Talku.AI is a headless Voice AI layer for B2B. Seamlessly connect advanced AI agents to your existing telecom infrastructure via standard SIP URIs. Automate reservations, support, and inbound calls without ripping out your current IP-PBX or CRM.
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Maker
📌
Hi Product Hunt! 👋 I'm Yusuf, the founder of Talku.AI.
I built Talku.AI because I realized a massive friction point in the B2B telecom space: Businesses (like hotels and clinics) want to automate their inbound calls with Voice AI, but they do not want to abandon their existing IP-PBX setups, phone numbers, or CRMs to do it.
Most AI telephony platforms force you into a walled garden. With Talku.AI, we took a "Headless" approach. It’s a Bring-Your-Own-PBX (BYOPBX) architecture powered by ultra-low latency WebRTC and advanced LLMs.
Here is how it works:
🔌 SIP Native: You don't need to change your telecom provider. Just route calls to the unique SIP URI we provide (sip:yourbusiness@talku.ai).
⚡ Ultra-Low Latency: Built on top of LiveKit and Groq, the AI responds in milliseconds, making conversations feel completely natural.
🔗 Webhook & n8n Ready: We don't force you into our CRM. When a call finishes, the AI extracts structured JSON data (caller intent, dates, names) and fires a standard POST webhook. You can catch this via n8n, Zapier, or your custom endpoint to update your existing CRM instantly.
🛡️ Zero PCI/HIPAA Friction: Because the AI acts as an assistant and your PBX handles the core routing, you can keep sensitive card transactions out of the AI layer.
Whether you are automating hotel reservations, qualifying leads, or handling after-hours support, Talku.AI acts as the intelligent edge of your current phone system.
I would love to hear your feedback on our architecture, use cases, or any questions you have about integrating Voice AI into legacy telecom systems! Let me know what you think in the comments. 👇
Cheers!
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How does the SIP integration handle latency on really long customer calls, and is there a usage-based pricing model or do you charge per seat?
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Maker
@ercan1811695 Hello, its usage-based pricing. There is no limit on sip calls
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Dropped a SIP URI into our existing FreePBX and the agent picked up inbound calls without any weird audio issues. Latency felt close enough to a human handoff that customers probably won't notice.
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How does the latency compare to having a human pick up, especially on longer customer support calls where context matters a lot?
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Plugged it into our FreePBX over SIP and the agent picked up on the second ring without any weird codec issues. Latency feels close enough to a human to not spook callers.
How does the SIP integration handle latency on really long customer calls, and is there a usage-based pricing model or do you charge per seat?
Dropped a SIP URI into our existing FreePBX and the agent picked up inbound calls without any weird audio issues. Latency felt close enough to a human handoff that customers probably won't notice.
How does the latency compare to having a human pick up, especially on longer customer support calls where context matters a lot?
Plugged it into our FreePBX over SIP and the agent picked up on the second ring without any weird codec issues. Latency feels close enough to a human to not spook callers.