I'm having what is probably a simple configuration issue. Calls between two SIP clients (zoiper) are successful.
When I start a call between a WebRTC client (sipml5) and a SIP client (Zoiper) is the connection active, but there is no audio in both directions available.
So, I have latest Asterisk 16, latest Chrome (with Firefox & Chrome Beta the same problem), sipml5 and a local network - no nat or firewall.
Any ideas on what we may be doing wrong? Thank you in advance for any help!
/etc/asterisk/sip.conf
[general]
realm=192.168.11.31;
udpbindaddr=192.168.11.31;
transport=udp
[1060] ; This will be WebRTC client
type=friend
username=1060
host=dynamic
secret=password
encryption=yes
avpf=yes
icesupport=yes
context=default
directmedia=no
transport=udp,ws,wss;
force_avp=yes
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/cert/asterisk.pem
dtlssetup=actpass
rtcp_mux=yes
[6002] ; This will be the legacy SIP client
type=friend
username=6002
host=dynamic
secret=password
context=default
[6001]
type=friend
username=6001
host=dynamic
secret=password
context=default
/etc/asterisk/extensions.conf
[default]
exten => 1060,1,Dial(SIP/1060)
exten => 6002,1,Dial(SIP/6002)
exten => 6001,1,Dial(SIP/6001)
