Warheads Stacked in the Kitchen

The day has finally come, I have been able to place a SIP call directly to and from the Skype network via a Skype service. I received my Skype for SIP (SfS) closed beta credentials yesterday and immediately set to work configuring endpoints to start making calls. 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...

Today Skype announced Skype for SIP (SFS). Put simply, enterprise telephone systems may now interconnect with the Skype network to receive calls from the Skype network and place calls to SkypeOut. All without the need to install any special hardware or software on most modern enterprise phone systems (IP-PBXs to be more specific). Skype's new enterprise targeted connectivity uses SIP, the industry standard for VoIP interconnection. SIP already powers the bulk of Skype's revenue, via SkypeIn/SkypeOut, so this is a logical progression to take advantage of the large scale infrastructure already in place at Skype. This is a tremendous move by Skype and one I have contended for years was necessary for them to make headway in the enterprise. I applaud this step. There are plenty of great posts out there covering this already, including the one by @danyork on Disruptive Telephony. What does this mean for Skype for Asterisk (SFA) announced last September? At best the value of SFA has been signficantly reduced by this announcement. Previously SIP interconnection to the Skype cloud was given to the rarified group of larger players such as Voxeo, Tellme, Genesys and others. SFA was the first time this access was going to be brought to the world of open source telephony developers through Asterisk. This provided an immense opportunity for the Asterisk developer community to create new applications to take advantage of this, which lead me to invest time to participate in the closed beta for SFA still underway. The SFS announcement this morning has just marginalized SFA to applications that benefit from direct dialing of Skype users from Asterisk and from basic presence updates from the Skype network. Gone are the benefits of providing Skype/SkypeIn inbound calls to the enterprise, SkypeOut trunking, etc. More so, SFA is at a disadvantage since you will have to pay a per channel (simultaneous call) license fee on top of any SkypeIn/SkypeOut costs. Further, I suspect that the number of SFA channels available to a single account will be limited for the same reason that SFS does not do SIP to Skype dialing, so that no one may provide large scale alternatives to SkypeIn. All of this has really taken the wind out of the SFA sails before it even had a chance to make it to a public beta. Digium must now look to quickly add new features. Such as advanced presence information, instant messaging, the SILK codec and others, if they hope to salvage their own investment in the development of SFA to date. While I understand these things take time, the lethargy of getting the SFA to market does not bode well for rapidly trumping the SFS announcement. Time will tell.

Gizmo5 has recently announced their OpenSky service, allowing SIP dialing into the Skype network. OpenSky allows calling from:

• Your Gizmo5 desktop client
• Nokia mobile phone software client
• Any mobile phone using a text message to kick-off the call
• Any SIP endpoint such as Asterisk

Calls to the Skype network are free up to 5 minutes. If you want longer calls then it is $20 for one user per year on a discounted scale based on number of users thereafter.

It is great to see networks beginning to interconnect with Skype and force open their closed and proprietary network. There is still a long way to go. In this case Skype and eBay do not support this interconnection, so there is no apparent way to call from Skype to the Gizmo5 network. Further, dialing to the Skype network from outside is cumbersome, although Gizmo5 has added aliasing to mitigate this issue.

With initiatives like these and the upcoming (fingers crossed) release of the Asterisk/Skype channel, we will begin to see some interesting innovations leveraging the Skype cloud.

When I moved back to the US after several years in Europe, just under 2 years ago, I had the unique situation of having no legacy phone number. At this time GrandCentral was still an independent startup in beta offering one phone number to rule them all. I jumped on the opportunity to have a GrandCentral number and then only gave that number out both professionally and personally. Having all of the GrandCentral features was a breath of fresh air at the time, but quickly became stale after the Google acquisition of GrandCentral in July of 2007.

At first I thought the Google acquisition was great, as it seemed to guarantee that number could be relied upon 'for life'. But after over a year and a half of silence, I began to think that all that had happened was that GrandCentral had been forgotten. Would GrandCentral go the way of so many other previous services? This appeared to be a certainty when WebWorkerDaily reported that the GrandCentral SSL Cert was expiring...

Then an eagle eye over at the Google Operating System blog noted that Jeff Huber, SVP of Engineering at Google, commented on a FriendFeed post about the SSL cert story:

Writely + XL2Web + TonicSystems -> Google Docs, Keyhole -> Google Earth/Maps, Urchin + MeasureMap -> Google Analytics, JotSpot -> Google Sites, Zingku -> Google FriendConnect, Android -> Android, DoubleClick -> DoubleClick, Feedburner -> AdSense for Feeds (in-process); sorry about Dodgeball.

... and David Pogue's comment after the post gets it right -- a new version on new infrastructure will be coming soon. Apologies to anyone who has run into issues on the legacy version.

Will this be simply a back-end infrastructure change, or will we see all the great features we have been waiting for in a 2.0 release? I am anxiously awaiting an API for call routing, hooking up my own SIP endpoints and having integration with all of my other Google Apps. Has Google created the killer VSB/SME application platform by adding voice? I certainly hope to find out soon...

  

Those who know me are well aware that I have long been a detractor of Skype. I do respect and admire much of what they have done, but the end does not justify the means. While Skype did bring VoIP to the global masses, they did so by creating a walled garden reminiscent of the early days of AOL.

Of course Skype had to come up with an alternative to SIP to resolve NAT issues, but they did not have to keep it closed and proprietary. Google achieved the same thing with GTalk. Google then not only published their Jingle extension to XMPP, but even provided a library for everyone to use. Skype's commercial interest is clearly served by keeping their own protocol under lock and key, but this has stifled innovation. In fairness Skype did release a desktop API and allowed limited access to their SkypeIn/SkypeOut SIP Gateways for large vendors such as Tellme. This is not the same as allowing full peers in their network.

There was a glimmer of hope at Astricon in Glendale last September that I attended. Digium and Skype announced connectivity between the P2P Skype network and Asterisk. This would allow open source developers to begin providing solutions leveraging Skype and its large network of users. But that was September and four months later an imminent release does not appear to be on the horizon.

The other major issue that stifles Skype is their corporate parent, Ebay. Ebay's acquisition of Skype for billions in 2005 was one of the larger blunders in recent tech history. With the latest Ebay earnings call the long held rumors that eBay is about to sell Skype are increasing. The Times published an article today highlighting the reason for the uptick in speculation:

Industry insiders believe that eBay signalled its intent last week after John Donahue, its chief executive, described Skype as a “great stand-alone business”.

Ebay's shareholders would be well served by selling Skype now and putting the cash back in the bank. While Skype themselves, and their users, would be better served by new ownership and vision.

AT&T would do well by acquiring Skype, opening the network by publishing the protocols and instantaneously becoming the dominant player in the space. The synergy that never existed between Skype and Ebay, would be replaced by synergy that could drive real innovation.