I can not disconnect the active sessions

1. Brekeke Product Name and Version:

2. Java version:

3. OS type and the version:

4. UA (phone), gateway or other hardware/software involved:

5. Your problem:
I'm using push notification (FCM/APNs) to announce an incoming call.
User A make a phone call to User B.
On the B side, after received a push notification, B will do registration to SIP server and waiting for an Incoming Call, but the INVITE from Brekeke SIP server was not sent to B somehow (it happens rarely).
So that, the active session between A and B still in "Initializing" and hanging up there for a long time.
User A can not send INVITE to User B again.

I used Brekeke web admin to disconnect the active session, but it can't help.

Please teach me for this practice.

Is it Brekeke PBX or Brekeke SIP Server?
What kind of SIP client product are you using?
Which transport protocol are you using for SIP? (UDP? TCP? or TLS?)

> B will do registration to SIP server and waiting for an Incoming Call,

Can you find the above registration record at Brekeke SIP Server's [Registered Clients] page?

Dear Niloc,

Sorry for lack of information.
I give more information inline.

Niloc wrote:Is it Brekeke PBX or Brekeke SIP Server?

Brekeke SIP Server

Niloc wrote:What kind of SIP client product are you using?

I used SIP client of my company.
It's still in developing, not public yet

Niloc wrote:Which transport protocol are you using for SIP? (UDP? TCP? or TLS?)

TLS

Niloc wrote: > B will do registration to SIP server and waiting for an Incoming Call,
Can you find the above registration record at Brekeke SIP Server's [Registered Clients] page?

Yes, it has registration record.

Regards,
Liem

Does the same problem persist even if you use UDP instead of TLS?

redroof wrote:Does the same problem persist even if you use UDP instead of TLS?

My SIP client application don't use UDP mode (this is spec), so I don't know whether it works fine in UDP mode or not.
But I think this problem is due to TLS connection keeping its session.

*Note:

nsliem wrote:but the INVITE from Brekeke SIP server was not sent to B somehow (it happens rarely).
So that, the active session between A and B still in "Initializing" and hanging up there for a long time.

My application will wait for about 15 seconds for the INVITE.
If it's timeout, app will un-register (de-register) from SIP server.
But I don't know why TLS connection was keeping in this case.
And that, I can not make other phone call.

Are there any NAT between the SIP Server and the client B?
Does the same issue happen even if you test at LAN without NAT/router?

Which version of Brekeke SIP Server is it?

Niloc wrote:Are there any NAT between the SIP Server and the client B?

Yes, I used router for testing.

Niloc wrote:Does the same issue happen even if you test at LAN without NAT/router?

I'm not test yet. So, I'm not sure about this case.
Once again, this problem is happening rarely.

Niloc wrote:Which version of Brekeke SIP Server is it?

Brekeke SIP Server, Version 3.6.1.8, Advanced, Push Notification


---
My question is:
1. When will the record in active sessions disappear?
(Although connection between A and B was lost, and the status of them might be "Initializing", or "Closing", but it took too much time to disappear, mybe an hour).

2. What is the purpose of this setting?
Configuration > RTP > RTP Session Timeout (ms)

> Brekeke SIP Server, Version 3.6.1.8, Advanced, Push Notification

Try the latest version. It is ver 3.7.7.8.
http://www.brekeke.com/downloads/sip-server.php


> 1. When will the record in active sessions disappear?

After the Inviting timeout. It is 20sec in the default.


> 2. What is the purpose of this setting?
> Configuration > RTP > RTP Session Timeout (ms)

It checks whether RTP packets are received or not.
Refer to the document for more details.

Dear Niloc,

Thank you for your support,
I will consider doing version up later.