#1383 Can't call another number after standby period (Call reliability issue)

[Anonymous]

Originally created by: ursu.adr...@gmail.com

What steps will reproduce the problem?
1.Register with your sip account
2.Wait 1 hour (for example)
3.Make a call

What is the expected output? What do you see instead?
The expected output:

The other party receives the call, answers it and a conversation is established.

What do I see instead:

My call isn't received at all (in most of the cases), or the other party's phone starts ringing very late (rarely) and after the call is answered, I cant't hear the person from the other end .
If I want to make a call, I have to try 4 or five times to successfully
establish a connection. 

What version of the product are you using? On what operating system?
CSipSimple [r1105] on Android 2.3.3 ( HTC Desire HD ) and Android 2.2.1 (Samsung Galaxy Mini). The calls are made using 3G connecion with the option "create ZRTP".Also, I have installed the app without WEBRTC support.

Please provide any additional information below.

I am using a server running Freeswitch (in default configuration) as a SIP provider.The computer never enters in standby, so that isn't the problem. Also, I have tried other SIP accounts and had the same problem. It seems that something enters in standby or something like that.
Can this problem be fixed by modifying  settings from CSipSimple (like Register Timeout -> from Accounts), or is this a software problem?

This seems to me like a very important problem, because the possibility to call someone should be a very basic feature.

I have attached the logs. Please let me know if I can do anything else to help. I will continue testing.

Based on the GPL CSipSimple version : 0.03-00 [r1105]
Here are important informations about Device :
android.os.Build.BOARD : spade
android.os.Build.BRAND : htc_wwe
android.os.Build.DEVICE : ace
android.os.Build.ID : GRI40
android.os.Build.MODEL : Desire HD
android.os.Build.PRODUCT : htc_ace
android.os.Build.TAGS : release-keys
android.os.Build.VERSION.INCREMENTAL : 87995
android.os.Build.VERSION.RELEASE : 2.3.3
android.os.Build.VERSION.SDK : 10 

Keep up the good work.

[Anonymous]

Originally posted by: ursu.adr...@gmail.com

I will try different configurations and I will post tonight a new update.

[Anonymous]

Originally posted by: ursu.adr...@gmail.com

Also, I wanted to ask something: witch is better to use while registering with the server: TCP or UDP?

[Anonymous]

Originally posted by: r3gis...@gmail.com

I guess that you are using the tls version (since you talk about zrtp).
Unfortunately that's currently a broken version. I had to broke that for releasing the latest stable release (without tls).

Next versions of csipsimple (the one I will produce after I'll update the automatic build server to build with the new way), will support  ZRTP and TLS as a core feature. And will also fix the problem of ZRTP SAS that is not always shown and other various problems there were with zrtp integration.
So, for now, if you want something to test on, use the version 0.03-01 available on download section of this website or on the android market.

Nightly builds wil come back to live really soon. Well it will maybe be less stable than the stable 0.03-01 but ZRTP and TLS will be part of the future unique build and part of my mainstream target ;)

About TCP/UDP actually it should not really matter. Normally TCP is better. But on current version of csipsimple the keep alive value of TCP is 90 sec (on [r1150] actually was disabled - another bug fixed for the stable release). On UDP it's something that you can modify in settings.
Normally TCP is better cause it's less necessary to try to keep alive the socket. However since for now you can't go upper than 90 from settings it's not very useful to use TCP cause your network maybe supports an higher value even for UDP. In this case UDP may be better. But it's only due to the current way it works on csipsimple and should not be the case (fixing that has a high priority on my todo list even if it does not affect mainstream users cause main sip providers only support udp).

Another important point about tcp is that it must be correctly manager by the sip provider. If not it can break the hangup and subsequent dialog command. It should be fine with freeswitch, but some providers such as pbxes.org are just buggy on TCP (and it only works with their client sipdroid and client that are not strictly compliant with sip rfc).

Status: NextRelease

[Anonymous]

Originally posted by: ursu.adr...@gmail.com

Thank you for the prompt reply.

I am a student at the Computer Science University and I am doing a project about the importance and advantages of open source software, and the advantages and disadvantages of VOIP ( the biggest disadvantage -> security, that's why I am interested in TLS support).
A very important part of the project is making a large number of people ( we already have 80 people from school with android phones enrolled in the project) use open source software. I've chosen CSipSimple because it's practical use, and because we can give feedback for the developers (which I know it's important) .
So, my question is: when will the next version ( with TLS support) be available ?
I ask that because my project is due in 3 weeks.

Thanks.

[Anonymous]

Originally posted by: r3gis...@gmail.com

By the end of the week-end should be done.

I can produce something before with from my build environement, but I prefer to build that from the source to be sure all source code necessary to build it is published :).

You can track the ZRTP issue (issue 262). I'll publish something on the thread when it will be available on the nightly build website.