Update to GStreamer v1.20

[danisla]

Notable changes:

• Build gstreamer 1.20 from monorepo
• Include gstwebrtcbin patch to add rtx-time to SDP per #7 comment.
• Enable do-nack on transceiver to reduce stuttering/freezing on packet loss, thanks to @xhejtman.
• Set RTP cap aggregate-mode to zero-latency to reduce RTP packetization overhead.
• coturn-web now uses "?transport=tcp" param for turn URI to enable traversal of networks with UDP blocked.

New dependencies:

• Need to apt-get install x264 in target images to use software encoder.

TODO:

• Verify x264enc encoder option still works.
• Verify nvh264enc encoder option still works.

[xhejtman]

rtph264pay_caps.set_value("aggregate-mode", "zero-latency")
rtph264pay_caps.set_value("rtcp-fb-ccm-fir", True)

perhaps set these two as well

[xhejtman]

are you seeing significant number of packet loss? my observation is that 10% of packet loss is way too much for smooth experience. does it change with bitrate? and are you seeing that nacks/pli is actually sent? this is available in chrome://webrtc-internals in statistics. Also you should see that only 2-3 key frames are sent in total.

[ehfd]

@xhejtman Feels much better after improving my internet (resetting my router firmware).
It's smooth, with only very occasional tearing for a very brief time which doesn't impact the overall experience.

[xhejtman]

well, we cannot do much if connection sucks ;) perhaps we can optionaly enable FEC? But latency will suffer and I am not sure it actually helps a lot. But Geforce Now stream offers also FEC enabled stream. I wonder if and how we can offer several stream options?