def test_capabilities(self):
# audio
capabilities = RTCRtpSender.getCapabilities('audio')
self.assertTrue(isinstance(capabilities, RTCRtpCapabilities))
self.assertEqual(capabilities.codecs, [
RTCRtpCodecCapability(
mimeType='audio/opus', clockRate=48000, channels=2),
RTCRtpCodecCapability(
mimeType='audio/PCMU', clockRate=8000, channels=1),
RTCRtpCodecCapability(
mimeType='audio/PCMA', clockRate=8000, channels=1),
])
self.assertEqual(capabilities.headerExtensions, [
RTCRtpHeaderExtensionCapability(
uri='urn:ietf:params:rtp-hdrext:sdes:mid'),
])
# video
capabilities = RTCRtpSender.getCapabilities('video')
self.assertTrue(isinstance(capabilities, RTCRtpCapabilities))
self.assertEqual(capabilities.codecs, [
RTCRtpCodecCapability(mimeType='video/VP8', clockRate=90000),
RTCRtpCodecCapability(mimeType='video/rtx', clockRate=90000),
RTCRtpCodecCapability(mimeType='video/H264',
clockRate=90000,
parameters=OrderedDict(
[('packetization-mode', '1'),
('level-asymmetry-allowed', '1'),
('profile-level-id', '42001f')])),
RTCRtpCodecCapability(mimeType='video/H264',
clockRate=90000,
parameters=OrderedDict(
[('packetization-mode', '1'),
('level-asymmetry-allowed', '1'),
('profile-level-id', '42e01f')])),
])
self.assertEqual(
capabilities.headerExtensions,
[
RTCRtpHeaderExtensionCapability(
uri='urn:ietf:params:rtp-hdrext:sdes:mid'),
RTCRtpHeaderExtensionCapability(
uri=
'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time'
), # noqa
])
# bogus
capabilities = RTCRtpSender.getCapabilities('bogus')
self.assertIsNone(capabilities)
async def __offer(self, request):
"""
Generates JSON Response with a WebRTC Peer Connection of Video Server.
"""
# get offer from params
params = await request.json()
offer = RTCSessionDescription(sdp=params["sdp"], type=params["type"])
# initiate stream
if not (self.__default_rtc_server is
None) and not (self.__default_rtc_server.is_launched):
if self.__logging:
logger.debug("Initiating Video Streaming.")
self.__default_rtc_server.launch()
# setup RTC peer connection - interface represents a WebRTC connection
# between the local computer and a remote peer.
pc = RTCPeerConnection()
self.__pcs.add(pc)
if self.__logging:
logger.info("Created WebRTC Peer Connection.")
# track ICE connection state changes
@pc.on("iceconnectionstatechange")
async def on_iceconnectionstatechange():
logger.debug("ICE connection state is %s" % pc.iceConnectionState)
if pc.iceConnectionState == "failed":
logger.error("ICE connection state failed.")
# check if Live Broadcasting is enabled
if self.__relay is None:
# if not, close connection.
await pc.close()
self.__pcs.discard(pc)
# Change the remote description associated with the connection.
await pc.setRemoteDescription(offer)
# retrieve list of RTCRtpTransceiver objects that are currently attached to the connection
for t in pc.getTransceivers():
# Increments performance significantly, IDK why this works as H265 codec is not even supported :D
capabilities = RTCRtpSender.getCapabilities("video")
preferences = list(
filter(lambda x: x.name == "H265", capabilities.codecs))
t.setCodecPreferences(preferences)
# add video server to peer track
if t.kind == "video":
pc.addTrack(
self.__relay.subscribe(self.config["server"]) if not (
self.__relay is None) else self.config["server"])
# Create an SDP answer to an offer received from a remote peer
answer = await pc.createAnswer()
# Change the local description for the answer
await pc.setLocalDescription(answer)
# return Starlette json response
return JSONResponse({
"sdp": pc.localDescription.sdp,
"type": pc.localDescription.type
})
def test_capabilities(self):
# audio
capabilities = RTCRtpSender.getCapabilities("audio")
self.assertTrue(isinstance(capabilities, RTCRtpCapabilities))
self.assertEqual(
capabilities.codecs,
[
RTCRtpCodecCapability(
mimeType="audio/opus", clockRate=48000, channels=2),
RTCRtpCodecCapability(
mimeType="audio/PCMU", clockRate=8000, channels=1),
RTCRtpCodecCapability(
mimeType="audio/PCMA", clockRate=8000, channels=1),
],
)
self.assertEqual(
capabilities.headerExtensions,
[
RTCRtpHeaderExtensionCapability(
uri="urn:ietf:params:rtp-hdrext:sdes:mid")
],
)
# video
capabilities = RTCRtpSender.getCapabilities("video")
self.assertTrue(isinstance(capabilities, RTCRtpCapabilities))
self.assertEqual(
capabilities.codecs,
[
RTCRtpCodecCapability(mimeType="video/VP8", clockRate=90000),
RTCRtpCodecCapability(mimeType="video/rtx", clockRate=90000),
RTCRtpCodecCapability(
mimeType="video/H264",
clockRate=90000,
parameters=OrderedDict([
("packetization-mode", "1"),
("level-asymmetry-allowed", "1"),
("profile-level-id", "42001f"),
]),
),
RTCRtpCodecCapability(
mimeType="video/H264",
clockRate=90000,
parameters=OrderedDict([
("packetization-mode", "1"),
("level-asymmetry-allowed", "1"),
("profile-level-id", "42e01f"),
]),
),
],
)
self.assertEqual(
capabilities.headerExtensions,
[
RTCRtpHeaderExtensionCapability(
uri="urn:ietf:params:rtp-hdrext:sdes:mid"),
RTCRtpHeaderExtensionCapability(
uri=
"http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time"
),
],
)
# bogus
with self.assertRaises(ValueError):
RTCRtpSender.getCapabilities("bogus")