6 Sngrep Jobs

As a skilled FreeSWITCH Developer with deep experience in OpenSIPS and/or Kamailio, you will be joining the core VoIP engineering team at our company. Your role will involve architecting, building, and maintaining robust SIP-based communication systems. If you are excited about solving complex signaling challenges and scaling VoIP infrastructure, this opportunity is perfect for you. **Key Responsibilities:** - Build and maintain scalable VoIP services using FreeSWITCH. - Configure and manage SIP proxies (OpenSIPS, Kamailio) for routing, load balancing, and NAT traversal. - Develop and fine-tune dialplans, call flows, and scripts in Lua, Python, or JavaScript. - Troubleshoot signaling/media i...

Cyara is the world's leading Automated CX Assurance Platform provider, enabling leading brands across the globe to build better customer experiences faster. Through automated testing and monitoring, Cyara accelerates the delivery of flawless customer journeys across digital and voice channels while reducing the risk of customer-facing defects. Every day the most recognizable brands, including Airbnb and NAB trust Cyara to deliver customer smiles at scale. Our promise is Customer Smiles - Delivered at Scale, and as a member of Cyara's team, you'll be given the opportunity to bring that mission to fruition alongside our amazing community of fun-loving forward thinkers. Interested to find out m...

Job Description 7+ years of hands-on VoIP engineering experience, with a strong focus on carrier SIP trunking and interoperability. Expert-level knowledge of SIP signaling (INVITE, BYE, REGISTER, re-INVITE, UPDATE, PRACK, REFER, etc.), SDP negotiation, RTP/RTCP, and SIP headers. Proven ability to troubleshoot complex SIP interoperability issues across different carrier platforms and SBCs.Strong experience with Session Border Controllers (SBCs) and SIP proxies (e.g., Ribbon, AudioCodes, Oracle/Acme Packet, OpenSIPS,Kamailio, etc.). Proficiency with diagnostic tools: Wireshark, sngrep, Homer/HEP, SIPp, Empirix, etc. Solid networking knowledge: IP, DNS, subnetting, VLANs, routing protocols. Str...

Role Overview: You have 7+ years of hands-on VoIP engineering experience, with a specific focus on carrier SIP trunking and interoperability. Your expert-level knowledge includes understanding SIP signaling such as INVITE, BYE, REGISTER, and other related protocols, SDP negotiation, RTP/RTCP, and SIP headers. You must have proven abilities to troubleshoot complex SIP interoperability issues across various carrier platforms and Session Border Controllers (SBCs). Your experience with SBCs and SIP proxies like Ribbon, AudioCodes, Oracle/Acme Packet, OpenSIPS, Kamailio will be beneficial in this role. Proficiency in diagnostic tools such as Wireshark, sngrep, Homer/HEP, SIPp, and Empirix is requ...

As a skilled FreeSWITCH Developer with experience in OpenSIPS and/or Kamailio, you will join our core VoIP engineering team. In this hands-on role, you will be responsible for architecting, building, and maintaining robust SIP-based communication systems. If you are excited about solving complex signaling challenges and scaling VoIP infrastructure, this is the right opportunity for you. Your responsibilities will include building and maintaining scalable VoIP services using FreeSWITCH, configuring and managing SIP proxies (OpenSIPS, Kamailio) for routing, load balancing, and NAT traversal. You will also develop and fine-tune dialplans, call flows, and scripts in Lua, Python, or JavaScript. T...

About the Role We're seeking a skilled and proactive VoIP Engineer with at least 2 years of hands-on experience in SIP-based systems, Linux server management, and VoIP troubleshooting. If you've ever found beauty in a perfect SIP handshake or debugged NAT hell like a champ, youll feel right at home here. Youll be responsible for maintaining, optimizing, and expanding our IP telephony and unified communication infrastructureworking across Asterisk, FreeSWITCH, Kamailio, and other open-source tools. Key Responsibilities Deploy, configure, and manage VoIP infrastructure (Asterisk, FreePBX, Kamailio, OpenSIPS, or similar). Monitor and troubleshoot call quality issues, SIP signaling, RTP media st...