I’m trying to configure Asterisk 13 to work with webRTC. I installed it like described here and configured it like described here. When i try to call from a browser with sipML5 to another sip user, i get the following error:
[Jun 30 14:47:29] WARNING[C-0000000f]: chan_sip.c:10807 process_sdp: Rejecting secure audio stream without encryption details: audio 42029 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
When i call from softphone like
blink with two instances from demo-bob to demo-alice, the call rings, can be accepted and audio is working. Only calling from browser (sipML5) to e.g. blink does not work. The certificates seem to be valid (except that its self-signed) in the check via openssl. I also tried to remove encryption from the asterisk config to have unencrypted communication, but the error persisted.
Can someone please tell me what is wrong here?
My sip.conf (other configuration is provided in my linked wiki documentation):
[friends_internal](!) type=friend host=dynamic context=from-internal disallow=all allow=ulaw [demo-alice](friends_internal) secret=demo-alice ; put a strong, unique password here instead [demo-bob](friends_internal) secret=demo-bob ; put a strong, unique password here instead [general] udpbindaddr=0.0.0.0:5060 realm=192.168.1.91 ;replace with your Asterisk server public IP address or host transport=udp,ws,wss tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1  host=dynamic secret=test2 context=from-internal type=friend encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=no disallow=all allow=ulaw dtlsenable=yes ;dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlscafile=/etc/asterisk/keys/ca.crt dtlssetup=actpass rtcp_mux=yes transport=tls,udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets dtlsverify=no ; Tell Asterisk to not verify your DTLS certs  host=dynamic secret=test1 context=from-internal type=friend encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=no disallow=all allow=ulaw dtlsenable=yes ;dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlscafile=/etc/asterisk/keys/ca.crt dtlssetup=actpass transport=tls,udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets dtlsverify=no ; Tell Asterisk to not verify your DTLS certs rtcp_mux=yes
If necessary, i can provide my sip debug log on pastebin, but i wasn’t allowed to paste more than 2 links in one topic
I’m able to start a call from softphone Blink to logged in sipml5 in Chrome, the browser request access to microphone and then the call is terminated. The other way, calling from SipML5 to Blink result in the above stated error.