VoIP/SIP Integration Engineer

VoIP/SIP Integration Engineer (SBC, RTP/SDP, Realtime TTS/ASR)PART-TIME HOURLY BASED WORK,ObjectiveDesign and implement a bidirectional SIP/RTP integration layer between enterprise SIP trunks and Realtime voice AI platforms (e.g. **ElevenLabs Realtime**, **Retell AI**), enabling low-latency, high-fidelity audio exchange for interactive voice applications.Scope of Work1. Core IntegrationEstablish a two-way audio session between a **SIP trunk** and **Realtime AI voice endpoints** (ElevenLabs, Retell AI, or similar).Configure and normalize **SIP/SDP** negotiation (offer/answer, m=audio, a=rtpmap, fmtp, directionality).Ensure **codec compatibility** (G.711 µ-law/A-law Opus 48kHz) with transcoding where necessary.Implement **B2BUA** or **RTP relay** via **FreeSWITCH**, **Asterisk**, or **MediaProxy**.Support **SRTP/TLS encryption** and **NAT traversal** (rport, symmetric RTP, ICE-Lite, STUN/TURN).Manage **reINVITE/UPDATE** cycles and session refresh for long-running audio sessions.2. SBC / Media ControlConfigure **OpenSIPS** or **Kamailio** as a **Session Border Controller (SBC)** for:- SIP normalization, routing, authentication, and topology hiding.- RTP anchoring and media proxying.- Header rewriting, dialog management, and security enforcement.Optimize **RTP flow** for minimal jitter and latency.Implement **session timers**, **keep-alives**, and **failover logic**.Capture and analyze SIP and RTP using **Wireshark**, **sngrep**, or **Homer**.3. Audio & Media Layer ConfigurationIntegrate with **ElevenLabs Realtime** and **Retell AI Realtime Voice** APIs:- Understand supported **protocols** (WebSocket, SIP/TLS, RTP, PCM16).- Configure **sampling rates** (8kHz, 16kHz, 24kHz, 48kHz) and **audio formats** (WAV, PCM16, µ-law, Opus).- Manage latency, jitter, and synchronization in both directions.Build **transcoding pipelines** with FreeSWITCH or external RTP relays.Tune:- ptime, **jitter buffers**, and **packetization** for low latency.- **VAD**, **AGC**, and silence suppression where required.- Channel configuration (mono/stereo) and bit-depth conversions.4. Testing & ValidationConduct full **end-to-end test calls** verifying:- Audio flow between SIP trunk SBC Realtime AI endpoint.- Codec negotiation and SRTP compliance.- Latency, jitter, and MOS quality scoring.Analyze SIP dialogs and RTP streams via **Wireshark/sngrep/tcpdump**.Produce **call-flow diagrams**, **SDP traces**, and **debug logs** for documentation.Deliverables**Working integration**: SIP trunk SBC/B2BUA Realtime voice API (verified via bidirectional test call).**Final configuration package**:- SBC and B2BUA config files.- Codec/port/security matrix.- TLS/SRTP settings and NAT traversal documentation.**Supporting documents**:- SIP flow diagrams and media topology.- Troubleshooting procedures and reference captures.Optional: **Docker/Ansible deployment** for reproducibility.QualificationsDeep knowledge of **SIP**, **SDP**, **RTP/SRTP**, **ICE/DTLS**, and **NAT traversal**.Advanced experience with **FreeSWITCH**, **Asterisk**, **OpenSIPS**, or **Kamailio**.Experience integrating SIP trunks with **Realtime AI platforms** (e.g. ElevenLabs, Retell AI, Deepgram).Proficiency with **Wireshark**, **sngrep**, **tcpdump**, and **SIP ladder analysis**.Familiarity with **TLS/SRTP profiles**, ptime, **RFC2833 DTMF**, **Symmetric RTP**, and **rport**.Bonus:- Experience with **WebRTCSIP gateways** (Janus, Pion, Mediasoup).- Familiarity with **Python** or **** for WebSocketRTP bridges.- Understanding of **audio pipelines**, **transcoding**, and **TTS/ASR systems**.Preferred ExperienceImplemented Realtime AI integrations using **ElevenLabs** or **Retell AI**.Built or maintained **SIP-to-AI voice bridges** or **AI-driven IVR systems**.Tuned media stacks for **