Peace In VoIPland: Bridging the SIP & Skype Kingdoms

opensky_logoLast year a number of people lead a charge to get Skype to open their network to greater interoperability with the open standards-based VoIP world. Gizmo5’s Michael Robertson, ever the instigator, was perhaps the most vocal of the group. Various others weighed in with opinions, including such notables as Dan York, Andy Abramson, Phil Wolff, Alec Saunder & others. At the time it sounded not unlike Mr Reagan telling Mr Gorbachev to “tear down these walls.”

Then word came out of Astricon that Digium had forged a partnership with Skype. The result would be “Skype For Asterisk,” a channel module allowing the integration of Skype into the leading open source IP-PBX platform. A great cheer went up all around VoIP-land. There was much rejoicing.

skypeforasterisklogo160px-723331Then time passes.

Details about “Skype For Asterisk” were initially sketchy, but have eventually become known. It will be a closed source channel module licensed on a per channel basis. Word is that it will eventually support both SMS and wideband audio capability, both of which are very interesting. There’s been a very hush-hush private beta program that remains underway even today. Rumors are that stage two of the beta, with broader community involvement, will be starting shortly.

More time will certainly pass before Skype For Asterisk is released into the wild.

Earlier this month Gizmo5 fired another shot in this saga by launching OpenSky, a SIP-to-Skype gateway service. It’s interesting. SIP clients can make calls to Skype users using simply formatted SIP URIs. There’s also an SMS based callback facility intended to allow cell phone users to place calls to Skype users. Calls under 5 minutes are free, but users can pay $20/year for greater access and then have calls up to 2 hours in duration.

Last weekend, as an experiment, I placed a call to Switzerland to VUC member Maxim. I called his Skype account from the Gizmo5 client. Then later called I called his Skype account from my Skype client, but he received the call using SIP-to-Sis to route the call into his Asterisk server. As Nerd Vittle’s Ward Mundy pointed out, SIP-to-Sis is the software behind Gizmo5’s OpenSky service.

In general call quality through the OpenSky gateway was marginal. However, call quality through the local instance of SIP-to-Sis was fairly good.

Of course, there have been a number companies offering Skype gateways. Some are cheapish software products, like PSGw which I actually purchased long ago. Of these many really interface to Skype at the client API level, essentially passing baseband audio streams to/from a running instance of the Skype client. This can work but often incurs a significant quality loss and adds latency to the calls.

Others companies offer hardware gateways like those from VoSky or Portech. There are even large, modular, multi-channel Skype gateways capable of connecting T-1/E-1 or greater capacities to the Skype realm. Such dedicated hardware is costly but it is available, and has been for some time.

I must admit that upon hearing the announcement of Skype For Asterisk I was caught up in the burst of enthusiasm of the Asterisk user community. As time passed I found myself wondering why. In truth I have little use for such a gateway today. I had more application for it 18-24 months ago, when many of my UK associates were making daily use of Skype. More recently we only use Skype for IM.

In surveying our operation today I find that for voice we have moved to using Polycom and Siemens hardware in conjunction with OnSIP. Our traveling staff occasionally use the Eyebeam or X-lite soft phones. It seems that we have, as a company, moved in the direction of using SIP over Skype. It wasn’t a policy decision. It was driven by the availability of very nice SIP hard phones and the adoption of a standards-based hosted IP-PBX.

One thing about working in a company dominated by engineers, when presented with an instrument that is truly a pleasure to use…we use it. We appreciate fine, dedicated hardware, be it Porsche or Polycom.

I’m still interested in Skype For Asterisk on a purely technical level, especially the wideband aspect. Will we be able to pass wideband calls between Asterisk & Skype without a major transcoding cost? Will call quality be sustained? Will Skype implement G.722 to minimize the transcoding requirement? Or will Asterisk get a codec_silk?

On the strategic level I find myself wondering who will benefit more; the Asterisk user community, or Skype? Some have pointed out that while Skype has a gazillion users that doesn’t translate linearly into revenues. Personal users are very different from SMB or enterprise users. Digium has certainly been getting traction in serious business application, taking ports from major PBX vendors with startling regularity.

Asterisk and Skype. Like a good wine and food the pairing they should be complementary. I wonder when we’ll be served?

And finally, with so many gateway options becoming available, will all the sabre-rattling about SIP vs proprietary finally stop?

  • http://www.teamforrest.com Fred Posner

    Great article! I too was very very excited at first, but truly also expected more by now. I have many dreams of integrating skype with phone systems… from call centers to IVR’s. I still remain optimistic… only the optimism is beginning to fain.

  • Carlos Talbot

    If you’re curious about wideband audio and Skype you might want to check out the new module that’s part of FreeSWITCH: http://wiki.freeswitch.org/wiki/Skypiax

    It works with G.722 and Skype’s 16khz codec. It’s included in the latest Windows MSI build for FreeSWITCH:
    http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries

    Carlos

  • http://www.mgraves.org mjgraves

    Which codec is that? The new SILK? Or something older?

  • Carlos Talbot

    Good question. I suppose it depends on which version of the skype client you are running on the Freeswitch host. The right person to ask would be Giovanni Maruzzelli, the author of the module. He initially wrote it for Asterisk and has since ported to Freeswitch. From what he wrote on the wiki page above I’m lead to believe it support’s Skype’s 16khz audio codec:

    “Skypiax works in FreeSWITCH (FS) on both Linux and Windows, at both 8khz and 16khz (Skype client has 16khz audio I/O). Skypiax works on Asterisk too, at 8khz, on Linux and Windows (through CygWin). ”

    Giovanni’s web site is http://www.celliax.org/

    Carlos

  • Carlos Talbot

    Here’s the response from the developer:

    http://jira.freeswitch.org/browse/MODSKYPIAX-26

    “the new SILK codec is standard on skype client 4.x for windows.
    If you use that skype client version as interface, you are using SILK (I believe), no intervention needed.
    With Mac and Linux skype clients, no SILK available. “

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  • http://digg.com/tech_news/Cheap_Voip_Service_from_the_top_VOIP_carrier_Phone_Power Cheap Voip Service

    Great article. The husband and I dumped landlines for voip about 2 years ago and haven’t looked back since. Our friends are cell phone only but we only have prepaid so this works out. Thanks!