Some people seem to have this burning desire to connect sip calls through to Skype users. I’m not sure why you wouldn’t just give them a native SIP/IAX client such as Zoiper Free, but for those who feel they can only have one voice communications client running on their desktop, and that client has to be Skype, then head on over to Tom Keating’s VoIP & Gadgets Blog:
Ok, so Skype, Inc. won’t let you dial into your Skype username using a SIP URI. Yeah, Skype claims their proprietary protocol is better than SIP. I’ve written about one SIP-to-Skype workaround, but still I hear the cries of VoIP fans everywhere clamoring for SIP access to Skype. So what’s a SIP-lovin’ VoIPster to do? Well, head on over to Scopezoom and check out the step-by-step guide that lets you deliver SIP calls directly to Skype. Essentially you can have a SIP DID number ring your Skype client. The workaround uses Net2Max.com’s One Click Contact number (1CC number) to make this possible.
Read the rest here: SIP to Skype Calls
Mr. Keating does mention that once you have this set up, you can then use PontiVoce, which “lets you turn your PC into a Skype gateway, allowing you to dial into your PC running Skype and then initiate an outbound call at SkypeOut rates.” So I guess that might be useful to some folks.
I could get a lot more excited about Skype if their protocol was open, and people could develop codecs that would allow (for example) Asterisk to interface with Skype without a bunch of hacks. In the alternative, I wish that Asterisk would start including a wide-bandwidth codec (something that would go out to 8,000 Hz at a minimum) for intra-Asterisk calls – obviously this couldn’t be used for calls to the PSTN, but it would be great if VoIP users had the option to use a wider bandwidth, instead of being limited to the (approximately) 3,000 Hz bandwidth that’s been the standard in the telephone industry since the days of vacuum tube amplification.
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