Contact person: Fix android app Audio Bridging from sip to gsm and from gsm to sip

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Location: oran, Algeria

Budget: Recommended by industry experts

Time to start: As soon as possible

Project description:
"Android Default Dialer App – Fix GSM ↔ SIP Audio Bridging (ConnectionService / Conference)

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Background:
We are developing a custom Android Default Dialer app that must handle both SIP and GSM calls.

The intended flow is:
1. A SIP call arrives (INVITE from Asterisk/VoIP provider).
2. The app triggers a GSM call to the B number.
3. Once the GSM call is connected, the app answers the SIP call.
4. Both calls are then merged into a Conference so that audio flows between SIP ↔ GSM (two-way audio bridge).

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What works now:
• SIP registration and incoming SIP calls (using Linphone/PJSIP).
• Outgoing GSM call initiation via [login to view URL]().
• SIP and GSM calls connect separately.
• Call states and control logic are functioning (answer, reject, hang up).

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Problem:
• When we merge SIP and GSM legs, no audio is passed in either direction (SIP ↔ GSM).
• Sometimes we see Cannot merge AUDIO in logs.
• We already tried:
• TinyALSA / PCM hooks
• Native JNI audio hooks
• Java AudioRecord/AudioTrack bridging
• LSPosed/system-level modules
• ConnectionService + [login to view URL]()

None of these approaches resulted in two-way audio bridging on stock Android (non-rooted).

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Goal:
We need an expert Android telecom developer who can:
• Debug the current ConnectionService + Conference implementation.
• Fix the audio path so SIP ↔ GSM works (two-way audio).
• Provide a working proof-of-concept APK (with source code).
• Explain the changes so our in-house team can continue development.

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Environment:
• Device: OnePlus 5T (Android 9) – stock ROM.
• SIP Server: Asterisk (we will provide account & config).
• Test SIM card inserted in device.
• Remote debugging access available via AnyDesk.

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Deliverables:
• Working two-way audio bridge between SIP and GSM inside the app.
• Clean, reproducible code (Java or Kotlin).
• Documentation or notes explaining the solution.

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What we provide:
• Test phone with SIM and SIP account.
• AnyDesk live access for real-time debugging.
• Full logs (adb logcat, Asterisk traces, SIP traces).

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Engagement:
• Paid project (hourly or fixe" (client-provided description)