SIP Calls Dropping At 15 Minutes
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”
|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”
|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”
|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?
3) Should OpenSIPS be responding differently to the INVITE at 15 minutes?
4 thoughts on - SIP Calls Dropping At 15 Minutes
Looks like session timers are kicking in and a Re-Invite is being sent.
I would disable them in sip.conf and try again:
session-timers=refuse
http://doxygen.asterisk.org/trunk/sip_session_timers.html
This appears to work, but it feels wrong. Shouldn’t I be configuring Asterisk or OpenSIPS to respond or receive the re-invite correctly?
probably opensips isn’t forwarding the re-invite to the endpoint.. set re-invites up and run sip tracing on your opensips and asterisk box and see what happens when the reinvites arrive.
Maybe but I would not lose sleep over having session timers disabled if it fixes your problem.