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?
3) Should OpenSIPS be responding differently to the INVITE at 15 minutes?
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards [email protected] Voice: +1-760-468-3867 PST
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