PJSIP outgoing call audio not ok in the first call but OK in the immediate second call using webrtc client

pjsip.conf
;==============TRANSPORTS

[simpletrans]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.10.0/24
external_media_address=103.248.12.228
external_signaling_address=103.248.12.228

;===============TRUNK

[mytrunk]
type=registration
outbound_auth=mytrunk
server_uri=sip:103.248.12.227
client_uri=sip:202@103.248.12.227
retry_interval=60

[mytrunk]
type=auth
auth_type=userpass
password=xxxxxx
username=202

[mytrunk]
type=aor
contact=sip:103.248.12.227:5060
allow=ulaw
outbound_auth=mytrunk
aors=mytrunk
rtp_keepalive=1

[mytrunk]
type=identify
endpoint=mytrunk
match=103.248.12.227

[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
; All other transport parameters are ignored for wss transports.

[webrtc_client]
type=aor
max_contacts=5
remove_existing=yes

[webrtc_client]
type=auth
auth_type=userpass
username=webrtc_client
password=webrtc_client ; This is a completely insecure password! Do NOT expose this
; system to the Internet without utilizing a better password.

[webrtc_client]
type=endpoint
aors=webrtc_client
auth=webrtc_client
dtls_auto_generate_cert=yes
webrtc=yes
; Setting webrtc=yes is a shortcut for setting the following options:
; use_avpf=yes
; media_encryption=dtls
; dtls_verify=fingerprint
; dtls_setup=actpass
; ice_support=yes
; media_use_received_transport=yes
; rtcp_mux=yes
context=default
disallow=all
allow=opus,ulaw

extensions.conf
exten => _+.,1,Dial(PJSIP/${EXTEN}@mytrunk)
same => n,Hangup()

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