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We are currently working on our mobile application, GCM Dialer, which handles both SIM and SIP calls. We have successfully implemented the conference screen for the SIM and SIP merge option, but we are encountering an issue where voice is not being exchanged between all parties. Below is the scenario we are testing: - User-1 has the GCM Dialer app installed and set as the default dialer. - User-2 calls User-1 via SIM, and they can exchange voice successfully. - User-3 then calls User-1 via SIP. User-1 answers, which puts User-2 on hold. User-3 and User-1 can exchange voice successfully. - User-1 clicks the "Merge" button, and all three parties are brought into one call screen. The issue is that while User-1 can hear and speak to both User-3 and User-2, User-2 and User-3 cannot hear each other. User-2 can only hear User-1, and User-3 can only hear User-1. Essentially, User-1 can exchange voice with both parties, but the two external parties cannot exchange voice between themselves. Could you please review this scenario and provide guidance on how to resolve this audio routing issue during the merge?
Project ID: 40686713
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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
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