Job Description
<p><p><b>Role Overview :</b><br/><br/>We are seeking a highly skilled SIP Server Engineer/Asterisk Developer to design, configure, and maintain SIP-based communication systems.
The candidate should have in-depth knowledge of Asterisk, VoIP technologies, and real-time communication protocols, with strong expertise in performance tuning, security, and integration with enterprise platforms.<br/><br/><b>Key Responsibilities :</b><br/><br/>- SIP Server Setup & Configuration: Deploy, configure, and manage SIP servers (Asterisk, FreeSWITCH, or similar PBX systems).<br/><br/>- WebRTC Integration: Implement WebRTC solutions for browser-based real-time communication.<br/><br/>- VoIP Features & Signaling: Develop and maintain SIP signaling flows, call routing, IVR, conferencing, and advanced telephony features.<br/><br/>- Network & Firewall Management: Configure firewalls, NAT traversal, and optimize network setups for SIP/VoIP traffic.<br/><br/>- SIP Event Management: Monitor, troubleshoot, and optimize SIP events, call flows, and media streams.<br/><br/>- Security & Compliance: Implement security mechanisms such as TLS/SRTP, anti-fraud measures, and system hardening.<br/><br/>- Performance Optimization: Continuously monitor server performance, debug issues, and fine-tune systems for scalability.<br/><br/>- Integration: Work on API-based integrations of SIP/Asterisk systems with Salesforce and other enterprise platforms.<br/><br/>- Monitoring & Maintenance: Use tools for real-time monitoring, logging, and analytics to ensure high availability.<br/><br/><b>Required Skills & Qualifications</b><br/><br/>- Strong experience in Asterisk PBX and other open-source SIP/VoIP platforms.<br/><br/>- Hands-on knowledge of SIP, RTP, WebRTC, TCP/UDP, NAT traversal, and QoS mechanisms.<br/><br/>- Proficiency in Linux server administration (CentOS, Ubuntu, etc.).<br/><br/>- Familiarity with VoIP codecs (G.711, G.729, Opus, etc.) and media handling.<br/><br/>- Strong understanding of network protocols (DNS, DHCP, TLS/SSL, Firewall rules).<br/><br/>- Experience with SIP debugging tools (Wireshark, sngrep, tcpdump).<br/><br/>- Programming/scripting knowledge in Python, Bash, or C for automation and customization.<br/><br/>- Good understanding of real-time communication security standards.<br/><br/>- Prior experience integrating with CRM/enterprise applications is a plus.<br/><br/><b>Preferred Qualifications :</b><br/><br/>- Experience with FreeSWITCH, Kamailio/OpenSIPS, or similar SIP servers.<br/><br/>- Knowledge of cloud telephony platforms (Twilio, Plivo, etc.).<br/><br/>- Exposure to Salesforce integrations.<br/><br/>- Familiarity with containerized deployments (Docker, Kubernetes).</p><br/></p> (ref:hirist.tech)