Showing posts with label IBM. Show all posts
Showing posts with label IBM. Show all posts

Tuesday, 7 March 2023

Deleting all documents from a Watson Discovery collection


Sometimes when you are cleaning up a collection, you might want to delete all of the documents. It is tedious to do this by clicking on the files in the UI. Therefore I wrote a simple script del_docs.ipynb to do this for you.

To run this script, edit the panel containing the collection details and then jus run the notebook.

By default the script will delete all documents, if you want to only delete a subset of documents then alter the query variable to specify the selection formula for the documents you want to delete.

Friday, 14 September 2018

Connecting IBM Watson Speech services to the public phone system

Many use cases for IBM Watson speech services involve connecting phone calls. This can be tricky so I decided that it might be useful to publish a sample which shows such a connection in action. This application is very simple, it uses Speech to Text (STT) to understand what the caller says and when the caller pauses it uses Text to Speech (TTS) to read it back to them. This simple application can easily be used as a starting point for building a more complex application which does more with the received speech.

Flow Diagram


I chose to use the NEXMO service because it is the easiest way to connect a phone call to a websocket. You can visit their documentation site if you want to learn details of how this works. The short summary is that it acts as a broker between the phone system and a web application of your choice. You need to provide two URLs that define the interface. Firstly nexmo will do a GET on the '/answer" URL  every time a call comes in to the number you configure - the way your application handles this request is the key part of the application experience. Secondly nexmo will do a POST to your '/events' URL anytime anything happens on your number (e.g. a call comes in or the person hangs up) in our case we don't do anything interesting with these except to write them to the log for debugging purposes.

To get this working for yourself, the first thing you need to do is deploy my sample application. You can get the code from this GIT repository. Before you deploy it to IBM Bluemix, you need to edit the manifest.yml file and choose a unique URL for your instance of the application. You also need to create instances of the IBM Watson STT and TTS services and bind them to your application.

Next you need to configure nexmo to connect to your application. You need to login to the nexmo website and then click on the 'Voice' menu on the left side and then 'Create Application. This pops up a form where you can enter details of the /event and /answer URLs for the web application you just deployed. After you fill in this form, you will get an nexmo application id.


Unfortunately connecting to the phone system costs money. Nexmo charges different amounts of money for numbers depending upon what country they are associated with. In my case I bought the number +35315134721 which is a local number in Dublin, Ireland. This costs me €2.50 per month so I might not leave it live too long or maybe swap for a US based number at a reduced cost of US$0.67 per month.

Once you get out your credit card and buy a number, you must tell nexmo which application id you want to associate with the number. Visit the your numbers page and enter the details (like you see below).



Having done this, you can now ring the number and see it in action. When we receive a call, we open a websocket interface to the STT service and start echoing all audio received from the phone line to the STT service. Although the TTS service supports a websocket interfacer, we don't use the websocket interface because the chunks of audio data from the TTS service won't necessarily be returned evenly spaced the nexmo service will produce crackly audio output. Instead we use the REST interface and we write the returned audio into a temporary file before streaming it back into the phone call as a smooth rate.

The bulk of the code is in a file named index.js and it is fairly well explained in the comments, but here are a few more explanatory notes:

  • The first 70 lines or so are boilerplate code which should be familiar to anyone who has experience of deploying node.js application to BlueMix. First we import the required libraries that we use and then we try and figure out the details of the Watson service instances that we are using. If running on the cloud, this will be parsed from the environment variables. However, if you want to run it locally, you will need a file named vcap-local.json that contains the same information. I have included a file named vcap-local-sample.json in the repository to show you the required structure of the file.
  • Next comes a function named tts_stream which acts as an interface to the TTS service. It takes two parameters, the text to synthesise and the socket on which to play the result. We use the REST interface instead of opening a websocket to the TTS service (like we do with the STT service). The reason for this choice is that it results in crackly audio as the audio chunks coming back from the TTS service are not evenly spaced. The way it works is that it saves the audio to a temporary file and then pipes the file smoothly into the nexmo socket before deleting the temporary file. This approach introduces a slight delay because we need to wait for the entire response to be synthesised before we start playing. However, the problem is not as bad as you might think because a 15 second audio response might get sent back in under a second.
  • Next comes the two functions which respond to the /events and /answer URLS. As mentioned earlier the  /event  handler is very simple because it just echos the POST data into the log.  The /answer function is surprisingly simple also. Firstly it creates a websocket and then it sends a specially formatted message back to nexmo to tell it you want to connect the incoming phone call into the new websocket.
  • The real meat of the code is in the on connect method which we associate with the websocket that we created.
    • The first thing we do is stream audio from a file named greeting.wav which explains to the user what to do. While this message is helpful to the user, it also gives the application some breathing room because it might take some time to initialise the various services and the greeting will stop the user talking before we are ready.
    • Next we create a websocket stt_ws which is connected to the Watson STT service. 
      • As soon as the connection is established, we send a special JSON message to the service to let it know what type of audio we will send and what features we want to enable.
      • When the connection to STT is started, a special first message is sent back saying that it is ready to receive audio. We use a boolean variable stt_connected to record whether or not this message is received. This is because attempting to send audio data to the websocket before it is ready will cause errors.
      • When starting the STT service, we specify that we would like to receive interim results i.e. when it is transcribing some audio and it thinks it knows what was said, but it does not yet consider the results to be final (because it might change its mind when it hears next).  We do this because we want to speed up responses, but we don't want to echo back a transcription which might later be revised. For this reason we check the value of the final variable in the returned JSON and only call the tts_stream function when the results are final.
    • For the nexmo websocket, we simply say that all input received should automatically be echoed to the STT websocket (once we have received confirmation that the STT service link has been initialised properly.
    • When the nexmo websocket closes we also try to close the STT web socket
To give credit, I should point out that my starting point was this sample from nexmo. I should also point out that the code is currently only able to deal with only one call at a time. It should be possible to solve this problem, but I will leave this as a learning exercise for some reader of the blog.

Friday, 7 September 2018

Watson Speech to Text supports language and acoustic customizations, but do you need both?

The Watson Speech to Text service has recently released a feature whereby customers can customize the service so that it works better for their specific domain. These customizations can be in the form of a Customized Language Model and/or an Customized Acoustic Model. Customers sometimes get confused by these two different model types and they wonder which they need or if they need both.

The quick summary is that the language customization model tells Watson how the words spoken in your domain are different from normal English (or whatever other base language you are using). For example you might be transcribing speech where the speakers use a specialised terminology. On the other hand, the acoustic customization model tells Watson that the words spoken in your domain might be spoken quite differently than they were spoken in the corpus initially used to train Watson STT. For example, you might have audio samples where the users are using a strong regional accent.

Depending upon your domain, you may need both types of customization, but lets look at them in more detail first.

Customized Language Models

Customized language models tell Watson what words are likely to occur in your domain. For example, when you specify that you wand to use the en-US language, Watson will have a very large (but fixed) list of possible words that can be spoken in US English. However, in your domain, the users might use a specialised vocabulary.  The purpose of the customized language model is to teach Watson how the language in your domain is different from normal English.

The way you build a customized language model is that you provide one or more corpora which are simple text files containing a single utterance per line. It is important to give complete utterances, because accurate speech to text transcription requires that the service knows not only what words might be seen but also in what context the words are likely to occur.

If you are building a model to be used in transcribing film reviews, your corpus might include words like movie release star and blockbuster. These words are already in the Watson standard dictionary, but including them in your model tells Watson that these words are more likely to occur in your domain than normal (which increases the chance that they will be recognised).

  • You might also include the word umpa lumpas in your corpus since people will be discussing them and you need to tell Watson it is a valid word. Since this word is pronounced like it is written, all you need to do is tell Watson that it is a valid word. 
  • However, if you are interested in Irish movies, it is likely that people will speak about an actress named Ailbhe or Caoimhe. These common Irish forenames wouldn't be in the Watson dictionary, but it is not enough to tell Watson that they exist. You also need to tell Watson that Ailbhe is pronounced like Alva and Caoimhe is pronounced like Keeva.

The building of this customized language model is probably a relatively simple job. Nevertheless, this customization will probably bring about a dramatic reduction in word error rate. If your audio contains examples of people saying words not in the standard Watson dictionary, then you will never transcribe these properly without a customized language model. In addition, when your speakers say words in the Watson dictionary, the language customization model will increase the chances of these being properly transcribed.

Many users find that the language customization by itself will meet their needs and there is not necessarily any need to combine it with an acoustic model.

Customized Acoustic Models

Customized acoustic models allow you to tell Watson what words sound like in your domain. For example, if speakers in your  region consistently pronounce the word there as if they are saying dare you might need need to build a customized acoustic model to account for this.

At one level building a customized acoustic model is even easier than building a customized language model. All you need to do is upload between 10 minutes and 50 hours of sample audio which is typical of the type of speech that you will be  trying to transcribe with the model that you are building. And then you train the model.

However, if you read the documentation carefully you will see that they say "you will get especially good results if you train with a language model built from a transcription of the audio".  Transcribing 50 hours of speech is a lot of work and so many people ignore this advice. However, I think the advice should read "you are extremely unlikely to get good results unless you train with a language model built from a transcription of the audio". In my experience, training without a language model containing the transcription can very often produce a model whose word error rate (WER) is significantly worse than having no model at all.

To understand why this is the case, you need to look a little closer at how acoustic model training works. For illustration purposes, assume that the problem you are trying to solve is that the utterance a-b-c is sometimes being erroneously transcribed at x-y-z.

  • If you train with a language model, Watson will encounter an ambiguous utterance in your training data which it thinks is 70% likely to be x-y-z and 55% likely to be a-b-c. Since your language model doesn't contain x-y-z it will know that it must be a-b-c and it will make adjustments to the neural network to make it more likely that this utterance will be transcribed as a-b-c in the future. Hence, the model gets better.
  • On the other hand, if you train without a language model. Watson will encounter an ambiguous utterance in your training data which it thinks is 70% likely to be x-y-z and 55% likely to be a-b-c. Since it has no other information will assume that it must be x-y-z. However, the confidence score is not very high so it will make adjustments to the neural network to make it even more likely that this utterance will be transcribed as x-y-z in the future. Hence, the model gets worse.

Of course the chance of such an error happening is related to the word error rate. In my experience, users rarely put the effort into building a customized model when the WER is low. Mostly people build customized models when they are seeing very hight WER and hence they often see carelessly built acoustic models making the problem even worse.

Another problem people encounter is building an acoustic model from speech which is not typical of their domain. For example, user might be tempted to get a single actor to read out their entire script and get then to record the samples in a recording studio with a good microphone. When their application goes live they might have to deal with audio recorded on poor phone lines, in a noisy environment by people with different regional accents.

Summary

Language and acoustic customizations serve a different purpose - the first deals with non-standard vocabulary while the other deals with non-standard speech sounds. It is possible that you can build a customized language model very easily and this may be enough for your domain. An acoustic model can improve your WER even further, but you should be careful to ensure you build a good one. In particular you should use transcribed data rather than just collecting random samples.

Wednesday, 20 September 2017

Adding a speech interface to the Watson Conversation Service

The IBM Watson Conversation Service does a great job of providing an interface that closely resembles a conversation with a real human being. However, with the advent of products like the Amazon Echo, Microsoft Cortana and the Google Home, people increasingly prefer to interact with services by speaking rather than typing. Luckily IBM Watson also has Text to Speech and Speech to Text services. In this post we show how to hook these services together to provide a unified speech interface to Watson's capabilities.

In this blog we will build upon the existing SpeechToSpeech sample which takes text spoken in one language and then leverages Watson's machine translation service to speak it back to you in another language. You can try the application described here on Bluemix or access the code on GitHub to see how you can customise the code and/or deploy on your own server.

This application has only one page and it is quite simple from the user's point of view.
  • At the top there is some header text introducing the sample and telling users how to use it. 
  • The sample uses some browser audio interfaces that are only available in recent browser versions. If we detect that these features are not present we put up a message telling the user that they need to choose a more modern browser. Hopefully you won't ever see this message.
  • In the original sample there are two drop down selection boxes which allow you to specify the source and target language. We removed these drop downs since they are not relevant to our modified use case.
  • The next block of the UI gives the user a number of different ways to enter speech samples:
    • There is a button   which allows you to start capturing audio directly from the microphone. Whatever you say will be buffered and then passed directly to the transcription service. While capturing audio, the button changes colour to red and the icon changes  - this is a visual indication that recording is in progress. When you are finished talking, click the button again to stop audio capture.
    • If are working in a noisy environment or if you don't have a good quality microphone, it might be difficult for you to speak clearly to Watson. To help solve this problem we have provided you with some ample files hosted in the web app. To play one of these samples click on one of the buttons to play the associated file and use it as input.
    • If you have your own recording that you can click on the  button and select the file containing the audio input that you want to send to the speech-to-text service.
    • Last, but not least, you can drag and drop an audio file onto the page to have it instantly uploaded
  • The transcribed text is displayed on an input box (so you can see if Watson is hearing properly) and sent to either the translation service (in the original version) or the conversation service in our updated service. If there is a problem with the way your voice is being transcribed, see this previous article on how to improve it.
  • When we get a response from the conversation or translation service we place the received text on an output text box and we also call the text-to-speech service to read out the response and save you the bother of having to read.
I know that you want to understand what is going on under the covers so here is a brief overview:
  • The app.js file is the core of the web application. It implements the connections between the front end code that runs in the browser and the various Watson services. This involves establishing 3 back-end REST services. This indirection is needed because you don't want to include your service credentials in the code sent to the browser and because your browser's cross site script protections will prohibit you from making a direct call to the Watson service from your browser. The services are
    • /message - this REST service implements the interface to the Watson Conversation service. Every time we have a text utterance transcribed, we do a POST on this URL with a JSON payload like {"context":{...},"input":{"text":"<transcribed_text>"}}. The first time we call the service we specify an empty context {} and in each subsequent call we supply the context object that the server sent back to us the last time. This allows the server to keep track of the state of the conversation.
      Most conversation flows are programmed to give a trite greeting in response to the first message. To avoid spending time on this the client code sends initial blank message when the page loads to get this out of the way.
    • /synthesize - this REST service use used to convert the response into audio. All that this service does to convert a get on http://localhosts:3000/synthesize?voice=en-US_MichaelVoice&text=Some%20responsevoice=en-US_MichaelVoice&text=Some%20response into a get on the URL  https://watson-api-explorer.mybluemix.net/text-to-speech/api/v1/synthesize?accept=audio%2Fwav&voice=en-US_MichaelVoice&text=Some%20response this will return a .wav file with the text "some response" being spoken in US English by the voice "Michael". 
    • /token - the speech to text transcription is an exception to the normal rule that your browser shouldn't connect directly to the Watson service. For performance reasons we chose to use the websocket interface to the speech to text service. At page load time, the browser will do a GET on this /token REST service and it will respond with a token code that can then be included in the URL used to open the websocket. After this, all sound information captured from the microphone (or read from a sample file) is sent via the websocket directly from the browser to the Watson speech to text service.
  • The index.html file is the UI that the user sees. 
    • As well as defining the main UI elements which appear on the page, it also  includes main.js which is the client side code that handles all interaction in your browser.
    • It also includes the JQuery and Bootstrap modules. But I won't cover these in detail.
  • You might want to have a closer look at the client side code which is contained in a file public/js/main.js:
    • The first 260 lines of code are concerned with how to capture audio from the client's microphone (if the user allows it - there are tight controls on when/if browser applications are allowed to capture audio). Some of the complexity of this code is due to the different ways that different browsers deal with audio. Hopefully it will become easier in the future. 
    • Regardless of what quality audio your computer is capable of tracking, we down sample it to 16bit, mono at 16 Khz because this is what the speech recognition is expecting.
    • Next we declare which language model we want to use for speech recognition. We have hardcoded this to a model named "en-GB_BroadbandModel" which is a model tuned to work with high fidelity captures of of speakers of UK English (sadly there is no language model available for Irish English). However, we have left in a few other language models commented out to make it easy for you if you want to change to another language. Consult the Watson documentation for a full list of language models available.
    • The handleFileUpload function deals with file uploads. Either file uploads which happen as a result of explicitly clicking on the "Select File" button or upload that happen as a result of a drag-and-drop event.
    • The initSocket function manages with the interface to the websicket that we use to communicate to/from the speech_to_text service. It declares that the showResult function should be called when a response is received. Since it is not always clear when a spaker is finnished talking, the text-to-speech can return several times. As a result the msg.results[0].final variable is used to deremine if the current transcription is final. If it is an intermediate result, we just update the resultsText field with what we heard. If it is the final result, the msg.results[0].alternatives[0].transcript variable is also used as the most likely transcription of what the user said and it is passed on to the converse function.
    • The converse function handles sending the detected text to the Watson Conversation Service (WCS) via the /message REST interface which was descibed above. When the service gives a response to the question, we pass it to the text-to-speech service via the TTS function and we write it on the response textarea so it can be read as well as listened to.
  • In addition there are many other files which control the look and feel of the web page, but won't be described in detail here e.g. 
    • Style sheets in the /public/css directory
    • Audio sample files in the /public/audio directory
    •  Images in the public/images directory
    • etc.
Anyone with a knowledge of how web applications work, should be able to figure out how it works. If you have any trouble, post your question as a comment on this blog.
At the time of writing, there is an instance of this application running at https://speak-to-watson-app.au-syd.mybluemix.net/ so you can see it running even if you are having trouble with your local deployment. However, I can't guarantee that this instance will stay running due to limits on mypersonal Bluemix account.

Tuesday, 5 September 2017

Watson Speech to Text with Node.js

Talking to a computer makes you feel like your living in the future. In this post I will show how using node.js to expand on the Watson Speech to Text (STT) example to improve the accuracy of the transcription.
As a base I will take the cognitive car demo and try write a STT system for that. Once Speech to text can correctly interpret what a person asks the car we will want to send those commands to The Watson Conversation Service Car demo. But I will deal with connecting STT to WCS in another post.
Log into bluemix and set up a STT service. Create new credentials and save them as you will use them in all the following programs. This username and password are used everywhere below.
Next assuming you already have node installed you also need the watson developer cloud SDK
npm install watson-developer-cloud --save
and the speech-to-text-utils
npm install watson-speech-to-text-utils -g

Get an audio file of the sorts of things you would say to this Car demo. I asked questions like the ones below

turn on the lights
where is the nearest restaurant
wipe the windscreen
When is the next petrol station
When is the next gas station
does the next gas station have rest rooms
How far are the next rest rooms
please play pop music
play rock music
play country music


I have made an audio file of that (I had a cold). You can get it here or record your own version.
In speech_to_text.v1.js in the sdk examples file put in the username and password you got in 1. above. And point at the sound file of commands from 3.


const speech_to_text = new SpeechToTextV1({
  username: 'INSERT YOUR USERNAME FOR THE SERVICE HERE',
  password: 'INSERT YOUR PASSWORD FOR THE SERVICE HERE'
});

fs.createReadStream(__dirname + '/DavidCarDemo.wav').pipe(recognizeStream);

Now look at transcription.txt. We want to improve the accuracy of this transcription. Now create a model that we will train so that the STT understands our speech better.

Running node createmodel.js on this code gives you a custom_id needed from here on.

{ "customization_id": "06da5480-915c-11e7-bed0-ef2634fd8461" }
5. If we tell STT to take in the car demo corpus. It learns from this the sort of words to expect the user to say to the system
watson-speech-to-text-utils set-credentials
and give it the username and password same as above.
watson-speech-to-text-utils customization-add-corpus
Will ask you for a conversation workspace. Give it car_workspace.json downloaded from the car demo. Adding a conversation workspace as a corpus improves accuracy of the words that are unusually common in our conversation.
Now we want to improve accuracy on words that are not common in the corpus. For example "Does the next" is heard as 'dostinex" currently.
watson-speech-to-text-utils corpus-add-words -i 06da5480-915c-11e7-bed0-ef2634fd8461
and give words.json which looks like


{
  "words": [{
    "display_as": "could",
    "sounds_like": [ "could" ],
    "word": "culd"
  }, {
    "display_as": "closeby",
    "sounds_like": [ "closeby" ],
    "word": "closeby"
  }, {
    "display_as": "does the next",
    "sounds_like": [ "does the next", "dostinex" ],
    "word": "does_the_next"
  }
]
}
The speech utils allow you to see what words you have in your model. Words added to not overwrite old ones so sometimes you need to delete old values. watson-speech-to-text-utils customization-list-words -i 06da5480-915c-11e7-bed0-ef2634fd8461
watson-speech-to-text-utils corpus-delete-word -i 06da5480-915c-11e7-bed0-ef2634fd8461 -w "word to remove"

Finally in transcribe.js you need to tell it to use the model we have just made


const params = {
  content_type: 'audio/wav',
  customization_id: '06da5480-915c-11e7-bed0-ef2634fd8461'
};
The code I have used is up on github here.
Now we get a more accurate transcription of the voice commands. By training on the Conversation corpus. Then fixing other common errors using a words.json file.


























Thursday, 31 August 2017

Combining other services with IBM Watson Conversation Service

It is becoming increasingly popular to offer an interface to computer applications which resembles the way that we converse with a fellow human. The IBM Watson Conversation Service is an excellent way to program such an interface because it allows the developer an easy way to specify the conversation flow and is also very good at doing fuzzy matching on input text to guess what the user is really trying to find out. However, the graphical way that conversation flows are specified doesn't allow the user to make calls to external services in order to get information to be included in the reply.

People often need to call external services to get the information that their users are looking for and so in this article I describe a simple sample application written by myself and my colleague David Curran which shows a common pattern whereby the conversation service provides a template response along with parameters which can be used by the calling application to retrieve the necessary information to give the end user the answer that they are looking for.

This pattern is useful in a lot of different situations, but we will use a fictitious application of where people want to use a conversational interface to track their parcels. We will leverage the simple conversation application as a starting point to minimise the amount of work. You can either download that sample and follow the steps below to add the interface to the conversation agent, or if your prefer you can download the completed example from our GitHub repository.

Adding the Parcel Intent to the conversation

In order to modify the conversation agent to handle parcel requests, you first need to add a parcel intent to the list of intents. The original sample contains 25 intents which is the maximum allowed with the free plan, so you will need to delete one of the existing intents. I deleted the weather intent since it is not being used and then I added a parcel intent with a few sample inputs as you can see below,


The next step is to add a node to the dialog to specify how parcel queries are to be dealt with. Our logic is quite simple. If a number is detected in the input we assume that this is the parcel number so we set a context variable parcel_num with this value and then we send back a response message with placeholders where the parcel location should be inserted. However, if no number is detected in the input stream, we simply reply saying that they need to supply us with a parcel number. For simplicity sake we won't consider holding context from one question to the next.



Implementing the dummy parcel lookup service

We don't want to use a real parcel lookup service for this sample, because when testing we won't know the parcel number for parcels in transit. Instead we will implement a very simple lookup service.

To implement the parcel lookup service you need to add the following function near the end of app.js
 what this does is respond to get requests on /api/parcel and respond with one of the sample location names e.g. requesting http://localhost:3000/api/parcel?parcel_num=6 will return the string "Buckingham Palace". Just to illustrate how we should deal with errors, we have implemented the rule that if the parcel number is divisible by 13 it will return a status code 404 and an error message saying that the parcel number is unlucky.

/**
 * A dummy parcel tracking service
 */
 app.get('/api/parcel', function(req, res) {
   var parcel_num = parseInt(req.query.parcel_num);
   if (!req.query.parcel_num || isNaN(parcel_num)) {
     return res.status(400).end("Not a valid parcel number "
                                  +req.query.parcel_num);
   }
   if (0 == (parcel_num %13)) {
     return res.status(404).end("We can't find parcel number "
                                  +parcel_num+" it is unlucky!");
   }

   var locations = [
     'Anfield', 'Stamford Bridge', 'Old Trafford', 'Parkhead',
     'Heathrow Airport', 'Westminister, London', 'Buckingham Palace',
     'Lands End, Cornwall', 'John O\'Groats'
   ];

   parcel_num = parcel_num % locations.length;
   var location  = locations[parcel_num];
   res.end(location);
});

You should experiment with this service and/or customise it before moving on to the next steps.

Recognising a parcel location request and filling in the details

The main code  modification we need to do is in the app.post('/api/message',  function in app.js. However we first need to do some housekeeping changed due to the fact that we will be using the requestify library.

Add the following line to the dependencies section of package.json:
    "requestify": "^0.2.5",

Then add this line near the top of app.js
var requestify = require('requestify');

The nub of the code is contained in the function below. You should paste this into app.js to replace the call to conversation.message which is around line 56 of the original file.

  // Send the input to the conversation service
  conversation.message(payload, function(err, data) {
    if (err) {
      // the conversation service returned an error
      return res.status(err.code || 500).json(err);
    }
    var parcel_num = data.context.parcel_num;
    if (data.intents && (data.intents.length>0) && data.intents[0].intent
                  && (data.intents[0].intent === 'parcel') && parcel_num) {
      var server = 'localhost';
      var port = process.env.PORT || process.env.VCAP_APP_PORT || 3000;
      var url = 'http://' + server + ':' + port +'/api/parcel?parcel_num='+parcel_num;
      requestify.get(url)
        .then(function(response) {
          var location = response.body;
          data.output.text[0] = data.output.text[0].replace( /\{0\}/g, location);
          return res.json(data);
        })
        .catch(function(err){
          data.output.text[0] = "Parcel lookup service returned an error: "+err.body;
          return res.json(data);
        });
    } else {
      return res.json(data);
    }
  });

The original code did nothing other than calling the updateMessage function before passing the data received from the Conversation service back to the UI layer. However, the updateMessage function didn't do anything useful so we can delete it and instead we will call our dummy parcel location service to find the location of the parcel whose number appears in the context variable.

If the http call succeeds we assume that we have a good location and we replace any placeholder {0} strings in the message received from the conversation service with this location. If we get an error status from the http call, we replace the entire string received from the conversation service with a message saying we failed to locate the parcel.

Summary

You have a conversation service which can reply to questions such as "where is my parcel number 543210" with details of where the parcel is located. It is currently only using a toy implementation which pseudo-randomly selects locations in the UK. However, it should be relatively easy to extend it to any real parcel tracing service you want. In fact, the methods used can easily be applied to interfacing with any 3rd party service.

You can go to https://github.com/bodonova/conversation-parcel to download the complete working sample.

Sending an Email from Watson Conversation Service

Sending emails is a simple way to connect a chatbot to a business process. This post shows how to extend the Watson Conversation Sample Application to get it to send email from a Gmail account. We will use the nodemailer library and a slightly modified version of the code from w3 schools to help us.

The first step is to download the original sample application from GitHub. Follow the instructions in the Readme file in the repository to get this application running on your local machine and/or on the BlueMix service. Make sure you have the original application working correctly before you go on to make any changes.

Modify the code below to reference the username and password of an Gmail account you have access to (or create a new account) and then add it near the top of the app.js file.

var nodemailer = require ('nodemailer');

var transporter = nodemailer.createTransport({  
  service: 'Gmail',  
  auth: {  
   user: 'chatbot@gmail.com',  
   pass: 'secretpassword'  
  }  
 });  
 var mailOptions = {  
  from: 'chatbot@gmail.com',  
  to: 'name@email.com',  
  subject: 'Sending Email from the chatbot',  
  text: 'That was easy!'  
 }  

If you use a gmail email and two factor authentication in Gmail you have to get an app password for your gmail address. Google are worried you will share your Gmail password with people so they give you a special password just for your app that has limited powers.

Next update the Watson application's package.json to say that you want to install the nodemailer package and force the server to use a version of node.js that supports nodemailer i.e.:

  "dependencies": {
...,
  "nodemailer": ">0.3.x"
  },
...
  "engines": {
    "node": ">= 6.9.x",
    "npm": "> 5.30.x"
  }


Having done this, the next step is to change the conversation flow in the Watson Conversation Service so that when the user says ‘send an email’, we will set a context variable called 'email' to say an email should be set. (A production version would probably send different emails to different addresses depending upon context, but we will stick to a simple example for now)

If your WCS has an intent that you want to send an email. This contains example user utterances like

#sendEmail
Can you email for me?
email dave important info
Ping dave an email to tell him to do important business things
Send an email to find my package
Send an email reminder message
Message Dave a mail to tell him the stuff he needs to do

In WCS dialog tab we want a node that recognises this intent and sets a context variable to signal to the node application to send an email.

In the json editor of the dialog node add a context variable for the email.

{
  "context": {
    "email": "TRUE"
  },
  "output": {
    "text": {
      "values": [
        "Ill email david now"
      ],
      "selection_policy": "sequential"
    }
  }
}



Next, you need to change app.js to say that when this ‘send an email' context variable is send it. At the start of the function updateMessage add in the following code check to see if the context variable email from WCS' response has been set and send an email if it has.

if(response.context.email){
    response.context.email=null;//set send variable to null so we don't send email every time
  transporter.sendMail(mailOptions, function(error, info){

    if (error) {
      console.log(error);
    } else {
      console.log('Email sent: ' + info.response);

    }
  });  
}


You can download the complete sample from here

This code just sends the same email every time. In practice you would probably have to include some information from the conversation in the mail. Usually this would something from context variables. You would add in the email information the user had told the chatbot such as the value in response.context.query

I am not a node developer and this code needs some error correction and security checks added to it. But for a quick demo it should allow you to show your chatbot emailing.