### CURRENT SITUATION ###
I am working on a project related to streaming implemented with GStreamer, and I've run into an issue that I would like to solve.
It's a webrtc audio streaming data flow. I have two clients, one is C# based on a PC and the other is in Android on phone.
In WebRTC both client share their connection information in form of SDP and ICE candidates.
SDP describes the device capability regarding whether they have send, recv and both capabilities.
At C# side, I use the follwing pipeline to send receive audio stream from remote client:
webrtcbin name=webrtcbin audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=97 ! webrtcbin
I send this offer to Android Device.
On Android side I use:
webrtcbin name=webrtc openslessrc ! queue ! audioconvert ! audioresample ! audiorate ! queue ! opusenc ! rtpopuspay ! webrtc
Here I first set remote offer SDP and based on it create the answer.
I wrote android specific start/receive stream pipeline, and provided the interface to set remote offer SDP and create answer.
From PC to Android Audio streaming is working fine and I am able hear the voice signals sent by PC
### PROBLEM ###
I am unable to send AUDIO from Android to PC.
I can provide error log file, etc.
### NEEDED ###
I need expertise to debug this - guidance on how to solve this issue and possibly to write a bit of code, but it shouldn't be much.
Please bid only if you are expert in this and you have extensive experience in dealing with such streaming issues.