Hi there, I'm already using Kamailio as a simple proxy and it worked flawlessly.
This time, I need to have a new setup like this: [UAC (sip phone)] <--- register & has to deal only with ----> | PUBLIC | [Kamailio + RtpEngine] | PRIVATE | <----> [Asterisk] Client = 37.100.100.100 (public) Asterisk = 172.17.200.20 (private) Kamailio/RtpEngine = 172.17.203.102 (private, advertising 3.99.99.99 public IP) Kamailio and RtpEngine will behave as a SBC/B2BUA, the client will deal only with kamailio using public access (internet), and asterisk will handle the calls but stays 100% private behind kamailio. Kamailio will handle registrations to reduce the load put on the asterisk boxes. The setup is running on AWS, my Kamailio has 1 Private IP address tied to 1 Public IP address (an Elastic public IP) so, no multi-homing, one network interface. I made a "simple" work in progress configuration, the registration is working as a PoC, and my calls are *almost* working but with some nonsense. My Main issue is that when my UAC (SIP phone) make a call, to let's say +1000, the call enters kamailio, then kamailio handle the invite, trigger rtpEngine, but the IP addresses in Record-Route and Via headers are wrong. >From Kamailio to UAC Sip Phone, it seems OK. But from Kamailio to Asterisk, the record-route/via headers are set with Kamailio Public IP when it should be the Kamailio private IP. So, when Asterisk sends a BYE, it tries to send it to the public IP of Kamailio, which is wrong. There are probably other quirks in the configuration but this one, I can't figure out what's wrong. I'm attaching a capture file, a log file and my configuration file (some informations are redacted). I am missing something and sadly I can't figure out what, a little help would be appreciated :) Regards
## Network capture from Kamailio box
08:43:33.472017 IP (tos 0x0, ttl 102, id 48545, offset 0, flags [none], proto
UDP (17), length 30)
100.100.100.37.rev.sfr.net.56065 >
ip-172-17-203-102.eu-central-1.compute.internal.sip: [udp sum ok] SIP
08:43:35.592685 IP (tos 0x0, ttl 102, id 48546, offset 0, flags [none], proto
UDP (17), length 1021)
100.100.100.37.rev.sfr.net.56065 >
ip-172-17-203-102.eu-central-1.compute.internal.sip: [udp sum ok] SIP, length:
993
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
37.100.100.100:56065;rport;branch=z9hG4bKPja0fcc9509b4c43269259445a4912e1a3
Max-Forwards: 70
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To: <sip:[email protected]>
Contact: <sip:[email protected]:56065;ob>
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
CSeq: 12028 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE,
NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.20.7
Content-Type: application/sdp
Content-Length: 339
v=0
o=- 3846649415 3846649415 IN IP4 37.100.100.100
s=pjmedia
b=AS:84
t=0 0
a=X-nat:1
m=audio 4000 RTP/AVP 8 0 101
c=IN IP4 37.100.100.100
b=TIAS:64000
a=rtcp:4001 IN IP4 37.100.100.100
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:801129593 cname:67e4069c62f84408
08:43:35.593656 IP (tos 0x10, ttl 255, id 5473, offset 0, flags [none], proto
UDP (17), length 572)
ip-172-17-203-102.eu-central-1.compute.internal.sip >
100.100.100.37.rev.sfr.net.56065: [bad udp cksum 0x732b -> 0x5b71!] SIP,
length: 544
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
37.100.100.100:56065;rport=56065;branch=z9hG4bKPja0fcc9509b4c43269259445a4912e1a3;received=37.100.100.100
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=3cd483beb4ee8a26c384c7540ab382f6.a416a2e2
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
CSeq: 12028 INVITE
WWW-Authenticate: Digest realm="sip1.redacted.com",
nonce="YZyq42GcqbcosEWlY5prHZTM2GIvDdBx", qop="auth"
Server: kamailio (5.4.7 (x86_64/linux))
Content-Length: 0
08:43:35.621964 IP (tos 0x0, ttl 102, id 48547, offset 0, flags [none], proto
UDP (17), length 435)
100.100.100.37.rev.sfr.net.56065 >
ip-172-17-203-102.eu-central-1.compute.internal.sip: [udp sum ok] SIP, length:
407
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
37.100.100.100:56065;rport;branch=z9hG4bKPja0fcc9509b4c43269259445a4912e1a3
Max-Forwards: 70
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=3cd483beb4ee8a26c384c7540ab382f6.a416a2e2
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
CSeq: 12028 ACK
Content-Length: 0
08:43:35.622643 IP (tos 0x0, ttl 102, id 48548, offset 0, flags [none], proto
UDP (17), length 1278)
100.100.100.37.rev.sfr.net.56065 >
ip-172-17-203-102.eu-central-1.compute.internal.sip: [udp sum ok] SIP, length:
1250
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP
37.100.100.100:56065;rport;branch=z9hG4bKPjf0384dab541e481f9b5dfdb3c4dd6ed2
Max-Forwards: 70
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To: <sip:[email protected]>
Contact: <sip:[email protected]:56065;ob>
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
CSeq: 12029 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE,
NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.20.7
Authorization: Digest username="test", realm="sip1.redacted.com",
nonce="YZyq42GcqbcosEWlY5prHZTM2GIvDdBx", uri="sip:[email protected]",
response="178e3ba525529cd187af7f83f7676091",
cnonce="e19a1640cbe347229a2074ba58f3e099", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 339
v=0
o=- 3846649415 3846649415 IN IP4 37.100.100.100
s=pjmedia
b=AS:84
t=0 0
a=X-nat:1
m=audio 4000 RTP/AVP 8 0 101
c=IN IP4 37.100.100.100
b=TIAS:64000
a=rtcp:4001 IN IP4 37.100.100.100
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:801129593 cname:67e4069c62f84408
08:43:35.626298 IP (tos 0x10, ttl 255, id 5480, offset 0, flags [none], proto
UDP (17), length 448)
ip-172-17-203-102.eu-central-1.compute.internal.sip >
100.100.100.37.rev.sfr.net.56065: [bad udp cksum 0x72af -> 0x8554!] SIP,
length: 420
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP
37.100.100.100:56065;rport=56065;branch=z9hG4bKPjf0384dab541e481f9b5dfdb3c4dd6ed2;received=37.100.100.100
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To: <sip:[email protected]>
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
CSeq: 12029 INVITE
Server: kamailio (5.4.7 (x86_64/linux))
Content-Length: 0
08:43:35.626499 IP (tos 0x10, ttl 255, id 48278, offset 0, flags [none], proto
UDP (17), length 1153)
ip-172-17-203-102.eu-central-1.compute.internal.sip >
ip-172-17-200-20.eu-central-1.compute.internal.sip: [bad udp cksum 0xf01c ->
0x9a8f!] SIP, length: 1125
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:3.99.99.99;lr>
Via: SIP/2.0/UDP
3.99.99.99:5060;branch=z9hG4bK37f1.d3ae697f45b71811bfa3d1a574ffe93d.0
Via: SIP/2.0/UDP
37.100.100.100:56065;received=37.100.100.100;rport=56065;branch=z9hG4bKPjf0384dab541e481f9b5dfdb3c4dd6ed2
Max-Forwards: 69
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To: <sip:[email protected]>
Contact: <sip:[email protected]:56065;ob>
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
CSeq: 12029 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE,
NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: MicroSIP/3.20.7
Content-Type: application/sdp
Content-Length: 317
v=0
o=- 3846649415 3846649415 IN IP4 3.99.99.99
s=pjmedia
b=AS:84
t=0 0
a=X-nat:1
m=audio 23000 RTP/AVP 8 0 101
c=IN IP4 3.99.99.99
b=TIAS:64000
a=ssrc:801129593 cname:67e4069c62f84408
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:23001
08:43:35.627367 IP (tos 0xb8, ttl 64, id 26920, offset 0, flags [DF], proto UDP
(17), length 563)
ip-172-17-200-20.eu-central-1.compute.internal.sip >
ip-172-17-203-102.eu-central-1.compute.internal.sip: [udp sum ok] SIP, length:
535
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
3.99.99.99:5060;rport=5060;received=172.17.203.102;branch=z9hG4bK37f1.d3ae697f45b71811bfa3d1a574ffe93d.0
Via: SIP/2.0/UDP
37.100.100.100:56065;rport=56065;received=37.100.100.100;branch=z9hG4bKPjf0384dab541e481f9b5dfdb3c4dd6ed2
Record-Route: <sip:3.99.99.99;lr>
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To: <sip:[email protected]>
CSeq: 12029 INVITE
Server: Asterisk PBX 16.9.0
Content-Length: 0
08:43:35.629478 IP (tos 0xb8, ttl 64, id 26921, offset 0, flags [DF], proto UDP
(17), length 751)
ip-172-17-200-20.eu-central-1.compute.internal.sip >
ip-172-17-203-102.eu-central-1.compute.internal.sip: [udp sum ok] SIP, length:
723
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
3.99.99.99:5060;rport=5060;received=172.17.203.102;branch=z9hG4bK37f1.d3ae697f45b71811bfa3d1a574ffe93d.0
Via: SIP/2.0/UDP
37.100.100.100:56065;rport=56065;received=37.100.100.100;branch=z9hG4bKPjf0384dab541e481f9b5dfdb3c4dd6ed2
Record-Route: <sip:3.99.99.99;lr>
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
CSeq: 12029 INVITE
Server: Asterisk PBX 16.9.0
Contact: <sip:172.17.200.20:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length: 0
08:43:35.629750 IP (tos 0x10, ttl 255, id 5481, offset 0, flags [none], proto
UDP (17), length 628)
ip-172-17-203-102.eu-central-1.compute.internal.sip >
100.100.100.37.rev.sfr.net.56065: [bad udp cksum 0x7363 -> 0x00a9!] SIP,
length: 600
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
37.100.100.100:56065;rport=56065;received=37.100.100.100;branch=z9hG4bKPjf0384dab541e481f9b5dfdb3c4dd6ed2
Record-Route: <sip:3.99.99.99;lr>
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
CSeq: 12029 INVITE
Server: Asterisk PBX 16.9.0
Contact: <sip:172.17.200.20:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length: 0
08:43:37.632035 IP (tos 0xb8, ttl 64, id 26992, offset 0, flags [DF], proto UDP
(17), length 1144)
ip-172-17-200-20.eu-central-1.compute.internal.sip >
ip-172-17-203-102.eu-central-1.compute.internal.sip: [udp sum ok] SIP, length:
1116
SIP/2.0 200 OK
Via: SIP/2.0/UDP
3.99.99.99:5060;rport=5060;received=172.17.203.102;branch=z9hG4bK37f1.d3ae697f45b71811bfa3d1a574ffe93d.0
Via: SIP/2.0/UDP
37.100.100.100:56065;rport=56065;received=37.100.100.100;branch=z9hG4bKPjf0384dab541e481f9b5dfdb3c4dd6ed2
Record-Route: <sip:3.99.99.99;lr>
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
CSeq: 12029 INVITE
Server: Asterisk PBX 16.9.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:172.17.200.20:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 263
v=0
o=- 3846649415 3846649417 IN IP4 172.17.200.20
s=Asterisk
c=IN IP4 172.17.200.20
t=0 0
m=audio 10786 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
08:43:37.633396 IP (tos 0x10, ttl 255, id 5619, offset 0, flags [none], proto
UDP (17), length 1029)
ip-172-17-203-102.eu-central-1.compute.internal.sip >
100.100.100.37.rev.sfr.net.56065: [bad udp cksum 0x74f4 -> 0x502e!] SIP,
length: 1001
SIP/2.0 200 OK
Via: SIP/2.0/UDP
37.100.100.100:56065;rport=56065;received=37.100.100.100;branch=z9hG4bKPjf0384dab541e481f9b5dfdb3c4dd6ed2
Record-Route: <sip:3.99.99.99;lr>
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
CSeq: 12029 INVITE
Server: Asterisk PBX 16.9.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, MESSAGE, REFER
Contact: <sip:172.17.200.20:5060>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 271
v=0
o=- 3846649415 3846649417 IN IP4 3.99.99.99
s=Asterisk
c=IN IP4 3.99.99.99
t=0 0
m=audio 23020 RTP/AVP 8 0 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:23021
a=ptime:20
08:43:37.679055 IP (tos 0x0, ttl 102, id 48552, offset 0, flags [none], proto
UDP (17), length 448)
100.100.100.37.rev.sfr.net.56065 >
ip-172-17-203-102.eu-central-1.compute.internal.sip: [udp sum ok] SIP, length:
420
ACK sip:172.17.200.20:5060 SIP/2.0
Via: SIP/2.0/UDP
37.100.100.100:56065;rport;branch=z9hG4bKPj1df5c9a4fada4c28a9ad5f7540513287
Max-Forwards: 70
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
CSeq: 12029 ACK
Route: <sip:3.99.99.99;lr>
Content-Length: 0
08:43:37.679671 IP (tos 0x0, ttl 102, id 48553, offset 0, flags [none], proto
UDP (17), length 943)
100.100.100.37.rev.sfr.net.56065 >
ip-172-17-203-102.eu-central-1.compute.internal.sip: [udp sum ok] SIP, length:
915
UPDATE sip:172.17.200.20:5060 SIP/2.0
Via: SIP/2.0/UDP
37.100.100.100:56065;rport;branch=z9hG4bKPj553ecda2f38547e884efd5bba4cc313d
Max-Forwards: 70
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
Contact: <sip:[email protected]:56065;ob>
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
CSeq: 12030 UPDATE
Route: <sip:3.99.99.99;lr>
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length: 315
v=0
o=- 3846649415 3846649416 IN IP4 37.100.100.100
s=pjmedia
b=AS:84
t=0 0
a=X-nat:1
m=audio 4000 RTP/AVP 8 101
c=IN IP4 37.100.100.100
b=TIAS:64000
a=rtcp:4001 IN IP4 37.100.100.100
a=ssrc:801129593 cname:67e4069c62f84408
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
08:43:37.679712 IP (tos 0x10, ttl 255, id 48327, offset 0, flags [none], proto
UDP (17), length 536)
ip-172-17-203-102.eu-central-1.compute.internal.sip >
ip-172-17-200-20.eu-central-1.compute.internal.sip: [bad udp cksum 0xedb3 ->
0x24e3!] SIP, length: 508
ACK sip:172.17.200.20:5060 SIP/2.0
Via: SIP/2.0/UDP
3.99.99.99:5060;branch=z9hG4bK37f1.a5e72f99089ff1b4e282c09caaaf3a43.0
Via: SIP/2.0/UDP
37.100.100.100:56065;received=37.100.100.100;rport=56065;branch=z9hG4bKPj1df5c9a4fada4c28a9ad5f7540513287
Max-Forwards: 69
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
CSeq: 12029 ACK
Content-Length: 0
08:43:37.681851 IP (tos 0x10, ttl 255, id 48328, offset 0, flags [none], proto
UDP (17), length 1021)
ip-172-17-203-102.eu-central-1.compute.internal.sip >
ip-172-17-200-20.eu-central-1.compute.internal.sip: [bad udp cksum 0xef98 ->
0x0f9f!] SIP, length: 993
UPDATE sip:172.17.200.20:5060 SIP/2.0
Via: SIP/2.0/UDP
3.99.99.99:5060;branch=z9hG4bKc6f1.36bcd0fe1b2325ed4f511c4581b7e118.0
Via: SIP/2.0/UDP
37.100.100.100:56065;received=37.100.100.100;rport=56065;branch=z9hG4bKPj553ecda2f38547e884efd5bba4cc313d
Max-Forwards: 69
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
Contact: <sip:[email protected]:56065;ob>
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
CSeq: 12030 UPDATE
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800;refresher=uac
Min-SE: 90
Content-Type: application/sdp
Content-Length: 305
v=0
o=- 3846649415 3846649416 IN IP4 3.99.99.99
s=pjmedia
b=AS:84
t=0 0
a=X-nat:1
m=audio 23000 RTP/AVP 8 101
c=IN IP4 3.99.99.99
b=TIAS:64000
a=ssrc:801129593 cname:67e4069c62f84408
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:23001
a=ptime:20
08:43:37.682503 IP (tos 0xb8, ttl 64, id 27003, offset 0, flags [DF], proto UDP
(17), length 1085)
ip-172-17-200-20.eu-central-1.compute.internal.sip >
ip-172-17-203-102.eu-central-1.compute.internal.sip: [udp sum ok] SIP, length:
1057
SIP/2.0 200 OK
Via: SIP/2.0/UDP
3.99.99.99:5060;rport=5060;received=172.17.203.102;branch=z9hG4bKc6f1.36bcd0fe1b2325ed4f511c4581b7e118.0
Via: SIP/2.0/UDP
37.100.100.100:56065;rport=56065;received=37.100.100.100;branch=z9hG4bKPj553ecda2f38547e884efd5bba4cc313d
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
CSeq: 12030 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:172.17.200.20:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 16.9.0
Content-Type: application/sdp
Content-Length: 239
v=0
o=- 3846649415 3846649418 IN IP4 172.17.200.20
s=Asterisk
c=IN IP4 172.17.200.20
t=0 0
m=audio 10786 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
08:43:37.683675 IP (tos 0x10, ttl 255, id 5625, offset 0, flags [none], proto
UDP (17), length 970)
ip-172-17-203-102.eu-central-1.compute.internal.sip >
100.100.100.37.rev.sfr.net.56065: [bad udp cksum 0x74b9 -> 0xf0ba!] SIP,
length: 942
SIP/2.0 200 OK
Via: SIP/2.0/UDP
37.100.100.100:56065;rport=56065;received=37.100.100.100;branch=z9hG4bKPj553ecda2f38547e884efd5bba4cc313d
Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
From: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
To:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
CSeq: 12030 UPDATE
Session-Expires: 1800;refresher=uac
Require: timer
Contact: <sip:172.17.200.20:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE,
CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Server: Asterisk PBX 16.9.0
Content-Type: application/sdp
Content-Length: 247
v=0
o=- 3846649415 3846649418 IN IP4 3.99.99.99
s=Asterisk
c=IN IP4 3.99.99.99
t=0 0
m=audio 23020 RTP/AVP 8 101
a=maxptime:150
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp:23021
a=ptime:20
## Call is established (RTP media between asterisk<->kamailio flowing through
incorrect network too, should be private).
But if I let the IPBX hangup, nothing shows here, because asterisk is trying to
send BYE to the public IP
## Capture from Asterisk box:
[Nov 23 09:43:53] <--- Transmitting SIP request (488 bytes) to
UDP:3.99.99.99:5060 --->
[Nov 23 09:43:53] BYE sip:[email protected]:56065;ob SIP/2.0
[Nov 23 09:43:53] Via: SIP/2.0/UDP
172.17.200.20:5060;rport;branch=z9hG4bKPjbcb73473-6c8c-4d2d-9e19-58b4f1d282e5
[Nov 23 09:43:53] From:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
[Nov 23 09:43:53] To: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
[Nov 23 09:43:53] Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
[Nov 23 09:43:53] CSeq: 28535 BYE
[Nov 23 09:43:53] Route: <sip:3.99.99.99;lr>
[Nov 23 09:43:53] Reason: Q.850;cause=0
[Nov 23 09:43:53] Max-Forwards: 70
[Nov 23 09:43:53] User-Agent: Asterisk PBX 16.9.0
[Nov 23 09:43:53] Content-Length: 0
[Nov 23 09:43:53]
[Nov 23 09:43:53]
[Nov 23 09:43:53] -- Remote UNIX connection
[Nov 23 09:43:53] -- Remote UNIX connection disconnected
[Nov 23 09:43:53] -- Remote UNIX connection
[Nov 23 09:43:53] -- Remote UNIX connection disconnected
[Nov 23 09:43:55] <--- Transmitting SIP request (488 bytes) to
UDP:3.99.99.99:5060 --->
[Nov 23 09:43:55] BYE sip:[email protected]:56065;ob SIP/2.0
[Nov 23 09:43:55] Via: SIP/2.0/UDP
172.17.200.20:5060;rport;branch=z9hG4bKPjbcb73473-6c8c-4d2d-9e19-58b4f1d282e5
[Nov 23 09:43:55] From:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
[Nov 23 09:43:55] To: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
[Nov 23 09:43:55] Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
[Nov 23 09:43:55] CSeq: 28535 BYE
[Nov 23 09:43:55] Route: <sip:3.99.99.99;lr>
[Nov 23 09:43:55] Reason: Q.850;cause=0
[Nov 23 09:43:55] Max-Forwards: 70
[Nov 23 09:43:55] User-Agent: Asterisk PBX 16.9.0
[Nov 23 09:43:55] Content-Length: 0
[Nov 23 09:43:55]
[Nov 23 09:43:55]
[Nov 23 09:43:59] <--- Transmitting SIP request (488 bytes) to
UDP:3.99.99.99:5060 --->
[Nov 23 09:43:59] BYE sip:[email protected]:56065;ob SIP/2.0
[Nov 23 09:43:59] Via: SIP/2.0/UDP
172.17.200.20:5060;rport;branch=z9hG4bKPjbcb73473-6c8c-4d2d-9e19-58b4f1d282e5
[Nov 23 09:43:59] From:
<sip:[email protected]>;tag=826bcfed-1430-4c66-b17a-f66ca997066e
[Nov 23 09:43:59] To: "test user"
<sip:[email protected]>;tag=e75ca7ae0a8748e9bb6be82d523a568d
[Nov 23 09:43:59] Call-ID: 445085acf1d7477a9e0b51f25e60c1dd
[Nov 23 09:43:59] CSeq: 28535 BYE
[Nov 23 09:43:59] Route: <sip:3.99.99.99;lr>
[Nov 23 09:43:59] Reason: Q.850;cause=0
[Nov 23 09:43:59] Max-Forwards: 70
[Nov 23 09:43:59] User-Agent: Asterisk PBX 16.9.0
[Nov 23 09:43:59] Content-Length: 0
kamailio-log.log
Description: Binary data
kamailio0.cfg
Description: Binary data
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