.If you always dreamed about designing the KIIT of the future... Your place is here! We are looking for a VoIP Systems Developer with strong C++ skills and experience in VoIP technologies. You will be part of a multidisciplinary team in charge of developing and optimising telecommunication solutions based on IP communication protocols for railway systems. If you have experience in the design, implementation and optimisation of VoIP systems and a strong command of C++, this is your opportunity! WHAT CHALLENGE WILL YOU BE TAKING ON? Your main responsibilities will be: Design, develop and maintain VoIP solutions using C++ as the main language. Implement and optimise VoIP protocols such as SIP (Session Initiation Protocol), RTP (Real-time Transport Protocol), and RTCP (Real-time Transport Control Protocol) for voice and video communications. Collaborate with network teams to ensure efficient integration with VoIP servers and devices. Perform debugging and troubleshooting in VoIP environments , ensuring high availability and quality of service. Implement cybersecurity measures in VoIP communications . WHAT DO WE NEED IN OUR TEAM? For this position, we are looking for Graduates in Computer Engineering or similar with experience in: VoIP technologies: SIP (Session Initiation Protocol) : You must be proficient in this protocol both in theory and in practice. This includes the ability to implement SIP clients and servers, as well as interaction with other SIP-based communication platforms. RTP (Real-time Transport Protocol) and RTCP (Real-time Transport Control Protocol) : Experience working with these protocols for real-time voice and video transmission. The ability to manage aspects such as media synchronisation, latency and quality of service (QoS) will be an asset. SDP (Session Description Protocol) : Knowledge in the use of SDP to describe multimedia sessions and negotiate codec and media parameters in VoIP systems. PJSIP or OpenSIPS : Experience with these libraries or frameworks for the development of VoIP solutions. Ability to work with PJSIP for the management of multiple SIP channels is an asset. Asterisk or FreeSWITCH server : Experience in configuring and customising VoIP servers based on Asterisk or FreeSWITCH for call management, conferencing and other communication services. C++ and Familiarity with libraries for multimedia processing (e.G. GStreamer, FFmpeg). WHAT DO WE OFFER? Hybrid working model and 8 weeks per year of teleworking outside your usual geographical area. Flexible start and finish times , and intensive working hours on Fridays and in summer. Personalized career plan development, training and language learning support. National and international mobility . Do you come from another country? We can offer you a relocation package . Competitive compensation with ongoing reviews , flexible compensation and discount on brands