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 softphone or WebRTC client and physical such as a desk phone or an onpremise PBX) and providing fixes.
- Support customers during EST timezone during critical releases or emergency incidents 5 yrs of supporting global VoIP services and/or applications on cloudbased servers.
- Expertise in SIP call flow analysis and debugging Expertise in setup and maintaining SIPbased monitoring debugging and alerting services
- Experience scripting call flow dialplan and custom routing with FreeSwitch using LUA and XML
- Experience in debugging Kamailio and Freeswitchbased applications is a must Good problemsolving and analytical skills Excellent written and verbal communication
- Experience working with opensource 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 loadbalanced applications of FreeSwitch and Kamailio with Call Center functionality Presence and SIP Registrations
- Working FreeSWITCH carrier experience to handle 10000 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/modeventsocket
- 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 georedundant 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 failover 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 realtime RTP processing for transcription and predictive response handling using internal applications or third party services
high availability and load balancing,sip security,mysql and/or postgressql database infrastructure,voip,code deployment automation,amqp protocol with kamailio and freeswitch,sip,docker,debugging,mysql,cloud media infrastructure,postgresql,kubernetes,sql backend setup and maintenance,voice over ip (voip),sip call flow analysis,freeswitch scripting with lua and xml,open source voip applications,voip protocols sip,voip infrastructure support,voip protocols/platforms knowledge,kamailio and freeswitch debugging,rtp proxy and audio conferences,containers and automation tools,freeswitch,real-time rtp processing,network usage scenarios,lua