VoIP Audio Reliability Fix
Budget: ₹12,500 – ₹37,500 INR
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.
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.