$120,000.00 - $180,000.00 /Yea
Job is located in Westchester, CA.
**MEMBERS ONLY**SIGN UP NOW***. is currently seeking an IP Telephony Engineer with extensive skills and experience to develop, deploy, operate and maintain an advanced VoIP service for a global telecom provider.
The successful candidate will design and build a SIP network using FreeSWITCH, Open SIPS, Kamillo, or similar technologies. They will also work with vendors and internal tech support for customer inquiries to deliver advanced voice solutions. The individual must bring a passion for all aspects of VoIP networking and engineering.
Duties & Responsibilities:
Design, Implement, and maintain VoIP telephony.
Interface with business customers and peer IT teams to develop & analyze requirements and deliver solutions.
Work with carriers to diagnose and fix issues.
Develop process and procedures for implementation and operational support of systems.
Assist in the architecture and implementation of our new platform.
Assist in capacity planning, switch layout design, IP addressing for reliability, performance and quality.
Work to establish and test new carrier connections.
Analyze and identify issues that would hinder integration of databases.
Verify, test and document all call routes.
Skills & Specifications
Strong engineering and operational experience in service provider networks.
In-depth understanding of SIP, RTP and RTCP
Experience configuring and maintaining open source SIP server software including patching, upgrades, logs, debugs and virtual traces.
Team and cooperative work ethic and approach.
In-depth understanding of open source PBX features, configuration, deployment and support (Asterisk or FreeSWITCH).
Knowledge 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
Excellent verbal and written communication skills
**MEMBERS ONLY**SIGN UP NOW***. is the number one provider of international virtual phone numbers and call forwarding services in the world. Founded in 2002, **MEMBERS ONLY**SIGN UP NOW***. is a technology-driven, international telecommunications provider based in Los Angeles, California.
**MEMBERS ONLY**SIGN UP NOW***. leverages “best of breed” technology to empower businesses with state-of-the-art capabilities and services—which is why clients from our first month of operations are still using the service today, 11 years later!
With over 50 employees, a cloud-based communications management system is under development that will change the way businesses handle customer communications.
**MEMBERS ONLY**SIGN UP NOW***. is experiencing off the chart growth. We anticipate even more massive gains in the coming months.
BS in Computer Science, Electrical Engineering, Telecommunications Engineering or equivalent required
Minimum 8 years experience in telecommunications, networking, or programming
Minimum 5 year’s supporting SIP in large / enterprise scale production environments
Experience with FreeSWITCH or Asterisk is required
Expert knowledge on Linux
Experience with G.711, G.729, and Speex
Experience with WebRTC is desirable
This position offers a competitive base salary, plus employee benefits plan (medical, dental, vision) partially paid by company, 401(k) with company matching, PTO, paid company holidays and paid garage parking or MTA allowance.