Hello,
as already discussed, it seems to be more a problem of the SIP retransmission timer.
This can be changed with TM parameter. But I doubt this is really the solution to your
problem, usually it is not necessary to change this timer.
My suggestion would be to look into the network trace what actually happens on the SIP
level. Then you can probably better understand what the problem is.
You can differentiate between a retransmitted INVITE and a real re-INVITE easily – the
re-INVITE has a “To tag”, the retransmitted INVITE (for the initial call setup) not.
Cheers,
Henning
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Kamailio services –
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From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Anuran Barman
Sent: Wednesday, December 18, 2019 7:30 AM
To: Patrick Wakano <pwakano(a)gmail.com>
Cc: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>
Subject: Re: [SR-Users] How to reduce re-INVITE time in Kamailio
@Henning Westerholt
So I loaded the sst.so module and setup the params it said to set like below:
modparam("dialog", "timeout_avp", "$avp(i:10)")
# Set the sst modules timeout_avp to be the same value
modparam("sst", "timeout_avp", "$avp(i:10)")
modparam("sst", "min_se", 10)
and also set the flag in INVITE method in request_route {}
# account only INVITEs
if (is_method("INVITE")) {
setflag(FLT_ACC); # do accounting
setflag(5);
}
and also set the t_set_retr in route{}
route {
dlg_manage();
t_set_retr(10, 0);
if (is_method("BYE")) {
$var(elapsed) = ( $Ts - $dlg(start_ts) );
xlog ("L_INFO", "Completed $dlg(from_uri) to $dlg(to_uri)
-
$var(elapsed) duration");
}
}
but still I am getting the INVITE after 57-59 seconds later. How do I do this?
On Wed, Dec 18, 2019 at 9:40 AM Anuran Barman
<anuranbarman@gmail.com<mailto:anuranbarman@gmail.com>> wrote:
Thank you all for the detailed information. Let me try both changing the transmission time
and also suspending the invite transaction. That NAT port close did not hit my mind.
Thanks for putting light on that.
On Wed, 18 Dec 2019 at 9:35 AM, Patrick Wakano
<pwakano@gmail.com<mailto:pwakano@gmail.com>> wrote:
As I understood Anuran is after the SIP retransmission timer.
From the description, I think the first INVITE, which I assume triggers the pushkit voip
to IOS stuff, is missed, then (with default timer T1) after 500ms the first retransmission
is done, then after another 500ms you get the 2nd, then after 1s the 3rd, after 2s the
4th, after 4s the 5th, after 8s the 6th and finally after 16s the 7th. This is probably
why if you wait 10/15s it works, because probably the 5th or 6th retransmission hits your
client. If you wait just 5s you can fall under one window without retransmission (if you
miss the 5th you have to wait 8s for the next retransmission).
You can change the timer T1 in the TM module and hope the retransmissions do the job for
you. It might work but doesn't look safe to me. I have no idea how this pushkit voip
event works, but you may have problems of NAT ports being closed due to lack of SIP
activity, then the INVITE will never reach your destination anyway.
The way to go seems what David has suggested about suspending the Invite transaction while
your client start-up and register, but I don't know details about this....
Cheers,
Patrick Wakano
On Wed, 18 Dec 2019 at 07:34, Henning Westerholt
<hw@skalatan.de<mailto:hw@skalatan.de>> wrote:
As mentioned, it is more a User agent related topic. But there is a module that provide
related functions in Kamailio:
https://kamailio.org/docs/modules/stable/modules/sst.html
Cheers,
Henning
Am 17.12.19 um 19:51 schrieb Anuran Barman:
Thanks for the reply. Sharing doc links of Session timer settings related to Kamailio
specifically will be greatly appreciated.
On Wed, 18 Dec 2019 at 12:18 AM, Yuriy Gorlichenko
<ovoshlook@gmail.com<mailto:ovoshlook@gmail.com>> wrote:
Session-Expires header ( Just remembered right name) will help you
See rfc 4028
https://tools.ietf.org/html/rfc4028#page-6
On Tue, 17 Dec 2019, 19:44 Yuriy Gorlichenko,
<ovoshlook@gmail.com<mailto:ovoshlook@gmail.com>> wrote:
Hi. It is no a kamailio question but sip question itself. You can manage how often te
invite will be by using session-timers. It will help you to setup time range between
reInvite ir whatever message you will use as keepalive
On Tue, 17 Dec 2019, 19:34 Anuran Barman,
<anuranbarman@gmail.com<mailto:anuranbarman@gmail.com>> wrote:
Hi, I am using Kamailio for my SIP Phone app. My main SIP client is the iOS app. Till now
everything is working fine regarding the calling. But the problem is when the app is
killed I am not able to pick up incoming call. The thing is, when app is killed I am
sending PushKit VOIP message to my ios app to wake it up. App is waking up. But by the
time app is waking up, the original INVITE message got missed up. So according to SIP
protocol it will re-INVITE after some time. Currently if I keep my app open for 10-15 secs
after it wakes up, I am able to get the re-INVITE and everything works fine just as it was
supposed to. But waiting 10-15 sec on the app is not practical in any sense.
So my question is there any way I can reduce the re-INVITE time to like 5/6 seconds so
that as soon as the app open it will wait for 5/6 seconds (practical upto some extent) and
it will get the call ? Or what kamailio is using as the interval time is a SIP standard
and can not be changed? If possible to do so, how can I do that? Please help.
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