WebRTC

VoIPmonitor sniffer is able to analyse SIP over WebSocket encrypted or unencrypted. For unencrypted WebSocket just configure WebScoket port as sipport:

voipmonitor.conf:

sipport = 5060, 8088

this example will analyse SIP TCP/UDP and SIP over WebSocket on port 8088

For encrypted webscoket see following examples for Freeswitch and Asterisk:

Freeswitch

vars.conf

<param name="tls-version" value="tlsv1.2"/> 
<param name="tls-ciphers" value="AES128-SHA" >

voipmonitor.conf

ssl = yes
ssl_ipport = 192.168.0.1 : 7443 /etc/voipmonitor/privkey.pem

Asterisk

http.conf

;
; Asterisk Builtin mini-HTTP server
;
;
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
;prefix=asterisk
;sessionlimit=100
;enablestatic=yes
;redirect = / /static/config/index.html
tlsenable=yes          ; enable tls - default no.
tlsbindaddr=0.0.0.0:8089    ; address and port to bind to - default is bindaddr and port 8089.
tlscertfile=/etc/asterisk/keys/asterisk.pem  ; path to the certificate file (*.pem) only.
tlscipher=AES128-SHA
;tlsprivatekey=</path/to/private.pem>    ; path to private key file (*.pem) only.

rtp.conf

add at the end of this file:

icesupport=yes
stunaddr=stun.l.google.com:19302

pjsip.conf

[general]
allowguest = no 
  [global]
type = global
user_agent = VoIPsun PBX
realm=192.168.2.107
bindport=5060
transport=udp,ws,wss

[transport-udp]
type = transport
protocol = udp
bind = 192.168.2.107:5060
tos = cs3
cos = 3 

[transport-ws]
type=transport
protocol=ws
bind=192.168.2.107 

[transport-wss]
type=transport
protocol=wss
bind=192.168.2.107
cipher=0x002f
method=tlsv1_2
 [101]
type=aor
max_contacts=1
remove_existing=yes

[101]
type=auth
auth_type=userpass
username=101
password=1234

[101]
type=endpoint
disallow=all
allow=opus
allow=alaw
allow=ulaw
context=from101
auth=101
aors=101
media_encryption=dtls
dtls_verify=fingerprint
dtls_cert_file=/etc/asterisk/keys/asterisk.pem
dtls_ca_file=/etc/asterisk/keys/ca.crt
dtls_setup=actpass
use_avpf=yes
ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes

[102]
type=aor
max_contacts=1
remove_existing=yes

[102]
type=auth
auth_type=userpass
username=102
password=1234

[102]
type=endpoint
disallow=all
allow=opus
allow=alaw
allow=ulaw
context=from102
auth=102
aors=102
media_encryption=dtls
dtls_verify=fingerprint
dtls_cert_file=/etc/asterisk/keys/asterisk.pem
dtls_ca_file=/etc/asterisk/keys/ca.crt
dtls_setup=actpass
use_avpf=yes
ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes

extensions.conf

[from101] exten => _X.,1,NooP(Call from 101 to ${EXTEN}) same => n,Dial(PJSIP/102/${EXTEN}) 
exten => i,1,Goto(other,${EXTEN},1)
[from102]
exten => _X.,1,NooP(Call from 102 to ${EXTEN})
same => n,Dial(PJSIP/101/${EXTEN})

exten => i,1,Goto(other,${EXTEN},1)
[other]
exten => X,1,NooP(Call from ${CALLERID(num)} to ${EXTEN})
same => n,DumpChan()
same => n,Ringing()
same => n,Wait(3)
same => n,Playback(/var/lib/asterisk/sounds/cz/queue-periodic-announce)
same => n,Hangup()

modules.conf

noload => chan_sip.so

keys

mkdir /etc/asterisk/keys
cd /etc/asterisk/keys
openssl genrsa -des3 -out ca.key 4096
openssl req -new -x509 -days 365 -key ca.key -out ca.crt
openssl genrsa -out key.pem 1024
openssl req -new -key key.pem -out req-sip_server.csr
openssl x509 -req -days 365 -in req-sip_server.csr -CA ca.crt -CAkey ca.key -set_serial 01 -out cert-sip_server.crt
cat key.pem > asterisk.pem
cat cert-sip_server.crt >> asterisk.pem

Sipml5

https://www.doubango.org/sipml5/call.htm?svn=170#

Display name: 102   
Private Identity: 102 
Public Identity: sip:102@192.168.2.107 
Password: 1234Realm: 192.168.2.107

click on expert mode:

Disable video: on
Enable RTCWeb breaker
WebSocket Server URL: wss://192.168.2.107:8089/ws
ICE servers: [{ url: 'stun:stun.l.google.com:19302'}]  (this can be maybe empty) 
Disable 3GPP Early IMS: on
Disable debug messages: on
Cache media stream: on
Disable Call button options: on

• after settings go back to first tab and click login

• open https://192.168.2.107:8089/ws in browser and accept the certificate - otherwise the webrtc will not login.