I am trying a webrtc-sip via Asterisk call with Asterisk 14 and WCS Server version FlashphonerWebCallServer-5.0.2159. I am using the demo application for WebRTC(Phone and Phone Video) and Bria for SIP End Point. I am getting the following issue in the console of Asterisk
[Apr 5 15:36:51] ERROR[C-00000001]: chan_sip.c:5933 dialog_initialize_dtls_srtp: No DTLS-SRTP support present on engine for RTP instance '0x7f3614009500', was it compiled with support for it?
[Apr 5 15:36:51] NOTICE[C-00000001]: chan_sip.c:26201 handle_request_invite: Failed to authenticate device "1060" <sip:email@example.com>;tag=fce5ad16
Can anyone please help me on this since I am unable to debug and the same and I'm stuck with this since a couple of days.
This is my http.conf
bindaddr=0.0.0.0 ; Replace this with your IP address
bindport=8088 ; Replace this with the port you want to listen on
realm=192.168.30.156 ; Replace this with your IP address
udpbindaddr=192.168.30.156 ; Replace this with your IP address
 ; This will be WebRTC client
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=password ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=no ; Tell Asterisk to not verify your DTLS certs
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
 ; This will be the legacy SIP client
exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060
exten => 1061,1,Dial(SIP/1061) ; Dialing 1061 will call the SIP client registered to 1061
Have checked the configs in Asterisk and it seems to be fine.
Also I just wanted to know if the FlashPhoner client can support SRTP since I don't see the Offer/Answer having the fingerprint.
Getting the following error in the Asterisk debug Logs
han_sip.c:10726 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
Attached the logs for the WEB-SIP and SIP-WEB logs.
Can anyone give an insight on this?Thanks for the help in advance.Really appreciate the help.