Cooling signalNo paid features detectedMedium-risk permissions

DialConnection Phone

DialConnection Phone is a WebRTC client which works using WebRTC & SIP protocol

Users624Current public install base

Rating--Store average score

Reviews0Public review volume

Manifest versionV3Extension platform version

7-day growth-23Net users gained this week

7-day growth rate-3.55%Relative weekly velocity

Preview

DialConnection Phone Media preview

2 assets

Trend

User growth trend

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User Growth Over Time

Rating trend

7-day rating trend

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Growth overview

Daily, weekly, and monthly growth

Compare 1-day, 7-day, and 30-day net growth and growth rate.

1-day growthDeclining

-7-1.11%

7-day growthDeclining

-23-3.55%

30-day growthGrowing

+44+7.59%

Technical snapshot

Version, languages, and crawl freshness

Review publication date, version, supported languages, and crawl timestamps.

Version0.3.0

ManifestV3

Size1.74MiB

Languages1English

Published--

Store updated--

Last crawled--

Overview

Product summary

Review the store description, core capabilities, and common use cases.

DialConnection Phone is a WebRTC client that operates using WebRTC & SIP protocols. It serves as a replacement for Soft (VoIP) Phones, making communication more efficient and cost-effective.

Key Features:

Do Not Disturb (DND)

Auto Answer Incoming Calls

Hold/Unhold

Mute/Unmute

Reviews

Recent review snapshot

Inspect the latest comments and rating distribution.

Store average score--Chrome Web Store aggregate rating, including ratings without synced review text

Synced text average--Average computed only from synced review bodies below

Reviews with text0Synced review bodies

Total ratings / reviews0Chrome Web Store public rating/review count

Store average score: --. The bars below are calculated from synced review text only, so they may be empty for extensions that have public ratings but no synced comments yet.

5★

4★

3★

2★

1★

No synced review text is available yet.

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AI product read

WebRTC SIP Client

Provides a softphone solution for making VoIP calls directly from the browser using WebRTC and SIP protocols, replacing traditional desktop softphones.

User installs extension -> Configures SIP server details (username, password, server URL) in options page -> Extension connects to SIP server via WebSocket -> User can make/receive calls through browser UI.

Target users: Businesses using SIP/VoIP services / Remote workers needing a browser-based phone / Users wanting to replace desktop softphones

The extension relies on external SIP servers which may have their own paid plans, but the extension itself is free.
Static analysis heuristic flagged 'upgrade' but it was a false positive from React library code.

Monetization

No paid features detected

No evidence of paid features, subscriptions, or billing logic within the extension code. The 'upgrade' keyword found in static analysis was a false positive from React library code ('react.production.js'). The extension appears to be a free tool for connecting to a user's own SIP server.

Confidence: 90
Payment platform: --
Source: AI / High
Login required: Yes
Reason: The extension is a WebRTC/SIP client that requires the user to input SIP server credentials (username, password, server URL) in the config page (config.html, assets/config-CgDSLsb7.js). The core workflow of making calls depends on authenticating with an external SIP/VoIP provider. The extension itself does not have its own account system but acts as a thin client for a destination service.

Permission risk

Medium-risk permissions

This estimate uses declared permissions and host permissions to assess access scope and sensitivity.

扩展能力较强：storage

Publisher

Independent publisher

Email: [email protected]
Phone: --
Address: --

Public contact fields are shown above. Developer portfolio pages are available on paid accounts.

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Permissions

Host access