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Alejandro Caires - ABBAN | 4 Nov 2009 11:00
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Sofia-SIP & Java

Hi,

I would like to know, if exist an add-on, plugin or some other way to make Sofia-SIP "interact" with the Java programing language? Above those already known, JNA and JNI.

Best regards,

--
Alejandro Caires

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jonathan augenstine | 5 Nov 2009 17:45
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Re: Sofia-SIP & Java

Alejandro,


I do not know if you have received a reply on this yet, but basically, the short answer I am not aware of any add-on or plugin available for Java.  The Sofia-SIP library is C based, and other than the standard interfaces you already mentioned, Java is not really an option.  If you require Java, you would probably be in a better position by investigating the Java based SIP stacks that are out there.

Jonathan

On Wed, Nov 4, 2009 at 2:00 AM, Alejandro Caires - ABBAN <alejandro.caires-yctX8EbsscI@public.gmane.org> wrote:
Hi,

I would like to know, if exist an add-on, plugin or some other way to make Sofia-SIP "interact" with the Java programing language? Above those already known, JNA and JNI.

Best regards,

--
Alejandro Caires

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Abhilash K | 7 Nov 2009 00:07
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Sofia-Sip Video Call Example

Hi Guys,

We are really excited to be working on Sofia-Sip. But we right now
find some difficulties in placing video call through Sofia-Sip. Does
anyone have sample codes to get video call working using C and
Gstreamer?
Any help will be appreciated.

Thanks,

Best Regards,

Abhilash

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Meftah Tayeb | 9 Nov 2009 00:08
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Re: Sofia-Sip Video Call Example

hi,
the best way is to see an existing applications like freeswitch (
http://www.freeswitch.org)
is using Sofia Sip a a SIP Stack and provide a  Video Format (.FSV) to 
play and record
thank you
Abhilash K a écrit :
> Hi Guys,
>
> We are really excited to be working on Sofia-Sip. But we right now
> find some difficulties in placing video call through Sofia-Sip. Does
> anyone have sample codes to get video call working using C and
> Gstreamer?
> Any help will be appreciated.
>
> Thanks,
>
> Best Regards,
>
> Abhilash
>
> ------------------------------------------------------------------------------
> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day 
> trial. Simplify your report design, integration and deployment - and focus on 
> what you do best, core application coding. Discover what's new with
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> _______________________________________________
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> Sofia-sip-devel@...
> https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
>
>
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>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>   

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Bernhard Suttner | 10 Nov 2009 18:27

Codec Change

Hi,

I am using sofia-sip 1.12.9 for an client and have the following
problem:

- The call between A and S (S is using sofia sip) is up and running and
using PCMA (for both directions)
- A sends a ReINVITE to S with a new (fresh) SDP (all codecs exists but
PCMU is the first one)

Sofia sip does now automatically send the 200 OK (without accessing the
application) with PCMA. Therefore the call would be A using PCMA, S
using PCMU (or A receives PCMU and sends PCMA, S receives PCMA and sends
PCMU). 

Is it somehow possible to change that sofia sip will answer
automatically and "ask" the application before it will send the "old"
SDP?

Thanks in advane.

Best regards,
Bernhard Suttner

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Andrew Gibbons | 11 Nov 2009 11:29

Building sofia-sip with Xcode

Hi , we are trying to build Sofia library using Xcode tools. We have created an Xcode project 

and added the sofia files. We have run ./configure and it seems to have generate most of the
headers.

However there are some headers not created and thus giving problems  when we build ie

msg_protos.h not created, we can see there is an associated .in file

msg_mime_protos.h  not created

http_protos.h, http_tag.h   not created

sip_tag.h, sip_protos.h, sip_hclasses.h, sip_extra.h  not created

Also io.h and process.h cannot be found.

Are there some other scripts or somethng we need to run to get these headers, thanks

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Bergfreund, Eli | 12 Nov 2009 10:10
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Build and use sofia-cli

Hi,

I compiled the latest sofia-sip software on Linux.
I downloaded also sofia-cli (sofsip-cli-0.16). I want to run it , if possible , without a media.

“make” of sofia-cli  (after ./configure) fails (“make: *** No targets specified and no makefile found.  Stop.”)
How can I compile sofia-cli and link and run it with sofia-sip?
Should they have same root directory?

I didn’t find the sofsip-cli/src/test_media tool to “verify that all components are found”:
See in http://wiki.opensource.nokia.com/projects/SofSipCli:
To verify that all components are found succesfully, run the sofsip-cli/src/test_media tool. It will print a list of required elements and whether they were found or not.


Thanks in advane.
Best Regards,
Eli Bergfreund 

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Bernhard Suttner | 12 Nov 2009 13:34

Re: Codec Change

Hi,

I think I explained the scenario a little bit wrong. It's a server connect (so nua_invite_server_XXXX is
used) and the ReINVITE will call the nua_inivte_server_respond function which will not access the the
application but only re-use the already stored local SDP (this was previously saved with
SOATAG_USER_SDP). Is it somehow possible to force NUA to NOT re-use the stored SDP but access the
application which will then re-do the sdp negotiation and then answer with the SDP created from the application?

Thanks for every hint or idea.

Best regards,
Bernhard Suttner

-----Ursprüngliche Nachricht-----
Von: Bernhard Suttner [mailto:suttner@...] 
Gesendet: Dienstag, 10. November 2009 18:28
An: sofia-sip-devel@...
Betreff: [Sofia-sip-devel] Codec Change

Hi,

I am using sofia-sip 1.12.9 for an client and have the following
problem:

- The call between A and S (S is using sofia sip) is up and running and
using PCMA (for both directions)
- A sends a ReINVITE to S with a new (fresh) SDP (all codecs exists but
PCMU is the first one)

Sofia sip does now automatically send the 200 OK (without accessing the
application) with PCMA. Therefore the call would be A using PCMA, S
using PCMU (or A receives PCMU and sends PCMA, S receives PCMA and sends
PCMU). 

Is it somehow possible to change that sofia sip will answer
automatically and "ask" the application before it will send the "old"
SDP?

Thanks in advane.

Best regards,
Bernhard Suttner

------------------------------------------------------------------------------
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Timo Bruhn | 13 Nov 2009 07:31
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Re: Codec Change

Hi,

did you try so set NUTAG_AUTOANSWER(0) on the nh used for that call?

Best regards,
Timo Bruhn

> -----Ursprüngliche Nachricht-----
> Von: "Bernhard Suttner" <suttner@...>
> Gesendet: 12.11.09 14:53:36
> An: <sofia-sip-devel@...>
> Betreff: Re: [Sofia-sip-devel] Codec Change

> Hi,
> 
> I think I explained the scenario a little bit wrong. It's a server connect (so nua_invite_server_XXXX is
used) and the ReINVITE will call the nua_inivte_server_respond function which will not access the the
application but only re-use the already stored local SDP (this was previously saved with
SOATAG_USER_SDP). Is it somehow possible to force NUA to NOT re-use the stored SDP but access the
application which will then re-do the sdp negotiation and then answer with the SDP created from the application?
> 
> Thanks for every hint or idea.
> 
> Best regards,
> Bernhard Suttner
> 
> -----Ursprüngliche Nachricht-----
> Von: Bernhard Suttner [mailto:suttner@...] 
> Gesendet: Dienstag, 10. November 2009 18:28
> An: sofia-sip-devel@...
> Betreff: [Sofia-sip-devel] Codec Change
> 
> Hi,
> 
> I am using sofia-sip 1.12.9 for an client and have the following
> problem:
> 
> - The call between A and S (S is using sofia sip) is up and running and
> using PCMA (for both directions)
> - A sends a ReINVITE to S with a new (fresh) SDP (all codecs exists but
> PCMU is the first one)
> 
> Sofia sip does now automatically send the 200 OK (without accessing the
> application) with PCMA. Therefore the call would be A using PCMA, S
> using PCMU (or A receives PCMU and sends PCMA, S receives PCMA and sends
> PCMU). 
> 
> Is it somehow possible to change that sofia sip will answer
> automatically and "ask" the application before it will send the "old"
> SDP?
> 
> Thanks in advane.
> 
> 
> Best regards,
> Bernhard Suttner
> 
> 
> ------------------------------------------------------------------------------
> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day 
> trial. Simplify your report design, integration and deployment - and focus on 
> what you do best, core application coding. Discover what's new with
> Crystal Reports now.  http://p.sf.net/sfu/bobj-july
> _______________________________________________
> Sofia-sip-devel mailing list
> Sofia-sip-devel@...
> https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
> 
> ------------------------------------------------------------------------------
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> _______________________________________________
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> Sofia-sip-devel@...
> https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
> 

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Bernhard Suttner | 13 Nov 2009 11:44

Re: Codec Change

Hi,

thanks for that hint. I will test that. 

BR,
Bernhard Suttner

-----Ursprüngliche Nachricht-----
Von: Timo Bruhn [mailto:voip_voip@...] 
Gesendet: Freitag, 13. November 2009 07:32
An: Bernhard Suttner
Cc: sofia-sip-devel@...
Betreff: Re: [Sofia-sip-devel] Codec Change

Hi,

did you try so set NUTAG_AUTOANSWER(0) on the nh used for that call?

Best regards,
Timo Bruhn

> -----Ursprüngliche Nachricht-----
> Von: "Bernhard Suttner" <suttner@...>
> Gesendet: 12.11.09 14:53:36
> An: <sofia-sip-devel@...>
> Betreff: Re: [Sofia-sip-devel] Codec Change

> Hi,
> 
> I think I explained the scenario a little bit wrong. It's a server connect (so nua_invite_server_XXXX is
used) and the ReINVITE will call the nua_inivte_server_respond function which will not access the the
application but only re-use the already stored local SDP (this was previously saved with
SOATAG_USER_SDP). Is it somehow possible to force NUA to NOT re-use the stored SDP but access the
application which will then re-do the sdp negotiation and then answer with the SDP created from the application?
> 
> Thanks for every hint or idea.
> 
> Best regards,
> Bernhard Suttner
> 
> -----Ursprüngliche Nachricht-----
> Von: Bernhard Suttner [mailto:suttner@...] 
> Gesendet: Dienstag, 10. November 2009 18:28
> An: sofia-sip-devel@...
> Betreff: [Sofia-sip-devel] Codec Change
> 
> Hi,
> 
> I am using sofia-sip 1.12.9 for an client and have the following
> problem:
> 
> - The call between A and S (S is using sofia sip) is up and running and
> using PCMA (for both directions)
> - A sends a ReINVITE to S with a new (fresh) SDP (all codecs exists but
> PCMU is the first one)
> 
> Sofia sip does now automatically send the 200 OK (without accessing the
> application) with PCMA. Therefore the call would be A using PCMA, S
> using PCMU (or A receives PCMU and sends PCMA, S receives PCMA and sends
> PCMU). 
> 
> Is it somehow possible to change that sofia sip will answer
> automatically and "ask" the application before it will send the "old"
> SDP?
> 
> Thanks in advane.
> 
> 
> Best regards,
> Bernhard Suttner
> 
> 
> ------------------------------------------------------------------------------
> Let Crystal Reports handle the reporting - Free Crystal Reports 2008 30-Day 
> trial. Simplify your report design, integration and deployment - and focus on 
> what you do best, core application coding. Discover what's new with
> Crystal Reports now.  http://p.sf.net/sfu/bobj-july
> _______________________________________________
> Sofia-sip-devel mailing list
> Sofia-sip-devel@...
> https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
> 
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> what you do best, core application coding. Discover what's new with
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> _______________________________________________
> Sofia-sip-devel mailing list
> Sofia-sip-devel@...
> https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
> 

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