Android-iOS VOIP Voice Call App

@crtvecode

It sounds like you're facing a challenging audio routing issue in the GCM Dialer app during the merge of SIM and SIP calls. This scenario, where User-2 and User-3 cannot hear each other despite being merged into the same call, likely stems from how the audio streams are being managed during the merge process. To resolve this, I would recommend reviewing the audio session configurations and ensuring that the media routing is correctly set up to allow bidirectional audio between all parties. This might involve checking the handling of audio channels and ensuring that the mixer is correctly configured to allow audio from both User-2 and User-3 to be routed through User-1. I can help you troubleshoot this issue by analyzing the current implementation and providing a solution that ensures all parties can communicate effectively. I'm ready to get started on this right away. Rajavel Mobile App Developer | VoIP | Audio Routing Solutions

@Ak230598

Hello, I can create/modify the app/web as you wish. I have 6.5+ years of experience in this field. Let's connect and discuss more about your requirements. Thank you.

@MuhammadAsad108

✔ I deliver 100% work — 99.9% is not for me. ✔ Workflow Call Flow Analysis ⟶ SIP/SIM Audio Path Review ⟶ Conference Bridge Analysis ⟶ RTP/Media Routing Debugging ⟶ Merge Logic Fix ⟶ 3-Party Testing ⟶ Audio Quality Verification Key Highlights ✔ Experienced Android/VoIP approach for diagnosing complex SIM + SIP conference audio routing issues. ✔ Trace the complete media path between User-1, User-2, and User-3 during hold and merge states. ✔ Identify whether the issue is caused by Android Telecom, SIP/RTP routing, Asterisk conference bridging, or application-level audio mixing. ✔ Verify that the merge creates a true three-way media bridge rather than simply placing two independent calls through User-1. ✔ Review SIP session state, RTP streams, codecs, direction attributes, and bridge configuration. ✔ Ensure User-2 and User-3 receive bidirectional media directly through the conference bridge. ✔ Test hold → resume → merge transitions and prevent duplicate/muted audio paths. ✔ Validate microphone, speaker, Bluetooth, codec, and audio-focus behavior on real Android devices. ✔ Provide clean implementation guidance and fixes without disturbing the existing working SIM/SIP call flows. Best Regards, Asad Android Developer | VoIP/SIP | Asterisk | Real-Time Audio Specialist

@technologiesit

Having worked in Mobile App Development for over 8 years, I believe I can tackle and resolve the audio routing issue that you're encountering. Being a Full Stack Developer, I bring a comprehensive understanding of how different components of an application interact with each other. Having traversed the issues faced during call merging across different mobile platforms, I'm confident that my experience will help to successfully resolve this scenario in your GCM Dialer app. Not only do I possess technical prowess needed for resolving this issue, but my business-first approach ensures that I prioritize solutions that resonate with your needs as well as your users' expectations. This means the resolution strategy we implement together will not only accurately address the audio routing problem, but it will be tailored to align with the architecture and goals of your app. Finally, my commitment is not limited to project completion - I'm focused on building long-lasting relationships with my clients by providing comprehensive ongoing support. Choose me, and let's resolve this issue together while creating a stable and scalable VOIP Voice Call App that will take GCM Dialer to greater heights. Let's get started today!

@GobindSawarn

With over 6 years of expertise in Mobile Development, I have a proven ability to solve complex issues and develop innovative solutions. I understand the intricacies of voice call apps and can leverage my experience to tackle the audio routing issue you're experiencing during the merge. Beyond just fixing the problem, I will ensure all parties -users one, two and three- can conveniently hear each other. Having successfully worked on similar projects involving SIP and SIM calls, I am deeply familiar with the challenges that can arise. By carefully analyzing your scenario, I plan to devise a solution that eliminates any form of discrepancy in audio handling between users two and three. My strategy would revolve around comprehensive testing, optimization of code and coordination between conduits. To express the importance of having this issue fixed, voice clarity is integral in communication applications such as yours; that's why my dedication to providing uninterrupted, high-quality voice calls makes me a strong contender for your project. Let us build an app whose users have no concerns about VoIP connection quality!

@adityakhuman006

Drawing from my 6+ years of mobile app development experience including proficient knowledge of Android development, I would be more than ready to help your team tackle the audio routing issue you're facing in your GCM Dialer. Known for my problem-solving skills, I thrive in resolving technical conundrums and providing effective solutions promptly. My expertise in building robust and user-friendly applications would enable me to fully comprehend your project needs. My mastery in utilizing version control systems such as Git guarantees that I can seamlessly integrate with your existing technology stack and adapt to your work culture with ease. Additionally, my knowledge and grasp of APIs lend me the necessary proficiency to analyze and offer precise insights on how to fix any challenges related to SIP calls versus SIM calls.

@shayankhan333

Hi, I can fix the SIM and SIP merge audio routing in GCM Dialer so User-2 and User-3 hear each other, not just User-1. The issue is that Android does not natively bridge SIM and SIP streams, so I will mix them locally through an audio bridge using the SIP stack's media session. I can start today. Questions: 1) Which SIP library are you using (PJSIP, Linphone, or custom)? 2) Can I get access to the current merge code to review? Looking forward to discussing further. Regards, Shayan.

@Visharaa2024

Hi, The key issue appears after the SIM and SIP calls are merged: User-1 maintains two-way audio with both parties, but User-2 and User-3 cannot exchange audio with each other. That suggests the conference UI/merge state is being reached, while the underlying audio/media path between the two remote parties is not being bridged correctly. We can investigate this systematically by tracing the call state and audio path during the merge, including how the SIM call and SIP media session are being connected, how audio routing is handled on the device, and whether the conference implementation is creating a true two-way media path for all participants. Rather than assuming the cause, we'd first reproduce the issue and inspect the existing merge/audio implementation to identify where the remote-to-remote path is being lost. Could you share which SIP stack/SDK and Android telecom or audio APIs are currently being used for the SIM/SIP integration? If the existing implementation and environment are available, we can review the merge flow and work toward an isolated fix. Best regards, Deepak

@bhargav922002

Hello, I’m bharghav, and I bring over a decade of experience in matching job skills to meet unique project requirements. My expertise lies primarily in Android, Asterisk PBX, and VoIP, which align well with your needs for the GCM Dialer app. I understand that you’ve made strides in implementing the conference screen for both SIM and SIP calls but are facing a significant challenge with audio routing during the merge of calls. The scenario you've outlined highlights that while User-1 can communicate with both User-2 and User-3, there’s a lack of direct communication between the two external parties. To resolve this issue, I will conduct a thorough review of your current implementation, focusing on the audio routing logic for merged calls. I aim to pinpoint the root cause and suggest clear modifications to ensure seamless communication for all parties involved. Let’s initiate a chat to discuss this issue in detail, as well as any other aspects of the project you may want to explore further. Best regards, bhargav922002

@Geetam00

I am Geetam, and I specialize in end-to-end product development for startups and businesses, creating applications that are not just interactive but rebuilt to withstand long-term growth. From planning and UI/UX design to AI implementation, backend engineering, cloud deployment, and ongoing support, I handle it all so that you don’t have to manage multiple teams. Having mentioned this, I am confident that I can help you resolve the audio routing issue in your GCM Dialer app's conference screen. My proficiency in Android & iOS mobile development alongside my experience with building AI-powered products makes me a strong candidate for the job. I understand the complexity involved when multiple parties are on a single call screen and will use my skills to find an efficient solution for the voice exchange between User-2 and User-3. My commitment to clean code, scalable architecture, and performance optimization ensures secure and high-performing applications at all times. By leveraging my expertise in OpenAI, LangChain, LangGraph, CrewAI & more for creating AI-powered systems including chatbots and recommendation engines, I'll ensure any limitation is swiftly overcome. Let's connect and discuss further on how we can make your VOIP voice call app truly reliable and efficient.

@mohaum0

I am the best candidate for this project because I have proven experience developing secure, cross-platform mobile banking applications for iOS and Android. I specialize in implementing secure fund transfer functionality, intuitive UX/UI design, and robust security protocols including encryption and authentication. I deliver clean, scalable, and maintainable code on time, and I understand the critical importance of reliability and compliance in financial applications.

@SoneJ

Hello, I appreciate the opportunity to assist with the audio routing issue in your GCM Dialer app. It sounds like a pivotal feature for your users, and I understand how frustrating it can be when functionality doesn't meet expectations. To address the problem, I would start by closely examining the audio session configurations for both SIM and SIP calls during the merging process. It’s crucial to ensure that all parties are properly added to the audio session so they can communicate with each other. I would implement a debugging process to identify where the audio stream is failing, ensuring that the correct audio paths are established for User-2 and User-3. This approach not only aims for a swift resolution but also enhances the reliability of future calls. You can expect clear communication throughout the process, with regular updates and early discussions on any questions that arise. Could you share any existing logs or error messages that might help pinpoint the issue? Additionally, what testing environments are you currently using? Looking forward to discussing this further. Regards, Soné Joubert

@rncmishra

Hi, We can help investigate and resolve the SIM + SIP conference/merge audio routing issue in your GCM Dialer application. From your description, the merge itself is working and User-1 is successfully connected to both User-2 (SIM) and User-3 (SIP). The issue appears to be specifically with audio bridging between the two remote parties—User-2 and User-3 can each communicate with User-1 but cannot hear each other. We have experience working with Android telephony, SIP/VoIP, WebRTC/audio routing, Telecom APIs, CallManager/ConnectionService, and real-time audio flows and can review your current implementation to identify where the audio path is being isolated during the merge. We can help with: Reviewing your existing SIM/SIP call architecture Analyzing the conference/merge implementation Checking Android Telecom/ConnectionService call handling Reviewing SIP audio routing and media streams Identifying why the two remote audio streams aren't being bridged Implementing the required audio mixing/bridging logic Testing the complete 3-party scenario Ensuring User-2 ↔ User-3 audio works while maintaining User-1 communication with both parties Please share the relevant conference/merge code, SIP implementation, ConnectionService/Telecom code, and audio routing logic, and we can investigate it.

@sheikhmateen

Hi, this comes down to how Android separates SIM and SIP audio paths. SIP gives you raw RTP you can mix freely, but the SIM leg runs through the carrier's baseband stack, which the OS doesn't expose for software mixing. So when you merge, both legs are likely just routing through User-1's mic/speaker path rather than being mixed into one shared stream, that's why User-1 is the only common point. The real fix is building an actual audio bridge: pull the SIP leg's RTP stream, get access to the SIM leg's audio (limited on stock Android, easier with a telephony SDK that exposes it), then mix and push the combined stream back to both parties. If direct SIM audio access isn't available on your target devices, routing that leg through a SIP trunk/PBX so all three legs conference natively is usually the more reliable path. I have built SIP-based voice systems before and run into this exact class of bug. Is the SIM leg going through a telephony SDK or just a native call intent right now? That'll tell me which fix applies.

@AlphaaT

Hi, I understand the issue with your GCM Dialer’s SIM + SIP conference merge. From the behavior described, it appears that the merge is successfully combining the call states, but the audio media paths between User-2 and User-3 are not being bridged correctly. User-1 can communicate with both parties, while the two remote parties remain isolated. I can review the existing Android/SIP/Asterisk implementation, trace the audio/media routing during the merge, and identify where the bidirectional audio path is being broken. I would specifically investigate the SIP media session, Asterisk bridge/conference configuration, Android audio routing, RTP/media handling, and how the SIM call audio is connected to the SIP call. I can work with your existing implementation rather than rebuilding the application from scratch. Once I review the relevant code, logs, and call flow, I can determine whether the issue is in the Android audio layer, SIP/RTP handling, or Asterisk-side bridging. Please share the current merge implementation and any relevant Asterisk/SIP logs so I can start by tracing the media flow. Best regards, Fahimur