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SIP Server Engineer/Asterisk Developer - Network Protocols

0 years

0 Lacs

noida uttar pradesh india

Posted:19 hours ago| Platform: Linkedin logo

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Skills Required

sip server developer network design communication voip tuning security integration configuration routing firewall management compliance optimization api salesforce monitoring maintenance logging analytics tcp udp qos linux ubuntu dhcp debugging wireshark programming scripting python automation crm kamailio twilio docker

Work Mode

On-site

Job Type

Full Time

Job Description

Role Overview

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.

Key Responsibilities

  • SIP Server Setup & Configuration: Deploy, configure, and manage SIP servers (Asterisk, FreeSWITCH, or similar PBX systems).
  • WebRTC Integration: Implement WebRTC solutions for browser-based real-time communication.
  • VoIP Features & Signaling: Develop and maintain SIP signaling flows, call routing, IVR, conferencing, and advanced telephony features.
  • Network & Firewall Management: Configure firewalls, NAT traversal, and optimize network setups for SIP/VoIP traffic.
  • SIP Event Management: Monitor, troubleshoot, and optimize SIP events, call flows, and media streams.
  • Security & Compliance: Implement security mechanisms such as TLS/SRTP, anti-fraud measures, and system hardening.
  • Performance Optimization: Continuously monitor server performance, debug issues, and fine-tune systems for scalability.
  • Integration: Work on API-based integrations of SIP/Asterisk systems with Salesforce and other enterprise platforms.
  • Monitoring & Maintenance: Use tools for real-time monitoring, logging, and analytics to ensure high availability.

Required Skills & Qualifications

  • Strong experience in Asterisk PBX and other open-source SIP/VoIP platforms.
  • Hands-on knowledge of SIP, RTP, WebRTC, TCP/UDP, NAT traversal, and QoS mechanisms.
  • Proficiency in Linux server administration (CentOS, Ubuntu, etc.).
  • Familiarity with VoIP codecs (G.711, G.729, Opus, etc.) and media handling.
  • Strong understanding of network protocols (DNS, DHCP, TLS/SSL, Firewall rules).
  • Experience with SIP debugging tools (Wireshark, sngrep, tcpdump).
  • Programming/scripting knowledge in Python, Bash, or C for automation and customization.
  • Good understanding of real-time communication security standards.
  • Prior experience integrating with CRM/enterprise applications is a plus.

Preferred Qualifications

  • Experience with FreeSWITCH, Kamailio/OpenSIPS, or similar SIP servers.
  • Knowledge of cloud telephony platforms (Twilio, Plivo, etc.).
  • Exposure to Salesforce integrations.
  • Familiarity with containerized deployments (Docker, Kubernetes).
(ref:hirist.tech)

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