Some of you that have been following along closely, may already have noticed our involvement with Tropo.com. We worked closely with Voxeo at the inception of Tropo which was subsequently launched at eComm in March of 2009. The goal of Tropo.com is to bring a scalable telephony cloud infrastructure to the broadest group of developers by allowing development directly in dynamic languages. Languages included are: Javascript, Groovy, PHP, Python and Ruby. We will now be focused on bringing Tropo and Adhearsion together to offer the power of Adhearsion with the scale of Tropo, while extending capabilities in the other languages.

Jay and I as founding members of Voxeo Labs is no accident. The DNA of Voxeo Labs is a commitment to open-source. Adhearsion will continue to evolve as an open-source framework with continued support for Asterisk along with Tropo and others. Our goal is to provide the widest possible set of deployment choices for the Adhearsion community, while eliminating friction for those who want effortless deployment and scaling.

Stay tuned as we have lots of interesting announcements just around the corner. In the meantime, thanks to the entire Adhearsion community for the support that allowed us to pursue this next step.

Here are the slides:

While attending Euruko several weeks ago, I had the opportunity to see @aslak_hellesoy present the Cucumber Behavior Driven Development (BDD) framework. BDD is a part of Agile software development focused on bringing the domain expert into the specification process through the use of natural language. This is done through the users and developers creating a story that is directly parsed by the code for tests.

Aslak’s presentation was well done, giving some compelling reasons to adhere to BDD principals, showing that using Cucumber made it a joy. Thing is, Cucumber is already a part of testing for  the Adhearsion framework and some specific projects we are working on. What was new to me, was a lightning talk given at the end of the day on Cucumber-Nagios.

Cucumber-Nagios is a project developed by @auxesis that bridges Cucumber test results into the Nagios monitoring tool. This allows you to take any test you have written and allow the steps to become part of an on-going systems monitoring strategy. While Cucumber-Nagios is geared towards web application testing, you may use this beyond to test virtually any application.

Naturally I have started using Cucumber-Nagios to create a series of tests to do application level monitoring of Adhearsion and Asterisk. This now allows one to not only ensure that these processes are running, but they are working as defined by the BDD specification.

To start, I created a couple of features that will connect to Adhearsion and/or Asterisk. Then ping the Asterisk Manager Interface (AMI) looking for a corresponding pong, ensuring the systems are still responsive.

The feature to ping the AMI uses the straight forward Gherkin language to create a feature file with the test specification:

The feature file is then parsed by the steps definitions as set out in the step file:

The methods called in the step definitions then invoke the system testing library I wrote specifically for these telephony systems. The result is having a standalone test framework that may provide direct input at an application level to Nagios, as well as provide plain English feedback to anyone looking to track down the cause of the problem.

First, if Asterisk is down we may see the result of running the Cucumber plain English test:

Then the corresponding Cucumber-Nagios test that also fails with a ‘Critical’ error:

Now, if Asterisk is up and all is fine we may see these same tests pass. First, for Cucumber all is green:

Then for Cucumber-Nagios we see all 3 tests passed:

The great thing about this approach is that it may be used to test systems using a standard open source dynamic language like Ruby. These tests are not limited to testing Ruby systems though, as you may see in this case I am testing the Asterisk open source telephony engine. Further, I may test not only the standard interfaces of various systems, but that the behavior of my application deployed within these systems is behaving as required from a specification perspective. This is a powerful approach to systems monitoring and troubleshooting, pulling the system administrators directly into the development process.

You may check out my telephony-system-tests project on Github. My plan is to greatly extend this system testing framework over time in order to have a stable of standard tests for Adhearsion, Asterisk, Freeswitch and others in the future. One of the next steps will be to integrate SIPr into the framework for testing SIP dialogs as well.

Another interesting item from Euruko that I will be posting about soon is Chef, which was presented by Joshua Sierles of 37Signals at the conference. So much more to come on this front.

“…I needed to leverage ‘all the work’ done in .Net…As I discovered, the marriage of .Net and Adhearsion was lovely, thanks to the RESTful RPC [API of Adhearsion]“

So David created an example and has graciously shared it with the rest of us. The example includes a component in Adhearsion that has this method extending the ‘methods_for :rpc’:

def launch_call_rpc(vars)
  ahn_log.ami vars   

  channel = vars["src"]
  exten = vars["dest"]
  options = { "Channel" => channel,
              "Context" =>  "callback",
              "Exten" =>    exten,
              "Priority" => “1”,
              "Callerid" => “33333” }
  result = Adhearsion::VoIP::Asterisk.manager_interface.originate options

  ahn_log.ami "status: call Queued"
  ahn_log.ami result
end

Following the full README and example here, this C# code may then be used to invoke the Adhearsion REST API:

using System;
using System.Collections.Generic;
using System.Linq;
using System.Text;
using JSONSharp;
using System.Net;
using System.IO;

namespace adhearsion
{//adhearsion server is http://192.168.1.62:5000
    class Program
    {
        static void Main(string[] args)
        {
            Program.connect();
        }
        static void connect()
        {
            try
            {
                ahnparams ahn = new ahnparams();
                ahn.src = @"SIP/4031234567@192.168.1.200";
                ahn.dest = @"7801234567";
                //Pass it to our static reflector, which will build
                JSONReflector jsonReflector = new JSONReflector(ahn);   //JSONSharp converts the class to JSON format
                //Console.WriteLine(jsonReflector.ToString());
                NetworkCredential myCred = new NetworkCredential("jicksta", "roflcopterz");

                CredentialCache myCache = new CredentialCache();

                myCache.Add(new Uri("http://192.168.1.62:5000"), "Basic", myCred);

                WebRequest request = WebRequest.Create("http://192.168.1.62:5000/launch_call_rpc");
                request.Credentials = myCache;
                string postData = "["+jsonReflector.ToString()+"]";  //had to add the [] for it to work with restful_rpc just like php json
                Console.WriteLine(postData);
                request.Method = "POST";
                byte[] byteArray = Encoding.UTF8.GetBytes(postData);
                // Set the ContentType property of the WebRequest.
                request.ContentType = "application/x-www-form-urlencoded";
                // Set the ContentLength property of the WebRequest.
                request.ContentLength = byteArray.Length;
                // Get the request stream.
                Stream dataStream = request.GetRequestStream();
                // Write the data to the request stream.
                dataStream.Write(byteArray, 0, byteArray.Length);
                // Close the Stream object.
                dataStream.Close();

                WebResponse response = request.GetResponse();
                // Display the status.
                Console.WriteLine(((HttpWebResponse)response).StatusDescription);
                // Get the stream containing content returned by the server.
                dataStream = response.GetResponseStream();
                // Open the stream using a StreamReader for easy access.
                StreamReader reader = new StreamReader(dataStream);
                // Read the content.
                string responseFromServer = reader.ReadToEnd();
                // Display the content.
                Console.WriteLine(responseFromServer);
                // Clean up the streams.
                reader.Close();
                dataStream.Close();
                response.Close();
            }
            catch (Exception e)
                {
                    throw e;
                }

        }
    }
}

This is a great example of the cross-platform capabilities of Adhearsion that you get for free via REST. You do not have to be a Ruby-ist to take advantage of adding voice capabilities to your web applications. You may use virtually any language and platform that may consume web services.

We definitely appreciate it when our community takes the time to share their Adhearsion experiences with the rest of us. Thanks David, and for the rest of you the full example with his README is available here.

Of course you could setup routing rules in your phone system each time you move to a new timezone, but good luck remembering to do this. Tripit now makes a blog badge available that reports your current location based on city, state or country:

My Tripit Adhearsion Comonponent on Github allows you to mashup the details from your badge and use them in your dialplan. Now, you may add this component to your project and add something like this to your Adhearsion dialplan:

adhearsion {
  user_time = tripit_user_time?('jdoe')
  if user_time
    ahn_log.tripit.debug user_time
  else
    ahn_log.tripit.debug 'Error'
  end
}

You may then use this information to decide whether to dial your mobile at your destination, or send it straight to voicemail. This is a great example of using Adhearsion to mashup data available on the web and use it to control your communications.

The first step was to use an XLite Softphone to connect to the Skype SIP proxy. Then I graduated to connecting an Asterisk server. While there were some initial issues, Skype worked with the beta testers and had all of the major elements working by the end of the first day. I am impressed by how few issues there actually are, but I guess that is to be expected since this service leverages the same infrastructure behind SkypeOut and SkypeIn. Those services have been around for years and make up some of Skype’s core revenue generating business.

Here was what I was able to do so far:

• Asterisk/XLite -> SIP -> SkypeOut -> PSTN (w/G711 codec)
• PSTN -> SkypeIn -> SIP -> Asterisk/XLite (w/G729 codec)
• Skype User -> SIP -> Asterisk/XLite

(Note: What I will not be able to try is ‘Asterisk/Xlite -> SIP -> Skype User’ since this is intentionally blocked by Skype. I presume this is to protect their SkypeIn business by blocking the ability to create a competing alternative.)

If you have used the SkypeIn/SkypeOut services before, then you already know the quality of the calling. So far I have not had any issues with the call quality or dropped calls once the kinks were worked out.

While Skype is supporting the freely available G711 codec via its SIP gateways, it is dependent on what codecs are supported by the carrier that is being used to terminate a particular call. So while on SkypeOut I was able to use the G711 codec to terminate to numbers in the San Francisco Bay Area (415/650), on SkypeIn to a San Jose number (408) the only available codec was g729. So the reality is, you will need to have a SIP endpoint that supports the licensed G729 codec for reliable use.

The current calls are not encrypted. Skype has already stated they intend to support TLS/SRTP in the near future as encryption is considered a core feature for SIP as much as for their P2P calls.

As time permits I will be continuing to do more tests as well as contrasting with the Skype for Asterisk (SfA) beta software. Skype is definitely on the right track by opening their network to key standards with multiple interfacing options. I believe Skype is now poised to become a key global player in the business VoIP market, as well as bringing in a broader range of developers.

I began to think this day would never come, but it has…

To this end we have now migrated the Adhearsion website, wiki and API docs to Amazon EC2. Leveraging S3 and EBS we have a great solution that allows us to quickly fire up more servers if and when needed. The cloud is great.

We have also moved the Sandbox to a new hosted server. Our friends at Voxeo have provided a great server for us and are now sponsoring the Adhearsion Sandbox. It is great to have a visionary leader in the telephony cloud space involved with the Adhearsion community.

I would like to take this opportunity to remind folks what the Adhearsion Sandbox provides. We have provided a hosted system that takes away the need to install your own Asterisk to get started developing Adhearsion applications. Simply install Adhearsion, sign-up for a Sandbox account and get started writing apps. Our goal is to lower all the barriers to make it easy for developers to realize the possibilities of voice in modern web development.

Enjoy the new servers!

I used a combination of Rack and Passenger, since they make the process so easy and production ready. To this end I have updated the README and Click To Call component I covered in my previous blog post to include the how to and Rack configuration file required.

To get this up and running do the following:

• Install the Apache Webserver
• sudo gem install rack
• sudo gem install passenger (follow instructions here for compiling the Apache module)

Copy the files and directories in ~ahn-project/components/restful_clicktocall/web directory to the appropriate file system location for your Apache web server. Add these virtual host settings to your Apache configuration (ie – /etc/httpd/conf/httpd.conf):

<VirtualHost *:80>
  ServerName ec2-174-129-89-20.compute-1.amazonaws.com
  DocumentRoot /var/www/restful_clicktocall_web/public
</VirtualHost>

Change line three to reflect your configuration. Restart your Apache web server and then connect to the site you configured. For more options I recommend referring to the Passenger documentation found here.

Stay tuned for the upcoming Amazon Machine Image!

As of now Skype publishes a web page of calling rates based on the country name and the per minute rate including or excluding the tax. A few additional items are needed to make this usable for LCR systems:

• The associated country code for each country (ie – ’34′ for Spain, ’1′ for the US, etc)
• More granular prefixes where calling rates may differ (ie – ’346′ for Spanish mobiles, ’336′ for Frech mobiles, ’1212′ for NYC, ’1712′ for Iowa, etc)
• Billing intervals
• A file download in CSV, or similar format, for import into LCR systems

Of course, in the meantime it is easy enough to scrape the website and convert the available data into a more appropriate format. Here is an example, in Ruby, of how this may be done in a trivial way:

require 'rubygems'
require 'open-uri'
require 'nokogiri'
require 'json'

skype_rates = Hash.new
skype_url = 'http://www.skype.com/prices/callrates/#allRatesTab'
skype_htmldoc = Nokogiri::Hpricot(open(skype_url).read)
(skype_htmldoc/'table.listing//tr.r1').each do |country|
  country_name = country.at('td').inner_html
  skype_rates.merge!({ country_name => { 'amount' => country.at('span.amount').inner_html.split('<!')[0].gsub('$ ', '').to_f,
                                         'vat' => country.at('span.vat').inner_html.split('<!')[0].gsub('$ ', '').to_f } })
end

p skype_rates.to_json

Which produces JSON output as follows:

{
  "Bolivia-La Paz": {
     "amount":0.122,
     "vat":0.14
  },
  "Sweden - Mobile": {
     "amount":0.292,
     "vat":0.336
  },
  "Hong Kong": {
     "amount":0.021,
     "vat":0.024
  }
}

You may then perform a Regular Expression against another data source to derive the appropriate country codes/prefixes and store those in your LCR system. A good example of the additional detail needed is provided by Flowroute.

I have on my list of actions to create an Adhearsion component to provide LCR capabilities for any Adhearsion application. The plan is to support a wide number of VoIP providers and other data inputs as a part of this plug-in.

In the meantime, it will be interesting to see how Skype goes about publishing their rates with additional details and formats for download.

UPDATE @JimCanuck points out it is not just about least cost, but also about quality of termination. Skype has some interesting approaches to call quality. More here.

The first application I participated in writing was a Computer Telephony Integration system using COBOL on HP3000s. This system is still alive and well today and now may be adapted to take advantage of all the clouds have to offer. Considering the codebase has not been touched since 1999, and then only for refactoring, this is a tremendous leap forward.

Here is an example to get you started, I will be posting to GitHub shortly:

      $ SET SOURCEFORMAT"FREE"
IDENTIFICATION DIVISION.
PROGRAM-ID.  CallingGrace.
AUTHOR.  Jason Goecke.
<ul>
  <li>Demonstrates the use of Tropo.com</li>
<li>for writing voice applications from</li>
<li>all the world's systems that use</li>
<li>COBOL in the clouds.</li>
</ul>

DATA DIVISION.
WORKING-STORAGE SECTION.
01 MenuStr             PIC X(99)      VALUE "Please tell us what you would like to know about Grace Hopper.".
01 RetryCntr           PIC 9          VALUE ZEROS.
01 MenuOptions         PIC X(99)      VALUE "war('Where did Grace work?'), bug('Where is that moth?')".
01 MenuResult          PIC X(99)      VALUE "".
01 SayGoodbye          PIC X(99)      VALUE "Thank you for calling, goodbye.".
01 PleaseTryAgain      PIC X(99)      VALUE "Invalid choice, please try again."
01 WarDeptPhone	       PIC 9999999999 VALUE 7035456700.
01 SmithsonianBugs     PIC 9999999999 VALUE 2026331000.

LINKAGE SECTION
01 TROPO-PROCEDURES

PROCEDURE DIVISION USING TROPO-PROCEDURES.
000-MAINLINE.
    CALL 'ANSWER'.
    PERFORM UNTIL RetryCntr = 3
       MenuResult = CALL 'ASK' USING MenuStr
                                     MenuOptions.
       IF MenuResult ALPHABETIC
         PERFORM 200-CALL-DESTINATION.
		 RetryCntr = 3.
       ELSE
         CALL 'SAY' USING PleaseTryAgain.
         ADD 1 TO RetryCntr.
    END-PERFORM.

    GO TO 300-EXIT.

200-CALL-DESTINATION.
    IF MenuResult EQUAL 'war'
      CALL 'TRANSFER' USING WarDeptPhone.

	IF MenuResult EQUAL 'bug'
	  CALL 'TRANSFER' USING .

300-EXIT.
	CALL 'SAY' USING SayGoodbye.
	CALL 'HANGUP'.
	STOP RUN.

It is a fact, there are more lines of COBOL in use in the world today than any other single computer programming language. I applaud the visionary leadership behind Tropo.com siezing this giant opportunity to bring voice from the clouds to all of those systems out there.