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Our production VoIP app runs on Kamilio/Asterisk is suffering from three clear symptoms: call dropping, audio quality issues. The trouble appears only on a fraction of calls rather than every connection, so I need someone who can track down the underlying triggers—whether they sit in the signalling layer, jitter buffer, codec choice or network traversal logic—and then eliminate them. You’ll start by reviewing the current SIP/WebRTC stack, server logs, packet captures and any client-side metrics I provide. From there, pinpoint the root cause and apply code, configuration or infrastructure tweaks to stabilise audio flow. Deliverables • Detailed diagnosis outlining where packets or media streams fail • Fixed client or server code / configs ready for deployment • A concise test plan that proves calls connect consistently without drops, one-way audio or choppiness across the scenarios that used to fail Acceptance is simple: test calls that previously exposed the issue must now play cleanly end-to-end.
Proje No: 40042831
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8 freelancer bu proje için ortalama ₹24.094 INR teklif veriyor

Drawing from an extensive background in both web and mobile development, I leverage expertise in Android technology to bring creative solutions to the table. Having successfully executed projects involving SIP/WebRTC stack, server logs and packet captures in the past, I am confident in my ability to diagnose and resolve issues plaguing your VoIP application. My expertise ranges from code modifications to infrastructure tweaks and optimizations which will ensure a stable audio flow on your app. One of the things that sets me apart is a strong commitment to delivering quality work on time and within budget. As part of our package, we also offer free post-delivery support for three months to guarantee seamless performance even after the project is completed. In addition, we can provide cost-effective hosting and domain options to further alleviate any worry. I am dedicated to turning ideas into realities, which precisely mirrors your purpose with this project. From fixing call drops to ensuring consistent audio, you can trust in my competence. Let us take on this challenge together- turning potential bugs into smoother and stable user experiences!
₹25.000 INR 7 gün içinde
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.
₹25.000 INR 7 gün içinde
5,5
5,5

Hi Alexandre F., Good morning! Already have something live to show you I am professional mobile programmer with skills including Network Security, Asterisk PBX, VoIP, Android, SIP, Linux, WebRTC and Audio Engineering. Please send a message to discuss more about this project. Feel free to contact us to discuss your project
₹12.500 INR 4 gün içinde
0,4
0,4

Hello, As a seasoned software engineer with a comprehensive knowledge within the disciplines of C++, Python, and networking (including UDP and RTP), I believe I am precisely what your project needs to conquer these VoIP audio issues. Over my 8-year career, I've been solving complex technical challenges similar to yours with great success. My understanding of Linux, Asterisk PBX, and VoIP in general will be sterling attributes for this venture. But my expertise lies not solely in troubleshooting problems; I'm equally adept at crafting long-term, integral solutions. Additionally, as someone who's contributed actively towards building high-performing, scalable systems - your project aligns perfectly with my skill set. Furthermore, my familiarity with the Kamilio and Asterisk base will only hasten our progress. Choosing me for this mission means selecting a professional who excels concurrently in C++, Python development for server-side interworking and WebRTC for the frontend interface. With me on board, you have a multi-faceted teammate who can harmonize and optimize code across all layers of your project to give you results that exceed your expectations. Let's eliminate these issues together and build a stable foundation for your VoIP app! Thanks!
₹27.750 INR 1 gün içinde
0,0
0,0

IMPOSSIBLE! SO I DON'T EVEN USE AI FOR BIDDING. Hi there! If I were to share my perspective on the project, it's virtually impossible to grasp the entire project in a short period of time. What I can honestly say right now is that the technology stack is suitable for me, and I can implement it based on my previous experience. I'm an honest developer, so I don't even use AI for bidding. I want to work with clients who value honesty.
₹25.000 INR 7 gün içinde
0,0
0,0

Hello, I can help you diagnose and fix the intermittent call drops and audio issues in your Kamailio/Asterisk environment. I specialize in SIP/WebRTC troubleshooting and real-time media reliability, and I’ve solved similar problems caused by signalling errors, jitter buffer behavior, codec negotiation, ICE handling, and NAT traversal. I will start by reviewing your SIP/WebRTC stack, Kamailio routing, Asterisk media settings, server logs, PCAP traces and client metrics. From there, I will pinpoint the root cause and apply the required configuration or code adjustments to stabilize RTP flow and ensure consistent audio quality. Deliverables: • Clear diagnosis showing where SIP or RTP fails • Corrected configs/code ready for deployment • Test plan proving clean end-to-end calls without drops or one-way audio I’m available to start immediately and can provide fast, verifiable results. Best regards, J. Fonca
₹27.000 INR 5 gün içinde
0,0
0,0

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