Thanks for your help in advance 😃
has anyone used aiortc for a proxy server? or care to sanity check that it makes sense to do that? my goal is to manage separate connections with streaming users (in which case the server is receiving) and viewing users (in which case the server is sending), and be able to write logic re: who receives what. if anyone has even a super simple example of just forwarding video received from one connection to another, that would be super helpful.
@jlaine Is there any example you can point me for a simple streaming speech to text ; I am struggling to do that following the https://github.com/aiortc/aiortc/tree/main/examples/server ; where on the server side, I am trying to add speech to text api call and sending it back via data channel ; any blog , direction will be appreciated
Hi Everyone,
I hope you are well
I have this issue with audio tracks in which some of the audio frames/packets are missing and not sent to the user or encoder.
More details in GitHub issue:
Any help is appreciated
i'm streaming audio, it says connected, but I hear no audio on the iPhone
Hi. I've been working in a webrtc streaming and everything It's working fine. However, I want to send videostreamtrack to youtube. I looked into the aiortc examples and what I'm trying to do is something similar to this
from av import VideoFrame
from aiortc.contrib.media import MediaRecorder
from aiortc.mediastreams import AudioStreamTrack, VideoStreamTrack
import av
import numpy as np
import asyncio
def run(coro):
return asyncio.get_event_loop().run_until_complete(coro)
class RTCServer(VideoStreamTrack):
def __init__(self,):
super().__init__()
self.image = np.random.randint(
0, 255, (640, 360, 3),
dtype='uint8')
async def recv(self):
pts, time_base = await self.next_timestamp()
image = np.random.randint(
0, 255, (640, 360, 3),
dtype='uint8')
av_frame = VideoFrame.from_ndarray(image)
av_frame.pts = pts
av_frame.time_base = time_base
return self.frame
def terminate(self):
try:
if not (self.stream is None):
self.stream.release()
self.stream = None
except AttributeError:
pass
rtc_video = RTCServer()
url_youtube = 'rtmp://a.rtmp.youtube.com/live2/'
url_youtube += 'SECRETE-KEY'
recorder = MediaRecorder(url_youtube, format='flv')
recorder.addTrack(rtc_video)
recorder.addTrack(AudioStreamTrack())
run(recorder.start())
run(asyncio.sleep(2))
run(recorder.stop())
I would like to be able to separate the datachannels using decorators like
@pc.on('pingchannelmessage')
def ping_channel(message):
#do something
@pc.on('tempchannelmessage')
def temp_channel(message):
#do somethingCurrently I do this:
@pc.on("datachannel")
def on_datachannel(channel):
@channel.on("message")
def on_message(message):
print(f"Message is {message}")
if isinstance(message, str) and message.startswith("ping"):
#Do something
if isinstance(message, str) and message.startswith("Temp"):
#Do something
async def on_track(track): ?? what to do here ???
Hello I am looking to implement the the "server" example of aiortc with the heroku hosting service. I have managed to clone the "server" example that is on the GitHub repo in the examples folder onto my local machine and run it on my local network successfully. I am just stumped on actually implementing this into the cloud. Thank you.
I'm using the "server" example
Initially, the parameter "transform == edges" ( VideoTransformTrack(
relay.subscribe(track), transform=params["video_transform"]
))
But when you click on the button in the browser, I want to change this parameter to "transform == cartoon"
How can I do this without breaking the connection?
Initially, the parameter "transform == edges" ( VideoTransformTrack(
relay.subscribe(track), transform=params["video_transform"]
))
But when you click on the button in the browser, I want to change this parameter to "transform == cartoon"
How can I do this without breaking the connection?
i am trying to send frames from a server python instance to client instance. Here i am trying to construct frames locally and send them over MediaStreamTrack but when i receive the frames in my client instance its only loading few frames and looping over. Has anyone faced this issue ?
How can we use google's speech to text API with aiortc (audio being received over webrtc). Since I have tried using it but aiortc recv() function is never called when trying to use google's code with aiortc. Even somehow if I manage to use google's api , the audio coming from aiortc becomes extremely stuttered.
Hey, all. I'm trying to port the client.js script in the aiortc examples to Python, so I'll be able to run two Python scripts in parallel and stream the video (from a file instead of the webcam) from one to the other. I seem to be missing something in my translation, though and the ICE connection is never fully established (gets hung at WAITING). I'm wondering if anyone has done anything similar and could offer some suggestions?