FreeSwitch Deepgram Voice AI Agent Integration
Budget: $30 – $250 USD
We are building a real-time voice agent using Deepgram's WebSocket API and need an expert in Freeswitch to help us bridge our current call flow to the Deepgram voice agent.
Requirements: - Proven experience with Freeswitch core and module development - Experience with `mod_audio_fork` and `mod_audio_stream` - Deep understanding of SIP/RTP/media flows
What you will do: - Connect our existing Freeswitch server with Deepgram’s WebSocket-based voice agent using `mod_audio_fork` and `mod_audio_stream`, we need both to be configured.
Enable seamless, real-time, bi-directional audio between the caller and the voice agent .
Stream audio to Deepgram in real-time and handle incoming transcription/command messages.
Maintain high availability and low latency across multiple concurrent sessions.
Ensure voice agent can: - Execute in-call commands like: - End the call - Transfer the call to a human agent - Trigger DTMF or SIP-based routing actions - Play custom messages or handle call hold -
Implement and maintain synchronized call recording (merged caller and agent audio in a single file) without disrupting existing call flows - Set up retry/reconnect mechanisms on WebSocket failure - Collaborate with our backend team to ensure correct message routing between Freeswitch and voice agent via secure WebSocket communication - Optimize system performance and provide documentation for deployment and maintenance
Our FreeSWITCH is the latest version.
This project is by bid but we are also looking for a new telecom engineer who does good work at a good price, we need a lot of other things done and our OpenSIPs updated month many other things, if we work well together there will be plenty more work, we are a small startup that is getting ready to launch soon.
Thank you!
Requirements: - Proven experience with Freeswitch core and module development - Experience with `mod_audio_fork` and `mod_audio_stream` - Deep understanding of SIP/RTP/media flows
What you will do: - Connect our existing Freeswitch server with Deepgram’s WebSocket-based voice agent using `mod_audio_fork` and `mod_audio_stream`, we need both to be configured.
Enable seamless, real-time, bi-directional audio between the caller and the voice agent .
Stream audio to Deepgram in real-time and handle incoming transcription/command messages.
Maintain high availability and low latency across multiple concurrent sessions.
Ensure voice agent can: - Execute in-call commands like: - End the call - Transfer the call to a human agent - Trigger DTMF or SIP-based routing actions - Play custom messages or handle call hold -
Implement and maintain synchronized call recording (merged caller and agent audio in a single file) without disrupting existing call flows - Set up retry/reconnect mechanisms on WebSocket failure - Collaborate with our backend team to ensure correct message routing between Freeswitch and voice agent via secure WebSocket communication - Optimize system performance and provide documentation for deployment and maintenance
Our FreeSWITCH is the latest version.
This project is by bid but we are also looking for a new telecom engineer who does good work at a good price, we need a lot of other things done and our OpenSIPs updated month many other things, if we work well together there will be plenty more work, we are a small startup that is getting ready to launch soon.
Thank you!