Archive | 9:42 am

PBX replacement with MS Lync (with Dual Forking) Part 2

9 Jan
As i mention in my last post (part 1) we can choose to use the Voice gateway in pass-through as shown here :

PBXReplacementPart2_Example_base scenario

but to do this , we have to consider various steps before moving the IP-PBX in production  and insert voice gateway between PSTN and the enviroment, so let start from the beginning,  these are necessary steps :

1.  Voice gateway configuration for PSTN trunk (only configuration not yet connected), in the other side (to lync and to PBX) configuration of SIP trunk to Lync and SIP trunk to IP-PBX (IMPORTANT : to obtain a dual forking of calls in this scenario we must use only sip trunk to and from  PBX , so we must have an IP-PBX with sip trunk enabled ).

At this step we don’t have any disservice on IP-PBX production environment but we are ready to switch PSTN from IP-PBX to Voice gateway .PBXReplacement_part2_Example_step1

2.  We can schedule session tests to verify that all configuration made before work    fine (for example during time range in which we couldn’t have any outages to users, for example during the night?), in this way if not all scheduled tests list will be fine , we can rollback easily and move PSTN trunk to IP-PBX again.

To achieve this we must locate the Voice gateway properly sized, to mantain compliance on  requirement in terms of business continuity, for example redundant power supply , right number of SIP channels to/from lync, and to/from IP-PBX.

In this way we can also test a SIP trunking from a provider instead of PSTN (or buy another PSTN trunk to do a test pilot for Lync voice for example) , because until we switch the PSTN from IP-PBX to Voice Gateway, we can work easily in Voice Gateways side without give any problem in IP-PBX side.

About call flow management and dual forking we have the same behavior as i wrote in my previous post (part 1) , unique difference is that all configuration for dual forking is made in Voice gateways side , and in PBX side we have only to switch all inbound and outbound call to the new SIP trunk instead of PSTN trunk already switched off.

At this point we can assert that we have a lot of way to do a soft migration or simply to use the existence telephony infrastructure for Lync and generally Microsoft UC. We know that Microsoft Lync just for presence/audio/video/conference is really wasted and with a good approach and a small effort we can implement an excellent Lync Voice project for a really Unified Communications experience.

Source: http://msucblog.wordpress.com/2013/01/08/pbx-replacement-with-ms-lync-with-dual-forking-part-2/

PBX replacement with MS Lync (with Dual Forking) Part 1

9 Jan
Talking about PBX replacement with MS Lync can be a difficult argument when proposed to customers. But as the nature of MS Lync we have a lot of ways to do it. Usually we can meet two different type of customers, one can think that his employees must change how they work day by day, and for this reasons we can explore solution with direct switch to new technology providing a direct cut-off ; the other one,  not so confident,  prefer a soft migration and possibly a true coexistence between old and new phone system, the last one obviosly is more complicated,  but surely the most funny for us:-), i want to explain  you how we can do a soft migration also with a good coexistence, for now i can mention 2 type of IP-PBX or TDM-PBX: ALCATEL OXE and Cisco CCM.

The first important thing is that all of this project must provide a Voice Media Gateway to ensure that all translation and, eventually transcoding,  from one system to Lync and viceversa don’t drive us crazy…:-)

Actual Infrastructure Enviroment without integration

PBXReplacementExample_base scenario

Based infrastructure consider that we have a fully up and running Lync enviroment and a consolidate Phone infrastracture .

Scenario  (Coexistence with Dual Forking)

If we have a ALCATEL OXE (with remote extension license), CISCO CCM (sip forking with extension mobility license) or a TDM/IP-PBX that support forking to another number not included in its dial-plan (for example to a sip trunk or TDM trunk connected) we can consider this scenario :

PBXReplacementExample_scenario1

Using  Voice Gateway between Lync and Phone infrastructure give us a lot of configuration that otherwise we could not easily do without provide a big effort from the Phone system team .

In this scenario we can consider this events :

Inbound call from PSTN : When we receive a call from PSTN to +3906….4444 , call arrive to PBX , PBX at this stage send the call to the extension in its dialplan and see that there’s also another number associated (for example 9994444) and , in parallel , fork this call to that number with 999 (a prefix trunk associated to the Voice gateway).

When the call arrive to Voice Gateway with destination number 9994444 , it translate called number in +3906….4444 and send to lync .

Result  :  Lync client (or lync phone) and PBX phone ring at the same time, and when one of this two pick up the call ,the other one stop ringing .

Inbound call from another PBX phone : When we receive a call from another PBX phone  to 4444 , call arrive to PBX , PBX at this stage send the call to the extension 4444 and see that there’s also another number associated (for example 9994444) and , in parallel , fork this call to that number with 999 (a prefix trunk associated to the Voice gateway).

When the call arrive to Voice Gateway with destination number 9994444 , it translate called number in +3906….4444 and send to lync .

Result  :  Lync client (or lync phone) and PBX phone ring at the same time, and when one of this two pick up the call ,the other one stop ringing .

Outbound call from Lync to other PBX phone : In lync we have two way to make a call to a contact, if we make a classic Lync call , this call remain inside Lync enviroment but if we make a work phone the call is translated for example in extension format :

– Digited : +3906……4444 , normalized in 4444  so the call are sent outside Lync through the Voice Gateway  and arrive to the extension 4444 , as i mention before in pbx enviroment 4444 have another extension configured (9994444) that corrisponds to the Voice gateway trunk and the same call was also diverted to Lync client .

Result  :  Lync client (or lync phone) and PBX phone ring at the same time, and when one of this two pick up the call ,the other one stop ringing .

Yes i know , a little cumbersome but it’s work fine .

Inbound call from lync to PBX and dual forked to lync

Outbound call from Lync to other PSTN: All external calls made from Lync follow the classic flow to PSTN (Voice Gateway –> PBX –> PSTN) , it’s important to know that all calls made by Lync can have the same Calling number as the associated extension in PBX dial plan :

– for example if i make a call from PBX phone my external DID is : +3906……4444, but PBX add instead of me the +3906….. (* maybe that +39 is not considered in a national call) .

When i make a call from lync if i want that it must be the same calling number as the PBX phone ,  i have to configure on Voice Gateway a good format for PBX to accept DID so for example in ALCATEL enviroment i must pass to it the call in this format :

calling number (Lync side) +3906…..4444  — >  Translated by VG in  : 06…..4444 , in this way ALCATEL recognize the call as one from its dial-plan, otherwise can appean that my calling number is only +3906……. without the extension.

Result : the call appear to PSTN exactly from one number shared by Lync and PBX and we can realize a true Single Number Reach

Requirement for this scenario

We have to consider that if we make a QSIG trunk between PBX and Voice Gateway my advice is to use a QSIG-GF (Generic Function) not basic because there are a lot of service such as call diversion, line identification, etc.. that is not implemented on QSIG-BC (Basic Call).

If we choose a SIP trunk between PBX and Voice Gateway we have to consider in Voice Gateway side licenses for IPtoIP and eventually transcoding with DSP onboard because if we configure trunk from/to PBX in a RTP codec different from G711, for example G729 , all calls are trascoded (Lync Mediation server work only in G711).

In part 2 we’ll consider a scenario in which I’ll describe the positioning of Voice Gateway in passthrough between PSTN and PBX to prepare a clean migration phase.

Source: http://msucblog.wordpress.com/2012/12/09/pbx-replacement-with-ms-lync-with-dual-forking-part-1/

Learn how a telecoms provider takes strides to make applications security pervasive

9 Jan

BriefingsDirect

Listen to the podcast. Find it on iTunes. Read a full transcript or download a copy. Sponsor: HP.

Welcome to the latest edition of the HP Discover Performance Podcast Series. Our next discussion examines how a major telecommunications provider is tackling security, managing the details and the strategy simultaneously, and extending that value onto their many types of customers.

Here to explore these and other enterprise IT security issues, we’re joined by our co-host for this sponsored podcast, Raf Los, who is the Chief Security Evangelist at HP Software.

And we also welcome our special guest, George Turrentine, Senior IT Manager at a large telecoms company, with a focus on IT Security and Compliance. George started out as a network architect and transitioned to a security architect and over the past 12 years, George has focused on application security, studying vulnerabilities in…

View original post 3,502 more words

BYOD with QoS

9 Jan
Even though this is not a CCIE wireless topic, I had to spent lot of time to test BYOD devices for its QoS capability. My company is planning to deploy Cisco Jabber on those sort of devices to provide voice services.Officially Cisco is not supporting those sort of deployment  over BYOD, but most of customers  keen to deploy voice/ video services through BYOD.

As stated in my previous wireless QoS posts, it is purely up to end device to correctly classify traffic within the wireless media. AP does not have control how end devices tagging user priority (UP) in 802.11 frame. AP simply use this UP value in 802.11 frame to map QoS values in CAPWAP headers and subsequently this will determine how packet will get QoS treatment within wired media.

If end device (BYOD) is WMM certified we would expect that to classify traffic into  one of Voice, Video, Best Effort, Background classes and tag correct UP value in 802.11 frame. But in reality it will not work as we expect. Some time application developing vendors & BYOD manufacturers are not collaborative (as they are competing for same market segment).

I have done extensive testing with different types of BYOD with Cisco Jabber & Polycom Real Presence client installed  on it. Here is my testing topology. BYOD-QoS-01

I have made a call from wireless device to wired phone & capture packets at 3 different location shown as “A” “B” & “C” in the diagram. I have mentioned the device type & operating system used as things heavily depend on hardware/driver platform & operating system used.Also shown the 802.11 band where the end device associated. WLAN configured with Platinum QoS profile & WMM settings as “Allowed”.

Here is the result for wireless phones. You can see Cisco wireless phones mark UP value 6 for its voice traffic & UP-4 for its signalling traffic. iPhones mark voice traffic with UP  value of 5 (treat as video), but signalling traffic go as UP-0. Galaxy SII mark voice traffic as UP of 4 & signalling traffic as UP of 3. BYOD-QoS-02

If you look at Tablet devices observation is quiet different. Most of the apple devices does not correctly classify traffic with required UP value. In this case no prioritization within the wireless cell & further in wired media as well. All type of traffic go as “Best Effort” in to wired media. Noticed SIP traffic mark as CS3 in original IP header, but no UP other than “0″.BYOD-QoS-03

Here is the result for Laptops. Windows 7 does not add correct UP values for its traffic. Interestingly MacBook Pro classify its wireless traffic with UP values (only RTP with UP-5)BYOD-QoS-04

As you can see behavior is different in each type of BYOD and you need to verify end to end QoS can be delivered prior to deploying  voice/video services over wireless network.

To overcome this issue (BYOD does not classify traffic properly) cisco has come up with traffic classification (feature called Application Visibility & Control) in 7.4 software release for WLC. In this way at least traffic can be re-classified at the WLC, so wired media will get correct QoS treatment for the different type of traffic. Still this won’t help to prioritize upstream traffic from wireless client to Access point. That’s why you should select appropriate BYOD models to deliver voice/video services over wireless.

I Will write another post on AVC feature in this WLC 7.4 code.

Source: http://mrncciew.com/2013/01/08/byod-with-qos/

Achieving retail operation excellence through unified communications

9 Jan

Huawei Enterprise Retail Solution Blog

Retail eCommerce and mobile channels are not restricted by space limitations of physical store front and thus can make the full inventory of merchandise available for shoppers to view and compare, and shoppers can easily find third party or peer reviews that they trust. On the other hand, traditional retail stores bring that personal touch and more upselling opportunities. Huawei’s retail video solutions boost retailers’ omni-channel strategy and operation efficiency by increasing the level of product expertise among sales representatives, promoting tight and real-time collaborations across the full supply chain, and bridging the gap between stores and other retail channels by enhancing physical experience with virtual content.

Huawei Digital Signage Solution

Huawei’s digital signage solution is a centralized management system to control audio, video or graphic content displayed on devices located in physical retail stores. It consists of media creation, media distribution, media management and media display functionalities, and can…

View original post 501 more words

T-Mobile’s next steps: no-contract unlimited 4G, free 4G for laptops and tablets, and HD Voice calls

9 Jan

T-Mobile’s next steps: no-contract unlimited 4G, free 4G for laptops and tablets, and HD Voice calls

T-Mobile just unleashed a flurry of news at CES: The company’s popular Unlimited Nationwide 4G Data plan is going contract free tomorrow, it’s launching a new program to bring 4G wireless to laptops and tablets, and it’s also the first U.S. carrier to launch HD Voice for clear calls.

Oh, and it has also partnered with Major League Baseball to power a new on-field communication system. Whew.

First launched last September, T-Mobile says its Unlimited Nationwide 4G data plan has been a smashing success, attracting nearly 46 percent of new subscribers last month. Starting tomorrow, T-Mobile is going to make the plan even more appealing by offering it for $70 a month with no contract.

Such a move could help T-Mobile stand apart from AT&T and Verizon Wireless, which have both moved away from unlimited plans. But it’s hard to say if it will be enough to stem T-Mobile’s subscriber bleed. T-Mobile remains the only U.S. carrier without the iPhone, which has led to some dismal earnings for the last several quarters.

T-Mobile is also launching “4G Connect,” an effort to equip more laptops and tablets with 4G access. The program will kick off on a few Windows 8 laptops and Ultrabooks, including the Dell Inspiron 14z and HP Pavilion dm1. The amount of free data on every device will vary, but T-Mobile says you’ll be able ot get up to 200 megabytes of free monthly 4G service when you pick up on of the 4G Connect computers. For additional data, you can choose from any of the carrier’s monthly plans.

HD Voice, which offers a significant improvement on call quality, has been flipped on for T-Mobile’s network today, the carrier announced. You’ll need a compatible 4G smartphone to take advantage of HD Voice, and it’ll only work with other HD Voice capable T-Mobile subscribers for now. But it’s still a major get for T-Mobile. For all of its competitors LTE network advancements, call quality hasn’t improved much.

T-Mobile also said that it has launched support for 1900 megahertz spectrum in Denver, Los Angeles, San Diego, and Virginia Beach, which will allow unlocked iPhone owners to utilize the full speed of its network. The carrier now supports the iPhone-ready spectrum in 46 metro areas.

When it comes to its own LTE network, T-Mobile plans to kick off its coverage with Las Vegas in the next few weeks, the Verge reports. The carrier plans to cover more than 100 million people with its LTE network by the middle of the year.

Source: http://venturebeat.com/2013/01/08/tmobile-no-contract-unlimited-4g/