Managing Heterogeneous VoIP Networks

The Nightmare of Managing Heterogeneous VoIP Networks in Medium to Large Enterprises

There is an Answer

Managing Heterogeneous VoIP Networks

Managing heterogeneous VoIP networks in medium to large enterprises can be a nightmare. Due to the large number of vendors participating in providing the solution, there can be problems at every stage of the implementation and maintenance issues can arise in a variety of aspects including:

  • Network design and configuration
  • Device discovery
  • Distributed routing & policy enforcement
  • Distributed PSTN breakouts
  • Multiple VoIP network elements configuration: SBC and MGW
  • Multiple dial plans: SfB, IP-PBX, SBC and MGW
  • SIP interworking between IP-PBXs A large number of end user policies

Distributed networks present numerous challenges and when it comes to heterogeneous networks where the devices and applications are provided by different vendors, the situation is even worse. Unified Communication systems (e.g. Skype for Business) and the various components in the network such as IP-PBX, SBC and MGW, each have their own static routing, manipulation and dial plan tables. In actuality, we are talking about a distributed system whose elements don’t communicate with each other.  As for troubleshooting, there is no single throat-to-choke and each vendor puts the blame on the other.

AudioCodes’ new White Paper entitled, “Call Routing and Policy Management in Heterogeneous VoIP Networks” presents the challenges of heterogeneous VoIP networks and related applications currently existing in the market. The paper describes a new solution – the “Centralized Dynamic Routing and Policy Manager” – a holistic dynamic routing manager whose design is based on Software-defined Networking principles.

The Centralized Dynamic Routing and Policy Manager decouples the device layer from the network routing and policy layer, automatically creates complex VoIP networks, and simplifies routing rules, monitoring and management configuration. The Centralized Dynamic Routing and Policy Manager does not enforce modifying the network to a star formation and rather manages the network as is.

For more information, click here to download the “Call Routing and Policy Management in Heterogeneous VoIP Networks” White Paper.

VoIP Quality Jitter Delay and Packet Loss

Is VoIP Jitter, Delay and Packet Loss the Best Measure of Voice Quality?

VoIP Quality Jitter Delay and Packet Loss

“One of the values that I think men in particular have to pass on is the value of empathy. Not sympathy, empathy. And what that means is standing in somebody else’s shoes, being able to look through their eyes.” Barack Obama

Empathy – being able to feel sorry for someone else’s troubles – is a basic ingredient for good mental health. However, sometimes, when irony is involved, empathy might turn to ambivalence.  For example, you will naturally feel empathy when you see or hear about a house going up in flames, but you might even crack a smile if the house in question is the fire department. (Of course that would happen only if no one was injured, right?)

We recently had voice quality issues in our corporate telephony system. Now that might happen in any organization, but when it happens in AudioCodes, the company which has become the hallmark of HD VoIP, that’s ironic.

Our IT guys brought their usual armour to battle and started taking on the packet loss with Wireshark (network protocol analyser), upgrading this and downgrading that.  Then they started looking at the AudioCodes products on the VoIP network; scrutinizing the latest IP phone firmware, restarting the SBCs, disconnecting the MGW.  And still, they couldn’t find anything to help pinpoint the problem.

After a week or so, one of my colleagues placed a call and couldn’t stand the noise on the line.  Naturally, he cut short the call as cellular SP has “trained” us to do over the past two decades. And since he is the Product Manager for AudioCodes Management Solutions, he decided to try troubleshooting the problem on his own (15 years in R&D can come handy sometimes), using the AudioCodes Session Experience Manager.

Now I have to apologize up front since at this stage the story gets a little corny.  He found that a substantial amount of the off-net calls originated by the UC system were faulty with a poor quality score.  After conducting a short analysis with our IT department, he found a misconfiguration of the VMware machine running one of the IPPBX servers.  He wondered why our own IT guys weren’t using the AudioCodes Session Experience Manager. Could it be as a famous man once said, “familiarity breeds contempt”?

I am not relating this story to point shortcomings on the part of our IT department (God forbid!) or alternatively to glorify an AudioCodes product.  A vendor who speaks about his own products is almost as bad as a mother who talks about her own children.  My point, rather, is to elaborate on the concept of VoIP quality monitoring.

It is human nature to reach a conclusion by reviewing the consequences of actions. The human brain is constantly on an endless quest to solve riddles and reach conclusions (Gestalt psychology or Gestaltism).  Therefore, it’s perfectly normal for an IT engineer to analyse the VoIP network, search for the causes of poor quality such as packet loss, jitter and delay and based on the analysis, evaluate the call quality.

From my perspective, this analysis method is problematic.  It’s like commenting on the taste of a casserole by evaluating its ingredients – before even cooking the dish. Voice quality can’t be accurately evaluated by reviewing only the relevant network impairments (jitter, delay and packet loss). An accurate measure of voice quality needs to be taken at the VoIP edges (MGWs, SBCs, IP phones, soft clients) while taking into account the vocoder, jitter buffer type and depth (static, adaptive), packet loss metric (Network Packet Loss Rate, Jitter Buffer Discard Rate) and more. This information can’t be obtained by only analysing the traffic.

Epilogue

There is a mistaken belief in the industry that VoIP Quality Monitoring applications are like insurance policies -you buy them hoping not to use them. This belief is due to decades of trusted legacy telephony and basic trust in the organizational network, devices and solutions. In Reality, VoIP Quality Monitoring should be considered for ongoing use for the following:

  • Day-to-day, end-to-end quality monitoring and assurance operations with the ability to perform proactive actions to prevent quality issues
  • Root cause analysis for diagnostics when VQ issues arise

So if you are having VoIP quality issues, I empathize, even though after reading this, you might smile at the irony.

Yaniv Christmas tree Germany

How IT Managers Can Better Manage Skype for Business IP Phones

It was a freezing, snowy night, just a few days before Christmas.

I was on one of my road trips in Germany looking at the biggest Christmas tree in the world, while trying to warm my soul with a good, local Gluh Wein. Then my mobile phone disturbed the tranquility of the moment. One of my IT manager customers was calling. I had to answer.

Yaniv Christmas tree Germany

The customer started to describe his day to day challenges and concerns. His enterprise was beginning to migrate from an IPPBX to Skype for Business and Unified communications and his end user satisfaction was poor.  His multi-site roll-out plan took him much more time and consumed more effort than expected. And he was yet to go beyond deploying at Headquarters

where he couldn’t understand what was causing some of the IP Phones to reboot several times a day. He was unable to control the end users’ IP Phones in an efficient way and solve their issues and concerns.

I reminded him about AudioCodes’ IP Phone Manager that he had been considering and I updated him about our free of charge Express Edition. I explained how in our vision, we empower IT with a full life-cycle IP Phone operation management platform and that we view the IP Phone as an IT managed device, essentially turning the IP Phone into an IT Phone.

Feeling warmer by the minute (the Gluh Wein was no doubt having its effect…) I pointed out our multi-tenant IP Phone Manager’s day to day management and maintenance capabilities with a monitoring dashboard showing the phone operation status, active registered IP Phones, non-registered IP Phones and the disconnected devices, allowing IT to proactively detect issues before they are noticed by the end user.  I also described the smart device and user search. He just needed to search on the user name and the IP Phone provided all the necessary information (IP address, subnet, VLAN, software version and more) with just one click.

In that call I convinced my customer to give our IP Phone Manager a try and let the system become the eyes into his network in order to figure out the problems he was facing. And so he did. In just a few hours after he installed the IP Phone manager, he was already able to put his finger on the problem and track the root cause. He immediately saw that when a bulk of users were disconnected it was always in the same network switch and it seemed his POE unit was not functioning well either. He replaced the switch and the issue was resolved.

Several months after installing the IP Phone Manger he wanted to migrate additional branches to Skype for Business. I reminded him that his IP Phone Manager has zero touch installation and provisioning and that he can pre-define his configuration for an automatic zero touch rollout. Once the IP phone is plugged in, it gets the proper configuration automatically.

Together, we invested a half hour in defining his open space area, lobby area and the different company branches with the proper time offset and IP Phone menu language.

A week later he called me and expressed his amazement from the simplicity of the roll out. His ability to proactively detect and solve his end users concerns easily and efficiently was well appreciated in his organization.

And it all started with a phone call while sipping some Gluh Wein……..

Download the AudioCodes IP Phone Manager Express for Windows today and enjoy it free of charge

Telephone-Exchange

Simplified Call Routing for Complex Networks

What makes VoIP level routing hard to manage?

Telephone-Exchange

One of the main achievements in the world of data routing over the last two decades is the emergence of the distributed routing architecture.

Using various routing protocols (RIP, OSPF, EGRP, BGP, etc.), networks are created and modified dynamically and automatically. VoIP routing is much more complicated than data routing, but with no routing protocols.

Traditionally, enterprise VoIP networks are controlled by the IP-PBX and call routing is typically based on static routing tables. The situation becomes complicated for off-net calls. The IP-PBX deals with such calls by directing them to the nearest media gateway or session border controller (SBC).

In the case of multi-branch organizations, a single VoIP network may have several IP-PBXs from different vendors, as well as media gateways and SBCs. Each IP-PBX, SBC and gateway in the network has its own static routing table resulting in a distributed system in which the included elements do not communicate with each other.

In theory, one could configure all the network elements once and leave everything unchanged. In reality, however, organizations’ VoIP networks are constantly evolving due to a variety of factors: mergers & acquisitions, new locations, integration of new & old IP-PBXs (or even legacy PBXs) and integration of SBCs and gateways. In addition, there is the organic growth of an organization and the accompanying technology evolution, such as introducing unified communications to the network and consolidating IP-PBXs. All of this makes managing the organization’s VoIP network a nightmare. A static approach is no longer fit for the job – there is a need for a centralized routing management system.

Centralized VoIP Routing System

System administrators should be able to design and modify their voice network topologies and call routing policies from a single location, resulting in significant time and cost savings. Time-consuming tasks such adding a new PSTN or SIP trunk interconnection, adding a new branch office or modifying individual users’ calling privileges should be carried out simply and rapidly.

The AudioCodes Advanced Routing Manager

AudioCodes-Routing-Manager-Dynamic-Creation-of-Logical-Network-Topology

AudioCodes Routing Manager (ARM) is a holistic, scalable, dynamic routing manager based on software-defined networking (SDN) principles. The AudioCodes Routing Manager decouples the device layer from the network routing and policy layer, automatically creates complex VoIP networks, and simplifies routing rules, monitoring and management configuration.

  • The ARM learns about the SBCs and gateways in the network automatically and dynamically. Every change in connectivity and configuration is reported to the ARM. Gradually, the ARM builds up a complete picture of the network topology, connectivity and health.
  • The ARM also assists with the design and creation of the VoIP network. With the ARM, a network can be created with one-click. The administrator can choose between mesh, star or dual star formations and all the connections are automatically configured in the SBCs and gateways. This can significantly reduce the time needed for professional services as there is no need to configure hundreds of classification rules, trunk groups, profiles and routing rules. And all of this is provided through an intuitive and simple graphical user interface.
  • The AudioCodes Routing Manager also looks at user attributes to optimize routing calculations. It imports and aggregates user information and huge dialing plans from different sources (e.g. LDAP server and CSV files) and groups users and dial-groups for routing calculation and implementation of number portability.

The AudioCodes Routing Manager is a critical solution. It reduces the operational time spent on designing and provisioning the network topology; it dramatically reduces OPEX by avoiding routing configurations of many VoIP network elements; it lowers the need to rely on telephony experts; and it reduces the time spent on adopting evolutions in the network.

Learn more about the ARM: http://www.audiocodes.com/audiocodes-routing-manager-arm

The ARM was recently featured in an article on No Jitter: http://www.nojitter.com/post/240171460/audiocodes-tackles-simpler-session-management

Routing

Super Creativity in Routing Diversity

Routing

“Creativity is just connecting things. When you ask creative people how they did something, they feel a little guilty because they didn’t really do it, they just saw something. It seemed obvious to them after a while. That’s because they were able to connect experiences they’ve had and synthesize new things”- Steve Jobs.

Recent researches show that “super creativity” is the way the brain makes the connection between different segments of the brain. Furthermore, it’s all about amount and diversity of connectivity. What’s connectivity? Brain anatomy (circuitry) or routing?

Perhaps both?

Karl H. Pfenniger (origins of creativity; Oxford 2001) claims that creativity involves the association of diverse and apparently unrelated entities and that the brain’s plasticity allows for neurons to adapt their connectivity to function. In other words, new semantic connections will not only be associated or correlated but over time will become hard-wired in the network of associations. So creativity is related to routing diversity.

Routing in the network layer

Data routing’s main pride over the last two decades is the distributed routing architecture. Using various routing protocols (RIP, OSPF, EGRP, BGP, etc.), networks are created and shattered dynamically and automatically. Routers are totally independent and are allegedly the ultimate “plug and play” products. Hmmm…. that’s almost true. For every product there are usually two parties – the vendor and the user. Routers are the first products which users can’t use without the assistance of a mediator – “network engineers”. That’s because while a single router configuration may be simple, designing an architecture of a complete network of distributed routers is very complex.  Over the years there have been several failed initiatives (e.g. FORCES RFC, AT&T RCP) to standardize centralized routing architectures. The emergence of Software-Defined Networking (SDN) reshuffles the cards. SDN architecture is dynamic, manageable and mainly cost-effective. SDN decouples the network control and forwarding functions, enabling the network control to be centralized.  The underlying infrastructure is abstracted and used by applications and network services.

VoIP level routing

Routing in VoIP architecture lags considerably behind data routing. VoIP routing is much more complicated than data routing. Data routing is based on layers 2-4, while VoIP routing rules includes parameters from layers 2-7.  Since the initial design of the enterprise VoIP network gave control to the Soft-Switch or IP PBX, routing was typically based on static routing tables. The situation becomes complicated for off-net calls. IP-PBX/Soft Switches “get rid” of such calls by directing them to the nearest GW or SBC.

In the case of multi-branch organizations, a single VoIP network may have several IP PBXs from different vendors, as well as MGWs and SBCs. The complexity of such heterogeneous VoIP networks is rising all the time. Each IP-PBX, SBC and MGW in the network has its own static routing table; a distributed system in which the included elements don’t communicate with each other.

SBC vendors have tried to solve this complicated situation in two ways:

  • Routing Manager – a static application which helps to construct huge dialing plans and routing tables and then download them to the SBCs in the network as static tables. These files are large and complicated, allowing vendors to charge a high price for the professional services that build them.
  • Session Router -a “primitive” SIP proxy located in the center of a star formation network that performs hop-to-hop routing of the call signaling. The Session Router doesn’t solve the problem as it dictates that all the VoIP signaling will be directed via one route and leaves the routing to the VoIP peers to the SBCs and MGWs.

“Shoot for the moon. Even if you miss, you’ll land among the stars” – Norman Vincent Peale.

The scalable & manageable VoIP routing solution

VoIP Routing Controller (VRC) is a holistic, dynamic routing manager based on Software-defined Networking principles. The VoIP routing controller decuples the device layer from the network routing and policy layer, automatically creates complex VoIP networks, and simplifies routing rules, monitoring and management configuration.

Organizations’ VoIP networks are constantly evolving due to a variety of factors; mergers & acquisitions, new locations, integration of new & old IP-PBX (or even PBX) and integration of SBCs and GWs. In addition, there is organic growth of an organization and accompanying technology evolution such as introducing UC to the network and consolidating IP-PBX. All of this makes managing the organization’s VoIP network a nightmare.

Unlike the routing manager and the session router mentioned above, the VoIP routing controller is neither a configuration element nor a SIP proxy (or any other SIP element for that matter). The VRC is a dynamic routing controller which calculates end-to-end routes.

All SIP network elements (i.e. SBCs, GWS) register to the VoIP Routing Controller automatically as they boot-up. The SBC and MGWs update the VRC with all the peer connections, (the SBC / MGW connections to IP-PBX, PSTN, SIP trunk, user interfaces, etc.) The VRC learns about the SBC and MGW automatically, and it’s done dynamically. Every change in connectivity and configuration is reported to the VRC.  Eventually, the VRC has the complete network topology connectivity and health picture.

The VRC also assists with the VoIP network design and creation. With the VRC, a network can be formed in one-click. The administrator can choose between mesh, star or dual star formations and all the connections are automatically configured in the SBCs and MGWs.. This literally saves days of professional services as there is no need to configure hundreds of classification rules, trunk groups, profiles and routing rules. Everything is done automatically by VRC. For customized network connections, the administrator just draws a line and the connection is configured automatically in the SBC or MGW. The same is true regarding erasing and modifying connections. Configuring connections between SIP elements is as simple as drawing a line. And all of this is provided through an intuitive graphical and simple user interface.

The VoIP Routing Controller deals with users’ attributes to optimize routing calculations. It imports and aggregates users’ information and huge dialing plans from different sources (i.e. LDAP server and csv files) and groups user and dial-groups for routing calculation and implementation of number portability.

How does this work? When an SBC or MGW receives a call, it sends a query to the VoIP Routing Controller by REST API. The query includes the entire set of call information (i.e. sip:invite) The VRC calculates the entire route end-to-end and sends the entire route back by REST API. From there, SBCs and MGWs perform the call and when the call terminates, the last node in the route sends a notification to the VRC. The routing calculation may be done using a variety of decision criteria, including: priority, time based, least cost, quality, connectivity, user and dial group..

Testing Creativity

Close your eyes and imagine ten or more different animals. OK, now open your eyes and write down the imaginary list.. This simple test can differentiate between Walt Disney and Steve Jobs and the rest of us mere mortals. Where most of us will imagine cows, donkeys, sheep and maybe a lion, in the “Super Creative” list you will find anything but “regular” animals, instead you will find the likes of Capybaras, Fennec Foxes, Kinkajous, and Servals.

Routing rules and manipulations are complex and can’t be implemented on-line. Therefore, one of the VRC’s important features is the “test route” and “test manipulation” which allows for testing any rule and policy that was created in the VRC and immediately seeing the impact on the routing and policy.

Conclusion

The VoIP Routing Controller is a very much needed solution.  It reduces the operational time spent on designing and provisioning the network topology; it dramatically reduces OPEX by avoiding routing configurations of many VoIP network elements; it lowers the need to rely on telephony experts; and it reduces the time spent on adopting evolutions in the network.