With the passing of Gizmo5, so goes one of the last and largest open-standards based VoIP services. Gizmo5 supported the interoperable SIP Internet standard protocol. That meant it worked with any SIP hardware, software, PBX, etc. Most VoIP services, especially the consumer-focused ones, including Google Voice, do not support standards, and instead require proprietary, closed, hardware or software.

As can be seen below, from a snapshot of the SIPphone.com site in 2004, the company started off being an advocate for SIP and open-standards, with an emphasis on free SIP to SIP calls, using any compliant device or software that you want.

Gizmo5 could be used with off-the-shelf hardware devices, ATAs, SIP phones, Wi-fi phones, and standard software on smartphones and PCs.  Google Voice can’t.

In fact, back in 2008, Michael Robertson (CEO of Gizmo) wrote a long open-letter on the subject: Gizmo Project’s Michael Robertson Sounds Off explaining about how Skype was closed (it still is) and why being interoperable was a good thing for competition, etc. You could pretty much take that letter today, and replace “Skype” as the target with “Google Voice” – for instance:

If Skype truly believes there should be open competition then they should start by enabling other networks such as Gizmo5 to call Skype users in an official and supported capacity.

Reads just as well as:

If [Google Voice] truly believes there should be open competition then they should start by enabling other networks such as [Insert Third-party-Provider Here] to call [Google Voice] users in an official and supported capacity.

I guess it was convenient for Gizmo’s business objectives to “wave the flag of openness” back then, but it apparently isn’t so for Google and Google Voice now, who conveniently ignore competitors, as well their own customers’ requests for openness and interoperability.

It sucks.  The suits won. I expected better from Google.

Maybe I’ll have more to say on this when I have more time.

When I first considered all the moving pieces to getting it working, I gave myself about a 25% chance of success. To make it even more challenging, I decided to try to set it up on an Ubuntu Linux VM running under VirtualBox on Mac OS X.

Well, to my pleasant surprise, the whole thing went swimmingly well. I followed the excellent SipToSis Linux Tips documentation on the SipToSis site and it all went smoothly. The first trick is to find a static version 2.1.0.81 download of Skype for Linux – to ensure compatibility with SipToSis. (I can’t put a link here because it moves around and I don’t think anyone is allowed to host it officially – you have to get Skype from Skype’s own website.)

Once the proper version of Skype is installed, it’s just a matter of setting it up to use the snd-dummy fake sound drivers and getting SipToSis installed and configured.

The Skype wire protocol is not open nor documented, so one cannot simply connect to Skype directly via the network from a third-party program. Third party apps can, however, interact with the running Skype (binary) process on the same machine via an inter-process communications API.  So to use SipToSis, you first start the normal Skype application on the machine and then start the SipToSis application (on the same machine) – SipToSis then connects to the Skype application via the inter-process communications API (on the same machine) to manage calls on the Skype side via the Skype App, while speaking the open-standard SIP protocol on the other side directly.

In this basic configuration, there is one instance of Skype running, with one Skype username, which means it’s basically a one-channel gateway, one call at time. The call flow for a call from SIP to a Skype user looks like this:

SipToSis - SIP to Skype Call Flow

Note that SipToSis and Skype (representing the running Skype App) are shown inside one box because they are on the same machine, not connected over the network. The arrows inside that box, between SipToSis and the Skype App, represents Skype API inter-process communication – not network traffic. The other arrows in the diagram, the arrows between boxes, represent network traffic flows,where those elements could be on the same machine, or on a separate machine across the room or miles way. The Skype user SkypeUser will see an inbound call from the user that was used to sign in with the Skype instance that SipToSis is attached to. All calls will appear to originate from the same Skype user in this simple single-channel setup.

The SipToSis app talks SIP protocol directly on one side and uses the Skype App on the other side as a “black box” to speak the proprietary, undocumented, Skype protocol.

As shown above, I have SipToSis registering to a SIP proxy, where I can send calls from SIP to Skype via SipToSis – in this case, I have speed-dial codes that I dial from a plain handset that place Skype calls to specific Skype users, but depending on your SIP proxy/PBX capability (and your skills) you could setup click-dial or SIP uris, perhaps something like sip:skype-{username}@mypbx.com that ring through to the specified user on Skype.

Inbound calls from Skype work just the opposite of the diagram above, where SipToSis is configured to forward inbound calls arriving from Skype to a specified SIP address (if you’re connecting to Asterisk, that means an extension on the PBX).

As mentioned previously, my setup has just a single instance of Skype running and, therefore, provides only one Skype gateway channel, supporting just one Skype-to-SIP or SIP-to-Skype call at a time.  The SipToSis website has documentation about running multiple SipToSis and multiple Skype instances and even chaining multiple Skype accounts so you can advertise a single Skype Userid and still take multiple incoming Skype calls. Based on my experience with SipToSis so far, I have no doubt this stuff actually works.  However, it’s way more than I want to mess with – and Skype is a pretty heavy, resource intensive app – you would want to have a pretty beefy machine to run multiple instances of it. The SipToSis site has some information on performance on various hardware and OSes running multiple Skype users.

Overall, SipToSis is quite cool.  Skype charges $6.95 per month per channel for a SIP PBX interface that only allows inbound calls (from Skype) and calls from the PBX to SkypeOut (paid calls) – it does not support calls from the SIP PBX to Skype users, as SipToSis does.  Why Skype still doesn’t offer (preferably FREE) SIP support after all these years is just sad, but it’s nice to see third-parties like the SipToSis folks putting together practical solutions like this. The fact that it is cross-platform, and runs on Linux (and even under a VirtualBox VM), is very impressive and greatly improves the server-friendliness and cloud-based friendliness of SipToSis.

SipToSis is not quite for the mainstream, but it does work, and if one follows their excellent documentation, it’s not that hard to setup (assuming you already have the SIP side in place, or why would you want SipToSis in the first place).  If you already have a machine available, why not run SipToSis rather than pay Skype $6.95 per month for the same thing (whether you use it or not) – plus, if you want to call Skype users from SIP, SipToSis will do that, where Skype’s own product won’t.

A few notes:

• Since you must have Skype running to use SipToSis, and Skype is a GUI app, you need X Windows setup on the machine where you want to use SipToSis. This was not a problem for my simple test case, but it could be an issue on a real server setup running in a data center. The SipToSis docs talk about running Xvfb in the “background” but I didn’t test that trick.
• I had to remove the pulseaudio drivers on my Ubuntu machine to get this working
• Because I wanted to test it on a lightweight setup, I only gave the VM 512MB of RAM – and it works – impressive
• SipToSis needs Java – I used Java 6.

Over on the PhoneGnome blog, we look at where it is, and where to go from here:

http://www.phonegnome.com/blog/2009/12/02/phonegnome-participatory-marketing-challenge/

I guess I’ll find a way to make it through another day, somehow.

No calls unless you can get Wifi.

In general, incoming calls are impractical, even if you’re on wifi, since the Skype app has to be the one and only active app in order to receive calls.  If you’re doing something else on the phone, like browsing or checking email (or twitter), you cannot receive calls.

In test calls, I found the app unreliable even when all the conditions are met.  Trying to call the iPhone Skype from a PC, the calling side just continued to ring, even after I answered the call on the iPhone.  The Skype for iPhone app then seemed “frozen” where I couldn’t end the call or do anything except hit the big button.

When calls did connect (requires the iPhone to be connected via Wi-fi), the call quality was fine.

Not being able to make Skype calls except when connected to wi-fi is a pretty big limitation for me. Ironically, when a friend had to call their wife on Skype in Costa Rica recently, I had to let them use PhoneGnome and OpenSky on my iPhone to do so, because that was the only combination that worked on iPhone without wifi.

It looks like Skype For SIP supports calling from Skype into a SIP PBX, but not calling a Skype user (free call) from a SIP PBX.  It supports placing calls from SIP using SkypeOut, though, so we’ll see.

Skype For SIP now available in Beta

UPDATE: via Skype Journal, I see that “Skype for SIP” maps one Skype username to one IP address (SIP address?) – this is basically for “Skype Me” buttons and to let Skype users ring your PBX.

It also lets a businesses use SkypeOut for outgoing calls.  As Phil notes, Skypeout rates are higher than what businesses can already get from typical SIP termination providers, so I’m not sure what the win is there.

So, all in all, “Skype for SIP” is not the SIP interoperability everyone is asking for.  For better SIP/Skype interoperability, we still need to turn to third parties like OpenSky. On the other hand, while “Skype for SIP” appears to be a pretty small step toward Skype/SIP interoperability, it is at least a step, and we should give them some credit for that.

Today, Gizmo5 launched OpenSky, a SIP to Skype gateway. if you want to call a Skype user named echo123 you simply call SIP address  sip:echo123@opensky.gizmo5.com

The free version can be used by anyone for calls up to 5 minutes long. For longer calls, you need to use their paid service, which is $20 per year for individuals.

I guess this will test Skype’s claim that no one is asking for this service.

[in regards to Skype demanding that wireless companies open their networks], Skype’s actions do not mirror their words to the commission which diminishes credibility for Skype to demand openness.

…

It appears that when it is convenient for Skype’s business objectives Skype waves the flag of openness, at the same time conveniently ignoring competitors requests for openness.

Here, here.  You can read the entire letter over at Andy’s site.