Desarrollador de voz (SIP)

Ingeniería y tecnología

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.

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.