date:20191218

Re: [SR-Users] TLS testing (with sipp)

2019-12-18 Thread Patrick Wakano

You just need to inform the certificate and key. It is informed in the docs
(http://sipp.sourceforge.net/doc/reference.html#TLS+mono+socket)
Use sipp options: t -l1 -tls_cert user-cert.pem -tls_key user-privkey.pem



On Thu, 19 Dec 2019 at 03:33, Sebastian Damm  wrote:

> Hi,
>
> I'm trying to construct an end-to-end encrypted signalling test
> through our setup. I thought I could use sipp for that, as it supports
> TLS according to the man page. However, when I try to run it, I get
> this error:
>
> FI_init_ssl_context: SSL_CTX_use_certificate_file failed.
>
> I searched the web; however, all similar questions end up without
> answers. I tried specifying a local key and cert without success. I'd
> think I should not need a cert for my client, though.
>
> Has anyone ever successfully conducted an automated TLS test? I'm open
> to using a different tool if necessary.
>
> Thanks for all hints or examples.
>
> Regards,
> Sebastian
>
> --
> Sebastian Damm
> Voice Engineer
> __
> sipgate GmbH
> Gladbacher Straße 74 | 40219 Düsseldorf
>
> ___
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> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
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Re: [SR-Users] TLS testing (with sipp)

2019-12-18 Thread Julien Chavanton

Hi Sebastian, this is off topic for the Kamailio mailing list.

You can use Voip_patrol :
https://github.com/jchavanton/voip_patrol

Once you have your certificate, key and ca_list in default location
./voip_patrol -c ./xml/tls.xml

[18:24:51.800][INFO] main: TLS tcfg.tlsConfig.ca_list  :tls/ca_list.pem
[18:24:51.800][INFO] main: TLS tcfg.tlsConfig.certFile
:tls/certificate.pem
[18:24:51.800][INFO] main: TLS tcfg.tlsConfig.privKeyFile  :tls/key.pem

tls.xml














On Wed, Dec 18, 2019 at 8:34 AM Sebastian Damm  wrote:

> Hi,
>
> I'm trying to construct an end-to-end encrypted signalling test
> through our setup. I thought I could use sipp for that, as it supports
> TLS according to the man page. However, when I try to run it, I get
> this error:
>
> FI_init_ssl_context: SSL_CTX_use_certificate_file failed.
>
> I searched the web; however, all similar questions end up without
> answers. I tried specifying a local key and cert without success. I'd
> think I should not need a cert for my client, though.
>
> Has anyone ever successfully conducted an automated TLS test? I'm open
> to using a different tool if necessary.
>
> Thanks for all hints or examples.
>
> Regards,
> Sebastian
>
> --
> Sebastian Damm
> Voice Engineer
> __
> sipgate GmbH
> Gladbacher Straße 74 | 40219 Düsseldorf
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
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Re: [SR-Users] Kamailio SIP TLS Troubleshooting

2019-12-18 Thread Henning Westerholt

Hello,


Sorry for the short previous reply, my fault.

Kamailio can act as TLS Server and Client with the TLS Module. So it can of 
course process and handle encryption and decryption for TLS Transport

Cheers,

Henning



--
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://skalatan.de/services

From: Sudhakar Parasher 
Sent: Wednesday, December 18, 2019 6:16:35 PM
To: Henning Westerholt 
Cc: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Kamailio SIP TLS Troubleshooting

Ok, thanks. But do you mean to say Kamailio doesn't provide any module to 
decrypt the Encrypted SIP  packets?

On Wed, Dec 18, 2019 at 10:22 PM Henning Westerholt 
mailto:h...@skalatan.de>> wrote:

Hello,



if the requirement is to send this from android – then maybe ask at the android 
client project or vendor. I can’t help you here, sorry.



If the requirement is to send this from Kamailio, there were already some 
suggestions done in this e-mail thread.



Cheers,



Henning



--

Henning Westerholt – https://skalatan.de/blog/

Kamailio services – https://gilawa.com



From: Sudhakar Parasher 
mailto:sudhakarparas...@gmail.com>>
Sent: Wednesday, December 18, 2019 5:49 PM
To: Henning Westerholt mailto:h...@skalatan.de>>
Cc: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Kamailio SIP TLS Troubleshooting



Hi Henning



My SIP message which are received by my kamailio is already TLS Encrypted from 
my android Client ( Handset) but I have requirement to forward

these SIP messages from android client to a port mirrored destination ( say a 
Probe).



Cheers

Sudhakar Parasher



On Wed, Dec 18, 2019 at 9:39 PM Henning Westerholt 
mailto:h...@skalatan.de>> wrote:

Hello,



Not sure if I understood you correctly. The SIP message data is not encrypted 
inside Kamailio, otherwise it would not work to parse the packet etc.. So you 
can access it and also forward it unencrypted to another destination.



Cheers,



Henning



--

Henning Westerholt – https://skalatan.de/blog/

Kamailio services – https://gilawa.com



From: Sudhakar Parasher 
mailto:sudhakarparas...@gmail.com>>
Sent: Wednesday, December 18, 2019 5:06 PM
To: Henning Westerholt mailto:h...@skalatan.de>>
Cc: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Kamailio SIP TLS Troubleshooting



Thanks Henning

But i need to send unencrypted TLS packets to mirrored destination

Any ideas??



On Wed, 18 Dec 2019, 19:15 Henning Westerholt, 
mailto:h...@skalatan.de>> wrote:

Hello,



you could mirror the SIP traffic also with t_replicate function from the TM 
module in the cfg.



Cheers,



Henning



--

Henning Westerholt – https://skalatan.de/blog/

Kamailio services – https://gilawa.com



From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Sudhakar Parasher
Sent: Tuesday, December 17, 2019 9:01 AM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] Kamailio SIP TLS Troubleshooting



Hi Experts.



Need your opinion on below part related to kamailio



I am using SIP Kamailio module Version for the SIP Registrations from the  
Client mobile phones using Android and IOS which is Over TLSv1.2 , So in 
nutshell the SIP Signaling is TLS version 1.2 encrypted and Customer is 
requesting for the forward of the  unencrypted SIP signaling messages 
especially the SIP Registration messages  to their probe solutions without 
storing the SIP messages locally in the Kamailio Database



Question  :Would it not be possible for Kamailio   imply “mirror” all the SIP 
messages related to all SIP REGISTER dialogs (after removing TLS for inbound 
messages and prior to adding TLS for outbound message) to a designated IP 
Address and Port?  No need for local database storage.  No need for 
encapsulation.  No mirroring of other messages/dialogs.



Cheers

Sudhakar
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Re: [SR-Users] Kamailio SIP TLS Troubleshooting

2019-12-18 Thread Henning Westerholt

Hello


--
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://skalatan.de/services


From: Sudhakar Parasher 
Sent: Wednesday, December 18, 2019 6:16:35 PM
To: Henning Westerholt 
Cc: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Kamailio SIP TLS Troubleshooting

Ok, thanks. But do you mean to say Kamailio doesn't provide any module to 
decrypt the Encrypted SIP  packets?

On Wed, Dec 18, 2019 at 10:22 PM Henning Westerholt 
mailto:h...@skalatan.de>> wrote:

Hello,



if the requirement is to send this from android – then maybe ask at the android 
client project or vendor. I can’t help you here, sorry.



If the requirement is to send this from Kamailio, there were already some 
suggestions done in this e-mail thread.



Cheers,



Henning



--

Henning Westerholt – https://skalatan.de/blog/

Kamailio services – https://gilawa.com



From: Sudhakar Parasher 
mailto:sudhakarparas...@gmail.com>>
Sent: Wednesday, December 18, 2019 5:49 PM
To: Henning Westerholt mailto:h...@skalatan.de>>
Cc: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Kamailio SIP TLS Troubleshooting



Hi Henning



My SIP message which are received by my kamailio is already TLS Encrypted from 
my android Client ( Handset) but I have requirement to forward

these SIP messages from android client to a port mirrored destination ( say a 
Probe).



Cheers

Sudhakar Parasher



On Wed, Dec 18, 2019 at 9:39 PM Henning Westerholt 
mailto:h...@skalatan.de>> wrote:

Hello,



Not sure if I understood you correctly. The SIP message data is not encrypted 
inside Kamailio, otherwise it would not work to parse the packet etc.. So you 
can access it and also forward it unencrypted to another destination.



Cheers,



Henning



--

Henning Westerholt – https://skalatan.de/blog/

Kamailio services – https://gilawa.com



From: Sudhakar Parasher 
mailto:sudhakarparas...@gmail.com>>
Sent: Wednesday, December 18, 2019 5:06 PM
To: Henning Westerholt mailto:h...@skalatan.de>>
Cc: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Kamailio SIP TLS Troubleshooting



Thanks Henning

But i need to send unencrypted TLS packets to mirrored destination

Any ideas??



On Wed, 18 Dec 2019, 19:15 Henning Westerholt, 
mailto:h...@skalatan.de>> wrote:

Hello,



you could mirror the SIP traffic also with t_replicate function from the TM 
module in the cfg.



Cheers,



Henning



--

Henning Westerholt – https://skalatan.de/blog/

Kamailio services – https://gilawa.com



From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Sudhakar Parasher
Sent: Tuesday, December 17, 2019 9:01 AM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] Kamailio SIP TLS Troubleshooting



Hi Experts.



Need your opinion on below part related to kamailio



I am using SIP Kamailio module Version for the SIP Registrations from the  
Client mobile phones using Android and IOS which is Over TLSv1.2 , So in 
nutshell the SIP Signaling is TLS version 1.2 encrypted and Customer is 
requesting for the forward of the  unencrypted SIP signaling messages 
especially the SIP Registration messages  to their probe solutions without 
storing the SIP messages locally in the Kamailio Database



Question  :Would it not be possible for Kamailio   imply “mirror” all the SIP 
messages related to all SIP REGISTER dialogs (after removing TLS for inbound 
messages and prior to adding TLS for outbound message) to a designated IP 
Address and Port?  No need for local database storage.  No need for 
encapsulation.  No mirroring of other messages/dialogs.



Cheers

Sudhakar
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Re: [SR-Users] Kamailio SIP TLS Troubleshooting

2019-12-18 Thread Henning Westerholt

Hello,

if the requirement is to send this from android – then maybe ask at the android 
client project or vendor. I can’t help you here, sorry.

If the requirement is to send this from Kamailio, there were already some 
suggestions done in this e-mail thread.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: Sudhakar Parasher 
Sent: Wednesday, December 18, 2019 5:49 PM
To: Henning Westerholt 
Cc: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Kamailio SIP TLS Troubleshooting

Hi Henning

My SIP message which are received by my kamailio is already TLS Encrypted from 
my android Client ( Handset) but I have requirement to forward
these SIP messages from android client to a port mirrored destination ( say a 
Probe).

Cheers
Sudhakar Parasher

On Wed, Dec 18, 2019 at 9:39 PM Henning Westerholt 
mailto:h...@skalatan.de>> wrote:
Hello,

Not sure if I understood you correctly. The SIP message data is not encrypted 
inside Kamailio, otherwise it would not work to parse the packet etc.. So you 
can access it and also forward it unencrypted to another destination.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: Sudhakar Parasher 
mailto:sudhakarparas...@gmail.com>>
Sent: Wednesday, December 18, 2019 5:06 PM
To: Henning Westerholt mailto:h...@skalatan.de>>
Cc: Kamailio (SER) - Users Mailing List 
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] Kamailio SIP TLS Troubleshooting

Thanks Henning
But i need to send unencrypted TLS packets to mirrored destination
Any ideas??

On Wed, 18 Dec 2019, 19:15 Henning Westerholt, 
mailto:h...@skalatan.de>> wrote:
Hello,

you could mirror the SIP traffic also with t_replicate function from the TM 
module in the cfg.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Sudhakar Parasher
Sent: Tuesday, December 17, 2019 9:01 AM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] Kamailio SIP TLS Troubleshooting

Hi Experts.

Need your opinion on below part related to kamailio

I am using SIP Kamailio module Version for the SIP Registrations from the  
Client mobile phones using Android and IOS which is Over TLSv1.2 , So in 
nutshell the SIP Signaling is TLS version 1.2 encrypted and Customer is 
requesting for the forward of the  unencrypted SIP signaling messages 
especially the SIP Registration messages  to their probe solutions without 
storing the SIP messages locally in the Kamailio Database

Question  :Would it not be possible for Kamailio   imply “mirror” all the SIP 
messages related to all SIP REGISTER dialogs (after removing TLS for inbound 
messages and prior to adding TLS for outbound message) to a designated IP 
Address and Port?  No need for local database storage.  No need for 
encapsulation.  No mirroring of other messages/dialogs.

Cheers
Sudhakar
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[SR-Users] TLS testing (with sipp)

2019-12-18 Thread Sebastian Damm

Hi,

I'm trying to construct an end-to-end encrypted signalling test
through our setup. I thought I could use sipp for that, as it supports
TLS according to the man page. However, when I try to run it, I get
this error:

FI_init_ssl_context: SSL_CTX_use_certificate_file failed.

I searched the web; however, all similar questions end up without
answers. I tried specifying a local key and cert without success. I'd
think I should not need a cert for my client, though.

Has anyone ever successfully conducted an automated TLS test? I'm open
to using a different tool if necessary.

Thanks for all hints or examples.

Regards,
Sebastian

-- 
Sebastian Damm
Voice Engineer
__
sipgate GmbH
Gladbacher Straße 74 | 40219 Düsseldorf

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Re: [SR-Users] Kamailio SIP TLS Troubleshooting

2019-12-18 Thread Henning Westerholt

Hello,

Not sure if I understood you correctly. The SIP message data is not encrypted 
inside Kamailio, otherwise it would not work to parse the packet etc.. So you 
can access it and also forward it unencrypted to another destination.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: Sudhakar Parasher 
Sent: Wednesday, December 18, 2019 5:06 PM
To: Henning Westerholt 
Cc: Kamailio (SER) - Users Mailing List 
Subject: Re: [SR-Users] Kamailio SIP TLS Troubleshooting

Thanks Henning
But i need to send unencrypted TLS packets to mirrored destination
Any ideas??

On Wed, 18 Dec 2019, 19:15 Henning Westerholt, 
mailto:h...@skalatan.de>> wrote:
Hello,

you could mirror the SIP traffic also with t_replicate function from the TM 
module in the cfg.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: sr-users 
mailto:sr-users-boun...@lists.kamailio.org>>
 On Behalf Of Sudhakar Parasher
Sent: Tuesday, December 17, 2019 9:01 AM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] Kamailio SIP TLS Troubleshooting

Hi Experts.

Need your opinion on below part related to kamailio

I am using SIP Kamailio module Version for the SIP Registrations from the  
Client mobile phones using Android and IOS which is Over TLSv1.2 , So in 
nutshell the SIP Signaling is TLS version 1.2 encrypted and Customer is 
requesting for the forward of the  unencrypted SIP signaling messages 
especially the SIP Registration messages  to their probe solutions without 
storing the SIP messages locally in the Kamailio Database

Question  :Would it not be possible for Kamailio   imply “mirror” all the SIP 
messages related to all SIP REGISTER dialogs (after removing TLS for inbound 
messages and prior to adding TLS for outbound message) to a designated IP 
Address and Port?  No need for local database storage.  No need for 
encapsulation.  No mirroring of other messages/dialogs.

Cheers
Sudhakar
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Re: [SR-Users] Kamailio SIP TLS Troubleshooting

2019-12-18 Thread Henning Westerholt

Hello,

you could mirror the SIP traffic also with t_replicate function from the TM 
module in the cfg.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

From: sr-users  On Behalf Of Sudhakar 
Parasher
Sent: Tuesday, December 17, 2019 9:01 AM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] Kamailio SIP TLS Troubleshooting

Hi Experts.

Need your opinion on below part related to kamailio

I am using SIP Kamailio module Version for the SIP Registrations from the  
Client mobile phones using Android and IOS which is Over TLSv1.2 , So in 
nutshell the SIP Signaling is TLS version 1.2 encrypted and Customer is 
requesting for the forward of the  unencrypted SIP signaling messages 
especially the SIP Registration messages  to their probe solutions without 
storing the SIP messages locally in the Kamailio Database

Question  :Would it not be possible for Kamailio   imply “mirror” all the SIP 
messages related to all SIP REGISTER dialogs (after removing TLS for inbound 
messages and prior to adding TLS for outbound message) to a designated IP 
Address and Port?  No need for local database storage.  No need for 
encapsulation.  No mirroring of other messages/dialogs.

Cheers
Sudhakar
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Re: [SR-Users] Kamailio SIP TLS Troubleshooting

2019-12-18 Thread Fred Posner

You can do this with the siptrace module and send to a homer system or other 
receiver. You can choose what messages to mirror/trace. 

-- Fred
direct/sms +1 (336) 439-3733


> On Dec 18, 2019, at 7:36 AM, Sudhakar Parasher  
> wrote:
> 
> 
> Hi Experts.
> 
> Need your opinion on below part related to kamailio
> 
> I am using SIP Kamailio module Version for the SIP Registrations from the  
> Client mobile phones using Android and IOS which is Over TLSv1.2 , So in 
> nutshell the SIP Signaling is TLS version 1.2 encrypted and Customer is 
> requesting for the forward of the  unencrypted SIP signaling messages 
> especially the SIP Registration messages  to their probe solutions without 
> storing the SIP messages locally in the Kamailio Database
> 
> Question  :Would it not be possible for Kamailio   imply “mirror” all the SIP 
> messages related to all SIP REGISTER dialogs (after removing TLS for inbound 
> messages and prior to adding TLS for outbound message) to a designated IP 
> Address and Port?  No need for local database storage.  No need for 
> encapsulation.  No mirroring of other messages/dialogs.
> 
> Cheers
> Sudhakar
> ___
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> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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[SR-Users] Kamailio SIP TLS Troubleshooting

2019-12-18 Thread Sudhakar Parasher

Hi Experts.

Need your opinion on below part related to kamailio

I am using SIP Kamailio module Version for the SIP Registrations from the
Client mobile phones using Android and IOS which is Over TLSv1.2 , So in
nutshell the SIP Signaling is TLS version 1.2 encrypted and Customer is
requesting for the forward of the  unencrypted SIP signaling messages
especially the SIP Registration messages  to their probe solutions without
storing the SIP messages locally in the Kamailio Database

Question  :Would it not be possible for Kamailio   imply “mirror” all the
SIP messages related to all SIP REGISTER dialogs (after removing TLS for
inbound messages and prior to adding TLS for outbound message) to a
designated IP Address and Port?  No need for local database storage.  No
need for encapsulation.  No mirroring of other messages/dialogs.

Cheers
Sudhakar
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Re: [SR-Users] How to reduce re-INVITE time in Kamailio

2019-12-18 Thread Raúl Alexis Betancor Santana

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"  
To: "Patrick Wakano"  
Cc: "Kamailio (SER) - Users Mailing List"  
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:anuranbar...@gmail.com 
| anuranbar...@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:pwak...@gmail.com | 
pwak...@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:h...@skalatan.de | 
h...@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:ovoshl...@gmail.com | ovoshl...@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:ovoshl...@gmail.com | 
ovoshl...@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:anuranbar...@gmail.com | 
anuranbar...@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

Re: [SR-Users] How to reduce re-INVITE time in Kamailio

2019-12-18 Thread Henning Westerholt

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.com

From: sr-users  On Behalf Of Anuran Barman
Sent: Wednesday, December 18, 2019 7:30 AM
To: Patrick Wakano 
Cc: Kamailio (SER) - Users Mailing List 
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:anuranbar...@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:pwak...@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 
mailto:h...@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 
mailto:ovoshl...@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, 
mailto:ovoshl...@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, 
mailto:anuranbar...@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 
a

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