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route_phones
2015 Sep 15
3
Asterisk 13 WebRTC Status report
...:)
p.s. many thanks to my colleague martin tomec for debugging support
p.s.2 relevant part of pjsip.conf
[global]
[transport-wss]
type=transport
protocol=wss ;udp,tcp,tls,ws,wss
bind=0.0.0.0
;===============ENDPOINT TEMPLATES
[endpoint-basic](!)
type=endpoint
transport=transport-wss
context=route_phones
disallow=all
allow=alaw
allow=ulaw
force_avp=yes
use_avpf=yes ; Determines whether res_pjsip will use and enforce usage of
media_encryption=dtls ; Determines whether res_pjsip will use and enforce
dtls_verify=no ; Verify that the provided peer certificate is valid
(default:
dtls_rekey=0...