VoIP and SIP Telephony Engineer with Asterisk and Kamailio
We are looking for an experienced VoIP / SIP Telephony Engineer with strong hands-on experience in Asterisk, Kamailio, SIP infrastructure, and mobile application integration. The ideal candidate will be responsible for reviewing our existing telephony architecture, performing technical audits, tracing communication flows between components, troubleshooting system issues, and identifying opportunities to improve reliability, performance, and security.
Key Responsibilities:
- Review and audit existing Asterisk and Kamailio architecture, configurations, and integrations.
- Analyze and document end-to-end SIP communication and call flows.
- Trace communication between mobile applications, backend services, APIs, Kamailio, Asterisk, SIP providers, and media services.
- Identify configuration issues, routing problems, integration failures, bottlenecks, and potential points of failure.
- Troubleshoot SIP registration, call setup, call routing, dropped calls, authentication, RTP, and audio-related issues.
- Analyze SIP messages, system logs, network traffic, and configurations to identify root causes.
- Review NAT, firewall, networking, RTP, codecs, and SIP routing configurations.
- Investigate how Android and iOS applications connect to and interact with the SIP infrastructure.
- Determine whether issues originate from the mobile application, backend systems, SIP infrastructure, network, or telephony configuration.
- Recommend improvements to system reliability, scalability, security, and performance.
- Collaborate with backend, mobile, infrastructure, and product teams to resolve technical issues.
Experience and Skills Needed:
- 5+ years of relevant experience in VoIP, SIP, telecommunications, telephony, or real-time communication systems.
- Strong hands-on experience with:
- Asterisk
- Kamailio
- SIP
- RTP
- VoIP infrastructure
- Strong understanding of SIP signaling, call flows, registration, routing, authentication, and media communication.
- Experience troubleshooting real-world VoIP and SIP issues.
- Strong Linux server and networking knowledge.
- Good understanding of TCP/IP, NAT, firewalls, routing, ports, and network troubleshooting.
- Experience with mobile applications that interact with VoIP/SIP infrastructure.
Soft Skills:
- Strong analytical and troubleshooting skills.
- Excellent attention to detail.
- Ability to investigate complex technical problems independently.
- Strong communication and technical documentation skills.
- Proactive and solution-oriented mindset.
- English proficiency: Upper-Intermediate (B2) or higher.
Equipment requirements:
- High-speed Internet with at least 50 Mbps upload and download speeds
- Reliable computer suitable for technical troubleshooting and development, noise-cancelling headset with microphone,
- High-quality video conferencing camera to participate in meetings with clients and company management.
Academic requirements:
Bachelor's Degree from a recognized University in Computer Science, Software Engineering, Telecommunications, Information Technology, Electrical Engineering, or a related technical field. Preferences will be granted to candidates with Master's qualifications or relevant telecommunications, VoIP, networking, or engineering certifications.
Please ensure your CV and supporting documentation, including a copy of your diploma and grade transcripts, clearly demonstrate your relevant experience with Asterisk, Kamailio, SIP, VoIP, networking, and mobile application integration.
Candidates must have AI certification from Anthropic (Claude) or equivalent accreditation institutions, demonstrating practical experience in developing with COWORK, AGENTS, SKILLS, LLM, workflows, automations, or other AI solutions. Link for the required courses: https://claude.com/resources/courses or equivalent.
Candidates must have worked with clear understanding utilizing the Anthropic Claude Co-working workspace and tools.
If you are interested in this opportunity, we encourage you to apply now.