SIP to OpenAI #8
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Pull Request Overview
This PR adds SIP (Session Initiation Protocol) support to the OpenAI WebRTC library, enabling SIP calls to be bridged to OpenAI's real-time API. The implementation introduces the capability to route audio from SIP callers to OpenAI and responses back to the caller, creating a voice gateway for phone-based conversations with OpenAI.
Key changes:
- Added
ConnectRTPSessionextension method to bridge RTP sessions with WebRTC endpoints - Created a complete SIP-to-OpenAI example application with call management and transcription logging
- Updated documentation to include the new SIP gateway functionality
Reviewed Changes
Copilot reviewed 6 out of 6 changed files in this pull request and generated 6 comments.
Show a summary per file
| File | Description |
|---|---|
src/Extensions/WebRTCExtensions.cs |
Adds RTP session connection methods and fixes a typo in existing documentation |
examples/GetStartedSIP/README.md |
Comprehensive documentation for the SIP gateway example with setup and usage instructions |
examples/GetStartedSIP/Program.cs |
Complete SIP server implementation that bridges calls to OpenAI WebRTC |
examples/GetStartedSIP/GetStartedSIP.csproj |
Project configuration for the SIP example application |
examples/SIPSorcery.OpenAI.WebRTC.Examples.sln |
Solution file updated to include the new SIP example project |
README.md |
Main documentation updated to reference the new SIP gateway example |
| /// <summary> | ||
| /// Connects an RTPSession to a WebRTC end point. The standard use case is to connect the audio packets from an RTP session established by a SIP call | ||
| /// to an OpenAI We. | ||
| /// </summary> |
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The documentation comment is incomplete. 'We.' should be completed to describe what OpenAI component this connects to (e.g., 'WebRTC endpoint').
Suggested change
| /// </summary> | |
| /// <summary> | |
| /// Connects an RTPSession to a WebRTC end point. The standard use case is to connect the audio packets from an RTP session established by a SIP call | |
| /// to an OpenAI WebRTC endpoint. | |
| /// </summary> |
src/Extensions/WebRTCExtensions.cs
Outdated
| /// </summary> | ||
| /// <param name="webRTCEndPoint">The WebRTC end point to connect.</param> | ||
| /// <param name="rtpSession">The RTPSession to connect.</param> | ||
| /// <param name="audioFormat">THe audio format being used for both audio streams.</param> |
There was a problem hiding this comment.
Typo in parameter documentation: 'THe' should be 'The'.
Suggested change
| /// <param name="audioFormat">THe audio format being used for both audio streams.</param> | |
| /// <param name="audioFormat">The audio format being used for both audio streams.</param> |
Co-authored-by: Copilot <[email protected]>
Co-authored-by: Copilot <[email protected]>
Co-authored-by: Copilot <[email protected]>
Co-authored-by: Copilot <[email protected]>
…OpenAI.WebRTC into sip-to-openai
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