Hello,
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.
rtp.conf
[general]
rtpstart=19000
rtpend=21000
icesupport=yes
stunaddr=stun.l.google.com:19302
sip.conf
[general]
context=guest
transport=udp,ws,wss
rtcachefriends=yes
allowguest=yes
limitonpeers=yes
callcounter=yes
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
udpbindaddr=0.0.0.0 ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
;externhost=set_your_externhost_here
externrefresh=150
;localnet=set_your_localnet_here ;i.e. 10.0.1.0/255.255.255.0
disallow=all
;allow=g729 ; First disallow all codecs
allow=gsm
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
callcounter=yes
limitonpeers=yes
callevents=yes
useragent=Digital-Merge_UA
realm=107.170.172.69
;nat=force_rport,comedia
[5005]
type=friend
host=dynamic ; Allows any host to register
secret=password ; The SIP Password for SIP.js
type=friend
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
nat=force_rport,comedia