Asterisk Webrtc DTLS-SRTP policy

I work with Asterisk 12 and Webrtc ( is use sip.js) . When Call is answered by Chrome browser (caller is zoiper) , the call imediatelly hangup and shows error

[Aug  4 10:45:16] WARNING[30235][C-0000001f]: res_rtp_asterisk.c:1667 dtls_srtp_setup: Could not set policies when setting up DTLS-SRTP on '0x7ff22802dff0'
[Aug  4 10:45:16] WARNING[30235][C-0000001f]: res_rtp_asterisk.c:3906 ast_rtp_read: RTP Read error: Unspecified.  Hanging up.

my pears is here

[1060] ; This will be WebRTC client
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=1060 ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS

[6003] ; This will be the legacy SIP client

Can anybody help me?