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Hi I'd like to use the Skype developer tools to create this functionality using Javascript or PHP or both but am having trouble starting:

I host Skype calls with users from around the world. The language makeup of these calls is overwhelmingly Spanish followed by English. There are also participants who speak Bengali, Indian, Chinese and some slightly lesser represented dialects. There are translators for English, Chinese and Spanish.

I'd like to improve the efficiency and accessibility of those calls by making it feel more organic in 3 key areas:

My goals:

  • Maintain a system whereby everyone hears the first untranslated voice speaking. (For the purpose of maintaining a sense of solidarity and authenticity).
  • Decrease the time interval everyone has to wait each translation after first speaker has finished. (Make it possible for everyone to hear their own translation privately and simultaneously).
  • Allow everyone to see when to see when all translations have been completed and follow respond.

Challenges:

  • It’s not always the same person speaking, someone else may follow up (possibly in a different language).

Any ideas and is this even possible? Thanks in advance.

Wahyu Kristianto
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1 Answers1

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Skype developer tools

Ouch. Microsoft has discontinued the Skype API and I believe you can't even download the developer tools any more. Trillian, the IM messenger, will probably drop Skype support because the API is not supported anymore, and Digium no longer supports it's chan_skype extension.

Even if it's still available, I don't think there ever was an API for controlling Skype calls via JavaScript or even PHP for that matter.

What you can do

Try Asterisk.

Asterisk is a software PBX, and it's free. I recommend you try the FreePBX variant of Asterisk, because it comes with a nice browser GUI. Also free. While you can't connect it to the Skype network via the chan_skype module anymore, you can either:

a) Purchase a Skype Connect subscription. It's official, it's provided by Skype itself. It's a SIP channel that you can easily connect to any software PBX, such as FreeSwitch, Cisco telephony softare, and, yes, Asterisk. But, it costs 5-6$ I belive PER CHANNEL. If you purchase 5 channels, you can have 5 inbound calls to your Asterisk PBX system.

b) Use mod_skypopen with FreeSwitch to provide Skype connectivity to your Asterisk box. It works by starting a number of hidden Skype clients and tunneling their audio in/out to the FreeSwitch PBX, which you can later use to connect to Asterisk via SIP. It's free, and I believe it supports up to 10 channels. No SkypeConnect subscription necessary. It's tricky to set up, though, and you require a dedicated machine for it.

You don't have to connect Asterisk with Skype, though. You can tell your clients to install a SIP client, such as XLite or Linphone. If not, you can place a SIP client on your own website with HTML5: http://sipml5.org/

After you're done with all that, you can use Asterisk to host conference calls easily, and you can mute participants at will. You can script your own call routines, you can detect silence, you can execute *nix commands, you can detect DTMF tones (e.g. "push * when you're done"), you can use TTS (Text to speech). If Asterisk's scripting language does not provide what you need, you can always write a special plugin for your needs using the Asterisk API (of which there are multiple types).

The Asterisk Manager API will be useful: http://www.voip-info.org/wiki/view/Asterisk+manager+API

And, yes, you can send it commands via PHP. Muting is the command what you're most likely seeking.

I can't go into depths of what Asterisk is capable of, because there are whole books written for it, this would be one hell of a long post. But here is a simple demo of how you can set up a conference system:

https://www.youtube.com/watch?v=KTnlIv592uw

Don't have a dedicated machine to host Asterisk/FreePBX? You can rent one. Search online for "PBX hosting". Yeah, it isn't easy, but that's telephony for you.

Summary: Connect Skype to an Asterisk PBX and control your calls from there via scripting or using the Asterisk Manager API.

Normal's overrated
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