Ice support problem? (WebRTC)


First, sorry for any mistakes in English. I’m using a translator to help me.
I know there are hundreds of topics around the internet with people struggling with WebRTC and the asterisk and great guides help (as done by the user navaismo) , who can guide us a lot, but I’m a few days trying to get the work in WebRTC my platform and unsuccessful. I tried more complex scenarios involving the PSTN, but decided to try to solve a simpler scenario first.

Looking for any tips that can assist me in solving:

No audio and error (“Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd.”) on Chrome with sipML5 client (last release) and HTTP connect (not secure).

Asterisk 11.11 with srtp and with libs libuuid libuuid-devel uuid uuid-devel installeds and icesupport enabled.

I will add the greatest amount of information and if it goes missing something, feel her hand to ask.


[general] rtpstart=19000 rtpend=21000 icesupport=yes



allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
udpbindaddr=             ; IP address to bind UDP listen socket to ( binds to all)
;localnet=set_your_localnet_here   ;i.e.
;allow=g729                   ; First disallow all codecs
allow=ulaw                     ; Allow codecs in order of preference
allow=alaw                     ; Allow codecs in order of preference
language=en                    ; Default language setting for all users/peers

host=dynamic ; Allows any host to register
secret=password ; The SIP Password for SIP.js
username=5005 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=5005 ; 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=from-internal; risk 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

Provide the sip debug and the Chrome JS debug.

Thanks for the reply!

JS Chrome Debug:
SIP Debug:

Indeed Asterisk isn’t sending the ice ufrag and ice pwd. Is the latest version or is the RC? Which OS?

Run this command and paste the output:

[size=85] rpm -qa | grep uuid && echo && ldd /usr/lib/asterisk/modules/ && echo && ls -lha /lib/libuu* && cat /etc/issue[/size]