Posts Tagged: pbx

SIP Trunking – In depth Guide and Tips

 

Sip Trunking guide navigation:

  1. What is SIP Trunking
  2. How does SIP Trunking work
  3. Benefits of SIP Trunking
  4. Main Disadvantages
  5. SIP Trunk Migration Tips
  6. Is SIP the Right Choice

 

What is SIP Trunking?

SIP Trunking uses VoIP to connect a PBX and the Public Switched Telephone Network (PSTN). It is an application layer protocol that sets up audio or video sessions between two devices. As such it replaces the traditional phone trunking. SIP requires on-site PBX and a connection to your Internet telephony provider.

Trunking to a Hosted PBX is typically done using SIP. Another solution would be SIP PRI (Primary Rate Interface). When compared, SIP is the more scalable and profitable service. While PRI offers better quality of voice transfer but has some limitations to its usability.

With SIP trunks, companies can connect multiple channels to a cloud PBX. This allows companies to make simultaneous calls over the existent Internet connection. Without restrictions to the number of incoming or outgoing concurrent calls.

Why SIP Trunking:

  • Costs less
  • Scales easier
  • Is more reliable
“SIP is a well-established technology with minimal upfront investment costs”

 

How does SIP Trunking work?

SIP manages signalling and access to a Voice IP network. Trunks are a mediator between your phone systems and your Internet Telephony Provider. In other words it provides a connection between your telephony system and your provider of choice.

With SIP Trunking need of a physical connection to a phone company is obsolete. You no longer need that old wiring, or the circuit boxes used for connection to the PSTN.

“Just think of SIP trunking as a virtual version of an analog phone line.”

Benefits of SIP Trunking

“The cost savings and benefits of SIP trunking are hard to neglect.”

Main sell points of Sip:

  • Cost Savings
  • Disaster Recovery & Security
  • Cheap Long distance calls
  • Extremely Scalable
  • No infrastructure
  • Mobility
  • Remote access to lines
  • Reliability
  • Less constraints
  • Cost Savings

 

SIP trunking saves a significant sum of money reaching from 20 up to 60% of company telecommunications costs. Additionally it saves time and if implemented correctly reduces stress.

SIP trunks are virtual and provided over Internet making it easy to add or remove capacity. It’s easy to scale, add or remove team members when required, often at minimal costs and without hardware requirement.

Hardware and maintenance

Hardware and maintenance are almost virtual as well. No wiring is required, no on premise hardware is needed and maintenance is not mandatory. Even tech support is often remote.

Recovery and Security

In case of emergencies, Sip trunking is actually quite helpful. It is easy to setup a rerouting, and backing up with a spare internet provider is not that expensive. The service is Cloud based and does not have many of the limitations of on site PBX systems, especially when it comes to power losses. Almost every provider has a failover countermeasure and recovery strategy.

As every technology Sip trunking has its security exploits and vulnerabilities. Data safety is a big concern for most companies and should not be neglected. You can find a list of possible glitches here. Any Sip trunk provider should walk you through the possible breaches as part of the provided service.

Mobility

SIP can be used with VoIP apps that come a very high degree of mobility. It allows company employees’ to connect with mobile devices or from remote computers. Some VOIP/SIP providers have even developed VoIP applications that require no data connection. Those apps can be used everywhere a mobile connection is available. With them mobile workers can remain connected to the communication system, and be reachable at all times.

Remote access to lines.

SIP supports remote access which is particularly handy for companies with offices based in multiple locations or countries. For outgoing calls it allows the use of all available numbers, no matter where you are located at the moment. Thus redirecting the incoming calls to your office of choice. SIP is extremely scalable, allowing startups and fast growing companies to open new sites or use full-time remote workers regardless of geographical location.

“You can make calls and still assure professional pick up in a different office or even in a different country.”

No Physical Infrastructure

“SIP trunking does not require additional hardware investment.”

Moving and expanding the company infrastructure before SIP trunking ware challenging. Both costed a hefty sum and ware lengthy processes. With SIP adding a new line is as simple as putting on a new pair of headphones and scaling your service agreement. Off-site workers are easily integrated with applications and can be reached with a simple extension number. Moving is fast and virtually free as no on-site hardware is required.

Reliability

Traditional telephone service is often reliable. But it can still interrupt at times, because of construction works, bad weather conditions, or simple human errors. And if the line is broken there is no viable workaround. SIP trunking is a bit more reliable solution. Internet backup line is almost a must and guarantees calls will get through even if the main data provider fails. Cloud servers often have backups and can be restored within minutes, and vendors often offer mobile failover. Mobile VOIP applications can take over the SIP trunking functions until the service is restored.

Simple Management

Extensions management and functions like call pickup, call routing can be managed through simple administrative portals with a few clicks.

Most providers offer full control of all the outgoing and incoming calls, as well as real time monitoring and statistics. It is easy to check who used which phone with which extension, who was called and now long the calls lasted.

In addition, SIP trunking also:

  • Works with most modern phones
  • Can use use your old phone numbers
  • Can help you rationalise your telephony
  • Is perfect for seasonal companies
  • Can have flat plans

Main Disadvantages of SIP

 

Bandwidth Costs

Migrating all your communications to SIP always comes with the need of better bandwidth. Depending on the number of simultaneous lines that are to be used the additional cost might vary from a small sum to a small fortune. Despite the ne Internet connection costs, the final balance is still in SIP Trunkings favor.

Simultaneous Calls

When a company starts using SIP trunking it often chooses the wrong plan. The number of simultaneous calls is mostly ill-considered and needs to be re-assessed. Companies often miss-judge how many lines they will need and need to change their SIP and data plans.

“To determine the number of simultaneous calls you have to take into account both incoming and outgoing calls. Calls that exceed the limit will receive a busy signal and won’t wait in line for pickup.”

Porting Numbers

When switching to SIP all your DID numbers can be kept but have to ported to the new system. This process is not complicated but can take a good amount of time.

Possible SIP Trunking problems if it is not implemented correctly:

  • Call recording – can become messy
  • Fax – highly dependant on providers and might not be supported
  • 911 – proper implementation is required
  • Long carrier lead times
  • Number porting can take months
  • DTMF might not work

Successful SIP Trunk Migration Tips

 

Migration to SIP-Trunk

 

SIP trunking migration might become a complicated process if not planned and executed correctly. Here are some tips that will help the implementation:

Planning

Plan everything. Begin with inventory of the existing infrastructure. Number of phone lines, DIDs, check the overall usage both outgoing and incoming. Note if you will need to use the system from other locations. Don’t forget to include the backup system.

Key Decisions

SIP Trunking plans and carriers are flexible but you should make some firm decisions.

    1. Choosing the right voice compression between clearer sound and reduced bandwidth?
    2. Choosing the backup bandwidth (to fully cover or to be just for emergency)?
    3. Choosing what to do with the old hardware
    4. Choosing the carrier
“Choosing the Carrier for your SIP Trunking should not be left as the last decision. You should carefully review the available carriers, judging to what grade the offers they make are in line with your demands”

Expected Carrier Cost

In the end it is not all about the money. But the cost should not be neglected. Not all carriers list the following costs for comparison yet if asked they should provide the numbers:

Non recurring:

  • Initial Implementation
  • DID porting
  • Initial Training

Recurring:

  • DID numbers
  • Toll-free numbers and features
  • Number of lines
  • Caller Identification
  • Cloud Services
  • Support

 

Preparing for SIP

When you have determined what you need from your provider, and who it will be you should start planning the implementation. Every SIP provider has its course of action and you should make sure it is lined up with your needs. Be sure to take a test flight before you start the full implementation:

  1. Create a test plan
  2. Test the quality of service
  3. Create and test new DIDs
  4. Port few of your existing DIDs for testing
  5. Assign few checkers form your team that will gradually and troughtly test all new changes and options
  6. Test the internet bandwidth and backup
  7. To test the provided support, check time to action and availability
  8. Check the toll-free and 911 numbers for traffic and availability
  9. Check the control panel, reports and call logs

 

Potential Problems

As with every new undertaking, it is common for problems to arise. Consider several potential pitfalls areas that should not be neglected:

  1. Call recording, and storage might not work correctly or be messy
  2. Faxing might not work or need to stay on traditional line
  3. Tall free numbers and 911 numbers might not be connected or misbehaving
  4. Carrier screening or spoofing of outbound Caller IDs
  5. Carrier lead times might be too long
  6. Number porting might take too long
  7. Control panel might not work correctly
  8. Call logs reports and statuses might misbehave or be hard to read
  9. DTMF might not work

Is SIP Trunking the Right Choice?

SIP Trunking isn’t an alternative to hosted or on-premise PBX. It’s an alternative to publicly-switched telephone network. Is well-established and can lead to significant improvement in reliability and cost savings. The offered flexibility is a big plus but as any other solution it has more than a few disadvantages.
Still SIP Trunking is the right choice for a lot of companies which have left high priced and unreliable phone lines in the past and moved their business communications to the Cloud

 

Advice – Communications for Enterprise

Lets address some points that should be taken under consideration for IT managers looking at packetized communications for their Enterprise.  Below are the highlights and my thoughts:

Highlights

Look For SIP Support

I couldn’t agree more!  SIP is a protocol used to establish, teardown, modify, etc communication sessions.  It’s very diverse and relatively simple when compared to past mechanisms.  Most importantly, it has become the defacto standard within the world of telephony.  There’s native SIP support in nearly all the major vendors that supply VoIP gear. (Cisco, Avaya, Siemens, Microsoft)

Consider The Benefits Of Hosted PBX

This topic has be discussed numerous times in the past, and even before that within a TDM context (PBX vs. Centrex).  The thing that’s different within an IP context is the feature and functionality available.  When comparing a PBX to a Centrex offering, one key difference was additional feature and functionality in a PBX.  Centrex offerings didn’t have the same “whiz-bang” features.  In today’s Hosted Telephony offerings, there’s near feature parity, so the key determining factor becomes cost of ownership.

Think Unified Communications

VoIP (or Telephony) MUST be seen as a stepping stone to the ultimate goal of Unified Communications.  IT managers should consider the roadmap to UC when choosing a Telephony solution.  Real-time communications need to become multi-modal, meaning there should be options to transition communications from IM to voice to video to online collaboration on a document, and then back again – all within the same context and within a common look/feel.

Traversing NAT

Though the issue of Network Address Translation (NAT) is well known to negatively impact SIP sessions, the real point for consideration here for the IT Manager should be around considering the deployment of a Session Border Controller (SBC) within their Enterprise as part of an overall design.

Know & Apply Codecs

There are more ways to packetize voice and video communications than one can shake a stick at.  The author points out the predominant technologies of G.711 and G.729.  Issues of bandwidth consumption and quality of user’s experience must be balanced.  Generally speaking, the more bandwidth consumed, the better the experience.  But the more bandwidth used, the greater the cost to upgrade the LAN/WAN infrastructure to accommodate.  If you skimp on cost, the result would be poor quality, and then adoption and experiences would suffer.  It’s a delicate balancing game.

Some further comments :

• Make sure to have 100k in bandwidth free and available for every conversation when determining whether the enterprise really has enough bandwidth for VoIP.  With multiple calls made from one location a simple DSL  won’t cut it.

• Get VoIP phones that are both wired for Ethernet and wireless for Wi-Fi connectivity. That way, people can wander, and all internal calls within the building are free of charge because they stay on the network. Check the mobile voip solutions, few of them even work without need of data plan.

• Make sure the vendor is going to be around to support the purchase.  An older vendor with roots, commitments, and financial means is an obvious choice. A new vendor with strong management, skills, and reputation who proves out through considered research can also be a sharp choice.

VoIP Considerations for IT Decision-makers

VoIP Decisions

Lets address some points that should be taken under consideration for IT managers looking at packetized communications for their Enterprise.  Below are the highlights and my thoughts:

Look For SIP Support

I couldn’t agree more!  SIP is a protocol used to establish, teardown, modify, etc communication sessions.  It’s very diverse and relatively simple when compared to past mechanisms.  Most importantly, it has become the defacto standard within the world of telephony.  There’s native SIP support in nearly all the major vendors that supply VoIP gear. (Cisco, Avaya, Siemens, Microsoft)

Consider The Benefits Of Hosted PBX

This topic has be discussed numerous times in the past, and even before that within a TDM context (PBX vs. Centrex).  The thing that’s different within an IP context is the feature and functionality available.  When comparing a PBX to a Centrex offering, one key difference was additional feature and functionality in a PBX.  Centrex offerings didn’t have the same “whiz-bang” features.  In today’s Hosted Telephony offerings, there’s near feature parity, so the key determining factor becomes cost of ownership.

Think Unified Communications

VoIP (or Telephony) MUST be seen as a stepping stone to the ultimate goal of Unified Communications.  IT managers should consider the roadmap to UC when choosing a Telephony solution.  Real-time communications need to become multi-modal, meaning there should be options to transition communications from IM to voice to video to online collaboration on a document, and then back again – all within the same context and within a common look/feel.

Traversing NAT

Though the issue of Network Address Translation (NAT) is well known to negatively impact SIP sessions, the real point for consideration here for the IT Manager should be around considering the deployment of a Session Border Controller (SBC) within their Enterprise as part of an overall design.

Know & Apply Codecs

There are more ways to packetize voice and video communications than one can shake a stick at.  The author points out the predominant technologies of G.711 and G.729.  Issues of bandwidth consumption and quality of user’s experience must be balanced.  Generally speaking, the more bandwidth consumed, the better the experience.  But the more bandwidth used, the greater the cost to upgrade the LAN/WAN infrastructure to accommodate.  If you skimp on cost, the result would be poor quality, and then adoption and experiences would suffer.  It’s a delicate balancing game.

Some further comments captured in the article.

• Make sure to have 100k in bandwidth free and available for every conversation when determining whether the enterprise really has enough bandwidth for VoIP, according to Andy Abramson, blogger at VoIP Watch and founder and CEO of Comunicano (www.comunicano.com). With this understanding, it’s easy to see that DSL or a cable modem line won’t cut it.

• Get VoIP phones that are both wired for Ethernet and wireless for Wi-Fi connectivity, notes Abramson. “That way, people can wander, and all internal calls within the building are free of charge because they stay on the network.”

• “Make sure the vendor is going to be around to support the purchase,” Abramson says. An older vendor with roots, commitments, and financial means is an obvious choice. A new vendor with strong management, skills, and reputation who proves out through considered research can also be a sharp choice.

 

How to Choose a VOIP Provider and How Important Is It For Your Business

 

VOIP is a telecommunications technology that replaces and upgrades the legacy Public Switched Telephone Network cabling. Its only drawback is that it requires Internet, but in return VOIP provides secure access and lots of extras important to business telephony.

VOIP BENEFITS

 

Transferring to VOIP is almost mandatory if you are running business that relies on telephony or if you are simply planning to change your telecom system. VOIP offers important benefits, both financial and operational. It simplifies the infrastructure and reduces call, maintenance and moving costs.

A growing trend

Growing numbers of businesses have already moved to VOIP realizing these benefits. The trend is not surprising considering the lower costs for businesses. According to analysis switching from legacy systems to SIP trunking lowers the total cost by 50%.

Upgrade legacy systems

VOIP gives the business access to the all the benefits of Internet telephony. With the new infrastructure Voip connects your PBX to an Internet Telephony Provider, benefiting you from the lower cost of long distance calls and professional pbx functions. You can even use rented cloud PBX to lower maintenance and infrastructure costs.

Connect your branches

If have interstate branch offices and want to connect them to the communication facilities of head office, you can connect them with VOIP. This can be done by connecting the sites to your infrastructure or using a service provider that can connect your facilities with cloud PBX. Both ways, you provide your branches with improved communications eliminating duplicate infrastructure while effectively halving the support staff needed.

Simplify infrastructure

Operate separate networks for data and voice is not cost effective and is even harder to backup. You can make significant savings with VOIP transition. using a single channel for data, voice and video will reduce network complexity and you will cut the costs for upgrades, backup lines and maintenance.

Mobile communications

As mentioned before the main disadvantage of VOIP communications is the requirement for internet connection. It is not much of a problem when you are calling from the office but field workers don’t have that privilege. Some VOIP providers like MIXvoip recently solved that issue by using DTMF allowing mobile VOIP calls to be made even without internet connection.

Selection factors

With that many benefits, how would you proceed when choosing a VOIP provider? Which are the most important factors to consider?
As people that have been in the VOIP business for quite some time now, we recommend you to choose by:

  1. Security
  2. Quality of Support
  3. Extended Services
  4. Pricing
  5. Mobility
  6. Call Quality

 

Security

Security the most important factor because VOIP is an Internet technology. As such it can be subject attacks and other risks that come with that environment. Service providers offer different security measures and techs. Software measures include authentication, encryption and the use of secure real time transport protocols. Hardware security is mostly done with the use of separate network access routers it is called transport-layer security and is used to protect your network against attacks that can disrupt the phone service.

Quality of support

Quality of support really important too and definitely should be considered. As with every service, at some point problems will arise and you should be confident that your provider can minimize the downtimes. VOIP is an Internet service often hosted on cloud servers, as such businesses don’t need support staff on site, the service provider support teams have the responsibility to keep everything running.

Pricing and mobility

Pricing and mobility are mostly self explanatory. They are both very business specific and depend on individual business structure and needs. Some providers charge flat, based on number of devices, others charge based on the number of calls your business makes. Most VOIP providers do not offer unlimited calls but there are exceptions. If you have lots of field workers you should make sure the company offers a mobile app.

Call quality

Call quality is not that important but shouldn’t be neglected. Most providers promote on call quality but you should know that sound quality requires more data. When more data is used you can make less simultaneous calls without call interferences and you will need to upgrade your internet plan.

VOIP is essential if you want to enjoy the advantages of business communications, save money and achieve business, financial and operational benefits.

Top 10 Free PBX Solutions – detailed comparison

Which are the top free PBX business solutions

Adopting an existing VOIP service from the top PBX providers will get the job done for most businesses. But no one said it is easy to find the right one. Providers do their best to fit the needs of every business on a case-by-case basis. Nevertheless, with gaps in usage cases, business team sizes, required features or safety, it may be about impossible satisfy everyone. For all those cases where a service won’t fill in the gaps, businesses can turn to open source solutions or platforms. And yes there are open source PBX software solutions out there. With them any business can develop and tailor their own in house PBX applications. Since these platforms are open source, all the source is available for free online, and can be tailored for every specific scenario.

Asterisk

Asterisk is the godfather of all open source PBX and VOIP solutions. And it continues to function as gold standard. It is the leading open source telephony platform, with an enormous feature lists which only continues to grow each year. The Asterisk tool kit is used by a mass quantity of developers around the world. Many of the suppliers on our list have either begun with, or are based completely on the Asterisk project. It is packed with the standard PBX VOIP features. Consisting of automatic telephone, an interactive voice response menu, conference calling, and voicemail. Asterisk makes it easy workable to turn any computer. The program is free and open source. To help get you started, Asterisk provides live web classes, as well as an Asterisk Definitive guide. Asterisk can be considered The Platform when it comes to creating your own VoIP or PBX.

SIPFoundry

SIP Foundry provides much of the solutions that the Asterisk engine can power. With SIPFoundry you can construct your own voice and video communications. In addition it support conference calls, messaging and chat. Like with Asterisk, the platform includes everything you might need to build your own PBX solution. Nevertheless, whilst Asterisk is 100% free and open source depending on guides. SIPFoundry has a somewhat different spin. It offers professional paid assistance to developers based upon customer needs. While an excess cost to think about, adopting a support team might be a essential step for some businesses looking to build their own system. With a focus on the marketplace, the tool is free for commercial or private use. The project has also brought in features from other open source projects. HylaFAX, FreePBX, Openfire and Postfix.

Elastix

Elastix intends to bring in Asterisk’s features and other such projects, all under 1 easy-to-use interface. Elastix boasts support for a wide range of hardware consisting of Yeastar, Dinstar, Digium, Yealink and Snom. It was actually 1 of the first distributions that included a call center module. And it proceeds to supply the solutionfree under the GNU General Public License. It’s come to our attention that Elastix appears to provide up to 8 SIM calls for approximately 25 users, since writing this post. FreeSWITCH was also based off the Asterisk platform, and was created and developed by 3 of the original programmers of the Asterisk platform. Anthony Minessale II, Brian West and Michael Jerris.

FreeSWITCH

FreeSWITCH is with a focus on modulator, cross-platform service, availability and stability. It provides 1 of the most flexible platforms to construct your own UC package. FreeSWITCH supports SIP, H.323 as well as WebRTC to leverage the latest advancements in the technology. It can integrate and interface with other some of the other open source PBX platforms. For less complexity FreeSWITCH utilizes open software libraries that preform the essential functions. FreeSWITCH delivers the calling features and some extras such as speech recognition. It even offers PSTN ports for digital and analogue circuits.

Voicetronix

Voicetronix is equipment supplier and a solutions that offers an open source platform, but also an assortment of hardware. The do-it-yourself OpenPBX of Voicetronix is a web enabled PBX program. It comes with a web based user management portal, in addition to a management GUI for easy and rapid configuration. It is feature rich with car attendant, automatic call distribution call routing search groups and even voicemail. Unique features like call hunt groups, music on hold and call records are good solution for businesses in need of a basic call center software. With CRM baked and enabled into the platform, users may not require to adopt a separate CRM solution, saving time and money.

PBXInAFlash

PBXInAFlash main feature is the ability set up your own PBX server in no time. The project has everything needed to set up a PBX system in under one hour. It uses CentOS, with integrated Apache web server, SendMail server, and MySQL database. In addition to firewalls and all essential protocols. Users have the option to pick from dozens of add-ons to tailor the system. Backups, Caller ID look up services, SSL keys, Google Voice integration, and fax support to name a few. With the number 1 goal of no bloat and no bugs, PBXInAFlash seems to be easiest and the quickest solution to adopt.

FreePBX (GUI)

FreePBX combines the best of both worlds, and leverages the work. While the project uses the Asterisk system, users may download either just the GUI to add on to their current system, or the whole package. It consists of a per-configured program OS, Asterisk, and the FreePBX GUI. So while by adopting Asterisk, some knowledge may be required to take advantage of, or to create your own GUI, FreePBX brings it all together. FreePBX makes it feasible to establish your SIP Trunks that are part of the platform thanks to the integration. FreePBX also contains a long list of commercial modules and add-ons to enhance your system with even more features.

OpenSIPs

With a focus on open source implementation of a SIP server, with OpenSIPs its easy to set up your own PBX. The platform supports video, voice, IM and presence services. It is using modular design, it is scalable, and very much customizable. OpenSIPs enterprise class SIP server solution and a very fast one at that . OpenSIP has made a list of benchmarks and performance tests to back their claim up. Similar to Asterisk, OpenSIPs often records webinars, and makes in depth manuals for configuration. A web interface makes it easy to collect data and shows on the fly configurations.

Kamailio

Kamailio is an open source project with 15 years of constructive development. And while the original company left the project, it continues to expand, both the SIP server and Kamailio project continue to build on. With features like UDP asynchronous TCP and SCTP, TLS to ensure secure communications. VoIP data consisting of voice video and text, and even WebRTC support the hard work can clearly be seen. Kamailio also supports instant messaging, least cost routing, load balancing, routing fail-over. Realizing that security features are important they are offering the strongest level of security on this list. Authentication and authorization for enhanced security as well as the level of encryption that the platform gives makes it a good recommendation for any staff or business that needs to keep everything as locked down and protected as possible. As a result of all that, Kamailio may be a bit more challenging to adopt.

3CX

The 3CX Phone System is the last open source PBX based upon the SIP standard on the list. This solution allows extensions to make calls on the PSTN or standard services. The platform also offers an easy to understand web based GUI, and the process to is actually simple – an executable file. 3CX supports iOS and Android for mobile customers, of coarse Windows and Mac softphones are supported too. It appears to take out the hassle of development required to establish your own PBX server. WebRTC adoption makes web conferencing possible. Click2Call and CRM are also part of the features. As most others, internet training academy is readily available for users to understand how to manage the platform.