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Day 2.                              (Day1 blog is here)

Today’s focus is all around Voice Design. First cab off the ranks is the telephony lingo & then dialling requirements.

Microsoft gave a really good description of E.164 -- it's a format for manipulating phone numbers to become globally unique. When designing an OCS infrastructure substantial time needs to go into a plan for normalising dialled numbers (and extensions) into E.164 format so that the call commonly understood and can be routed outside of the organisation to a PSTN.

Exchange 2007 RTM didn't support E.164 dial plans, but SP1 does. 

<http://technet.microsoft.com/en-us/library/bb803637.aspx and http://technet.microsoft.com/en-us/library/bb676323.aspx>

The new codec with OCS 2007 is RTAudio. This codec can be both wide or narrow band (WB/NB), and the WB codec used on average 45Kbps bandwidth per channel. This figure doesn't take into account the various network overhead components. For planning it’s more accurate to use 57Kbps as your average planning number for each one-way channel -- 57Kbps send and 57Kbps receive for each endpoint. While this may seem like a lot, in a typical conference 80% of the time only 1 person speaks (57kbps), 7% of the time 2 people speak (74kbps) and 13% of the time nobody speaks (almost 0kbps – empty RTP). However, be sure to plan on 57kbps, especially if you have different upstream/downstream capabilities! The estimations for signalling traffic are not included in the figures we were given, hence this will affect the estimation in a constrained network environment.   <http://forums.microsoft.com/unifiedcommunications/ShowPost.aspx?PostID=2697675&SiteID=57> -- this is not an “official“ guide, but the calculations are there and include the overhead so check it out; the official planning guide still has the numbers that don’t account for overhead, and the overhead can change based on network conditions. These numbers can be reduced further if you use silence suppression, however this can affect the end user experience.

E.164 is an ITU standard for normalizing phone number, as mentioned before. However, RFC 3966 defines the tel: URI scheme, which is basically a superset of E.164. E.164 applies only to public numbers; RFC 3966 applies to private numbers as well. Anytime you see a tel: prefix on a number in OCS, you're dealing with RFC 3966. <http://www.ietf.org/rfc/rfc3966.txt, http://en.wikipedia.org/wiki/Telephone_number, and http://en.wikipedia.org/wiki/URI_scheme>

OCS 2007 normalization rules use .NET regular expressions, the OCS 2007 Resource Kit provides some great tools for this job – the main gun in the arsenal is the Enterprise Voice Route Helper, which includes a normalization tool that helps you build your regular expressions. It also allows you to input other organisations configuration to assist them remotely with normalisation and routing issues. *very nice* <http://www.microsoft.com/downloads/details.aspx?FamilyID=b9bf4f71-fb0b-4de9-962f-c56b70a8aecd&displaylang=en>

PSTN breakout is a cool process -- use VoIP across your network, then break out the call to the PSTN at your location closest to the destination so you reduce or eliminate long-distance costs. OCS achieves this through the use of strategically placed mediation servers. However, in some cases the countries laws prevent you from doing so -- in India, for example, you must be registered as an ISP in order to switch calls back onto the PSTN from VoIP; point to point Communicator calls are just fine. <http://www.microsoft.com/downloads/details.aspx?FamilyId=24E72DAC-2B26-4F43-BBA2-60488F2ACA8D&displaylang=en and http://www.ilocus.com/2008/01/bsnls_voip_will_kill_the_grey.html>

This fact reminds me of something....If your OCS 2007 infrastructure spans regions be very clear on the laws and regulations that are enforced in each country – Do your research !

The labs that were constructed allowed us to further familiarise ourselves with Dial Plans, and call routing. Also we get to look at how to configure a dial plan to break-out to a PSTN – which in our case is just an analog FXS port on a Audiocode MP-114 gateway. *easy peasy*

For Day 3 blogs go here -> Day 3 - UC Voice Ignite conference – OCS 2007 Voice Architecture, Topologies and Infrastructure