One of the great things about OCS is the development capability that is available with both the client and server API’s. Wave 14 UCMA 3.0, which is a server side managed API, is now taking things to the next level and making it easier to develop with more functionality. This of course requires custom development but when compared to an off the shelf option it can be a significant cost saving. Just recently I had a customer solve a call center recording requirement using the previous version of UCMA which ended up saving them a significant amount of money compared to an off the self-purchase.
Below are some of things that UCMA 3.0 can enable for the contact center and other applications:
Alerts and notifications systems
Multi-channel
Self Service
Query/Response BOTs
Interactive Voice Response
Contact Center / Helpdesk
Multi-channel ACD
Presence aware
Supervisor functions, such as listen in, whisper
Recording
Conferencing Portals
Reach gateways
Silverlight (Web and Mobile)
Single Number Reach/Click to Call
Enough of me talking about it. Here is a video with Albert Kooiman from the OCS product group to talk about the new features coming in UCMA 3.0 for contact center.
Comments welcomed.
VoIPNorm
Device Review: Savi W430-M “DECT Goes Mobile”
While most mobile devices are Bluetooth, I do have something of a thing for DECT devices. The Savi W430-M is a DECT 6.0 device that can easily go on the road with you. I have struggled with Bluetooth devices over the years and as such I do not use a wireless headset for my cell phone. I either use the speaker phone function or phone to the ear. The fact I am not a heavy user of the voice function on my cell means that this isn’t to much of a problem for me.
Not to say that I am not on the phone much because I am. It’s just that it is usually a Communicator call rather than a cell or deskphone based call. On average I only use my cell phone 40 minutes a month to make calls. So having a roaming headset device that can pair to my cell phone and my laptop is not a high priority to me, I would rather have a high quality wireless headset that I can take on the road. Not to say there isn’t high quality headsets on the market that use Bluetooth but I have always found pairing with dongles and everything else that goes along with them a pain.
So when I received the Plantronics Savi W430-M in the mail I was pleasantly surprised and very anxious to try it out. The first day in the office was going to be it. I don’t have a cube as such at the Microsoft office but a shared space that is first come first serve. So to me being in the office is as good as being on the road. I have only what I can carry. The first thing I noticed when I plugged in the W430 is the LED indicator around the edge of the dongle device to let me know that the dongle and headset are paired and when I am on a call. Now being DECT there was no need to manually pair the devices, which is fine by me and once the headset began to charge they paired immediately and the dongle turned from red to solid green. When on a call it flashes green.
Next was call quality. I have so far made about 10 calls with the device and the call quality has been excellent. I actually had several people comment on it with no prompting from me. I have not tried out the 300ft distance capability which is a one of the bonuses of DECT but I have no doubt that I will.
The web page shows the device in the stand for charging but I pretty much don’t use it. I stick with the USB cable charger that can come with me on the road and fits nicely with the dongle and headset in the carry case. Not to say that this is a delicate device but if I was spending $250+ on a wireless headset I would like to protect it a little. While it fits nicely into the bag for travel, it offers no protection or padding. A padded bag would have been nice to protect the device.
Overall this device is solid. The earpiece wears just like the rest of the Plantronics devices series that use the same form factor and is extremely comfortable. Overall this is a great road warrior device with great call quality and easy to use. I will forever be a fan of DECT 6.0.
Comments welcomed.
VoIPNorm
Not to say that I am not on the phone much because I am. It’s just that it is usually a Communicator call rather than a cell or deskphone based call. On average I only use my cell phone 40 minutes a month to make calls. So having a roaming headset device that can pair to my cell phone and my laptop is not a high priority to me, I would rather have a high quality wireless headset that I can take on the road. Not to say there isn’t high quality headsets on the market that use Bluetooth but I have always found pairing with dongles and everything else that goes along with them a pain.
So when I received the Plantronics Savi W430-M in the mail I was pleasantly surprised and very anxious to try it out. The first day in the office was going to be it. I don’t have a cube as such at the Microsoft office but a shared space that is first come first serve. So to me being in the office is as good as being on the road. I have only what I can carry. The first thing I noticed when I plugged in the W430 is the LED indicator around the edge of the dongle device to let me know that the dongle and headset are paired and when I am on a call. Now being DECT there was no need to manually pair the devices, which is fine by me and once the headset began to charge they paired immediately and the dongle turned from red to solid green. When on a call it flashes green.
Next was call quality. I have so far made about 10 calls with the device and the call quality has been excellent. I actually had several people comment on it with no prompting from me. I have not tried out the 300ft distance capability which is a one of the bonuses of DECT but I have no doubt that I will.
The web page shows the device in the stand for charging but I pretty much don’t use it. I stick with the USB cable charger that can come with me on the road and fits nicely with the dongle and headset in the carry case. Not to say that this is a delicate device but if I was spending $250+ on a wireless headset I would like to protect it a little. While it fits nicely into the bag for travel, it offers no protection or padding. A padded bag would have been nice to protect the device.
Overall this device is solid. The earpiece wears just like the rest of the Plantronics devices series that use the same form factor and is extremely comfortable. Overall this is a great road warrior device with great call quality and easy to use. I will forever be a fan of DECT 6.0.
Comments welcomed.
VoIPNorm
New Avaya and Tandberg Interop Documents
Tandberg VCS X5 to OCS 2007 R1 and R2:
http://www.tandberg.com/collateral/documentation/Deployment_Guides/TANDBERG%20VCS%20Deployment%20Guide%20-%20Microsoft%20OCS%202007%20(R1%20and%20R2)%20and%20VCS%20Control%20(X5).pdf
Configuring SIP Trunking between Microsoft Office Communications Server 2007 R2, Avaya Aura™ Session Manager and Avaya Aura™ Communication Manager:
https://devconnect.avaya.com/public/download/interop/OCSR2-SM-CM.pdf
http://www.tandberg.com/collateral/documentation/Deployment_Guides/TANDBERG%20VCS%20Deployment%20Guide%20-%20Microsoft%20OCS%202007%20(R1%20and%20R2)%20and%20VCS%20Control%20(X5).pdf
Configuring SIP Trunking between Microsoft Office Communications Server 2007 R2, Avaya Aura™ Session Manager and Avaya Aura™ Communication Manager:
https://devconnect.avaya.com/public/download/interop/OCSR2-SM-CM.pdf
Communications Server 14: Standard Edition Disaster Recovery
In my last VoIPNorm myth busters I spoke briefly about the ability to have a disaster recovery scenario using CS 14 Standard Edition. In this post I am going to break this down a little further. This is a great option not just for small business but any business looking to deploy local resources with disaster recovery for voice features as well as some other peer to peer features.
The main concept with the new disaster recovery option on CS 14 is the ability to separate the SIP registrar allowing for a primary and secondary registrar. This concept has been talked about somewhat with survivable branch appliance with an Enterprise Pool but it is also an option with dual Standard Edition servers.
When Communicator 14 registers it receives the primary and secondary register in-band in the SIP signaling along with other information about meeting policy’s etc. Although there has been some mention of SRV DNS records it is defined in topology builder (another blog post for another time)as the backup registrar pool and passed via in-band signaling to the client when logging on to Communication Server.
This feature is really about keeping voice up and available in the case of a data center failure. Although I have showen the concept of two remotely located datacenters this doesn’t have to be the case. In the case of a small enterprise this could be the same rack for all practical purposes in the case of wanting to provide some hardware resiliency to voice.
Features during DR:
What’s great about this solution is its simplistic nature. No load balancing required. It also provides the ability to continue doing peer to peer sessions for datasharing, IM and video. When in failover mode the user receives a clear indication that there is an issue and to expect limited services. A big red banner is displayed across the client until services are restored. If you want to restore full services quickly to your users you can migrate your users on to the still operational SE server and they will be back up and working with full UC features.
As you can see I have also placed the mediation server functionality on the SE server so with the addition of two gateways you have a full UC solution of 5000 users with five 9’s for voice with two servers. Now of course if you want to add additional functionality like remote VPNless access to Communicator you will need additional servers but even with every service and role deployed your server count will remain relatively low. This also doesn’t account for virtualization which is still to be announced.
This scenario is great for up to 5000 users. Even though each server is capable of housing 5000 users on its own placing 5000 users on each server would mean that in a DR scenario that 10,000 users would be registered to one server which may not be supported. My thought here is what if you wanted to migrate the users from the failed server over to the still functioning server you would need spare capacity. Of course in a larger environment you could have an SE server actually fail over to a fully blown Enterprise pool so scaling to a much larger deployment in the future is also an option.
In summary, this is a great option for small business looking to expand its UC capabilities. It simplifies the deployment while improving uptime. It is also an option for the larger enterprise trying to provide some local hardware resiliency for a large remote site where an SE server can fail back to a pool or over to another SE server. The beauty in all this is flexibility to build the UC deployment that meets the needs of the business.
Comments welcomed.
VoIPNorm
The main concept with the new disaster recovery option on CS 14 is the ability to separate the SIP registrar allowing for a primary and secondary registrar. This concept has been talked about somewhat with survivable branch appliance with an Enterprise Pool but it is also an option with dual Standard Edition servers.
When Communicator 14 registers it receives the primary and secondary register in-band in the SIP signaling along with other information about meeting policy’s etc. Although there has been some mention of SRV DNS records it is defined in topology builder (another blog post for another time)as the backup registrar pool and passed via in-band signaling to the client when logging on to Communication Server.
This feature is really about keeping voice up and available in the case of a data center failure. Although I have showen the concept of two remotely located datacenters this doesn’t have to be the case. In the case of a small enterprise this could be the same rack for all practical purposes in the case of wanting to provide some hardware resiliency to voice.
Features during DR:
What’s great about this solution is its simplistic nature. No load balancing required. It also provides the ability to continue doing peer to peer sessions for datasharing, IM and video. When in failover mode the user receives a clear indication that there is an issue and to expect limited services. A big red banner is displayed across the client until services are restored. If you want to restore full services quickly to your users you can migrate your users on to the still operational SE server and they will be back up and working with full UC features.
As you can see I have also placed the mediation server functionality on the SE server so with the addition of two gateways you have a full UC solution of 5000 users with five 9’s for voice with two servers. Now of course if you want to add additional functionality like remote VPNless access to Communicator you will need additional servers but even with every service and role deployed your server count will remain relatively low. This also doesn’t account for virtualization which is still to be announced.
This scenario is great for up to 5000 users. Even though each server is capable of housing 5000 users on its own placing 5000 users on each server would mean that in a DR scenario that 10,000 users would be registered to one server which may not be supported. My thought here is what if you wanted to migrate the users from the failed server over to the still functioning server you would need spare capacity. Of course in a larger environment you could have an SE server actually fail over to a fully blown Enterprise pool so scaling to a much larger deployment in the future is also an option.
In summary, this is a great option for small business looking to expand its UC capabilities. It simplifies the deployment while improving uptime. It is also an option for the larger enterprise trying to provide some local hardware resiliency for a large remote site where an SE server can fail back to a pool or over to another SE server. The beauty in all this is flexibility to build the UC deployment that meets the needs of the business.
Comments welcomed.
VoIPNorm
Exchange 2010 UM and CUCM 7
Download the full configuration document from Microsoft here:
http://www.microsoft.com/downloads/details.aspx?FamilyID=d5db0297-7850-4f52-b965-b8006b4c05f5&displaylang=en
http://www.microsoft.com/downloads/details.aspx?FamilyID=d5db0297-7850-4f52-b965-b8006b4c05f5&displaylang=en
OCS Billing and Reporting
A really common question is around the reporting and billing options available in OCS. What made me write about this topic now is the question was raised on the Technet Forums and there was a reply that OCS wasn’t customizable. I was beside myself for a second as this is the total opposite of the reality, then I prepared my response which I have reposted here.
I tried to fill in all the options I could think of but I probably missed a bunch. If you have some good refference for OCS reporting please feel free to post a link in the comments of this post.
Comments welcomed.
VoIPNorm
So you have a number of options in regards to reporting on calls.
Not sure what Shafaquat is referring to. Reporting is completely customizable if you create your own SQL queries and reports. You really have three options.
A) There is some limited canned reporting available with OCS R2 that is set to expand with 14. When you deploy the OCS Monitoring server there is an option to deploy these reports to an SQL reporting server:
http://technet.microsoft.com/en-us/library/dd425085(office.13).aspx
This is a good blog describing some of the available options:
http://blog.insideocs.com/2009/04/27/ocs-usage-reporting-options/
B) Create your own SQL queries and reports. The CDR/Monitoring database is nothing more than a SQL database that is easily accessible
Blog on creating custom queries
http://blogs.technet.com/b/perez/archive/2009/07/18/application-sharing-monitoring-in-ocs-r2.aspx
CDR and Monitoring database scheme
http://technet.microsoft.com/en-us/library/dd572606(office.13).aspx
QoE query samples
http://technet.microsoft.com/en-us/library/dd819976(office.13).aspx
C)Buy reporting capabilities from a company like Unify2 or Time billing by Convergent Solution
http://www.unifysquare.com/powerview.aspx
http://www.convergent-solutions.com/cgi-bin/convergent/file/Convergent%20UC%20Professional%20Services.pdf
Wave 14 there are going to be a bunch of new partners developing billing solutions and other reporting capabilities
http://www.microsoft.com/communicationsserver/cs14/en/us/partners.aspx
Here is a thread on Technet about some different billing options
http://social.technet.microsoft.com/Forums/en-US/ocsmonitoring/thread/e5ed8c30-13e1-4ce4-b1e6-458a69eae7ad
Lots of options I hope this helps.
I tried to fill in all the options I could think of but I probably missed a bunch. If you have some good refference for OCS reporting please feel free to post a link in the comments of this post.
Comments welcomed.
VoIPNorm
Outlook Social Connector for Facebook
I know I don’t normally talk about Outlook on VoIPNorm but I really like the Outlook Social Connector. They have just added Facebook. Here is the link to download the new plugin for Facebook:
http://www.microsoft.com/downloads/details.aspx?FamilyID=ce8b7517-234c-48a1-a655-324a88893b02&displaylang=en
More about the OSC:
http://blogs.office.com/b/office_blog/archive/2010/07/13/connect-to-facebook-and-windows-live-with-the-outlook-social-connector.aspx
If you are on Office 2010 this will also comedown in the Windows Update. OSC is also available for 2003 and 2007.
http://www.microsoft.com/downloads/details.aspx?FamilyID=ce8b7517-234c-48a1-a655-324a88893b02&displaylang=en
More about the OSC:
http://blogs.office.com/b/office_blog/archive/2010/07/13/connect-to-facebook-and-windows-live-with-the-outlook-social-connector.aspx
If you are on Office 2010 this will also comedown in the Windows Update. OSC is also available for 2003 and 2007.
Video: Communicator 14 Enterprise Voice
BJ from the OCS product group runs through some of the new CS 14 Enterprise voice features.
KB article 981218: RTCP Timer Confusion
Sometimes you think your right only to be proven wrong. It’s annoying but I would rather be wrong and correct the information at hand than to let it hang about. After spending some time working through an issue with someone here in VoIPNorm I have had to correct an error in a previous post and in this post I will also expand more on what we found and what you can do to correct it.
I was working with a customer recently and we came across some interesting wording in KB article 981218 that led to some confusion around the setting required to disable RTCP timers on the Gateway side of the Mediation Server. The article shows the setting as “true” in the example, later in the article it states that setting it to true will enable session timers which when reading the article isn't very clear and provides no example of this other setting. The misleading statement:
This statement should read:
Confused yet. Just to recap why this setting is used. The issue we were trying to solve was with Cisco Unity when you are recording a voicemail message, Unity does not send RTCP packets and therefore the recording stops at 30 seconds when the Mediation server RTCP timers time out. Now that this is disabled a session timer which was previously disabled is now enabled by default. This means that calls between a Cisco IP Phone and Communicator will drop after 30 minutes if the session timer is not disabled. Also see my previous post on the topic for more information.
The error for the RTCP issue from the Mediation server looks something like this:
BYE
sip:chris.norman@rtctest.contoso.com;opaque=user:epid:U_CefZQFsVm9gVJSmY_DWgAA;gruu SIP/2.0
FROM: ;epid=B9723D714F;tag=f823396c4d
TO: ;epid=9277a8242a;tag=6e417bfd42
CSEQ: 1 BYE
CALL-ID: 8110143875c0465ea6add6729c04f800
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 130.42.108.237:1620;branch=z9hG4bK8941dab2
ROUTE:
CONTENT-LENGTH: 0
USER-AGENT: RTCC/3.5.0.0 MediationServer
Ms-diagnostics: 10011;source="Server.RTCTEST.TL.CONTOSO.COM";reason="Media Gateway side stream timeout";component="MediationServer"
To fix this issue the following steps need to be taken:
Step 1: Patch Mediation Server according to http://support.microsoft.com/kb/977937/
Step 2: Alter MediationServerSvc.exe.config file to include the new settings:
<?xml version="1.0" encoding="utf-8" ?>
<configuration>
<appSettings>
<add key="forwardDisplayName" value="True" />
<add key="GatewayIgnoreMediaTimeout" value="True" />
<add key="GatewaySessionTimer" value="False" />
</appSettings>
</configuration>
I have left in the “forwarddisplayname” setting as an example to show how to apply the change when other settings already exist in the file. If the file does not exists you will need to create it as the default setting is false.
Step 3: Reboot mediation server to have setting take effect.
Thanks to Mike D (you know who you are),Nemo and Anon for testing and validating the configuration.
Apologies to all those that read this article and were lost or confused. Hopefully these corrections have cleared the air.
Comments welcomed.
VoIPNorm
I was working with a customer recently and we came across some interesting wording in KB article 981218 that led to some confusion around the setting required to disable RTCP timers on the Gateway side of the Mediation Server. The article shows the setting as “true” in the example, later in the article it states that setting it to true will enable session timers which when reading the article isn't very clear and provides no example of this other setting. The misleading statement:
“If the file does not exist, the default setting for GatewayIgnoreMediaTimeout is set to False. If the Mediation Server is set to ignore media timeouts for its gateway side, the Mediation Server will automatically enable session timers for its interactions with its gateway side peer. If the session timers expire, the call will automatically be cleared. If the Mediation Server has to ignore media timeouts for the gateway side interactions of the Mediation Server but the session timers does not been enabled, a configuration file setting is used. This configuration file setting is called GatewaySessionTimer. This setting contains one of two values: True or False. A False value disables session timers for gateway side interactions.”
This statement should read:
"If the file does not exist, the default setting for GatewayIgnoreMediaTimeout is set to False.
If the Mediation Server GatewayIgnoreMediaTimeout (RTCP) is set to True the Mediation Server will automatically enable session timers (SIP) for its interactions with its gateway side peer. If the session timers expire, the call will automatically be cleared as per the default behavior. The session timer will expire after 1800 seconds (30 minutes) by default. This configuration file setting to change the default behavior is called GatewaySessionTimer. This setting contains one of two values: True or False. A False value disables session timers for gateway side interactions."
Confused yet. Just to recap why this setting is used. The issue we were trying to solve was with Cisco Unity when you are recording a voicemail message, Unity does not send RTCP packets and therefore the recording stops at 30 seconds when the Mediation server RTCP timers time out. Now that this is disabled a session timer which was previously disabled is now enabled by default. This means that calls between a Cisco IP Phone and Communicator will drop after 30 minutes if the session timer is not disabled. Also see my previous post on the topic for more information.
The error for the RTCP issue from the Mediation server looks something like this:
BYE
sip:chris.norman@rtctest.contoso.com;opaque=user:epid:U_CefZQFsVm9gVJSmY_DWgAA;gruu SIP/2.0
FROM: ;epid=B9723D714F;tag=f823396c4d
TO: ;epid=9277a8242a;tag=6e417bfd42
CSEQ: 1 BYE
CALL-ID: 8110143875c0465ea6add6729c04f800
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 130.42.108.237:1620;branch=z9hG4bK8941dab2
ROUTE:
CONTENT-LENGTH: 0
USER-AGENT: RTCC/3.5.0.0 MediationServer
Ms-diagnostics: 10011;source="Server.RTCTEST.TL.CONTOSO.COM";reason="Media Gateway side stream timeout";component="MediationServer"
To fix this issue the following steps need to be taken:
Step 1: Patch Mediation Server according to http://support.microsoft.com/kb/977937/
Step 2: Alter MediationServerSvc.exe.config file to include the new settings:
<?xml version="1.0" encoding="utf-8" ?>
<configuration>
<appSettings>
<add key="forwardDisplayName" value="True" />
<add key="GatewayIgnoreMediaTimeout" value="True" />
<add key="GatewaySessionTimer" value="False" />
</appSettings>
</configuration>
I have left in the “forwarddisplayname” setting as an example to show how to apply the change when other settings already exist in the file. If the file does not exists you will need to create it as the default setting is false.
Step 3: Reboot mediation server to have setting take effect.
Thanks to Mike D (you know who you are),Nemo and Anon for testing and validating the configuration.
Apologies to all those that read this article and were lost or confused. Hopefully these corrections have cleared the air.
Comments welcomed.
VoIPNorm
Communications Server 14 Devices and Licensing
What’s new in devices for wave 14 video:
Licensing changes for wave 14. BJ from the product group has done a great job of walking through the licensing changes on the UC group team blog.
Licensing changes for wave 14. BJ from the product group has done a great job of walking through the licensing changes on the UC group team blog.
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