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VoIPCall is a developer proof of how a native Android app can place and receive regular phone-network (PSTN) calls through Vobiz. Its call path combines an Android app, a SIP endpoint assigned to the device, a Vobiz number and backend, and Firebase push notifications for incoming calls when the app is in the background. It is not presented as a replacement for Android’s default dialer, and the available project description does not establish a public Google Play release.
What VoIPCall does—and what it does not
The project gives an Android app a keypad and uses SIP registration to connect that device to Vobiz. Outbound calls use the device’s Vobiz endpoint and configured caller ID. Incoming calls are routed to the registered endpoint; when the app is not in the foreground, a backend and Firebase push path are used to wake it and handle the incoming-call experience.
The app does not register itself as Android’s default phone app. Another app can open it with Android’s ACTION_DIAL intent, or a caller can use the app’s own call action. The walkthrough also describes a call-history content provider protected by a permission, with a URI through which authorized code can read call records. These integrations do not make VoIPCall the system dialer.
The Tool Desk
Outbyte PC Repair FREERepair Windows errors before they cause bigger problemsFix Now →Outbyte Driver Updater FREEFix the driver behind crashes, sound loss and screen glitchesFind Drivers →The project is described in a developer walkthrough by Niladri Prasad Padhy. That source describes an implementation and setup, not an independently validated product release, benchmark, or guarantee that it works on every Android device.
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- 20 SIP lines, hotspot
- 7-inch color touch screen with 112 one-touch DSS keys
- External Fanvil USB camera (optional)
- Built-in 2.4G/5G WiFi & Bluetooth
- Support EHS headset
Components in the call path
- Android app: keypad, SIP registration, call controls, and status indicators for SIP registration and backend/Firebase health.
- Per-device SIP endpoint: the SIP username and password used by that handset to register with the Vobiz registrar.
- Vobiz number and caller ID: a DID (direct inward dialing number) for inbound routing and a caller ID in E.164 international format for outbound calls.
- Backend: receives voice-application webhook traffic from Vobiz and supports the incoming-call flow.
- Firebase: supplies push notifications used to wake the app for incoming calls when it is backgrounded.
- Network access: a public HTTPS address for the backend during development, typically exposed through a tunnel; production media may also need TURN relay service for networks that STUN alone cannot traverse.
Requirements and setup values
The walkthrough lists these prerequisites for trying its implementation; they are not a claim of independent testing:
- JDK 17 and Android SDK 36.
- A physical Android 12 or later phone for testing.
- Node.js 22 or later and a tunnel that exposes the backend over HTTPS.
- A Vobiz number and a SIP endpoint reserved for the device.
- A Firebase project configured for the Android app’s package identifier.
Configuration separates credentials intended for the handset from secrets that should remain server-side. The app settings include the SIP username and password, registrar WSS URL wss://registrar.vobiz.ai:5063/, SIP domain registrar.vobiz.ai, public HTTPS backend URL, device token, Vobiz caller ID, and a default country for cases where the device has no SIM. The app walkthrough says it encrypts SIP credentials with an Android Keystore AES-GCM key. Vobiz Auth ID/Auth Token and webhook secrets belong on the backend, not in the mobile app.
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Permissions listed for the implementation include microphone, notifications, contacts, phone state, and nearby devices/Bluetooth. The exact permission prompts and behavior can vary with Android version and app configuration.
Build, install, and verify the app
The walkthrough supplies Gradle and ADB commands for building and installing the project. Because the actual command lines and project-specific package details are not established here, use the commands in the project’s own walkthrough rather than substituting guessed paths or identifiers.
Rank #3
- 16 lines with up to 16 SIP accounts
- Built-in 2 mega-pixel CMOS camera for video calling with privacy shutter
- Runs on the Android 11 operating system
- Built-in Bluetooth 5.0 for syncing with mobile devices and connecting Bluetooth headsets
- Dual-switched auto-sensing 10/100/1000Mbps network ports
- Configure the SIP account, registrar, domain, caller ID, backend URL, Firebase project, and device token for the handset.
- Build the Android app using the project’s Gradle instructions and install it on the physical test phone with the documented ADB command.
- Open the app and confirm that SIP registration and backend/Firebase health are reported as available before placing calls.
- Test an outbound call from the keypad, then test an inbound call while the app is open.
- Repeat the inbound test with the app backgrounded, and confirm that the notification and incoming-call screen behave as intended.
- Verify that the DID routes to the handset configured to receive it, particularly when more than one device is involved.
How an endpoint registers and why each phone needs its own
Each handset registers to registrar.vobiz.ai over the configured WSS registrar URL using its SIP username and password. The Vobiz endpoint associated with that device is then the target for calls routed to it. The app’s SIP status indicator is the first useful check: if it remains disconnected, verify the username, password, registrar URL, and SIP domain.
Assign a separate SIP endpoint to each phone. Sharing one endpoint across multiple handsets can create registration conflicts, because devices compete to register under the same identity; the source warns that this can disrupt which device receives calls. Pair the intended device’s endpoint with its DID routing rather than treating one shared login as a multi-phone setup.
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Background incoming calls, Android limits, and network failures
Firebase and notification delivery
For an incoming call while the app is not in front, the described flow relies on Vobiz routing, the backend webhook, Firebase push handling, and Android’s notification/full-screen incoming-call configuration. A call may fail to ring if any link is misconfigured. Check that the public tunnel URL is current, the backend is reachable, the Firebase service-account key loaded successfully, and the app has its required notification and background-call configuration.
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Android battery management can restrict background work and delay or prevent the app from responding promptly to a push. The walkthrough advises unrestricted battery settings for reliable inbound behavior. The precise menu names vary by manufacturer and Android version; find the app’s battery-use setting in system settings and remove restrictions if background calls are not arriving.
Best Value
- 16 lines with up to 16 SIP accounts
- Built-in 2 MP CMOS camera for video calling with privacy shutter
- Runs on Android 11 OS
- Built-in Bluetooth 5.0
- Integrated dual- band Wi-Fi 5 (2.4GHz & 5GHz)
NAT traversal and media
Registration and call signaling do not guarantee that audio media can traverse every network. The walkthrough cautions that STUN alone may fail on restrictive NATs and says production deployments should add TURN for media relay. If a call connects but audio is missing or one-way across certain networks, investigate media traversal and TURN configuration rather than assuming the SIP endpoint is disconnected.
Quick failure checks
- SIP stays disconnected: recheck the SIP username/password, registrar WSS URL, and domain.
- Backend reports offline: verify that the tunnel is running and that the configured public HTTPS URL still points to it.
- Background call does not ring: inspect Firebase credentials and delivery, notification/full-screen configuration, and battery restrictions.
- Wrong handset rings or registration drops: use a unique SIP endpoint per device and confirm the DID is assigned to the intended handset.
- Call connects without usable audio: assess NAT traversal and TURN relay support for the affected networks.
VoIPCall’s SIP approach versus Vobiz WebRTC
VoIPCall’s described Android proof is SIP-based and centers on a native dialer-style interface. Vobiz’s separate WebRTC offering is an SDK approach for embedding voice calling in browser or mobile products. Vobiz says its WebRTC infrastructure includes STUN/TURN, codec translation between Opus and G.711, and PSTN bridging; those vendor capabilities should not be mistaken for features of this particular SIP app.
| Question | VoIPCall Android proof | Vobiz WebRTC offering |
|---|---|---|
| Implementation model | Device-specific SIP endpoint registered with Vobiz. | Vobiz WebRTC SDK for browser and mobile apps. |
| Where calls fit | Native Android keypad and call flow; callable by other apps through ACTION_DIAL, but not the system default dialer. |
Voice embedded inside a browser or mobile product. |
| Network/media responsibility | The walkthrough warns that STUN alone may not traverse restrictive NAT and identifies TURN as a production need. | Vobiz says its managed WebRTC infrastructure includes STUN/TURN and codec translation. |
| Published price reference | Not stated for the VoIPCall project; carrier voice and number charges depend on the Vobiz services used. | Vobiz’s pricing page displayed US WebRTC at $0.0034 per minute when accessed on 2026-10-07; check the page for current rates. |
Neither model is universally preferable: the SIP proof is relevant when the desired interface is a native Android calling app with device-level SIP registration, while WebRTC is the more direct fit when voice is to live inside a browser or mobile product. Vobiz’s WebRTC page describes the SDK and infrastructure; its pricing page is the source for current rates.
What Vobiz telephony may cost
Vobiz’s pricing page, accessed on 2026-10-07, listed these US rates. They are vendor-published prices, not independent estimates, and can change:
| US service listed | Vendor-listed rate |
|---|---|
| WebRTC | $0.0034 per minute |
| Local outbound voice | $0.0115 per minute |
| Local inbound voice | $0.0055 per minute |
| Local number rental | $1.00 per month, plus a one-time new-number activation fee equal to one month’s rental |
These are US-specific published figures. They do not establish the total cost of operating VoIPCall, which depends on the selected service, call usage, number, and any additional infrastructure. Confirm live rates on Vobiz’s pricing page.
Quick Recap
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