SIP Integration Developer

Kooya Inc; Taguig; On-site; PHP 900,000 - 1,300,000; Full time

Job summary

Kooya Inc. in Taguig is seeking a skilled SIP Integration Developer with a minimum of 3 years of hands-on experience in voice engineering and telecommunications software development.

The role focuses on architecting scalable SIP routing, ensuring high availability, and optimizing for high-volume traffic. You will tackle core architectural issues, implement scalable SIP strategies, and improve performance across the platform, with responsibilities spanning from load balancing to API optimization

Qualifications

• 3+ years hands-on SIP voice engineering experience.
• Bachelor's degree in a related field.
• Strong background in SIP, RTP/RTCP, SDP, WebRTC, and TCP/IP networking.

Responsibilities

• Diagnose, isolate, and resolve architectural issues in SIP routing and media engines.
• Design scalable SIP routing to handle millions of calls and high CPS.
• Architect and optimize failover, clustering, and load-balancing for 99.999% uptime.
• Tune performance: reduce latency and packet loss in RTP handling.
• Optimize backend APIs and webhooks for asynchronous, high-throughput processing.

Skills

High-Volume SIP Routing

Protocols SIP/RTP/WebRTC

Concurrency & Systems Programming

Diagnostic Tools

Performance Testing

Education

Bachelor's degree

Tools

Kamailio

OpenSIPS

SIPp

Wireshark

sngrep

Hep/Homer

Job description

On-site - Taguig 3-5 Yrs Exp Bachelor Full-time

Job Description

Role Overview

We are seeking a skilled and detail-oriented SIP Integration Developer with a minimum of 3 years of hands‑on experience in voice engineering and telecommunications software development.

In this role, your primary mission will be to tackle core architectural issues, eliminate system bottlenecks, and optimize our platform to handle high-volume concurrent voice traffic. You will ensure our SIP infrastructure is highly available, resilient, and capable of scaling seamlessly under heavy loads.

Key Responsibilities

• Core Issue Resolution: Diagnose, isolate, and resolve deep‑seeded architectural issues, memory leaks, and concurrency bugs within our core SIP routing and media engines.
• High-Volume Scaling: Design and implement highly scalable SIP routing strategies capable of handling millions of concurrent calls, managing high CPS (Calls Per Second), and mitigating spikes in traffic.
• Load Balancing & Redundancy: Architect and optimize failover mechanisms, clustering, and load‑balancing solutions using tools like Kamailio or OpenSIPS to ensure 99.999% uptime.
• Performance Tuning: Monitor system metrics under heavy load, optimize RTP media handling, reduce latency, and minimize packet loss across the network.
• API & Integration Optimization: Ensure backend APIs and webhooks handling call state events are built for asynchronous, high‑throughput processing.

Required Qualifications & Skills

Technical Expertise

• High-Volume SIP Routing: Proven experience configuring proxy servers and load balancers (Kamailio, OpenSIPS) to manage massive call volumes.
• Protocols: Deep understanding of SIP, RTP/RTCP, SDP, WebRTC, and TCP/IP networking, specifically how they behave under heavy stress.
• Concurrency & Systems Programming: Strong backend programming skills in a language built for high concurrency (Go, C/C++, Python with AsyncIO, or Java).
• Diagnostic Tools: Advanced mastery of Wireshark, sngrep, and Hep/Homer for capturing and analyzing distributed SIP traces in real time.
• Performance Testing: Experience with SIP stress‑testing tools like SIPp to simulate high‑volume call traffic and identify breaking points.

What Success Looks Like in This Role

• Reduced Downtime: Quick resolution of complex, intermittent core platform bugs that affect call quality or routing logic.
• Increased Throughput: Measurable improvements in the platform's ability to handle high CPS without degrading Voice Quality (MOS scores).
• System Resilience: Seamless, automated failover behavior during infrastructure or carrier outages.

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