Disruptive Telephony

NOTE: I have a few updates to the post that I am putting at the bottom of the text.

---

Would you like Skype users to be able to call your business' phone system? Would you like to connect your phone system to Skype's network and make use of their cheap calling rates? If you have an IP-PBX or other call server that supports the Session Initiation Protocol (SIP), you may now have those options.

For a company that only a bit over a year ago was saying that customers weren't asking for interconnection, today Skype has done something rather dramatic and lowered their walls a bit more with the announcement of the beta program of "Skype For SIP". With this announcement from the "Skype For Business" group, companies with SIP-enabled phone systems will be able to receive calls from Skype users - and make calls using Skype's network at Skype rates. The news release (and blog post announcement) highlights these four aspects:

• Receive and manage inbound calls from Skype users worldwide on SIP-enabled PBX systems; connecting the company Web site to the PBX system via click-to-call
• Place calls with Skype to landlines and mobile phones worldwide from any connected SIP-enabled PBX; reducing costs with Skype’s low-cost global rates
• Purchase Skype’s online numbers, to receive calls to the corporate PBX from landlines or mobile phones
• Manage Skype calls using their existing hardware and system applications such as call routing, conferencing, phone menus and voicemail; no additional downloads or training are required

Let's take those one at a time - and then take a look at some details and what's missing.

[NOTE: For the ease of writing this post I am going to refer to the SIP system as an "IP-PBX", but it could be a "call server", "call manager", "application server" or anything else than can send and receive SIP signaling. It could be open source or commercial - that doesn't really matter.]

---

INBOUND CALLS FROM SKYPE USERS

In a pre-announcement briefing, Chris Moore, a senior product manager at Skype, indicated that when the "Skype For SIP" service is fully released, the current "Business Control Panel" will be revamped a bit and will have an area where you can sign up for the service, identify your IP-PBX and associate one or more Skype names with your IP-PBX.

Calls from Skype users to those Skype names would then be routed across the SIP connection to your IP-PBX, where the IP-PBX would deal with those incoming connections exactly as it would any other incoming SIP connection. Call routing will be handled in the IP-PBX. Perhaps the incoming call will go to an auto-attendant, IVR, call-center software or other application. Perhaps it will be routed to a person. From the IP-PBX point-of-view, it's just another incoming call.

From a Skype users point-of-view, they are simply calling another Skype ID. It's a free call that seems just like any other Skype call.

You will, Moore stated, be able to associated multiple Skype IDs with your single SIP connection. Now I'm not sure what kind of call info you get across the SIP connection, but if you do get to see the Skype name the person is calling you could do some interesting call routing based on the Skype ID called. For instance, if someone made a Skype call to "companyname-help" it could be routed one way and calls to the Skype ID "companyname-sales" could go another way.

In any event, this aspect of the service makes it so that any Skype user can call your IP-PBX.

SIP TRUNKING, SKYPE-STYLE

The second aspect of the "Skype For SIP" service is that you can use Skype's global network for connections out to the PSTN... what we have been generically calling "SIP trunking" for some time now. As part of the Business Control Panel registration you will apparently indicate which Skype ID is to be charged for outbound calling (what we used to call "SkypeOut") and then any outbound calls via SIP will be charged to that account.

Effectively, Skype just opened up cheap international calling to businesses everywhere using Skype's cheap rates.

The already crowded "SIP trunking" market just got another big player. Configure your IP-PBX to send calls out across Skype's network to the PSTN... and start calling. Right now the plan as I understand it is that you would pay the regular SkypeOut rates, without the subscription plans that are available to individuals. Skype's Moore did say that they may evaluate some form of plans during the beta period. As we see all these various details, we'll have to see what impact this will have on the existing players.

ESTABLISHING PHONE NUMBERS AROUND THE WORLD

The third aspect of the announcement is that you can establish phone numbers around the world that will route to your IP-PBX via the Skype For SIP connection. Since, as I mentioned above, you are associating one or more Skype accounts with the SIP connection to your IP-PBX, you can also associate the "Online Numbers" (what we used to call "SkypeIn" numbers) of those Skype accounts with your IP-PBX.

So if you want a phone number in one of the 20 countries where Skype currently supports Online Numbers, you just buy the number for one of the Skype accounts connected to your IP-PBX.

For $60 per year per online number.

Not a bad deal if you want phone numbers in other area codes or other countries that will ring your business' phone system. Again depending upon what kind of caller ID and called ID info you receive, some interesting call routing might be possible.

(This assumes, of course, that Skype keeps the current pricing for Online Numbers and doesn't charge any additional costs.)

USE EXISTING HARDWARE

Skype's point here is really just that there's no additional software... it's just an inbound SIP connection to your IP-PBX that you deal with in the same way that you deal with all other inbound SIP connections.

---

COMMENTS/CONCERNS

Based on the conversations I had with folks from Skype about this new service, I do have a couple of comments and concerns:

SECURITY

Yes, okay, you would expect this of me. Obviously we'll have to wait to see the implementation, but it sounds like Skype has thought this through a good bit. They'll support the standard SIP digest authentication, but more interestingly they will support restricting connections based on IP addresses. If your IP-PBX - or more likely the SIP-aware firewall or SBC or edge proxy - has a fixed address you will apparently be able to enter this in and use that to limit inbound SIP connections. Skype also indicated that when they service moves out of beta into full production they intend to support TLS-encrypted SIP as well.

Similarly, on the media side the beta will support regular RTP but Skype is looking to support SRTP in the full production release.

CODECS

In a somewhat bizarre move (to me), Skype is initially releasing this in the beta program with only support for the G.729 codec. For those who don't follow audio codecs (used to encode the audio of your voice to send across the network), the G.729 codec compresses audio and results in lower bandwidth usage. It also, unfortunately, requires the payment of royalties which typically then require the purchase of additional licenses from the IP-PBX vendor. (This is true even in the case of Asterisk, where you can purchase G.729 licenses from Digium.)

Given that the very people likely to want to use Skype's services for low-cost calling are also the same people who are probably not going to pay for G.729 licenses, it seems that there is a bit of a disconnect here.

The good news is that the folks at Skype say that they are going through the testing to make the much more widely used (and royalty-free) G.711 codec available and expect to have that ready within the first couple of weeks of the beta program.

On the wideband side, Skype folks indicated that at the point in time where there are SIP endpoints that support the SILK codec, which Skype recently said they will make available in binary form for free, those SIP endpoints should be able to make and receive calls to/from Skype users with wideband audio.

(I'll just note that Skype's rationale when I asked them about why G.729 vs G.711 was that they currently use G.729 with all their many SIP termination providers and so using that codec just seemed to make sense to them. For someone with the high volume of calls that they have who are looking to send as many as possible over limited bandwidth, that probably does make sense. However, in this era of more and more available bandwidth, I've seen many people, especially on the SMB side, less concerned about conserving bandwidth and just using G.711.)

SIPconnect

I was pleased to hear Skype's Moore mention that they are looking at specifications like the SIP Forum's SIPconnect initiative as a way to help with interoperability from premise IP-PBX's out to Skype's service. Having been peripherally involved with SIPconnect, this is exactly the kind of situation that it's trying to address (interop between a premise SIP system and a SIP Service Provider). It would be great if Skype would formally get behind that initiative. (I can see certain SIP Forum people typing email as soon as they read this... ;-)

OUTBOUND *TO* SKYPE USERS

It's interesting to note what this release does NOT have - the ability to call from your IP-PBX out to a Skype user. You can call out to PSTN numbers via the SkypeOut connection... but you can't call a Skype ID. This isn't surprising, on one level, because this is a MUCH harder problem to solve. Basically every SKYPE ID would need a SIP address (a "SIP URI" in SIP-speak) that the IP-PBX could use to connect.

Several of us (myself included) have been asking for that kind of interconnection for years - and perhaps at some point we'll see that. Meanwhile, this "Skype For SIP" release gets us closer.

---

SOME CLOSING WORDS...

Over the years, I have written a good deal about Skype on this blog - and I have certainly been critical of Skype's closed network in the past (such as here and here). I don't like "walled gardens" in general and Skype has definitely had high walls.

That's certainly changing. I've definitely been pleased to see what they started last fall with "Skype For Asterisk" (which I wrote about here, here and here).

And now... "Skype For SIP".

If Skype implements this well, I think there is great potential. Suddenly, the millions and millions of Skype users are able to call your phone system... just via Skype. Forget dealing with long-distance or international calls. Just have someone use Skype to call your Skype ID.... ta da, it winds up on your corporate phone system via SIP. Similarly there is now the ability to easily project your presence geographically with phone numbers in different regions or countries - all through Skype's easy UI. Let alone the SIP trunking via Skype's network.

Granted, you can get the phone numbers and trunking through existing SIP service providers today. Skype just makes this a bit easier because they have the existing programs, user interfaces, etc.

Skype For SIP isn't perfect - it still doesn't get me the outbound calling to Skype users that I want - but it's definitely a step in the right direction as we continue building the interconnect between all these IP communications systems. (See my rant here and my Park Bench Manifesto presentation out at eComm)

Kudos to the folks at Skype for taking this step. For lowering their walls a bit more and letting others connect in. I definitely will look forward to seeing what evolves out of this.

Meanwhile, of course, I'll be heading over to skypeforsip.com and signing up for the beta program. :-)

---

UPDATE #1: (a few minutes after posting) Two items:

• GigaOm has a post up that I'm sure will attract some good comments
• The skypeforsip.com website is apparently not live yet (oops!) but will be soon. The skypeforsip.com is now live and accepting applications for the beta program

UPDATE #2: (6 hours later)

• Moshe Maeir disputes my characterization of G.729 and says most of his customers use it.
  
  
• Phil Wolff over at Skype Journal came out with his contrarian piece: Skype For SIP: Big Money, Skypeless, Brand Destroyer that argues that Skype For SIP is a negative thing for Skype. I disagree with a number of Phil's points (and would note that some of his info relates to the SFS *beta* versus the full announced release (such as the 1 Skype ID only limitation), but it's definitely worth a read.
  
  
• Phil does make the point, reinforced in the SFS beta application, that during the beta you can only have one Skype ID associated with your Skype connection. Per the beta application, it also needs to be a "temporary" one, i.e. you may not be able to continue using that after the end of the beta period. Chris Moore at Skype indicates that this has to do with different Terms of Service and EULAs for business users right now versus regular Skype users and they are looking to rationalize all of that before the full product launch.

UPDATE #3:

• Jason Goecke has a view that the release of Skype For SIP greatly reduces the value of Skype For Asterisk.

UPDATE #4:

• Rich Tehrani thinks that Skype should just call a spade a spade and refer to SFS as "SIP trunking", which is what we in the industry would certainly call the PSTN connectivity side of what Skype is offering. I think, though, that as much as we use that term, it is not widely used outside of our own industry. So as Skype seems to promote itself to the larger business audience, it makes sense to speak of it more openly as "placing calls", etc.
  
  
• Irv Shapiro over at IfByPhone weighs in on "Why Skype for Asterisk is more important than Skype for SIP". Irv believes because SFA lets out make outbound calls TO Skype users it is ultimately more important than SFS. I agree with Irv that SFA definitely has advantage and value because of this.

---

If you found this post interesting or useful, please consider either subscribing to the RSS feed or following me on Twitter or identi.ca.

---

Technorati Tags: skype, sip, standards, voice, voip