[sip-comm-dev] Interface Java GUI with Asterisk PBX


#1

Hello,

As a general question, I am looking to create a custom
GUI that will basically act as a softphone. This GUI
will interact with an Asterisk server to complete the
calls. Instead of having the users dial a call, they
will choose a certain channel in the GUI which
constitutes a certain frequency and then I will most
likely open a conference call dynamically. I believe
the Asterisk-Java library can be used for creating the
connection to the Asterisk server and initiating the
conference calls. The issue I am having is the
real-time transmission of the audio. I can see that I
will most likely have to use JAIN SIP to initiate the
session and JMF to transmit the audio stream via RTP.
I have been researching this for a while and have
found a few examples. Here are my questions:

1.Am I on the right track with regards to the overall
system and where each component is used(ie: JAIN SIP,
JMF)?
2.Is Sip-Communicator just a more complex version of
the system that I want to build?
3.Do you know of any examples of anything similar to
what I am trying to build?
4.What is a good place to learn about receiving and
transmitting audio using RTP? (Are there any tutorials
that you know of?)

Any help or advice that you can provide would be
greatly appreciated.

Thanks very much,

Denis Kutman

···

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#2

Hello Denis,

Denis Kutman wrote:

Hello,

As a general question, I am looking to create a custom
GUI that will basically act as a softphone. This GUI
will interact with an Asterisk server to complete the
calls. Instead of having the users dial a call, they
will choose a certain channel in the GUI which
constitutes a certain frequency and then I will most
likely open a conference call dynamically. I believe
the Asterisk-Java library can be used for creating the
connection to the Asterisk server and initiating the
conference calls. The issue I am having is the
real-time transmission of the audio. I can see that I
will most likely have to use JAIN SIP to initiate the
session and JMF to transmit the audio stream via RTP.
I have been researching this for a while and have
found a few examples. Here are my questions:

1.Am I on the right track with regards to the overall
system and where each component is used(ie: JAIN SIP,
JMF)?

Seems like it, though I am not sure I completely understand your idea
and more precisely that part that says:

Instead of having the users dial a call, they
will choose a certain channel in the GUI which
constitutes a certain frequency and then I will most
likely open a conference call dynamically.

2.Is Sip-Communicator just a more complex version of
the system that I want to build?

SIP Communicator is way too complex for what you need. Many of the
things that we have are used for handling modularity, presence, instant
messaging, message history, and a meta contact list. You could of course
easily remove most of them by simply commenting them in the
felix.client.run.properties file.

3.Do you know of any examples of anything similar to
what I am trying to build?

The old version of SIP Communicator was very much what you needed ...
but unfortunately it is not around any more. I am not aware of other
java projects that do the same thing.

4.What is a good place to learn about receiving and
transmitting audio using RTP? (Are there any tutorials
that you know of?)

Try the JMF programmers guide and the sample code available on the sun site:

JMF pages:
http://java.sun.com/products/java-media/jmf/

JMF sample code:
http://java.sun.com/products/java-media/jmf/2.1.1/solutions/index.html

Programmers guide:
http://java.sun.com/products/java-media/jmf/1.0/guide/index.html

Hope this helps
Emil

···

Any help or advice that you can provide would be
greatly appreciated.

Thanks very much,

Denis Kutman

____________________________________________________________________________________
Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out.
http://answers.yahoo.com/dir/?link=list&sid=396545469

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