[sip-comm-dev] p2p connection, possible in existing version


#1

Hi all,
as the one or other probably already know, my research group and I are
trying to write a p2p-grid extension for sip-communicator.
The last days, we have been occupied with evaluating NAT-traversal,
hole-punching,and firewall-piercing techniques.
We tried some tests with JXTA and NUTSS with very limited success.
But as we intend to write an extension for sip-communicator, we may not
to worry about these?

Does anybody here on the list have an idea, if it would be possible to
setup a real direct p2p-connections between two computers which are both
behind a NAT firewall, but have a communication IM connection, with the
provided means of the current sip-communicator.
To simplify it, we could suppose, that they both have access to a
stun-server and all firewalls in between do not block udp traffic.

Can anybody give a small outline, what classes we would need to
establish such a connection?

Let the scenario be the following:
B: linux, computer outside IP:172.17.36.198 (A is running on B, B has IP
in vm 10.0.2.2)
C: windows, notebook in university wlan: IP for example 192.168.10.73

B is NATed over a server 193.40.36.20 to the internet
C is NATed over the same server to the internet

Sip-communicator with our extension is running on A and C. On computer A
Alice is logged in to her msn-account and on Computer C Charly is logged
into his msn account.
Alice and Charly are buddies, so they see themselves in the buddy-list.

For a proof of concept, we just want to sent one (or multiple) direct
message (can also be udp), not via the IM-channel, from A to C and an
acknowledge back.

As we are pretty stalled at this point, we would be very grateful for
any hints.

Cheers,
Uli

···

A: linux, qemu virtual machine:10.0.2.15

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#2

Hello Uli,

Sorry for the late reply, things have been quite busy around here lately.

In order to handle such issues I'd suggest you use the solutions that
the IETF is currently working on: STUN, TURN, and ICE. You could use the
instant messaging protocol, MSN in your case, to start the ICE
negotiation and exchange the candidates that you have discovered through
STUN, TURN and on the local host.

It would be even easier to do the same with Jabber (or GoogleTalk) since
ICE semantics have already been clearly defined for XMPP in [4].

Hope this helps.

Emil

[1] STUN - http://tools.ietf.org/html/draft-ietf-behave-rfc3489bis
[2] TURN - http://ietf.org/internet-drafts/draft-ietf-behave-turn
[3] ICE - http://tools.ietf.org/html/draft-ietf-mmusic-ice
[4] XEP-0176 - http://www.xmpp.org/extensions/xep-0176.html

Ulrich Norbisrath (sip-communicator) wrote:

···

Hi all,
as the one or other probably already know, my research group and I are
trying to write a p2p-grid extension for sip-communicator.
The last days, we have been occupied with evaluating NAT-traversal,
hole-punching,and firewall-piercing techniques.
We tried some tests with JXTA and NUTSS with very limited success.
But as we intend to write an extension for sip-communicator, we may not
to worry about these?

Does anybody here on the list have an idea, if it would be possible to
setup a real direct p2p-connections between two computers which are both
behind a NAT firewall, but have a communication IM connection, with the
provided means of the current sip-communicator.
To simplify it, we could suppose, that they both have access to a
stun-server and all firewalls in between do not block udp traffic.

Can anybody give a small outline, what classes we would need to
establish such a connection?

Let the scenario be the following:
A: linux, qemu virtual machine:10.0.2.15
B: linux, computer outside IP:172.17.36.198 (A is running on B, B has IP
in vm 10.0.2.2)
C: windows, notebook in university wlan: IP for example 192.168.10.73

B is NATed over a server 193.40.36.20 to the internet
C is NATed over the same server to the internet

Sip-communicator with our extension is running on A and C. On computer A
Alice is logged in to her msn-account and on Computer C Charly is logged
into his msn account.
Alice and Charly are buddies, so they see themselves in the buddy-list.

For a proof of concept, we just want to sent one (or multiple) direct
message (can also be udp), not via the IM-channel, from A to C and an
acknowledge back.

As we are pretty stalled at this point, we would be very grateful for
any hints.

Cheers,
Uli

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#3

Hi Emil (and all others),

Sorry for the late reply, things have been quite busy around here lately.

I know this problem.

In order to handle such issues I'd suggest you use the solutions that
the IETF is currently working on: STUN, TURN, and ICE. You could use the
instant messaging protocol, MSN in your case, to start the ICE
negotiation and exchange the candidates that you have discovered through
STUN, TURN and on the local host.

Thanks, this is actually, what we are investigating. We were just
looking, if some work in terms of programmed libraries had been already
done in this area, but we actually only found your stun4j and jstun. We
are currently using the latter.

It would be even easier to do the same with Jabber (or GoogleTalk) since
ICE semantics have already been clearly defined for XMPP in [4].

As we want to enable this with all protocols, we have registered a
listener to every protocol and now intercept tagged messages - it's just
a pity that there is now way to do real interception of messages in SC.
Or is there any possibility to install a filter without touching every
protocol?
Or are there any plans to offer such an interface in the future?

Hope this helps.

Thanks, we are implementing our own STUN-based variation of some of a
UDP based NAT-traversal method. As we have a reliable communication
between two peers over the IM-protocol, we use our own modified scheme.
As soon as we have a first prototype running, we will publish this, but
I expect first results in the beginning of next years.

Cheers,
Uli

Emil

[1] STUN - http://tools.ietf.org/html/draft-ietf-behave-rfc3489bis
[2] TURN - http://ietf.org/internet-drafts/draft-ietf-behave-turn
[3] ICE - http://tools.ietf.org/html/draft-ietf-mmusic-ice
[4] XEP-0176 - http://www.xmpp.org/extensions/xep-0176.html

Ulrich Norbisrath (sip-communicator) wrote:

Hi all,
as the one or other probably already know, my research group and I are
trying to write a p2p-grid extension for sip-communicator.
The last days, we have been occupied with evaluating NAT-traversal,
hole-punching,and firewall-piercing techniques.
We tried some tests with JXTA and NUTSS with very limited success.
But as we intend to write an extension for sip-communicator, we may not
to worry about these?

Does anybody here on the list have an idea, if it would be possible to
setup a real direct p2p-connections between two computers which are both
behind a NAT firewall, but have a communication IM connection, with the
provided means of the current sip-communicator.
To simplify it, we could suppose, that they both have access to a
stun-server and all firewalls in between do not block udp traffic.

Can anybody give a small outline, what classes we would need to
establish such a connection?

Let the scenario be the following:
A: linux, qemu virtual machine:10.0.2.15
B: linux, computer outside IP:172.17.36.198 (A is running on B, B has IP
in vm 10.0.2.2)
C: windows, notebook in university wlan: IP for example 192.168.10.73

B is NATed over a server 193.40.36.20 to the internet
C is NATed over the same server to the internet

Sip-communicator with our extension is running on A and C. On computer A
Alice is logged in to her msn-account and on Computer C Charly is logged
into his msn account.
Alice and Charly are buddies, so they see themselves in the buddy-list.

For a proof of concept, we just want to sent one (or multiple) direct
message (can also be udp), not via the IM-channel, from A to C and an
acknowledge back.

As we are pretty stalled at this point, we would be very grateful for
any hints.

Cheers,
Uli

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- --
title+name: Dr. Ulrich Norbisrath
web: http://ulno.net; addr.+phone+fax+gpgkey: http://contact.ulno.net
googletalk/jabber:unorbisrath@googlemail.com; icq:46786247
mailto:u@ulno.net

···

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#4

Hi Emil (and all others),

Sorry for the late reply, things have been quite busy around here lately.

I know this problem.

In order to handle such issues I'd suggest you use the solutions that
the IETF is currently working on: STUN, TURN, and ICE. You could use the
instant messaging protocol, MSN in your case, to start the ICE
negotiation and exchange the candidates that you have discovered through
STUN, TURN and on the local host.

Thanks, this is actually, what we are investigating. We were just
looking, if some work in terms of programmed libraries had been already
done in this area, but we actually only found your stun4j and jstun. We
are currently using the latter.

It would be even easier to do the same with Jabber (or GoogleTalk) since
ICE semantics have already been clearly defined for XMPP in [4].

As we want to enable this with all protocols, we have registered a
listener to every protocol and now intercept tagged messages - it's just
a pity that there is now way to do real interception of messages in SC.
Or is there any possibility to install a filter without touching every
protocol?
Or are there any plans to offer such an interface in the future?

Hope this helps.

Thanks, we are implementing our own STUN-based variation of some of a
UDP based NAT-traversal method. As we have a reliable communication
between two peers over the IM-protocol, we use our own modified scheme.
As soon as we have a first prototype running, we will publish this, but
I expect first results in the beginning of next years.

Cheers,
Uli

···

Emil

[1] STUN - http://tools.ietf.org/html/draft-ietf-behave-rfc3489bis
[2] TURN - http://ietf.org/internet-drafts/draft-ietf-behave-turn
[3] ICE - http://tools.ietf.org/html/draft-ietf-mmusic-ice
[4] XEP-0176 - http://www.xmpp.org/extensions/xep-0176.html

Ulrich Norbisrath (sip-communicator) wrote:

Hi all,
as the one or other probably already know, my research group and I are
trying to write a p2p-grid extension for sip-communicator.
The last days, we have been occupied with evaluating NAT-traversal,
hole-punching,and firewall-piercing techniques.
We tried some tests with JXTA and NUTSS with very limited success.
But as we intend to write an extension for sip-communicator, we may not
to worry about these?

Does anybody here on the list have an idea, if it would be possible to
setup a real direct p2p-connections between two computers which are both
behind a NAT firewall, but have a communication IM connection, with the
provided means of the current sip-communicator.
To simplify it, we could suppose, that they both have access to a
stun-server and all firewalls in between do not block udp traffic.

Can anybody give a small outline, what classes we would need to
establish such a connection?

Let the scenario be the following:
A: linux, qemu virtual machine:10.0.2.15
B: linux, computer outside IP:172.17.36.198 (A is running on B, B has IP
in vm 10.0.2.2)
C: windows, notebook in university wlan: IP for example 192.168.10.73

B is NATed over a server 193.40.36.20 to the internet
C is NATed over the same server to the internet

Sip-communicator with our extension is running on A and C. On computer A
Alice is logged in to her msn-account and on Computer C Charly is logged
into his msn account.
Alice and Charly are buddies, so they see themselves in the buddy-list.

For a proof of concept, we just want to sent one (or multiple) direct
message (can also be udp), not via the IM-channel, from A to C and an
acknowledge back.

As we are pretty stalled at this point, we would be very grateful for
any hints.

Cheers,
Uli

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--
title+name: Dr. Ulrich Norbisrath
web: http://ulno.net; addr.+phone+fax+gpgkey: http://contact.ulno.net
googletalk/jabber:unorbisrath@googlemail.com; icq:46786247
mailto:u@ulno.net

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#5

Hi!

The ChatRoom interface has a method called isSystem().
Could you please explain what a system chat room is?

Best
Keio Kraaner

···

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#6

Hi everybody!

I am looking at Sip Communicator code source in order to understand its architecture and how the components interact with each other. I have understood that each protocol registers a Protocol Provider Factory = service, I have also understood that each single account in a given protocol registers a Protocol Provider Service. But, from the code source I haven't seen where a Protocol Provider Service is used. More
explanations: In SIP when initiating a call where the Protocol Provider Sevice corresponding to the account I am calling with is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, how , when and where this attribute is initialized?

Best,
Fatoumata.

-----Message d'origine-----

···

De : Keio Kraaner [mailto:keio@ut.ee]
Envoyé : mardi 11 décembre 2007 17:01
À : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] system chat room

Hi!

The ChatRoom interface has a method called isSystem().
Could you please explain what a system chat room is?

Best
Keio Kraaner

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#7

Hi Keio,

the system chat room corresponds to the server channel in the case of IRC. We have chosen to represent server channels as chat rooms, which are indicated as system.

Hope this helps.

Regards,
Yana

Keio Kraaner wrote:

···

Hi!

The ChatRoom interface has a method called isSystem().
Could you please explain what a system chat room is?

Best
Keio Kraaner

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#8

Hi again,
I wrote two years ago to ask how Protocol Provider Service are used I have no answer yet.
Can someone help me please? I am in hurry I have to do a kind of diagram showing how the registered services interact with each other. My first email is below:
"---------------------------"
Hi everybody!

I am looking at Sip Communicator code source in order to understand its = architecture and how the components interact with each other. I have = understodd that each protocol registers a Protocol Provider Factory = service, I have also understood that each single account in a given = protocol registers a Protocol Provider Service. But, from the code = source I haven't seen where a Protocol Provider Service is used. More =
explanations: In SIP when initiating a call where the Protocol Provider = Sevice corresponding to the account I am calling is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, = how , when and where this attribute is initialized?

Your help is really appreciated,
Thanks in advance,
"-----------------------------"
Best regards,
Fatoumata.

-----Message d'origine-----

···

De : zze-CAMARA Fatoumata RD-MAPS-GRE [mailto:fatoumata2.camara@orange-ftgroup.com]
Envoyé : mardi 11 décembre 2007 17:58
À : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] Protocol Provider Service

Hi everybody!

I am looking at Sip Communicator code source in order to understand its architecture and how the components interact with each other. I have understood that each protocol registers a Protocol Provider Factory = service, I have also understood that each single account in a given protocol registers a Protocol Provider Service. But, from the code source I haven't seen where a Protocol Provider Service is used. More
explanations: In SIP when initiating a call where the Protocol Provider Sevice corresponding to the account I am calling with is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, how , when and where this attribute is initialized?

Best,
Fatoumata.

-----Message d'origine-----
De : Keio Kraaner [mailto:keio@ut.ee]
Envoyé : mardi 11 décembre 2007 17:01
À : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] system chat room

Hi!

The ChatRoom interface has a method called isSystem().
Could you please explain what a system chat room is?

Best
Keio Kraaner

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#9

Hi,
Can anyone give me some help about using JMF in Felix. I have not been able to make it work. Your help is appreciated.

Regards,
Fatoumata.

-----Message d'origine-----

···

De : zze-CAMARA Fatoumata RD-MAPS-GRE [mailto:fatoumata2.camara@orange-ftgroup.com]
Envoyé : mardi 11 décembre 2007 17:58
À : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] Protocol Provider Service

Hi everybody!

I am looking at Sip Communicator code source in order to understand its architecture and how the components interact with each other. I have understood that each protocol registers a Protocol Provider Factory = service, I have also understood that each single account in a given protocol registers a Protocol Provider Service. But, from the code source I haven't seen where a Protocol Provider Service is used. More
explanations: In SIP when initiating a call where the Protocol Provider Sevice corresponding to the account I am calling with is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, how , when and where this attribute is initialized?

Best,
Fatoumata.

-----Message d'origine-----
De : Keio Kraaner [mailto:keio@ut.ee]
Envoyé : mardi 11 décembre 2007 17:01
À : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] system chat room

Hi!

The ChatRoom interface has a method called isSystem().
Could you please explain what a system chat room is?

Best
Keio Kraaner

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#10

Hi,

You can find these informations on our website (www.sip-communicator.org) in the developer documentation. Here is the exact link : http://www.sip-communicator.org/index.php/Documentation/PPBasics

Hope that helps,
Ben.

zze-CAMARA Fatoumata RD-MAPS-GRE a �crit :

···

Hi again,
I wrote two years ago to ask how Protocol Provider Service are used I have no answer yet.
Can someone help me please? I am in hurry I have to do a kind of diagram showing how the registered services interact with each other. My first email is below:
"---------------------------"
Hi everybody!

I am looking at Sip Communicator code source in order to understand its = architecture and how the components interact with each other. I have = understodd that each protocol registers a Protocol Provider Factory = service, I have also understood that each single account in a given = protocol registers a Protocol Provider Service. But, from the code = source I haven't seen where a Protocol Provider Service is used. More =
explanations: In SIP when initiating a call where the Protocol Provider = Sevice corresponding to the account I am calling is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, = how , when and where this attribute is initialized?

Your help is really appreciated,
Thanks in advance,
"-----------------------------"
Best regards,
Fatoumata.

-----Message d'origine-----
De : zze-CAMARA Fatoumata RD-MAPS-GRE [mailto:fatoumata2.camara@orange-ftgroup.com] Envoy� : mardi 11 d�cembre 2007 17:58
� : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] Protocol Provider Service

Hi everybody!

I am looking at Sip Communicator code source in order to understand its architecture and how the components interact with each other. I have understood that each protocol registers a Protocol Provider Factory = service, I have also understood that each single account in a given protocol registers a Protocol Provider Service. But, from the code source I haven't seen where a Protocol Provider Service is used. More
explanations: In SIP when initiating a call where the Protocol Provider Sevice corresponding to the account I am calling with is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, how , when and where this attribute is initialized?

Best,
Fatoumata.

-----Message d'origine-----
De : Keio Kraaner [mailto:keio@ut.ee]
Envoy� : mardi 11 d�cembre 2007 17:01
� : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] system chat room

Hi!

The ChatRoom interface has a method called isSystem().
Could you please explain what a system chat room is?

Best
Keio Kraaner

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#11

Hello Fatoumata,

Yes we have JMF running in an OSGi bundle. In our case we include the
whole framework inside the bundle jar.

Running it is a bit trickier since, depending on the Operating System,
we need to know which set of native libs to include in the library path.
You can have a look at our "bundle-media" and "run" targets in the
build.xml file in order to get an idea of how we handle the whole thing.

Hope this helps

Emil

fatoumata2.camara@orange-ftgroup.com написа:

···

Hi,
Can anyone give me some help about using JMF in Felix. I have not been able to make it work. Your help is appreciated.

Regards,
Fatoumata.

-----Message d'origine-----
De : zze-CAMARA Fatoumata RD-MAPS-GRE [mailto:fatoumata2.camara@orange-ftgroup.com]
Envoyé : mardi 11 décembre 2007 17:58
À : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] Protocol Provider Service

Hi everybody!

I am looking at Sip Communicator code source in order to understand its architecture and how the components interact with each other. I have understood that each protocol registers a Protocol Provider Factory = service, I have also understood that each single account in a given protocol registers a Protocol Provider Service. But, from the code source I haven't seen where a Protocol Provider Service is used. More
explanations: In SIP when initiating a call where the Protocol Provider Sevice corresponding to the account I am calling with is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, how , when and where this attribute is initialized?

Best,
Fatoumata.

-----Message d'origine-----
De : Keio Kraaner [mailto:keio@ut.ee]
Envoyé : mardi 11 décembre 2007 17:01
À : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] system chat room

Hi!

The ChatRoom interface has a method called isSystem().
Could you please explain what a system chat room is?

Best
Keio Kraaner

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#12

Hi,
Thanks. I had already red the information. Reading again, It seems to me that it is the UI Service that uses Protocol Provider Service. Is it right?
Can someone tells me where the session parameters negociation is done in the SIP Protocol?

Regards,
Fatoumata.

-----Message d'origine-----

···

De : Benoit Pradelle [mailto:ze_real_neo@yahoo.fr]
Envoyé : jeudi 13 décembre 2007 09:37
À : dev@sip-communicator.dev.java.net
Objet : Re: [sip-comm-dev] Protocol Provider Service Again

Hi,

You can find these informations on our website
(www.sip-communicator.org) in the developer documentation. Here is the exact link :
http://www.sip-communicator.org/index.php/Documentation/PPBasics

Hope that helps,
Ben.

zze-CAMARA Fatoumata RD-MAPS-GRE a écrit :

Hi again,
I wrote two years ago to ask how Protocol Provider Service are used I have no answer yet.
Can someone help me please? I am in hurry I have to do a kind of diagram showing how the registered services interact with each other. My first email is below:
"---------------------------"
Hi everybody!

I am looking at Sip Communicator code source in order to understand
its = architecture and how the components interact with each other. I
have = understodd that each protocol registers a Protocol Provider
Factory = service, I have also understood that each single account in
a given = protocol registers a Protocol Provider Service. But, from
the code = source I haven't seen where a Protocol Provider Service is
used. More =
explanations: In SIP when initiating a call where the Protocol Provider = Sevice corresponding to the account I am calling is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, = how , when and where this attribute is initialized?

Your help is really appreciated,
Thanks in advance,
"-----------------------------"
Best regards,
Fatoumata.

-----Message d'origine-----
De : zze-CAMARA Fatoumata RD-MAPS-GRE
[mailto:fatoumata2.camara@orange-ftgroup.com]
Envoyé : mardi 11 décembre 2007 17:58
À : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] Protocol Provider Service

Hi everybody!

I am looking at Sip Communicator code source in order to understand
its architecture and how the components interact with each other. I
have understood that each protocol registers a Protocol Provider
Factory = service, I have also understood that each single account in
a given protocol registers a Protocol Provider Service. But, from the
code source I haven't seen where a Protocol Provider Service is used.
More
explanations: In SIP when initiating a call where the Protocol Provider Sevice corresponding to the account I am calling with is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, how , when and where this attribute is initialized?

Best,
Fatoumata.

-----Message d'origine-----
De : Keio Kraaner [mailto:keio@ut.ee]
Envoyé : mardi 11 décembre 2007 17:01
À : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] system chat room

Hi!

The ChatRoom interface has a method called isSystem().
Could you please explain what a system chat room is?

Best
Keio Kraaner

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#13

Hello,
Thanks for your answer, it heps, now I have JMF working on Felix.
I would like the understand the logic behind the protocol sip in sip communicator. I looked at the code and notice that:
-A sip call has a state, an Id, mediaSession and may have more than one callParticipant,...
-A callParticpant has a call (the call it owns), an Id, a state as well, a Dialog,...
Can you please explain to me why do we need a state for both the call and the callParticipant?
A call may have more than one participant, but since now, It seems to me that a new call is created for each incoming ot outgoing call.The fact that a call may have more that one participant is intended for videoconferencing for example (three people conversation), isn't? Is this functionnality working?

Thanks,
Regards,
Fatoumata.

-----Message d'origine-----

···

De : Emil Ivov [mailto:emcho@sip-communicator.org]
Envoyé : lundi 4 février 2008 14:49
À : dev@sip-communicator.dev.java.net
Objet : Re: [sip-comm-dev] JMF in OSGi

Hello Fatoumata,

Yes we have JMF running in an OSGi bundle. In our case we include the whole framework inside the bundle jar.

Running it is a bit trickier since, depending on the Operating System, we need to know which set of native libs to include in the library path.
You can have a look at our "bundle-media" and "run" targets in the build.xml file in order to get an idea of how we handle the whole thing.

Hope this helps

Emil

fatoumata2.camara@orange-ftgroup.com написа:

Hi,
Can anyone give me some help about using JMF in Felix. I have not been able to make it work. Your help is appreciated.

Regards,
Fatoumata.

-----Message d'origine-----
De : zze-CAMARA Fatoumata RD-MAPS-GRE
[mailto:fatoumata2.camara@orange-ftgroup.com]
Envoyé : mardi 11 décembre 2007 17:58
À : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] Protocol Provider Service

Hi everybody!

I am looking at Sip Communicator code source in order to understand
its architecture and how the components interact with each other. I
have understood that each protocol registers a Protocol Provider
Factory = service, I have also understood that each single account in
a given protocol registers a Protocol Provider Service. But, from the
code source I haven't seen where a Protocol Provider Service is used.
More
explanations: In SIP when initiating a call where the Protocol Provider Sevice corresponding to the account I am calling with is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, how , when and where this attribute is initialized?

Best,
Fatoumata.

-----Message d'origine-----
De : Keio Kraaner [mailto:keio@ut.ee]
Envoyé : mardi 11 décembre 2007 17:01
À : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] system chat room

Hi!

The ChatRoom interface has a method called isSystem().
Could you please explain what a system chat room is?

Best
Keio Kraaner

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#14

Hi Fatoumata,

zze-CAMARA Fatoumata RD-MAPS-GRE wrote:

Hi, Thanks. I had already red the information. Reading again, It seems to me that it is the UI Service that uses Protocol Provider Service. Is it right?

ProtocolProviderService-s are providing access to protocols. They are used in the ui bundle and also in some of the plugins (account registration wizards for example).

Can someone tells me where the session parameters negociation is done in the SIP Protocol?

You could have a look in the OperationSetBasicTelephonySipImpl and in
media.CallSessionImpl.

Hope this helps.

Regards,
Yana

···

Regards,
Fatoumata.

-----Message d'origine-----
De : Benoit Pradelle [mailto:ze_real_neo@yahoo.fr] Envoy� : jeudi 13 d�cembre 2007 09:37
� : dev@sip-communicator.dev.java.net
Objet : Re: [sip-comm-dev] Protocol Provider Service Again

Hi,

You can find these informations on our website
(www.sip-communicator.org) in the developer documentation. Here is the exact link : http://www.sip-communicator.org/index.php/Documentation/PPBasics

Hope that helps,
Ben.

zze-CAMARA Fatoumata RD-MAPS-GRE a �crit :

Hi again,
I wrote two years ago to ask how Protocol Provider Service are used I have no answer yet.
Can someone help me please? I am in hurry I have to do a kind of diagram showing how the registered services interact with each other. My first email is below:
"---------------------------"
Hi everybody!

I am looking at Sip Communicator code source in order to understand its = architecture and how the components interact with each other. I have = understodd that each protocol registers a Protocol Provider Factory = service, I have also understood that each single account in a given = protocol registers a Protocol Provider Service. But, from the code = source I haven't seen where a Protocol Provider Service is used. More =
explanations: In SIP when initiating a call where the Protocol Provider = Sevice corresponding to the account I am calling is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, = how , when and where this attribute is initialized?

Your help is really appreciated,
Thanks in advance,
"-----------------------------"
Best regards,
Fatoumata.

-----Message d'origine-----
De : zze-CAMARA Fatoumata RD-MAPS-GRE [mailto:fatoumata2.camara@orange-ftgroup.com]
Envoy� : mardi 11 d�cembre 2007 17:58
� : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] Protocol Provider Service

Hi everybody!

I am looking at Sip Communicator code source in order to understand its architecture and how the components interact with each other. I have understood that each protocol registers a Protocol Provider Factory = service, I have also understood that each single account in a given protocol registers a Protocol Provider Service. But, from the code source I haven't seen where a Protocol Provider Service is used. More
explanations: In SIP when initiating a call where the Protocol Provider Sevice corresponding to the account I am calling with is gotten?
I've seen that the CallManager has a selectedProtolProvider attribute, how , when and where this attribute is initialized?

Best,
Fatoumata.

-----Message d'origine-----
De : Keio Kraaner [mailto:keio@ut.ee]
Envoy� : mardi 11 d�cembre 2007 17:01
� : dev@sip-communicator.dev.java.net
Objet : [sip-comm-dev] system chat room

Hi!

The ChatRoom interface has a method called isSystem().
Could you please explain what a system chat room is?

Best
Keio Kraaner

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#15

Hey Fatoumata,

fatoumata2.camara@orange-ftgroup.com написа:

Hello, Thanks for your answer, it heps, now I have JMF working on
Felix. I would like the understand the logic behind the protocol sip
in sip communicator. I looked at the code and notice that: -A sip
call has a state, an Id, mediaSession and may have more than one
callParticipant,... -A callParticpant has a call (the call it owns),
an Id, a state as well, a Dialog,... Can you please explain to me why
do we need a state for both the call and the callParticipant? A call
may have more than one participant, but since now, It seems to me
that a new call is created for each incoming ot outgoing call.The
fact that a call may have more that one participant is intended for
videoconferencing for example (three people conversation), isn't?

Yes, that's exactly the point.

Is
this functionnality working?

No, unfortunately not yet.

Emil

···

Thanks, Regards, Fatoumata.

-----Message d'origine----- De : Emil Ivov
[mailto:emcho@sip-communicator.org] Envoyé : lundi 4 février 2008
14:49 À : dev@sip-communicator.dev.java.net Objet : Re:
[sip-comm-dev] JMF in OSGi

Hello Fatoumata,

Yes we have JMF running in an OSGi bundle. In our case we include the
whole framework inside the bundle jar.

Running it is a bit trickier since, depending on the Operating
System, we need to know which set of native libs to include in the
library path. You can have a look at our "bundle-media" and "run"
targets in the build.xml file in order to get an idea of how we
handle the whole thing.

Hope this helps

Emil

fatoumata2.camara@orange-ftgroup.com написа:

Hi, Can anyone give me some help about using JMF in Felix. I have
not been able to make it work. Your help is appreciated.

Regards, Fatoumata.

-----Message d'origine----- De : zze-CAMARA Fatoumata RD-MAPS-GRE
[mailto:fatoumata2.camara@orange-ftgroup.com] Envoyé : mardi 11
décembre 2007 17:58 À : dev@sip-communicator.dev.java.net Objet :
[sip-comm-dev] Protocol Provider Service

Hi everybody!

I am looking at Sip Communicator code source in order to understand
its architecture and how the components interact with each other.
I have understood that each protocol registers a Protocol Provider
Factory = service, I have also understood that each single account
in a given protocol registers a Protocol Provider Service. But,
from the code source I haven't seen where a Protocol Provider
Service is used. More explanations: In SIP when initiating a call
where the Protocol Provider Sevice corresponding to the account I
am calling with is gotten? I've seen that the CallManager has a
selectedProtolProvider attribute, how , when and where this
attribute is initialized?

Best, Fatoumata.

-----Message d'origine----- De : Keio Kraaner [mailto:keio@ut.ee]
Envoyé : mardi 11 décembre 2007 17:01 À :
dev@sip-communicator.dev.java.net Objet : [sip-comm-dev] system
chat room

Hi!

The ChatRoom interface has a method called isSystem(). Could you
please explain what a system chat room is?

Best Keio Kraaner

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