SIP/VoIP Engineer
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Before you apply
LocationRiyadh, Saudi Arabia
Work modeon-site
Typefull-time
SeniorityMid
Experience4+ yrs
DepartmentEngineering
Company size16+ people
PostedOct 26, 2026 · 1w ago
Verified live1d ago
At a glanceSummarised by Seekless from the posting.
Bachelor's degree in Software Engineering, Computer Science, or a related field
Professional proficiency in Arabic and English
4+ years of experience in SIP/VoIP engineering or telephony integrations
Strong knowledge of SIP, RTP, SDP, call routing, and VoIP networking
Software development experience, with the ability to build integrations and work with APIs beyond configuring existing telephony platforms
Hands-on experience with Asterisk, FreeSWITCH, Kamailio, or commercial SBCs
Experience working with Saudi telecom carriers, WebRTC, and enterprise telephony platforms such as Genesys, Avaya, or Cisco
A proactive, hands-on approach to problem-solving and ownership
Skills
SIP
RTP
SDP
VoIP
Asterisk
FreeSWITCH
Kamailio
WebRTC
Genesys
Avaya
Cisco
About Us
Built in KSA and leveraging Silicon Valley expertise, we’re building innovative solutions in the realm of Voice AI and AI agents. As we continue our rapid growth trajectory in an exciting and dynamic market, we’re searching for creative talent to join our team.
Role Overview
As a SIP/VoIP Engineer, you’ll help build Sarj’s telephony capabilities and own custom integrations connecting our Voice AI agents to client infrastructure. You’ll work closely with our Operations and Engineering teams and B2B and B2G clients to develop, deploy, and troubleshoot reliable voice solutions.
Key Responsibilities
•
Build and maintain telephony capabilities within Sarj’s platform alongside our engineering team.
•
Integrate Voice AI agents with client PBXs, SBCs, SIP trunks, and contact-centre platforms.
•
Develop custom call flows, SIP routing, and WebRTC-based human handoffs.
•
Lead technical client discussions, gather requirements, and drive integrations through to production.
•
Troubleshoot call quality, signalling, connectivity, and interoperability issues.
Requirements
•
Bachelor’s degree in Software Engineering, Computer Science, or a related field.
•
Professional proficiency in Arabic and English, with strong communication and client-facing skills.
•
4+ years of experience in SIP/VoIP engineering or telephony integrations.
•
Strong knowledge of SIP, RTP, SDP, call routing, and VoIP networking.
•
Software development experience, with the ability to build integrations and work with APIs beyond configuring existing telephony platforms.
•
Hands-on experience with Asterisk, FreeSWITCH, Kamailio, or commercial SBCs.
•
Experience working with Saudi telecom carriers, WebRTC, and enterprise telephony platforms such as Genesys, Avaya, or Cisco.
•
A proactive, hands-on approach to problem-solving and ownership.
Why Join Us?
•
Opportunity to be part of a high-growth, innovative tech start-up.
•
Collaborative, mission-driven culture.
•
Hands-on experience working on impactful AI products.
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