I neglected to point out in my morning Dell Picks Fonality VoIP System post that I first discovered Dell got into the VoIP game with their late 2007 announcement that they would be offering Microsoft OCS 2007 pre-bundled on Dell servers. Thus, the Fonality/Dell deal marks the 2nd VoIP platform that Dell is offering. I noticed on Dell's VoIP page that they are also carrying Nortel as well. Microsoft OCS is noticeably absent. Interesting... I even used the Dell search for OCS and came up empty. Though OCS 2007 needs to work in conjunction with a PBX so it might be in a different section on Dell's site. In fact, Nortel has worked very closely with Microsoft in tightly integrating OCS with Nortel IP-PBXs. Still, I'm surprised the Dell VoIP page makes no mention of Microsoft OCS. That could be a major selling point when choosing between Fonality or the Nortel system.
Of course, you can get OCS to work with Fonality, an Asterisk-based PBX as well. As I wrote previously, unfortunately, Asterisk-based PBXs only supports SIP over UDP and OCS 2007 & Exchange 2007 Unified Messaging (UM) requires SIP over TCP. There is one unsupported patch to enable SIP over TCP on Asterisk, however. Another workaround is to use sipX, an open-source SIP proxy which handles both SIP/TCP and SIP/UDP and acts as a gateway between Asterisk and Exchange UM. You can use OpenSER as a SIP UDP to SIP TCP gateway as well. Can't really blame Microsoft for this lack of Asterisk integration since usually SIP uses TCP and only the RTP audio stream uses UDP. In fact, according to the voip-info wiki, "Making TCP mandatory for the UA is a substantial change from RFC 2543. It has arisen out of the need to handle larger messages, which MUST use TCP."
So like I've said before, the Asterisk community needs to get on the ball and add SIP over TCP functionality.
Of course, you can get OCS to work with Fonality, an Asterisk-based PBX as well. As I wrote previously, unfortunately, Asterisk-based PBXs only supports SIP over UDP and OCS 2007 & Exchange 2007 Unified Messaging (UM) requires SIP over TCP. There is one unsupported patch to enable SIP over TCP on Asterisk, however. Another workaround is to use sipX, an open-source SIP proxy which handles both SIP/TCP and SIP/UDP and acts as a gateway between Asterisk and Exchange UM. You can use OpenSER as a SIP UDP to SIP TCP gateway as well. Can't really blame Microsoft for this lack of Asterisk integration since usually SIP uses TCP and only the RTP audio stream uses UDP. In fact, according to the voip-info wiki, "Making TCP mandatory for the UA is a substantial change from RFC 2543. It has arisen out of the need to handle larger messages, which MUST use TCP."
So like I've said before, the Asterisk community needs to get on the ball and add SIP over TCP functionality.
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"Can't really blame Microsoft for this lack of Asterisk integration since usually SIP uses TCP and only the RTP audio stream uses UDP. "
Really? If "usually" means about .01% of SIP calls, then maybe.
The fact it is not normal is pretty clear by your next sentences:
'In fact, according to the voip-info wiki, "Making TCP mandatory for the UA is a substantial change from RFC 2543. It has arisen out of the need to handle larger messages, which MUST use TCP."'
Not saying it is a bad idea to go to TCP, but your contention that TCP is something common, let alone normal, is a huge stretch.
Using TCP for SIP is not common at all, I've never ever seen it in the field. It runs over UDP since it that was how it was designed, for some very good reasons (you don't want three way handshakes for real time signalling such as answering a call if you can avoid it), but you have to switch if your messages get too large (which they don't when running "normal" telephony but may do when there is other stuff involved).
Fonality has some good offerings, including their PBXtra package which is pretty reasonably priced. It's only a matter of time until we are all using VoIP, so I think we are going to start to see a lot of companies making the switch.