Voltar a todas as vagasDesenvolvedor de voz (SIP)
Engenharia e tecnologia
China
Responsabilidades
- Develop core SIP/RTP-based voice modules, including call control, signaling, media negotiation, and session management.
- Customize and extend open-source platforms such as FreeSWITCH and Kamailio to build SBC, softswitch, and media server components.
- Integrate with upstream and downstream telecom carriers to support voice routing, SIP Trunk interoperability, and international voice termination.
- Ensure voice quality (QoS) and system stability under high-concurrency workloads, and establish call quality monitoring capabilities.
Requisitos
- Bachelor’s degree or above, with at least 5 years of VoIP development experience.
- Strong expertise in SIP, RTP, and SDP protocols, with an in-depth understanding of signaling and media flows.
- Hands-on experience with at least one of the following: FreeSWITCH, Asterisk, Kamailio, or OpenSIPS, including customization or secondary development.
- Proficiency in C/C++ or Go and strong familiarity with Linux environments.
- Familiarity with mainstream codecs, NAT traversal, and voice security technologies such as SRTP and TLS.