Location: Los Angeles, CA The Voice Over Internet Protocol (VoIP) Engineer will be responsible for designing, analyzing and implementing projects with the Voice Carrier infrastructure. The VoIP Engineer will work closely with the operations and engineering teams to deploy our SIP-based telephony solutions to a multitude of vertical markets. The position requires working with both hardware (in our data centers) and software to provide a Carrier Grade Voice cloud-based solution while interoperating with customer requirements. Responsibilities *The VoIP Engineer will work with internal and external NOC staffs to identify and mitigate escalated (tier 3) call completion issues *The VoIP Engineer will work individually and in a team environment with operations and engineering to drive new initiatives and complete existing projects for VoIP PBX offerings *Supports the existing network infrastructure and its need for security, stability and effective use of existing and future technology *Provides documentation and planning for all VoIP and infrastructure work *Delivers new projects using strong interpersonal, written and oral communications skills *Conducts research into networking issues and products as required *Ensures the stability, integrity, source control, change control and schedule for both existing and new products *Works with operations to resolve any ongoing implementation issues related to gateways and existing legacy systems. Requirements: VoIP Engineer Must have experience with: *SIP (OpenSIPS or FreeSWITCH) 3-5 years with VoIP 3-5 years supporting SIP G.711 in large / enterprise scale production environments 3-5 years with Unix Linux networking *Strong engineering and operational experience in service provider networks *In-depth understanding of SIP, RTP and RTCP interworking *Experience configuring and maintaining Open Source SIP Server Software including patching, upgrades, logs, debugs and traces (OpenSIPS or FreeSWITCH) *In-depth understanding of Open Source PBX features, configuration, deployment and support *Understanding of network server policies, routing network elements and collocation modeling *Ability to perform SIP debugs, capture and analyze SIP and RTP to isolate and resolve issues *Understanding of MOS Score, Jitter, Delay, Echo, Gaps, and how to troubleshoot these occurrences *Hands-on experience configuring routers and switches to maximize SIP / RTP pass through *Experience working directly with carriers and trunk providers *In-depth understanding of switching for a telephony standpoint into the PSTN *Must possess strong organizational, verbal, written and documentation skills *In-depth understanding of legacy telephony and legacy PBX interworking *Experience with IP PBX Trunking Media Gateways in NAT Environments Please send your resume to: voicejobs@gogii.net