Feat/rtp transport for SIP

[marverlous811]

Pull Request

Description

I added the RTP transport for the SIP call—it is a transport for processing the raw RTP packet without Webrtc. I also added a server using the ng-control protocol, such as an RTP engine, for request calling from the SIP server.

Checklist

• Simple Ng Control Server
• Rtp Transport
  • Process RPC from Ng Control
  • Generate Sdp
  • Transmit Rtp packet
  • Encode audio data
  • End Call
• using configuration param for setting. Currently, I'm fixing all settings.
• Gateway route to best node
• Secure
  • Token
  • Fixed IP

Additional Notes

Some problems I faced:

• I generate the answer SDP by fixing the media session and using the public IP of the current node. But I think it is another way to generate a more generic SDP

• For the transmitted RTP packet flow, I will convert the UDP data to an RTP packet and send it to the Media Endpoint. Is my way okay?

[codecov]

Codecov Report

Attention: Patch coverage is 9.40841% with 1271 lines in your changes missing coverage. Please review.

Project coverage is 39.94%. Comparing base (b63e5cf) to head (a49b745).

Files	Patch %	Lines
packages/media_codecs/src/opus/opus_wrap.rs	0.00%	376 Missing ⚠️
packages/transport_rtpengine/src/transport.rs	0.00%	190 Missing ⚠️
packages/media_runner/src/worker.rs	0.00%	91 Missing ⚠️
packages/media_codecs/src/pcma.rs	42.30%	90 Missing ⚠️
bin/src/ng_controller.rs	0.00%	81 Missing ⚠️
packages/protocol/src/protobuf/cluster_gateway.rs	0.00%	60 Missing ⚠️
packages/transport_rtpengine/src/worker.rs	0.00%	59 Missing ⚠️
packages/protocol/src/transport/rtpengine.rs	0.00%	37 Missing ⚠️
bin/src/server/gateway/local_rpc_handler.rs	0.00%	32 Missing ⚠️
bin/src/http/api_token.rs	0.00%	30 Missing ⚠️
... and 19 more

Additional details and impacted files

@@            Coverage Diff             @@
##           master     #359      +/-   ##
==========================================
- Coverage   42.81%   39.94%   -2.87%     
==========================================
  Files         134      149      +15     
  Lines       14534    15890    +1356     
==========================================
+ Hits         6223     6348     +125     
- Misses       8311     9542    +1231     

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[giangndm]

@marverlous811

• I generate the answer SDP by fixing the media session and using the public IP of the current node. But I think it is another way to generate a more generic SDP

You can use local-ip-address lib to get current node IP. May be with sip we only need single default IP

• For the transmitted RTP packet flow, I will convert the UDP data to an RTP packet and send it to the Media Endpoint. Is my way okay?

You need transcode it to opus, which is used to send over cluster

[giangndm]

NodeJs sample is moved to separated repo:

https://github.com/8xFF/atm0s-media-sip-call-sample