Jitter Buffer: Instead of fixed 120ms legacy buffers, our engine uses an adaptive NetEQ jitter buffer that dynamically recalculates delay on a 10ms frame basis, stabilizing streams under 50ms on average.
Dialxa natively supports modern interactive audio standards as defined by the IETF:
OPUS Fullband (48,000 Hz, 20ms): The primary HD standard, delivering studio-grade voice clarity optimized for human ears and speech AI models.G.722 Wideband (16,000 Hz): Supported for enterprise IP-PBX systems and HD conference endpoints.G.711 µ-law / A-law (8,000 Hz): Standard PSTN fallback for legacy telecom lines and mobile network handoffs.
Dialxa incorporates three redundant layers of network resilience:
- In-band Forward Error Correction (FEC): Redundant low-bitrate parity data is attached to subsequent packets, allowing the receiver to reconstruct lost frames without waiting for a retransmission.
- Algorithmic Packet Loss Concealment (PLC): Synthesizes micro-gap acoustic waveforms using algorithmic predictive interpolation to eliminate clicks and robotic stuttering during network drops of up to 30%.
- Dynamic Bitrate Throttling: Seamlessly scales audio bitrate between 16 kbps and 64 kbps based on RTCP feedback reports.
No downloads or installations are needed. Dialxa is a 100% browser-native Web Dialer and WebRTC softphone. Unlike legacy phone software (such as Zoiper, MicroSIP, Cisco Jabber, or bulky desktop apps) that require .exe/.dmg downloads, administrator privileges, and constant manual updates:
- Zero Installation: You simply open your web browser (Chrome, Edge, Safari, Firefox, or Brave), log in, and you are ready to make and receive HD phone calls immediately.
- Zero Disk Clutter: It runs entirely in memory without taking up hundreds of megabytes of storage or background CPU cycles when idle.
- Any Device, Anywhere: Works seamlessly on Windows PCs, MacBooks, Chromebooks, tablets, and Linux workstations with zero IT setup.
- Optional PWA Desktop App: If you prefer a desktop icon on your dock or taskbar, you can click "Install App" directly in your browser address bar to run it as a standalone Progressive Web App (PWA) with zero background clutter.
Yes. Dialxa offers bidirectional SIP Trunking over TLS (RFC 3261). You can authenticate via:
- IP Access Control Lists (ACL): Whitelist your PBX public static IPs for zero-signaling overhead.
- SIP Registration / Digest Auth: Standard username and password credentials.
Once configured, calls arriving at your PBX can be forwarded directly to Dialxa WebRTC browser agents or dispatch workstations via standard SIP URI invitations (e.g. sip:session-id@sip.dialxa.com).
No. Dialxa uses standard native browser WebRTC (W3C standard). It works out-of-the-box across 100% of modern browsers, including:
- Google Chrome (Desktop & Android)
- Apple Safari (macOS & iOS 14.3+)
- Mozilla Firefox & Microsoft Edge
- React Native, Flutter, and native iOS/Android WebRTC binaries
Dialxa connects directly through native WebRTC browser APIs without requiring proprietary plugins or external desktop software. Your web application can initiate sessions and manage audio streams directly via standard JavaScript and our secure REST API.
Yes, Dialxa supports BYOC. You can keep your existing telecom contracts and toll-free numbers. Simply point your SIP trunk to our edge SBC IP addresses for seamless routing into the Dialxa web ecosystem.
Yes, shortcuts can be customized in application settings, but they only function when the application window or tab is active:
- Customizable Key Bindings: In your settings panel under Keyboard Shortcuts (Skróty klawiszowe), you can reassign quick keys for essential call operations — including answering incoming calls (default
Enter), ending or rejecting calls (defaultEscape), and muting/unmuting your microphone (defaultM). Click any assigned key button to record a replacement. - Active Application Focus Only: Shortcuts respond only when the Dialxa application has active focus. They do not operate globally when you are typing or navigating inside other desktop programs or when the browser window is minimized.
- Why this architectural design? Dialxa runs as a zero-installation, web-native VoIP solution using modern browsers and standard WebRTC as its execution core. Because we do not force users to install invasive desktop background software or system-wide hooks, browser security sandboxes intentionally restrict keystroke interception to active windows to protect user privacy and system security.
No. Integration takes less than 60 seconds and requires absolutely zero engineering or API development.
Thanks to the Dialxa Windows Client and our Chrome Extension, your existing platforms transform into voice-enabled communications centers instantly:
- Windows Native Client: Automatically intercepts operating-system
tel:,callto:, andsip:protocol links. Clicking any phone number in desktop apps like Excel, Outlook, SAP, or legacy on-premise TMS instantly initiates a call via Dialxa. - Universal Chrome Extension: Scans active web pages (Salesforce, HubSpot, Zendesk, or custom web CRM portals) and converts raw phone numbers into clickable WebRTC launchpads without altering your host system's code.
Dialxa uses transparent, predictable per-user licensing billed on a monthly basis — without arbitrary minute counting or hidden fees. With volume discounts, our most cost-effective tiers start from:
- Dialxa Solo (from €6 / user / mo): Essential VoIP core for standard communication workflows, including basic admin panel, account creation, and number management (€8 / user / mo standard, down to €6 / user / mo with volume discount).
- Dialxa Velocity (from €11 / user / mo): Most popular tier for dynamic teams, featuring advanced routing, intelligent call groups, call-back panel, and detailed analytics (from €11 to €13 / user / mo).
- Dialxa Pro (from €18 / user / mo): Maximum enterprise performance with availability-based routing, SLA reporting & monitoring, and priority support (from €18 to €20 / user / mo).
Automatic volume discounts scale at 100, 150, and 200 users. Every plan includes a 7-day free trial with no credit card required. Visit our Pricing Page to test the live calculator across EUR, USD, and PLN.
Our core engine is built on ultra-efficient C/C++ architectures (the same tech powering major telcos). A single standard deployment comfortably handles hundreds of concurrent encrypted WebRTC sessions with zero degradation. For extreme volume, we offer dedicated edge instances.
No. There are zero setup fees, zero charges for adding SIP credentials, and zero egress bandwidth surcharges. You only pay for your predictable per-user monthly software license as detailed on our pricing page.
All audio streams are protected by mandatory cryptographic protocols:
- DTLS 1.3: Datagram Transport Layer Security enforces mutual authentication and ephemeral key exchanges.
- SRTP with AES-GCM 256: Audio payload packets are encrypted from the client directly to our edge core, with zero plain-text fallback permitted.
- Signaling Security: HTTPS and WSS (WebSockets Secure) utilizing TLS 1.3 with Perfect Forward Secrecy (PFS).
Yes. Dialxa is fully compliant with EU GDPR (RODO) regulations:
- Zero Data Retention by Default: Real-time audio streams are routed in volatile memory and never persisted to disk unless you explicitly configure encrypted cloud recording buckets.
- Data Processing Agreements (DPA): We execute standard EU Model Clauses (SCCs) for enterprise clients.
Dialxa backs its enterprise services with a contractual >99.5% monthly uptime SLA. Our infrastructure utilizes redundant upstream carrier transits, automated health monitoring, and fast route failover to protect core voice network availability.
We believe developers should speak directly with infrastructure engineers:
- Free & Developer: Comprehensive documentation and direct email support (support@dialxa.com) with <12h turnaround.
- Enterprise: Dedicated Slack/Teams connect channel, direct access to the founding engineering team, and priority incident response (P1) during European business hours.
Still have technical questions?
Our voice infrastructure engineers are available to review your architecture, run SIP packet captures, or provide custom volume quotes.