I have this chat system that's using SIP for voice transmission (no phones, only browser to browser in the same server). The sound goes just fine in both ends, but asterisk gives a warning twice (one for each client probably):
[Mar 11 09:01:27] WARNING[614][C-00000002]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
[Mar 11 09:01:33] WARNING[614][C-00000002]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
The keys where generated with:
./usr/src/asterisk-*/contrib/scripts/ast_tls_cert -C IpAddress -O "Company" -d /etc/asterisk/keys
sip.conf
file:
[general]
udpbindaddr=0.0.0.0:5060
realm=IpAddress
transport=udp,ws,wss
videosupport=no
icesupport=yes
allowguest=no
allwaysreject=yes
rtptimeout=30
My peers are set in sip.conf
as:
[000001]
host=dynamic
secret=000001
context=from-internal
type=friend
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=no
disallow=all
allow=ulaw,alaw,gsm,all
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
And I'm using sipML5 for the client. Any insights to clear the warnings?