June 2011 Archives

Recently I decided to explore Cisco Unified Communication Manager's Mobile Connect feature. This used to be called Single Number Reach and is exactly what you would expect: one number rings multiple phones--your VoIP set and an off-system number. It is straightforward to set up; follow this helpful guide on the Cisco Learning Network.

The nice thing about Mobile Connect is that it isn't simply a multi-ring scheme; rather, it's effectively a shared line appearance with your cell phone or other remote number. Communications Manager maintains supervision of the line, even if you take the call on your remote line (mobile phone), which allows you to easily switch between the remote and the desk phone. I believe, though I have not tested it, that this configuration would also allow the Mobile Connect line to participate in a hunt group.

In CUCM 7.x, you can configure an on-hook screen Mobility softkey to enable or disable Mobile Connect. This way, it does not have to be an always-on feature.

Enabling a Mobile Connect number consumes two additional device license units (DLUs) if you are licensing devices a la carte. The functionality is very nice and probably worth the licensing cost for those who want to use it.

Has anyone implemented a video-capable SIP-H.323 gateway using free, open-source software?

If so, please comment.

Asterisk, FreeSWITCH and YATE all have some ability to connect SIP and H.323 endpoints to one another. Asterisk acts as a back-to-back user agent (B2BUA) and the other two act as proxies. All can switch audio calls. Video seems to be another story.

In theory, it should be straightforward. The difficult part is translating the signaling; the media streams are the same. Thus the interworking component (PBX, switch, or proxy) should be able to translate signaling and then proxy the media between the endpoints. If video is just another media stream, why doesn't it work just the same as an audio-only call?

I've tested YATE (built-in modules), Asterisk's chan_ooh323, a custom chan_ooh323 for Asterisk 1.4 that specifically enables video (but crashes/disconnects, and is definitely not supported unless you buy the unnamed company's video IVR product, which I haven't), FreeSWITCH's mod_h323 and mod_opal. I never really got the FreeSWITCH mods working stably. YATE and Asterisk worked fine in audio mode. Neither would translate video (H.263 protocol, nothing fancy).

Additional reading: RFC 4123, Session Initiation Protocol (SIP)-H.323 Interworking Requirements

Three years ago I wrote a blog post here titled Greener IT through bicycling and VoIP. I believe little has changed since then, except that a greater number of people are thinking "green." Indeed, the bicycle paths from my house to work are seeing more traffic during the morning commute, and the bikes or riders are laden with bags that seem to indicate they're riding to work and not just for fun.

Today was World IPv6 Day, the day when many web sites and Internet service providers were supposed to test IPv6 for 24 hours. I doubt most people noticed. My workstation is IPv6-connected and I kept a connection tracker up for most of the day to see when IPv6 was used. I was disappointed to see that Akamai did not go over IPv6 today, at least from here. Google and Facebook (which uses "face:b00c" in its IPv6 addresses) showed up on the list. A number of Penn State web sites showed up, but they had been IPv6-enabled long before today.

In other words, ho-hum. From my perspective, World IPv6 Day was a blip. The IT world just needs to keep pressing on diligently in deploying IPv6.

I played with Asterisk 1.8 a little, to see what needs to be done to enable IPv6. Just bind it to the v6 stack with bindaddr=:: in sip.conf (or sip_general_custom.conf if you use FreePBX). That's it; you're ready to communicate over IPv6.

Check out the free IPv6 Certification from he.net. You get an IPv6 Sage T-shirt when you have completed all of the exercises that test your IPv6-configuration skills.

There has been a lot of talk about Skype connectivity in the VoIP blogs lately. Digium announced that they will no longer be selling their Skype channel driver, and this news rekindled interest in free/DIY methods for connecting Skype to Asterisk. Nerd Vittles has a good writeup, specific to the PBX-in-a-Flash/Incredible PBX implementation. His method relies on the SipToSis gateway application (read the SipToSis how-to for more generic setup instructions).

If you are only wanting to receive Skype calls, you can transfer them to your PBX via Tropo developer account, free.

I decided to add Skype to my home PBX last weekend, and chose FreeSWITCH's mod_skypopen as the connector.

Why?

• The PBX is Asterisk 1.4, and I'm holding steady on this version until 1.8 becomes a bit more stable. Thus, because I want to use Google Voice,
• FreeSWITCH is already in place, on the same machine, as a gateway to Google Voice. It has been working well in this role.
• mod_skypopen requires only the FreeSWITCH module and Skype. No extra connector software is required.
• It works and was easy to install!

Asterisk 1.4 continues to be a rock-solid choice and is supported (security patches will be provided) for another year. For me, the only killer feature that 1.8 provides is Google Voice connectivity. There are bunch of other new features too, but I don't need them. Stick with that which works, until you need the features or the support is gone.

Back to mod_skypopen. There are basically two ways to install this to FreeSWITCH. The easy way, and the hard way.

Today, the easy way. If you visit the FreeSWITCH wiki page on mod_skypopen, you'll find some notes on building and installing the module. If you're running a fairly standard install of FreeSWITCH that utilizes the autoload_configs structure, you can follow through the wiki instructions to build the module and the fake audio driver and then run the Perl installer found in the source directory to automatically build out your configs, configure the Skype client(s) and set up a shell script to get everything going at startup.

If, like me, you have set up FreeSWITCH only to act as a feature gateway for Asterisk, and have aggressively minimized the configuration, you'll want to avoid the installer script and perform the necessary steps by hand. More on that in the next posting: "the hard way." (Not really that hard.)