Early impressions of Skype for SIP (SfS)

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…