What is a PRI Card?

A PRI Line is a Digital Trunk Line that is provided by a Telecom service provider which enables users behind a PBX to make 30 simultaneous incoming/outgoing calls using a single line. You can read more about PRI Lines from here.

A PRI Card is used to connect PRI lines to IP PBX/ IP Telephony Server so that all the IP Phones/ Analog phones (extensions) can make outgoing calls or receive incoming calls using it.

In an Analog/IP PBX, one needs to procure a specialized PRI Card that fits into one of the empty slots of the PBX in order to connect the PRI Line. This Card is mostly proprietary to the specific PBX vendor. But with Soft-Switches (that run using standard server hardware), one can purchase a generic PRI Card to interface/ connect the PRI line.

The generic PRI Cards are generally vendor neutral and they are inserted into PCI 3.3V/ PCI 5V/ PCI Express (empty) Slots in the server. There are different PRI Cards for each type of PCI interface.

One PRI Card can have 1,2 or 4 Slots to connect to 1,2 or 4 PRI lines. Some PRI Cards come with echo cancellation modules (at extra cost) in order to reduce the echo generated when Digital Signals are converted to IP & vice versa. It is recommended to buy PRI Cards along with Echo Cancellation modules.

What is an Analog Interface Card?

An Analog Interface Card is used to connect Analog Trunk lines and/or Analog Extensions (Analog Phones) to the IP PBX/ IP Telephony Server.

Well, there are two types of Analog Interface Cards – Analog Trunk Cards & Analog Extension Cards. The Analog Interface cards are similar to  PRI Cards and connect to the PCI Interface of a Server. Of course, they are applicable for a Soft-Switch (Software based PBX) run on a generic Server.

Analog Trunk Card connects the Analog Trunk Lines from your Telecom service provider to the Soft-Switch (IP Telephony Server). 4-Port, 8-Port, 24 Port Analog Trunk Cards are popular and they connect to 4,8,24 Trunk Lines respectively in the same card which occupies one PCI slot. An Analog Trunk Line is similar to the Land-line phone connection that we have at home – Each Trunk line can transmit/receive a maximum of one phone call at a time.

Analog Extension Card connects the normal Analog Phones directly to the Soft-Switch (IP Telephony Server). 4-Port, 8-Port, 24-Port Analog Extension Cards are popular and they connect to 4,8,24 Analog Phones respectively using the same card which occupies one PCI Slot.

There are Certain Hybrid Cards that have various combinations of Analog Trunk ports & Analog Extension Ports. For example, an 8-Port Hybrid card may contain 4 Analog trunk ports and 4 Analog extension ports (or) any other combination.

In case of 4-Port/8-Port Cards, the Trunk Lines and Analog Phones can be directly plugged in. But in case of 24-Port cards, a Cable with 48 connections need to be patched on a Patch panel and the phone lines need to be patched at the other end of the same patch panel to complete the connection.

Apart from Analog Interface Cards, there are specialized stand-alone devices called ATA’s – Analog Telephony Adapters that can connect to Analog Trunks or Analog Extensions at one end & the IP Network (and hence the IP Telephony Server) at the other end.

There are VOIP / IP Trunk Lines called SIP Trunks and no special interface cards are required to connect them to the IP Telephony Server. But SIP Trunks are not available in all countries.

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What are Telephone Trunk lines?

Telephone trunk lines are the ones that enable users to make outgoing calls / receive incoming calls in an organization. We are talking about land-line calls, not cell phones!

So, in an organization there might be many users with either IP Phones or Analog Phones and they connect to the IP PBX or Analog-Mixed PBX. This is on the user end.

On the other side, you can get some trunk lines (Analog Trunks, ISDN Lines/ PRI-E1-T1 Lines, etc) from a Public Telephone Exchange or Telecom Service Provider (PSTN) and terminate them to the PBX as well.

So, when a user makes an outgoing call with his analog phone/ IP phone, the call reaches the PBX and the PBX will connect the call to any of the free Trunk Lines. Similarly, when there is an incoming call, it lands up on the PBX and the PBX routes it to the appropriate extension.

Click here to read the difference between Analog Trunks, PRI/E1/T1 Lines and Cellular Gateways. They are all Telephone Trunk Lines and they are also called PSTN Trunks (Public Switched Telephone Network).

What is a SIP Trunk?

One of the biggest advantage of VOIP / IP Telephony is its ability to enable users to make inexpensive long-distance calls as these calls are transported on the Internet and hence it can by-pass the normal toll charges applicable for analog & digital trunks for inter-state / inter-country calling.

So, a SIP Trunk is a virtual IP based trunk line that uses the Internet to make calls / receive calls to land-line numbers, cell phone numbers and other VOIP numbers. The SIP Trunk is terminated on the PBX (both IP PBX and supported Analog-Mixed PBX models) and the user’s call can be sent over the Internet.

There is a SIP Trunk provider called ITSP – Internet Telephony Service Provider who has connections with other VOIP Service providers/ PSTN Service providers to enable VOIP phones to call PSTN phones and vice versa.

So, when an international call is made by the user, the PBX connects this call to the SIP Trunk, it reaches the ITSP over the SIP Trunk (via the Internet), it is carried to the destination country/city over the Internet / packet switched network by the ITSP and then transferred to the local PSTN exchange over there to reach the destination number.

What are Private SIP Trunks?

SIP Trunks also refer to the Interconnectivity of two private IP PBX’s located at different places over the Internet / IP Network. This creates a (Virtual) Private Network between the two of them so that users at one location can call/receive calls from users in the other location without incurring additional call charges.

For example, if there are two branches of the same company in City A and City B, the PBX in City A can be Trunked over the Internet/ Leased Lines/ IP Networks (using SIP Trunks) with the PBX in the City B. The calls between the users of these two locations do not incur additional cost as these are internal calls carried over existing network.

Salient Features of SIP Trunks (Public/ ITSP):

• SIP Trunks connect to the Enterprise PBX over the Internet through an ITSP – Internet Telephony Service Provider/ Business VOIP Service provider. They enable VOIP users to call PSTN numbers and vice-versa.
• SIP Trunks are virtual circuits that use the existing Internet connection (Internet Leased Lines) for physical transport of the voice packets.
• Using SIP Trunks, users can make calls to local PSTN numbers, long distance land line numbers, cell phone numbers and other VOIP numbers.
• SIP Trunks come with their own telephone numbers, just like PSTN phone numbers which can be reached from anywhere. It is possible to get a local phone number with a local area code (and) toll free numbers.
• For each IP call, it is recommended to provision at least 80 kbps of upload / download Internet bandwidth. So, symmetric Internet connection like Internet Leased Lines are always better than business broadband.
• ITSP’s offer calling plans for SIP Trunks like normal telecom/ PSTN service providers. There is either a fixed rental for each line every month  offering some free calls and charging for additional calls (or) there are unlimited calling plans for each line with a flat monthly rental.
• SIP Trunks can be obtained quickly (usually within a single business day) and individual line increments/ decrements are possible.
• SIP Trunks can provide Local Number Portability, Direct Inward Dialing, Enhanced 911 Services, and other services provided by E1/T1 trunks.
• SIP Trunks can co-exist along with Telecom/PSTN trunks in a PBX and the PBX can choose the best trunk for each call, based on certain pre-programmed parameters.

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Advantages of IP PBX when compared to Analog/Digital PBX (IP PBX Vs Traditional PBX):

1. Single Network: This is one of the biggest advantages of of IP Telephony systems. Traditional PBX require their own network and it is quite expensive to build a separate telephone network spanning the entire premises. With IP Telephony, the Computer Network can be used to carry voice calls along with data and the IP Phones connect directly to Network Switch Ports using Cat 5/6 Cables. Most of the IP Phones come with an inbuilt two-port switch that eliminate additional switch ports and cables required for their connectivity. Redundancy is an issue, and the IP Phones stop working if the computer network is down, but redundancy can be built into IP networks using technologies like Link Aggregation, RSTP, etc.

2. Inter-branch Calls: If IP Telephony has been deployed in multiple branches (in different locations) of the same company, it is possible to use MPLS Networks/ Internet Leased Lines (With Unlimited Usage plans) to transmit voice calls over the WAN IP Network. This way, the inter-branch calls would not incur additional costs.

3. Long Distance Calls: Its possible to terminate SIP Trunks from ITSP’s (Internet Telephony Service Providers) directly to the IP PBX. So, international calls and long distance calls can be made through the Internet for lower cost. Concepts like DISA (Direct Inward Station Access) allow a user to dial to the corporate IP PBX (from anywhere using a PSTN Phone/ GSM Cell Phone) and access the IP Trunks connected to it, to make long distance calls at reduced rates.

4. Easier Management: The Analog/ Digital PBX is difficult to manage. Some of them can only be managed using complex CLI Commands that are proprietary to each vendor. But an IP PBX generally has a web based GUI (Graphical User Interface) console to manage/ configure/ make changes to many of its functions. This makes it easier for administrators to manage an IP PBX. Users may even be given a custom web-page which they could use to login and set their own preferences.

5. Soft-Switch: Some IP PBX models come as down-loadable software that can run on standard computer servers. These are called Soft-Switches. Soft Switch based IP PBX have a lot of advantages. There are some open-source based Soft-switches that can be downloaded free of cost (Like Asterisk, Trixbox, FreePBX, etc).

6. Cell Phone/ Land Line Integration: You can download a SIP Client on your Cell Phone (that supports this feature) to receive land line calls on your cell phone itself, through a Wi-Fi Network. So, your cell phone can become your mobile land-line extension!

7. Fixed Mobile Convergence: As an enhancement to the above mentioned point, it is possible to automatically shift between Wi-Fi Networks and Cellular Networks by using a technology called Fixed Mobile Convergence. So, when you are attending a land line call on your cell phone using the Wi-Fi network and you suddenly move out of the office, the call can continue on a cellular network!

8. Wi-Fi Phones: Wireless Networks (Wi-Fi) are very popular, and can be found almost in every company. It is possible to use special Wi-Fi based Phones to attend to land line calls from where ever you are, within the company premises (Wi-Fi Zone). The DECT standard followed by Digital PBX also allows to do something similar, but a separate and a dedicated digital wireless network is required to enable the same. There are a lot of advantages of VoWLAN (Voice Over Wireless LAN) technology when compared to DECT (Digital Enhanced Cordless Technology).

9. IP Phones/ Soft Phones: IP Phones may be costlier than analog phones, but IP Phones have a lot of advantages over analog phones like easy movement from one place to another (while still retaining the extension number), Connecting to Internet using inbuilt browsers, Down-load ring-tones, etc.  IP PBX support Soft-Phones that are software programs which can be run on a computer and they have a number of advantages over IP Hard-phones. These soft-phones can be used along with headset/mic to receive all your land-line calls directly from your desktop PC.

10. Encryption: IP PBX / IP Phones are capable of encrypting conversations by using techniques like sRTP (Secure Real Time Protocol). Though this technique is not used often, it can be used to deter hackers from listening to voice calls over the IP Network.

11. IP Faxing: IP PBX support IP Faxing which, among other things, can be used to receive and send faxes directly from a computer.

12. Analog Trunks/ Analog Phones: Analog/ Digital Trunks like ISDN / PRI Lines / FXO Lines etc from the Telephony Service provider can be directly terminated on an IP PBX. Similarly, Analog Fax Machines and Analog Phones can also be connected to an IP PBX. Both use devices called Analog Telephony Adapters (ATA). The main advantage with ATA is the fact that it can be present anywhere on the network. For example, if an Optical Fiber Cable connects to an individual department, the ATA can be placed in that department to connect the FXO/FXS trunk and subscriber terminals directly, while communicating back to the IP PBX over the IP Network (Using the Optical Fiber Cable).

13. Video Calls: With an IP PBX, it is possible to make video calls along with audio calls over the IP network. There are Video Phones that can be used for this purpose, and some IP PBX vendors support this functionality.

14. Unified Communications: Unified Communications (Or UC) is an emerging field in IP Telephony that integrates multiple ways of communications like voice calls, video calls, voice mail, email, fax messaging, Instant Messaging, Cell Phones, etc and allows the user to use one mode of communication to communicate with other modes, seamlessly.

15. Call Recording: Often, you may want to record certain voice calls for future reference. In analog/digital PBX, a separate line needs to be connected from each phone (in parallel) to an expensive equipment called Voice Logger. But with certain IP PBX models, call recording is an inbuilt function and can be activated by the user through their IP phone/ PC interface whenever required.

16. Presence/ Instant Messaging (IM): Some IP PBX models come with built-in Instant messaging function that can be used along with a PC based interface. This is similar to the web based IM that we are familiar with. The instant messaging function includes presence information which enables the caller to see if the user is available and their preferred mode of communication at that point of time.

17. Speech Recognition: Speech Recognition, is another emerging technology that has successfully been integrated with IP PBX which recognizes voice input (speech) directly from a caller to perform a desired action. For example, a caller could call the board number of a company and say the name of the person whom he wants to talk to, for the IP PBX to fetch the extension number from a database and forward the call to that extension automatically.

18. Meet Me Conference: Only simple built-in conference facilities are available with Analog/ Digital EPABX. Even that is restricted to 3-8 party conference calls at a given point of time. But IP PBX can enable a Meet-Me Conference (sometimes included as a default functionality) which allows many users to dial into a conference room to enable multi-conference calls. There can be multiple such conference rooms and multiple callers in each conference room. The users could even be given a security pass code that authorizes them to enter the conference call.

19. QoS & Voice Compression: Since IP Telephony shares the same network as computers and other IT equipments, the available bandwidth often needs to be shared between multiple type of devices. But fortunately, up to 10 Gigabit Ethernet bandwidth is available for the IP Network backbone today, which is sufficient for most enterprise applications. Further, Network Switches support QoS policies to be applied to IP Phones and voice related applications so that real time latency sensitive voice traffic can be given priority over data traffic. Various Voice Compression CODECS are available to compress voice signals over the IP Network.

20. Remote Maintenance: Both IP PBX and IP Phones can be accessed remotely (with sufficient authorization, perhaps over VPN Networks) for making configuration changes and monitoring purposes from anywhere over the Internet. The analog PBX might support remote access in a limited way, but analog/digital phones cannot be accessed from a remote location.

21. Hosted IP PBX: With an IP PBX, its possible to host the IP PBX Software (with a service provider) and register your IP Phones to the hosted IP PBX in their premises over the Internet. The service providers take care of the hardware/ software and maintenance of the hosted IP PBX, while charging a monthly fee for the Service.

22. Voice Mail/ IVR: Though Voice Mail functionality is available in an analog/ digital PBX as well, there is a limitation on the number of hours of voice mail that can be recorded. IP PBX use Servers/ Computer based disks to store voice mail and hence have more voice mail storage capacity. An IP PBX can even send a notification email to the user when a new voice mail is recorded. Advanced Interactive Voice Response (IVR) creation / customization can be done relatively easier and the tree structure changed frequently using IP PBX.

23. Help-desk/ Call Center functionalities: Some basic call center/ help desk functionalities can be built in to IP PBX like Call Queuing, Group Ringing, Automatic Call Distribution, etc.

24. Database Integration: A Number of interesting applications can be enabled by integrating databases (like MySQL, etc) with IP PBX. An IP PBX can be programmed to fetch certain database entries, when requested by the users, by pressing certain key combinations guided by an IVR – Interactive Voice Response. Mobile Banking is a good example of an application enabled by IP PBX through Database Integration.

25. Application Programming Interface (API): External programs and applications can interface with IP PBX using the Application Programming Interface that is provided by many IP PBX vendors. For example, when a customer is ringing the help-desk of a certain company, the IP PBX recognizes that it is a regular customer (via the phone number), fetches recent orders and their current status and displays them on the screen of the help-desk employee. So, even before the employee picks up the phone, he/she might be acquainted with the required information to answer the call! This is enabled by integrating the Customer Relationship Management Application with an IP PBX using API.

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The above diagram shows a basic IP PBX set up. LAN, is represented by a network switch (in the center). The IP PBX (or IP SoftSwitch) connects to the LAN (from the left) & also connects to both analog trunk lines (FXO /PRI Trunks) and IP Trunks (ITSP) through which the users can call the outside world. There are also some IP Phones (shown on the bottom) that connect to the LAN (through network switches) and gets registered to the IP PBX as extensions.

There is also a Wireless (Wi-Fi) Access Point that connects to the network switch (LAN) for enabling wireless (wi-fi) network access. And finally, there is a cell phone (smart phone) that connects to the Wi-Fi network (and hence LAN) through a built-in Wi-Fi adapter.

If this cell phone can be loaded with a SIP Client, it can also get registered to the IP PBX as an extension. SIP Client, is a software that can be downloaded from the Internet for most of the Wi-Fi based Cell Phones (If it is supported by the cell phone/PBX manufacturer). Sometimes, the IP PBX manufacturers make the SIP Client available to some leading brands/models of cell phones by themselves – Especially, if an IP PBX uses a proprietary protocol (Other than SIP).

For an Open Source based IP PBX like Asterisk, there are quite a number of SIP clients available for various smart phone models.

So, when the SIP Client is installed on to a Smart Phone & connected to the corporate Wireless Network, it could be registered with the IP PBX as an extension. So, from the normal land line IP Phone, you could communicate with the smart phone over the Wi-Fi network by just dialing the extension number. Similarly, you could call out (using the trunk lines attached to the IP PBX) by dialing the phone number you want to reach – The IP PBX will select whether to route the call over an analog/ digital / IP trunk. You can also receive land line calls on your cell phone.

It is to be noted here that the land line calls are received on the cell phone using the Wi-Fi network and not the cellular network. So, you should decide whether you want to dial out any call using the Wi-Fi network, or the cellular network when you are in the office. Of course, you can receive calls that come from both land line (IP PBX through Wi-Fi network) or the Cellular network.

The main advantage of having a set up like this is the fact that you could receive your land line calls where ever you are in the office on your cell phone (Provided that the Wi-Fi network coverage is present in that area). There might not be a requirement for a Wired IP Phone on your desk, as you are receiving all your land line calls on the cell phone. So, at least for some users, there might not be a need to buy separate IP Phones. But do note that the SIP clients may be charged by some vendors. For open source IP PBX, its mostly free provided the phone supports it.

Its even possible to roam across the office while you are attending that land line call with your cell phone (This requires the Wireless Network to support roaming. So, a wireless controller needs to be used to enable roaming on all the wireless access points).

You can also use the 3G/4G network to connect the cell phone to Internet and use the SIP client to make outgoing calls through an ITSP – Internet Telephony Service Provider. But, in this case, the calls go through the Internet and not your company IP PBX. So, unless you have a direct corporate account with an ITSP, the calls will be charged to you.

The next step to making and receiving land line calls on your cell phones, is Fixed Mobile Convergence. This technology is basically about transferring a call between your Wi-Fi network and cellular network (and vice versa), with out disconnecting it! So, when you are moving out of the office, the call gets automatically transferred to your cellular network, without you having to disconnect the line!

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CENTREX:

CENTREX refers to Central Exchange. Companies need an Analog/Digital PBX or IP PBX to give land-line telephone connectivity to all their employees. Click here to know more about the various components of a PBX. Instead of companies buying, installing and managing a PBX in their premises, the Telephony Service Providers (Local Exchange – CO) offer the PBX services directly from their switching equipment. Which means, without buying a physical PBX, customers can still enjoy many PBX functions at their premises. These functions are generally provided through direct (multiple) lines from the Centrex Service Providers. Some Service Providers might place an CPE equipment in the customer premises, if the number of subscriptions are high. But the CPE equipment is owned and maintained by the Service Provider.

Advantages of CENTREX:

• No capital expenditure incurred for buying the PBX systems.
• No Power bills which might have been incurred if PBX systems were bought.
• No Maintenance hassles / No need for technically trained manpower/ No AMC charges for PBX.
• Intercom dialing (between the various users in the same company) is free of cost.
• Sometimes, the free Intercom dialing is extended to various locations around the city/ country.
• Normal analog phones can be used with Centrex (mostly).
• There is no risk of obsolescence of PBX equipment/ features as new features are added regularly in the Switching equipment present in Exchange (CO premises).
• The maximum number of extensions that can be accommodated by a Centrex can be huge – So, the number of subscribers can be increased without having to replace/upgrade PBX hardware.
• Pay-As-You-Use model which enables companies to avail many of the PBX features/facilities with minimum up-front costs.

Limitations of CENTREX:

• Adds/ Moves and Changes would generally incur separate costs, each time they are done.
• There is a fixed contract period within which the customer cannot terminate the CENTREX facility.
• Charges are Per-Month Per Line (Extension).
• Some features like Voice Mail/ Message Waiting might incur additional cost, sometimes per user.
• Not all PBX features are supported by CENTREX.
• It takes time to make changes in the existing set up as help desk ticket needs to be raised with the Service Providers.
• Cables / MDF and other accessories required in the customer premises fall under customer scope.
• Analog Trunks/ Digital Trunks can be taken only from the same CENTREX service provider.

Hosted IP PBX:

IP PBX uses the Internet Protocol/ Infrastructure to carry and switch calls. Click here to learn more about IP PBX. Hosted IP PBX is like the CENTREX system mentioned above except that the PBX is IP based (IP PBX) which is hosted in a server (mostly) at the Service Provider premises. At the customer end, there are IP Phones that connect to the computer network directly using RJ-45 ports instead of the analog phones that connect to the telephony network using RJ-11 ports.

Advantages of Hosted IP PBX:

• Hosted IP PBX uses IP Phones (at the customer end) mostly. These IP Phones directly connect to the computer network and hence there is only one network to set up and maintain thereby reducing the costs involved in setting up/ maintenance of a separate voice network.
• In addition to IP Phones, Soft-phones (software that can be loaded on to a computer and used with headset-mic) & Wi-Fi Phones (Mobile land-line extensions that use the company Wi-Fi network) can also be used with Hosted IP PBX.
• Advanced features like Single Number Reach, Unified Messaging, Click-to-Dial etc can be supported by a Hosted IP PBX Service Provider.
• SIP Trunks can be used (either from the same service provider or external SIP Trunk service providers) to make outbound calls over the Internet and hence saving costs (especially for long distance calls). SIP Trunks can be taken from multiple service providers.
• Analog Phones/ Analog Trunks/ Analog Fax machines can be connected to the IP Network through ATA’s and hence may be allowed to be registered with some Hosted IP PBX service providers.
• Hosted IP PBX is a scalable solution, but license costs might still apply for every extension.
• There may not be much upfront costs, but there are monthly charges for using the Hosted IP PBX service. There may be a minimum lock-in period.
• The IP Phones (SIP Phones) can be anywhere across the world and still be registered on the same Hosted IP PBX over the Internet. So, the inter-branch calls might not incur additional costs.
• Adds/ Moves and Changes (for the IP PBX Settings) are easier and can be done (for limited features) remotely using an Internet Browser. Even the users can configure/ manage their own call/ voice mail settings similarly.
• Only the control signals go to the Hosted IP PBX, while the RTP (payload) flows directly between the two IP Phones.

Limitations of Hosted IP PBX:

• IP Phones are much more expensive than the analog phones. Moreover, they occupy one network switch port each.
• Analog phones can be connected to the Hosted IP PBX through ATA’s but each port of ATA is more expensive than the analog phone.
• Each call takes up a certain bandwidth (like 64 Kbps – at least). So, both the LAN network and WAN network needs to be planned to accommodate this additional bandwidth.
• QoS Policies need to be set up end to end on the network and all the devices ought to support such QoS policies in order to get optimum performance for voice over the data network. If there is excessive load on the data network, there may be some delays for the voice traffic.
• If the computer network fails, the IP Phones would also not work.
• For large installations, it may be better to use VOIP Codecs which might incur additional cost.

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• Cost: The higher cost of IP Phones have been the most important factor prohibiting their mass adoption. Analog/ Digital phones, in comparison are available in all ranges (including highly cost effective models). But there are reasons (supported functionalities) for that added cost.
• Diversity: There are different types of IP Phones. There are Soft-phones that can run on your computer, there are Voice over Wi-Fi phones that can operate using the wireless network, Soft-clients on mobile phones, and of course there are hard IP Phones on the desk. But analog phones aren’t available in so many forms.
• Single Network/ Cable: With IP Phones there is a single network to build and maintain – The IP network. But with analog-digital phones, you need a separate network with telephone cables in addition to the IP network for data. A single cable from the network switch could connect to the IP phone’s two port switch and the computer could be connected using another network cable from there, but many companies have two cables coming from the network switch for redundancy.
• Power: The good old analog phones do not require much power – what ever they need they can take it from the line itself! But the IP Phones need to be provided with a separate AC Power (or) Expensive Power Over Ethernet injectors/ switches that can send power along with data using the network cable.
• Mobility: IP Phones can just be moved from one desk to another, and they will still pick up the config. information and retain your extension there. The Voice Over Wi-Fi Phones can work using the wi-fi network and hence can be carried around the building – making it a mobile land-line! Even the Digital Phones could do this using DECT technology, but it is proprietary, and one cannot access the data network using DECT base stations.
• Usage over the WAN Network: You can use an IP Phones to register with an ITSP service provider to make long distance calls over the Internet economically. There is also no additional call charges when you use the IP Network (Connected with Leased Lines/ Internet) for calling between the branches, provided your data plan has unlimited bandwidth.
• Shared Bandwidth: Since IP Phones share the network with computer data networks, the bandwidth is shared between computers and IP Phones. Its possible that over usage of either can affect the performance of the other, especially if end-to-end QoS policies are not enforced on the network. But analog/digital phones are on a separate network and even if the computer network goes down, the telephone network is  still On. In large scale deployments, it may be prudent to deploy some voice compression protocols.
• Open Standards/ Multi-vendor connectivity: You could buy the analog phones from any vendor (not digital phones though) and the EPABX from any vendor and be assured that all the features/functionalities of the PBX would work on the phones. But with IP Phones, SIP is the open standard protocol, and only the PBX and IP Phones that support SIP would work together, and that too with slightly limited functionality. Some vendor’s IP Phones are proprietary and work only with their version of the IP PBX.
• Security: While the only issue with analog phones is the possibility of ‘phone-tapping’ of external calls, IP Phones are vulnerable to some network attacks that can be directed on the VOIP system.
• Remote Maintenance: IP Phones can be accessed and its configuration could be checked/changed from a remote network (or over the Internet), if appropriate permissions are given by the network administrator. This makes making configuration changes / maintenance easier.
• Multiple Lines: While analog/digital phones could have one extension number (max), IP Phones could have multiple extensions (one for desktop & one for ITSP account over the Internet, for example).
• Video Calling: IP Phones are moving in a big way to integrate video along with telephony. Users could use their web-cam to see the other person over the computer monitor and talk to them over the IP Phone, simultaneously; Or use special purpose video phones to see and talk over the large screen available over the IP Video Phones itself. This is available (in a limited way) with ISDN phones as well, but they are not very popular and need the digital ISDN network on both calling/receiving end.
• Connecting to the Internet: With certain higher end IP phones, you could directly connect to the Internet right from your IP Phone with the built in browser. It may be possible to download and view pictures / download and listen to songs and ring-tones right from your IP Phone!
• Computer Telephony Integration: This feature is available with Digital Phones (Making a call by clicking the contacts in your email client, etc), but IP Phones take the integration many steps forward – You can have a softphone right on your computer, you can click on a phone number embedded on a webpage to make an outgoing call through your corporate PBX, you can receive email notifications of your voice mails, you can even know who is calling/ what was their latest transaction details before picking up the call by integrating your IP PBX with business processes like the CRM systems, you can use API’s to integrate Google Maps to pop up the location of a caller – for example, and much more!
  

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Related Article: Voice over IP & IP telephony – An Introduction

As you can see in the above diagram representing IP Softphones Vs IP Hardphones, the connectivity architecture is very similar for both. Both are SIP extensions that can be registered in the (SIP based) VOIP / IP telephony server and can be used to make/ receive outbound calls both over the Internet (SIP Trunks, etc) & over the PSTN Networks (Analog trunks, E1/T1 trunks, etc).

IP Softphones are software programs that can be installed on a computer/ laptop. They are assigned extension numbers (like 301, 302, 303….) just like how IP Hardphones are. So, calls can be transferred to these Softphones using their extension numbers, and if DID is available, outside callers can directly call the Softphone extension numbers. Softphones have software based visual dial-pads on the monitor which can be used (by clicking on them using a mouse) to dial out/ receive calls, and for normal telephony functions like caller ID display, call hold, message waiting, call history, redial, etc.

So far, so good. But, since they need a computer to work on, they can work only when the computer is switched on. But since people receive  land line calls only when they are around the computer, this might be fine. Moreover as most of the calls are now received on cell phones, the investment on IP hardphones can be reduced by using the IP softphones.

IP Softphones may require a user to use headset with mic/ use a speaker to attend to a call, and the users might feel uncomfortable putting on a headset/ speaker every time a call comes in, or they need to dial out. The built-in (or external) speaker/ mic in the laptops can also be used for hands free operation, though the voice quality may not be that good and in this case, maintaining privacy in a shared room/ hall becomes difficult. There are USB based headset only devices that plug-in to the USB port of the computer and work along with Softphones to make and receive calls, but these are not manufactured by standard VOIP/IP Telephony vendors.

Two of the major advantages of IP Softphones (even some free ones) are – Instant Messaging & Video Calling. But only the users who use similar IP softphones can use these features. Some vendors even show presence information (availability information) of the contacts on IM that is built into the IP softphones & some vendors support H.264 Video Compression standards for video calling to get best video quality, with least bandwidth.

Of course, IP Softphones can call IP hardphones that are directly or indirectly (through a service provider) connected to an VOIP/ IP Telephony Server as Softphones are like any other extension of the IP Telephony server. By using IP Softphones, companies can not only save on the IP Hardphone costs, but they also save on cabling/ switchport/ electrical power costs.

Depending on the vendor, some of them support a lot of additional features required in an enterprise like 802.1p QoS packet tagging (so as to prioritize voice traffic on the network), recording the voice conversations, etc.

Since IP Softphones are software programs, they are Operating System dependent. They pop up on the screen when an incoming call is coming, so that the user could be informed on the same and pick up the call. Since most of the users these days work on their desktop/ laptop computers  for doing almost all of their work, IP Softphones can be considered as a replacement to IP hardphones for many. These Softphones are even available for many mobile devices and can be used with a Wi-Fi network to attend to/ make landline calls anywhere within the company Wi-Fi network.

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]]> http://www.excitingip.com/1490/voip-softphones-vs-ip-hardphones-ip-telephony/feed/ 0 http://www.excitingip.com/1088/licenced-open-source-voice-codecs-used-in-ip-telephony-voip/ http://www.excitingip.com/1088/licenced-open-source-voice-codecs-used-in-ip-telephony-voip/#comments Tue, 18 Jan 2011 17:24:18 +0000 admin http://www.excitingip.com/?p=1088 In this article, we will see why its important to select the right voice codec for IP Telephony (VOIP) implementations and a short introduction to some of the popular voice codec’s – G.711, G.729, G,723.1, iLBC, Speex & GSM codec. These are narrow band codec’s.

What is a Voice Codec?

A codec digitally encodes and compresses analog audio signals using complex mathematical models. A codec’s primary role is to seek a balance between the transmission efficiency (bandwidth) and the quality of voice signals. That is to say, transmitting the best quality of digital voice signals at the lowest bandwidth possible.

A codec refers to both coder/ decoder & compression/ decompression.

Why is a Voice Codec required?

A voice codec is required mainly for compressing the digital voice signals so that they can be transmitted across IP networks (including lossy networks like the Internet) with least possible bandwidth and affordable quality. When codec’s are used in IP Telephony, they generally introduce processing delays as complex mathematical formulas are used for encoding/compression are CPU intensive.

Codec’s are especially required in large IP Telephony/ multi-location VOIP installations as each call on the IP network consumes a certain amount of bandwidth and the total bandwidth consumed for uncompressed voice packets could be enormous.

If you are going with a default IP Telephony installation, there is a good chance that you might be dealing with G.711 Voice Codec for all your calls. But its important to note that, on larger/ multi-site VOIP installations it might be better to go with one of the licensed codec’s or even the open-source codec’s that are mentioned below (even if only partially). These codec’s make a considerable difference to the overall voice quality and bandwidth consumption for IP Telephony implementations.

Some techniques used to compress the digital voice data:

• VAD – Voice Activity Detection: In IP Telephony, both the conversations as well as the silence in between the conversations are digitized. So, we have both – packets containing voice as well as packets containing silence. Using VAD, packets of silence can be discarded after their duration is appropriately marked. So, the total number of packets transmitted after compression is lesser (generally around 30% lesser).
• CNG – Comfort Noise Generation: This is not a compression technique, but when voice is compressed using VAD, the awkward silence between the speech might be interpreted as lost connection and hence white noise is generated locally at both ends using CNG. This makes the call appear connected to both the parties during silence, as some background noise is audible during that duration.
• CELP – Code Excited Linear Prediction: In this method, various human sounds are mathematically modeled and a code book of all possible sounds is produced. So, instead of sending the actual sound packets across, only their codes are sent across. This is a very simplistic explanation, and a lot more techniques are involved.
• In some compression techniques, the headers can be compressed separately like the payload compression which provides additional bandwidth efficiency while transmission.

These are some methods by which voice is compressed (to be decompressed at the other end), and they are by no means comprehensive. But this is given here to get a hang of how voice compression is done in digital networks.

G.711:

G.711 is considered to be the base codec in IP Telephony. It gives the highest quality of voice but takes up largest possible bandwidth. A small compression is provided by a technique called companding, and for this codec CPU utilization is least. This technique is recommended for smaller IP Telephony implementations and in places with large network backbone bandwidths where high quality voice is required. This could also be used in multi-vendor IP Telephony projects.

G.711 uses a technique called Pulse Code Modulation (PCM) to give a bit-rate of around 64 Kbps theoretically. Practically though, after the addition of all overheads, it might consume around 87.2 Kbps per channel. G.711 is susceptible to packet loss when compared with other codec’s. Further, the end to end delay increases with increasing number of concurrent calls as processing delay (due to larger packet sizes) increases with each additional concurrent call that has to be processed by the VOIP Server.

There are two types of G.711 compression – Mu(U) Law, popular in North America and Japan, and A Law popular elsewhere. There are additional types and annexe to this standard that gives more flexibility and variation.

G.729:

G.729 uses a technique called as CS-ACELP (Conjugate Structure Algebraic Code Excited Linear Prediction). This technique of compression, as you might guess, uses complex mathematical formulas and hence takes more CPU processing resources. So, it has the highest value of delay in packetization. G.729 is recommended for many users and heavy data but the quality of voice may not be as good as G.711 compression. The theoretical bit-rate achievable by G.729 is 8 Kbps, and bandwidth of 31.2 Kbps with all overheads. As you can see, that is much lesser than G.711.

G.729 is a licensed codec and is not free. The codec needs to be purchased either by the end user or manufacturer based on the total no. of concurrent calls expected in the network (for example). G.729 has annexes like G.729A, G.729D & G.729E which support 8 Kbps, 6.4 Kbps, 11.8 Kbps bit-rates respectively. G.729D & E provide variable bit-rate (support for multiple bit-rates).

G.723.1:

G.723.1 is a dual rate speech codec supporting bit rates of 6.3 Kbps and 5.3 Kbps with a practical bandwidth consumption of  around 20-22 Kbps per channel. Though it is recommended that both sides transmit at same rate, the codec will still work if one side transmits at 5.3 Kbps and the other side at 6.3 Kbps. G.723.1 has the lowest transmission (propagation) delay among the three of them and hence is suitable for very large installations across unreliable networks with low bandwidth.

G.723.1 provides good immunity against network imperfections like packet loss, lost frames & bit errors. This codec is quite popular with audio/ video conferencing applications as well. This is not a free codec and is licensed. Either the end users or the manufacturers would have to purchase them based on the maximum number of concurrent calls expected in the network.

Open Source and Free Codec’s:

GSM:

GSM is the same as Global System for Mobile Communications that are used by cell phone providers, who are also shifting to IP based networks,  by the way. The bit rate is 13 Kbps and the speech signals are divided in to blocks of 20 ms. This is a free and open source codec with a good compression rate but average voice quality. Since it is already widely used in cell phone communications, large enterprise networks could use them if they are expecting a huge number of concurrent calls within their VOIP networks.

iLBC:

iLBC refers to Internet Low Bit-rate Codec. iLBC results in a payload bit-rate of 13.3 Kbps & 15.2 Kbps with encoding frame lengths of 30 ms & 20 ms respectively. It provides a mix of low bandwidth usage and decent quality especially in lossy networks like the Internet. However, its compatibility with common VOIP systems needs to be checked before implementation and this codec is quite CPU intensive. iLBC is license free and can be used without paying royalty fees. It is also open source. Anyways, its better to check the special terms and conditions given in their site before using this one in your network.

Speex:

Speex is a variable bit-rate codec which can operate with bit-rates from 2.15 to 22.14 Kbps. It can dynamically modify its bit-rate in that range according to the changing network conditions. There are both narrowband and wideband versions of this codec, available free of cost. It is also an open source based codec. Speex uses CELP as its encoding technique and is designed for VOIP and packet based networks. Speex supports a number of features like Intensity Stereo Encoding, integration of multiple sample rates in the same bit stream, Variable Bit rate Operation, etc in addition to the other usual compression features. Be sure to check if you can implement this codec, before purchasing the licensed codecs. Open Source PBX systems like Asterisk offers native support for this codec.

Limitations of Voice Compression techniques:

As mentioned earlier, almost all the compression techniques achieve lower bandwidth transmission by sacrificing some amount of voice quality. This is inevitable. Also, some compression methods may not support DTMF (Dual Tone Multi Frequency) touch functions, Fax Transmissions, High quality audio and MOH – Music On Hold functions. So, it is better to check for these before selecting one.

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]]> http://www.excitingip.com/1088/licenced-open-source-voice-codecs-used-in-ip-telephony-voip/feed/ 0 http://www.excitingip.com/536/can-open-source-pbx-be-scaled-to-carrier-grade/ http://www.excitingip.com/536/can-open-source-pbx-be-scaled-to-carrier-grade/#comments Tue, 01 Sep 2009 18:24:40 +0000 admin http://www.excitingip.com/536/can-open-source-pbx-be-scaled-to-carrier-grade/ As we know, open source PBX (Soft Switch – like Asterisk, for example) has already been accepted as a stable and preferred voice switch for enterprises and business establishments. In this article, we will see if the open source PBX meant for enterprises can be scaled to carrier grade – to handle protocols like SS7 after analysing the current limitations of current open source PBX which are meant for enterprise applications. We will also have a look at Woomera protocol which enables such a scaling through a distributed voice switch model.

Audio Codec:

Every system implementing VOIP/IP Telephony uses an audio codec to compress the audio signals at one end and de-compress the same at the other end. Although most of them are standardised, VOIP vendors implement proprietary codec’s too. Some examples of popular standardized Codec’s include G.723, G.729a etc.

The type of Codec used is an important factor that affect the VOIP call quality as higher the compression, lesser the size of data to be transmitted over the other side. But there is a flip side too – the voice quality generally suffers with higher compression rates. Most Codec’s can accommodate different target compression rates like 8 Kbps, 6.4 Kbps, 5.3 Kbps etc (Standard 64 Kbps required to transmit voice over T1 lines – Single channel, PCM). The bit rates mentioned are for audio only, and protocol overheads is added over that and hence the actual bit rate realized is quite higher.

The Codec’s also introduce a digitizing delay as each algorithm requires a certain amount of data to be buffered before it is processed. If the Codec is very complex to be implemented, more CPU resources would be required and hence this too affects the VOIP call quality.

Latency:

Network latency is caused both due to the distance that the packet needs to travel and also due to the changing network conditions. More the distance needed for a packet to traverse (Eg. across continents), higher the delay. The delay also depends on the number of router hops that a packet needs to take to reach the destination. Higher the number of hops, more the delay.

The Compression algorithms also cause their own delays. For example, G.723 Codec generally adds a fixed 30 ms delay. The total network latency (two way round trip delay), for the VOIP call to be clear is around 150 ms to 500 ms. Although more than 250-300 ms delay is not preferred for most VOIP systems.

Jitter and Jitter buffer:

When the packets are sent from the Codec after compression, they are sent at a constant rate with equal spacing between them. But when they are received at the other end, the decompression algorithm also expects the packets to arrive with equal spacing between them and in the same order as they were sent. But since network imposes delays at packet level, the packets may arrive at different time intervals and they may not arrive in the same order, as they were sent. To compensate for this, there is a small Jitter buffer at the receiving end, which induces a certain calculated delay before sending the packets for decompressing. The Jitter buffer induces a small delay to collect a certain number of packets for rearranging them in the proper order as well as inducing equal spacing between them before sending them for decompression.

Packet Loss:

Some of the packets are always lost in an IP Network. It may be due to a lot of reasons like excessive collisions, physical media errors, overloaded links etc. Some protocols like TCP account for such packet losses and allow for recovery of lost packets, while some other protocols like UDP doesn’t allow recovery of lost packets.

The Codec’s perform certain operations to compensate for the lost packets (like using the previous packet instead of the lost packet or perform more sophisticated interpolations to approximate for these losses, etc). Generally packet losses up to 5% are compensated for, and the user may not experience a sufficient degradation in voice quality. But a packet loss of more than 5% might lower the quality of voice or induce noticeable delays.

Packet Size:

The packet size poses an interesting situation. If the packet size (RTP) is higher, the overall bandwidth is reduced as more information can be packed in to a single packet and there is a substantial amount of overhead control packets (header information) that needs to be added to every packet that goes out. This overhead control information is almost thrice the size of the original payload packet (RTP) itself! So, it is better that the packet size is bigger, but if the packet size is too big then there is a packetization delay which is induced as the sender needs to wait for some time for filling up the payload.

It is better to send bigger sized packets anyway as the overall bandwidth required is reduced. But that is generally done by increasing the inter-arrival timing so it is better to check if the delay budget allows for it. In certain Point to Point links, cRTP (Compressed RTP) is preferred as it compresses the header information required to send the control signals across. cRTP almost brings down the size of each packet by almost half, but it generates additional processing overload for the routers and used only for certain types of point to point WAN links as it does not contain the IP address information in the packet and hence not rout-able.

Silence suppression:

Since only one person talks in a two-way communication at any given point of time, it is better not to transfer any packets for the other person who is silently listening. Several vendors take advantage of this attribute to reduce the overall bandwidth required for the transportation of the voice packets across WAN links.

Echo:

IP Telephony invariably involves the conversion of IP media to analog/digital and vice versa. There is an echo induced due to this conversion at various points in the network. There are two types of echo. Hybrid echo is generated due to the impedance mismatches at the various analog/digital points in the network. Acoustic echo is generated at the phone. It happens as the voice leaving the speaker is picked up by the microphone. It is generally difficult to monitor and contain echo, but certain vendors provide echo cancellation (hardware and software) modules at the gateway level where the translation takes place, to contain echo.

Most of the above parameters can be monitored by specialized tools and adjustments can be made accordingly

Network Parameters:

The overall network load is one important parameter that determines the quality of voice communications. More the network load, more collisions and lesser quality of transmission. Though this aspect may not always be under the control of the network administrators, following things could be done to increase the efficiency of transportation of the voice packets.

It is always recommended to set a higher priority to the voice packets traversing through the network, than the data packets (like mail traffic, etc). This is because voice/video packets are delay sensitive and even a slight delay might cause a degradation in quality. But even if the data packets like mail (SMTP) are delayed, it doesn’t make a noticeable difference to the user. The prioritization of real time packets needs to be done at every stage of transporting them (like switches, routers, WAN links etc).

Another alternative is to use bandwidth reservation or bandwidth limiting techniques in the network based on the application/protocol. This would ensure that some bandwidth is always reserved for voice packets and the sudden sprout in the usage of certain applications (like P2P) does not interfere with the sending of voice packets over the IP network.

Other parameters like Call set-up times (time taken for initial dialing of digits to establishing a voice connection), Call success ratio (ratio of successful connects to dial attempts) and Call set-up rate (the number of calls that can be set up per second, in the network) are also important factors that affect the VOIP call quality. Other factors like the type of protocol used – like SIP or H.323 may also affect the performance as various processes are handled differently by each of them.

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