I finally managed to get it working somehow.
WebRTC to SIP is translated by oversip (
http://oversip.net/) and yate is patched in several aspects:
- proper SDP attributes handling
- bug 0000367 fixed
- ICE-Lite (rfc5245) sipport added
- Fake rtcp-mux (rfc5761) support added
Now i can call from Google Chrome with sipml5 to Linksys sip phone, connected to yate.
After more testing i'll prepare patches, and may be, one day, they will accepted by Null team...
If someone is interesting in current project status, please see
https://github.com/vir/yate, branch
ice.