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Python VoIP library of the AI age

VoIP 최신버젼 다운로드

최종 버전 다운로드 (.zip)

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# Python VoIP

Async VoIP Python library for the AI age.

> [!WARNING]
> This library is in early development and may contain breaking changes. Use with caution.

## Sponsors

[![Sponsors](https://django.the-box.sh/sponsors/codingjoe/VoIP.svg)](https://github.com/sponsors/codingjoe)

## Usage

To get started, you will need a SIP account. One is usually included with ISP.
Check your ISP's documentation or router for details.

You will need a SIP AOR (URI), which looks like this:

```INI
sip:USER:PASSWORD@SIP_SERVER
```

> [!NOTE]
> This library defaults to **UDP transport on port 5060** for `sip:` URIs, which
> is the most widely supported configuration. To use TCP or TLS, add an explicit
> `transport` parameter, e.g. `sip:user@host;transport=TCP` or
> `sips:user@host` (SIPS always uses TLS on port 5061).

### CLI

A simple echo call can be started with:

```console
uvx 'voip[cli]' sip sip:alice:********@sip.example.com echo
```

Each command supports an optional `--dial` argument to initiate an
outbound call instead of waiting for an inbound one.

To dial a number, say a message, and hang up automatically:

```console
uvx 'voip[cli]' sip sip:alice:********@sip.example.com say sip:+15551234567@sip.example.com "Your package has arrived."
```

You can also talk to a local agent (needs [Ollama]):

```console
uvx 'voip[cli]' sip sip:alice:********@sip.example.com agent --initial-prompt "Hi, I am looking for a Mr. Ron, first name Mo?"
```

### MCP

The `voip` package ships a ready-made [Model Context Protocol (MCP)](https://modelcontextprotocol.io/) server
that exposes tools to make phone calls on your behalf to any MCP client.

```json
{
  "mcpServers": {
    "VoIP": {
      "type": "stdio",
      "command": "uvx",
      "args": [
        "voip[mcp]",
        "mcp"
      ],
      "env": {
        "SIP_AOR": "sip:number:password@example.com"
      }
    }
  }
}
```

### Python API

```console
uv add voip[audio,ai,pygments]
```

Subclass `TranscribeCall` and override `transcription_received` to handle results.
Pass it as `session_class` when answering an incoming call:

```python
import asyncio
import dataclasses

from voip.ai import TranscribeCall
from voip.sip.protocol import SIP
from voip.sip.types import SipURI
from voip.sip.transactions import InviteTransaction
from voip.rtp import RealtimeTransportProtocol
from faster_whisper import WhisperModel

@dataclasses.dataclass(kw_only=True, slots=True)
class TranscribingCall(TranscribeCall):
    def transcription_received(self, text) -> None:
        print(text)

class TranscribeInviteTransaction(InviteTransaction):
    def invite_received(self, request) -> None:
        self.ringing()
        self.answer(
            session_class=TranscribingCall,
            stt_model=WhisperModel("kyutai/stt-1b-en_fr-trfs", device="cuda"),
        )

async def main():
    loop = asyncio.get_running_loop()
    _, rtp_protocol = await loop.create_datagram_endpoint(
        RealtimeTransportProtocol,
        local_addr=("0.0.0.0", 0),
    )
    await loop.create_connection(
        lambda: SIP(
            rtp=rtp_protocol,
            aor=SipURI.parse("sip:alice:********@example.com"),
            transaction_class=TranscribeInviteTransaction,
        ),
        host="sip.example.com",
        port=5060,
    )
    await asyncio.Future()

asyncio.run(main())
```

For raw audio access without transcription, subclass `AudioCall` and override
`audio_received(self, audio: np.ndarray)` instead.

[ollama]: https://ollama.com/

VoIP

VoIP 최신버젼 다운로드

오늘의 추천 프로젝트

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VoIP

VoIP 최신버젼 다운로드

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