cervajs at fpf.slu.cz
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 Posted: Sun May 24, 2015 11:58 am Post subject: [asterisk-users] [SOLVED] Re: asterisk 13 webrtc |
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dtlsenable=yes was missing
thank you joshua
Dne 21.5.2015 v 22:53 Marek Cervenka napsal(a):
| Quote: | hi,
is there someone with working asterisk13+chan_sip+SIP.js/sipml5 ?
or is chan_pjsip better supported?
or the recommended way for asterisk is use respoke.io?
my problem with asterisk13+chan_sip+sipml5(the same problem is with SIP.js)
chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer "
sip.conf (important parts)
[vr1a882]
...
nat=force_rport,comedia
canreinvite=no
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
directmedia=yes
transport=wss,ws
dtlsrekey=60
dtlsverify=no
dtlscertfile=/etc/pki/tls/certs/rapidssl.crt
dtlsprivatekey=/etc/pki/tls/private/rapidssl.key
dtlssetup=actpass
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Marek Cervenka
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