Webphone (formerly Mizu Webphone): Complete Feature Overview

Comparing Webphone to Mizu Webphone: Improvements & What’s New

Summary

This article compares Webphone (formerly Mizu Webphone) with its predecessor, highlighting functional improvements, architectural changes, user experience updates, security enhancements, and deployment considerations. It’s written assuming a typical enterprise WebRTC/SIP deployment and focuses on practical differences that matter to admins, developers, and end users.

Key improvements at a glance

• Rebranded platform name with clearer product positioning.
• Modernized UI/UX for faster workflows and improved accessibility.
• Enhanced WebRTC and SIP interoperability, reducing call failures and improving media quality.
• Stronger security defaults, including updated encryption and stricter permissions handling.
• Improved extensibility and developer APIs for easier integration into web apps.
• Better performance and scalability, lowering CPU and memory usage in high-concurrency scenarios.

Architecture and core technology

• Webphone retains the original Mizu Webphone architecture (browser-based SIP over WebRTC) but updates core libraries and protocols to current standards. Expect:
  • Upgraded WebRTC stack for better browser compatibility and codec support.
  • Cleaner separation between signaling and media layers, simplifying debugging and extensibility.
  • Optional use of updated TURN/STUN handling to improve NAT traversal.

Call quality and media handling

• Codec support: Webphone adds newer codec negotiation strategies and broader codec fallbacks, improving interoperability with modern SIP endpoints and mobile devices.
• Adaptive jitter buffering and improved packet loss concealment reduce perceived audio glitches on lossy networks.
• Lower-latency audio paths through optimized media pipelines; measurable MOS improvements in medium-to-high packet loss conditions.

Security and privacy

• Stronger defaults: Webphone enables secure signaling (WSS) and encrypted media (DTLS-SRTP) by default, whereas older configurations in Mizu Webphone commonly left some options optional.
• More restrictive browser permission flows and clearer prompts to users about microphone access.
• Hardened CSRF and XSS mitigations in the management/UI layers.
• Audit-friendly logging options with adjustable verbosity; sensitive fields are masked by default.

User interface and user experience

• Redesigned call controls and a simplified layout reduce clicks for common tasks (hold, transfer, conference).
• Improved accessibility: keyboard navigation, ARIA attributes, and better screen-reader support.
• The contact and call history interfaces are more responsive; filtering and search are faster and more flexible.
• Mobile-responsive UI provides near-native experience on phones and tablets.

Developer experience and APIs

• Newer, more consistent JavaScript SDK with typed definitions (TypeScript-friendly).
• Webhooks and event-driven callbacks expanded for richer integration (presence, call-state changes, media events).
• Plugin/extension points for custom UI components and call handling hooks.
• Better documentation and examples for common integration patterns (embedding, single-page apps, progressive web apps).

Deployment and operations

• Simplified container images and Helm charts for Kubernetes; official Docker images are smaller and more modular.
• Improved health checks, metrics (Prometheus), and tracing hooks (OpenTelemetry) for observability.
• Scalability improvements: lower resource usage per session and enhanced load-balancing guidance for large deployments.
• Backwards compatibility modes available to help migrate from Mizu Webphone setups, but administrators should plan for tested rollouts.

Migration considerations

• Configuration mapping: most settings have direct equivalents, but some security defaults changed—review TLS/WSS/DTLS settings when migrating.
• Codec and SIP header behavior: test interoperability with existing PBX/SIP proxies, as negotiation order and header normalization may differ.
• Custom UI integrations: update to the new SDK and test custom hooks or UI components against the updated event model.
• Data and logs: verify log formats and retention settings if you rely on external log parsers.

Known limitations and considerations

• If you depend on legacy browser versions, verify compatibility—Webphone targets modern browsers and may drop older browser support.
• Some third-party SIP endpoints or PBX systems with strict SIP stacks might require header or SDP tweaks.
• Feature parity: most core features are preserved, but niche platform-specific behaviors should be validated during testing.

Practical checklist for switching

1. Backup current Mizu Webphone configurations and logs.
2. Deploy Webphone in staging with production-like SIP backends.
3. Run SIP interoperability tests (audio, DTMF, transfers, hold, conferences).
4. Validate security settings (WSS/DTLS, permission prompts) and update firewall/TURN rules if needed.
5. Migrate custom integrations to the new SDK; run end-to-end tests.
6. Roll out gradually (canary/batch) and monitor metrics and user feedback.
7. Decommission legacy instances once stable.

Conclusion

Webphone builds on Mizu Webphone’s strengths while modernizing the stack, improving security, and offering a better developer and user experience. Administrators should plan a staged migration focused on interoperability testing, security configuration review, and updating custom integrations to the modern SDK.