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 at client To:<sip:317xxxxxxx at
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 at client To:<sip:317xxxxxxx at
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 at client To:<sip:317xxxxxxx
at 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 sedwards at sedwards.com Voice: +1-760-468-3867 PST
On 11/20/15 11:13 AM, Steve Edwards wrote:> 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 at client To:<sip:317xxxxxxx at 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 at client To:<sip:317xxxxxxx at 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 at client To:<sip:317xxxxxxx at 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?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> > 3) Should OpenSIPS be responding differently to the INVITE at 15 minutes? >-- Technical Support http://www.cellroute.net
> On 11/20/15 11:13 AM, Steve Edwards wrote:>> 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.>> 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?On Sat, 21 Nov 2015, Andres wrote:> 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>> 3) Should OpenSIPS be responding differently to the INVITE at 15 >> minutes?This appears to work, but it feels wrong. Shouldn't I be configuring Asterisk or OpenSIPS to respond or receive the re-invite correctly? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST