title: webrtc configuration to reduce sent traffic
body: <p>I am developing the function of audio video calls using webrtc technology. I faced with case when one of the participants is in an unstable network (mobile network). In this case, the participant’s audio and video starts to freeze, there is a delay, etc. I think that to solve this problem it is necessary to configure the application so that it sends less traffic.</p>
  
  <p>Please tell me what webrtc configurations exist to reduce sent traffic?</p>