Guest Post by Dave: PBXNSIP Trunking to OCS R2

This week I would like to introduce Dave Howe as guest technical contributor at VoIPNorm. Dave works at Microsoft as a UC Senior Support Engineer and is an aspiring technical writer (checkout Dave’s full profile under the new about page).). In my current role I don’t always get time to write in-depth technical articles so I will be leaning on Dave from time to time to keep technical content coming. Dave’s technical expertise is widely recognized at Microsoft and I am very privileged to have him contribute to the blog directly. You can also check out Dave’s blog at http://blogs.technet.com/daveh.

This week is a quick post with info provided by Dave on trunking a PBXNSIP to OCS R2 using direct SIP for the purpose of routing calls between the two systems. Below is the trunk settings required for the PBXNSIP box.

Create New Trunk for OCS R2 Mediation Server

The option for creating a new trunk can be found in the lower left corner of the Trunks menu.



Enter an appropriate name for the new trunk and choose the SIP Gateway trunk type. Next, click on the Create button, which will add a new OCS 2007 R2 Mediation Server entry to the list of available trunks. To begin editing the configuration settings for the OCS 2007 R2 Mediation Server trunk, click on the edit option and configure the following values:

Name – OCS 2007 R2 Mediation Server
Type – SIP Gateway
Direction – Inbound and Outbound
Domain – mediation-r2.contoso.com (DNS name of Mediation Server)
Outbound Proxy – 192.168.1.12:5060;transport=tcp (IP address of Mediation Server)
Override Codec Preference –G.711U (for North America) / G.711A (for Europe)
Lock Codec during Conversation – Yes
Accept Redirect – Yes
Interpret SIP URI always as Telephone Number – Yes
Is Secure – No
Send Call to Extension – {Leave Empty}
Assume that Call comes from User – {Leave Empty}
Ringback - Media

Be sure to scroll to the bottom of the screen and click Save. .
Finally, click on the List tab to return to the list of available trunks.

Look for more articles from Dave in the coming months.

VoIPNorm

Cisco and HP Part Ways

HP is no longer certified as a Cisco reseller or service partner. This was an interesting announcement that may have a reoccurring theme as the competition between HP and Cisco heats up. How long before installing CUCM on a HP server is no longer supported, only time will tell if this eventuates.

Death of the Voice VLAN

I ran across this paper from Duncan Blake at Unify Square a few weeks back. I think it presents an interesting argument about the use of voice VLANs in the enterprise as a security measure. This is not the first time that I have seen this discussed. I have talked about the same subject myself with coworkers at my previous employer. While this document doesn’t really go into the network considerations of voice VLANs it does make me think about the merit of both the network and security measures of voice VLAN’s when moving to a PC based VoIP network.

Lets take a look at some of the reasons that you would use voice VLANs beyond just security in a IP hardphone deployment.

Quality of Service – The number one reason IPT vendors advocate the use of voice VLAN’s. Traffic separation to ensure quality.

IP addressing – In a large IPT deployments IP addressing could be a major concern when deploying hundreds of new IP phone devices on the LAN. Rather than readdress whole segments of the network voice VLAN’s are sometimes the simpler answer allowing the use of a private IP address scheme and extending the network.

Security –Included for completeness but really the last on the list.

With the following in mind when moving to a PC based VoIP environment where can the PC make up where Voice VLANs left off. Firstly QoS has to be addressed. No argument that QoS is a requirement over low bandwidth WAN connections and in high traffic areas such as a datacenter but at the desktop and local switch closet I start to have some doubts over the value this method offers.

Seeing as we now have data and voice on the same VLAN some might propose we allow the PC to do 802.1Q tagging to extend the existence of voice VLAN’s to the PC. This idea does have some merit when working with standard G. codecs like G.711 where tolerance to jitter and packet loss is low. With the ability to take advantage of PC hardware more vendors are taking advantage of more flexible intelligent codecs such as RTaudio(Microsoft), SILK (Skype) and iLBC (various, freeware).

RTAudio -http://www.microsoft.com/downloads/details.aspx?FamilyID=5d79b584-79c9-42a8-90c4-4ab3f03d19c4&displaylang=en

SILK -http://www.wirevolution.com/2009/01/13/skypes-new-super-wideband-codec/

iLBC -http://ilbcfreeware.org/

While not all of the mentioned codec's can do wideband they do prove that intelligence in the codec is an industry direction and the current G. standards are not the preferred PC codec. What this equates to is less reliance on the network to provide and secure quality.

But wait we haven’t totally dealt with QoS. Marking and classifying is a big part of using a voice VLAN and QoS. Trusting the PC to correctly categorize and intelligently mark VoIP packets is a question most will ask. Well here are a few points. The first one will only be relevant in a Windows environment but using Group Policy’s to ensure that computers are correctly marking packets to match a DSCP architecture is a pretty easy answer. The biggest question here is convincing network people that it can be done correctly is only something each organization can answer. The second would be to mark, classify and queue at bottlenecks on the enterprise network. Unfortunately you aren’t going to get this on the internet so if you have a large remote community this isn’t going to be much help so it’s back to more intelligence in the endpoint.

Moving back to my original discussion about VLAN’s our next area is IP addressing. Well, we already have PC’s on the network so unless adding VoIP to PC’s means more PC’s it’s kind of a moot point. I would be surprised to hear someone say “we added VoIP which meant we needed more PC’s.” Upgrading PC’s, sounds more likely.

The last point is security and at the heart of Duncan’s paper. It’s been proven time and time again the value of VLAN’s as a security measure is nonexistent. A kiddie with a program can defeat this which really makes its value pretty low from a security perspective. Getting access to a switch port for a non-employee of an enterprise is more of a challenge than a voice VLAN from a security aspect.

Although Duncan mentions OCS I think the industry as a whole is recognizing that VoIP needs to be more secure and layering on security as an upgrade or an overlay is a poor approach. I am not going to argue one vendor versus another’s approach but if your deployment of IPT or Unified Communications isn’t using secure media (SRTP and SRTCP) and secure signaling (SIP/TLS as an example) the question is, why? It also makes the discussion of using a VLAN to provide security much less tangible. Is the effort of configuration worth it, if it can be so easily defeated by someone with network access when your media and signaling is already secure? It’s not.

So when you sit down and write your next Unified Communications RFP for enterprise X, think of the following points-

Is security a part of the initial configuration or an overlay?

What codecs will you use for a solid call quality on uncontrolled networks?

How reliant should VoIP be on your enterprise network to ensure call quality whether it is PC based or not?

Comments welcomed.

VoIPNorm

Why Microsoft?

“Why Microsoft?” is new take from Microsoft as the UC battle unfolds. There is a section aimed at its main rival in the UC space as well as other competitors across different markets. I am not going to make any comments about the content but I certainly encourage people to take a look and make up their own minds on what’s available.

Feel free to leave a comment on the content of the site. I understand these types of competitive materials can generate some emotional responses from passionate people but please keep it at a professional level.

Stove Pipes are Easier to Sell

This is going to be one of those retrospective posts that are going to make you think so just hang in there. Although the conclusion seems like a tangent to the title I think you will understand what I am talking about when I get there.
Recently I have been thinking (those of you who actually know me may find this hard to believe) a lot about how different vendors build and present their Unified Communications platforms. Some use the method of dividing their platforms into easily recognizable products with each workload requiring a server or more (depending on availability requirements) with their own management interfaces. Why do they do this when it is possible to do it on one platform with one management platform?

Sure there are technical reasons why this is the way it is but what about when there isn’t. Here’s one reason. If they are separate products they are easier to sell in segments, easier to license and more than likely easier to derive their own revenue from. I look at this as stove pipe selling. Seems pretty simple right, but what about the customer’s perspective?

If they buy a product that has its own management interface and presents itself as a separate IT tower, it means less change and an easier product to buy from the perspective of team integration. If I need 5 separate products to run IM, Presence, Mobility, Web/ video/ audio Conferencing and Telephony but it means I don’t need to bring my IT teams together as one seamless team that somehow makes my life easier because I don’t need to deal with those other guys. It is also easier to sell to one IT tower than a whole UC team so some vendors have little incentive to mesh products together seamlessly under one management platform, in my opinion. To me, this seems counterintuitive to the whole concept of UC.

So once stove pipe selling has taken place what happens next. The CIO wants UC in the work place and now we have to figure out how all this stuff works and integrates together but because we purchased in stove pipes and we have to figure out if vendor X works with vendor Y. Now there are plugins and all sorts of other complexities that in the end will end up in a whole but what if we started in a different direction. If managers from each IT tower came together and said before we make one more IT purchase let’s form a UC architecture team to derive our direction and partners. Now this doesn’t mean buying products from just one vendor. It just means setting the direction within an enterprise for UC.

I am not trying to sell a product or convince people to go with product xyz with this post but its more around creating the idea that seamless integration starts with team and not product. The product selection is a consequence of the team not the reason the team forms in the first place.

Comments welcomed.

VoIPNorm

Avaya SES and Microsoft OCS interoperability

Over the last few weeks I have been doing a bit of research into Avaya and OCS R2 interoperability. When you refer to the OCS Open Interoperability web page you will find Avaya under the supported IP-PBX section. This means Microsoft has tested the interoperability internally and this is not OIP Certified as with other gateways etc. This means Microsoft does support the configuration but the other vendor may not. In saying that Avaya has released some documentation on this interoperability but it has been a while since it was updated.

OCS Enterprise Voice - Avaya interoperability - Direct SIP

Firstly, documentation on this interoperability is hard to find. Avaya have released this document on call routing between the two systems. It is based around using the Avaya SIP Enablement Server version 4.x and OCS R1. This document is about a year and half old which means things have changed since its release but I was unable to find any updated documentation publicly available. One thing about this document is it makes references to Remote Call Control. Its is not applicable to this configuration and is somewhat distracting for it to be in this document.

Avaya did document issues which are listed below:

“On a call between an Avaya phone and an EV client, the EV client is not able to place the call on hold. The microphone and speakers on the EV client are muted, but the call is not actually placed on hold. In contrast, on a call between two EV clients, both EV clients are able to place the call on hold.”

“On a call between an Avaya phone and an EV client, attempts by the EV client to
conference in another Avaya phone or EV client fail” and “On a call between two EV clients, attempts by either EV client to conference in an Avaya phone fail.”

Microsoft have also listed known issues for SES 4.x:

Configuration requires setting "Alternate Route Timer(sec)" value from default of 10 sec to 30 sec. The configuration should show "Alternate Route Timer(sec): 30" in the corresponding SIP signaling group.

When an call is ringing to the Office Communicator user, the caller (either on an Avaya station or a PSTN line routed through the PBX) will not get ring back tone. This issue has been resolved by Avaya with the 5.x software releases.

Quality of Experience reports will not contain information regarding jitter and packet loss.

Comfort noise generation is not supported. As a result, comfort noise is not played on Office Communicator.

ISDN Failover is not supported from an OCS perspective. If the Avaya PBX is being used for PSTN connectivity and multiple T1's are being utilized, an OC client will not retry a call based on a T1 being unavailable. It may be possible to configure the Avaya to not assign outbound calls from OCS to an unavailable T1, but this configuration was not tested.

Summary

So while this may have some issues for 4.x it is recommended by Microsoft to use SES 5.x software to get the best results. If planning to use direct SIP I would recommend thoroughly testing before deployment of the service.

An Alternative Solution

Another configuration that would also be possible is to use a Hybrid Gateway from one of the certified gateway vendors http://technet.microsoft.com/en-us/office/ocs/bb735838.aspx that also supports IP-IP GW features. Considering that some of these vendors have completed interoperability testing with Avaya and Microsoft it may present a more flexible solution with fewer issues.

Avaya OCS Interoperability

NET buys Smart SIP

If you haven’t already seen the announcement Network Equipment Technologies has purchased Smart SIP from Evangelize. For those that don’t know, Smart SIP allows generic SIP phones to register with OCS and loads as an application on a OCS Mediation server. It can also support Cisco IP phones with a SIP load and provides a TFTP server to allow phone configuration and updates.

Reusing already acquired phones is a common request, so this was an intelligent purchase by NET to help fill a gap and possibly take Smart SIP to the next level.

Technet OCS Webcast Series

Office Communication Server Series

Voice Architecture and Planning for Microsoft Office Communications Server 2007 R2

Tuesday, Feb 16th: 11:00AM-12:30PM (PST)

Want to know what's behind the voice features in Office Communications Server 2007 R2, and what it would really take to bring it in-house? In this webcast we present the deployment scenarios for enterprise voice functionality, including sizing and topology considerations, call routing and management, and interoperability with existing telephony infrastructure.

Register here

Unified Communications Development for Non-Professional Developers

Tuesday, Feb 23rd: 11:00AM-12:30PM (PST)

This webcast walks through a set of easy-to-use applications anyone can build on the developer platform of Microsoft Exchange and Office Communications Server with basic programming skills. All code will be made available after the webcast on http://gotuc.net

Register here

This is a great webcast series I should have posted a long time ago but was not thinking. May be a higher level than some may like but if your just getting into OCS a great place to start.

UCDoers March Meeting

Once again North West UCDoers are back for their quarterly meeting.

As requested at the last event we are jumping in headfirst to OCS Voice!! We'll be talking voice quality - how to get it, how to measure it, and all the other juicy details that go along with using OCS to run the voice communication platform in your company!!

Once again, we are bringing in Duncan Blake who is an expert on the topic!!
What: UCDoers Voice Quality

Where: Microsoft Campus
Enterprise Engineering Center
Building 25 North Tower
TBC - 25/3034
Redmond, WA 98052

When: March 10th, 2010, 5pm - 7pm

More info: Feel free to forward to folks we may have missed, all are welcome and encouraged to attend!! The broader the community the better!

For this meeting, our speaker @DuncanBlake will review the ins and outs of OCS voice quality - how to get it, how to measure it, and all the other juicy details that go along with using OCS to run the voice communication platform in your company!!
Definitely a not to miss event. There will be give aways, food & drinks, and of course some great industry peers to meet and discuss implementations with.
We're looking forward to seeing you there!!

Please register here: http://www.pingg.com/rsvp/i73ex6ce3pj5qz87y

Pre Call Diagnostic tool

The Pre Call Diagnostic Tool is a really interesting tool that has been available in 64 and 32bit versions for quite some time now but it still surprises me how many customers don’t realize its available for their use for free. The 64 bit version is available when you down load the OCS 2007 R2 Reskit and the 32 bit version is a separate download since it was only available after the release of the Reskit.

This tool offers the ability for a user to check the current state of the network before making an enterprise voice call. It is most useful when users are using networks that are uncontrolled like the internet, so great for testing you network access back to the OCS AV edge.

The screen shot below is the tool open on the desktop showing a good connection back to my corporate network. This is just after starting the tool.



The next shot shows the tool after its been up and running on my desktop for a while. It is showing a little bit of jitter but still good conditions to make a call. Some variation delay is normal on any network so this small amount of jitter is none to be concerned about.



This next shot is of the allowed settings and log file location. Similar to Communicator it uses the same DNS methods to locate the edge or internal pool resources, hence the use of a user SIP URI to find the correct SIP domain. You do not need to be logged onto Communicator to use the tool.



The last shot shows the tool minimized in the Task Bar. So if the tool is left running you will be able to see network conditions before making a call.



Hopefully for those that didn’t already know of its existence this has been of some help.

OCS 2007 R2 Architecture Poster

If you haven’t already seen this elsewhere this is worth checking out. This architecture poster does a great job of showing in detail the signaling used within OCS 2007 R2.It was created by the same people that wrote OCS 2007 R2 Resource Kit.

http://www.microsoft.com/downloads/details.aspx?FamilyID=AF2C17CB-207C-4C52-8811-0ACA6DFADC94&displaylang=en

Blogging Resources

I have been running my blog over a year now and I have spent a lot of time and energy try to mold it into a better resource for people trying to deploy Unified Communications while not being a repeat of a lot of other blogs already on the web. A couple of resources have really helped me that I thought I would share. First is StatCounter.com. This is a great site that offers free statistic metrics for your blog. I have been using it for the last 7 months and it has really helped me keep track of the traffic coming to my blog.

Below is a capture of my weekly stats over the last 7 months. Not only can you capture general stats but you can also drill deeper and map these hits to specific regions around the world and also map it to which search engine is providing you with the most hits.



Had it not been for this free service I would have no idea of what was happening with traffic to my blog and where it was coming from. Truly a cool resource and I imaging can be used for any web site not just blogs.

The second is ProBlogger.This is a blogging site about making money from blogging which I have used to help me come up with ideas to help drive traffic to my blog and make it more interesting for readers. Although some of the content on this site requires payment to get access it also has loads of free content. My intention with my blog was never to make money but more of a resource and networking tool which it has done for me very successfully with the help of ProBlogger.

I guess this is my way of giving something back to these free resources which do deserve it.

Comments are welcomed.

VoIPNorm