Android-iOS VOIP Voice Call App

@mrresearcher99

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

@crtvecode

It sounds like you're facing a tricky audio routing issue with the GCM Dialer app during the merge process. From what you've described, User-1 can communicate with both User-2 and User-3, but they can't hear those other users, which suggests a problem with the audio mixing or routing logic when merging calls. So, to tackle this, I'll analyze the call handling flow and identify where the audio paths are breaking down, I'll provide specific guidance on how to adjust the audio routing so that all parties can communicate effectively once merged. I'm kind of ready to get started on this right away for 3199 INR. Looking forward to resolving this issue with you! Rajavel VoIP Developer | Android & iOS Apps

@Mayanks010

Hi, Thanks for sharing your project. It aligns well with work I’ve successfully completed for other clients, and I’m confident I can deliver a high-quality solution tailored to your requirements. Before providing an accurate timeline, scope, and budget, I’d like to clarify a few key details to ensure we're aligned from the start. Freelancer’s proposal character limit makes it difficult to cover everything thoroughly here. If you're available, let's connect via chat. I can walk you through similar projects I've completed, discuss the best approach for your specific requirements, and answer any questions you may have. Once I understand your expectations, I'll provide a clear execution plan with realistic timelines and pricing. I’m ready to get started as soon as we finalize the details and look forward to working with you. Best regards, Mayank Sahu

@kratika0510

With over three years of direct experience in mobile app development, including proficiency in Android and iOS platforms, I am confident in my ability to resolve the audio routing issue you're presently experiencing with GCM Dialer. Your challenge aligns perfectly with my expertise. Our team at Neuravolt Technology specialises in transforming challenges into solutions, and our successful track record with diverse businesses assures you of our skills in finding a fitting resolution. One aspect that sets us apart is our collaborative approach. We don't just singularly solve technical problems but consider the larger picture and objectives of your project. In this case, my proficiency in not only Mobile Application Development but also Cloud Infrastructure & Migration, DevOps & CI/CD Implementation will enable me to diagnose the problem right within the project's ecosystem—ambient factors that often affect SIM and SIP calls— allowing us to devise a lasting solution. Moreover, at Neuravolt we uphold industry best practices ensuring each solution is maintainable and future-ready. Given the evolving nature of technology, my team and I are not only committed to resolving your immediate issue but also incorporate long-term maintenance into our delivery. Choose us not just for this task, but for an enduring relationship that'll ensure your business remains competitive as we continue to innovate together.

@nabeeli18

Greetings I understand you are facing audio routing issues during the merge of your conference call feature in the GCM Dialer app. We have successfully completed similar projects involving VoIP applications and can effectively tackle this challenge. Our goal is to ensure seamless communication between all parties during merged calls by diagnosing the audio routing logic. We can analyze the call flow and implement necessary adjustments to enable direct audio exchange between User-2 and User-3 while maintaining the current functionality for User-1. I bring experience in developing VoIP applications and troubleshooting audio issues, ensuring reliable communication for users. I'd love to chat about your project and suggest a follow-up discussion to explore potential solutions. Regards, Nabeel Ismail

@bitcoinpot

The key issue is the audio path created when a SIP participant is merged with an active SIM call: User-1 can hear both sides, but the two remote parties are not receiving each other’s audio. I’d first trace the merge flow, audio routing, SIP media path, and conference handling to identify where User-2 and User-3’s streams stop being bridged. Then I’d isolate whether the issue is in the Android call layer, SIP/RTP handling, or the Asterisk conference configuration before changing the implementation. Using VoIP and SIP, I’d focus on the actual media path rather than the call UI, and provide a targeted fix or clear guidance based on the current architecture. Can you share the relevant merge-call code and the Asterisk/SIP configuration used when the SIM and SIP calls are combined?

@eternitech

Hi, New on Freelancer — 20 years of development experience behind us. We're taking our first few projects here at a fraction of our normal rate purely to build our review history. You get senior agency work at junior pricing; we get a review. Straight trade. integrating SIP call handling with Android and iOS often requires careful attention to network conditions and codec compatibility, which could affect call quality. My first step would be to ensure the SIP settings are optimized for the best performance across both platforms. Could you provide access to the current GCM Dialer build so I can review the implementation?

@anas389

Hi, I can investigate the audio routing issue in your GCM Dialer and help identify why the SIM and SIP legs are not exchanging audio after the merge. From your scenario, User-1 is acting as the bridge between the two call legs, while User-2 and User-3 cannot hear each other. This suggests the merge is creating the combined call UI/state correctly, but the underlying audio paths or media routing are not being connected bidirectionally between both legs. I’ll review the existing conference/merge implementation, including Android Telecom/ConnectionService handling, SIM call audio state, SIP media streams, audio focus/mode, call hold/resume behavior, and how the two call legs are bridged during merge. I’ll trace the actual audio path rather than changing UI code blindly. The goal will be to determine whether the issue is in Android audio routing, SIP media handling, the conference implementation, or the bridge logic, and then provide a clean fix or precise implementation guidance. I have a software engineering and QA background with API/integration debugging and testing experience. I won’t claim prior GCM Dialer-specific expertise without reviewing the code first. Please share the relevant merge/conference code, SIP library being used, Android version/target SDK, and logs from the merge operation. I can start by reviewing the existing implementation.

@adarshverma161

Hello dear. I have gone through the issue. There are possibly 3 case scenarios - Case 1 Conference/media architecture already exists; mixer connection missing. Case 2 SIP + SIM conference architecture partially implemented, but media bridge redesign required. Case 3 Android platform limitations prevent the intended direct SIM-SIP media mixing architecture. If you can provide the exact code, I can see through and find what's the problem.

@eugabrielmaximo

Hellom The merge itself is working, but the audio path is not being bridged correctly between the SIP and SIM legs: User-1 can hear both parties, while User-2 and User-3 remain isolated from each other. This suggests the conference/merge logic is creating a hub where User-1 is connected to both calls, but the media streams are not being mixed or forwarded bidirectionally between the two external participants. I’d first trace the audio flow for each call leg before and after Merge, specifically checking how the SIM audio path, SIP RTP/media stream, audio focus, and conference bridge are connected. I’d then verify whether the merge implementation is actually mixing both remote streams or simply keeping two separate call connections active. The fix would depend on your current architecture, but I would focus on: * Identifying where the SIM and SIP media streams terminate. * Verifying the conference mixer/bridge receives both remote streams. * Ensuring User-2 audio is routed into the SIP leg and User-3 audio into the SIM leg. * Checking Android Telecom/ConnectionService and audio-routing behavior during the merge. If you share the current merge/conference implementation and relevant audio-routing code, I can review the exact issue and point out what needs to change.

@abhilashr53

You need a true three-way audio bridge so User-2 and User-3 can hear each other, not just User-1. Right now your merge is mixing locally on the device instead of routing audio bidirectionally between all legs. My plan: audit your current AudioManager and SIP stack merge logic, implement a proper audio mixing bridge that captures and redistributes audio streams across both the SIM and SIP legs, and validate on real Android hardware. I'll also check iOS parity if needed. I've solved similar cross-protocol audio routing issues involving Telecom API, ConnectionService, and SIP stacks like PJSIP or Linphone. Realistic fix timeline is 5-10 days depending on your current codebase structure. What SIP library are you using, and do you have the ConnectionService implementation handling the SIM leg?

@karthikresonite

Hi, We at Resonite Technologies are excited about the opportunity to assist you with your GCM Dialer app. Our experienced team specializes in VOIP solutions and has a proven track record of tackling complex audio routing challenges. We understand the critical nature of your current issue where User-2 and User-3 cannot hear each other during the merged call. This typically stems from how audio streams are managed and routed during the merging process. It’s essential to ensure that the media server or the app's audio management system supports seamless communication between all parties involved. Our approach will involve: 1. Thorough Analysis: Reviewing the current implementation of your merge functionality to identify the root cause of the audio routing issue. 2. Testing and Debugging: Conducting tests to simulate various call scenarios and pinpointing discrepancies in audio flow. 3. Implementation of Solutions: Providing recommendations or code modifications to ensure all parties can hear each other during merged calls. We are confident in our ability to resolve this issue efficiently and enhance your application’s performance. Best regards, Karthik B Resonite Technologies

@rockfortglobal

Hi, My name is Param Ramanan form Rockfort Global based out of Adelaide. We can investigate and resolve the audio-routing issue in your GCM Dialer SIM + SIP conference merge. My team has experience with Android telephony, SIP/VoIP, audio streams, call bridging, WebRTC/SIP media handling, and conference-call architecture. From your scenario, the key issue appears to be the media path between the SIM and SIP participants after merging, where User-1 remains the central audio endpoint instead of establishing a true three-party bridge. We will trace the call/audio routing, inspect the merge implementation, verify AudioManager and SIP media configuration, and identify where User-2 and User-3 streams are being isolated. We’ll then implement and test a reliable bidirectional audio bridge across all participants. Please google the name of our company or my name for more information. I request you to please connect with on the freelancer chat so we can discuss your needs further. Kind Regards Param Ramanan

@divyapatel1891

Hi, I’m a Senior iOS Developer with 8+ years of experience and 5+ years specializing in VoIP and real-time communication. I have hands-on experience with SIP, PJSIP, Linphone, WebRTC, CallKit, VoIP audio routing, call transfer, conference calling and call recording. I understand the issue described: User-1 can communicate with both the SIM and SIP participant, but User-2 and User-3 cannot hear each other after the merge. This suggests that the merge is successfully combining the call UI/state, but the underlying audio paths are not being bridged between the two external call legs. I can review the existing call/conference implementation, audio session/routing, SIP media handling and bridge/conference configuration to identify where the audio path is breaking and implement the required fix. I can also help determine whether the issue is on the mobile audio-routing side or the SIP/Asterisk PBX bridging side. I’m available to start immediately and can work with your existing codebase. Thank you

@quinlannh

INTRO I recently assisted a client in resolving a similar audio routing issue within a VOIP application. We streamlined their communication process and enhanced user experience. I can help you ensure seamless voice exchange for all parties in your GCM Dialer app. Let's schedule a free consultation and initiate fast project planning to tackle this issue immediately. MIDDLE I understand that while User-1 can communicate with both User-2 and User-3, the lack of audio exchange between them is a critical problem. Addressing this clean and professional integration is essential. I specialize in VOIP solutions and have extensive experience troubleshooting complex audio routing issues. I can start right away and deliver a quick action plan tailored to your needs. Could you share which audio libraries or protocols you are currently using to facilitate these calls? Regards, Quinlann.

@Ankurhardia34

Hello, I'm Ankur, a freelance developer with a dedicated team of professionals. I read all your requirements for Android iOS VOIP voice call and I assure you that I will provide high-quality work at the proper time. Additionally, we also provide you 3 months of support from our side As a Full Stack Developer, I specialize in Web and App Development, boasting a portfolio of stunning projects with top-notch UI/UX design. My expertise spans Flutter (for both Android and iOS), PHP, and WordPress, and I bring over 7 years of experience to the table. Whether it's websites, applications, or e-commerce platforms, I've got you covered. But I'm not limited to just coding. My skill set extends to graphic design and logo creation, offering you a one-stop solution for all your project needs. With a track record of over 500 completed projects, I am committed to delivering nothing short of excellence. My ultimate goal is your complete satisfaction. For a closer look at my work, please visit my portfolio Thank you for considering me for your project. I'm ready to transform your vision into a reality that stands out in today's competitive landscape. Best Regards, Ankur

@safarudinns

⭐⭐⭐Hello⭐⭐⭐ I see you are facing an audio routing issue during the merge of SIM and SIP calls in your GCM Dialer app. The challenge where User-2 and User-3 cannot hear each other while User-1 can communicate with both is a common scenario that can arise from how audio streams are managed in the merging process. This project fully aligns with my skills, and I can provide effective solutions within your timeframe with 100% accuracy. I would love the opportunity to connect with you and discuss your project further. To summarize, you need assistance in resolving the audio exchange problem when merging calls between SIM and SIP users. I completely grasp the situation and can help you identify the root cause and implement a fix to ensure all parties can communicate seamlessly. I look forward to collaborating with you, as your timeline requires prompt attention. Best regards, Sararudin

@nehat049

I am working as a Telecom Engineer from last 10 years. I worked various testing domains e.g. RAN testing which includes SIM and mobile testing as well. I believe i can help in this project

@shuvadeep003

Hi, You've already implemented the SIM and SIP merge flow, so the remaining challenge is getting true three-way audio where User-2 and User-3 can hear each other after the merge. My team can help debug this conference-call flow by tracing the audio routing between the Android Telecom framework, SIP session handling, and the conference bridge to identify why the two external parties remain isolated. We'll review the merge implementation, verify hold/resume behavior, inspect audio focus and routing, and ensure the conference mixes audio correctly instead of keeping both legs connected only to User-1. We'll also help validate the fix across different call sequences and provide clear documentation of the root cause and implementation changes. Before we begin, I'd like to know which SIP stack you're using (PJSIP, Linphone, or another library), the Android version you're targeting, and whether the conference is managed through Android Telecom's ConnectionService or a custom audio bridge. Best regards, Shuvadeep