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7.0 - 14.0 years
0 Lacs
karnataka
On-site
The role available is a direct position with our esteemed client where you will lead the design and development of Voice Services over NB-IoT. Your responsibilities will include building and optimizing Voice Gateway signaling components and media session setup, as well as implementing CSFB-SS7 (I1 protocol) for integration with NTN carriers and 3GPP IMS Core. You will be tasked with developing a scalable Voice Gateway Server with CSFB-I1 and SIP/SDP interworking functions, integrating third-party SIP/SDP protocol stacks for seamless operation, and focusing on optimizing voice services for low-bandwidth networks. Additionally, you will support thousands of SIP sessions with interworking to MNO SBC and IMS networks, define and execute test cases, automation, and code coverage with QA teams, and oversee build and deployment processes using Jira, Jenkins, GCR/GKE while ensuring security best practices in development and deployment. Collaboration with platform engineering and cross-functional teams for streamlined development and establishing disciplined processes for timely, high-quality software delivery with robust testing will also be part of your role. Your skills and expertise should include expertise in CSFB, I1, SIP signaling, and 3GPP IMS standards for telephony services, skilled in building SIP-based solutions using SIP stacks for proxy clients and B2B servers, hands-on experience with VoIMS and SIP/IMS network implementation, proficiency in tools like SIPP and FreeSWITCH for test environments, programming expertise in C/C++, Go, or Java with strong SQL schema and query skills, knowledge of stream processing (GCP Pub/Sub, Kafka) and REST API design, experience with scalable distributed systems and horizontal scaling, familiarity with Kubernetes, Docker, Jenkins pipelines, and GCP tools, strong understanding of code quality, performance metrics, and CI/CD tools (e.g., JMeter, Postman, Grafana, Prometheus), and being a collaborative team player with excellent communication skills. The ideal candidate should have 7 to 14 years of proven experience as a Cloud QA Engineer in the software industry, along with a Bachelor's degree in Computer Science, Information Technology, Telecommunications, or a related field. The working model for this position is hybrid, with 3 days a week from the office.,
Posted 4 days ago
5.0 - 9.0 years
5 - 9 Lacs
Gurgaon, Haryana, India
On-site
What You ll Do: Implement core voice capabilities using FreeSWITCH, Kamailio/OpenSIPs, and RTPEngine. Build and optimize SIP call routing logic, RTP media relays , failover mechanisms, and NAT traversal. Develop and manage configurations for scalability, codec negotiation, SIP trunk registration . Implement and test features like call recording, IVR, voicemail, DTMF detection. Monitor live traffic and participate in 24x7 on-call rotation for critical escalations. Collaborate with QA on stress/load testing and with Backend teams on provisioning APIs. Document design decisions, configurations, and troubleshooting runbooks. What Makes You Qualified 5 to 9 years of experience building and operating VoIP systems or CPaaS platforms . Solid expertise with SIP signaling, RTP, and media relay techniques . Strong hands-on with FreeSWITCH, Kamailio/OpenSIPs, RTPEngine . Hands-on experience with Session Border Controller (SBC), Media Servers and WebRTC . Experience in managing telephony infrastructure for uptime, latency, and call quality optimization. Strong systems programming and debugging skills in C/C++ . Good scripting/debugging skills ( Bash, Python, or Lua for FreeSWITCH modules ). Proficiency with diagnostic tools ( Wireshark, tcpdump etc ). Experience working with geographically distributed infrastructure or HA deployments.
Posted 1 month ago
5.0 - 9.0 years
5 - 9 Lacs
Bengaluru, Karnataka, India
On-site
What You ll Do: Implement core voice capabilities using FreeSWITCH, Kamailio/OpenSIPs, and RTPEngine. Build and optimize SIP call routing logic, RTP media relays , failover mechanisms, and NAT traversal. Develop and manage configurations for scalability, codec negotiation, SIP trunk registration . Implement and test features like call recording, IVR, voicemail, DTMF detection. Monitor live traffic and participate in 24x7 on-call rotation for critical escalations. Collaborate with QA on stress/load testing and with Backend teams on provisioning APIs. Document design decisions, configurations, and troubleshooting runbooks. What Makes You Qualified 5 to 9 years of experience building and operating VoIP systems or CPaaS platforms . Solid expertise with SIP signaling, RTP, and media relay techniques . Strong hands-on with FreeSWITCH, Kamailio/OpenSIPs, RTPEngine . Hands-on experience with Session Border Controller (SBC), Media Servers and WebRTC . Experience in managing telephony infrastructure for uptime, latency, and call quality optimization. Strong systems programming and debugging skills in C/C++ . Good scripting/debugging skills ( Bash, Python, or Lua for FreeSWITCH modules ). Proficiency with diagnostic tools ( Wireshark, tcpdump etc ). Experience working with geographically distributed infrastructure or HA deployments.
Posted 1 month ago
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