Registration is successful. But unable make calls in Public IP

periyasamy · December 13, 2023, 3:58pm

Hi,
I have the following pjsip.config. Server is behind NAT and Client is running on mobile phone.

[general]
context=internal
allowguest=no
allowoverlap=no

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5061
local_net=10.1.2.0/24
external_media_address=XX.XXX.XX.XX:5061
external_signaling_address=XX.XXX.XX.XX:5061

[auth200]
type=auth
username=200
password=XXXX
auth_type=userpass

[200]

type=aor
max_contacts=3
qualify_frequency=60

[200]
type=endpoint
context=internal
auth=auth200
aors=200
disallow=all
allow=ulaw,alaw
transport=transport-udp
direct_media=no
force_rport=yes
rewrite_contact=yes
rtp_symmetric=yes
ice_support=yes

<— Received SIP request (560 bytes) from UDP:49.37.203.219:35896 —>
REGISTER sip:sip.xxx.in:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.29.14:35896;branch=z9hG4bK-524287-1—9f6f01536956e22b;rport
Max-Forwards: 70
Contact: sip:200@192.168.29.14:35896;rinstance=19d6c5f15e105a46
To: "200"sip:200@sip.xxx.in:5061
From: "200"sip:200@sip.xxx.in:5061;tag=4f33e837
Call-ID: nwh-lXJYQsRgT2PWC-g02A…
CSeq: 20403 REGISTER
Expires: 600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 7.0.4
Content-Length: 0

<— Transmitting SIP response (517 bytes) to UDP:49.37.203.219:35896 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.29.14:35896;rport=35896;received=49.37.203.219;branch=z9hG4bK-524287-1—9f6f01536956e22b
Call-ID: nwh-lXJYQsRgT2PWC-g02A…
From: “200” sip:200@sip.xxx.in;tag=4f33e837
To: “200” sip:200@sip.xxx.in;tag=z9hG4bK-524287-1—9f6f01536956e22b
CSeq: 20403 REGISTER
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1702479060/3ab18678e9b6ce2f0b68bf5bcf43248f”,opaque=“0c77b899300ceaa5”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 20.5.0
Content-Length: 0

<— Received SIP request (841 bytes) from UDP:49.37.203.219:35896 —>
REGISTER sip:sip.xxx.in:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.29.14:35896;branch=z9hG4bK-524287-1—1837fb668fa3df44;rport
Max-Forwards: 70
Contact: sip:200@192.168.29.14:35896;rinstance=19d6c5f15e105a46
To: "200"sip:200@sip.xxx.in:5061
From: "200"sip:200@sip.xxx.in:5061;tag=4f33e837
Call-ID: nwh-lXJYQsRgT2PWC-g02A…
CSeq: 20404 REGISTER
Expires: 600
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 7.0.4
Authorization: Digest username=“200”,realm=“asterisk”,nonce=“1702479060/3ab18678e9b6ce2f0b68bf5bcf43248f”,uri=“sip:sip.xxx.in:5061”,response=“414696efd290405718449cfa06502e5b”,cnonce=“72831051180d6d9a063ab60423063e8e”,nc=00000001,qop=auth,algorithm=MD5,opaque=“0c77b899300ceaa5”
Content-Length: 0
Added contact ‘sip:200@49.37.203.219:35896;rinstance=19d6c5f15e105a46;x-ast-orig-host=192.168.29.14:35896’ to AOR ‘200’ with expiration of 600 seconds

<— Transmitting SIP response (491 bytes) to UDP:49.37.203.219:35896 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.29.14:35896;rport=35896;received=49.37.203.219;branch=z9hG4bK-524287-1—1837fb668fa3df44
Call-ID: nwh-lXJYQsRgT2PWC-g02A…
From: “200” sip:200@sip.xxx.in;tag=4f33e837
To: “200” sip:200@sip.xxx.in;tag=z9hG4bK-524287-1—1837fb668fa3df44
CSeq: 20404 REGISTER
Date: Wed, 13 Dec 2023 14:51:00 GMT
Contact: sip:200@192.168.29.14:35896;rinstance=19d6c5f15e105a46;expires=599
Expires: 600
Server: Asterisk PBX 20.5.0
Content-Length: 0

<— Transmitting SIP request (467 bytes) to UDP:49.37.203.219:35896 —>
OPTIONS sip:200@49.37.203.219:35896;rinstance=19d6c5f15e105a46 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.114:5061;rport;branch=z9hG4bKPj79d2a96d-baed-4843-af82-9978810f7f27
From: sip:200@10.1.2.114;tag=4dc16994-8a4d-4c5b-9a6c-23bde5b399d8
To: sip:200@49.37.203.219;rinstance=19d6c5f15e105a46
Contact: sip:200@10.1.2.114:5061
Call-ID: 347454dd-2228-4048-9ae1-acbc4acd7075
CSeq: 33927 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length: 0

<— Received SIP response (593 bytes) from UDP:49.37.203.219:35896 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.114:5061;rport=5061;branch=z9hG4bKPj79d2a96d-baed-4843-af82-9978810f7f27;received=182.66.XXX.XX
Contact: sip:192.168.29.14:35896
To: sip:200@49.37.203.219;rinstance=19d6c5f15e105a46;tag=8566c618
From: sip:200@10.1.2.114;tag=4dc16994-8a4d-4c5b-9a6c-23bde5b399d8
Call-ID: 347454dd-2228-4048-9ae1-acbc4acd7075
CSeq: 33927 OPTIONS
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 7.0.4
Content-Length: 0

Endpoint 200 is now Reachable

Contact 200/sip:200@49.37.203.219:35896;rinstance=19d6c5f15e105a46;x-ast-orig-host=192.168.29.14:35896 is now Reachable. RTT: 99.616 msec

<— Transmitting SIP request (467 bytes) to UDP:49.37.203.219:35896 —>
OPTIONS sip:200@49.37.203.219:35896;rinstance=19d6c5f15e105a46 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.114:5061;rport;branch=z9hG4bKPjafc5b326-4429-48fa-b946-a226b627d304
From: sip:200@10.1.2.114;tag=692c1e77-4570-4f4d-bf68-e83d3156e811
To: sip:200@49.37.203.219;rinstance=19d6c5f15e105a46
Contact: sip:200@10.1.2.114:5061
Call-ID: 732bd749-f0b8-4827-a91a-d4ced2a38feb
CSeq: 24345 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length: 0

<— Received SIP response (593 bytes) from UDP:49.37.203.219:35896 —>

SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.2.114:5061;rport=5061;branch=z9hG4bKPjafc5b326-4429-48fa-b946-a226b627d304;received=182.66.XXX.XX
Contact: sip:192.168.29.14:35896
To: sip:200@49.37.203.219;rinstance=19d6c5f15e105a46;tag=51c3d67e
From: sip:200@10.1.2.114;tag=692c1e77-4570-4f4d-bf68-e83d3156e811
Call-ID: 732bd749-f0b8-4827-a91a-d4ced2a38feb
CSeq: 24345 OPTIONS
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, SUBSCRIBE, UPDATE, INFO, MESSAGE
Supported: path, replaces, timer, norefersub
User-Agent: SessionTalk 7.0.4
Content-Length: 0

<— Transmitting SIP request (466 bytes) to UDP:49.37.203.219:35896 —>
OPTIONS sip:200@49.37.203.219:35896;rinstance=19d6c5f15e105a46 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.114:5061;rport;branch=z9hG4bKPjdc2d41d0-3356-4817-a51c-e130f8e4c73d
From: sip:200@10.1.2.114;tag=44a95ce2-404e-4e32-847c-492830b3d9ab
To: sip:200@49.37.203.219;rinstance=19d6c5f15e105a46
Contact: sip:200@10.1.2.114:5061
Call-ID: 93f02d98-eef4-47d5-bcd2-22591a739470
CSeq: 4853 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.5.0
Content-Length: 0

Endpoint 200 is now Unreachable

Contact 200/sip:200@49.37.203.219:35896;rinstance=19d6c5f15e105a46;x-ast-orig-host=192.168.29.14:35896 is now Unreachable. RTT: 0.000 msec

TheMark · December 13, 2023, 6:48pm

you neet to look at way your client stop responding to options send by asterisk
one guess it that the client NAT router is closing the connection before Asterisk send options
you may need to look at the interval asterisk send options
and changet it so it will send it every 25s as many router will close the NAT connetion after 30s

david551 · December 13, 2023, 7:50pm

Also, there is no general section in pjsip.conf, and no allowguest option. I’m not sure about overlap dialling.

periyasamy · December 15, 2023, 11:37am

I removed General section and also my client is on mobile network. SIP on udp(localip and publicip) works fine. pjsip on localnetwork also works fine. Only through Public IP we’re facing problem. tls also not working bot SIP and pjsip

system · January 14, 2024, 11:38am

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Registration is successful. But unable make calls in Public IP

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