Category: SIP
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Skype tears down more walls with “Skype For SIP”
Continue Reading: Skype tears down more walls with “Skype For SIP”NOTE: I have a few updates to the post that I am putting at the bottom of the text.
Would you like Skype users to be able to call your business’ phone system? Would you like to connect your phone system to Skype’s network and make use of their cheap calling rates? If you have an IP-PBX or other call server that supports the Session Initiation Protocol (SIP), you may now have those options.
For a company that only a bit over a year ago was saying that customers weren’t asking for interconnection, today Skype has done something rather dramatic and lowered their walls a bit more with the announcement of the beta program of “Skype For SIP“. With this announcement from the “Skype For Business” group, companies with SIP-enabled phone systems will be able to receive calls from Skype users – and make calls using Skype’s network at Skype rates. The news release (and blog post announcement) highlights these four aspects:
- Receive and manage inbound calls from Skype users worldwide on SIP-enabled PBX systems; connecting the company Web site to the PBX system via click-to-call
- Place calls with Skype to landlines and mobile phones worldwide from any connected…
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IETF 74 starts next week in San Francisco…
Continue Reading: IETF 74 starts next week in San Francisco…The 74th meeting of the Internet Engineering Task Force (IETF) starts Monday morning out in San Francisco. As usual there is a packed agenda with a lot of great discussions going on. This one is particularly interesting for those of us involved in the “Real-time Applications and Infrastructure (RAI)” area – which is all the various working groups related to SIP and other real-time communications protocols – as there are some proposals moving forward to rather fundamentally restructure the ways in which SIP-related work moves through the IETF. I expect there will be many involved conversations going on out there next week.As much as I would like to be there, I won’t be physically out at IETF 74. It’s not my new role at Voxeo keeping me away, but rather this… oh… wee minor little detail that my wife is now five weeks from giving birth to our second child! 🙂 At this stage of things I’m severely limiting my travel – and flights across the country are definitely out.
Instead I’ll be participating remotely, listening to the audio streams and joining in the Jabber chat rooms. Probably writing about some of it over on the “Speaking…
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The Park Bench Manifesto – text coming soon, video and slides now up
Continue Reading: The Park Bench Manifesto – text coming soon, video and slides now upThis week out at the Emerging Communications Conference in San Francisco, I gave a 10-minute talk called “The Park Bench Manifesto: Why We Want To Kill Off The PSTN”. In the talk, I mentioned that the text would be available here soon… And it will be.
In the meantime though, I have put up both the video and the 54-slide deck over on <a href=”http://blogs.Voxeo.com/ett/”>blogs.voxeo.com/ett/</a>
More soon….. (need to fly home…)
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FWD launches “SIP to SIP” directory of apps that work with SIP…
Continue Reading: FWD launches “SIP to SIP” directory of apps that work with SIP…If you have a new SIP service or application, how can you find other services to which you can directly connect via SIP? That’s the idea behind the new “SIP to SIP” directory launched by the folks at FWD and now available at www.siptosip.net. From the main page:Why SIP to SIP VoIP?
SIPtoSIP lists applications and content that require SIP (Session Initiation Protocol) enabled devices on both ends of the connection. Realizing the promise of VoIP requires expanding real-time communication options beyond the functions already available with traditional telephones or cell phones. The ability of SIP based VoIP to support HDVoice, video, and click-to-connect requires SIP devices on both ends of the connection. Send suggestions for corrections and additional listings to Daniel Berninger at dan at danielberninger.com.
The directory is very obviously new and only has a few entries on the various pages:
As is noted, Daniel Berninger is looking for people to email him suggestions.
I do applaud the FWD folks for looking at another way to promote the further building of SIP interconnections and so I wish them well with this directory. I’d note, though, that the VoIP-Info.org wiki does already contain a great amount…
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Define “VoIP” – and then we can debate whether it is dead!
Continue Reading: Define “VoIP” – and then we can debate whether it is dead!There is a fundamental problem with the “VoIP is dead” debate continuing to rage across the VoIP/communications part of the blogosphere (see Alec Saunders part 1 and part 2, Jon Arnold, Andy Abramson, Ken Camp, Jeff Pulver part 1 and part 2, Om Malik, Shidan Gouran, Ted Wallingford, Dameon Welch-Abernathy (PhoneBoy), Rich Tehrani and a zillion others…)Aswath Rao and Luca Filigheddu came closest to the mark in their posts. The fundamental problem with this entire debate is simply this:
Define “VoIP”?
As I discussed in an Emerging Tech Talk video podcast I put up this morning, there are a range of definitions you could give to “VoIP”, including, but not limited to, the following:
- The underlying infrastructure, a.k.a. the “plumbing” – the mechanisms, protocols, etc. that are used for the transport of voice/video/etc. over IP. Things like SIP, H.323, RTP, various codecs, etc.
- Consumer “PSTN line replacement” services – Offerings like those of Vonage and so many others where the basic idea is that you can get cheaper telephone charges by going over the Internet and getting rid of your local landline. Also called “pure play” VoIP by some or “VoIP arbitrage” by others.
- Computer-to-computer/softphone offerings, often coming from the…
- The underlying infrastructure, a.k.a. the “plumbing” – the mechanisms, protocols, etc. that are used for the transport of voice/video/etc. over IP. Things like SIP, H.323, RTP, various codecs, etc.
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Slides from my ITEXPO security talk – SIP Trunking and Security in an Enterprise Network
Continue Reading: Slides from my ITEXPO security talk – SIP Trunking and Security in an Enterprise NetworkEarlier this month out at ITEXPO in Los Angeles, I participated in the Ingate SIP Trunking seminars as I have been doing for the last year or so. My talk was “SIP Trunking and Security in an Enterprise Network“. The slides are available for viewing or download from my SlideShare account and I’ll also embed them here in this post.I did record the presentation in both audio and video and hope to be making that available as a Blue Box podcast some time soon. I’ll then sync the slides to the audio. Meanwhile… enjoy the slides! SIP Trunking & Security in an Enterprise NetworkView SlideShare presentation or Upload your own. (tags: telephony danyork)
Technorati Tags: voip, voip security, sip, sip security, voice, security, dan york, itexpo, commdev, voipsa
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Clarifying how Asterisk could possibly be used as a Skype-to-SIP gateway
Continue Reading: Clarifying how Asterisk could possibly be used as a Skype-to-SIP gatewayAfter my post yesterday about “Skype for Asterisk” (and the update post) and the potential it allows for SIP interoperability via Asterisk, I’ve received a few comments that seemed to interpret what I wrote as somehow indicating that the Skype announcement somehow meant that there was new “Skype to SIP” functionality in the “Skype for Asterisk” announcement.
Just to be clear, there isn’t any new “Skype to SIP” functionality in the “Skype for Asterisk” piece announced yesterday by Digium and Skype. None.
It is purely a commercially-licensed software module (which most of us speculate will be a binary software module, i.e. we won’t be able to actually see the code) that provides two-way connectivity from Asterisk to and from the Skype cloud. Skype users can call into an Asterisk system. Users connected to an Asterisk system can call out to Skype users. Users on the Asterisk system can also call to the PSTN (via what was called “SkypeOut”) and receive calls from the PSTN (via what was called “SkypeIn”).
That’s it. That was the announcement yesterday. Period. End-of-story.
However, the point I was making in my post yesterday was this announcement has the potential to turn Asterisk into a two-way…
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More on how “Skype For Asterisk” actually works…
Continue Reading: More on how “Skype For Asterisk” actually works…As per usual, Tom Keating gets us more details on the “Skype For Asterisk” beta program I just wrote about… in his update post, Tom explains how it will work:Well, on an inbound call to your Skype username, both your Skype desktop client rings (if running) and your Asterisk IP phone rings. You can take the call using either your PC’s Skype software or your IP phone. Similarly, if someone calls your SkypeIn number, both will ring. Further, if someone dials your corporate auto-attendant, and then enters an extension number, it will still ring both your Skype client and your regular IP phone.
His post discusses how you can assign Skype names to Asterisk call queues and then includes this intriguing text:
When asked how Skype IP-PBX gateway appliances are affected by this announcement, Stefan Öberg VP & GM Telecom for Skype said, “The appliances that are out there now have built their solutions on standard Linux client. They’ve used the public API on that and basically are running many instances of Skype Linux client. Obviously, that’s not the way the Linux client was meant to be implemented. So those solutions are not scalable or reliable to the extend that…
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Does “Skype for Asterisk” tear down some of Skype’s walls? (and allow SIP-to-Skype?)
Continue Reading: Does “Skype for Asterisk” tear down some of Skype’s walls? (and allow SIP-to-Skype?)Does today’s announcement of a beta version of “Skype for Asterisk” signal a way to tear down some of Skype’s walls? And does this move Skype along toward better SIP interoperability?The announcement happened out at Astricon today and TMC’s Tom Keating had one of the first posts about it – updated with info from TMC reporters who are at Astricon. Both the Digium news release and the Skype blog post highlight these four points that Asterisk users will be able to do:
- Make, receive and transfer Skype calls with multiple Skype names from within Asterisk phone systems, using existing hardware.
- Complement existing Asterisk services with low Skype global rates (as low as 1.7€¢ / 2.1US¢ per minute to more than 35 countries worldwide).
- Save money on inbound calling solutions such as free click-to-call from a website, as well as receive inbound calling from the PSTN throughcreate virtual offices all over world using Skype’s online numbers.
- Manage Skype calls using Asterisk applications such as call routing, conferencing, phone menus and voicemail.
I want to focus on one part of the first bullet. Recall that in my last post about Skype and SIP interoperability I talked about how Skype currently…
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Do the IM protocol wars even matter? Adium and the continued *client* unification of IM…
Continue Reading: Do the IM protocol wars even matter? Adium and the continued *client* unification of IM…Do you care any more about zillion different IM services? Do you care about the IM protocol wars that have plagued the usage of IM for the last years?Odds are that if you are an IM user like me, you probably don’t. Why not? Simple… we’ve unified the IM services on the client side and basically stopped caring about the various services and protocols.
I was reminded of this fact this morning when I received a message saying that an update was available for Adium on my Mac that solved a really annoying disconnection problem with Yahoo!Messenger. (And if you are a Yahoo IM user, you really need to get the 1.3.2b1 beta.)
[NOTE: An equivalent to Adium for Windows or Unix/Linux users is Pidgin.]
Somewhat ironically, there was a discussion going on in a Skype groupchat in which I participate about the various IM protocols and whether anyone really used GTalk, etc. Since I was updating Adium at the time, I took a moment to look at all the different protocols that Adium now supports… as seen in the screenshot on the right side of this post. If I look at my own usage, I use Adium to…