No Erlangs for UC Part 2

 

FOX GROUP’s Helga Egan has spent several years implementing SIP solutions for multiple client projects. She devoted additional time researching SIP solution providers in Canada, and has put together a guided tour of the decision processes required for SIP adoption and deployment.  

 

Sharing Bandwidth, Myth and Reality

The initiative to make voice services part of the deliverables of IT department’s services was started during my working life, but it has not necessarily been a smooth transition for those who have made it, while many are still on the fence or keeping data and voice separated.  

 

The concept was to introduce a dynamic use of bandwidth resources between voice and data, in theory saving money and reducing complexity.  The elephant in the bandwidth room is voice and its’ needs for bandwidth can be elephantine. 

 

QoS and prioritized data are still foreign to many businesses networks. The IT groups I meet with surprisingly need to be educated on the integration of voice even today.  Vendors, may have to take some of the blame, as they may not be exercising a duty of care to inform buyers of the need for advanced, leveled-up network bases to run their products on. 

 

To use a Christmas analogy, the shiny bobble of Smartphone telephony applications tied to your enterprise, that update you with voicemail tallies, user presence, and conference and video calls don’t come with wired and wireless network builds (batteries extra), and can use a lot of data resources. (see Editor in Chief comments at end of article)

 

The price for a voice system can be only the tip of the iceberg.  Once you have handed out and trained users on the nifty technologies they will expect them to work as well as their Smartphones do.

 

SIP trunking as a UC service

The next technology that is trending for the larger market today, although it is a well tried product with years as a telephony service, is SIP or Ssession Initiated Protocol trunking.  The protocol, when used with a UC telephony system and deployed SIP end devices, in a LAN/WAN, removes the need to translate voice to digital or analogue signaling and back.

 

This ability to do SIP to SIP calls reduces overhead and complexity on the network.  The idea of using SIP from outside the firewall for trunking, can bring additional reductions in traffic, and if prioritized properly, improvements in quality as a complete SIP to SIP system.

 

The figure below illustrates the evolution of trunking (although these technologies are all very much in use today) scaling from complete separation of voice and data on diverse service accesses to virtual separation today using SIP.

 

 

Termination of services in the older analogue and digital technologies were separate and services went their separate ways to diverse devises on arrival at the business.  SIP voice delivery will arrive on the same pipe with your data and video traffic. 

 

The difference at this demarcation point for many businesses will be an upgrade to their routers, and with that a requirement for the skills to program and troubleshoot problems on those routers.  IT groups need to understand the use of VLAN’s and the distribution of the arriving data and its’ priorities and protocols routing.

 

The major delay our clients have experienced in transitioning to SIP is the provisioning from their last mile carrier of the required circuit(s) to complete the conversion to SIP trunking.

 

If you are located in a large urban center on a fibre loop the time to installation could be a relatively short 3 to 6 weeks.  If your business is not located in these prime infrastructure areas SIP may not be in the cards for your business at all or delayed for years.  Circuit build costs can increase your project costs substantially, and push out time-lines from months to years causing a need to go back to project sponsors for additional funding.  

 

Internal wireless networks built for stationary computers, or minimal Smartphone use, could experience extreme delays and slowness, which voice (and video) cannot tolerate.  These networks could require major overhauls for the SIP device users to move freely and remain on their calls. 

 

Make sure your IT department is aware, and in sync with your desired outcomes, as they will be the major stakeholder and possibly program managers for the resources you need to execute. 

 

Their service providers will likely be your voice circuit provider although they may not be your SIP provider.  You may have separate SIP, ISP and Network providers, or you may be able to find a service provider that does it all.  In the figure below we are illustrating a basic layout. 

 

The providers on the right side are all different, in addition you may be running with analogue or PRI as a 911 service provider or as a back-up, which could be another provider as well.

 

Once past these network, and equipment hurdles, you may be on your way to the freedom and joys of SIP Trunk services and applications.

 

Provisioning and the Erlang Model

Now that you can move ahead with your project, how do you provision your broadband network to be able to transport the new SIP service?

 

Most SIP carriers will do a 'like-for-like' replacement of your current analogue or digital trunks in channels, unless you are having issues, or feel you are over-provisioned.  I don’t feel there is much expertise in the market or at the SIP providers to model your traffic as was done with Erlangs.  From our observations at FOX GROUP, there is very little telecom design expertise left in the telecom market for provisioning altogether. 

 

In the past, I was a designated expert at traffic studies and designing PRI (Megalink) services at a national carrier.  It was quite scientific, and the traffic studies involved extensive industry knowledge and were useful to both the customers and their vendors providing the day to day telecom support. 

 

If you are coming from a place of having Primary Rate Interface PRI (digital) trunks, you could safely follow the general carrier rules.  Coming from analogue environments would require some ratcheting down of trunks depending on how close you were to 23 B-channel lines.

 

The most vital first step is to make sure you have upload speeds to match your requirements.  There are many bandwidth products you can use for your voice traffic. Asynchronous is the best starting platform, but if you have to use a basic DSL,  make sure your minimum upload bandwidth matches your use requirements for all of your various UC applications that you will require transport for. 

 

To determine requirements, you need to understand a little about bandwidth used per call.  Bandwidth for voice has a codec dependency.  Codec G.711 is the best quality and most commonly used codec for voice transmission, however a higher compression codec of G.729 is also standardized for voice.  G.729 has challenges transmitting fax, and tone.  

 

Check with your carrier to see if you need to use G.729 at any locations, if they have other provisions for tones and fax transmission. An accepted calculation of bandwidth per voice call is 85kbps, however this should be confirmed with your SIP carrier.  An example of a company or branch location with 10 users might expect 3-5 people on calls at a time, but need to again look at their normal voice usage to be sure. 

 

Using the above model:  85 x 5 = 425 kbps

Upload speed required = 425 kbps or 4.25 mbps

 

The minimum upload speed for this example would be roughly 5 mbps.  Bandwidth will limit any other changes you want to make such as increasing SIP channels.  It is best to project a reasonable length of time into the future (3-5 years), and expected growth levels of the business when making your bandwidth purchase, especially of you are considering locking into a contract for better rates.

 

The good thing is many SIP Carriers can deliver traffic reports to assess your provisioning in short order once installed and you can make adjustments to the number of SIP channels.  In addition, a perk of SIP is the ability to flex usage. 

 

Most SIP providers allow bursting to more than your stated number of channels.  This is great for all kinds of businesses that have busy and slack times within months, or seasons.  This allows you to pay a low rate and burst up to additional channels as and when required, whereas PRI operates in the opposite. You must provision and pay monthly for your peak requirements in order to have the maximum number of channels available as and when you need them.  This channel expansion is one of the main reasons you hear that SIP can save you money.  

 

For firms that require seasonal expansion and flexibility to reduce as and when needed, RUN, don’t walk, to migrate to SIP services. 

 

Note from Editor in Chief:  Based on our internal past experience of the past 2+ years using full UC applications including PSTN running on SIP trunks over public Internet broadband services, we generally recommend to our clients that each fully loaded UC user may require up to 2-2.5 Mbps of download and upload bandwidth available in order to have high quality performance across all communications methods and devices.

 

Don't try to skimp on the WAN network connectivity if you expect to get the benefits possible from UC applications!

 

If you would like to discuss if and why you should move to SIP solutions for your organization, feel free to contact Roberta.Fox@FOXGROUP.ca or +1.289.648.1981. 

 

If you would like to discuss if and when you should move to SIP solutions for your organization, feel free to contactBill.Elliott via email or his SIP DID no. at 289.648.1985.

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