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Firstly excellent library! I have been trying out the v1.6.8 with a SIP account, the call is picked up correctly but as soon as I try to fetch the audio, I keep getting:
ResourceWarning: Enable tracemalloc to get the object allocation traceback
pyVoIP/RTP.py:341: ResourceWarning: unclosed <socket.socket fd=5, family=AddressFamily.AF_INET, type=SocketKind.SOCK_DGRAM, proto=0, laddr=('0.0.0.0', 17372)>
self.sout = self.sin
I have also tried transcribing the audio using Assembly AI instead of just printing the raw bytes, same issue.
I took a peek into the library on line 341, but not too familiar with RTP so not sure what the issue is.
def start(self) -> None:
self.sin = socket.socket(socket.AF_INET, socket.SOCK_DGRAM)
# Some systems just reply to the port they receive from instead of
# listening to the SDP.
self.sout = self.sin
self.sin.bind((self.inIP, self.inPort))
self.sin.setblocking(False)
r = Timer(0, self.recv)
r.name = "RTP Receiver"
r.start()
t = Timer(0, self.trans)
t.name = "RTP Transmitter"
t.start()
Any suggestions?
The text was updated successfully, but these errors were encountered:
Firstly excellent library! I have been trying out the v1.6.8 with a SIP account, the call is picked up correctly but as soon as I try to fetch the audio, I keep getting:
I have also tried transcribing the audio using Assembly AI instead of just printing the raw bytes, same issue.
I took a peek into the library on line 341, but not too familiar with RTP so not sure what the issue is.
Any suggestions?
The text was updated successfully, but these errors were encountered: