Hi guys, verry simple setup, all is local
Registered an trunk , when call comes in, it starts calling endpoint 6001 (linphone user)
The problem is that approx 1 out of x calls, i cant hear two way audio
I did a RTP capture and grabbed some sip logs, i compared them, and all look exact the same?
What is wrong? any idea?
timestamp working call: 2023/02/22 13:07:01
timestamp non working call, no audio: 2023/02/22 13:07:12
pjsip config for trunk and endpoint:
[mytrunk-auth]
type=auth
auth_type=userpass
password=xxx
username=10000000005
[mytrunk-aor]
type=aor
contact=sip:192.168.0.71:5065
;qualify_frequency=300
[mytrunk-registration]
type=registration
outbound_auth=mytrunk-auth
server_uri=sip:192.168.0.71:5065
client_uri=sip:10000000005@192.168.0.71:5065
retry_interval=10
contact_user=10000000005
expiration=600
[mytrunk]
type=endpoint
context=default
disallow=all
allow=ulaw
allow=h264
outbound_auth=mytrunk-auth
aors=mytrunk-aor
rewrite_contact=yes
from_domain=mydomain.com
[mytrunk]
type=identify
endpoint=mytrunk
match=192.168.0.71
-------------------------------
[6000]
type=endpoint
context=default
disallow=all
allow=ulaw
allow=h264
auth=auth6000
aors=6000
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
direct_media=no
max_audio_streams=10
max_video_streams=10
from_domain=mydomain.com
[auth6000]
type=auth
auth_type=userpass
password=xxx
username=6000
[6000]
type=aor
qualify_frequency=30
max_contacts=1
remove_existing=yes
remove_unavailable=yes
dialplan:
exten => 10000000009,1,NoOp()
same => n,Progress()
same => n,Dial(PJSIP/6001)
Logs:
rtp logs
rtp first audio second no audio.txt (548.2 KB)
sip traces
audio.txt (10.1 KB)
noaudio.txt (10.1 KB)