This question has already been asked here. However based on a comment there, I am posting it here.
I am trying to set up Asterisk to work with webrtc.. On the client side I am using sipML5.
This is new to me so I am having some difficulties. Below are my config file
extensions.conf
[default]
exten=>bob,1,Dial(PJSIP/${EXTEN})
exten=>lucy,1,Dial(PJSIP/${EXTEN})
http.conf
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/asterisk/keys/asterisk.pem
rtp.conf
[general]
rtpstart=10000
rtpend=20000
stunaddr=stun.l.google.com:19302
pjsip.conf
[transport_wss]
type=transport
bind=0.0.0.0
protocol=wss
[bob]
type=aor
max_contacts=1
[bob]
type=auth
auth_type=userpass
username=bob
password=123456 ; This is an insecure password
[bob]
type=endpoint
context=default
direct_media=no
allow=!all,ulaw,vp8,h264
aors=bob
auth=bob
max_audio_streams=10
max_video_streams=10
webrtc=yes
dtls_cert_file=/etc/asterisk/keys/asterisk.pem
dtls_ca_file=/etc/asterisk/keys/ca.crt
[lucy]
type=aor
max_contacts=1
[lucy]
type=auth
auth_type=userpass
username=lucy
password=123456 ; This is an insecure password
[lucy]
type=endpoint
context=default
direct_media=no
allow=!all,ulaw,vp8,h264
aors=lucy
auth=lucy
max_audio_streams=10
max_video_streams=10
webrtc=yes
dtls_cert_file=/etc/asterisk/keys/asterisk.pem
dtls_ca_file=/etc/asterisk/keys/ca.crt
I am using sipml5 in browser to initiate a call. 
.
While the registration process is done without any hassle, whenever I try to call lucy, it shows call in progress... and then nothing. I am pasting the output from browser console.
SEND: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKJqNKVm8FAolyGCgAwqzodBv7mqnn1fMI;rport
From: "bob"<sip:[email protected]>;tag=zoXiEWFrIS8aWE8NsM73
To: <sip:[email protected]>
Contact: "bob"<sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: d3132f61-16b8-b1e1-a3a7-57d6e4a7c026
CSeq: 31217 INVITE
Content-Type: application/sdp
Content-Length: 1345
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
v=0
o=mozilla...THIS_IS_SDPARTA-91.0.2 7078761787079714000 0 IN IP4 127.0.0.1
s=Doubango Telecom - firefox
t=0 0
a=sendrecv
a=fingerprint:sha-256 D4:19:8F:2E:4B:09:9D:11:B1:BE:39:9E:C1:DA:4A:A0:F2:78:AB:3A:6F:85:70:7F:83:66:69:F7:F3:45:C8:69
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 48510 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 192.168.5.240
a=candidate:0 1 UDP 2122252543 192.168.5.240 48510 typ host
a=candidate:5 1 TCP 2105524479 192.168.5.240 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.5.240 33548 typ host
a=candidate:5 2 TCP 2105524478 192.168.5.240 9 typ host tcptype active
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:f783c8c7730e1a0fb0452874847c94bd
a=ice-ufrag:cc6d4770
a=mid:0
a=msid:{f9680322-af64-4e1c-9d4d-914ecb7e000f} {e5f3c53b-63f2-48d4-8543-2e123d3a0014}
a=rtcp:33548 IN IP4 192.168.5.240
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:2565114692 cname:{8199f724-189a-4907-9e58-479c402727c4}
__tsip_transport_ws_onmessage tsk_utils.js:116:65
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=52704;received=192.168.5.240;branch=z9hG4bKJqNKVm8FAolyGCgAwqzodBv7mqnn1fMI
From: "bob"<sip:[email protected]>;tag=zoXiEWFrIS8aWE8NsM73
To: <sip:[email protected]>;tag=z9hG4bKJqNKVm8FAolyGCgAwqzodBv7mqnn1fMI
Call-ID: d3132f61-16b8-b1e1-a3a7-57d6e4a7c026
CSeq: 31217 INVITE
Content-Length: 0
WWW-Authenticate: Digest realm="asterisk",qop="auth",nonce="1630570870/e1479902f9a76951e382002e033c97d6",opaque="2d987ff95facbd3f",stale=FALSE,algorithm=md5
Server: Asterisk PBX 18.5.1
SEND: ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKJqNKVm8FAolyGCgAwqzodBv7mqnn1fMI;rport
From: "bob"<sip:[email protected]>;tag=zoXiEWFrIS8aWE8NsM73
To: <sip:[email protected]>;tag=z9hG4bKJqNKVm8FAolyGCgAwqzodBv7mqnn1fMI
Call-ID: d3132f61-16b8-b1e1-a3a7-57d6e4a7c026
CSeq: 31217 ACK
Content-Length: 0
Max-Forwards: 70
State machine: x0000_Any_2_Any_X_i401_407_INVITE tsk_utils.js:116:65
SEND: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvOvxmLoVuDrOKCGHwcTUJnsa6W1pDd0o;rport
From: "bob"<sip:[email protected]>;tag=zoXiEWFrIS8aWE8NsM73
To: <sip:[email protected]>
Contact: "bob"<sip:[email protected];rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: d3132f61-16b8-b1e1-a3a7-57d6e4a7c026
CSeq: 31218 INVITE
Content-Type: application/sdp
Content-Length: 1345
Max-Forwards: 70
Authorization: Digest username="bob",realm="asterisk",nonce="1630570870/e1479902f9a76951e382002e033c97d6",uri="sip:[email protected]",response="392ab7a05965b49f6516d5622a92f209",algorithm=md5,cnonce="6ecee939225265170a96b8ebef0f88ec",opaque="2d987ff95facbd3f",qop=auth,nc=00000001
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
v=0
o=mozilla...THIS_IS_SDPARTA-91.0.2 7078761787079714000 0 IN IP4 127.0.0.1
s=Doubango Telecom - firefox
t=0 0
a=sendrecv
a=fingerprint:sha-256 D4:19:8F:2E:4B:09:9D:11:B1:BE:39:9E:C1:DA:4A:A0:F2:78:AB:3A:6F:85:70:7F:83:66:69:F7:F3:45:C8:69
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 48510 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 192.168.5.240
a=candidate:0 1 UDP 2122252543 192.168.5.240 48510 typ host
a=candidate:5 1 TCP 2105524479 192.168.5.240 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.5.240 33548 typ host
a=candidate:5 2 TCP 2105524478 192.168.5.240 9 typ host tcptype active
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:f783c8c7730e1a0fb0452874847c94bd
a=ice-ufrag:cc6d4770
a=mid:0
a=msid:{f9680322-af64-4e1c-9d4d-914ecb7e000f} {e5f3c53b-63f2-48d4-8543-2e123d3a0014}
a=rtcp:33548 IN IP4 192.168.5.240
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:2565114692 cname:{8199f724-189a-4907-9e58-479c402727c4}
__tsip_transport_ws_onmessage tsk_utils.js:116:65
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport=52704;received=192.168.5.240;branch=z9hG4bKvOvxmLoVuDrOKCGHwcTUJnsa6W1pDd0o
From: "bob"<sip:[email protected]>;tag=zoXiEWFrIS8aWE8NsM73
To: <sip:[email protected]>
Call-ID: d3132f61-16b8-b1e1-a3a7-57d6e4a7c026
CSeq: 31218 INVITE
Content-Length: 0
Server: Asterisk PBX 18.5.1
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:116:65
==session event = i_ao_request tsk_utils.js:116:65
After this, no 180 TRYING appears in the console. I am not sure why, but after the above message, there are again registration message in the console. I have seen multiple examples on the internet, but I could not manage to configure most of them.
Thanks
