Web RTC issue with asterisk 13.22

I am using ctxsip Java script client for web phone. It got registered with asterisk but immediately went on UNREACHABLE. Same result on asterisk 13.22 and asterisk 15.6. Following are some results of my testing

1:- I am using https for communication by using self signed certificates.
2:- It works good with pjsip but went to UNREACHABLE on sip
3:- I tried with SML5 web-phone and it got registered and calls are landing on it but it performs quite slow.
4:- I am using it on LAN i.e. client browser and asterisk server are on LAN
5:- Tested on chrome and Firefox.

Need guidance about it also if there some other free web phone , please let me know