webrtc - 尝试使用 gstreamer webrtc 进行流式传输时，“on-negotiation-needed”如何工作？

问题描述

webrtc 管道如何获取有关其对等点的任何信息？

这就是我假设 on_negotiation_needed 回调的作用？

def start_pipeline(self):
        self.pipe = Gst.parse_launch(PIPELINE_DESC)
        self.webrtc = self.pipe.get_by_name('sendrecv')
        **self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)**
        self.webrtc.connect('on-ice-candidate', self.send_ice_candidate_message)
        self.webrtc.connect('pad-added', self.on_incoming_stream)
        self.pipe.set_state(Gst.State.PLAYING)

我看到它有 on_negotiation_needed 回调，但不清楚元素变量来自哪里？我看了这里： http: //blog.nirbheek.in/2018/02/gstreamer-webrtc.html和这里：https ://github.com/centricular/gstwebrtc-demos我仍然对这个谈判的工作方式感到困惑? 据我了解，有 2 个（或更多）对等点，它们都必须连接到信令服务器，然后其中一个必须创建要约。

我在信号服务器上等待来自（我假设）gstreamer webrtcbin 的消息：

print (websocket.remote_address)
#get message from client
message = await asyncio.wait_for(websocket.recv(), 3000)

当管道启动时出现此错误：

('192.168.11.138', 44120)
Error in connection handler
Traceback (most recent call last):
  File "/usr/local/lib/python3.6/dist-packages/websockets/protocol.py", line 674, in transfer_data
    message = yield from self.read_message()
  File "/usr/local/lib/python3.6/dist-packages/websockets/protocol.py", line 742, in read_message
    frame = yield from self.read_data_frame(max_size=self.max_size)
  File "/usr/local/lib/python3.6/dist-packages/websockets/protocol.py", line 815, in read_data_frame
    frame = yield from self.read_frame(max_size)
  File "/usr/local/lib/python3.6/dist-packages/websockets/protocol.py", line 884, in read_frame
    extensions=self.extensions,
  File "/usr/local/lib/python3.6/dist-packages/websockets/framing.py", line 99, in read
    data = yield from reader(2)
  File "/usr/lib/python3.6/asyncio/streams.py", line 672, in readexactly
    raise IncompleteReadError(incomplete, n)
asyncio.streams.IncompleteReadError: 0 bytes read on a total of 2 expected bytes

The above exception was the direct cause of the following exception:

Traceback (most recent call last):
  File "/usr/local/lib/python3.6/dist-packages/websockets/server.py", line 169, in handler
    yield from self.ws_handler(self, path)
  File "signaling_server.py", line 34, in signaling
    message = await asyncio.wait_for(websocket.recv(), 3000)
  File "/usr/lib/python3.6/asyncio/tasks.py", line 358, in wait_for
    return fut.result()
  File "/usr/local/lib/python3.6/dist-packages/websockets/protocol.py", line 434, in recv
    yield from self.ensure_open()
  File "/usr/local/lib/python3.6/dist-packages/websockets/protocol.py", line 646, in ensure_open
    ) from self.transfer_data_exc
websockets.exceptions.ConnectionClosed: WebSocket connection is closed: code = 1006 (connection closed abnormally [internal]), no reason

标签： webrtcgstreamer

我不能说 Python（不幸的是，不能让 GStreamer 的 Python 绑定在 Windows 上工作），但是，演示可以从 C# 工作（我刚刚检查过）。

首先，您应该使用浏览器连接到https://webrtc.nirbheek.in/，并获取“我们的 id”值。

您的 Python Gstreamer 应连接到 wss://webrtc.nirbheek.in:8443，并使用浏览器中的 Id 值。

浏览器将从 GStreamer 获取测试图像流，GStreamer 应用程序将从浏览器获取网络摄像头图像。

HTH，汤姆

这是一个屏幕截图：

webrtc - 尝试使用 gstreamer webrtc 进行流式传输时，“on-negotiation-needed”如何工作？

问题描述

解决方案

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