WebRTC to WebRTC call on TLS , media relay via RTP engine
This is the next step from webrtc_to_webrtc_ws section. Note it doesnt not support sip client or ports on 5060 . Only works for websocket and wss port
installing RTpengine
echo 'deb http://deb.sipwise.com/spce/mr7.1.1/ stretch main' > /etc/apt/sources.list.d/sipwise.list
echo 'deb-src http://deb.sipwise.com/spce/mr7.1.1/ stretch main' >> /etc/apt/sources.list.d/sipwise.list
apt-get update
apt-get install -y --allow-unauthenticated ngcp-keyring
apt-get update
apt-get install -y ngcp-rtpengine
RTPengine configuration file
[rtpengine]
RUN_RTPENGINE=yes
listen-ng = 22222
listen-tcp = 25060
listen-udp = 12222
LISTEN-CLI=9900
interface = internal/private_ip!public_ip;external/public_ip
TIMEOUT=60
SILENT_TIMEOUT=3600
PIDFILE=/var/run/ngcp-rtpengine-daemon.pid
FORK=yes
TABLE=0
PORT_MIN=40000
PORT_MAX=60000
Debug help
Issues 1 : CRIT: Fatal error: Bad command line: Key file does not start with a group solution : add a group like [rtpengine] to the header of the file
Issues 2 : CRIT: Fatal error: Missing option –interface solution : Add interface from formats, such as a single interface:
interface = 123.234.345.456
separate multiple interfaces with semicolons:
interface = internal/10.20.30.40;external/123.234.345.456
for different advertised address:
interface = 12.23.34.45!23.34.45.56
Issue 3: CRIT: Fatal error: Missing option –listen-tcp, –listen-udp or –listen-ng Solution : Add listen option