Tag Archives: VoIP

VoIP Billing System

1 Aug

With the advancement of technology in telecommunication and improvement in the services of multimedia sessions, voice traffic has increased in huge amount, which presents a challenging situation for the service providers to meet the expected demands of customers and to optimize the network available.

voip billing systemvoip billing

The growing names in the industry has come up with feasible solutions for the arising issue, one such name is AdoreInfotech, which provides VoIP solutions to maximize the resources available by the latest technologies. VoIP is abbreviation of Voice –Over-Internet-Protocol, which is a technologically advanced method to be used through internet for delivering voice communication and multimedia sessions.


Adoreinfotech has developed with the VoIP billing solutions that has been designed keeping in mind the demands and expectation in a telecom sector. VoIP Billing software can be implemented as a Customer Management system to collects, rates, taxes and bills voice and related services. We cater all issues related to VoIP; we have experts in our team to handle all the software related issues and gives the best of VoIP solutions. VoIP software is a need of all telecom industry whether large or small. VoIP is also available on many smartphones, personal computers and on internet enabled devices and helps calls and SMS with the availability of Wi-Fi or 3G connections.


The VoIP billing Software of Adoreinfotech has come with a competitive edge in the market and has been enhanced with the blend and integration of various functionality. With a keen focus on the development of VoIP and the related products, company has specialized with a business strategy in the development of related VoIP products as per the customized demand of the market to make modular VoIP billing software.

Adoreinfotech has set standards to deal in VoIP billing software with a strategy to deliver the best, our focus lies on the specifications of our customers, with this in mind , Adoreinfotech has devised special designed software with competitive rates for the users.

Source: http://softphonevoip.wordpress.com/2014/07/30/voip-billing-system/


Telephony – Telco service or Internet application?

9 Jun


When comparing different forms of VoIP, one risk comparing “apples and oranges”. Broadly speaking, we can divide VoIP into two main categories. First, the service can be implemented as a faithful copy of circuit switched telephony; in a network with full control over performance and quality. Second, VoIP can be implemented as a standalone application used over the open Internet.

Originally published in NetworkWorld Norway.


3GPP and IMS

3GPP (3rd Generation Partnership Project) has played an important role when VoIP has become a recognised substitute for traditional telephony among telecom operators. 3GPP standardises the mobile technologies 2G, 3G and 4G, and they have done so based on the general IP technology standardised by IETF (Internet Engineering Task Force).

At first 3GPP concentrated on developing mobile networks as an evolving telecommunications architecture, following a vertically integrated model for provision of telephony. As the Internet revolution influenced the telecom market, the focus has shifted more towards IP-based services of various kinds.

IP networks and the Internet are not equivalent concepts. As IP technology was introduced in the mobile architecture, this was done in a way that maintained telecommunications networks’ support for QoS (Quality of Service). They had a clear view to continue provision of telecom services, as opposed to Internet applications, but based on a new IP-based network.

The service platform which was standardised as part of the mobile architecture was named IMS (IP Multimedia Subsystem). IMS is based on SIP (Session Initiation Protocol), the VoIP protocol from IETF, but extended with a comprehensive architecture for QoS. IMS has an “open” interface for service development, but requires a business agreement with the mobile operator. So this is a completely different kind of openness than the one found on the Internet where “everyone” can develop their own services.


The basic mobile architecture has undergone a tremendous development by 3GPP. Now we are in a phase where LTE (Long Term Evolution) is being adopted, often referred to as 4G despite the fact that it is not “real” 4G. LTE is the first 3GPP architecture that has eliminated the circuit switched domain, appearing as a pure IP network. Therefore there are great expectations for VoIP in this architecture, a functionality called VoLTE (Voice over LTE).

The transition from traditional telephony to VoIP has been going on for a long time. In mobile networks this has taken longer than expected. IMS has been around as a part of the mobile architecture for many years already. Furthermore, VoLTE includes options that could still delay this transition; LTE phones will initially combine LTE with older mobile technologies, allowing telephones to fall back to these older technologies. There is also a quasi-solution that transports traditional telecom protocols encapsulated in IP packets, so-called VoLGA (Voice over LTE via Generic Access).

The telecom industry also promotes advanced VoIP services that can stimulate the transition from traditional telephony and SMS to IP-based “equivalents” called RCS (Rich Communication Services). RCS provides services such as voice and video telephony, presence, instant messages and more, integrated in a unified user client for mobile phones that will provide seamless user experience of multimedia communication.

RCS is based on the IMS platform using SIP and SIMPLE (SIP for Instant Messaging and Presence Leveraging Extensions). Thus, the basis of this is IETF protocols, but implemented in an architecture that is intended to replicate the telecom network in the shape of an IP-based multimedia network. RCS is promoted by GSMA (GSM Association) and OMA (Open Mobile Alliance). OMA is the descendant of the WAP Forum, if there are still some who remember WAP.

QoS and Policy Control

RCS seems like an impressive technology, and what is the big deal? What distinguishes this from the applications that are already in use on the Internet? A major difference is that RCS can benefit directly from the mobile network built-in mechanisms for QoS. But it is difficult to predict what will give the best user experience, multimedia services integrated in the mobile architecture or free choice among different applications offered over the Internet.

A well-known characteristic of the Internet is that it is “best effort” and can’t guarantee the quality of the communication. In the mobile architecture, QoS is a key feature across the entire design. The underlying IP network will typically be based on DiffServ (Differentiated Services) and MPLS (Multiprotocol Label Switching), both well-known technologies from IETF supporting traffic management and QoS.

In the LTE architecture, QoS is policed by a function called PCC (Policy and Charging Control). As the name suggests, not unnaturally, management of QoS and charging are two sides of the same coin. PCC controls establishment of user sessions with various performance levels, and charging information is generated based to the capacity used by the different sessions.

Initially, IMS was specified for mobile networks, but in retrospect it has been found very useful extending the scope to include fixed networks, giving a combo solution which is often referred as NGN (Next Generation Networks). This facilitates convergence between fixed and mobile networks (Fixed-Mobile Convergence).

Over-the-top (OTT)

The traditional telcos are operating in a market that is completely changed because of the Internet. This leads to a situation where the business that telecom players envision, is facing strong competition from Internet players. The Internet model is based on decoupling of applications from the network layer, as opposed to the telecom model that relies on the services that are vertically integrated with the network.

Innovative solutions that can be used “over-the-top” without specific facilitation from telecom operators, enables virtually unlimited choices for end users. Internet applications, even real-time applications such as VoIP, work fairly well without the quality architecture of NGN. Congestion control mechanisms regulate traffic load of the Internet, sharing the available capacity between users.

However, users’ choice is not easy. Such innovative solutions in some cases evolve into isolated “islands” that are not compatible with each other. Major players are trying to create their own closed ecosystems consisting of operating systems or app stores for example. On the other hand, some traditional telecom operators introduce OTT solutions to meet the competition, making use of similar means.

The future will show which model is most adaptable. Net neutrality is tasked to ensure that the Internet model can develop freely. Meanwhile, the Norwegian guidelines for net neutrality are balanced, allowing the telecom model to evolve in parallel. This is often referred to as “specialised services”, as opposed to the Internet access service that works as a general electronic communication service.

Source: http://ipfrode.wordpress.com/2013/01/21/telephony-telco-service-or-internet-application/

SIP Trunk BandWidth Calculation

9 Jun


Bandwidth as per Wikipedia is

A measurement of bit-rate of available or consumed data 
communication resources expressed in bits per second or 
multiples of it (bit/s, kbit/s, Mbit/s, Gbit/s, etc.). 
According to Hartley's law, the digital data rate limit 
(or channel capacity) of a physical communication link is 
proportional to its bandwidth in hertz.

Calculating the bandwidth requirement for a TDM network is easy. That is because they are based on either Multiplexing techniques like TDM,  Where each (Digital Signal) DS0 would correspond to one call, and depending on whether you are using T1 or E1 the number of simultaneous calls would be either 24 or 32 respectively. And at the end it boiled down to the number of connected circuits we have between two sites. However the same isn’t applicable for a SIP network when it comes to identifying Bandwidth, as there are no TDM or FDM techniques employed to send data. We’r going to use a century old telecom calculation technique called erlang. An erlang calculator can be found at http://www.erlang.com/calculator/ Prior to determining the bandwidth requirement of a SIP network we’d need to

  1. Determine, maximum simultaneous calls we need to support at any given time.
  2. Busy Hour Traffic (BHT): BHT is the measure of the call traffic at the busiest operational hour. also known as erlang load the Calculation ofBHT = (Average Call Duration(s) * calls per hour )/ 3600
    For example if we have 4000 calls per hour, with an 
    average duration of 180 seconds then BHT would be 
    --> (360 x 180)/ 3600 = 200 Erlang
  3. Determining Blocking: Blocking is the measure of failure of call attempts due to insufficient available resources. (as per definition number of lines).
    For example, a Blocking of 0.05 indicates 5 calls 
    blocked per 100 calls attempted. These blocked 
    calls would hear a busy signal or re-order tone.

The resultant of feeding these numbers in the erlang calculator is the number of trunks required to support the number of calls we wanted at a certain desired Grade of Service. Now if we had been working with TDM, then our calculation would have been complete with this resultant number from the erlang calculator. But since we are dealing with IP telephony and SIP there are a few more steps to be taken. In the next steps we would be converting that number of trunks (which is also equal to the simultaneous calls had it been TDM) into bandwidth. Lets see how.. For doing that we need to identify what codecs we would use. Whether we would use g711, g729 etc. Each codec has its own set of characteristics, which could include the codec’s sampling size, payload type, tolerance etc.. A short comparative list of capabilities of various codecs Codec Bandwidth Sample period                     Frame size                   Frames/ packet                Ethernet Bandwidth G.711 (PCM)              64 kbps                         20 ms                                160 1 95.2 kbps G.723.1A (ACELP) 5.3 kbps                         30 ms                               20 1 26.1 kbps G.723.1A (MP-MLQ) 6.4 kbps                    30 ms                                24 1 27.2 kbps G.726 (ADPCM)      32 kbps                          20 ms                                80 1 63.2 kbps G.728 (LD-CELP)   16 kbps                           2.5 ms                               5 4 78.4 kbps G.729A (CS-CELP) 8 kbps                            10 ms                                10 2 39.2 kbps AMR (ACELP)          4.75 kbps                     20 ms                                12 1 36.0 kbps AMR (ACELP)          7.4 kbps                        20 ms                                19 1 38.8 kbps AMR (ACELP)          12.2 kbps                      20 ms                                31 1 43.6 kbps AMR-WB/G.722.2(ACELP)6.6 kbps       20 ms                                17 1 38.0 kbps Before moving ahead we would also need to know the following

Codec Bit Rate (Kbps) Based on the codec, this is the number of bits per second that need to be transmitted to deliver a voice call. (codec bit rate = codec sample size / codec sample interval).
Codec Sample Size (Bytes) Based on the codec, this is the number of bytes captured by the Digital Signal Processor (DSP) at each codec sample interval. For example, the G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval).
Codec Sample Interval (ms) sample interval at which the codec operates. For example, the G.729 coder operates on sample intervals of 10 ms, corresponding to 10 bytes (80 bits) per sample at a bit rate of 8 Kbps. (codec bit rate = codec sample size / codec sample interval).
MOS MOS is a system of grading the voice quality of telephone connections. With MOS, a wide range of listeners judge the quality of a voice sample on a scale of one (bad) to five (excellent). The scores are averaged to provide the MOS for the codec.
Voice Payload Size (Bytes) The voice payload size represents the number of bytes (or bits) that are filled into a packet. The voice payload size must be a multiple of the codec sample size. For example, G.729 packets can use 10, 20, 30, 40, 50, or 60 bytes of voice payload size.
Voice Payload Size (ms) The voice payload size can also be represented in terms of the codec samples. For example,a G.729 voice payload size of 20 ms (two 10 ms codec samples) represents a voice payload of 20 bytes [ (20 bytes * 8) / (20 ms) = 8 Kbps ]
PPS PPS represents the number of packets that need to be transmitted every second in order to deliver the codec bit rate. For example, for a G.729 call with voice payload size per packet of 20 bytes (160 bits), 50 packets need to be transmitted every second [50 pps = (8 Kbps) / (160 bits per packet) ]

Ok, The theater is all set now. Let’s jump into some calculations Bandwidth Calculation Total packet size = (L2 header) + (IP/UDP/RTP header) + (voice payload size) PPS = (codec bit rate) / (voice payload size) Bandwidth = total packet size * PPS Sample Calculation For example, the required bandwidth for a G.729 call (8 Kbps codec bit rate) with the default 20 bytes of voice payload is: Total packet size (bytes) = (MP header of 6 bytes) + ( compressed IP/UDP/RTP header of 2 bytes) + (voice payload of 20 bytes) = 28 bytes Total packet size (bits) = (28 bytes) * 8 bits per byte = 224 bits PPS = (8 Kbps codec bit rate) / (160 bits) = 50 pps => Note: 160 bits = 20 bytes (default voice payload) * 8 bits per byte Bandwidth per call = voice packet size (224 bits) * 50 pps = 11.2 Kbps Now let’s say we would have received a number 200 From the the Erlang calculator, then the required bandwidth requirement scales up to (bandwidth per call * total number of trunks that are needed) = 11.2 * 200 = 2240 kbps. On top of that there would be some percentage of additional (in a factor of 20-25%) bandwidth would be considered to compensate for factors like network re-transmissions, variance and collision. Which would eventually lead us to a figure of 28000 Kbps = 2.8 MegaBytes per second. So the bandwidth requirement would be an approximate 3MBps. However since each codec offers an entirely different sampling size it would be possible to achieve the same call rate with the desired GOM. Hope I was able to do some justice to the topic. If there is any way in which this article can be improved, please let me know. Thanks for stopping by. References: http://www.cisco.com/c/en/us/support/docs/voice/voice-quality/7934-bwidth-consume.html#formulae http://www.wekipedia.org http://www.cs.ru.ac.za/courses/honours/RTMM/software/52-VoIP-Bandwidth.pdf the image was taken from http://m.flikie.com/33582293/beautiful-night.html?cid=33554432&order=feellucky Source: http://abhishekchattopadhyay.wordpress.com/2014/06/07/sip-trunk-bandwidth-calculation/

Three Essential Telephony Services for Your Business

20 Jan

If you look back a few decades ago, the average corporate phone system was rather simple, involving a switchboard and several desk phones around the workplace. These days though, new technological developments have combined with greater competition to create an environment in which the simple telephony service of the past has evolved into a very different character. If you’re confused about which options to take, here are three of the best to benefit your business.

pbxHosted IP PBX Services

In the past, a large business would have to invest in an expensive private branch exchange (PBX): a system of telephones, switchboards & cables that ran throughout the office. These required external contractors for repair and maintenance as the typical corporate employee wouldn’t have the technical experience.

Needless to say, this was a bulky, inefficient option that has now been replaced by the hosted IP PBX service which requires no setup costs or technical expertise. With these packages, your provider has already established a state-of-the-art system on their premises. For a monthly fee, you can tap into this and gain a fully functional communications platform without any large installation and maintenance costs. These systems also offer other features such as:

  • Unified messaging
  • Softphones

This is the ideal solution for anyone seeking a better corporate telephone system without the hassle and cost of installing one directly onsite.

Portrait of customer service operators communicating in a call center

Call Answering Providers

When running a company, you’ll also have to determine how to successfully handle your incoming calls throughout the day and night. While you could hire a team of fulltime receptionists, the resultant salaries can be quite costly. A better solution is to outsource these services to a locally based agency instead. The good news is that anyone can find telephone answering packages tailored to their company. In this way, you can hire a third party to take your excess calls whenever you need it. They will be fully trained in your preferred manner, systems and corporate details too, allowing you to handle all incoming calls in the following situations:

  • Talking to customers after hours and on weekends
  • Temporarily replacing receptionists who call in sick
  • Hiring additional staff during high volume call periods

Thus, you can take important customer messages regardless of what’s happening without any reduction in your quality of service or client satisfaction.


Voice over IP (VoIP) Packages

The last essential service is the VoIP package, an option that allows you to make calls over the internet instead of through the traditional phone line. Not only will this save you money when it comes to overseas or interstate calls but the call quality will also be a lot better especially if you’re using fibre optics or a 4G data connection. If you’ve chosen an affordable call handling package for your business, you should then have enough money to consider a VoIP phone system and call plan as well! Of course, you’ll have to consider the following options before you sign the contract:

  • The cost per month or per call of the package
  • Whether you can call for free to any devices or apps
  • The change in rates for long distance phone calls
  • The type of softphone that you need to install
  • The compatibility of these with your current setup

Just think carefully and compare what’s on offer, and you should then be able to choose a VoIP provider and package that suits your corporate needs perfectly.

These three services are absolutely essential for the modern business. If you’ve yet to experience their benefits, we recommend you get in touch with the appropriate specialists as soon as possible. The success of your business depends on this!


Source: http://nancy-rubin.com/2014/01/16/three-essential-telephony-services-for-your-business/

Who’s Winning the Global VoIP Race?

12 Nov

VoIP global dynamic

As global telecommunication networks shifts from older, copper wire based systems towards Internet based communications, some nations and world regions are further ahead than others.

In the next few years, Voice Over Internet Protocol (VoIP) is set to be the largest growing industry worldwide. Just as the development of railroads and teelphone lines helped fuel the growth of industry in the 20th century, the pace at which nations are able to adapt to digital communications will play a big role in determining their success in the online marketplace of the 21st century. Because VoIP does not rely on expensive phone networks, but rather on Internet networks, there is a large opportunity for VoIP growth in countries that traditionally have not had strong infrastructures.

Countries like India and Indonesia are actually some of the fastest growing countries in terms of VoIP growth, particularly in the mobile VoIP space as more and more people are able to afford cell phones that can utilize VoIP through data connections. Countries that have heavily invested in Internet infrastructure, like Japan and South Korea, are seeing a strong return on investment as more of their people and businesses make the switch to digital communications. Additionally, as more people make the switch from traditional phone lines, there becomes less of a need for maintaining and expanding high cost telephone networks.

Taking a broad look at the changing global dynamic, the following graphic from WhichVoIP pulls statistics from recent studies of VoIP growth to map out which nations are leading the way with VoIP, which are lagging behind, and the socio-economic factors that influence their rise and fall.

Developed by WhichVoiP.com

VoIP changing global dynamic

Voice Over Internet Protocol (VoIP) is the future of global telecommunications

Source: http://itnews2day.com/2013/11/12/whos-winning-the-global-voip-race/

The Case for VoLTE

9 Sep

Despite the introduction of LTE with its heavy focus on improved mobile broadband speeds, mobile operators still rely on voice and SMS services for a large part of their revenues (roughly 70% globally). In previous generations of mobile technology these services were explicitly supported as part of the mobile network stack, with voice bearers supported in the radio access network and SMS making use of the voice signalling mechanisms. Indeed in 2G mobile standards data was originally only supported by sending data over a nailed up voice channel (high-speed circuit switched data or HSCSD) with the native data transport mechanisms of the general packet radio service (GPRS) introduced later as part of the so called 2.5G standards and in enhanced form as EDGE with 2.75G.

With the advent of 3G mobile standards (such as UMTS) voice and data were catered for on an equal footing, with distinct voice and data air interfaces being defined, each optimised for their respective payload. The lu-CS interface, used for voice, uses circuit switching to provide deterministic, guaranteed capacity for in-progress voice calls. In contrast, the lu-PS interface, used for data, uses packet switching on a shared data carrier to maximise the efficiency of data transport.

Compared to 2G, the 4G LTE standard turns the situation on its head and is optimised for data services without any specific native support (i.e. circuit switched) for voice transport. The rational for this is that broadband traffic is now the predominant use of mobile bandwidth and it is better to optimise the network for data transport and carry voice as an application over data using voice over IP. This is possible in an LTE network due to the substantial enhancements in the data plane, which compared to 3G has significantly reduced latency and has the quality of service mechanisms required to support a good quality voice service.

Although LTE was, from the outset, designed to support voice service via voice over IP (VoIP), the call flows, their associated signalling and media encoding where not defined or standardised. So although all the “hooks” where in place to support voice their was no voice standard that an operator could deploy and indeed there were many options, a number of which were debated at length by equipment vendors and network operators as part of the standards process.

Given the importance of voice (and SMS) this lack of standardised support for voice could be regarded as a serious omission. To mitigate this a mechanism called circuit switched fall-back (CSFB) has been introduced which entails the phone switching to 3G operation when a voice call is to be made or received.

As indicated above, there has been much debate about the standard approach for supporting voice (and SMS) over LTE. One approach, called VoLGA, which was proposed was to utilise existing UMA mechanisms such as those supported on some mobile phones (notably Blackberries) to tunnel voice across WiFi. Whilst this would have leveraged existing voice switching networks it had the disadvantage of being quite backward looking and not providing a future path to full multi-media communications. Instead the approach using native VoIP with SIP signalling and an IMS (IP multi-media subsystem) core found favour as the approach for voice over LTE (VoLTE) with circuit switched fall-back being used as a transitionary step.

As of today (September 2013) VoLTE standardisation is more or less complete and both handsets and network equipment are available. Nevertheless, there are very few deployments of VoLTE – to this author’s understanding one in the US (MetroPCS) and three in South Korea (SK Telekom, KT and LG U+). Whilst it is clear that operators will eventually embrace VoLTE what is not clear is how rapidly operators will move to do so.

On the one hand aggressively introducing VoLTE could enable network operators to refresh their voice product set, so as not to be left behind by the over the top (OTT) voice and messaging service providers such as Skype and WhatsApp. On the other hand, existing voice and SMS revenues are not yet necessarily under threat and it may be better to take a slow and steady approach. This paper sets out the different approaches network operators could take and the rational for preferring one approach to another.

Broadly speaking network operators have three approaches they may take for introducing VoLTE (with various shades of grey in between). They may:

  • Stick with the existing solution of circuit switched fall-back (CSFB) for the foreseeable future,
  • Gradually introduce VoLTE over a period of time thus operating CSFB and VoLTE in parallel, or
  • Aggressively migrate to VoLTE to obtain the benefits of improved service and network rationalisation sooner rather than later.

Circuit Switched Fall-Back (CSFB)

For an operator with an established 3G network and customer base, continuing to use CSFB (circuit switched fall-back) can make a lot of business sense, as it continues to exploit an existing asset that often will already have sufficient call carrying capacity. In this case, for those customers with LTE handsets, the downside of CSFB is slightly impacted voice service performance:

  • There is an increased call set-up time of several seconds as the phone connects to the 3G network, and
  • Data connections will likely be dropped as the phone reconnects to the network and is likely allocated a different IP address.

Whilst these issues do impact the service they are unlikely to be sufficiently troublesome to the customer to underpin a business case for the substantial investment required to enable VoLTE.

Introducing VoLTE into a network is not straightforward. It requires a new mobile voice core (IMS) and represents a substantial change in the way voice service is provided. This change from circuit switched to end-to-end VoIP as required by VoLTE means that new skill sets are required within the organisation and there is much to be learnt with respect to how to optimise the network to get the best voice performance. Putting aside business case considerations, these factors alone would suggest that an operator should approach the introduction of VoLTE cautiously.

OTT Service Competition

Whilst operators should approach the introduction of new technology cautiously, operators may also wish to seize the opportunity presented by the introduction of an IMS core as a enabler to more generally reinvigorate voice service, linking the launch of VoLTE service to wider reaching service improvements:

  • Introduction of a better quality voice service, with wide-band CODECs and faster call set-up time,
  • Linking to the introduction of RCS (rich communication services), often marketed as “joyn”, incorporating presence, instant messaging, group calling and video sharing,
  • Enabling service to be accessed more generally, for example, enabling calls to or from the mobile phone number to be made from a PC client, tablet or fixed line.

Nevertheless, it can be argued and often is, that these services will not attract additional revenue but instead just protect current revenues, giving customers additional value for their money, making them less inclined to use alternative services. In this view, customers will not pay more for voice services and consequently other revenues sources must be sought – of which the most likely candidate is video. For example a video sharing service called “See what I can see” is a service that we can all identify with. Additional revenues associated with such new services, however, are unlikely to pay for the cost of introducing an IMS core to support VoLTE and the rich communications services (RCS) server.

In the opinion of this author, much of the analysis presented in the previous paragraph is in all likelihood correct but the conclusion that there is little value in investing in voice services is questionable. If voice services are genuinely under threat from competition, especially from over the top (OTT) providers and device manufacturers such as Skype, WhatsApp, Google, Apple and even FaceBook then there is financial value in defending existing revenues. It should not be forgotten that much of the competition from OTT providers has heretofore impacted fixed voice revenues. However, with the increasing performance of mobile broadband networks, especially after the wide spread introduction of LTE (and in the future LTE-A) there will be little to stop OTT providers encroaching on traditional mobile operators’ revenue streams.

Compared to fixed operators, mobile operators are typically at an advantage as mobile calling is normally part of a bundle and if a subscriber makes fewer calls within the bundle it actually saves the operator money. This is fine in the short term, but customers will sooner or later perceive that when they are making few traditional voice calls, and not sending so many SMSs that they are getting poor value for money and migrate to operators who are selling “value based” mobile packages with a focus on charging just for data. In this world, operators will then fall into one of two categories:

  • Those that effectively compete with OTT providers and retain voice and messaging revenues, or
  • Those that accept that they cannot compete with OTT providers and focus on providing a mobile broadband pipe in the most cost effective manner.

Currently most operators see themselves as being in the first category but are doing very little to make this a reality. Much more could be written on this topic, and this will indeed be the topic for a future paper.

The problem for operators faced with producing a business case for VoLTE and by implication IMS is that there is much uncertainty to the extent of the risk from competition from OTT services and equally there is uncertainty into how well defensive measures against the threat may work. In most organisations it is difficult to get financial approval for such uncertain business cases. Indeed matters can be worse than this, for example, a strategy to replace SMS by an operator’s own equivalent of WhatsApp could well be perceived to undermine existing SMS revenue and consequently be unpalatable.

It is seen in industries again and again that the market leaders are unwilling to risk undermining existing revenue streams and investing in uncertain innovation. This is left to entrepreneurs who thrive on risk and are willing to take a chance, “having a go” at the established market players. Few typically succeed, but the ones that do reap rich rewards, whilst the original market leaders fade into insignificance. One only has to look at the computing industry where DEC and Sun are no more, and the situation in the mobile industry where Nokia and Blackberry have fallen from their positions of pre-eminence.

In summary, a small proportion of operators will believe that it is their strategic interest to deploy IMS and VoLTE and will find ways to make the business case work. However, most operators will not take this leap of faith and will take a much tougher view of revenue threats and opportunities, effectively only considering in the bottom line of the business case those factors that they can be definitive about.

Operator Benefits

Looking beyond the direct customer benefits, from an infrastructure perspective, VoLTE presents a number of opportunities to network operators:

  • Over time it will let them simplify their network as the old circuit switched voice infrastructure is closed down.
  • It will no longer be necessary for the operator to continue operating 3G service in all LTE coverage areas (as would be required by CSFB). This means that the 3G spectrum can potentially be re-farmed, for example to increase the available LTE spectrum.
  • The complexity of handsets may be reduced. At the present time, for example, there are not chipsets that support both LTE and CDMA. In this case dropping support for CDMA would reduce the cost of the handset – but would of course require the voice to be transported natively over LTE (via VoLTE). This is less of a driver for 3G UMTS operators as there are chipsets that support both this and LTE.

It will be clear that most of these benefits come from being able to turn circuit switched voice infrastructure off and re-farm 3G (or 2G) spectrum as LTE. The most significant barrier to this is likely to be the existing customer base and their handsets. Whilst the traditional upgrade cycle for mobile phones is quite rapid it is still in most markets too early too expect everyone to upgrade to the latest LTE SmartPhone that supports VoLTE. Moreover most operators will have a sizeable rump of low usage customers who change their handset infrequently, and whilst these will not be the highest paying users an operator will still be reluctant to loose these subscribers.


Another challenge at the present time with VoLTE is the support for roaming. Whilst the technical (3GPP) and interconnect (GSMA) standards are being progressed, even when they are complete, for the initial network deployments of VoLTE there will be few other networks to connect to that do have VoLTE – so for roaming users, even if VoLTE is present in the home network, circuit switched fall-back will have to be relied on for a substantial period yet. This means that handsets for users who expect to roam will have to support 3G with circuit switched fall-back for voice.

Example 1: MetroPCS

One of the few network operators to have launched VoLTE is MetroPCS in the US. What the author understands of their case, based on publically available information and assumptions is the following:

  • They are a pre-pay operator focused on specific markets in the US.
  • They have limited spectrum and are keen to re-farm existing 3G spectrum as LTE.
  • Their existing 3G network is CDMA requiring dual chipset in the handset to support both it and LTE.

MetroPCS’s public statements make it clear that they are aggressively launching VoLTE in their different markets with a view to:

  • Simplifying / reducing the cost of the handset as just including an LTE radio means that a CDMA chipset is not required.
  • Enhancing the service via the introduction of RCS based services.
  • Plan to shift investment to LTE infrastructure only with a view to re-farming CDMA spectrum as LTE when possible.

Given MetroPCS’s heavy promotion of the benefits of an LTE only handset with VoLTE. It must be assumed that the majority of its customers have limited need for coverage outside of their specific market area (though this will improve as MetroPCS rolls out LTE to more of its markets) and limited, if any roaming requirements. This may well be true for their target market segment – low cost, pre-pay users. It is assumed that users requiring great coverage flexibility will continue to use at least dual band LTE / CDMA handsets.

To enable it to cap investment in CDMA and eventually re-farm CDMA spectrum as LTE MetroPCS will need to rapidly shift its customers to LTE / VoLTE capable handsets. It is assumed that given the nature of the pre-pay market, there is a high degree of customer churn which could facilitate this. Nevertheless, to achieve it, MetroPCS will require an attractive range of LTE/VoLTE handsets, something which is still somewhat of a challenge.

Note that as of May 2013 MetroPCS has been acquired by T-Mobile. This may well change their approach to the aggressive introduction of VoLTE as it offers them a number of options to use T-Mobile’s existing coverage and extensive handset range. Nevertheless, MetroPCS has acted as a proving ground for VoLTE and this may well also lead to a change in T-Mobile’s approach.

Example 2: EE

Following the merger of Orange and T-Mobile in the UK to form EE, EE are now the largest mobile operator in the UK. They were the first operator to launch LTE in the UK using their existing 1800MHz spectrum prior to the auction of the “official” LTE spectrum.

EE’s public position is that whilst they will to continue to invest heavily in LTE to improve data rates they are not aggressively developing VoLTE. They state:

We found circuit-switched fallback [CSFB] for voice very stable from the beginning by putting deliberate aspects in for it with our core network vendors. We find that fallback is very reliable, and delays are minimal. It’s efficient as a solution, so we haven’t seen a need to rush to VoLTE. It will bring some additional benefits, but CSFB solution is good enough as a service.

It would be a fair guess to make that EE are actively investigating VoLTE and ensuring that new SIMs and handsets will be compatible with it wherever possible. It is also fair to say is that they do not see a business case for deploying VoLTE in the near future. This makes sense for EE:

  • They have more than sufficient spectrum holdings to operate 3G and LTE,
  • They have an existing voice network with sufficient capacity for their needs,
  • They are focussed on promoting mobile broadband as their “hero” product, exploiting the market lead they have over the other UK mobile network operators,
  • They believe the voice service provided by circuit switched fall-back is good enough for the time being.


It is clear that over a period of time most if not all operators will deploy VoLTE and migrate customers to it. What is less clear is how rapidly they will do so.  Clearly MetroPCS have proceeded rapidly in the US whereas EE in the UK are proceeding cautiously and are quite happy at present with circuit switched fall-back. These two examples could be viewed as extremes of network operators’ positions. That being said, the author would not be surprised if, in light of their merger with T-Mobile, MetroPCS rollout less aggressively, and no doubt behind the scenes EE are working on their approach to VoLTE. In reality most operators are likely to fall somewhere in between.

For an operator considering VoLTE the benefits will fall into two categories:

  • Voice product enhancements potentially also including multi-media and RCS based services, and
  • Network rationalisation opportunities, whereby 3G equipment and spectrum can eventually be retired.

Operators in general prefer concrete business cases with clear cost benefits. The financial benefits associated with product enhancements are notoriously difficult to quantify whereas those associated with network rationalisation more straightforward. In an ideal world an operator would justify the cost of VoLTE deployment based on the network rationalisation savings and consider the benefits gained from product enhancements as “up-side”. For most operators, a business case of this nature is unlikely to show a return.

For substantial savings to come from 3G infrastructure rationalisation, network operators will need to shift a significant majority of customers to LTE and VoLTE capable handsets. This includes not only consumers but also business customers who may have large deployments of 3G (or even 2G) devices. Whilst the upgrade cycle for mobile phones is often quite rapid there will nonetheless be a sizeable rump of customers that will continue to have 3G phones for a considerable period. This inevitably will limit the speed at which operators can force a migration to LTE and undermine business cases based on network savings.

Clearly, given that one of the major barriers to migration to VoLTE is the entrenched base of customer handsets, operators should be planning now to ensure that any as many of the new handsets and SIMs deployed over the next few years are LTE and VoLTE capable. Even this though will just be the SmartPhone customer base. Whilst SmartPhone penetration is growing rapidly it is not clear if and when SmartPhones will entirely replace more basic phones and at what point a basic, low cost LTE/VoLTE phone might be manufactured as an alternative to todays basic 2G/3G phones.

Considering now the potential of product enhancements: VoLTE requires the introduction of an entirely new IMS voice core and major changes in the way the network is designed and operated. Business case aside, for any operator, this is no small undertaking. As has been indicated the benefits from product enhancement will be difficult to quantify and may be considered as a way to protect existing revenue from OTT competition rather than as a way to gain new revenues. Therefore, for most operators there is going to be a marked reluctance to move rapidly to VoLTE.

Most large operators are naturally conservative in nature and would view the case of moving to VoLTE as having questionable financial benefits and significant technical risk. That is not to say that sticking with circuit switched fall-back is not without risk. Indeed the voice product offered by mobile and fixed voice operators has changed very little over the last 15 years. Although handsets and the features they provide have been transformed the basic service provided by network operators remains the same. In the long run, this is not a sustainable position – operators will either need to improve their products or see customers move to OTT voice and messaging providers.

VoLTE actually offers an opportunity to operators to break from the legacy voice and messaging services and innovate. True, it is not yet clear what will be successful in the market, and to what extent it will generate new revenue. Nevertheless, operators need to start planning for IMS and VoLTE, it is to their advantage at the moment that they can start small, substantially reducing the risk for something they will inevitably have to do. In the view of this author, the prudent approach for most mobile operators with respect to VoLTE is the following:

  • Push SmartPhones and make sure they are LTE and VoLTE capable. Ensure that all SIMs are IMS compatible.
  • Build a small IMS core with associated systems and experiment – both technically and product-wise. Do not just limit service to mobile devices, but allow users to access their phone service from any device.
  • Get IMS based VoLTE services in to the hands of a number of trial customers who are enthusiastic to try different services.
  • Start making plans and building the key network and IT building blocks that will be required in the future for communication services. Which services will be successful is not necessarily clear, but the building blocks of future services are easier to define.

This is not to say that operators should proceed in undue haste, but they need to spend the time learning how to deploy voice over IP based VoLTE services, the service opportunities this presents and the challenges to their current systems and practices of offering something other than a basic voice service.

It may well pay operators to wait for VoLTE to be more mature, for IMS systems to have further developed and come down in cost, and for IT systems to have developed to support a rich fixed-mobile product set. What will be true, however, is that the knowledge they can gain now by experimenting and trialling will pay them back handsomely when the time does come to put together a robust business case for VoLTE, define system architectures, select vendors and move to deployment.

The message of this paper is that for most operators the case for aggressively implementing VoLTE is likely to appear weak. Nevertheless it is very important that VoLTE should not be ignored. It is for good reason that vendors and operators chose an IMS based approach for voice over LTE – they saw this as the best way to evolve voice service, to enable them to incrementally enhance their product. The alternative is for an operator to focus on delivering mobile broadband in the most efficient manner, at the lowest possible cost and accepting that voice and messaging revenues will eventually drift elsewhere.

The question of what strategy an operator should adopt for VoLTE forces them to consider the broader strategic question of the nature of their core business. Should they continue to remain a vertically integrated company offering all communications services or focus on operating a low-cost bit-pipe network that other service providers can overlay services upon. This is a hard question to answer; most operators will say they want to adopt the former position but then do nothing about it. A well-considered project to trial and implement (albeit at small scale) VoLTE actually offers them the opportunity to jump-start this process: understanding the challenges of launching more complex communications services, the impacts on network and IT systems, and the likely customer interest in such services.

Operators that ignore this risk falling between two stools:

  • Not being ready to launch VoLTE and other VoIP based communications services and seeing increasing revenue erosion to other operators or OTT service providers.
  • Not aggressively simplifying and cutting cost to make the network as cost effective as possible for basic bit-pipe based services and struggling to meet the price point set by other more efficient operators.

If past history is anything to go by, many operators will struggle to adapt rapidly enough to the changing market. Quite how this will play-out is yet to be seen – unlike other high technology industries network operators have the entrenched asset of their deployed network. Whilst this will protect them somewhat, networks are increasingly capital intensive and an operator will need strong revenues to continue to fund developments such as LTE, VoLTE and the next generation of LTE (LTE-A).

Source: http://pscomms.wordpress.com/2013/09/07/the-case-for-volte/

Network Instruments’ survey shows growing adoption of trends, but concerns about visibility into the network remain.

30 Jul

Cloud, UC, BYOD Making Network Monitoring Difficult: Survey - See more at: http://www.eweek.com/networking/cloud-uc-byod-making-network-monitoring-difficult-survey#sthash.V8gnWYVg.dpuf

Cloud computing, unified communications and BYOD promise to bring big benefits to organizations, from greater collaboration and productivity to improved efficiency and lower costs.

However, the trends, which are hitting the data center at the same time, also pose some significant challenges, not the least of which is gaining enough visibility into the networks to ensure that the IT staff can properly manage and secure them, according to a survey by Network Instruments.

“The technologies are kind of being forced on them,” Brad Reinboldt, senior product manager at Network Instruments, told eWEEK. “They need the technology,” but need the tools to manage and monitor them properly.

Among the findings in Network Instruments’ Sixth Annual State of the Network Global Study were that organizations are saying that bring-your-own-device (BYOD) technology is the most difficult to monitor, and that bandwidth demand will continue to spike as these new services and technologies are incorporated.

The survey by Network Instruments, which makes and sells network management solutions, was released July 23. The results were drawn from responses from 170 network engineers, IT directors and CIOs in a number of regions, including North America, Asia, Europe, Africa, Australia and South America.

For the various data center trends, the company found that most IT professionals understood the benefits cloud computing, BYOD, unified communications (UC) and faster bandwidth will bring to their companies, but also worried about managing and securing the company’s data.

For many businesses, UC is quickly moving beyond voice over IP (VOIP) and into new areas, including videoconferencing, Web-based collaboration and messaging. VOIP deployments are staying around 70 percent, but 62 percent of respondents said they have deployed videoconferencing, and more than 60 percent have deployed instant messaging. Adoption of videoconferencing and instant messaging both grew more than 35 percent over the last four years, and more than half of organizations this year have deployed Web collaboration applications, such as Cisco Systems’ WebEx.

“Traditionally, UC was very focused on the voice aspect,” Charles Thompson, director of product strategy at Network Instruments, said in an interview with eWEEK. “We’re really seeing people adopting more than just voice.”

That’s bringing with it some monitoring problems, Thompson said. More than two-thirds of the respondents said their biggest challenge is gaining visibility into the user experience, and UC tools won’t be utilized to their full potential if users are reluctant to use them because of latency or jitter problems with the video, for example, he said.

Respondents also said they were concerned about the difficulties assessing bandwidth used by UC programs and the inability to view communications at the edge of the network.

In last year’s survey, 60 percent said their organizations had adopted cloud computing. That number jumped to 70 percent this year, with 39 percent having deployed private clouds and another 14 percent leveraging external private cloud services, such as Amazon Virtual Private Cloud, Savvis Symphony Dedicated and Citrix Systems’ Cloud.com.

Most organizations expect that about half of their applications will be in the cloud within the next 12 months, with the top cloud services being email at 59 percent, Web hosting at 48 percent, storage (45 percent), and testing and development (41 percent).

Twenty-three percent of respondents said they had moved VOIP into the cloud, though only 16 percent had migrated complex services, such as enterprise resource management, in that direction.
Data security remains the top concern about the cloud, with 80 percent calling it the number-one worry. Other concerns include compliance challenges, the lack of ability to monitor the user’s experience and to assess the impact cloud is having on network bandwidth. However, 43 percent said the availability of applications in the cloud had improved, and 37 percent said the end-user experience in moving to the cloud also improved.

The adoption of 10 Gigabit Ethernet in the data center is rising rapidly, with 77 percent of respondents saying they will use the technology within the next 12 months, a growth of 52 percent over the last four years. Twenty percent said they will migrate to 40GbE within the next year.

Businesses are anxious to get to 40GbE to help ease bandwidth issues caused by such trends as UC, BYOD and cloud, Network Instruments’ Reinboldt said. ”There’s just too much data,” he said. “There’s so much pushing through the pipe … they can’t wait anymore.”

With applications and networks growing in complexity, resolving problems increasingly becomes an issue. The biggest concern in this area was the inability to identify the source of the problem, according to 70 percent of respondents. Another third said they were still having trouble with bandwidth, according to the survey. 

Thanks to eWEEK for article.

Source: http://telnetnetworks.wordpress.com/2013/07/29/network-instruments-survey-shows-growing-adoption-of-trends-but-concerns-about-visibility-into-the-network-remain/

What is WEBRTC

1 Apr

WebRTC is a communications standard developed by the W3C in close cooperation with the RTCWeb standard developed by the IETF. RTCWeb functions at a lower protocol layer; WebRTC enables the embedding of this functionality in applications and websites.

The WebRTC standard solves a very common problem: incompatibilities for real-time communications. Today, to place audio or video calls from a computer, users need to download proprietary software and create accounts. WebRTC leverages the recent trend in which the web browser IS the “application”, and facilitates browser-to-browser communication, with no software downloads or registration needed. The browsers themselves include all the capabilities needed to support the standard. WebRTC standardizes communications between browsers, enabling audio and video communications, and data bridges to support text chat or file-sharing.


Though the WebRTC standard has obvious implications for changing the nature of peer-to-peer communication, it is also an ideal solution for customer care solutions to allow direct access to the contact center. For example, a user of a mobile customer care application could click one button to directly talk to an agent, without leaving the application or its context. Similarly, customers (or prospective customers) browsing a website, whether on a mobile device or at their computer, could easily initiative a chat with an agent. The overall effect is a seamless experience that eliminates the “context gap” – customers no longer have to search for a contact center phone number or wait for a call back, or re-explain the issue from the beginning to a contact center agent.

What does all this corporate speek mean?

Imagine clicking on a person name on your contact list and instantly being able to chat with them? No Downloads No Installs No hassles. No need for a degree in computer science to be able to use voice and video chat on the web.

All data, voice and video is seamlessly transmitted over the www backbone you never have to download a plug in or update anything ever again

SaCoderz is heavily invested in this new tech and will in the upcoming months be releasing a number of innovative products round this tech, watch this space

Source: http://sacoderz.wordpress.com/2013/04/01/what-is-webrtc/

Full Duplex Audio Engine for BlackBerry 10 VoIP Developers

3 Jan


Figure shows a high level audio block diagram of BlackBerry 10 audio subsystem and BlackBerry 10 APIs

If you haven’t already heard, RIM’s subscriber base just grew to 80 million users — and of those users, 60 million use BlackBerry Messenger (BBM). If you are thinking about growing your user base, why not just port your existing VoIP solution to BlackBerry 10?

For VoIP developers, how can this be possible? Well, it’s simple: BlackBerry 10 is powered by a QNX operating system, and you can benefit from this POSIX-compliant platform by porting open source libraries – whether that is PJSIP/PJMEDIA or your existing propriety VoIP stack.

As a VoIP developer, you may wonder if BlackBerry 10 provides the core ingredients for you to integrate your existing stack (SIP, RTP, Jitter buffer, SDP etc.) to BlackBerry 10’s underlying platform. Good news is that we have already done the work and validated this for you.

BlackBerry 10 audio subsystems consist of:

  • Full duplex audio support
  • Interface to the microphone, receiver, loudspeaker and headset
  • Analog-to-Digital Conversion for the microphone
  • Digital-to-Analog Conversion for the speaker
  • Hardware Audio Routing to select audio user interface (headset, speakerphone…)
  • Io-audio enables voice processing (AEC, NR, Gain Control) and routes audio to voice path
  • Automatic Least cost audio routing over Wi-Fi or cellular radio
  • Volume control

What’s important here is that these features are available to you today through the Audio Library APIs as part of the BlackBerry 10 Native SDK. The Audio Library is based on the QNX Sound Architecture API functions and has a lot of similarities to the Advanced Linux Sound Architecture (ALSA) APIs. These are not directly compatible, but for those of you that come from the Linux world and choose to use ALSA this would definitely be a good opportunity for you to port your existing code to BlackBerry 10.

So why wait – get started today! The knowledge base article below directs you to sample code that demonstrates how to integrate your existing VoIP stack with the BlackBerry 10 audio subsystem.


You can follow existing success stories at: http://blog.truphone.com/2012/05/blackberry-10-the-developers-advantage-1.html http://devblog.blackberry.com/2012/07/voip-development-on-blackberry-10/

PJSIP Blog: http://blog.pjsip.org/2012/06/14/initial-support-for-blackberry-10-bb10-now-available/ https://trac.pjsip.org/repos/wiki/Getting-Started/BB10

Source: http://devblog.blackberry.com/2013/01/blackberry-10-voip/

Features of Firmware Version 3.6.3 for Vigor2830 Series Router

30 Oct
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