[sip-comm-dev] Demo app using sip-communicator code


#1

Hello, I'm a long since sip advocate.

My interest started while at broadband operator Utfors in Sweden and I have
done some sipping in periods since 2002.
For a short period (some 5 months) I was involved in the Mobicents project.

I have just taken a quick look at sip-communicator and A LOT has happened
since I did the same in 2006!
Very nice job!

Now to the subject, a request for a little help!

I'm about to do a simple demo application serving a media file (mp3 , wav or
similar) to sip UA
More specifically I would like to use a suitable sip api to register with a
sip proxy (which is a "real" operator) and when an INVITE comes serve the
media file.

I don't have access to the operators environment so I don't really see sip
servlets or JSLEE as good candidates although it may be possible.
I'm looking for the simplest way to do this and then I thought about using
the code in sip-communicator for this.

What do you think about that?

In short again, DEMO APP is the sip ua serving the media file:

1. DEMO APP registers with proxy using sip address +46170170170@mydomain.se
2. Mobile phone with sip ua calls +46170170170
3. DEMO APP receives INVITE , responds and uses media file for RTP stream.

Could you give be basic hints about where to look in the sip-communicator
code to reuse for the sip UA stuff?

I am totally new to using mp3 or wav files sourcing the rtp stream, any
hints about this?

Thanks!


#2

Hey Niklas,

(inline)

Niklas Uhrberg написа:

Hello, I'm a long since sip advocate.

My interest started while at broadband operator Utfors in Sweden and I
have done some sipping in periods since 2002.
For a short period (some 5 months) I was involved in the Mobicents project.

I have just taken a quick look at sip-communicator and A LOT has
happened since I did the same in 2006!
Very nice job!

Thanks for the nice words!

[snip]
In short again, DEMO APP is the sip ua serving the media file:

1. DEMO APP registers with proxy using sip address
2. Mobile phone with sip ua calls +46170170170
3. DEMO APP receives INVITE , responds and uses media file for RTP stream.

Could you give be basic hints about where to look in the
sip-communicator code to reuse for the sip UA stuff?

We have a property that currently allows you to replace the default data
sources (web cam and mic) with a file. The property is:

net.java.sip.communicator.impl.media.DEBUG_DATA_SOURCE_URL

and you simply need to set it to the file that you'd like to use.
Something like:

file:/home/niklas/mydatasource.mov

Note that the property name prefix would appear as tags in the config
file. In other words you'd have something like
<net>
    <java>
        <sip>
            ...
                <DEBUG_DATA_SOURCE_URL value="file:/home/nik/myds.mov"/>
        </sip>
    </java>
</net>

You can then write a plugin that would listen for new calls received by
the protocol provider and automatically answer them when necessary.

Hope this helps
Emil

···

+46170170170@mydomain.se <mailto:46170170170@mydomain.se>

I am totally new to using mp3 or wav files sourcing the rtp stream, any
hints about this?

Thanks!

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#3

OK, I think this will do for purposes of the demo.

Thanks a lot!

/Niklas

···

On Wed, Oct 29, 2008 at 2:26 PM, Emil Ivov <emcho@sip-communicator.org>wrote:

Hey Niklas,

(inline)

Niklas Uhrberg написа:
> Hello, I'm a long since sip advocate.
>
> My interest started while at broadband operator Utfors in Sweden and I
> have done some sipping in periods since 2002.
> For a short period (some 5 months) I was involved in the Mobicents
project.
>
> I have just taken a quick look at sip-communicator and A LOT has
> happened since I did the same in 2006!
> Very nice job!

Thanks for the nice words!

> [snip]
> In short again, DEMO APP is the sip ua serving the media file:
>
> 1. DEMO APP registers with proxy using sip address
> +46170170170@mydomain.se <mailto:46170170170@mydomain.se>
> 2. Mobile phone with sip ua calls +46170170170
> 3. DEMO APP receives INVITE , responds and uses media file for RTP
stream.
>
> Could you give be basic hints about where to look in the
> sip-communicator code to reuse for the sip UA stuff?

We have a property that currently allows you to replace the default data
sources (web cam and mic) with a file. The property is:

net.java.sip.communicator.impl.media.DEBUG_DATA_SOURCE_URL

and you simply need to set it to the file that you'd like to use.
Something like:

file:/home/niklas/mydatasource.mov

Note that the property name prefix would appear as tags in the config
file. In other words you'd have something like
<net>
   <java>
       <sip>
           ...
               <DEBUG_DATA_SOURCE_URL value="file:/home/nik/myds.mov"/>
       </sip>
   </java>
</net>

You can then write a plugin that would listen for new calls received by
the protocol provider and automatically answer them when necessary.

Hope this helps
Emil
>
> I am totally new to using mp3 or wav files sourcing the rtp stream, any
> hints about this?
>
> Thanks!

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