مهندس SIP/VoIP (SIP/VoIP Engineer)

Job Summary

The SIP/VoIP Engineer will develop and maintain Sarj’s telephony systems, integrating Voice AI agents with client infrastructures and collaborating with various teams and clients to deliver reliable voice solutions.

Responsibilities

• Develop and support telephony features within Sarj’s platform alongside the engineering team.
• Connect Voice AI agents to client PBXs, SBCs, SIP trunks, and contact center systems.
• Create custom call flows, manage SIP routing, and implement WebRTC-based human handoffs.
• Lead technical discussions with clients, collect requirements, and manage integration projects through deployment.
• Diagnose and resolve issues related to call quality, signaling, connectivity, and interoperability.

Requirements

• Bachelor’s degree in Software Engineering, Computer Science, or a related discipline.
• Fluency in Arabic and English with strong communication and client interaction skills.
• At least 4 years of experience in SIP/VoIP engineering or telephony integration.
• In-depth knowledge of SIP, RTP, SDP, call routing, and VoIP networking.
• Software development skills for building integrations and working with APIs beyond telephony platform configuration.
• Practical experience with Asterisk, FreeSWITCH, Kamailio, or commercial SBCs.
• Experience with Saudi telecom carriers, WebRTC, and enterprise telephony platforms such as Genesys, Avaya, or Cisco.
• A proactive and hands-on approach to problem-solving and ownership.

What the company offers

• Chance to join a fast-growing, innovative tech startup.
• A collaborative and mission-focused work culture.
• Direct involvement with impactful AI products.

We refresh listings regularly, but some roles close early on the source platform.