[asterisk-users] SIP calls dropping at 15 minutes
Andres
andres at telesip.net
Sat Nov 21 12:19:48 CST 2015
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?
>
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