Hi, I am looking for a little bit of help from the community if possible please? I’m experimenting with Chrome (JSSIP), WebRTC and Asterisk 13.2.0-rc1. I currently have everything configured as per the latest recommendations and so far have made some promising progress.
I have two peers defined in Asterisk, which I am able to successfully register with. Signalling seems to be working nicely, as I am able to initiate and reject calls without any problems.
However, when I answer a call which is initiated, it terminates immediately. In the Asterisk logs, I see the following:-
[2015-05-12 13:03:33] WARNING[3931][C-00000009] res_rtp_asterisk.c: Could not set policies when setting up DTLS-SRTP on ‘0x7f752c001320’
[2015-05-12 13:03:33] WARNING[3931][C-00000009] res_rtp_asterisk.c: RTP Read error: Unspecified. Hanging up.
I have generated keys/certificates for Asterisk and they are declared in sip.conf (example):-
[XXX]
type=friend
username=XXX
host=dynamic
secret=XXXXXXXXXX
encryption=yes
avpf=yes
icesupport=yes
context=default
directmedia=no
transport=ws,wss,udp
force_avp=yes
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
idtlscafile=/etc/asterisk/keys/ca.crt
dtlssetup=actpass
dtlsrekey=60
videosupport=no
nat=no
disallow=all
allow=ulaw
Please could anyone shed some light on what I’m doing wrong? Thank you