[Freeswitch-users] rtp-timer-name=soft and WebRTC sound quality in Chrome

Hi, I've found very bad WebRTC sound quality in Chrome (in opposite to FF) and found that this issue can be fixed on FS side via some params in Sofia profile: rtp-timer-name=soft - no any sound with this param as is inbound-late-negotiation=true - good sound quality in Chrome with this and previous But these two params has no effect for outbound calls: bad quality without both and no any sound with them Can anybody explain why and how to fix this? -- WBR, Eugene Prokopiev