Talking Technologies

Ever asked yourself where to find files in Ubuntu packages?

2010-10-14T18:20:00.003+02:00

Have you ever asked yourself where to find the library ABC - and how to install it convenient using the apt-get install ... command in Ubuntu linux?

Well, I had the issue and almost went crazy trying to install the lib gtksourceview-2.0 ...

Then I found apt-file (see http://en.wikipedia.org/wiki/Apt-file). A really great tool which solved the issue described above immediately. apt-file search solved the issue in seconds.

A great tool!

Running Titanium from Appcelerator behind Proxy ...

2010-03-25T09:03:00.004+01:00

The Titanium X-mobile development framework looks nice - but is a bit goatish when it should run on a connected Ubuntu 9.10 - BEHIND a proxy.

I downloaded the latest (1.1 at this time) version, unzipped it and started it by clicking on the executable. This lead to a short installation first time and a login screen. Having created a login on the web site already, I tried to login immediately.
But the application said it couldn't connect to the internet to login.

On a plain internet connection everything went smoothly.

Browsing the web indicated that setting the http_proxy environment variable might help.
Setting the variable on the bash with "export http_proxy=:" helped. Starting the Titanium Developer Kit from this bash allowed me to login.

Interestingly, this is not really documented. Perhaps this post saves you some time ;)

-- Michael.

Setting proxies for ANT and IVY

2010-02-08T09:34:00.006+01:00

In a bash based environment simply type:

export ANT_OPTS="-Dhttp.proxyHost=set_your_proxy_here -Dhttp.proxyPort=set_your_port_here"

Eclipse and Proxies ...

2010-02-03T19:34:00.003+01:00

Edit eclipse.ini and append:

-Dorg.eclipse.ecf.provider.filetransfer.excludeContributors=org.eclipse.ecf.provider.filetransfer.httpclient
-Dhttp.proxyPort=1234
-Dhttp.proxyHost=yourproxygoeshere
-Dhttp.nonProxyHosts=localhost|127.0.0.1

Liferay 5.2.3 ... Compiling EXT and good-to-knows ...

2010-01-19T14:16:00.006+01:00

Currently, I'm having some looks at Liferay 5.2.3 and how stable, function-rich of an environment it is. Might be a good technology candidate for our portal site? Well, let's see.

I'm reading the book "Liferay Portal 5.2 Systems Developmen" by Jonas X. Yuan (Packt Publishing). Part of the book is to setup the development environment EXT and the SDK portion of Liferay. The author actually proposes to download the latest versions of EXT and SDK from the trunk. I had happy 2 hours downloading and re-doing the compilation and development process. Actually, it turned out that the trunk version of the author DID obviously compile - mine didn't ...

So, what I did is to download the latest release version 5.2.3. Building the liferay 5.2.3 portal worked fine - but ext failed ... But it was a quite simple "bug". Costed again 1 hour of searching, but well ...

In the file ext/build-common.xml locate the entry:

<path id="project.classpath">
   <pathelement path="${classpath}" />
   <path refid="lib.classpath" />
</path>

and add the highlighted elements below:

<path id="project.classpath">
   <pathelement path="${classpath}" />
   <fileset dir="${project.dir}/modules" includes="*.jar" />
   <fileset dir="${project.dir}/ext-service" includes="*.jar" />
   <path refid="lib.classpath" />
</path>

Without the above mentioned additions ext won't compile on my system.

As soon as you're ready to go, you need to take care about you XML-files and especially how you wrap values and the surrounding XML tags. The file ext/ext-web/docroot/WEB-INF/portlet-ext.xml is needed to configure your portlets developed during the course of the book. Be sure you write classes without tabs or spaces:

Not like this:

<resource-bundle>
  com.liferay.portlet.StrutsResourceBundle
</resource-bundle>

but like this:

<resource-bundle>com.liferay.portlet.StrutsResourceBundle</resource-bundle>

Otherwise you'll get nasty error messages leading to some quite frustrating experiences ...

I'll keep you posted!

Running PhoneGap on HTC Hero

2009-12-09T08:35:00.003+01:00

PhoneGap has quite a promise - do applications which are cross-mobile-platforms! How? Utilizing JavaScript glued together with native libs. For further information, have a look at the PhoneGap site.

I was seriously curious if that promise can be hold. I wanted to test PhoneGap on my Android 1.5 HTC Hero phone. First, it took me quite some attempts to identify the right version of PhoneGap, get the development environment up and play around. Initially, I wasn't successful at all.

I setup the environment as described here: Getting started with Android PhoneGap in Eclipse. Then I downloaded PhoneGap 0.8.2. Unpacked and imported it into Eclipse and did some modifications (since HTC Hero runs Android 1.5 ...).

What I did?

1) Open the project settings for PhoneGap (Alt+Enter). Navigate to "Android" (if it's not there your dev environment is most likely not properly setup ...). Select "Android 1.5".

2) Open PhoneGap's Manifest "AndroidManifest.xml". Here, locate the section:


   <supports-screens
       android:largeScreens="true"
       android:normalScreens="true"
       android:smallScreens="true"
       android:resizeable="true"
       android:anyDensity="true"
       />

and delete it.

3) In the same file, locate:


 <uses-sdk minsdkversion="4">

and modify it to


 <uses-sdk minsdkversion="3">

4) Deploy it onto the device!

Amazing!

OpenSSO loops infinite after showing login screen

2009-12-04T11:14:00.003+01:00

Interesting problem ...

I just integrated JBoss 5.1.0 with OpenSSO Express Build 7 (Apr 2009) via the Agent sitting in the application server. On Glassfish, everything went smooth and fine - without any issues. Now, running the very same example (see the Lab 6727 of JavaOne 2009) named "mini-agentsample" on JBoss - it leads to the login page, does the authentication and then results in FireFox (3.5.5 - on Ubuntu 9.10) saying:

The page isn't redirecting properly

Firefox has detected that the server is redirecting the request for this address in a way that will never complete.

* This problem can sometimes be caused by disabling or refusing to accept
cookies.

Try Again.

Nice. Looking for a solution took me some time. SUN's wiki talks about "Infinite redirection between OpenSSO server and OpenSSO client" - which actually describes the problem very well - and more importantly solved it!

You need to login to the OpenSSO server admin console and navigate to:
* Configuration
* Servers and Sites
* Default Server Settings
* Advanced

Find "com.iplanet.am.cookie.c66Encode" and set it to "true". For me it worked.

JSON and RESTful interfaces - best practices

2009-11-30T15:39:00.003+01:00

Looking around for JSON and REST and associated technologies and blogs, I found the blog of Edwin Khodabakchian. He talks about best practices learned during building their web application named "Feedly".

Very interesting read.

Comparison of Java Enterprise Edition (JEE) and Spring (Acegi) Security

2009-11-19T12:09:00.003+01:00

Looking around in the depth of the WWW, I found a good comparison on JEE and Spring based security. It's an article on JavaWorld by Dr. Xinyu Liu.

Definitely worth reading!

Solution: Eclipse 3.5.0 doesn't react on mouse interactions after updating Ubuntu 9.04 to 9.10

2009-11-18T10:30:00.004+01:00

Eclipse 3.5.0 on Ubuntu 9.04 worked fine - without any issues actually. When updating Ubuntu from 9.04 to 9.10 I experienced the odd behavior of some buttons not reacting on my mouse clicks any more. The workaround used to be to select the button with the mouse and then hit . But that's really odd.

It took some time until I found a blog entry in German. The essence is to start Eclipse with the prefix "env GDK_NATIVE_WINDOWS=1".

So, if you utilize a launcher in Ubunutu, just modify the launcher as shown in the image below:

Command: env GDK_NATIVE_WINDOWS=1 /opt/eclipse/eclipse

Android Audio Recording and PlayBack ...

2009-10-30T16:47:00.007+01:00

The Google Android SDK allows audio recording and playback as well. The code snippet below shows an Activity capable of recording and playing back audio on a device running Android.

Asuming you're using Eclipse in combination with the Android SDK, you need to create a layout named "recorder.xml".

<linearlayout android="http://schemas.android.com/apk/res/android"
orientation="vertical" layout_width="fill_parent"
layout_height="fill_parent">

<button text="Start Recording" id="@+id/btn_record"
layout_width="wrap_content"
layout_height="wrap_content"></button>

<button text="Stop Recording" id="@+id/btn_stop"
layout_width="wrap_content"
layout_height="wrap_content"></button>

<button text="Play Recording" id="@+id/btn_play"
layout_width="wrap_content"
layout_height="wrap_content"></button>
</linearlayout>

Then, the below shown activity responds to the three buttons displayed on your UI screen. You need to add a permission to your android Manifest file:

<uses-permission name="android.permission.RECORD_AUDIO">

Then deploy and run ... funny!

package com.maretzke.audio;

import java.io.FileInputStream;
import java.io.FileOutputStream;
import java.io.IOException;

import android.app.Activity;
import android.media.MediaPlayer;
import android.media.MediaRecorder;
import android.os.Bundle;
import android.view.View;
import android.view.View.OnClickListener;
import android.widget.Button;

public class RecorderUI extends Activity implements OnClickListener {
MediaRecorder recorder;
final String fileNameStr = "audio-android.3gp";

/** Called when the activity is first created. */
@Override
public void onCreate(Bundle savedInstanceState) {
 super.onCreate(savedInstanceState);
 setContentView(R.layout.recorder);

 ((Button) findViewById(R.id.btn_record)).setOnClickListener(this);
 ((Button) findViewById(R.id.btn_stop)).setOnClickListener(this);
 ((Button) findViewById(R.id.btn_play)).setOnClickListener(this);
}

@Override
public void onClick(View v) {
 int id = v.getId();

 switch (id) {
 case R.id.btn_record: {
     System.out.println("Record pressed.");
     try {

         recorder = new MediaRecorder();
         FileOutputStream fos = openFileOutput(fileNameStr, MODE_PRIVATE);

         recorder.setAudioSource(MediaRecorder.AudioSource.MIC);
         recorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP);
         recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB);
         recorder.setOutputFile(fos.getFD());
         recorder.prepare();
         recorder.start();

     } catch (IOException e) {
         System.out.println("IOException caught during recording.");
         e.printStackTrace();
     }
     return;
 }
 case R.id.btn_stop: {
     recorder.stop();
     recorder.release();
     System.out.println("Stop pressed.");
     return;
 }
 case R.id.btn_play: {
     System.out.println("Play pressed.");

     MediaPlayer mp = new MediaPlayer();
     try {
         FileInputStream fis = openFileInput(fileNameStr);
         mp.setDataSource(fis.getFD());
         mp.prepare();
     } catch (IllegalArgumentException e) {
         e.printStackTrace();
     } catch (IllegalStateException e) {
         e.printStackTrace();
     } catch (IOException e) {
         e.printStackTrace();
     }
     mp.start();

     return;
 }
 }

}
}

RFC 3428 SIP IM Client on android - Want one?

2009-10-30T16:43:00.002+01:00

Implementing a RFC 3428 SIP IM client on Android - A quick-starter's guide.

The step-by-step guide shows a JAIN SIP (JSR-32) based SIP IM client implementing RFC 3428 - on Google's Android operating system. Step-by-step guide, sources and compiled application (APK) - ready to download!

Go and grab the code!

Maven -- Where to find groupId, artifactId and version for a dependency in the POM?

2009-10-09T09:09:00.004+02:00

In Maven2 you sometimes need to lookup the groupId, artifactId and perhaps version of the tools you want to utilize in the scope of your project. There is a repository to ask provided by a company named Sonatype.

It provides a handy GUI and allows an easy search of the assets you need to name in the dependencies section of your POM file.

SubVersion -- Merging branches back into the main trunk

2009-10-07T13:40:00.003+02:00

In SW development, SubVersion is an established tool to manage source code and other artifacts. In SW development it's common practice to create private branches of the actual main trunk of a SW development project. After a while, merging the private branch into the main trunk is a common done task.
To accomplish this in SubVersion you utilize the merge command. It's important to follow the notion of [TRUNK] [BRANCH] when merging - otherwise the repository might be messed up - nothing really harming since everything is still stored in the repository - but you might end in a whole pile of work ...

svn merge --reintegrate svn://[TRUNK] svn://[BRANCH]

e.g.:

svn merge --reintegrate svn://localhost/projects/NumOps/trunk svn://localhost/projects/NumOps/branches/anotherbranch

WURFL - Mobile Device Parameter Database

2009-09-22T15:57:00.004+02:00

Rendering content onto specific devices is a painful activity since mobile devices differ significantly in their capabilities. It's not only having 2 or 3 mainstream browser technologies you need to test your content against - it's more like 400 devices ... Even if you're able to identify the device browser - where from do you know what capabilities the device eventually has build-in?

The answer is WURFL (Wireless Universal Resource File). This is an ambitious OPEN SOURCE project hosted on sourceforge.net. The initiator is Luca Passini. He also runs the maintenance of the XML document containing around 400 device descriptions. WURFL also provides API's to access the information from various programming languages.

Very nice!

API to access Networks needed? Nice overview found ....

2009-09-10T12:33:00.003+02:00

Out there in the Web 2.0 world - or shall I say Telco 2.0-world - there are quite some companies offering access to their core assets. Think about Amazon, Google, Skype, Ribbit and quite some others.

A colleague of mine pointed me to this site: www.programmableweb.com. Following the link, the site shows quite a big set of sites exposing API's for telephony purposes.

Telephony in the Internet is an awfully interesting topic to follow. I'm curious as hell to see who actually makes the race for the customer. Will the newentrants succeed? Or the giants from past?

We'll see!

JSR-309 is moving on ...

2008-10-03T22:03:00.005+02:00

The "JSR 309: Media Server Control API" ... what is it about? What's the problem it solves?

The JSR-309 starts the public review ballot on 7th of October. Let's see how fast the application server vendors pick up the idea of implementing connectors to the various media servers --- well, or the media server vendors start implementing connectors for various application servers.

Sounds very much like a chicken-egg-problem. Interesting to see what will happen!

JSR-289 finished!

2008-10-03T22:01:00.004+02:00

The JSR-289 made its way through standardization. It's finished. The JCP-site says it's finished since 21st August 2008.

Let's see when the first commercial products (beside the Oracle Communications Converged Application Server 4.0) will happen.

Who is best - JAIN™ SLEE or JEE/SIP Servlet?

2008-08-05T15:41:00.005+02:00

I'm following the religious fight between the two condenters JAIN™ SLEE and SIP Servlet since quite some time - and it very much reminds me on the fight between "Windows versus Unix" "black versus white" ...

If you watch googlefight on JAIN SLEE vs. SIP Servlet it's clear - 16.500 : 47.200 hits. However, that's pure counting - and ... well at least funny.

I've worked both on and with JAIN SLEE and with SIP Servlet application servers and hence thought it might be useful to share my thoughts on this technology competition. Have a look at the whitepaper!

JSR-289 Application Router based on the principles of Distributed Feature Composition (DFC)

2008-08-05T15:33:00.003+02:00

Eric Cheung posted just recently a howto on echarts.org. He explains how an application router - being compliant to JSR-289 and following the principles of Distributed Feature Composition - invented by Michael Jackson and Pamela Zave -can be combined with the latest SailFin JEE/SIP Servlet application server.

His howto is based on a paper that describes the DFC based application router in greater detail.

This is a signal - JSR-289 happens and brings value to the SIP Servlet community.

Trillium Poster - An excellent overview of network elements and interfaces

2008-07-23T14:54:00.004+02:00

During one of the famous Barcelona 3GSM / MWC events, I got in touch with Continous Computing / Trillium. They presented a great overview poster of network elements and interfaces. The poster addresses IMS, IPTV, VoIP, 2.5G, 2G, PSTN networks.

You can order the poster here at Continuous Computing.

JAJAH, Ribbit, GrandCentral ... the new communication providers?

2008-05-05T11:33:00.006+02:00

Browsing the web for communication offerings, three interesting offerings crossed my way.

The first one is quite since some time around - JAJAH . JAJAH is around since quite some time and allows placing calls without installing any clients on the PC - or on the phone.

JAJAH offers quite cheap international connections and advertises to connect people without installation needed + on cheap rates. JAJAH focuses on connecting people for voice conversation. I'd rate it as a very basic service - however already providing communication capabilities in competition to the incumbent telephony companies. With JAJAH, the communication device is less important than the actual communication. It works seamlessly with any telephone which is connected to either a mobile or landline network - and reachable via a number.

Ribbit with their application named "amphibian" is aiming for value-added phone services. They offer actually a quite rich IP Centrex like service which offers a dashboard to follow-up calls, manage contacts and voice mails in an efficient way.
Ribbit also aims at mixing mobile and landline phones with messaging and text capabilities. Interesting at Ribbit is the feature mix allowing to manage contacts, to initiate calls via clicking onto a button, searching your voice mail box and other cool features. Furthermore, even more impressive is the idea collection blog they run as the Ribbit Idea Wall.
Quite some cool ideas are collected with telephony services in mind. Cool!

At a recent business meeting somebody told me about GrandCentral.com. "Have you seen it?" I was asked. Nope. Not so far. Have you? If not, you should definitely look at it. It looks like the solution to what some EMEA based operators are wanting to offer to their customers. However, GrandCentral gives most of it for free. Hmmm. It was not really surprising to me that Google actually got hold of this - and now offers it as "perpetual beta" (I'd guess) to the public.
It offers you one central phone number which can be routed to either your fixed line phone(s) or your mobile phone(s) - all rules based. Furthermore, it offers you one centralized voice mail system - for all devices. It runs a web based address book and allows "Click2Call" features. On top of this it offers quite some further features like: personalized voice mail greetings, call recording, call blocking, notifications, web-call button and quite some more - see yourself.

So, why spending time blogging about these? Well, quite some operators - traditional or incumbent operators are currently trying to increase loyalty, stickiness. They introduce "Click2Service" features, offer IP Centrex services to their customers - but are always somewhat limited - or you have to pay for the service.
Here, internet players are about to revolutionize the voice market - as well. They offer real value-added services to the now-commoditized voice communication market. They offer most of the service for free and create another dimension of stickiness. People start utilizing their offerings and turn away from their traditional or incumbent communication service providers - making them - at best - a bit-pipe.
Why do I believe in this happening? Well, it took Apple to revolutionize the user interface for mobile devices. It was not the incumbent players in the market being brave enough to break paradigms - it was a new-entrant to the market. This principle, I'd guess, we'll see quite often in the future.

Let's see!

Requirements for SIP Servers on SIP-I / SIP-T handling ...

2008-04-25T11:50:00.003+02:00

From my discussions with various people being involved in VoIP networks on SIP basis it was highlighted that the SIP application server should not take care about SIP-I and SIP-T in theory - but practice is actually different.

If the signaling gateway or media gateway does proper ISUP-to-SIP conversion, the SIP application server shouldn't need to read, update or create SIP-I attachments. But sometimes, the signaling gateway is not acting as specified. In such cases a SIP server needs to create or read SIP-I attachments which then will be translated by the SG or MGW into proper ISUP messages into the PSTN.

Creating, parsing and modifying these attachments is a performance-hungry operation. SIP-I is binary encoded ASN.1 following BER rules. Especially the fact that the SG or MGW can only be configured to create these SIP-I attachment either allways or never. So, this will put quite some load on the SIP application server - if switched on.

Especially, in the case where the application server is a Java based application server it might be even harder to do these en-/decoding operations in Java.

RFC 3666's Call Flows -- SIP to PSTN Dialing

2008-03-31T12:52:00.003+02:00

Interesting to see there's even a whole RFC (RFC 3666) dealing with SIP to PSTN call flows. As usual, the textual descritption of RFC 3666 "Session Initiation Protocol (SIP) Public Switched Telephone Network (PSTN) Call Flows" is not that good to read.

Thanks to Tech Invite however, there is a nice graphical representation available at http://www.tech-invite.com/Ti-sip-CF3666.html.

Search ITU-T recommendations at ITU-T web site

2008-03-20T14:00:00.002+01:00

ITU-T recommendation papers are free to download. Simply browse to the website of ITU-T: http://www.itu.int and search for the papers needed.

Example Telecommunication Call Flows

2008-03-20T13:44:00.002+01:00

Looking for GSM, SIP, IMS, ISUP example call flows? Have a look at the site from EventHelix.com.

Next Generation Networks - SIP and ISUP interworking

2008-03-20T13:04:00.002+01:00

In SIP, everything looks quite simple and easy. Very human readable format, no ISUP binary encodings and bit-counting. In pure SIP network environments, everything is great and feasible. However, when it comes to interworking between SIP networks and ISUP based networks protocols are needed to carry the additional information provided by ISUP.

Here, the three protocols

BICC (Bearer Independent Call Control) by ITU-T
SIP-T (SIP for Telephones) by IETF
SIP-I (SIP with encapsulated ISUP) by ITU-T, ANSI, ETSI

provide different solution approaches.

The paper "Why SIP-I? A Switching Core Protocol Recommendation for GSM/UMTS Operators"at 3GAmericas.org provides an excellent comparison of the various approaches.

Key points
SIP-I is considered to be the superior approach for operators based on the following points:

Assumptions on trust and security environment
Encapsulation procedures & message mapping
Support of RFCs
User plane interoperability

The key areas where SIP-I and SIP-T differ are the following points:

Trust and Security
Encapsulation Procedures & Message Mapping
Support of RFCs
User Plane Interoperability
Support of Forking

Standards Overview
BICC

Standardization body: ITU-T
CS1 defined in Q.1901 (2000)
CS2 defined in Q.1902 (2001)

SIP-T

Standardization body: IETF
RFC 3204 "MIME media types for ISUP and QSIG Objects"
RFC 3372 "Session Initiation Protocol for Telephones (SIP-T): Context and Architectures"
RFC 3398 "Integrated Service Digital Network (ISDN) User Part (ISUP) to SIP Mapping"
RFC 3578 "Mapping of ISUP Overlap Signaling to SIP"

SIP-I

Standardization body: ITU-T
ITU-T Q.1912.5 covers SIP interworking with ISUP (Q.761-Q.761) and BICC (Q.1902.1-Q.1902.4)
Defines 3 Profiles:

Profile A does not encapsulate ISUP messages. ISUP information is mapped towards SIP headers. Standards of interest are: 3GPP TS 24.229, 3GPP TS 29.163
Profile B is also a pure SIP based solution generalizing some 3GPP specific details of Profile A to allow interworking with a range of ISUP networks.
Profile C (SIP-I) adds ISUP encapsulation to Profile B. Profile C is found where ISUP networks are interconnected via SIP backbones. The encapsulated ISUP messages are used for various purposes (e.g. meet regulatory requirements).

Sailfin V1b21 vs. Mobicents SIP Servlets 0.1

2008-02-28T10:48:00.002+01:00

I visited the website of sailfin (https://sailfin.dev.java.net/) quite some time ago - shortly after Ericsson and Sun announced the birth of the sailfin project. At this time, sailfin was not impressive at all. I guess the website didn't improve that much - but sailfin has!

Having a natural interest in SIP Servlet engines, I thought I should have a look at the two open source alternatives being in the market and see how far they are actually. I downloaded:

Sailfin version 1 build 21 from https://sailfin.dev.java.net/
Mobicents SIP Servlets 0.1 from https://sip-servlets.dev.java.net/

As a test-case I thought I should try the already developed and tested SIP Registrar I developed some time ago on the BEA WebLogic SIP Server. The idea is to try to execute the same code base on the other SIP Servlet engines.

Mobicents SIP Servlets 0.1
I checked-out the source version of Mobicents SIP Servlets 0.1 and had a very nice experience with the build process. Wow - went through smoothly. Maven did its job very, very well. The documentation and overall impression is great. I decided to use the Tomcat Servlet Engine. The container started as expected and I was ready to push the application package into the webapps directory for auto-deployment. The SAR was deployed correctly and the system seemed still quite stable. However, it showed that some statements in the deployment descriptor sip.xml needed modification. After those modifications, the application deployed correctly and was up. Impressive. Then, when trying to send SIP REGISTER messages to the SIP Server, there were some issues with the SIP stack. After seeking the error in the application, I figured that something was wrong in the SIP stack - or the wrapper. Not being successful - but still a very good start - especially for a 0.1 version! I will follow closely - if time allows.

Sailfin v1 build 21
I started as I did with Mobicents SIP Servlet - checked out the source. But there was no documentation available. The one I found did not really work like I expected. The ANT files didn't really work - a disappointing experience. After some time, I decided to give the binary a try. Download, unzip, runs. Hmmmm. I dropped the application into the autodeploy folder and it runs - without any changes whatsoever. Even the web-portion executed. Wow, I'm impressed.

So, both cases are very promising. I learned that my test-application needs some modifications - which will be done in SIP Registrar 2.0. From a very basic testing perspective, both SIP Servlet engines are interesting candidates to watch.

Let's see how the answer will look like for non-functional requirements like clustering, high availability and failover - the domain of commercially available products.

Mobicents to release SIP Servlet engine

2008-02-01T09:34:00.000+01:00

Have you seen this in the java.net forums? The Mobicents team announced the availability of a SIP Servlet compliant application server container. The team contains currently 4 members and they run a project web site: https://sip-servlets.dev.java.net/.

In my personal opinion this implementation will gain much more stability and credibility than other available open source implementations of the SIP Servlet standard.

And it's interesting to see ... the religious war about JSLEE and SIP Servlet as competing technologies seems to be over ...

Interestingly, the Mobicents team decided to build their own SIP Servlet implementation on Tomcat - a servlet-only implementation. From my perspective they're giving away quite some benefits from the JEE environment - but I guess the next step will be to offer a tight integration with the JBoss environment.

Definitely a project to watch!!!

BEA is no longer on its own - Oracle to acquire BEA

2008-01-22T13:31:00.000+01:00

Most likely, you've heard - or read about the deal already: BEA will be acquired by Oracle (http://www.oracle.com/bea/index.html). Wow, what a bang in the industry.

I think it was not the fact of being bought from BEA perspective - it's more the timing and WHO actually bought BEA. Oracle.

Take a look at the results when googling for "BEA ORACLE overlap". It's amazing. The announcement of Oracle / BEA made quite some customer of both companies hesitant to make the investment right now. Which product line will survive? This seems to be the million USD question.

I sincerely hope that the CEO's of the companies have already teams setup and are working on the product portfolio clean-up.

In an German newspaper I read about the overlap in the product portfolio:

High degree of overlap:

Java Application Server
Enterprise Service Bus
Business Process Management Server
Event Management
Software for Telecommunication

Low degree of overlap

Integration Server
SOA Management
Information Fabric
Collaboration
Security

No overlap

Transaction Monitor
Java Virtual Machine
Realtime Java
Business Process Modeling
Java Development Tools
Business Activity Monitoring
Business Rules
Identity Management

Well, time will tell!

Voice Applications? Just one click away!

2007-10-10T13:59:00.000+02:00

So far it has been fairly complex to actually develop voice centric applications. Do you have a text-to-speech engine running on your server? An interactive voice recognition system? Are you fluent in VoiceXML or all the other nice dialects? No, then you might want to visit a company named Voxeo.

They offer a Java based, web-centric tool to develop voice applications in a graphical way.

It took me 15 minutes to compose (okay - it's useless) an application - try it! Dial +1 347 901 5891 ("1" stands for "yes"; "2" stands for "no").

I am really impressed by the easy-to-use interface and the capabilities offered by this company. Why run your own highly complex and costly infrastructure? Outsource it and just purchase the services you want and need.

Great idea!

Mesh it up! Web 2.0 arrives @ Telecommunication

2007-10-10T13:54:00.000+02:00

Already earlier in 2007, O'Reilly Media and StrikeIron ran the first Telephony Meshup Contest. Amazing - Web 2.0 arrives at the Telecommunication World.

I highly recommend to have a look at the Contest Finalists.

It is astonishing what people invent when some interfaces ready-to-be-used are offered and they simply need to drag and drop these offered bricks together.

API for SIP Application Server -- Media Server Binding

2007-08-03T09:18:00.001+02:00

Today, talking to SIP Application Server vendors and to Media Server vendors it seems like they agree on the value of their existence - but if you start digging down into details it turns out that the support for each other is quite minimalistic.
In (IMS-) theory the SIP Application Server communicates via SIP with the Media Server, however on a practical level it seems close to none Media Server vendor is too serious about their SIP interface (at least if you want to unveil the whole capabilities of the server).
So, the reality sounds like MSCP, voiceXML, MSCML, netann, MSML, MOML, and whatever protocols are used between the Media Server and the Application Server. Not very appealing to dive into details of every protocol.

But there seems to be a light at the end of the tunnel - the JSR-309: Media Server Control API. The JSR aims for abstracting the protocol clutter and easing the live of developers. The API is introduced inside the (SIP) Application Server and allows the developers to interact with the Media Server via the well-defined and standardized API. She/he no longer needs to twiddle the protocol details - that's the task of the server vendors.
And it seems the JSR-309 gets real support from the industry. The expert group reads like the who-is-who in media server industry.
The JSR aims to finish early 2008 - let's see!

RFC 3261 compliant SIP Registrar as SIP Servlet Implementation

2007-07-27T12:46:00.000+02:00

Just finished the documentation work on the SIP Registrar entirely written in Java and SIP Servlet technologies. The aim of the implementation is to be as close as possible to the RFC 3261 description.
The implementation is well-documented and also well-tested. A whole bunch of SIPp test scripts ran against the implementation. The architecture, design and implementation are well described in the HowTo document. The Setup document explains how to compile and deploy the SIP Registrar into a SIP Servlet application server (here: BEA's WebLogic SIP Server).

You can access the source and documentation here.

Happy registering!

SIP Servlet 1.1 - JSR-289 moved to Early Draft Review

2007-02-01T14:06:00.000+01:00

JSR-289 moved to its next stage from private discussion in the expert group to the public review phase. The SIP Servlet 1.1 specification did evolve quite a bit and it's definitely worth browsing through the latest specification improvements.

The specification can be obtained from the JCP web site.

Especially the Application Router concept adds significant value to the SIP Servlet specification. In JSR-116 the composition and chaining of SIP Servlets being hosted in a JSR-116 SIP Application Server is very static and done through a XML definition. In JSR-289, the whole behavior changes significantly. Now, the standard specifies an API which allows the SIP Application Server to externalize the decision of where to route which invocation under what conditions.
A further point to concentrate is the shared session state between SIP sessions and e.g. HTTP session. The introduction of the SipApplicationSession object allows the bundling of various session information into one head object. That's very convenient in the case where sessions need to be correlated.

And obviously there is more to discover. A recommended reading!

Any questions regarding SIP traffic?

2006-10-18T15:43:00.000+02:00

A colleague of mine pointed me to a great website on SIP message flows. If you ever had a question on how SIP traffic does look like in certain scenarios you need to have a look at the site's SIP examples.

The site is operated by Joel Repiquet and is a superb collection around SIP signalling.
It's definitely worth visiting Tech-invite.

The 3G IP Multimedia Subsystem (IMS) - the yellow book ...

2006-04-07T15:56:00.000+02:00

Looking for a book which explains the IMS system? Surely you recognized the "red" book by Miikka Poikselka explaining the Nokia view on things and the "yellow" book by Gonzalo Camarillo talking the reader through the Ericsson understanding of IMS.

I was looking for a newly published book to polish up my understanding of the IMS system - and found the "yellow" book more attractive ...

The book gives a great overview of the system, talks through the basic call flows through an IMS-System, explains every system in great detail, talks about the various protocols, introduces the call flows for various applications and so on.

The good thing about the book is that it is very good readable. It is not boring technical writing (although it's also not a criminal or a novel ...). But it's good reading - and it helped me a lot to increase my understanding of Rf, Ro, Sh, ISC, ...

A good reading!

-- Michael.

KTS, PBX, Hosted PBX, IP Centrex, CTI, iPBX and WPBX ?????

2006-03-17T23:00:00.000+01:00

Ever asked yourself about these acronyms? A nice and short book for around 15€ gives a lot of answers and is really easy reading!

The author gives a historic introduction into telephony systems and moves to a description of various telephony systems (KTS, PBX, CTI, ..). After explaining them quite thoroughly, he moves on to describe typical call processing features like e.g. call forwarding, voice mail, call centers, conference call, ...

Nice book, easy reading - not only for telcom specialists!

-- Michael.

Book on Parlay - Introduction and Reference: Parlay/OSA by Musa Unmehopa, Kumar Vemuri and A

2006-03-17T22:43:00.000+01:00

The book, roughly 300 pages and around 90€ expensive, helps solving the puzzle around the Parlay standards. For me a lot of the pieces felt into place whilst reading this book. After giving a bit of a background on the overall scene, the authors explain the relationship between the various standards (ETSI, Parlay, 3GPP). Next, a good readable overview on the various capabilities - including an outlook - follows.

Now, the book gets a bit more exciting ... The authors start talking about migrating the theory into practice. They talk about building carrier-grade systems, a realization path for Parlay, interworking with network legacy and some other "advanced" topics.

Great book and nice to read - All you need to know about Parlay (and is not written into standards).

-- Michael.

Information about CIMD and SMPP

2006-01-13T20:17:00.000+01:00

In the messaging domain in GSM networks the both protocols CIMD (Computer Interface to Message Distribution) and SMPP (Short Message Peer-to-Peer Protocol) play a certain role.

The SMPP protocol is a binary protocol on TCP or X.25. More information is available at Wikipedia. The official website http://www.smsforum.com is abandoned and doesn't contain any useful information any more.

There is a lot of information about the Nokia proprietary CIMD protocol for their Artus SMSC. The protocol specification and a FAQ are available at the Nokia website.

SIP / IN Interworking - RFC3976 @ ietf.org

2006-01-13T20:00:00.000+01:00

The page @ CS deparment of the Columbia University lists some very interesting drafts which explain SIP / PSTN interworking on ISUP and even INAP level. Some of these drafts might be interesting reading in the future. Unfortunately, not all links are functional any more.

Still, I found the following RFC3976 looking through the IETF site ... Interworking SIP and Intelligent Network (IN) Applications, Gurbani, Haerens, Rastogi, January 2005 (RFC3976). That might be interesting!

Impressive blog on IMS technologies sighted

2005-12-01T15:01:00.000+01:00

Looking for further information on talking technologies I found a blog about IMS technologies on the web: www.ims-insider.blogspot.com. The blog is run by the editors of www.ims-insider.com.

Both are interesting sites to watch.

Great book on SS7 networks: Signalling System No. 7 (SS7/C7) by Lee Dryburgh and Jeff Hewitt

2005-11-27T12:48:00.000+01:00

Signaling System No. 7 (SS7/C7). Protocol, Architecture, and Services. A complete, practical guide to the world's most popular signaling system, including SIGTRAN, GSM-MAP, and Intelligent Networks by Lee Dryburgh and Jeff Hewitt.

The book explains the architecture in today's telecommunication networks thoroughly. The authors exlain the history of network infrastructure and why signaling networks are decoupled from the voice transport networks. They introduce protocols like MTP2, MTP3, SCCP, TCAP, INAP, AIN and explain the relations between fixed line and wireless networks.

I can highly recommend that book!

-- Michael.

Books on Telecommunication Technologies on MobileIN.com

2005-11-11T10:47:00.000+01:00

MobileIN.com offers a very comprehensive list of technologies, solutions, some descriptions, white papers and further information all around telecommunication technologies.
One of the most interesting information is their book list on various technologies.

Overview & Status of OSA API Specifications

2005-11-11T10:27:00.000+01:00

Looking for some OSA (Open Service Access) related information I came accross this overview site at ETSI.
It shows the various standards releases and the associated documents. So, having a free account at ETSI, it is easy to download the latest standards on OSA.

Looking for a quite comprehensive IN overview?

2005-10-25T20:54:00.000+02:00

Yet again, I found a quite interesting document at the International Engineering Consortioum (IEC). This time I was looking for an Intelligent Network (IN) overview - and found a 32 pages PDF document: Intelligent Network (IN).

If you ever asked yourself how the IN evolved - and especially why - you should have a look at the document. It also explains the features of the various AIN releases and the IN Call Model.

Cool stuff - and it comes for free!

The Parlay X interface specification

2005-10-25T13:13:00.000+02:00

Looking for the Parlay X interface specification I found them stored at the ETSI homepage. At ETSI you first have to register prior to access the information. That's odd! I found the same information on Parlay X interfaces at the Parlay.org homepage!

Don't know anything about JAIN SLEE Resource Adaptors?

2005-10-14T11:41:00.000+02:00

It took me some while - however it is finished:
The Quick-Starter Guide to JAIN SLEE (JSLEE) Resource Adaptors.

The 10-pager describes JSLEE, the role and principles of Resource Adaptors in JSLEE and how to write code which follows the standards and is understandable easy. Part of the Resource Adaptor is a pretty simple TCP/IP based protocol invented to make the principles and concepts described in the tutorial even better and easier to understand.

The guide is not only paperware - no, the Resource Adapter Sources and the described Service's Sources are shipped as well - for free!

Even better, the guide made it into the Mobicents project and could be checked out from the CVS repository as part of the "mobicents-examples" module.

Check it out!

Overview of GSM/GPRS/UMTS network elements needed?

2005-10-04T15:43:00.000+02:00

A recent visit of Tektronix resulted in handing over a poster of network elements in the GSM / GPRS / UMTS network. The poster is impressive and shows all elements with their interfaces. Furthermore, the acronyms are resolved and the long version of the element's name is shown.

If you're looking for a clear overview of network elements in mobile operator's networks, have a look at the poster ... It's free for download!

MULE - Open Source ESB messaging framework ... The solution for integration problems?

2005-10-04T15:08:00.000+02:00

MULE looks at least interesting. The fact and feature list is impressively long. Especially, JBI and BPEL support added in Mule 1.1 is very impressive.

It looks like Mule could be an interesting approach to integrate J2EE environments with JSLEE environments ... and perhaps adding some more environments like TX monitors or workflow engines.

Need to think about this!

SEDA - Staged Event Driven Architecture

2005-10-04T15:00:00.000+02:00

Surfing the web and looking for ESB related projects, I struggled with the acronym SEDA. Hmmm. Googling the web I found the website of Mark Welsh at the Harvard University, Cambridge. Currently Assistant Professor of Computer Science at the Harvard University he did his Ph.D. Thesis on SEDA when being with the University of Berkeley, CA.

What is SEDA? SEDA is an acronym for staged event-driven architecture, and decomposes a complex, event-driven application into a set of stages connected by queues. This design avoids the high overhead associated with thread-based concurrency models, and decouples event and thread scheduling from application logic. By performing admission control on each event queue, the service can be well-conditioned to load, preventing resources from being overcommitted when demand exceeds service capacity. SEDA employs dynamic control to automatically tune runtime parameters (such as the scheduling parameters of each stage), as well as to manage load, for example, by performing adaptive load shedding. Decomposing services into a set of stages also enables modularity and code reuse, as well as the development of debugging tools for complex event-driven applications.

SEDA is the fundament of several open source and commercial systems.

The Online Education of IEC offers a variety of communication related documents for free

2005-09-22T13:47:00.000+02:00

The International Engineering Consortium offers a huge set of documents provided by various companies on various topics around communication.

The Online Education of IEC addresses topics like Broadband Access, Business and Strategy Issues, Fixed Wireless, Mobile Wireless, Network Components and Design, Packet Networking, Services and Applications and some more.

To access the documents free registration is required - however it looks like it is definitely worth registering ;)

JAIN SLEE Introduction in Java Spektrum 05/2005

2005-09-21T13:45:00.000+02:00

The latest "Java Spektrum" magazine (German) published an introductory article on JAIN SLEE technology. The (german) article is freely available as PDF document:
"Ereignisorientierte Komponenten mit JAIN SLEE - Wenn sich Ereignisse überschlagen".

German readers wanting to learn more about this technology should definitely have a look ;)

People with less German language knowledge should definitely have a look at:
* JAIN SLEE Technology Overview
* JAINSLEE.org
* and Mobicents - the first open source implementation of JAIN SLEE v1.0

Ah, the founder of Mobicents runs a Blog as well --- ivelin ivanov - blog

P2P SIP website ...

2005-09-21T13:43:00.000+02:00

One of the authors of the SOSIMPLE paper runs a website dedicated to Peer-to-Peer networking on the fundament of the SIP protocol: http://www.p2psip.org

Another P2P SIP network approach from K. Singh and H. Schulzrinne

2005-09-21T13:28:00.000+02:00

Some time ago, when looking for a comprehensive protocol analysis of the Skype communication protocols I found a research paper from the Department of Computer Science at Columbia University, New York. The paper "An Analysis of the Skype Peer-to-Peer Internet Telephone Protocol" was published by Salman A. Baset and Henning Schulzrinne. I found the name Schulzrinne associated to SIP protocols and thought "Why is this guy examining proprietary protocols ...????".

I think I found the answer when reading "Peer-to-Peer Internet Telephony using SIP" from Kundan Singh and Henning Schulzrinne. From my understanding, this paper addresses exactly the same problem domain as the SOSIMPLE implementation with some different solution approaches. Most appealing is the possibility to connect also to dedicated SIP proxy/registrars. So, a dual-mode operation in P2P node and "static" binding is possible. That opens huge possibilities. The P2P network can co-exist to various commercial SIP gateway providers - and help to establish VoIP communication networks.

Unfortunately, as in the SOSIMPLE case, there is no implementation available. Hopefully, these approaches don't end on shelves ...

SOSIMPLE - challenging established proprietary P2P communication networks?

2005-09-20T14:08:00.000+02:00

Today, on a web search for "Peer-to-peer communication networks" - inspired by the topic "Skype" I found a very interesting paper from David A. Bryan and Bruce B. Lowekamp from The Department of Computer Science at The College of William & Mary.

The paper named "SOSIMPLE: A Serverless, Standards-based, P2P SIP Communication System" shows in an impressive manner how to implement P2P communication neetworks without centralized servers (!) in a standards-oriented way. Utilizing the SIP protocol not only for session signalling on user level but also for signalling purpose in the P2P overlay network allows their approach to inter-operate with existing SIP infrastructure. Various SIP implementations are available for free (think of the excellent NIST-SIP stack in Java) or even whole applications based on SIP are available. The SOSIMPLE approach has the potential to challenge the success of the very proprietary communicatonn solution from Skype networks.

So far, the SOSIMPLE approach is a study paper from the Computer Science Department at the College of William and Mary Williamsburg and Cisco Systems Inc. and they claim to have a prototype implementation available. Let's hope there will be a project homepage very soon and the implementation becomes one of the most famous open source projects ...

-- Michael