VOIP Engineer 

As a Senior SIP Engineer, you must take complete ownership of supporting all VoIP infrastructure, debugging issues related to specific servers or software, or remote clients such as SIP devices (both virtual such as soft-phone or WebRTC client, and physical such as a desk phone or an on-premise PBX), and providing fixes.

Key Skills:

Docker products, Docker swarm, Freeswitch, Kubernetes, Lua, MYSQL , PostgreSQL, VoIP protocols SIP, Voice over IP (VoIP)

Roles & Responsibilities: 

* Support customers during EST timezone during critical releases or emergency incidents 5+ yrs of supporting global VoIP services and/or applications on cloud-based servers.

* Expertise in SIP call flow analysis and debugging Expertise in setup and maintaining SIP-based monitoring, debugging, and alerting services

* Experience scripting call flow, dialplan, and custom routing with FreeSwitch using LUA and XML

* Experience in debugging Kamailio and Freeswitch-based applications is a must Good problem-solving and analytical skills Excellent written and verbal communication

* Experience working with open-source projects

* Exposure to SIP Carrier Integration

* Advanced Experience with cloud media infrastructure (load balancers, gateways, SBCs, STUN, TURN)

* Advanced Knowledge of all modern VoIP protocols/platforms including (SIP, RTP stack & SDP, RTCP, TCP, UDP, SIP, HTTPS, SSL/TLS)

* Working Knowledge of Network Usage Scenarios and understanding of Internet Traffic with the general flow of Routing, Ports, Firewalls, and Packet Flow

* Experience with Open Source VoIP applications such as Kamailio, OpenSIPS, FreeSWITCH, RTPEngine or RTPProxy, and open source tools such as Wireshark, sngrep, and Homer Experience with High Availability, geographically redundant, and load-balanced applications of FreeSwitch and Kamailio, with Call Center functionality, Presence, and SIP Registrations

* Working FreeSWITCH carrier experience to handle 10,000+ concurrent calls

* Good knowledge of RTP Proxy and routed audio conferences concept where media would flow via free switch RTP Proxy FreeSWITCH - Listening to all events generated by Kamailio or events from FreeSwitch such as those exposed using esl/mod_event_socket

* Experience with any load testing tools for FreeSwitch/Kamailio to ensure scalability and acceptable minimum load tolerances, such as automated dialplan testing, calls per second testing (CPS), transcoding validation, and playback verification

* Working understanding and knowledge of codecs such as PCMU, G722, and Opus and how to efficiently transcode codecs, or optimize and prevent call quality issues by signal updates for optimized codec renegotiation Ability to create and maintain geo-redundant and highly available and optimized MySQL and/or PostgresSQL based database infrastructure (with working understanding of vertical and horizontal sharding)

* Excellent troubleshooting skills and working knowledge of IPTables, Fail2ban, wireshark, tcpdum, sipp

* Understanding of SIP security such as acceptable or unacceptable requests, and how to respond/honeypot

* Experience with containers and automation tools such as Kubernetes, Docker, Ansible, Jenkins, Nomad.

* Advanced working knowledge and experience to set up and maintain a geographically redundant and highly scalable SQL backend

* Working experience implementing and testing HA scenarios and automated fail-over tests

* Experience with CloudFlare products (such as WebSockets, SIP, and RTP over Magic Transit)

* Experience working with AWS, GCS Kubernetes is a plus

* Experience with Linux, open source tools and shell scripting

* Experience with video conferences and video transcoding is a plus Develop and maintain automation of code deployment (AWS, k8s, CI/CD, etc.)

* Experience with AMQP protocol with Kamailio and FreeSwitch (such as RabbitMQ / Kafka)

* Experience with real-time RTP processing for transcription and predictive response handling using internal applications or third party services