Senior SIP / Kamailio / Asterisk / WebRTC Telecom Architect & Prototype Consultant

We are an advanced CPaaS / CCaaS company already operating large-scale telecom infrastructure and products. We are NOT looking for a junior VoIP engineer or somebody to install a PBX. We are looking for a highly experienced telecom expert with deep hands-on expertise in: SIP Kamailio Asterisk WebRTC Mobile VoIP applications Softphones Telecom security High-scale SIP infrastructure The purpose of this engagement is to help us validate and improve the architecture of a next-generation communication platform. This role is a mix of: High-level architecture consulting Technical validation Security review Prototype implementation We do NOT want pure consulting only. One of the biggest issues with traditional consultants is that they often recommend architectures without taking responsibility for whether they truly work in real-world environments. Because of this, we want the selected candidate to also build and demonstrate parts of the architecture through a working prototype/MVP to validate the recommendations technically. What We Are Building: We are building a secure, scalable telecom infrastructure that supports: SIP extensions WebRTC browser connectivity Softphones Mobile VoIP applications Public internet SIP registration Real-time communications at scale The architecture revolves around: Kamailio Asterisk WebRTC Secure SIP infrastructure Mobile SIP connectivity FreePBX integration/support where relevant MOST IMPORTANT REQUIREMENT — SECURITY Security is the most critical aspect of this project. We are looking for somebody who truly understands how to expose SIP infrastructure to the public internet securely. This includes real-world experience with: SIP DDoS mitigation SIP flood attacks SIP registration attacks Extension enumeration attacks Brute-force prevention RTP protection TLS/SRTP Kamailio hardening SBC architecture SIP rate limiting Fraud prevention Geo filtering Cloudflare Spectrum or similar telecom protection layers Secure WebRTC deployment Secure mobile SIP registration Carrier-grade telecom protection If your experience is mostly theoretical or limited to basic PBX setups, this project is not a fit. Scope of Work Phase 1 — Independent Architecture Proposal Before exposing our existing infrastructure, architecture, or internal implementations, we want the selected candidate to independently propose and explain how they would design a modern, secure, scalable telecom platform based on the requirements below. The goal of this phase is to evaluate: Architecture thinking Security mindset Scalability understanding SIP/WebRTC expertise Real-world operational experience Decision-making process The candidate should present their proposed approach for: SIP architecture Asterisk role/design WebRTC connectivity Mobile SIP connectivity Softphone registration Authentication flows NAT traversal Media handling Scaling strategy SBC/security architecture DDoS protection Public internet exposure protection High availability/failover Monitoring/logging strategy At this stage, we intentionally do NOT want to influence the candidate by exposing our current infrastructure or technical decisions. We first want to evaluate independent thinking and technical maturity. Phase 2 — Discovery, Validation & Prototype After the initial architecture evaluation, the selected candidate will gain visibility into parts of our current infrastructure, assets, constraints, and goals. The objective of this phase is to: Compare architecture approaches Validate assumptions Identify weaknesses or risks Recommend improvements Build or assist in building a working prototype/MVP The prototype should help validate: Secure SIP registration WebRTC connectivity Softphone connectivity Mobile SIP connectivity Secure signaling/media flow Authentication/security layers SIP protection mechanisms Scalability assumptions The purpose is to validate architecture decisions through real implementation — not only diagrams or theoretical consulting. Phase 3 — Documentation & GitHub Provide: Clean documentation Infrastructure diagrams Security recommendations GitHub repository with prototype/code examples Deployment guidance Architecture decision explanations Required Experience You MUST have strong real-world experience with: Kamailio Asterisk SIP internals RTP/SRTP WebRTC TURN/STUN SIP over WebSockets Telecom security Carrier-grade SIP routing Large SIP deployments Production VoIP systems Linux networking Docker/Kubernetes telecom deployments (advantage) Important — Candidate Evaluation Before engaging with a candidate, we will conduct detailed technical discussions. We are looking for: Excellent English communication Strong responsiveness Deep technical understanding Real production experience Ability to challenge ideas constructively Strong architecture thinking Security-first mindset Practical implementation capability We will ask many technical and architectural questions before moving forward. Long-Term Opportunity This may evolve into a long-term collaboration involving: Product architecture Telecom infrastructure optimization Security consulting Scalability planning Production deployment guidance To Apply Please include: Your experience with similar telecom systems you designed WebRTC/mobile VoIP experience Security-related SIP infrastructure experience Previous scaling challenges you solved GitHub examples or technical references Your thoughts on secure SIP architecture exposed to the public internet Only highly experienced telecom professionals should apply.