docs: add rtp-engine

docs/contributor-guide/transports/README.md

@@ -3,14 +3,16 @@
 We have some types of transports:
 
 - [WebRTC](./webrtc.md)
-- [SIP](./sip.md)
-- [RTMP](./rtmp.md)
+- [RTP-Engine](./rtp-engine.md)
 - [Whip-Whep](./whip-whep.md)
 
 Bellow transport will be implemented in next version:
 
+- [RTMP](./rtmp.md)
 - [Media over Quic](https://quic.video/)
 - [SRT](https://www.haivision.com/products/srt-secure-reliable-transport/)
 - [HLS](https://en.wikipedia.org/wiki/HTTP_Live_Streaming)
 
-If you don't find the transport you need, you can implement it by yourself by implementing the `Transport` traits. Please refer to [Architecture](../architecture.md) for more info.
+If you don't find the transport you need, you can implement it by yourself by
+implementing the `Transport` traits. Please refer to
+[Architecture](../architecture.md) for more info.

docs/contributor-guide/transports/rtp-engine.md

@@ -0,0 +1,3 @@
+# RTP Engine
+
+RTP Engine is implemented in 'packages/transport_rtpengine/'

docs/getting-started/quick-start/rtmp.md

@@ -1,20 +1 @@
 # RTMP
-
-You can use RTMP to publish to our media server. Currently, we don't support transcoding, so you need to publish with the same resolution and codec that correspond to your viewers.
-
-Preferred codecs:
-
-- Video: H264, baseline profile, bitrate 2500kbps
-- Audio: AAC, bitrate 128kbps
-
-URL: `rtmp://{gateway}/live/{token}`
-
-Demo configuration for OBS:
-
-![Config OBS](../../imgs/demo-rtmp-config.png)
-
-Pregenerated token for default secret and room `demo`, peer `publisher`:
-
-```jwt
-eyJhbGciOiJIUzI1NiJ9.eyJyb29tIjoiZGVtbyIsInBlZXIiOiJydG1wIiwicHJvdG9jb2wiOiJSdG1wIiwicHVibGlzaCI6dHJ1ZSwic3Vic2NyaWJlIjpmYWxzZSwidHMiOjE3MDM3NTIzMzU2OTV9.Gj0uCxPwqsFfMFLX8Cufrsyhtb7vedNp3GeUtKQCk3s
-```

docs/getting-started/quick-start/rtp-engine.md

@@ -0,0 +1,68 @@
+# RTP Engine
+
+The atm0s-media-server supports RTP transport for integration with external
+applications such as SIP Gateways. We have also developed a simple SIP Gateway
+that supports basic call flows, including outgoing and incoming calls:
+[atm0s-sip-gateway](https://github.com/8xFF/atm0s-media-sip-gateway).
+
+## Integration Guide
+
+Our RTP media endpoint and HTTP APIs enable integration with external SIP
+Gateways. The integration process involves three main steps:
+
+1. Creating an RTP token
+2. Creating an SDP Offer for outgoing calls or an SDP Answer for incoming calls
+3. Deleting the RTP media endpoint after the call ends
+
+For more detailed API documentation, please visit:
+
+`http://gateway/token/ui` and `http://gateway/rtpengine/ui`
+
+### RTPEngine APIs
+
+The RTPEngine APIs provide endpoints to manage RTP media endpoints for SIP
+integration. All endpoints are prefixed with `/rtpengine/`.
+
+#### Authentication
+
+Most endpoints require Bearer token authentication. Include your token in the
+Authorization header:
+
+```
+Authorization: Bearer <your-token>
+```
+
+#### Endpoints
+
+##### Create Offer
+
+- **POST** `/rtpengine/offer`
+- **Summary**: Creates an RTPEngine endpoint with an offer
+- **Security**: Bearer token required
+- **Response**:
+  - Status: 200
+  - Content-Type: application/sdp
+  - Schema: SDP string
+
+##### Create Answer
+
+- **POST** `/rtpengine/answer`
+- **Summary**: Creates an RTPEngine endpoint with an answer
+- **Security**: Bearer token required
+- **Request Body**:
+  - Content-Type: application/sdp
+  - Required: true
+  - Schema: SDP string
+- **Response**:
+  - Status: 200
+  - Content-Type: application/sdp
+  - Schema: SDP string
+
+##### Delete Connection
+
+- **DELETE** `/rtpengine/conn/{conn_id}`
+- **Summary**: Deletes an RTPEngine connection
+- **Response**:
+  - Status: 200
+  - Content-Type: text/plain; charset=utf-8
+  - Schema: string