7 Nat Traversal Jobs

As a UCaaS Developer, you will be responsible for developing and optimizing WebRTC-based calling solutions using platforms such as Jitsi, Janus, or Zoom APIs. Your expertise in media servers will be key in efficiently handling real-time audio/video streaming. You will implement SIP, VoIP, and RTC protocols to ensure seamless communication. Integrating third-party UCaaS platforms like Twilio, Vonage, and Microsoft Teams will be part of your daily tasks. Your role will also involve designing and maintaining scalable and secure real-time communication applications. You will develop Node.js-based microservices for signaling, session management, and media control. Optimizing latency, bandwidth usage, and call quality to enhance user experience will be crucial. Implementing security best practices for encryption, authentication, and data protection in WebRTC calls will also be within your scope. Troubleshooting and debugging real-time communication issues across various network conditions will be a part of your routine. Collaboration with frontend teams to ensure seamless WebRTC integrations is essential for success in this role. You should possess 3+ years of experience in WebRTC development and media server integration. Your strong expertise in Jitsi, Janus, Zoom SDK, or similar WebRTC platforms will set you apart. Proficiency in Node.js for backend implementation and API development is required. Experience with SIP, VoIP, STUN/TURN, ICE candidates, and signaling protocols is necessary. Hands-on experience with real-time streaming technologies like RTSP, RTP, HLS, or DASH is preferred. Knowledge of Unified Communications (UCaaS) platforms such as Twilio, Vonage, or Microsoft Teams is essential. Familiarity with containerization (Docker, Kubernetes) for deploying RTC applications is desired. Understanding load balancing and scaling strategies for media servers will be beneficial. Knowledge of networking, NAT traversal, and WebRTC optimization techniques will be advantageous. Preferred qualifications include experience with AI-driven noise cancellation, echo reduction, or speech analytics. Knowledge of real-time chat and messaging solutions is a plus. Prior experience with cloud-based media server deployments on platforms like AWS, GCP, or Azure is beneficial. Experience with server-side event handling using Web Sockets or Socket.IO will be an added advantage.,

You should have at least 3 years of experience in software testing, particularly focusing on VPN and WAN optimization products and networking technologies. Your responsibilities will include conducting thorough testing of various VPN protocols like OpenVPN, IPSec, IKEv2, WireGuard, etc. You will also execute end-to-end testing on VPN software across different platforms to ensure compatibility and stability. Furthermore, you will assess VPN performance under varying network conditions, focusing on latency, throughput, and connection stability. Identifying and evaluating potential security vulnerabilities in VPN implementations, testing WAN optimization technologies, and troubleshooting defects related to VPN and WAN optimization connectivity and performance will also be part of your role. Moreover, you will collaborate with development, product management, and other QA teams to provide feedback and contribute to product enhancements. You should have a Bachelor's degree in Computer Science, Information Technology, Engineering, or a related field, along with strong knowledge of VPN technologies, WAN optimization technologies, and networking protocols. Additionally, experience with performance testing tools, security testing techniques, automation testing tools, and strong troubleshooting skills are required for this role. Your ability to document test cases, test plans, results, and defects in detail is crucial to ensure traceability and accountability. Providing detailed reports on test results, issues, and improvements, ensuring transparency in the testing process, is also an essential part of this position.,

As a Unified Communications Engineer at Dassault Systmes in Pune, Maharashtra, you will have the opportunity to unlock your potential with a global leader in Scientific Software Engineering. In this role, you will collaborate with the WW Unified Communications team, specializing in various Collaboration Services such as Cisco Telephony, call center operations, and cloud telephony solutions. Reporting to the APAC UC Manager, you will play a key role in supporting and enhancing the corporate Unified Communication services, including Zoom Phone and Cisco Telephony. Working as part of a diverse team spread across Europe, Asia, and the US, you will be responsible for the daily administration operations, end-user support, and maintenance of the Unified Communications infrastructure within the Dassault Systmes eco-system. Your role will involve installing, configuring, and monitoring new Unified Communication Services, as well as participating in different projects to drive continuous improvements in work processes and methodologies. To succeed in this role, you should have 4 to 6 years of hands-on experience in designing, implementing, and deploying Voice technologies. An engineering degree in Computer Science or IT is a must, along with a strong knowledge of SIP Protocol and Call Flow Analysis using RTMT, translator, or LX tool. Experience in the administration of CUCM, Cisco Voice Gateway, and Cisco CUBE is essential, as well as familiarity with cloud telephony platforms like Zoom Phone, Webex, or Microsoft Teams. Additionally, a good understanding of Firewall, NAT traversal, Certificate, and DNS is required. Experience in managing Sonus SBC and UCCX, as well as Cisco Video technologies, would be advantageous. The ability to work in a multinational environment and excellent communication skills in English are mandatory for this role. Joining Dassault Systmes offers you the opportunity to work for one of the biggest software companies in a culture of collaboration and innovation. You will have opportunities for personal development, career progression, and collaboration with internal users and stakeholders on various projects. As a company committed to building inclusive and diverse teams globally, Dassault Systmes values empowering employees to bring their whole selves to work and fosters a harmonized Workforce of the Future.,

Sr. Software Engr–VoIP (3+ yrs), SIP, RTP, Asterisk/Freeswitch, Kamailio, WebRTC, AWS, Golang, REST APIs, DevOps, MySQL, Linux. Prod-grade VoIP dev exp a must. C2H via TE Infotech (Exotel), Convertible to Permanent, Loc:BLR @ ssankala@toppersedge.com

You will play a key role as a VOIP Engineer at Advantmed, a prominent player in the US Healthcare sector. Your interpersonal skills and ability to connect with colleagues will be crucial in this collaborative environment. You will be responsible for designing, deploying, and managing VoIP systems utilizing platforms such as FreePBX, Genesys Cloud, and RingCentral. Your expertise will be utilized in maintaining and troubleshooting SIP trunks, call routing, and number provisioning. Additionally, you will implement and oversee fax solutions and support call center operations by optimizing IVR, call queues, and call flows with systems like Queue Matrix. Collaboration with cross-functional teams will be essential as you integrate voice systems with CRM, ticketing, and other enterprise applications. You will take the lead or actively participate in Disaster Recovery (DR) and Business Continuity Planning (BCP) related to voice infrastructure. Monitoring system performance, ensuring uptime, and promptly addressing voice traffic issues will be part of your routine. Acting as a liaison for technical escalations with telephony vendors and documenting network topologies and system configurations will also be part of your responsibilities. To excel in this role, you are expected to hold a Bachelor's degree in information technology, Computer Science, or a related field, or possess equivalent experience. With at least 7 years of experience in VoIP Infrastructure management, your expertise in FreePBX administration, Genesys Cloud deployment, RingCentral CX configuration, and other related areas will be invaluable. Strong knowledge of SIP protocols, call routing, telephony concepts, network QoS, NAT traversal, and firewall configuration related to VoIP is required. Your soft skills will also be put to the test, including problem-solving abilities, communication, collaboration, attention to detail, and organization skills. Your ability to work independently and as part of a team will be key to your success. You will need exceptional verbal communication skills, effective task management, analytical and decision-making skills, delegation abilities, and strong interpersonal skills. Are you ready to take on this challenging yet rewarding role as a VOIP Engineer at Advantmed and contribute to the advancement of US Healthcare ,

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 streams, and NAT/firewall behavior. Manage Linux servers for VoIP applications service tuning, logs, security, cron jobs. Handle integrations with CRM, call recording, SIP gateways, SBCs, and media servers. Write and optimize dialplans, IVRs, and call routing logic using Asterisk or Lua/JSON for FreeSWITCH. Analyze PCAPs and SIP traces using Wireshark, sngrep, Homer, or similar tools. Work with networking and DevOps teams to ensure QoS, bandwidth, and latency optimization. Maintain documentation for configurations, deployments, and internal processes. Required Skills & Experience Minimum 2 years of VoIP experience in a production environment. Strong knowledge of SIP, RTP, SRTP, SIP registration, and NAT traversal techniques. Proficiency in Linux system administration (Ubuntu/RHEL) and bash scripting. Practical experience with VoIP monitoring and debugging tools (e.g., sngrep, tcpdump, Wireshark). Understanding of IP networking (TCP/UDP, DNS, DHCP, VLANs, routing). Ability to interpret SIP INVITE, 200 OK, BYE, REGISTER flows like poetry. Good-to-Have (Not Mandatory, but Gold) Familiarity with Kamailio, OpenSIPS, or other SIP proxy/registrar solutions. Experience with VoIP security measures (TLS, SRTP, fail2ban, SIP authentication). Exposure to SIPREC, TTS/STT, or AI-based call analytics. Understanding of number masking, multi-tenant PBX, or contact center integrations. Knowledge of monitoring stacks (Grafana/Prometheus) or VoIP-aware dashboards. DevOps-friendly skills: Docker, Git, CI/CD pipelines. Education Bachelors degree in Computer Science, Information Technology, Electronics, or relevant field.

Software Engr(Backend+VoIP, 3–5 yrs), SIP/SDP/RTP/RTCP, Asterisk/Freeswitch, Kamailio/Opensips,WebRTC,REST APIs,Golang/Ruby/C++, AWS/GCP/Azure, MySQL/Postgres,DevOps (K8s/Terraform/Ansible). C2H via TE Infotech(Exotel), BLR. @ssankala@toppersedge.com