Chair: All right, Skype and Asterisk, what does it mean for business communications? Founder and CTO of Digium, Mark.
Mark: Thank you. Asterisk and Skype is probably not a pairing that a lot of you guys might have thought about a year ago. You would have thought about, "Wow, it would be nice," but didn't seem much like a reality. It is exciting that we have it now getting closer to being a reality.
First of all, what is Skype for Asterisk? You're used to SIP and SIP let's you talk to SIP devices, and Skype let's you talk to Skype devices, which pretty much is, of course, just Skype. It is a generic channel driver for Asterisk, so it supports most of the things that Asterisk channel drivers already support. There is no special programming or anything; you just install it an you can use it.
It supports, of course, the usernames, encryption, end points, and it supports both talking to regular Skype names, any arbitrary Skype name, as well as talking to the SkypeIn, SkypeOut services.
It's really, the first practical Skype gateway from a PBX platform. It allows you to connect this really broad user base of people that are already using Skype, with Asterisk. If you think about Asterisk as a very pragmatic and practical platform for telephony, for business phone systems, Skype has been incredibly successful in the Voice over IP space because it's been a very pragmatic solution for customers to be able to use.
I've often said that the lesson of Skype is really that when you look at open source, open standards, and all those things, their value is really only as strong as their ability to deliver on ease of use, performance, and low cost to the consumer. Even though Skype wasn't open source, wasn't open standards, or any of that stuff, because they were able to deliver value to the customer they were able to get such a tremendous amount of following and a number of users who are obviously accessible to you, through Asterisk.
There have been some other very ugly hacks that involve virtual machines and emulating sound hardware and stuff like that, but the Skype for Asterisk is literally just native Skype code running within the Asterisk environment so it removes this huge, ugly hack factor. It's actually scalable and it integrates nicely in Asterisk with behaviors you would expect.
There are several use cases to go over. The business call centers, of course, is one. Now, anybody who has Skype can contact you and you could register a Skype name for your business and have people call directly into there. It integrates very nicely in and gives you a low-cost way for your people to call you without having to run up 800 number minutes. A lot of customers already have Skype and you could have click-to-dial, and all that kind of stuff.
The other thing, of course, is business PBX. If you like to use Skype for your business communications, which a lot of customers do, even though the IT people typically don't; you can use SkypeIn and SkypeOut minutes to associate with your PBX and you can also have your Skype username that both comes to your Skype client natively, and rings over to the Asterisk PBX. For example, you could have everything unified on the same voice mailbox. When someone calls you, it can ring both your hard phone and your Skype soft phone.
Of course for end users, any application that you have that you would want to expose to a wide variety of users, you could do so via Skype. If you wanted to make a call for the weather, or whatever it is, any kind of IVR, you could connect all that in and allow for calling people back via Skype, as well.
The status of the Beta - because of the complexity of the integration, we decided to start with a closed Beta. We were only going to open it up to a certain number of people, initially, so that we could have engineers working directly with the customers. That started in January of this year. We have over 100 people, but less than 1,000, so far, in the Beta. We will be going to a public Beta, hopefully, very soon. So far, it's been working reasonably well.
There are some big caveats that are very important. First of all, Skype is also in the process of releasing something called the Business Control Panel. Although it's not implemented in the current Beta, Skype is requiring that the usernames you use to register your device with Skype, in other words, the ones you use with the Skype for Asterisk, will all have to be business control panel accounts, which I believe means you are not going to be able to use existing accounts unless you are somehow able to make them part of the business control panel.
This is something that Skype has demanded, so feel free to go tackle him, over there, if you have any concerns about that. There is not much we can do on it, right now. However, there are areas that we are interested in hearing more about, like chat and video. We would definitely like to get your input on priorities about how those features would want to be integrated in, in later revisions of the product.
You dial it kind of as you would expect. It's just another channel driver and you do Skype slash the username. Presence is supported. We do have some AMI events that are generated, to give you a little bit more ability to kind of hook in for some of the Skype-specific stuff that wouldn't be present in other telephony interfaces.
You actually have access, as it turns out, to a lot of the variables in the Skype call. For example, you could use the language that is provided by the user in their configuration to give them IVR in their native language of choice. Some of the others, I guess you could wish them a happy birthday if they called on their birthday. I don't know exactly what you would do with that. You can see there is a lot of demographic information that may be able to be helpful, if for nothing else, from a logging and statistic gathering point of view.
There is a shortened URL, if you want to sign up for the Beta for when it does go to the public Beta. I will save the rest of the time for questions. Your questions can be about Asterisk generally, or specific to the Skype for Asterisk.
Audience 1: Is this going to use the Asterisk jitter buffer, or Skype's jittering, fancy de-jittering technology?
Mark: It's going to use the Skype de-jittering, but it will use the Asterisk native codecs. Essentially, all the existing codec work you do with Asterisk, already, will be used. That is all native. It doesn't have to get retranslated.
Audience 2: What caller ID will it actually pass?
Mark: The caller ID will be based on the Skype username that you are using to place the call. You can register multiple Skype usernames with it and then say, "I want this call to come from this particular Skype username". On inbound, it will obviously be the phone number associated with the caller. Typically, it would just be their Skype name.
Audience 3: So if you put it under the corporate name...
Mark: Right, it would go out, for example if it was Digium, the word Digium would be the caller ID and then the name would be whatever name you had associated with it.
Audience 4: You can use a mobile number as your Skype caller ID, instead of your regular Skype ID. Will the...
Mark: I'm not really sure if we've set it up for that. I hadn't thought about that, in terms of allowing your mobile number, whatever you have listed in Skype as your actual phone number...
Audience 4: It becomes handy for getting into things like CauliFlower, and so on.
Mark: Okay, I will have to get back to you on that one. You stumped me on that one.
Audience 5: Mark, if you're phoning someone like 1-800-go-fedex through Asterisk to Skype, does early media work, and can we get the DTMF into that?
Mark: The early media is only unidirectional, right now, in Skype. I don't believe you can transmit media before the call is through. FedEx, as a specific example, probably wouldn't work, although now that you mention it, I don't know how you do that through SkypeOut anyway. That's a good follow up question.
Audience 6: There was a question about the jitter buffer. Can you elaborate a little bit about the media exchange, going from a SIP endpoint to Skype? It's starting out on RTP, and then are you guys doing a virtual driver, or are you writing a wav file to a file and then playing it out?
Mark: The audio comes in over RTP, into Asterisk, and Asterisk routes it out the Skype interface.
Audience 7: I have two questions. One has to do with security. At what point does the Skype call cease to be encrypted? The other question is will you be allowing Skype-to-Skype re-routing, as opposed to a PSTN to Skype? What can I do, through this application, transfer a call to another Skype user?
Mark: The first question is when does it get unencrypted. It gets unencrypted when it hits Asterisk, obviously it has to be to be able to convert to other media. I think the second question was can you have a Skype call in and then send a Skype call back out. The answer is yes, you can. I don't know if it's going to be out in the first release, but there is a way to transfer that call off. Otherwise, you would be in the middle of that call and you would essentially be getting the media, decrypting it and re-encrypting it on the other leg of the call.
Audience 8: Mark, is there a charge for the [0:11:56.7 unclear] Skype? Has that been determined, yet?
Mark: The current plan is that they would be sold, more or less, like the G.729 licenses, but there hasn't been any kind of formal price announcement or anything like that. Hopefully, as we get closer to the final Beta, we'll be able to confirm what the terms of that will be.
Audience 9: I'm just curious; there is something I heard about, called SkyHost. Is that part of this solution?
Mark: That's basically a name for the API that's being used, yes.
Thank you very much.
Chair: Great job, thanks Mark.
Mark: Thank you. Asterisk and Skype is probably not a pairing that a lot of you guys might have thought about a year ago. You would have thought about, "Wow, it would be nice," but didn't seem much like a reality. It is exciting that we have it now getting closer to being a reality.
First of all, what is Skype for Asterisk? You're used to SIP and SIP let's you talk to SIP devices, and Skype let's you talk to Skype devices, which pretty much is, of course, just Skype. It is a generic channel driver for Asterisk, so it supports most of the things that Asterisk channel drivers already support. There is no special programming or anything; you just install it an you can use it.
It supports, of course, the usernames, encryption, end points, and it supports both talking to regular Skype names, any arbitrary Skype name, as well as talking to the SkypeIn, SkypeOut services.
It's really, the first practical Skype gateway from a PBX platform. It allows you to connect this really broad user base of people that are already using Skype, with Asterisk. If you think about Asterisk as a very pragmatic and practical platform for telephony, for business phone systems, Skype has been incredibly successful in the Voice over IP space because it's been a very pragmatic solution for customers to be able to use.
I've often said that the lesson of Skype is really that when you look at open source, open standards, and all those things, their value is really only as strong as their ability to deliver on ease of use, performance, and low cost to the consumer. Even though Skype wasn't open source, wasn't open standards, or any of that stuff, because they were able to deliver value to the customer they were able to get such a tremendous amount of following and a number of users who are obviously accessible to you, through Asterisk.
There have been some other very ugly hacks that involve virtual machines and emulating sound hardware and stuff like that, but the Skype for Asterisk is literally just native Skype code running within the Asterisk environment so it removes this huge, ugly hack factor. It's actually scalable and it integrates nicely in Asterisk with behaviors you would expect.
There are several use cases to go over. The business call centers, of course, is one. Now, anybody who has Skype can contact you and you could register a Skype name for your business and have people call directly into there. It integrates very nicely in and gives you a low-cost way for your people to call you without having to run up 800 number minutes. A lot of customers already have Skype and you could have click-to-dial, and all that kind of stuff.
The other thing, of course, is business PBX. If you like to use Skype for your business communications, which a lot of customers do, even though the IT people typically don't; you can use SkypeIn and SkypeOut minutes to associate with your PBX and you can also have your Skype username that both comes to your Skype client natively, and rings over to the Asterisk PBX. For example, you could have everything unified on the same voice mailbox. When someone calls you, it can ring both your hard phone and your Skype soft phone.
Of course for end users, any application that you have that you would want to expose to a wide variety of users, you could do so via Skype. If you wanted to make a call for the weather, or whatever it is, any kind of IVR, you could connect all that in and allow for calling people back via Skype, as well.
The status of the Beta - because of the complexity of the integration, we decided to start with a closed Beta. We were only going to open it up to a certain number of people, initially, so that we could have engineers working directly with the customers. That started in January of this year. We have over 100 people, but less than 1,000, so far, in the Beta. We will be going to a public Beta, hopefully, very soon. So far, it's been working reasonably well.
There are some big caveats that are very important. First of all, Skype is also in the process of releasing something called the Business Control Panel. Although it's not implemented in the current Beta, Skype is requiring that the usernames you use to register your device with Skype, in other words, the ones you use with the Skype for Asterisk, will all have to be business control panel accounts, which I believe means you are not going to be able to use existing accounts unless you are somehow able to make them part of the business control panel.
This is something that Skype has demanded, so feel free to go tackle him, over there, if you have any concerns about that. There is not much we can do on it, right now. However, there are areas that we are interested in hearing more about, like chat and video. We would definitely like to get your input on priorities about how those features would want to be integrated in, in later revisions of the product.
You dial it kind of as you would expect. It's just another channel driver and you do Skype slash the username. Presence is supported. We do have some AMI events that are generated, to give you a little bit more ability to kind of hook in for some of the Skype-specific stuff that wouldn't be present in other telephony interfaces.
You actually have access, as it turns out, to a lot of the variables in the Skype call. For example, you could use the language that is provided by the user in their configuration to give them IVR in their native language of choice. Some of the others, I guess you could wish them a happy birthday if they called on their birthday. I don't know exactly what you would do with that. You can see there is a lot of demographic information that may be able to be helpful, if for nothing else, from a logging and statistic gathering point of view.
There is a shortened URL, if you want to sign up for the Beta for when it does go to the public Beta. I will save the rest of the time for questions. Your questions can be about Asterisk generally, or specific to the Skype for Asterisk.
Audience 1: Is this going to use the Asterisk jitter buffer, or Skype's jittering, fancy de-jittering technology?
Mark: It's going to use the Skype de-jittering, but it will use the Asterisk native codecs. Essentially, all the existing codec work you do with Asterisk, already, will be used. That is all native. It doesn't have to get retranslated.
Audience 2: What caller ID will it actually pass?
Mark: The caller ID will be based on the Skype username that you are using to place the call. You can register multiple Skype usernames with it and then say, "I want this call to come from this particular Skype username". On inbound, it will obviously be the phone number associated with the caller. Typically, it would just be their Skype name.
Audience 3: So if you put it under the corporate name...
Mark: Right, it would go out, for example if it was Digium, the word Digium would be the caller ID and then the name would be whatever name you had associated with it.
Audience 4: You can use a mobile number as your Skype caller ID, instead of your regular Skype ID. Will the...
Mark: I'm not really sure if we've set it up for that. I hadn't thought about that, in terms of allowing your mobile number, whatever you have listed in Skype as your actual phone number...
Audience 4: It becomes handy for getting into things like CauliFlower, and so on.
Mark: Okay, I will have to get back to you on that one. You stumped me on that one.
Audience 5: Mark, if you're phoning someone like 1-800-go-fedex through Asterisk to Skype, does early media work, and can we get the DTMF into that?
Mark: The early media is only unidirectional, right now, in Skype. I don't believe you can transmit media before the call is through. FedEx, as a specific example, probably wouldn't work, although now that you mention it, I don't know how you do that through SkypeOut anyway. That's a good follow up question.
Audience 6: There was a question about the jitter buffer. Can you elaborate a little bit about the media exchange, going from a SIP endpoint to Skype? It's starting out on RTP, and then are you guys doing a virtual driver, or are you writing a wav file to a file and then playing it out?
Mark: The audio comes in over RTP, into Asterisk, and Asterisk routes it out the Skype interface.
Audience 7: I have two questions. One has to do with security. At what point does the Skype call cease to be encrypted? The other question is will you be allowing Skype-to-Skype re-routing, as opposed to a PSTN to Skype? What can I do, through this application, transfer a call to another Skype user?
Mark: The first question is when does it get unencrypted. It gets unencrypted when it hits Asterisk, obviously it has to be to be able to convert to other media. I think the second question was can you have a Skype call in and then send a Skype call back out. The answer is yes, you can. I don't know if it's going to be out in the first release, but there is a way to transfer that call off. Otherwise, you would be in the middle of that call and you would essentially be getting the media, decrypting it and re-encrypting it on the other leg of the call.
Audience 8: Mark, is there a charge for the [0:11:56.7 unclear] Skype? Has that been determined, yet?
Mark: The current plan is that they would be sold, more or less, like the G.729 licenses, but there hasn't been any kind of formal price announcement or anything like that. Hopefully, as we get closer to the final Beta, we'll be able to confirm what the terms of that will be.
Audience 9: I'm just curious; there is something I heard about, called SkyHost. Is that part of this solution?
Mark: That's basically a name for the API that's being used, yes.
Thank you very much.
Chair: Great job, thanks Mark.
