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.
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 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. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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 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 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. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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 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 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 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. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
It is not kamailio manage thing. It is end to end Negotiation. In other words it should be placed by UAc to another UAc. Not by sip-proxy
On Tue, 17 Dec 2019, 19:52 Anuran Barman, anuranbarman@gmail.com wrote:
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 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 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 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. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
You can also suspend the invite transaction while the other phome registers, and when it does verify it has a pending call in a suspended transaction, then resume the transaction
On Tue, 17 Dec 2019 at 19:13, Yuriy Gorlichenko ovoshlook@gmail.com wrote:
It is not kamailio manage thing. It is end to end Negotiation. In other words it should be placed by UAc to another UAc. Not by sip-proxy
On Tue, 17 Dec 2019, 19:52 Anuran Barman, anuranbarman@gmail.com wrote:
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 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 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 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. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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.commailto: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.commailto: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.commailto: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. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/
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 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 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 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 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. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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 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 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 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 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 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. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
@Henning Westerholt So I loaded the sst.so module and setup the params it said to set like below:
*modparam("dialog", "timeout_avp", "$avp(i:10)")# Set the sst modules timeout_avp to be the same valuemodparam("sst", "timeout_avp", "$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 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 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 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 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 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 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. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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
-- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.comhttps://gilawa.com/
From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Anuran Barman Sent: Wednesday, December 18, 2019 7:30 AM To: Patrick Wakano pwakano@gmail.com Cc: Kamailio (SER) - Users Mailing List sr-users@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.commailto: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.commailto: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.demailto: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.commailto: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.commailto: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.commailto: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. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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Kamailio services - https://skalatan.de/services
Henning Westerholt - https://skalatan.de/blog/ _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.orgmailto:sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
You are facing the problem the wrong way.
For working with softphones, that must be "wake up", your logic must be like this:
... if request goes to a push_client: hold_request() send_push_notification() endif ...
if request(REGISTER) and client is marked as push_client: if holded_requests(): unhold_request(): endif endif ...
All the workflow with push_clients, does nothing to do with SST or any other timer
From: "Anuran Barman" anuranbarman@gmail.com To: "Patrick Wakano" pwakano@gmail.com Cc: "Kamailio (SER) - Users Mailing List" sr-users@lists.kamailio.org Sent: Wednesday, December 18, 2019 6:30:16 AM 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 < [ mailto:anuranbarman@gmail.com | 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 < [ mailto:pwakano@gmail.com | pwakano@gmail.com ] > wrote:
BQ_BEGIN
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 < [ mailto:hw@skalatan.de | hw@skalatan.de ] > wrote:
BQ_BEGIN
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 | https://kamailio.org/docs/modules/stable/modules/sst.html ]
Cheers,
Henning
Am 17.12.19 um 19:51 schrieb Anuran Barman:
BQ_BEGIN
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 < [ mailto:ovoshlook@gmail.com | ovoshlook@gmail.com ] > wrote:
BQ_BEGIN
Session-Expires header ( Just remembered right name) will help you See rfc 4028 [ https://tools.ietf.org/html/rfc4028#page-6 | https://tools.ietf.org/html/rfc4028#page-6 ]
On Tue, 17 Dec 2019, 19:44 Yuriy Gorlichenko, < [ mailto:ovoshlook@gmail.com | ovoshlook@gmail.com ] > wrote:
BQ_BEGIN
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, < [ mailto:anuranbarman@gmail.com | anuranbarman@gmail.com ] > wrote:
BQ_BEGIN
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. _______________________________________________ Kamailio (SER) - Users Mailing List [ mailto:sr-users@lists.kamailio.org | sr-users@lists.kamailio.org ] [ https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users | https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ]
BQ_END
_______________________________________________ Kamailio (SER) - Users Mailing List [ mailto:sr-users@lists.kamailio.org | sr-users@lists.kamailio.org ] [ https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users | https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ]
BQ_END
_______________________________________________ Kamailio (SER) - Users Mailing List [ mailto:sr-users@lists.kamailio.org | sr-users@lists.kamailio.org ] [ https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users | https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ]
BQ_END
-
Anuran Barman -
David Villasmil -
Henning Westerholt -
Patrick Wakano -
Raúl Alexis Betancor Santana -
Yuriy Gorlichenko