Skype to Kill Asterisk VoIP Integration
There are reports that Skype will no longer license Asterisk integration into Skype after June 26 2011 (confirmed by PCWorld) Asterisk is one of the most popular open source software platforms for Voice over IP (VoIP) communications. Asterisk runs on commodity computers or purpose built hardware acting as a Private Branch Exchange (PBX) which serves as a centralized phone controller for business phone systems.
Skype for Asterisk was a software component developed by Digium (creator of Asterisk) and it allowed Asterisk clients to communicate natively with Skype clients without the need to go through the legacy Public Switched Telephone Network (PSTN). Asterisk PBX systems supported a number of standardized VoIP clients using the IETF standards IAX2 and SIP. Skype for Asterisk essentially allowed standard VoIP clients like IAX2 and SIP to talk to proprietary Skype clients, but if the reports are true, that integration will end this June. The result is that Skype clients will have to go through the Skype-Out service, which uses the legacy PSTN system. That means users will have to pay a phone toll or have a Skype-Out subscription to communicate with standards based VoIP clients.
It is unclear if this decision came as a result of Microsoft’s acquisition of Skype or whether the decision was already in the works, but Microsoft had previously shunned standardized VoIP technologies in their business communication products such as the ubiquitous G.722 wideband codec. This will result in more fragmentation in the VoIP market where VoIP phones from one vendor won’t speak to VoIP phones from another vendor. Even VoIP phones that speak the same language can’t communicate with each other because they sit behind separate corporate firewalls and Network Address Translations (NAT). The same type of protocol and firewall fragmentation afflicts the video conferencing telepresence market.
There are ways around these firewall barriers, but those workarounds aren’t typically deployed because they require additional configuration and coordination between IT engineers and managers from different businesses. Many corporations simply do not want to open up their VoIP phones up to the entire Internet. Skype managed to become one of the most successful VoIP platforms because it avoided client fragmentation by acting as the sole controller of all Skype VoIP clients, whereas SIP solutions gave everyone the freedom to set up their own VoIP controllers, but that freedom came with the price of fragmentation. All Skype clients talk to other Skype clients by default whereas most SIP clients don’t talk to other SIP clients without a lot of additional IT effort. This decision to kill Skype and Asterisk integration will likely hurt Asterisk SIP and IAX2 users, but it is a double edged sword that also reduces the network size and utility of Skype.
I wrote a brief story on this as well. (http://www.freeswitch.org/node/325) The PC World article I referenced in my update offers an explanation that is both plausible and does not require the tin foil hat. I still suspect that Microsoft had some influence over this decision but I’m not overly concerned at this point.
FTR, my first-ever blog post on my mostly dead blog was about Skype for Asterisk. (http://bit.ly/2mesy8) I don’t think we’ve lost much with Sfa getting the plug pulled. The real issue for me is the implication that maybe – just MAYBE – other FLOSSy or non-Microsoft Skype interfaces will suffer a similar fate. I hope not. One of our intrepid FreeSWITCH community members created SkypOpen which is the “hack” I spoke about in my blog. The good thing about it is that it lets you have the native Skype client doing all the encoding/decoding so you get the SILK codec and the higher quality audio (if you have the requisite bandwidth).
Anyway, thanks for blogging on this issue. Time will tell if this was “just a business decision” or if it was a harbinger of things to come for non-MS Skype services.
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Skype has terminated its partnership with Digium, effectively killing Skype for Asterisk, which integrated the VOIP service with the open-source telephony platform.
Skype decided not to renew its agreement with Digium, which allowed Asterisk, the open-source telephony system, to be integrated with the Skype service, Digium said May 25 in a letter to users. Digium is behind most of the work on Asterisk and sells commercial products based on the platform. Skype for Asterisk uses some proprietary code from Digium to connect the two products.
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