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I want custom firmware written for an ESP32-S3 SIP module ## Project Overview We are building a **hardware–software audio gateway** that converts a **live cellular call audio from an Android phone** into a **SIP call**, and back again, in real time. ### High-level flow: **Android phone , SIM call → USB OTG → **ESP32-S3 (USB Audio Class 1 device)** → **On-device SIP + Opus client** → SIP server / VoIP endpoint …and **two-way audio must work reliably** for extended calls. This is **not an Android app project**. This is an **embedded firmware + SIP/audio pipeline project**. ## Project Goals (Non-Negotiable) ✔ ESP32-S3 enumerates as a **USB headset (UAC1)** on Android ✔ Android routes **SIM call audio** to USB ✔ ESP32-S3 runs a **SIP client with Opus codec** ✔ Audio flows both ways: * SIM call → SIP * SIP → SIM call ✔ Stable audio for **30+ minutes without drift or dropouts** ## Hardware & Environment ### ESP32 Side * **ESP32-S3 with 8MB PSRAM** * ESP-IDF (latest stable) * TinyUSB for UAC1 * Wi-Fi for SIP/RTP ### Android Side * **Redmi 4A** * Android 6–7 range * USB OTG enabled * SIM call active (GSM/VoLTE) ## Project Scope (3 Mandatory Parts) # PART 1 — USB Audio Class 1 (UAC1) ### Objective ESP32-S3 must act as a **USB Audio Class 1 headset** when connected to Android via OTG. ### Requirements * ESP32 acts as **USB Device** * UAC1 compliant (not UAC2) * Mono PCM audio * 8 kHz or 16 kHz (configurable) * Isochronous IN + OUT endpoints * Ring buffers in PSRAM * No audio glitches under continuous load ### Acceptance Criteria Android detects device as **USB headset** SIM call audio routes to USB automatically or via standard Android routing USB loopback test works (speaker → mic) Audio plays continuously for **10+ minutes** # PART 2 — SIP Client on ESP32-S3 ### Objective ESP32 registers to a SIP server and handles VoIP calls. ### Requirements * SIP REGISTER / INVITE / ANSWER / BYE * Digest authentication * SDP negotiation * RTP transport * **Opus codec mandatory** * UDP SIP (TCP optional) * Configurable SIP credentials * Clean reconnection logic ### Acceptance Criteria ESP32 registers and stays registered Can place outgoing SIP calls Can receive incoming SIP calls Two-way audio works with a SIP softphone Stable for **10+ minutes** # PART 3 — Audio Transport Bridge ### Objective Real-time bridge between USB PCM audio and SIP RTP audio. ### Required Pipeline USB Speaker PCM → Opus Encode → RTP Send RTP Receive → Opus Decode → USB Mic PCM ```### Architecture Requirements * Separate FreeRTOS tasks: * USB RX * RTP TX * RTP RX * USB TX * Ring buffers + jitter buffer * Drift control (drop/insert or simple resampling) * Underrun protection (never stall USB) * Configurable latency buffers ### Acceptance Criteria SIM call audio heard on SIP endpoint SIP endpoint audio heard in SIM call No progressive drift over **30+ minutes** End-to-end latency ≤ 300 ms (reasonable target) ## Deliverables 1. **Complete ESP-IDF firmware project** * Clean build * PSRAM enabled 2. **Source code (no binaries only)** 3. **Build & flash instructions** 4. **Configuration documentation** 5. **Audio test & validation notes** 6. **Runtime logs/screenshots proving success** ## Required Repository Structure firmware/ ├── components/ │ ├── uac1_device/ │ ├── sip_client/ │ ├── rtp_opus/ │ ├── audio_bridge/ ├── main/ docs/ ├── [login to view URL] ├── [login to view URL] 1 — UAC1 Working * USB headset recognized * Loopback audio works **30%** ### Milestone 2 — SIP Client Working * SIP calls with Opus audio **30%** ### Milestone 3 — Full Gateway Bridge * SIM call ↔ SIP call fully bridged * 30+ minute stability test ## Final Acceptance Test (Mandatory) ✔ Place real SIM call on Redmi 4A ✔ Audio passes through ESP32 to SIP endpoint ✔ SIP audio returns to phone call ✔ Call lasts **30+ minutes without audio failure** No final payment without this test passing. Ideal Freelancer Profile * Strong **ESP32 / ESP-IDF** experience * USB Audio Class (UAC) implementation experience * SIP / RTP / Opus knowledge * Real-time audio processing background * Comfortable debugging timing & buffer issues When Applying, Include: 1. Relevant ESP32 + audio projects 2. USB Audio or SIP experience 3. Which SIP stack you plan to use 4. Estimated timeline
Proje No: 40068148
13 teklifler
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13 freelancer bu proje için ortalama $133 USD teklif veriyor

As a seasoned professional with extensive experience in embedded systems and audio processing, I assure you that I am perfectly equipped to fulfill your project requirements for the ESP32-S3 SIP firmware. My advanced proficiency in Python, C, and C++ makes me well-suited to translate your specifications into robust, efficient firmware. Moreover, my expertise with ESP-IDF is an invaluable asset for this project. To further convince you of my suitability, my track record includes developing and troubleshooting audio systems for various applications, including signal processing and audio engineering. I have tackled numerous projects where maintaining audio integrity was paramount over extended periods. This demonstrates my ability to deliver the stable audio performance your project necessitates. Another crucial advantage that sets me apart is my rich understanding of USB and networking protocols like SIP and RTP. Your project's need for a seamless bridge between USB audio and SIP calls aligns directly with my core competencies. With my attention to detail and passion for delivering highly reliable solutions, I guarantee a firmware package that not only meets the outlined specifications but also performs optimally in real-world scenarios. I look forward to being entrusted with your project!
$220 USD 7 gün içinde
3,2
3,2

Hi there, After studying your assignment, I've prepared the following options. I look forward to your feedback. Option A: Develop a custom firmware for an ESP32-S3 SIP module to enable real-time audio conversion between a live cellular call and SIP, ensuring stable two-way audio for extended calls. The project involves implementing a USB Audio Class 1 headset, SIP client with Opus codec, and an audio transport bridge. Option B: Custom firmware development for ESP32-S3 SIP module to facilitate seamless audio conversion between live cellular calls and SIP, guaranteeing reliable two-way audio for extended calls. The project encompasses creating a USB Audio Class 1 headset, SIP client with Opus codec, and a real-time audio transport bridge. Let's discuss in detail through chat.
$30 USD 3 gün içinde
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Hello Michael M., I am Maryam Abbas, a Software Development professional with over 4 years of experience. I have carefully reviewed the requirements for the ESP32-S3 SIP Firmware project. My approach involves implementing USB Audio Class 1 functionality, developing a SIP client with Opus codec support, and creating an audio transport bridge for seamless communication. With a proven track record of successful project deliveries and positive client feedback, I am confident in providing a reliable solution for this project. You can explore my portfolio at https://www.freelancer.pk/u/maryam951 Let's discuss further details in chat. Best regards, Maryam Abbas
$30 USD 2 gün içinde
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My name is "Usama Safdar" and I am a Ph.D degree holder which means I am highly-capable to tackle this project "Content Editor " with 100 percent accuracy. I am a professional writer with over 6 years of experience in writing; Essays, Research Summaries, Thesis, Dissertation, Lab Reports and Case Studies. I always provide High-Quality Solutions within the shortest possible time with all instructions followed against very reasonable prices. I can manage works even with shortest deadlines like; "2500 words work in just 6 Hours" with very reasonable time. As a pro academic writer I am also familiar with all the referencing styles; such as APA, Harvard, OSCOLA, IEE, MLA etc. I always provide plagiarism-free solutions and as a prove I also provide "FREE Turnitin reports". For Samples, please visit my profile https://www.freelancer.com/u/SolutionMart Please message me to start the discussion. Thank You
$30 USD 1 gün içinde
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My name is Rabia Faisal, I am working in the writing industry since 2011. During this time, I have served countless clients with a full amount of satisfaction by providing them with TOP Quality Solutions. I have command of all references APA, Harvard, IEEE, MLA & Chicago, etc. I will provide plagiarism-free work with 100 percent accurate grammar within your given deadline. Please message me to get Top Class Services. I am waiting; https://www.freelancer.com/u/TopWritingGuru
$30 USD 1 gün içinde
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With my broad spectrum of skills in Software Development, Electronic Engineering, and my considerable expertise in ESP32/ESP-IDF ecosystem, I am ideally positioned to tackle your project head-on. I understand the core goals of your project: implementing an efficient USB Audio Class 1 driver on the ESP32-S3, designing and integrating a stable SIP client with Opus codec, and building a robust Audio Transport Bridge between USB PCM audio and SIP RTP audio. Having built high-performance game systems that require stable audio interfaces with minimal glitches, I am familiar with the challenges we can potentially face. My experience entails developing clean codebases with comprehensive configuration notes and runtime logs for solid documentation. This ensures that your customized firmware solution would be well-documented and your team can easily maintain, scale, or debug it in due course. Additionally, my thorough mastery of ESP-IDF, TinyUSB, Wi-Fi protocols coupled with my exposure to buffering techniques including ring buffers and jitter buffers makes me aptly equipped for architecting your specific solution. And when it comes to performance benchmarking and validating system stability over long durations– two elements absolutely critical for this project– rest assured of my robust and disciplined approach to run extensive real-world tests aligned with industry standards.
$120 USD 4 gün içinde
0,0
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I can help you design and implement the ESP32-S3-based audio gateway for converting SIM call audio to a SIP call with real-time audio transfer, using Opus codec and USB Audio Class 1 (UAC1) for integration with Android. I have extensive experience in ESP32/ESP-IDF, USB audio implementation, and SIP/RTP protocols. Approach: USB Audio Class 1 (UAC1): ESP32-S3 as USB device: UAC1 (mono PCM, 8kHz/16kHz), Isochronous IN/OUT, no audio glitches. Loopback test: Speaker → Mic for validation on Android. SIP Client Implementation: SIP functions: REGISTER, INVITE, ANSWER, BYE with Opus codec for RTP transport. Reconnection logic and UDP SIP with digest authentication. Audio Transport Bridge: Real-time USB → SIP RTP bridge: Use separate FreeRTOS tasks for handling USB RX, RTP TX/RX, and USB TX. Implement buffering, jitter management, and latency control to ensure <300ms latency and 30+ minute stability. Deliverables: Complete ESP-IDF firmware with components for UAC1 device, SIP client, RTP/Opus audio handling, and bridge logic. Source code with build and flash instructions. Test plan and validation documentation, proving successful SIP call bridging. I plan to use a custom SIP stack (e.g., pjsip) for flexibility and optimal performance. Let me know if this timeline and approach align with your needs! we can discuss more details in chat thanks
$140 USD 2 gün içinde
0,0
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I saw your project and am confident I can deliver on this. I'm currently working on a similar project and understand the importance of stable two-way audio for extended calls. Ensuring the ESP32-S3 acts as a USB headset on Android, runs a SIP client with Opus codec, and maintains audio reliability aligns with my expertise, guaranteeing the successful completion of the required benefits. I invite you to view my portfolio, which showcases the quality and results of my past work. I look forward to hearing from you. Regards, Travis
$120 USD 7 gün içinde
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Hello, I’m an embedded systems engineer with strong experience in ESP32 development, USB Audio Class (UAC1), SIP clients, and real-time audio processing. I can develop the firmware for your audio gateway, enabling two-way SIP audio over USB from a SIM call to SIP call with Opus codec. For this project, I will deliver: USB Audio Class 1 (UAC1) implementation for continuous, glitch-free audio transmission SIP client handling all core protocols (REGISTER, INVITE, etc.) with Opus codec support Audio transport bridge: converting PCM audio to Opus and back, with low-latency and drift control I have experience in similar projects with USB audio, SIP, and Opus integration, ensuring 30+ minutes stability for continuous calls. Best regards,
$140 USD 2 gün içinde
4,1
4,1

Santo Domingo, Dominican Republic
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