Backend VOIP Developer

Zetexa Global Private Limited is a dynamic startup founded by visionary leaders from esteemed institutions like IIT and IIM. We are at the forefront of reshaping global Travel Tech, Consumer IoT, and M2M applications, with a focus on both B2B and B2C domains. Our mission transcends borders as we engineer innovative solutions that elevate digital experiences. Join us on a transformative journey as we redefine industries and revolutionize digital interactions.

As a Backend VoIP Developer at Zetexa, you will be responsible for designing, developing, and managing scalable VoIP solutions. This includes working with SIP services, virtual number solutions, and softphone development. You will collaborate closely with cross-functional teams to ensure seamless integration, troubleshooting, and maintaining a high level of service availability while leveraging open-source technologies and networking tools. The role also involves configuring switches, working with AudioCodes gateways, and providing detailed technical documentation.

Key Responsibilities:

• 6+ years of experience in Design and development of scalable solutions to deliver reliable SIP services to a multitude of distributed endpoints including on premise Call Processing Servers, SIP proxy servers, handsets, and soft-phones.
• Hands on Development experience on the virtual number solution and softphone solutions.
• Enhance and maintain existing code and participate in design reviews and code reviews with other team members.
• Create custom provisioning tools to automate configuration of voice stack
• Participate in projects intended to improve infrastructure, including the integration of new systems and software
• Draft technical documentation
• Facilitate knowledge transfer to deployment and support teams
• Work with the Production Services and Support leaders towards keeping Service Availability above target.
• Work experience on class 4 and class 5 switches.
• Able to configure the billing solution as well sip trunking in switches.
• Responsible for finalization of DID vendors and take forward for integration.
• Knowledge of SIP, RTP, SRTP, RTCP, SDP, audio codecs and video codecs
• Knowledge of AudioCodes gateways and SBCs
• Knowledge of Microsoft Direct Routing.
• Work experience on the ope source switches or inhouse developed switches.
• Expertise in Session Initiation Protocol, PSTN Connectivity, VoIP, and Contact Center Solution
• Experience in troubleshooting and supporting AudioCodes SBC, Microsoft Teams, Polycom
• Responsible for detailed documentation of the solution in the form of LLD/HLD
• Resolution ownership of technical challenges occurring during/after the implementation
• The successful candidate would be able to independently develop and manage complex VoIP environment.

Must-Have Skills:

SIP protocol and its use in VOIP Services

SIP proxies and/or SIP PBXs preferably open source products like Kamailio (OpenSER), FreeSWITCH, Asterisk, and FreePBX etc.

Networking with Switches, Routers, Firewalls, IPSs and Security Appliances

Programming experience with PHP / Perl / JAVA / C and custom modules with open source products

Relational Databases ( MySQL), JDBC / ODBC

Linux server operations, deployment and performance monitoring and shell scripting

Join us in our mission to reshape eSIM narratives and redefine digital communication. Be a valuable contributor to our dynamic team, fostering collaborative success and contributing to the transformation of industries and digital experiences!

To apply, please submit your resume and a well-crafted cover letter detailing your relevant experience and enthusiasm for our vision to [Confidential Information].