Staff Software Engineer
Join Our Back-end Engineering Team!
We are looking for Staff Software Engineer to set the technical direction for telecommunications platform. You will own the architecture of how calls enter, route through, and leave our system: SIP signaling at the edge (Kamailio / OpenSIPS), media and call control (FreeSWITCH, Asterisk, RTPengine), the Python services that bridge voice with our AI inference pipeline, and the carrier-facing integrations underneath.
What will need to do:
- Own end-to-end telecom architecture: SIP edge (Kamailio/OpenSIPS), media & call control (FreeSWITCH), Python services bridging voice to AI, carrier integrations
- Scale platform to 10,000+ concurrent calls with tight latency/SLOs
- Design HA/disaster recovery: anycast SIP, active/active media, geographic failover
- Set engineering standards for telecom team (testing, load testing, code review, on-call)
- Mentor senior/mid engineers, raise technical bar
- Define voice quality observability & SLOs (MOS, jitter, packet loss, latency)
- Lead carrier strategy: SIP trunk vendors, redundancy, LCR, STIR/SHAKEN
- Partner with AI/product to keep latency budgets honest
Requirements:
- 8+ years engineering, 4+ years in telecom/VoIP in production
- Technical leadership: design docs, mentoring, cross-team decisions
- Deep production experience with FreeSWITCH + Kamailio OR OpenSIPS (custom modules, routing scripts, at scale)
- Strong SIP/RTP/SRTP/SDP/ICE; debug via pcap and SIP trace
- Operated voice platform at 1,000+ concurrent calls (knows diff to 10k)
- Strong Python; can work in Lua/Go/C as needed
- Production integration with PSTN/SIP trunk providers (Twilio, Telnyx, etc.), DID management, LCR
- Hands-on HA/failover for real-time systems
- Production Kubernetes (GKE/EKS) and Terraform
- Language Proficiency: English at B2-C1 level, Russian at B2 level
- Location: any, except Russia and Belarus
It will be a plus:
- WebRTC at scale (ICE, TURN, media servers)
- RTPengine, Jitsi, mediasoup
- Low-latency AI voice pipelines (streaming ASR/TTS, barge-in)
- STIR/SHAKEN, robocall mitigation
- Compliance: SOC2, HIPAA, GDPR, PCI for voice
- Multi-region active/active voice
- Experience hiring and growing a telecom team
Why Join us
At Mad Devs, you will be part of a company that values creativity, strategic thinking, and continuous improvement. You will work on meaningful projects that impact millions of users worldwide alongside a talented and supportive team. By joining our team, you will have the opportunity to make a real impact, grow your career, and contribute to developing cutting-edge solutions in the world of backend engineering.
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Staff Software Engineer
Прямой работодатель Mad Devs( maddevs.io )
Бишкек, Кыргызстан
Сеньор
Информационные технологии
- Разработка
- Backend
- Python
- ML/AI
- Voice assistants
16 июля
Удаленная работа
Опыт работы более 5 летот 3 500 до 5 500 ₽
Работодатель Mad Devs
Короткая ссылка: geekjob.ru/hiZz
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Описание вакансии
Join Our Back-end Engineering Team!
We are looking for Staff Software Engineer to set the technical direction for telecommunications platform. You will own the architecture of how calls enter, route through, and leave our system: SIP signaling at the edge (Kamailio / OpenSIPS), media and call control (FreeSWITCH, Asterisk, RTPengine), the Python services that bridge voice with our AI inference pipeline, and the carrier-facing integrations underneath.
What will need to do:
- Own end-to-end telecom architecture: SIP edge (Kamailio/OpenSIPS), media & call control (FreeSWITCH), Python services bridging voice to AI, carrier integrations
- Scale platform to 10,000+ concurrent calls with tight latency/SLOs
- Design HA/disaster recovery: anycast SIP, active/active media, geographic failover
- Set engineering standards for telecom team (testing, load testing, code review, on-call)
- Mentor senior/mid engineers, raise technical bar
- Define voice quality observability & SLOs (MOS, jitter, packet loss, latency)
- Lead carrier strategy: SIP trunk vendors, redundancy, LCR, STIR/SHAKEN
- Partner with AI/product to keep latency budgets honest
Requirements:
- 8+ years engineering, 4+ years in telecom/VoIP in production
- Technical leadership: design docs, mentoring, cross-team decisions
- Deep production experience with FreeSWITCH + Kamailio OR OpenSIPS (custom modules, routing scripts, at scale)
- Strong SIP/RTP/SRTP/SDP/ICE; debug via pcap and SIP trace
- Operated voice platform at 1,000+ concurrent calls (knows diff to 10k)
- Strong Python; can work in Lua/Go/C as needed
- Production integration with PSTN/SIP trunk providers (Twilio, Telnyx, etc.), DID management, LCR
- Hands-on HA/failover for real-time systems
- Production Kubernetes (GKE/EKS) and Terraform
- Language Proficiency: English at B2-C1 level, Russian at B2 level
- Location: any, except Russia and Belarus
It will be a plus:
- WebRTC at scale (ICE, TURN, media servers)
- RTPengine, Jitsi, mediasoup
- Low-latency AI voice pipelines (streaming ASR/TTS, barge-in)
- STIR/SHAKEN, robocall mitigation
- Compliance: SOC2, HIPAA, GDPR, PCI for voice
- Multi-region active/active voice
- Experience hiring and growing a telecom team
Why Join us
At Mad Devs, you will be part of a company that values creativity, strategic thinking, and continuous improvement. You will work on meaningful projects that impact millions of users worldwide alongside a talented and supportive team. By joining our team, you will have the opportunity to make a real impact, grow your career, and contribute to developing cutting-edge solutions in the world of backend engineering.
Специализация
Информационные технологии
Разработка
Backend
Python
Отрасль и сфера применения
ML/AI
Voice assistants
Уровень должности
Сеньор
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