Hello,
i’m new to asterisk and tried my best to set up my server, and I have a problem to receive call from my SIP line.
Everything was working very fine for weeks, but suddenly : no audio when I receive a phone call, or the caller can hear me (using his cellphone), but I cannot hear him (he is listening a kind of echo of what he is saying.)
I read a lot of topic about audio trouble, and tried to investigate, I think I can see in the log where is the problem, but I have no idea how to solve it. I hope you can help me about it
Asterisk server is on a VPS, no NAT. Android client or windows client (zoiper / linphone), are either on a wifi network behind a nat, either directly using 4G/5G. Same problem in both (wifi or 4g).
First, what I did :
1. pjsip.conf
(2 local users : 200 and 201, and registration for my SIP line) :
[transport-udp]
type = transport
protocol = udp ; choisir le protocole utilisé (udp / tcp / tls)
bind = 0.0.0.0:16060 ; changer le port par défaut
; ==================== UTILISATEUR 1 ====================
[200]
type = endpoint
context = local
disallow = all
allow = g722
allow = ulaw
auth = 200
aors = 200
[200]
type = auth
auth_type = userpass
password = toto
username = 200
[200]
type = aor
max_contacts = 1
qualify_frequency = 60
; ==================== UTILISATEUR 2 ====================
[201]
type = endpoint
context = local
disallow = all
allow = g722
allow = ulaw
auth = 201
aors = 201
direct_media = yes
[201]
type = auth
auth_type = userpass
password = tata
username = 201
[201]
type = aor
max_contacts = 1
qualify_frequency = 60
**; FOR SIP LINE REGISTRATION**
[ovh-trunk]
type=registration
outbound_auth=ovh-auth
server_uri=sip:sbc6.fr.sip.ovh
client_uri=sip:xxxxxxxxxx@sbc6.fr.sip.ovh
retry_interval=60
max_retries=30
expiration=3600
[ovh-auth]
type=auth
auth_type=userpass
username=xxxxxxxxx
password=secret
[ovh-aor]
type=aor
contact=sip:xxxxxxxxxxx@sbc6.fr.sip.ovh
[ovh-endpoint]
type=endpoint
context=local
aors=ovh-aor
;auth=ovh-auth
transport=transport-udp
disallow=all
allow=g722
allow=ulaw
allow=alaw
What is working :
someone calls me using his normal smartphone, asterisk receive the call and an exten DIAL my Zopier (200).
If I hangup on zoiper on windows on my PC, it’s working fine, and when I log rtp set debug on I can see something like this :
Got RTP packet from 91.121.129.145:31266 (type 08, seq 030839, ts 581981323, len 000160)
Sent RTP packet to 82.67.9.77:49353 (type 08, seq 022038, ts 640632, len 000160)
Got RTP packet from 82.67.9.77:49353 (type 08, seq 050037, ts 1830630439, len 000160)
Sent RTP packet to 91.121.129.145:31266 (type 08, seq 024441, ts 1830630432, len 000160)
Got RTP packet from 91.121.129.145:31266 (type 08, seq 030840, ts 581981483, len 000160)
Sent RTP packet to 82.67.9.77:49353 (type 08, seq 022039, ts 640792, len 000160)
Got RTP packet from 82.67.9.77:49353 (type 08, seq 050038, ts 1830630599, len 000160)
Sent RTP packet to 91.121.129.145:31266 (type 08, seq 024442, ts 1830630592, len 000160)
Got RTP packet from 91.121.129.145:31266 (type 08, seq 030841, ts 581981643, len 000160)
Sent RTP packet to 82.67.9.77:49353 (type 08, seq 022040, ts 640952, len 000160)
everything looks fine.
But when I receive the call on my android phone, since few days or weeks, it suddenly stopped working, no audio or only one way audio, and the log is :
Got RTP packet from 91.121.129.155:37986 (type 08, seq 002500, ts 1300816139, len 000160)
Sent RTP packet to 192.168.1.91:50476 (type 08, seq 011218, ts 067368, len 000160)
Got RTP packet from 82.67.9.77:50476 (type 08, seq 000436, ts 2186678056, len 000160)
Sent RTP packet to 91.121.129.155:37986 (type 08, seq 017414, ts 2186678056, len 000160)
Got RTP packet from 91.121.129.155:37986 (type 08, seq 002501, ts 1300816299, len 000160)
Sent RTP packet to 192.168.1.91:50476 (type 08, seq 011219, ts 067528, len 000160)
Got RTP packet from 82.67.9.77:50476 (type 08, seq 000437, ts 2186678216, len 000160)
Sent RTP packet to 91.121.129.155:37986 (type 08, seq 017415, ts 2186678216, len 000160)
Got RTP packet from 91.121.129.155:37986 (type 08, seq 002502, ts 1300816459, len 000160)
Sent RTP packet to 192.168.1.91:50476 (type 08, seq 011220, ts 067688, len 000160)
Got RTP packet from 82.67.9.77:50476 (type 08, seq 000438, ts 2186678376, len 000160)
Sent RTP packet to 91.121.129.155:37986 (type 08, seq 017416, ts 2186678376, len 000160)
This time I get a local 192.x.x.x IP address on “SentRTP packet to”
So I think this is why I have only one way audio or no audio ?
But I don’t understand why this happened suddenly, and some few days I still have not found anyway to fix it.
I tried to enable stun, ice, r_port… I think I tried all combination in zoiper but nothing worked. My phone and the computer are on the same network.
192.168.1.91 is the local IP of my phone, but why ? I setup a stun server, how can asterisk get it ? why only on android ? (when it was working fine I even didn’t setup stun server, I set up stun few days ago when it stopped working).
And IP 91.121.129.155: no idea what it is.
I hope you could understand my poor english and someone can help me thank you very much