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:firstname.lastname@example.org>;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 [general] enabled=yes bindaddr=0.0.0.0 ; Replace this with your IP address bindport=8088 ; Replace this with the port you want to listen on ;sip.conf [general] realm=192.168.30.156 ; Replace this with your IP address udpbindaddr=192.168.30.156 ; Replace this with your IP address transport=udp  ; This will be WebRTC client type=friend 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 nat=no canreinvite=no qualify=no 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 dtlscafile=/etc/asterisk/keys/ca.crt allow=alaw media_encryption=dtls  ; This will be the legacy SIP client type=friend username=1061 host=dynamic secret=password context=default disallow=all allow=alaw qualify=no ;rtp.conf [general] rtpstart=10000 rtpend=20000 icesupport=yes stunaddr=stun.l.google.com:19302 ;extensions.conf ;extensions.conf [default] 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. I have the current architecture framework as of now. Firefox<->Web SDK Client(Flash Phoner SIP Phone Demo)<->WCS Server<->Asterisk<->Bria I am launching the WCS server with the demo app running on it. I understand that that WCS must be configured as a peer for both WebRTC and Asterisk. But can anyone please brief me about the framework again? How can one deploy the client sdk( and the path) from flashphoner which supports a video and an audio call through Asterisk and reach the SIP endpoint on the WCS server and run calls and register with Asterisk. Since I am fairly new to this framework, Really appreciate the help.... Thanking you for your help in advance..