Freeswitch Elevenlabs Voice Agent Integration

@farooqtahir2013

With over a decade of diverse experience as a Network, Cybersecurity, VoIP and System Engineer, I'm confident in my ability to make your Freeswitch Elevenlabs Voice Agent Integration project a resounding success. I possess prodigious comprehension of Freeswitch core and module development, in particular, `mod_audio_fork` and `mod_audio_stream` which are essential for your desired integration. This mastery extends to substantial knowledge about SIP/RTP/media flows - all vital for connecting your existing Freeswitch server with Elevenlab's WebSocket-based voice agent. Moreover, my ultimate goal is handling any project I undertake not just as an individual task but as part of an ongoing professional relationship. As a small startup with plans to scale rapidly, you need someone who can grow with you and deliver consistent results. If chosen for this project, I promise an unwavering commitment to high-quality work at a competitive price. And as we go forward from this milestone together, know that I am fully capable of handling everything from OpenSIPs updates to any other telecom engineering tasks that may arise along the way. Choosing me would not only ensure pitch-perfect delivery for this project but also lay a foundation for numerous future collaborations.

@AZTURK1

Hi there, I understand you need an expert to integrate your FreeSWITCH server with Elevenlab's WebSocket-based voice agent for seamless real-time voice interactions. With 7+ years in telecom and voice systems, I specialize in FreeSWITCH core and module development, including advanced audio streaming and SIP/RTP flow management. - Configure and optimize `mod_audio_fork` and `mod_audio_stream` for bi-directional, low-latency audio streaming - Implement real-time command handling for call control functions like hang-up, transfer, DTMF, and custom messaging - Ensure high availability and robust performance for concurrent sessions - Provide ongoing support and collaborate on your telecom stack including OpenSIPs upgrades and more **Skills:** ✅ FreeSWITCH core & module development ✅ SIP/RTP/media flow expertise ✅ mod_audio_fork & mod_audio_stream configuration ✅ Real-time streaming & voice command handling ✅ Linux server & telecom stack **Certificates:** ✅ Microsoft® Certified: MCSA | MCSE | MCT ✅ cPanel® & WHM Certified CWSA-2 I’m ready to start promptly and help ensure your voice agent integration is smooth and scalable. Let’s discuss your existing call flow details and timeline to kick off this collaboration. Could you share details about your current FreeSWITCH call flow architecture and any existing audio processing customizations? Best regards,

@arodrigue

I have ElevenLabs running as a fully functional AI agent over pure SIP, which is more reliable and faster to deploy than Stream Socket.

@usamae

Hey, I've worked with FreeSwitch before and have set up mod_audio_fork for streaming audio to external services. The ElevenLabs WebSocket integration is the same pattern - fork the audio stream to their WS endpoint, handle incoming transcription/command events, and trigger SIP actions based on those commands. For the in-call stuff (end call, transfer to agent, DTMF routing), I'd drive those via ESL or FreeSWITCH's HTTP API. Keeping audio pipeline tight is key to low latency - no unnecessary transcoding steps. The ongoing work sounds interesting too. Happy to discuss what else you have lined up after this is done. What does your current dialplan/ESL setup look like? - Usama

@SkillCrafts

I’ve architected low-latency voice AI pipelines using FreeSWITCH’s `mod_audio_fork` to stream raw PCM data to high-performance WebSocket endpoints. My recent work involves bridging SIP-based telephony with ElevenLabs’ "Turbo v2.5" model, where I successfully mitigated jitter and optimized buffer management to maintain the natural prosody of synthesized speech. By focusing on the direct exchange of raw audio frames between FreeSWITCH and the ElevenLabs API, I can ensure your agent achieves the sub-second response times required for fluid interaction without the lag typical of standard REST-based integrations. My approach involves configuring `mod_audio_fork` to capture bi-directional media, routed through a lightweight Python or Node middleware acting as the WebSocket bridge. This bridge will handle ElevenLabs’ binary framing and stream the resulting audio back into the FreeSWITCH session using `mod_raw` for minimal-latency injection. I will implement robust "barge-in" logic, utilizing VAD markers to instantly clear the ElevenLabs output buffer the moment the user interrupts. I’ll also optimize the transcoding layer to ensure high-fidelity audio is preserved while minimizing the CPU overhead on your FreeSWITCH instance. Are you handling the conversation orchestration within the same middleware, or is that already decoupled in your stack? I’m also curious if you have a specific latency threshold or a preferred cloud region for the media server to minimize RTT to ElevenLabs. I am open to a brief chat or a call to walk through your current dialplan and see how we can most efficiently integrate this real-time stream into your existing architecture.

@Hemantt841

Thank you for considering me for the Freeswitch Voice Agent Integration project. I was excited to see your need for an expert in Freeswitch to bridge your call flow with Elevenlab's voice agent. With over 7 years of software development experience, I have a deep understanding of Freeswitch core and module development. Specifically, I have worked extensively with `mod_audio_fork` and `mod_audio_stream`, ensuring seamless audio handling in real-time SIP/RTP/media flows. Here's how I plan to approach this project: - Configure `mod_audio_fork` and `mod_audio_stream` to connect Freeswitch with Elevenlab's WebSocket API - Implement bi-directional audio for real-time communication between the caller and voice agent - Stream audio and handle transcription/command messages effectively - Maintain high availability and low latency for multiple concurrent sessions - Enable in-call commands like call termination, transfer to human agent, DTMF/SIP routing, custom messages, and call hold I have successfully completed a similar project for a telecom client where I integrated Freeswitch with a third-party API, resulting in improved call handling efficiency and customer satisfaction. To ensure a successful collaboration, could you provide more details on the expected call volume and any specific security requirements for the voice agent integration? This will help me tailor the solution to y