I attended a great session from TechEd on Network Considerations given by Neil Deason. In it he discussed Audio, Video, and Application Sharing within CS14. When looking at audio traffic it’s important to review network conditions, acceptability quality of the call as well as optimal quality. Narrowband is typically used for traditional calling and it doesn’t span the human ear hearing spectrum but is good enough for the far end to understand our call and make error correction themselves.
Understanding the capacity that’s needed for these different experiences can be difficult to ascertain as Voice Quality itself is broken up into the areas of Network, core performance (performance of server h/w and s/w), gateways, and devices. Our goals for the network optimization are to provide voice based on acceptable and optimal quality. I need to have less than 10ms of jitter with a max of less than 80ms of jitter.
Packet loss might seem like a lot but this is against the entire experience and should not be looked at in consecutive packets as this would indeed impair call quality.
How much bandwidth is required is determined by the following:
Bandwidth Profiles
This chart gives us typical bandwidth and upper limits with and w/o Forward Error Correction (FEC). When planning you should work with typically numbers to start with.
These bandwidth profiles can be used with Call Admission Control (CAC) to provide group policies to be applied in logical sites (subnets). They can allow sessions to only go as big as the profile allows and setup rules for what to do when the users session exceeds that level such as reroute traffic or dropping the session. One cool thing that you can do is redirect the session to the Internet for overflow of traffic to avoid PSTN call charges and to support failover for things like video sessions.
This example used at technet describes a session where we are using a communicator to communicator call. We are using a RTAudio Wideband (no FEC) codec which as you saw above is about 35kbps. This is well within the CAC policy parameters above and the endpoints are intelligent and provide for a great wideband session. Now if we move the user outside the f/w to the Internet we will still use RT Audio WB(No FEC) if the network quality is good.
Now the Internet quality has degraded the session and we switch the call to RTAudioNB (+FEC) instead of RTaudioWB(+FEC) due to the CAC policy of 60kbps. .
The call continues and we now have error correction. Key to this discussion is that we will continue to check for call quality and adjust the codec as necessary. All in all very cool stuff. I would recommend sitting the entire session here if you can.
Jamie Stark recently posted that RCC will be included in the CS 14 release not only for existing customers but also for new ones.
“As an aside, I should explain that our decision to support “Click-to-call” for new customers in CS “14” is a change from previous plans. Until recently, we planned only to support “Click-to-call” for existing RCC customers. We made the change for several reasons, including some great work we’ve been able to accomplish on the updated management experience that will be an integrated part of CS “14”, being able to pull broader Click-to-Call support along with many more small but important enhancements into the release. The most important reason, though, is that we want to give those customers who, for whatever reason, require desktop control of existing phones, a better option than adding software from a PBX maker, like Cisco’s CUCIMOC, on every user desktop. This blog post about CUCIMOC outlines general issues to consider with plug-in offers; we think our “Click to call” functionality is better not only because of the reasons in that blog post, but also because our “Click to call” functionality works with nearly every PBX and IP-PBX model and version you own, not just those from whichever vendor is promoting their software bolt-on. With our “Click to call”, you provide a consistent user interface and can easily move users to Communications Server voice in the future without re-training users or re-imaging desktops.”
For more from Jamie please visit this post.
There are a lot of new features coming with Exchange SP1. Today I will provide a quick look into the OWA enhancements slated for SP1:
Tweaks to the OWA UI in SP1:
Reminders overhaul:
New OWA themes:
IM presence icons switched to squares to match the new CS14 jellybean overhaul.
Reply buttons overhauled:
Preview pane overhauled:
New alert pane on the toolbar:
Calendar overhaul:
Other OWA tweaks slated for SP1:
Solid story with Communication Server 14 – make it easier to expose CS14 presence within OWA – reduce complexities
embed pictures in an email
Publish your calendar on the internet
Import internet calendars (via .ics)
read IRM emails in OWA
Reading pane on the bottom or off option
Grab SP1 beta here to try out some of these new OWA features in your test labs (not production). We will blog about the other SP1 features coming in future blogs.
I was asked this by a university in Phoenix. Here is what I found from our Exchange product team:
Prerequisites to make pattern matching work well:
You need Exchange 2007 Sp1 with Rollup10 or Exchange 2007 Sp2 or Exchange 2010
How do I create a sample rule?
In Exchange Command Shell type:
PS] D:\Users\Administrator\Desktop>New-TransportRule testrule -SubjectOrBodyMatchesPatterns "\d\d\d-\d\d-\d\d\d\d\s|$"
The result of the rule firing is:
Asdkljf 349863-43-3454 fg
Not triggered – good
Asdf 568-45-45463477
Asdf 636-23-394867987 fgh
Asdfa asdfa 234-23-2345 rfy
Triggered - good
5675675747-56-34545645747
Not triggered- good
747-56-34545645747
More on transport rules here.
With all the openness (watch the video showing the new CS14 clients and new Silverlight conferencing client) the product team is revealing at TechEd 2010 this week around CS14, I figured I could now blog about a common question I get from customers.
This was a question from a university in the Chicago area (most diagrams courtesy of the CS product team). Note: this information is based on beta technologies and may change:
What does E-911 need to provide as a base requirement?
E911 needs to provide location with emergency calls (North American)
The dispatchers must know the civic/street address of the caller
Locations may need to be to specific building, floor, wing, office, etc.
What E-911 challenge does CS14 need to address?
CS14 needs to support the roaming nature of Communicator users
Inside the network (automatic or manual)
Outside the network (manual then automatic for frequent locations) Sample of entering a manual location: Sample of automatic frequent location outside the network. I type it in once and it always remembers that location based on subnet:
Outside the network (manual then automatic for frequent locations)
Sample of entering a manual location:
Sample of automatic frequent location outside the network. I type it in once and it always remembers that location based on subnet:
What standards is the CS14 E-911 architecture based on?
It is based on the National Emergency Number Association (NENA) i2 Enhanced 911 services architecture. More here.
What are some new features of E-911 with CS14?
Connecting to the appropriate authorities without having a PSTN gateway to each emergency network
Added enablement location to provide flexible deployment
“Network Sites” – can add in subnets or network ports tied to location
Users – can be associated with a floor, cube, address
Added a Location Information Server
Contains records of civic addresses associated with network identifiers
Renders locations to UC clients
Locations can be used independent of E911
Diagram of the location registration process:
Client registration
1-Provisioning
a) Populate LIS with network element and location records
b) SIP trunk connected to SP
c) Enable sites or users
2- Addresses are sent for validation
3-Report back valid/invalid addresses
Diagram of the client location process:
Client location
1- Client sends subnet information to registrar
2- Registrar returns LIS URI (and E911 Enablement data) during Registration. This is because Subnet 172.24.33.132 is enabled for E911
3- Client sends subnet to LIS – locations by subnet.
4- LIS does subnet/location match and returns the location in PIDF-LO format
Diagram of the end user dialing 911:
When 911 is dialed:
1-Client dials 911 – includes PIDF-LO in SIP INVITE
2-IM notification of emergency call, party, and location sent to security group (optional)
3-E911 call routed over SIP trunk
4-Routing Provider connects to appropriate PSAP
5-Voice path connected to security group (optional)
What partners are we working with for E911 call routing services?
Currently, we have announced two partners at VoiceCon:
911 Enable Intrado
Where can I get a better deep dive on the upcoming CS14 voice capabilities?
Read the CS14 VoiceCon RFP response here: XPS | PDF
I get asked this one quite a bit by my customers deploying OCS and Exchange UM to their faculty and staff. I found a partner, 4What Interactive, that provides some really nice end user CBT training for OCS R2 and Exchange UM OVA.
Click here to view some of 4What’s CBT Microsoft UC training. They also can develop custom training if required.
It has voice dubbed CBT training and can also be accessed via the Internet:
UM Outlook Voice Access training: