Question concerning the call-controller2 example - Media Server

[Hugo]

Hello everyone,

I have been testing the call-controller2 example (http://tinyurl.com/
428dxa) and I have
run into a minor difficulty which consequently made me wonder about a
few questions. I
would be glad if you could give me some help and also redirect me to
some links with more in depth information. Here it goes:

I have installed 'mobicents-all-1.2.0.CR2-jboss-4.2.2.GA-0809121432',
I am running the example in Ubuntu, and I have two SIP client phones
installed in the same machine, Twinkle (T) and X-Lite (X). Here are
some of the settings I am using in both phones:

SIP Incoming port: T - 5070 | X - 5050
UDP port: T - 8070 | X - 8000
proxy: T - 127.0.0.1 | X - 127.0.0.1
registrar: 127.0.0.1 | X - 127.0.0.1
outbound proxy: 127.0.0.1 | X - 127.0.0.1

I run the server using './run.sh' at the 'bin' folder level. I am able
to successfully
test the use of Mobicents Media Server in the Call blocking,
forwarding, and voicemail
cases. However, each time I try to establish a call to a user SIP
phone that is connected
and (should theoretically be) registered in the Mobicents Server, I
get the notification
that the user is not online, rather than the phone ringing. More
specifically:

1 - If I try to make a call from any user besides 'mobicents' and
'hugo' to the user
'torosvi', even if 'torosvi' is connected and registered in the
Mobicents Server, I am
automatically redirected to the voice mail. Shouldn't the 'torosvi'
SIP phone start
ringing, and shouldn't I be able to establish a conversation between
the two SIP phones?
Just in case you are wondering, I am able to leave a message in the
'torosvi' voice mail,
and listen to the voice mail message that was left. Furthermore, if
one of the blocked
users ('mobicents' or 'hugo') tries to establish a call with
'torosvi', I am able to see
the blocked call notification. Do you know why the direct call is not
working? Am I missing
something? Is there a link with the two SIP phones settings for this
example?

2 - If I make a call from any user to 'victor', I receive the
forwarding call
notification, but two unexpected (I think) behaviors occur. The first
is that even if
'torosvi' is online, it should start ringing since the call is forward
to him, but it
does not ring. What is even more strange is that not only 'torosvi'
does not ring, as I
also don't reach his voice mail. Shouldn't the forwarded call reach
the voice mail of the
phone to where the call was redirected?

It would be really helpful if you could give me some insight (and
pointers to documentation, e-books -if any-, etc...) on the procedure
of how the SIP phones get registered with the Mobicents Server, and
how the Mobicents Server handles the outbound call? In general, is
there any other good source of documentation besides the users guide
in the mobicents public group (http://tinyurl.com/6bc3ca)? In
particular, I am aiming to write a resource adapter for my specific
situation, so I would like to know where I can learn in detail how
resource adapters work, architecture, API's, and good practices.

Can you please send me the link where I can download the source code
of the 'call-controller2', 'conv-demo' examples, and of the 'mobicents-
all-1.2.0.CR2-jboss-4.2.2.GA-0809121432' server and corresponding
resource adapters?

Thank you very much for your help and congratulations on the great
work.
Hugo

[Amit Bhayani]

My answers inline

On Wed, Oct 15, 2008 at 3:25 AM, Hugo <hmcl...@gmail.com> wrote:

Hello everyone,

I have been testing the call-controller2 example (http://tinyurl.com/
428dxa) and I have
run into a minor difficulty which consequently made me wonder about a
few questions. I
would be glad if you could give me some help and also redirect me to
some links with more in depth information. Here it goes:

I have installed 'mobicents-all-1.2.0.CR2-jboss-4.2.2.GA-0809121432',
I am running the example in Ubuntu, and I have two SIP client phones
installed in the same machine, Twinkle (T) and X-Lite (X). Here are
some of the settings I am using in both phones:

SIP Incoming port: T  - 5070  |   X - 5050
UDP port: T  - 8070  |   X - 8000
proxy: T - 127.0.0.1   |   X - 127.0.0.1
registrar: 127.0.0.1  |   X - 127.0.0.1
outbound proxy: 127.0.0.1  |   X - 127.0.0.1

For Twinkle for outbound proxy use 127.0.0.1:5060 (assuming that SIP RA is listening on 5060) I think same  is true for X-Lite too.

I run the server using './run.sh' at the 'bin' folder level. I am able
to successfully
test the use of Mobicents Media Server in the Call blocking,
forwarding, and voicemail
cases. However, each time I try to establish a call to a user SIP
phone that is connected
and (should theoretically be) registered in the Mobicents Server, I
get the notification
that the user is not online, rather than the phone ringing. More
specifically:

1 - If I try to make a call from any user besides 'mobicents' and
'hugo' to the user
'torosvi', even if 'torosvi' is connected and registered in the
Mobicents Server, I am
automatically redirected to the voice mail. Shouldn't the 'torosvi'
SIP phone start
ringing, and shouldn't I be able to establish a conversation between
the two SIP phones?

yes the phone should ring. When you say the use is registered, do you get message in your SIP Phone about same? For example in Twinkle you will see the twinkle icon turning yellow from grey. Also you should see message something like "REGISTER Successfull expires in (3600 sec)" or something like this.

Just in case you are wondering, I am able to leave a message in the
'torosvi' voice mail,
and listen to the voice mail message that was left. Furthermore, if
one of the blocked
users ('mobicents' or 'hugo') tries to establish a call with
'torosvi', I am able to see
the blocked call notification. Do you know why the direct call is not
working? Am I missing
something? Is there a link with the two SIP phones settings for this
example?

We don't have setting for this example but for Converged Demo http://groups.google.com/group/mobicents-public/web/sip-phone-settings-for-converged-demo. However the setting should be the same. 

2 - If I make a call from any user to 'victor', I receive the
forwarding call
notification, but two unexpected (I think) behaviors occur. The first
is that even if
'torosvi' is online, it should start ringing since the call is forward
to him, but it
does not ring. What is even more strange is that not only 'torosvi'
does not ring, as I
also don't reach his voice mail. Shouldn't the forwarded call reach
the voice mail of the
phone to where the call was redirected?

I have faced this issue specifically with Twinkle 1.2, which version you are using? Try using different SIP Phone.

It would be really helpful if you could give me some insight (and
pointers to documentation, e-books -if any-, etc...) on the procedure
of how the SIP phones get registered with the Mobicents Server, and
how the Mobicents Server handles the outbound call? In general, is
there any other good source of documentation besides the users guide
in the mobicents public group (http://tinyurl.com/6bc3ca)? In
particular, I am aiming to write a resource adapter for my specific
situation, so I would like to know where I can learn in detail how
resource adapters work, architecture, API's, and good practices.

Can you please send me the link where I can download the source code
of the 'call-controller2', 'conv-demo' examples, and of the 'mobicents-
all-1.2.0.CR2-jboss-4.2.2.GA-0809121432' server and corresponding
resource adapters?

Please refer user-guide for all your questions above http://groups.google.com/group/mobicents-public/web/user-guide

[Hugo]

Hi Amit,

Thanks so much for your prompt reply. Unfortunately I did not have
much success so far establishing a call between any two users
connected through the Mobicents server. Nevertheless I noticed that if
the user is online he is indeed detected, but then some sort of "loop"
problem occurs, and consequently I get a lot of exceptions. The end
result is that the call is not established. I am going to try to walk
you through one of the specific examples that originated this
behavior, and hopefully you can help me find the error. I conjecture
that the problem might be related with the default listening ports of
the SIP phones, but I am not at all sure.

Phones settings:
SIP Phone 1 (SPh1):
Twinkle 1.3.2.
user: hlouro
Domain: 127.0.0.1
Registrar: 127.0.0.1
Outbound Proxy: 127.0.0.1:5060
SIP port: 5070
RTP port: 8070

SIP Phone 2 (SPh2):
X-Lite 2.0 release 1105d build stamp 99999
user: torosvi
Auto Detect IP: No
Listen on IP: 192.168.43.128 but I also tried with 127.0.0.1 and I
obtained the same results ...
Listen SIP port: 5050
Listen RTP port: 8050
Outbound SIP Proxy: 127.0.0.1
Domain/Realm: 127.0.0.1
SIP Proxy: 127.0.0.1
Outbound Proxy: 127.0.0.1:5060
Use Outbound Proxy: Always
Send internal IP: Always
Register: Always

It would be an amazing help if you could take a look at the short
logFile (http://www-personal.umich.edu/~hlouro/mobicents/) with the
details of all the test steps bellow.
Test steps:
0 - I am running Ubuntu on VMware in a PC running Windows XP.
1 - Connect to the JAIN-SLEE server by typing ./run.sh at the ..../
JBoss-xxx/bin folder level.
2 - Start Twinkle ( I get the message: " hlouro, registration
succeeded (expires = 3600 seconds)" )
3 - Start X-Lite
4 - To try to make hlouro (SPh1) establish a call with torosvi(SPh2).
I am trying to make the call by simply dialing sip:torosvi (or
sip:tor...@127.0.0.1) in SPh1. Should I dial 'torosvi' in any other
way? Should I specify the port? This way seems to follow the example
instructions, and it's the only one that lead to the (good - I guess)
log message:
" 04:51:48,484 INFO [SubscriptionProfileSbb] ########## User
sip:tor...@127.0.0.1 is available with contact
sip:tor...@192.168.43.128:5050 "
5 - The loop problem occurs right after and the call is not
established. Please take a look at the logFile that I have put in the
link above for details.

Answering your question of the previous post, the registration is
successful. I am sure of that when I run the example with Twinkle and
Ekiga (because both SIP phones give information about the number of
users registered). Furthermore, if while the server is running I add
and/or change one user in either Twinkle or Ekiga, the updated
information is immediately displayed on the phone. For example, if I
register 2/3/... users on a given phone, that phone will inform that
it it has 2/3/... users registered. Despite X-Lite does not "clearly"
give the information if the user is registered, it seems that it is
registering its users as well. At least when I change the user, that
change is reflected in the server's log.

I have tried to do a similar test with Ekiga and Twinkle, and with
Ekiga and X-Lite, and I ran into the same "loop" problems. Basically I
have tried everything that I could remember off to try to solve this
(including changing the Ekiga's default ports in the property file),
but I keep on running into this problem. I understood why the call
block and call forward work; it is because the CallBlockingSBB has
higher priority. I also confirm what you said regarding not listening
the voice mail when calling 'victor' from Twinkle, but listening it
when calling from another SIP phone.

It could also be really helpful if you could give me some details on
how you put the call-controller2 demo to work on your machine. Namely
which SIP phones you used, it's versions, it's settings, and how you
dialed the users (e.g. sip:tor...@127.0.0.1 or
sip:tor...@127.0.0.1:5050; i.e. to use port or non-port).

I have successfully tested the 'conv-demo' and everything worked
nicely. That seems a strong evidence that there is nothing wrong with
the way I installed the JAIN-SLEE server, RA's, nor with the SIP
phones themselves. It might indicate a problem with the proxy's...
but I am kinda out of ideas for now.

Thanks you very much for your help,
Best
Hugo

[Amit Bhayani]

Hi Hugo,

I just tested cc2 example and everything works as expected. The only difference being I bound my mobicents server to actual IP address like 192.168.0.100. I think sip proxy is looping the SIP Invite if bound to local bind address (127.0.0.1)

To call torosvi I dial tor...@192.168.0.100

However before you deploy this piece of code I realized that you have to make changes to ProfileCreator.java and set "domain" to actual IP Address

I am committing new code where "domain" will pick JBoss bind address automatically. So you can use the code from SVN

Amit.

[Eduardo Martins]

The proxy should not loop if 127.0.0.1 is used, that issue was fixed
in 1.2.0.GA sip services