[sip-comm-dev] Strange behaviour during "non-registrar" setup and calls


#1

All,

in my setup to test some stuff for SC I stumbled over several
problems that block the tests. Here my setup:

- SC runs on my host, Linux (openSUSE 10.3), Java 1.5 64 Bit
- the other SIP client runs in a VMware guest system, also a
   Suse, but this client is not a SC
- my system has several Ethernet adapters (two real physical adapters,
   several vmnet adapters)

To communicate I configured the SC account as non-registrar, that
is: only a simple name (test) in the username field (which then
sudenly expands to test@null). Using the advanced tab I edited the
proxy fileds to reflect my internal IP addresses and ports. At the
same time deleting the port number in "Server port" (which again is
strange because there is no "Server" mentioned before, just a
"Registrar", thus IMHO it would better be named "Registrar server
address" and "Registrar server port")

The address I use is 192.168.104.128 which is the IP address of the
system in the VMware player, the vmnet1 interface on the host is
accordingly set to 192.168.104.1, route is also set.

When I set up a call from SC to the other client running in the VMware
guest the SIP communication is ok, I hear the rining tone of SC, then
answering the call on the other client - then silence. Digging into
the problem using wireshark shows that SC does not setup the SDP data
correctly in the SIP invite:

INVITE sip:1234 SIP/2.0
Call-ID: 4349393b7839887818d699499dbd788e@0.0.0.0
CSeq: 1 INVITE

From: "test" <sip:test@192.168.104.1:5060;transport=udp>;tag=7c285c6b

Via: SIP/2.0/UDP 192.168.104.1:5060;branch=z9hG4bKbec4673063e42bb5ca12e9798d114e77
Max-Forwards: 70
User-Agent: SIP Communicator 1.0-alpha3-0.build.by.SVN Linux
Contact: "test" <sip:test@192.168.104.1:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 202

v=0
o=test 0 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 110 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
m=video 5002 RTP/AVP 34 26 31
a=recvonly

I would like to draw your attention to the "c=" line: it states 127.0.0.1
instead of 192.168.104.1. I skimmed thru SC configuration but couldn't see
a way to force this setting. Using this information the other client is
obviously not able to send data to SC.

How does SC determine its address used in the SDP? Setting some info
in /etc/hosts?

Another issue: after the above thing happend a call to FWD's echo test did
not work anymore - that is the connection was setup, but I could not hear the
announcement. After restarting SC it work as normal. Seems that the audio
setup was/is confused after the above mentioned first failure.

Some additional side notes:
- Configuring SC show some errors: when selecting the tools, user account,
   and then modifiying an account: you can modify some data, then you can
   select "next", the next window shows "back", "register", "Cancel" - but
   no "store" or "save". in a non registrar mode you cannot "register". Hiting
   cancel: seems that the modifications are store but the user account list
   window does not show the modified account anymore - restart of SC
   is required - the the modified account is displayed again

- during SC call setup an enormous amunt of data is displayed in the terminal
   window (I use "ant run" to start it). I would propose to diable these message
   in the log4j prpoerties by default, only those who work with media streams
   should enable this to perform debugging.

- In general SC emits a large number of exception during a run - which makes
   me nervous and gives me an overall impression of "very alpha" :slight_smile:

An idea for the GUI:
- at the top level GUI there is a info in the lower right that shows which
   account is in use (in my sace I have 2 account, one for FWD, one for my tests)
   The selection drop down just says "SIP" and a number. Would it makes sense
   to show the full name so that I can see what SIP1 really is?
   To get the info about the active account I need to posiiton the mouse over
   the field and after 1-2 seconds I can see it (hover). IMHO not very user
   friendly.

That's it for today folks :slight_smile: .

Regards,
Werner

···

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#2

Hi Werner,

Werner Dittmann написа:

All,

in my setup to test some stuff for SC I stumbled over several
problems that block the tests. Here my setup:

- SC runs on my host, Linux (openSUSE 10.3), Java 1.5 64 Bit
- the other SIP client runs in a VMware guest system, also a
   Suse, but this client is not a SC
- my system has several Ethernet adapters (two real physical adapters,
   several vmnet adapters)

To communicate I configured the SC account as non-registrar, that
is: only a simple name (test) in the username field

(which then sudenly expands to test@null).

This is normal and shouldn't be causing any problems

Using the advanced tab I edited the
proxy fileds to reflect my internal IP addresses and ports.

I am afraid that the server-less mode has not been previewed for use
with a proxy. I had a quick look at the code and I actually see no
reason why it shouldn't work. It's just that it's never been tested so
some of your problems might be coming from there.

At the
same time deleting the port number in "Server port" (which again is
strange because there is no "Server" mentioned before, just a
"Registrar", thus IMHO it would better be named "Registrar server
address" and "Registrar server port")

Agreed for "Registrar port". We could also add a field for the proxy
port. Please log an issue for 1.0-rc1 if you feel this is important and
we'll try to resolve it.

The address I use is 192.168.104.128 which is the IP address of the
system in the VMware player, the vmnet1 interface on the host is
accordingly set to 192.168.104.1, route is also set.

When I set up a call from SC to the other client running in the VMware
guest the SIP communication is ok, I hear the rining tone of SC, then
answering the call on the other client - then silence. Digging into
the problem using wireshark shows that SC does not setup the SDP data
correctly in the SIP invite:

INVITE sip:1234 SIP/2.0
Call-ID: 4349393b7839887818d699499dbd788e@0.0.0.0
CSeq: 1 INVITE
From: "test" <sip:test@192.168.104.1:5060;transport=udp>;tag=7c285c6b
To: <sip:1234>
Via: SIP/2.0/UDP 192.168.104.1:5060;branch=z9hG4bKbec4673063e42bb5ca12e9798d114e77
Max-Forwards: 70
User-Agent: SIP Communicator 1.0-alpha3-0.build.by.SVN Linux
Contact: "test" <sip:test@192.168.104.1:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 202

v=0
o=test 0 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 110 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
m=video 5002 RTP/AVP 34 26 31
a=recvonly

I would like to draw your attention to the "c=" line: it states 127.0.0.1
instead of 192.168.104.1. I skimmed thru SC configuration but couldn't see
a way to force this setting. Using this information the other client is
obviously not able to send data to SC.

Indeed, this is not configurable but should select the right address
automatically just as it did for the addresses in the SIP part of the
message. I'll look into this as soon as I find the time. You might also
want to log an issue just so that we are sure we don't forget about it.

How does SC determine its address used in the SDP? Setting some info
in /etc/hosts?

Another issue: after the above thing happend a call to FWD's echo test did
not work anymore - that is the connection was setup, but I could not hear the
announcement. After restarting SC it work as normal. Seems that the audio
setup was/is confused after the above mentioned first failure.

Possible but let's take them one at a time.

Some additional side notes:
- Configuring SC show some errors: when selecting the tools, user account,
   and then modifiying an account: you can modify some data, then you can
   select "next", the next window shows "back", "register", "Cancel" - but
   no "store" or "save". in a non registrar mode you cannot "register". Hiting
   cancel: seems that the modifications are store but the user account list
   window does not show the modified account anymore - restart of SC
   is required - the the modified account is displayed again

Well, you should use the register button to save the account. Granted,
it won't really register with a registrar in the case of a server-less
account. We could change the button name to "save" whenever a form is
opened in order to modify an account. It would still remain "register"
for new server-less accounts but then again these are mostly used by
experienced users/developers so I am not that worried about it.

- during SC call setup an enormous amunt of data is displayed in the terminal
   window (I use "ant run" to start it). I would propose to diable these message
   in the log4j prpoerties by default, only those who work with media streams
   should enable this to perform debugging.

All such messages are disabled when running SC from any of its
installation packages. Running it from ant automatically puts it in
verbose mode as we are assuming that people that do this would mostly be
developers. I'll have a look at the media package and try to see whether
this would be an easy fix.

- In general SC emits a large number of exception during a run - which makes
   me nervous and gives me an overall impression of "very alpha" :slight_smile:

Shouldn't make you nervous. Most of these are not actual errors. Every
attempt to SRV resolve a host that has no SRV records would cause a
stacktrace to be dumped. Reasons for this are mostly historic and we
could remove it but I'd like us to first address the more serious issues
that you mention. Could you please log an issue or simply remind me when
we're done with the rest?

An idea for the GUI:
- at the top level GUI there is a info in the lower right that shows which
   account is in use (in my sace I have 2 account, one for FWD, one for my tests)
   The selection drop down just says "SIP" and a number. Would it makes sense
   to show the full name so that I can see what SIP1 really is?
   To get the info about the active account I need to posiiton the mouse over
   the field and after 1-2 seconds I can see it (hover). IMHO not very user
   friendly.

You should be seeing the full user name when you click on the button to
select it. Isn't this the case?

That's it for today folks :slight_smile: .

Thanks!

Emil

···

Regards,
Werner

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#3

Emil,

thanks for the fast answer. Just an additional comment:
the first few item I mentions are more (or less) issues
regarding handling and so on.

The most serious problems is the topic show in the SIP INVITE because
this is really blocking. SOme time ago (maybe 2 months or so)
I did a non-registrar test using the source from Emanuel's
SVN branch (the encryption branch) and it worked. I've noticed that
quite a lot of changed were done in the main trunk but couldn't
find the part which causes this problem.

Emil Ivov schrieb:

Hi Werner,

<SNIP ---- SNAP>

Some additional side notes:
- Configuring SC show some errors: when selecting the tools, user account,
   and then modifiying an account: you can modify some data, then you can
   select "next", the next window shows "back", "register", "Cancel" - but
   no "store" or "save". in a non registrar mode you cannot "register". Hiting
   cancel: seems that the modifications are store but the user account list
   window does not show the modified account anymore - restart of SC
   is required - the the modified account is displayed again

Well, you should use the register button to save the account. Granted,
it won't really register with a registrar in the case of a server-less
account. We could change the button name to "save" whenever a form is
opened in order to modify an account. It would still remain "register"
for new server-less accounts but then again these are mostly used by
experienced users/developers so I am not that worried about it.

SC performs to register when I click the "Register" button - which does
not make sense in this case :slight_smile: and results in some error message .

···

Thanks!

Emil

Regards,
Werner

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#4

Hey Werner,

Werner Dittmann написа:

The most serious problems is the topic show in the SIP INVITE because
this is really blocking.

Yup I got that. I'll be looking into it as soon as possible and will let
you know when ready.

Cheers
Emil

···

SOme time ago (maybe 2 months or so)
I did a non-registrar test using the source from Emanuel's
SVN branch (the encryption branch) and it worked. I've noticed that
quite a lot of changed were done in the main trunk but couldn't
find the part which causes this problem.

Emil Ivov schrieb:

Hi Werner,

<SNIP ---- SNAP>

Some additional side notes:
- Configuring SC show some errors: when selecting the tools, user account,
   and then modifiying an account: you can modify some data, then you can
   select "next", the next window shows "back", "register", "Cancel" - but
   no "store" or "save". in a non registrar mode you cannot "register". Hiting
   cancel: seems that the modifications are store but the user account list
   window does not show the modified account anymore - restart of SC
   is required - the the modified account is displayed again

Well, you should use the register button to save the account. Granted,
it won't really register with a registrar in the case of a server-less
account. We could change the button name to "save" whenever a form is
opened in order to modify an account. It would still remain "register"
for new server-less accounts but then again these are mostly used by
experienced users/developers so I am not that worried about it.

SC performs to register when I click the "Register" button - which does
not make sense in this case :slight_smile: and results in some error message .

Thanks!

Emil

Regards,
Werner

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#5

Hi,

I noticed that SC 1458 nightly shows under "CALL VIA"
* account name
* connection good/bad info

The account names include the top-level-domain, like jabber.ccc.de
**ONLY** for XMPP accounts and drops this info for SIP accounts.

That means if I have several SIP accounts such as:

Earl@123.com
Earl@456.net
Earl@789.org

there are problems with CALL VIA and account settings.

I would like to see the *complete* SIP address displayed under
CALL VIA and account settings, the same as with XMPP..

Regards, Earl

···

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#6

Hi,

I tried deleting an XMPP/Jabber contact by right-clicking.

It did not work. The contact is still there.

SC nightly 1458

Regards, Earl

···

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#7

Hey Werner,

I have just committed some extra logging into the
NetworkAddressManagerServiceImpl class. That's where our local host
address is picked.

What I am thinking is that we would probably need to extend our address
selection algorithm a bit. We should fall back to a user query for cases
where we are unable to select a localhost address by ourselves. I
believe KPhone are doing something quite similar.

Anyways, before we get there I'd like us to have a better understanding
of the cases where our process currently fails and would therefore like
to ask you to try it again and send us the logs. You would need to add
the following at the bottom of your lib/logging.properties file:

net.java.sip.communicator.impl.netaddr.level = FINEST

We would then be able to see what happened through your log files.

Cheers
Emil

Emil Ivov написа:

···

Hey Werner,

Werner Dittmann написа:

The most serious problems is the topic show in the SIP INVITE because
this is really blocking.

Yup I got that. I'll be looking into it as soon as possible and will let
you know when ready.

Cheers
Emil

SOme time ago (maybe 2 months or so)
I did a non-registrar test using the source from Emanuel's
SVN branch (the encryption branch) and it worked. I've noticed that
quite a lot of changed were done in the main trunk but couldn't
find the part which causes this problem.

Emil Ivov schrieb:

Hi Werner,

<SNIP ---- SNAP>

Some additional side notes:
- Configuring SC show some errors: when selecting the tools, user account,
   and then modifiying an account: you can modify some data, then you can
   select "next", the next window shows "back", "register", "Cancel" - but
   no "store" or "save". in a non registrar mode you cannot "register". Hiting
   cancel: seems that the modifications are store but the user account list
   window does not show the modified account anymore - restart of SC
   is required - the the modified account is displayed again

Well, you should use the register button to save the account. Granted,
it won't really register with a registrar in the case of a server-less
account. We could change the button name to "save" whenever a form is
opened in order to modify an account. It would still remain "register"
for new server-less accounts but then again these are mostly used by
experienced users/developers so I am not that worried about it.

SC performs to register when I click the "Register" button - which does
not make sense in this case :slight_smile: and results in some error message .

Thanks!

Emil

Regards,
Werner

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#8

Hi,

No matter how I try to add a new SIP contact, I get an error: failed to add contact:

net.java.sip.communicator.service.contactlist.MetaContactListException: Failed to create a contact with address: first.last@abcde.net I'm terribly sorry, server error occurred (1/SL)
    at net.java.sip.communicator.impl.contactlist.MetaContactListServiceImpl.addNewContactToMetaContact(MetaContactListServiceImpl.java:449)
    at net.java.sip.communicator.impl.contactlist.MetaContactListServiceImpl.createMetaContact(MetaContactListServiceImpl.java:663)
    at net.java.sip.communicator.impl.gui.main.contactlist.addcontact.AddContactWizard$CreateContact.run(AddContactWizard.java:121)

Using nightly 1458

I exchanged the real ID to first.last@abcde.net in the above text.

I do not understand why there should be a server error.
I should be able to add a SIP contact even when not connected to the Inet.

Regards, Earl

···

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#9

Under CALL VIA, all 4 of my XMPP servers have a yellow light bulb
so I assume that I am connected to all.
But I can not delete a XMPP contact.

Earl

Earl wrote:

···

Hi,

I tried deleting an XMPP/Jabber contact by right-clicking.

It did not work. The contact is still there.

SC nightly 1458

Regards, Earl

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#10

Hey Earl,

I am afraid I don't see this behaviour here. Could you please give us
the step necessary to reproduce it starting from a clean installation
(i.e. no .sip-communicator directory)?

Your log files would also help.

Cheers
Emil

Earl написа:

···

Hi,

I tried deleting an XMPP/Jabber contact by right-clicking.

It did not work. The contact is still there.

SC nightly 1458

Regards, Earl

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#11

Hey Earl,

This is a good point. Could you please enter an enhancement request in
the issue tracker and schedule it for 1.0-rc1?

Emil

Earl написа:

···

Hi,

I noticed that SC 1458 nightly shows under "CALL VIA"
* account name
* connection good/bad info

The account names include the top-level-domain, like jabber.ccc.de
**ONLY** for XMPP accounts and drops this info for SIP accounts.

That means if I have several SIP accounts such as:

Earl@123.com
Earl@456.net
Earl@789.org

there are problems with CALL VIA and account settings.

I would like to see the *complete* SIP address displayed under
CALL VIA and account settings, the same as with XMPP..

Regards, Earl

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#12

Dear SIP Dev. community,

We are currently trying to use the SIP Communicator for a project, based on the alpha3 build 989
SIP Server> 3CX 6.1.0, and a normal sip communicator instance

in general we rely on the SIP Bundles but we create also our own bundles.

We encounter some problems when trying to get the State of a subsbribed contact:

The problems are:
when we add a contact to our contactlist, we subscribe to it.
    (when debugging it seems to be fine)

// getting the presence of the ProtocolProv
OperationSetPresenceSipImpl presence = (OperationSetPresenceSipImpl)getProtocolPresenceOpSet(p);
//subscribing
presence.subscribe(mycontact.getParentContactGroup(), mycontact.getAddress());

...

in another bundle>
presence.addContactPresenceStatusListener(new MyContactPresenceStatusListener());
    private class MyContactPresenceStatusListener implements
            ContactPresenceStatusListener
    {

    public void contactPresenceStatusChanged(
                ContactPresenceStatusChangeEvent evt)
        {
            log.debug("StatusService ContactPresenceStatusChanged: " +evt.getSourceContact().getAddress() +", " + evt.toString()+", "+ evt.getPropertyName());
            String contactID = evt.getSourceContact().getAddress();
            ProtocolProviderService protocolProvider = evt.getSourceProvider();
            OperationSetPresenceSipImpl presence = (OperationSetPresenceSipImpl)getProtocolPresenceOpSet(protocolProvider);
            PresenceStatus contactStatus = null;
            ContactSipImpl contact = (ContactSipImpl)presence.findContactByID(contactID);
            presence.forcePollContact(contact);
                       try{
                contactStatus = presence.queryContactStatus(contactID);
            log.debug(Status of contact " + contactID +" is now: state " +contactStatus.getStatus() +" stateName: "+ contactStatus.getStatusName() + "Timestamp: "+ getDateString()); }
            catch (Exception e) {
                log.error("Error on getting Status of contact :" +e);
            }
            ....
                  }
    }

somehow this listener does not react to any status changes of the subscribed contacts.

also requesting a contact state does not work (when not requested by the listener)
      contactState = presence.queryContactStatus(contactSipId);
   most of the time I get the value 65, even the contact of the contactSipId is offline

I would be thankfull for any hints where I can continue to look for the problem.

KEEP Communicationg!

Mamimo

···

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#13

Emil,

thanks. Attached is the log for the test run.

There seems to be a problem with STUN, it asks for resolve my local address.
Somehow it tries to resolve my local network address, look at the lines:

...
17:40:48.166 FEIN: impl.netaddr.NetworkAddressManagerServiceImpl.getPublicAddressFor().393 Stun is disabled for destination 1234/0.0.4.210, skipping mapped address recovery (useStun=false, IPv6@=false).
17:40:48.168 FEINER: impl.netaddr.NetworkAddressManagerServiceImpl.getLocalHost().242 Querying a localhost addr for dst=1234/0.0.4.210
17:40:48.169 FEINER: impl.netaddr.NetworkAddressManagerServiceImpl.getLocalHost().258 Socket returned the AnyLocalAddress. Trying a workaround.
17:40:48.171 FEINER: impl.netaddr.NetworkAddressManagerServiceImpl.getLocalHost().311 Will return the following localhost address: linux-black/127.0.0.1
...

Several times the get LocalHost works ok, IMHO then it tries to resolve
the local address using STUN. the number "1234" is the number I dialed to
setup the call, see the SIP INVITE below. Of course I do not uuse NAT in my
local network, also I don't use NAT to connect to FWD as you can see in the log
(direct Internet connection).

Here is the according SIP INVITE:

INVITE sip:1234 SIP/2.0
Call-ID: 9ddbea4d91c70e5660756eee52ce4950@0.0.0.0
CSeq: 1 INVITE

From: "test" <sip:test@192.168.104.1:5060;transport=udp>;tag=54380294

Via: SIP/2.0/UDP 192.168.104.1:5060;branch=z9hG4bKf772ee634a074106a0c505299663522a
Max-Forwards: 70
User-Agent: SIP Communicator 1.0-alpha3-0.build.by.SVN Linux
Contact: "test" <sip:test@192.168.104.1:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 202

v=0
o=test 0 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 110 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
m=video 5002 RTP/AVP 34 26 31
a=recvonly

Hope this helps a bit.

Regards,
Werner

PS: an additional question regarding 64bit libs - the log shows some problems
here. Quite a number of 32bit libs are included in SC, but only 3 or 4 64bit
libs - what about the rest?

Werner

Emil Ivov schrieb:

Hey Werner,

I have just committed some extra logging into the
NetworkAddressManagerServiceImpl class. That's where our local host
address is picked.

What I am thinking is that we would probably need to extend our address
selection algorithm a bit. We should fall back to a user query for cases
where we are unable to select a localhost address by ourselves. I
believe KPhone are doing something quite similar.

Anyways, before we get there I'd like us to have a better understanding
of the cases where our process currently fails and would therefore like
to ask you to try it again and send us the logs. You would need to add
the following at the bottom of your lib/logging.properties file:

net.java.sip.communicator.impl.netaddr.level = FINEST

We would then be able to see what happened through your log files.

Cheers
Emil

Emil Ivov написа:

<SNIP --- SNAP>

sip-communicator0.log (8.08 KB)

···

To: <sip:1234>


#14

Hey Earl,

Earl написа:

net.java.sip.communicator.service.contactlist.MetaContactListException:
Failed to create a contact with address: first.last@abcde.net I'm
terribly sorry, server error occurred (1/SL)

Seems like the server returned an error.

I do not understand why there should be a server error.
I should be able to add a SIP contact even when not connected to the Inet.

We do expect you to be registered before adding contacts. I don't think
your problem is related to connectivity though since the error messages
seems to be sent from the server.

Cheers
Emil

···

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#15

Emil,

I did a fresh install to a new directory and was able to delete the XMPP contact
using nightly 1462.

This appears to be a "strange thing" and not a bug.

Regards, Earl

Emil Ivov wrote:

···

Hey Earl,

I am afraid I don't see this behaviour here. Could you please give us
the step necessary to reproduce it starting from a clean installation
(i.e. no .sip-communicator directory)?

Your log files would also help.

Cheers
Emil

Earl написа:
  

Hi,

I tried deleting an XMPP/Jabber contact by right-clicking.

It did not work. The contact is still there.

SC nightly 1458

Regards, Earl

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#16

Hey Werner,

Actually the debug logs you see only tell you that STUN is disabled. We
are not using STUN by default.

I noticed something else though. We can handle user-name-only URIs only
in cases where you have a registrar. In such cases we simply attach
behind them the address/fqdn of our registrar. This is based on the
assumption that if we are registered as alice@example.com and we try to
call "bob" then we would actually mean "bob@example.com". In the case of
a server-less account we would consider non-uri strings to actually be a
host name, so we try to determine a local address that we could use when
sending packets to the "1234" host and that fails which is why you get
127.0.0.1.

Could you please try calling 1234@ipaddress and let me know what happens?

Emil

Werner Dittmann написа:

···

Emil,

thanks. Attached is the log for the test run.

There seems to be a problem with STUN, it asks for resolve my local address.
Somehow it tries to resolve my local network address, look at the lines:

...
17:40:48.166 FEIN: impl.netaddr.NetworkAddressManagerServiceImpl.getPublicAddressFor().393 Stun is disabled for destination 1234/0.0.4.210, skipping mapped
address recovery (useStun=false, IPv6@=false).
17:40:48.168 FEINER: impl.netaddr.NetworkAddressManagerServiceImpl.getLocalHost().242 Querying a localhost addr for dst=1234/0.0.4.210
17:40:48.169 FEINER: impl.netaddr.NetworkAddressManagerServiceImpl.getLocalHost().258 Socket returned the AnyLocalAddress. Trying a workaround.
17:40:48.171 FEINER: impl.netaddr.NetworkAddressManagerServiceImpl.getLocalHost().311 Will return the following localhost address: linux-black/127.0.0.1
...

Several times the get LocalHost works ok, IMHO then it tries to resolve
the local address using STUN. the number "1234" is the number I dialed to
setup the call, see the SIP INVITE below. Of course I do not uuse NAT in my
local network, also I don't use NAT to connect to FWD as you can see in the log
(direct Internet connection).

Here is the according SIP INVITE:

INVITE sip:1234 SIP/2.0
Call-ID: 9ddbea4d91c70e5660756eee52ce4950@0.0.0.0
CSeq: 1 INVITE
From: "test" <sip:test@192.168.104.1:5060;transport=udp>;tag=54380294
To: <sip:1234>
Via: SIP/2.0/UDP 192.168.104.1:5060;branch=z9hG4bKf772ee634a074106a0c505299663522a
Max-Forwards: 70
User-Agent: SIP Communicator 1.0-alpha3-0.build.by.SVN Linux
Contact: "test" <sip:test@192.168.104.1:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 202

v=0
o=test 0 0 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 5000 RTP/AVP 0 8 97 3 110 5 4
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no;bitrate=6.3
m=video 5002 RTP/AVP 34 26 31
a=recvonly

Hope this helps a bit.

Regards,
Werner

PS: an additional question regarding 64bit libs - the log shows some problems
here. Quite a number of 32bit libs are included in SC, but only 3 or 4 64bit
libs - what about the rest?

Werner

Emil Ivov schrieb:

Hey Werner,

I have just committed some extra logging into the
NetworkAddressManagerServiceImpl class. That's where our local host
address is picked.

What I am thinking is that we would probably need to extend our address
selection algorithm a bit. We should fall back to a user query for cases
where we are unable to select a localhost address by ourselves. I
believe KPhone are doing something quite similar.

Anyways, before we get there I'd like us to have a better understanding
of the cases where our process currently fails and would therefore like
to ask you to try it again and send us the logs. You would need to add
the following at the bottom of your lib/logging.properties file:

net.java.sip.communicator.impl.netaddr.level = FINEST

We would then be able to see what happened through your log files.

Cheers
Emil

Emil Ivov написа:

<SNIP --- SNAP>

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#17

Emil,

Under send via, I see all 4 SIP servers with a small green ON icon.

Does the green ON mean that I am connected and registered to the
SIP server?

Does the gray OFF mean that there is no connection/registration?

If I am connected to SIP server 123.com and want to add a contact
JOE@456.net, does this work? Assume that I have no account
at 456.net.

If the above comments mean that I am connected to all 4 of my SIP
servers, but can not add a new SIP contact, then this is a severe bug
making SC useless for SIP.

Can I do any other tests here to help find out what is happening?
A log file somewhere that might help?
Can this be replicated by others or am I the only one?

Again, this is using nightly 1458.

Regards, Earl

Emil Ivov wrote:

···

Hey Earl,

Earl написа:
  

net.java.sip.communicator.service.contactlist.MetaContactListException: Failed to create a contact with address: first.last@abcde.net I'm terribly sorry, server error occurred (1/SL)
    
Seems like the server returned an error.

I do not understand why there should be a server error.
I should be able to add a SIP contact even when not connected to the Inet.
    
We do expect you to be registered before adding contacts. I don't think
your problem is related to connectivity though since the error messages
seems to be sent from the server.

Cheers
Emil

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#18

Emil,

thanks for the hint - this works and I can hear the other client.

What makes me wonder is that the SIP address is correct - and the
content of the SDP is incorrect if I just use 1234. Don't use both
the same data to resolve the address.

ZRTP tests will follow - I first need to update the zrtp.jar somewhow
because the Zfone uses an updated version of the protocol. Should be a
big problem though :slight_smile: .

Regards,
Werner

Emil Ivov schrieb:

···

Hey Werner,

Actually the debug logs you see only tell you that STUN is disabled. We
are not using STUN by default.

I noticed something else though. We can handle user-name-only URIs only
in cases where you have a registrar. In such cases we simply attach
behind them the address/fqdn of our registrar. This is based on the
assumption that if we are registered as alice@example.com and we try to
call "bob" then we would actually mean "bob@example.com". In the case of
a server-less account we would consider non-uri strings to actually be a
host name, so we try to determine a local address that we could use when
sending packets to the "1234" host and that fails which is why you get
127.0.0.1.

Could you please try calling 1234@ipaddress and let me know what happens?

Emil

Werner Dittmann написа:

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#19

Hey Earl,

Earl написа:

Emil,

Under send via, I see all 4 SIP servers with a small green ON icon.

Does the green ON mean that I am connected and registered to the
SIP server?

Yes, it does.

Does the gray OFF mean that there is no connection/registration?

Yes, absolutely.

If I am connected to SIP server 123.com and want to add a contact
JOE@456.net, does this work? Assume that I have no account
at 456.net.

Depends on whether or not you configured your account for P2P presence.
If you did then it should work. If you didn't then it would depend on
whether 123.com would let you do it.

If the above comments mean that I am connected to all 4 of my SIP
servers, but can not add a new SIP contact, then this is a severe bug
making SC useless for SIP.

I think I already mentioned this but here goes again: your server seems
to be returning an error message when you try to add a subscription. You
might want to contact them and ask them what they are unhappy with.
Alternately, since you mention inter-domain contacts, and since most
providers don't normally authorize this you could also try to configure
your account not to use SIMPLE.

Emil

···

Can I do any other tests here to help find out what is happening?
A log file somewhere that might help?
Can this be replicated by others or am I the only one?

Again, this is using nightly 1458.

Regards, Earl

Emil Ivov wrote:

Hey Earl,

Earl написа:
  

net.java.sip.communicator.service.contactlist.MetaContactListException:
Failed to create a contact with address: first.last@abcde.net I'm
terribly sorry, server error occurred (1/SL)
    

Seems like the server returned an error.

I do not understand why there should be a server error.
I should be able to add a SIP contact even when not connected to the Inet.
    

We do expect you to be registered before adding contacts. I don't think
your problem is related to connectivity though since the error messages
seems to be sent from the server.

Cheers
Emil

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#20

Emil, all,

just updated the zrtp4j library and did a first test - and had
success. SC was able to setup an encrypted connection to a
client system that uses Phil Zimmermann's Zfone as front-end to
another SIP client.

More detailled reports follow: I noticed some glitches that we need
to address. But this requires some more investigation at my side
here.

Regards,
Werner

Werner Dittmann schrieb:

···

Emil,

thanks for the hint - this works and I can hear the other client.

What makes me wonder is that the SIP address is correct - and the
content of the SDP is incorrect if I just use 1234. Don't use both
the same data to resolve the address.

ZRTP tests will follow - I first need to update the zrtp.jar somewhow
because the Zfone uses an updated version of the protocol. Should be a
big problem though :slight_smile: .

Regards,
Werner

Emil Ivov schrieb:

Hey Werner,

Actually the debug logs you see only tell you that STUN is disabled. We
are not using STUN by default.

I noticed something else though. We can handle user-name-only URIs only
in cases where you have a registrar. In such cases we simply attach
behind them the address/fqdn of our registrar. This is based on the
assumption that if we are registered as alice@example.com and we try to
call "bob" then we would actually mean "bob@example.com". In the case of
a server-less account we would consider non-uri strings to actually be a
host name, so we try to determine a local address that we could use when
sending packets to the "1234" host and that fails which is why you get
127.0.0.1.

Could you please try calling 1234@ipaddress and let me know what happens?

Emil

Werner Dittmann написа:

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