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

[Eugene Prokopiev]

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