Fix WhatsApp Calling API Error

@vladang

Hello... I am interested

@JAYLEN001

EXPERT ((Node.js, JavaScript, API, Audio Processing, WebRTC and Debugging)) DEAR EMPLOYER, I’ve completed the exact same projects before successfully. Awarding me will be the fastest way to complete your task with the best rates possible. I CAN ASSURE YOU 100% THAT WE ARE FULLY CAPABLE OF EXECUTING ANY LEVEL OF TASK/PROJECT BASED ON THE SKILL REQUIRED. I am fully confident about our skills and my understanding of the project description and we are ready to go through any test or sample task you assign to acquire your trust. Let me know when are you available for an initial 15-30-minute discussion (FREE OF CHARGE) so we can discuss the requirement in detail and I can walk you through the mentioned systems to acquire your trust in my skill. REST ASSURED YOUR WORK IS IN VERY SAFE AND PROFESSIONAL HANDS. THANK YOU

@Vishvajeet02

I have strong expertise in WebRTC, real-time communication, and backend integrations with Node.js and APIs. I reviewed the WhatsApp Cloud API Calling documentation as well as your Stack Overflow question, and I understand that the issue lies in the Media Stream handling, where audio works locally but fails in production despite similar logs. Such problems often relate to peer connection configurations, ICE candidates, signaling, or media permissions on live servers. With my experience in WebRTC signaling, peer-to-peer audio/video streams, and debugging production-level audio issues, I can efficiently trace the root cause and apply a stable fix. I have also worked with audio codecs, STUN/TURN servers, and real-time media optimization, ensuring smooth communication across environments. My goal is to deliver a reliable and production-ready fix so your WhatsApp calling feature works seamlessly.

@mohamede816

Hi — I’m Mohamed. I build real-time audio systems and AI voice assistants and I’m confident I can fix your WhatsApp WebRTC audio issue. I’ll reproduce the bug in production, collect logs/SDP/ICE, verify getUserMedia + codecs (OPUS), add/validate TURN, fix signaling/SDP mismatches, and deploy a tested patch. Delivery in 7 days with daily updates and a final report + README. Proposed milestones: 1) Repro & diagnosis (1–2 days), 2) Fix & test (3–5 days), 3) Deploy & handover (1–2 days). I’ve optimized backends, deployed to Vast/Hostinger and debugged real-time audio in production. Ready to start — I’ll need access to logs and a test account.

@ketan2217

i am computer science student and i have well knowledge about internet and error solving like mobile apps , and computers]

@VertexiaWeb

I am a perfect fit for your project as I have extensive experience with WhatsApp Calling API and WebRTC integration. I've successfully resolved similar audio issues following the official documentation you provided. While I am new to freelancer, I have tons of experience and have done other projects off site. I would love to chat more about your project! Regards, Ebraheem

@kamrana118

Drawing on my extensive experience as an AI, Mobile, and Full-Stack Developer, I offer the necessary expertise to effectively troubleshoot and resolve any issues you are currently facing with the WhatsApp Calling API and WebRTC. Having worked with various complex systems like blockchain, mobile development using React Native and Flutter, along with backend skills in Node.js - your project’s technology stack aligns perfectly with my skill set. Over the years, I have displayed a consistent knack for problem-solving, debugging, and optimizing systems to improve efficiency. My track record of deeply understanding systems such as Media Stream in WebRTC and finding tailored solutions will bring immense value to your project. Additionally, I maintain clean code by industry standards, thoroughly document my work, and strongly emphasize testability - all important facets for mission-critical projects like this one. Furthermore, my strength in System Integration & DevOps ensures that not only will your immediate problem be addressed and resolved promptly, but we can also ensure that the root of the issue is fixed to prevent it from recurring in the future. I am dedicated to reliable and maintainable solutions that demonstrate long-term value. Finally, my commitment to clear communication and timely delivery ensures a smooth collaboration cycle. Let's hop on a call to discuss further how we can get your project back on track efficiently!

@tristanb09

I am a perfect fit for your project, specifically with expertise in WhatsApp Calling API and resolving WebRTC issues like Media Stream for seamless audio integration. While I am new to freelancer, I have tons of experience and have successfully tackled similar projects offsite. I would love to chat more about your project! Regards, Tristan

@philiph13

Hello, As an experienced developer and cybersecurity professional, I bring a unique set of skills to the table that make me an ideal fit for your project. My extensive knowledge in API integration, debugging, JavaScript, Node.js, and specifically WebRTC lends itself perfectly to resolving your WhatsApp Calling API issue. Having resolved multiple complex issues similar to yours in the past has made me proficient with signaling and peer connection problems as well as debugging audio issues within the Media Stream component of WebRTC. I understand the urgency of fixing this issue on your live production environment and will provide timely and efficient solutions. Please review my portfolio/resources on these relevant projects on Kaggle/StackOverflow where you find detailed solutions I have provided to similar tasks. You can trust me with high-tech project like this, knowing you have the very best in the field working for your best interest. Thank you for considering my services; I look forward to resolving your WhatsApp Calling API error expeditiously. Thanks!

@satyajeet18

I will debug and resolve your WhatsApp Calling API + WebRTC audio issue. The problem occurs only in production where MediaStream audio fails despite similar logs locally. Scope: review signaling flow, check SDP/ICE candidates, confirm STUN/TURN servers, and ensure tracks are unmuted and enabled. Deliverables: permanent fix for live audio, tested end-to-end on production, plus added logging to prevent future regressions. I have strong experience with WhatsApp Cloud API, WebRTC MediaStream, and peer connection debugging.