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We have an Android mobile dialer application that is set as the default phone app. The app supports: GSM/SIM-based calling SIP/VoIP calling We need to implement a feature that allows conference calling between: One GSM call One SIP call Requirement When a GSM call is active and a SIP call is active, the user should be able to merge them into a single conference call so that all participants can talk together. Technical Expectations Experience with Android Telecom framework Experience working with: Connection Service In Call Service SIP stack (PJSIP or Android SIP API) Handling audio routing between GSM and SIP Managing call states and audio focus Proper call merge / conference implementation Clean and stable solution Important Notes The app is already functioning for individual GSM and SIP calls. We need proper audio bridging between GSM and SIP. Should work on modern Android versions (Android 11+ preferred). Deliverables Working conference implementation Source code integration Documentation of implementation Testing support Please share: Similar projects you have worked on Your experience with Android Telecom API Estimated timeline and cost
ID Projek: 40227080
26 cadangan
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Aktif 19 hari yang lalu
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26 pekerja bebas membida secara purata $24 USD/jam untuk pekerjaan ini

Hello, This requirement sits at the intersection of Android Telecom integration and real-time audio engineering, and it needs to be handled carefully to ensure stability and low latency. Since your app is already functioning as the default dialer with working GSM and SIP calls, the primary challenge is implementing a proper Conference object within the Telecom framework while building a reliable audio bridge between the GSM and SIP call streams. My approach would involve: • Extending your existing ConnectionService to create a custom Conference instance when both calls are active • Managing call state synchronization through Telecom APIs to ensure proper merge behavior • Implementing controlled audio routing and focus handling • Designing an audio bridge layer to mix GSM and SIP streams with minimal latency and echo issues • Ensuring compatibility with Android 11+ permission and audio restrictions • Thorough testing across speaker, Bluetooth, and wired headsets The critical part is stable bidirectional audio mixing without glitches or echo, so this will require careful handling of audio sessions and possibly native-level optimization depending on your current SIP implementation. Estimated timeline: 3 to 5 weeks depending on current architecture. If you can share your current SIP stack details (PJSIP or Android SIP API) and confirm how GSM audio is currently accessed within your app, I can provide a more precise estimate. Best, Jenifer
$20 USD dalam 40 hari
9.3
9.3

Hello, We understand your need to seamlessly merge active GSM and SIP calls into a single conference within your Android dialer. Our expertise in the Android Telecom framework, Connection/In Call Services, and SIP stack enables us to implement robust audio bridging and call state management for a clean, stable solution on modern Android. I’m Waqas from Eclairios, a professional software engineer with over 7 years of experience in app and web development. I have successfully completed 128 projects, earning a 5.0 rating from satisfied clients. I specialize in mobile apps (Android, iOS, Flutter), website development, custom APIs, and backend solutions. My goal is to deliver high-quality, scalable solutions that meet your business needs. Why hire me? ★ 100+ Projects Completed with 5-star rating. ★ 3 months of free post-launch support ★ Expertise in advanced technologies and systems Let’s connect and discuss how I can help you with your project. Best regards, Waqas
$18 USD dalam 40 hari
8.3
8.3

Hello We are Ominfowave Software Services Pvt. Ltd., and we have an expert mobile application development team with 10+ years of experience in building reliable, high-quality apps. ? Our Core Expertise •Android App Development (Kotlin / Native) •iOS App Development •Flutter & FlutterFlow (Cross-platform) Our team understands not just development, but also performance, user experience, scalability, and timely delivery. ? What We Can Deliver ✔ Social media integrations (Facebook, Google, Twitter, etc.) ✔ Google APIs (Maps, Places, FCM, YouTube, Analytics) ✔ Offline & cloud database solutions (SQLite, MySQL, Realtime DB) ✔ Secure payment integrations (Stripe, PayPal, Braintree) ✔ Ad monetization (AdMob, Appodeal, MobVista) ✔ In-App Purchases ✔ Location tracking & real-time features ✔ Chat & messaging applications ? App Types We’ve Built •Social Media Apps •Shopping & eCommerce Apps •Job Portals •Taxi / Carpooling Apps •VPN & Utility Apps •Travel & Booking Apps •Live Streaming Apps •AI & Machine Learning–based Apps ? Why Work With Us •10+ years of hands-on experience •Clean, scalable & secure code • Clear communication & regular updates • On-time delivery with quality assurance • Long-term support after project completion We would love to work with you and build a long-term, trusted business relationship. Let’s discuss your idea and turn it into a successful application Warm regards, Vishal Nasit Ominfowave Software Services Pvt. Ltd.
$18 USD dalam 40 hari
8.4
8.4

Hi there, I’ve reviewed your Android app requirements and would love to collaborate on your project. With 5+ years of experience in native Android development, I specialize in building high-performance, user-friendly apps with clean UI, optimized architecture (MVVM/MVI), and seamless API integration. I’ll start with a clear project roadmap, provide regular progress updates, and ensure the app is thoroughly tested for stability and performance before launch. Let’s connect to discuss your app idea in detail — I’m ready to bring your vision to life! Best, Bhargav Android Developer | Kotlin & Java Expert
$15 USD dalam 40 hari
6.9
6.9

Hi there, Your project to seamlessly merge GSM and SIP calls into a single FreeSWITCH conference room is clear and technically exciting. I have extensive experience with FreeSWITCH and VoIP infrastructure, ensuring seamless bridging and media handling. - I will configure FreeSWITCH as the bridge and mixer with precise dial-plan and conference profiles. - Implement Lua/ESL scripts for automatic recording and dynamic mute/unmute control via API or console. - Provide a step-by-step configuration guide with dial-plan snippets and mod_conference settings. - Conduct a thorough validation session covering call bridging, audio verification, recording control, and mute toggling. - Offer security enhancement advice, including voice encryption for future-proofing. **Skills:** ✅ FreeSWITCH conference configuration & dial-plan customization ✅ Lua and Event Socket Library scripting for call control ✅ VoIP integration with GSM gateways and SIP trunks ✅ SIP signaling and media flow optimization ✅ Linux server management and root-level FreeSWITCH tuning **Certificates:** ✅ Microsoft AE Certified: MCSA | MCSE | MCT ✅ cPanel AE & WHM Certified CWSA-2 Ready to deliver a secure, smooth conferencing experience on FreeSWITCH promptly. What specific FreeSWITCH version are you using, and do you have any existing custom dial-plan or script modifications in place? Best regards,
$30 USD dalam 20 hari
6.6
6.6

⭐⭐⭐⭐⭐ Valuable Client, we at CnELIndia, led by Raman Ladhani, can ensure your FreeSWITCH setup seamlessly bridges GSM and SIP into a single live conference. Our approach includes creating a tailored conference profile with mod_conference, configuring dialplan snippets for automatic joining, and scripting Lua/ESL routines to trigger recording and enable mute/unmute via API or console. We will provide step-by-step configuration documentation, perform a live validation session to confirm two-way audio, recording, and mute functionality, and suggest optional voice encryption for future security. With our hands-on experience in GSM-SIP integration, Linux, and VoIP systems, we guarantee a smooth, imperceptible user experience where both callers perceive a single seamless conversation. All deliverables will be fully tested and production-ready.
$20 USD dalam 40 hari
6.6
6.6

I have hands‑on experience with Android Telecom, ConnectionService/InCallService, and SIP stacks like PJSIP. I can deliver a clean GSM–SIP conference merge with proper audio bridging, documentation, and testing support within a stable timeline
$20 USD dalam 40 hari
5.4
5.4

With over 9 years of experience as a Full Stack developer, I offer a unique blend of technical expertise and creativity that makes me highly qualified for your project. My area of specialty includes Android development, and I've developed an array of applications that rely on GSM gateways and VoIP/SIP technologies. I’m also well-versed in working with tools such as FreeSWITCH for conferencing functionalities. This gives me an edge in understanding your project's requirements and implementing them seamlessly. Additionally, my proficiency in languages like Java, PHP, Python and understanding of database management systems ensures a robust architecture in every aspect of the software. Apart from the technical skills, I prioritize timely delivery, effective cost, and extended support to ensure my clients' satisfaction. The deliverables you mentioned can be easily accomplished through my experience leveraging dial-plan snippets, conference profiles, mod_conference settings in FreeSWITCH, as well as Lua/ESL scripts for trigger recordings and participant muting. To further fortify your system against potential threats, I'll also provide brief notes on optional security add-ons like voice encryption. Overall, I promise to make your vision a reality while ensuring quality and growth-friendly solutions.
$120 USD dalam 40 hari
5.2
5.2

Yes, we’ve successfully completed the same projects before and are confident we can deliver this one with the highest quality and within your timeline. We specialize in this type of work and would love to discuss your requirements in detail to ensure a smooth and timely delivery.
$30 USD dalam 40 hari
5.5
5.5

With my extensive 13+ years of experience in Android development and ability to navigate the complexities of the Android Telecom framework, I believe I am the perfect fit for your project. Moreover, having worked on numerous projects involving Connection Service, In Call Service, and configuring mobile dialers, I understand the intricacies of keeping GSM and SIP calls separate and merging them seamlessly into a conference call setting which is reliable and stable. One such example includes developing an enterprise-level healthcare application with VoIP capabilities that allowed clinicians to conduct conference calls using multiple lines. In terms of technical proficiency, I am well-versed with both PJSIP and Android SIP API, which aligns with your project requirement of proper audio bridging between GSM and SIP. Additionally, my experience in handling call states, audio focus, and managing audio routing will enable me to create an optimal solution for your use case. I have successfully developed apps across various Android versions, including Android 11+, and this expertise ensures your app's functionality even with the latest updates. Lastly, as a developer who prioritizes clean coding practices while eliminating technical debts, you can expect not just a working conference feature but also reliable source code integration along with well-documented implementation and testing support.
$20 USD dalam 40 hari
4.8
4.8

Since the early days of my career, I have been immersed in the world of communications technology and web development, including over 20 years of experience. Specifically, I have worked extensively with Android operating systems, which gives me a unique and valuable perspective for this project. My expertise navigating the intricacies of the Android platform allows me to seamlessly bridge disparate systems - such as GSM and SIP conferencing - which is exactly what you require. Moreover, my past projects have had similar components to your job description; merging different communication systems and ensuring smooth integration for a seamless user experience. Your requirements also include call recording and muting functionalities which I am fully versed in thanks to previous instances enriching my skillset. To ensure that my work is up to standards and in line with your expectations, I propose following a meticulous step-by-step configuration process, that includes dial-plan snippets and mod_conference settings as you outlined in your project requirements. Finally, I appreciate your suggestion about incorporating optional security add-ons like voice encryption into the deliverables for future-proofing the system extending its competency beyond current needs if necessary.
$20 USD dalam 40 hari
2.2
2.2

I’ve reviewed your requirement and understand that you need a true audio-level conference bridge between one active GSM call and one active SIP call inside your custom default dialer app, using the Android Telecom framework — not just UI grouping, but proper two-way audio mixing and stable call state management on Android 11+. With strong experience in Android Telecom (ConnectionService & InCallService) and VoIP integrations using SIP stacks such as PJSIP, I specialize in building carrier-grade call flows including call hold, swap, merge, audio focus control, and custom conference implementations inside default dialer environments. I’m ready to begin immediately and can prioritize stability and clean architecture. Best Regards, Deepak
$20 USD dalam 40 hari
1.2
1.2

Hello, I have carefully reviewed your project requirements, and as per my understanding, you are looking to implement a conference calling feature in your Android dialer app that merges one GSM call and one SIP/VoIP call into a single active call. The goal is to enable seamless communication between all participants while maintaining stable audio and proper call state management. My approach will focus on delivering a robust and production-ready solution by: Utilizing the Android Telecom framework, including ConnectionService and InCallService, to manage call lifecycles and UI integration. Bridging audio streams between GSM and SIP, ensuring proper audio routing, focus management, and echo prevention. Integrating with your existing SIP stack (PJSIP or Android SIP API) to handle conference merging reliably. Handling edge cases like call hold, call drop, and participant state changes to maintain a clean and stable conference session. Providing full documentation and source code integration guidance so your team can maintain and extend the feature. I have previous experience working on Android dialers with GSM/SIP integration, including multi-line call handling and conference merging, and am familiar with the complexities of audio bridging and telecom API behavior on Android 11+. Let’s connect to review your current architecture so we can define the conference logic, finalize the audio bridging approach, and set clear milestones for delivery. Best regards, Prateek
$15 USD dalam 40 hari
1.0
1.0

Hello There, I've more than 10+ years of experience and ready to work on lower rate as well as I've not enough history to show you here on freelancer but we can have call and you can judge my experience. I totally understand your requirement—you already have a working default dialer handling GSM and SIP independently, and now you need a stable conference bridge between one GSM call and one SIP call using Android Telecom, with proper audio routing and state management on Android 11+. For your case, I’ll: • Extend ConnectionService to support cross-technology conferencing. • Implement controlled audio bridging between GSM (Telecom) and SIP media streams. • Manage audio focus, routing (speaker/Bluetooth), and lifecycle states properly. • Ensure compatibility with Android 11+ Telecom restrictions. • Deliver integrated source, documentation, and testing support. Let me know when you’re available to discuss this further I’d be happy to walk you through my approach or showcase examples relevant to this project. Looking forward to hearing from you! Best regards, Mulayam
$15 USD dalam 40 hari
0.0
0.0

Hi dear, I am Khushboo. I have strong expertise in Android Telecom framework, Connection Service, InCall Service, and SIP stack integration (PJSIP/Android SIP API). I can implement stable audio bridging between GSM and SIP calls, ensuring proper call states, audio routing, and seamless conference merging on Android 11+. Deliverables will include source code integration, documentation, and testing support for a clean and reliable solution. Thank You Khushboo Kumari
$20 USD dalam 40 hari
0.0
0.0

I have about 10 years of experience with web app developer and I'm pretty much a fast learner and I pretty much am very fast-paced when it comes to it and then I pay attention the detail and I'm a very hardworking and I make sure that the projects are perfect 2-inch. On top of that, I have extensive knowledge in Android Telecom framework and have successfully implemented similar features in the past. For instance, I recently worked on an app that required me to integrate a SIP stack (using PJSIP) with the default phone app for seamless call management. This involved handling audio routing, managing call states, and ensuring proper audio focus during conference calling - all of which are directly relevant to your project. Moreover, my experience in troubleshooting and maintaining networks would be a valuable asset as I understand the need for stability and clean solutions, especially when it comes to something as crucial as a phone dialer app. I have a deep understanding of GSM and SIP call processes and how they can be harmoniously merged into smooth conference calls - without compromising the overall functioning of the app.
$20 USD dalam 40 hari
0.0
0.0

Vijayawada, India
Ahli sejak Jun 30, 2020
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₹100-400 INR / jam
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