I have a problem where SIP calls from some providers are dropping at
15 minutes.
The topology is: Client sends calls to a host running OpenSIPS,
OpenSIPS sends calls to an Asterisk server.
Below,
'Client' is the IP address of the client's host (running
FPBX-2.8.1(1.8.20.0)
'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls
'Asterisk' is the IP address of my host running Asterisk 11.17.1.
The relevant snippet of opensips.cfg is:
# 317
if ($rU =~ '317*')
{
ds_select_dst(
'02' # set-id (in dispatcher.list)
, '4' # algorithm (4 = round-robin)
);
forward();
return;
}
where set-id 02 is 'sip:Asterisk:5061'
The 'Flow' diagram from Wireshark from a tcpdump on the OpenSIPS host
follows, hopefully the email clients will not mung it too much.
|Time | Client | Asterisk |
| | | OpenSIPS | |7.158764
| INVITE SDP (g711U g7 | |SIP From:
"760xxxxxxx" <sip:760xxxxxxx@client To:<sip:317xxxxxxx@OpenSIPS
| |(5060) ------------------> (5060) | |
|7.159003 | | INVITE SDP (g711U g7
|SIP Request
| | |(5060) ------------------> (5061) |
|7.161857 | | 100 Trying|
|SIP Status
| | |(5060) <------------------ (5061) |
|7.161958 | 100 Trying| | |SIP Status
| |(5060) <------------------ (5060) | |
|7.538268 | | 200 OK SDP (g711U te
|SIP Status
| | |(5060) <------------------ (5061) |
|7.538411 | 200 OK SDP (g711U te | |SIP Status
| |(5060) <------------------ (5060) | |
|7.585703 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|7.585941 | | ACK | |SIP
Request
| | |(5060) ------------------> (5061) |
|7.586548 | INVITE SDP (g711U te | |SIP
From: "760xxxxxxx" <sip:760xxxxxxx@client To:<sip:317xxxxxxx@OpenSIPS
| |(5060) ------------------> (5060) | |
|7.586726 | | INVITE SDP (g711U te
|SIP Request
| | |(5060) ------------------> (5061) |
|7.587792 | | 100 Trying|
|SIP Status
| | |(5060) <------------------ (5061) |
|7.587922 | 100 Trying| | |SIP Status
| |(5060) <------------------ (5060) | |
|7.588003 | | 200 OK SDP (g711U te
|SIP Status
| | |(5060) <------------------ (5061) |
|7.588081 | 200 OK SDP (g711U te | |SIP Status
| |(5060) <------------------ (5060) | |
|7.635401 | ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|7.635674 | | ACK | |SIP
Request
| | |(5060) ------------------> (5061) |
|907.588019| | INVITE SDP (g711U te
|SIP Request
| | |(5060) <------------------ (5061) |
|907.590138| | 100 Giving a try
|SIP Status
| | |(5060) ------------------> (5061) |
|907.590261| | INVITE SDP (g711U te
|SIP Request
| | |(5060) ------------------> (5061) |
|907.591294| | 481 Call/Transaction
|SIP Status
| | |(5060) <------------------ (5061) |
|907.591420| | ACK | |SIP
Request
| | |(5060) ------------------> (5061) |
|907.591467| | 481 Call/Transaction
|SIP Status
| | |(5060) ------------------> (5061) |
|907.592140| | ACK | |SIP
Request
| | |(5060) <------------------ (5061) |
|907.867923| | BYE | |SIP
Request
| | |(5060) <------------------ (5061) |
|907.868231| | BYE | |SIP
Request
| | |(5060) ------------------> (5061) |
|907.869337| | 481 Call leg/transac
|SIP Status
| | |(5060) <------------------ (5061) |
|907.869412| | 481 Call leg/transac
|SIP Status
| | |(5060) ------------------> (5061) |
|1140.290782| INVITE SDP (g711U te | |SIP
From: "760xxxxxxx" <sip:760xxxxxxx@client To:<sip:317xxxxxxx@OpenSIPS
| |(5060) ------------------> (5060) | |
|1140.291032| | INVITE SDP (g711U
te |SIP Request
| | |(5060) ------------------> (5061) |
|1140.292338| | 481
Call/Transaction |SIP Status
| | |(5060) <------------------ (5061) |
|1140.292445| 481 Call/Transaction | |SIP
Status
| |(5060) <------------------ (5060) | |
|1140.339890| ACK | | |SIP Request
| |(5060) ------------------> (5060) | |
|1140.340011| | ACK | |SIP
Request
| | |(5060) ------------------> (5061) |
|1140.452758| BYE | | |SIP Request
| |(5060) ------------------> (5060) | |
|1140.452893| | BYE | |SIP
Request
| | |(5060) ------------------> (5061) |
|1140.453470| | 481 Call
leg/transac |SIP Status
| | |(5060) <------------------ (5061) |
|1140.453541| 481 Call leg/transac | |SIP
Status
| |(5060) <------------------ (5060) | |
My knowledge of SIP is limited, but it appears that Asterisk is
sending an INVITE at 907.588019, OpenSIPS responds with an INVITE at
907.590261, but Asterisk thinks the call doesn't exist and sends a BYE.
1) Is a 'ds_select_dst()' followed by a 'forward()' the right way to
route calls in OpenSIPS? It works most of the time.
2) Can (or should) I configure Asterisk to not send the INVITE at 15
minutes?
