Category: IP Multimedia Subsystem

Hosted IP-PBX and SBC

SBC ( Session Borde Controllers ) are basically gateways that provide interconnectivity between the hosted IP-PBX of the enterprise to the outside world endpoints such as telco service provider, PSTN/ TDM , SIP trunking providers or even third party OTT provider apps like skype for business etc. If you have a hosted IPPBX or PBX in your data-centre or on premise and you need controlled but heavy outflowing traffic, it is a good idea to integrate a resilient and efficient SBC to provide seamless interconnectivity.

Hosted PBX

For an enterprises such as an Trading floor or warehouse with multiple phone types , softphones , hardphones , turrets etc distributed across various geographies and zones a device agnostic architectural setup is prime . Listing the essentials for setting up such a system. Note supplementary services are data-services , logging , licensing etc are important but kept out of scope to keep focus on functional aspects .

An enterprise application usually is structured in tiers or layers

  • Client tier – the networks clients communication to the central java programs . Runs on client machines
  • web tier – state full communication between client and business tier . Runs in server machine.
  • business tier- handles the logic of the application. The business tier uses the Enterprise Java Bean (EJB) container, which manages the execution of the beans
  • data tier – encompasses DB drivers . Runs on separate machines for database storage

Event services for Line status notifications

providers lines status notification across enterprise for inter zone and softphone to hardphone .

Routing services

routing calls within enterprise and hardphone sites read more about resource zones later in the article

Call Control Manager (CCM)

Consolidated set of all service and component that make up the VOIP platform besides media handlers. It includes SIP adapters, bridge managers, call processing frameworks, API frameworks, healthchecks etc.

Call processing framework ( CPF)

Signalling and call routing logic, mostly in SIP and trunks. Manages identities such as Call Line information, Called Party Information, line status etc in shared memory.

Multiple shared Lines and their statuses

Incases where there is a need to process multiple calls from a single User agent device such as a softphone or hardphone ( common scenario for a turret phone) , the design involves assigning it multiple sip uris and each sip uri will establish a line. When caller calls callee , the line is said to be BUSY , otherwise said to be IDLE. Transition of a shared sip line from IDLE to BUSY is transmitted to others via SIP PUBLISH as other UAs holding the same sip Similarly any other event like transfer is propagated to other via SIP UPDATE

Clustering Call control managers

A Call Communication manager (CCM) from various zones should be able to cowork on call and session management and advanced features such as routing from home guest zone to home zone , call transfer , refer , barge etc. Designing a clustered setup will also provide elasticity , fail-over and high availability. Can use clustered , HA compliant framework such as Oracle Communication Application Server , suited for enterprise level deployments.

Call Replication and distributed memory management

A node will store two types of data: active sessions and passive sessions. The active sessions are used by the node and stored in cache. The passive sessions are the replicas from the other nodes’ active sessions. The passives sessions are stored on a persistent storage.

Controlling Line Calls using AOR and Resource Zones

When dealing with many SIP endpoints , now referred to as resource, it is best to assign the resources to their respective zones. Thus a resource’s status updates will be only updated by its active resource zone while can be read by any resource zone.

Incoming request Zone vs Active Resource Zone

For an Incoming request such a INVITE , check whether the zone sending the request is its active resource zone or not .If the Active Resource Zone is the same zone on which the INVITE came in, then the call is handled by that zone. If the Active Resource Zone is a different zone, then the call needs to be forwarded to the Active Resource Zone.

Bridges for Local Media connections

Although call signalling is handled by a resources active resource zone only, we can still create media bridges in local zone of the resource .

Local MM bridges are used to auto answer an incoming sip line call and create trunk , especially from hardphones which do not support provisional responses.

Interzone proxy Handler

proxies call control messages between active and non active resource zones. Primarily mapping the sip messages with all custom headers inbetween the communication device interfaces.

Dial Trunk using multiple dedicated SIP lines and connect via Media Bridge

To save up on call routing /connection time and to support te ability to add as many users on call at runtime , a dedicated media bridge is established for every call.

  • A sip line activated is auto-answered by MM , creates a trunk and waits for other endpoint to join the bridge. The flow is as follows :
  • As INVITE arrives for an IDLE sip line , it is connected to a trunk and auto answered by a local MM bridge .
  • Since the call is already answered , when caller dials number for callee , collect the DTMF digits over RTP using RFC 2833 DTMF events.
  • Run inter-digit timer for digit collection and detect end of dialing on timeout.
  • The dialed trunk connection is made and call is added to media bridge
  • When provisional responses are received on the trunk connection, generate in-band call progress tones (ringing, proceeding etc) via the MM
  • When the line answers, the progress tones have to be stopped and the called party gets bridged to the calling party via the media bridge.

Call Diversion involves forwarding calls from zone to another zone. joinjed parties get call UPDATE status and forward response.

Call barge is the processing of joining an ongoing call . The barge event is usually propagated to joined parities via SIP INFO. Private lines do not allow barge in and are exclusively reserved for only few users.

Interconnectivity provided by an SBC ( Session Border Controller)

Hold-Resume and Music on Hold in multi-line evironment

While a regular p2p call involves simple reinvite based hold and resume with varrying SDP, the scenario is slightly more detailed for hold resume on bridged trunk connection , as explained below.

As the calls made are on bridge , a hold signal involves a RE-INIVITE with held-SDP to media manager (MM). If hold status on trunk is 200 OK the hold status will be sent to other call interfaces connected on the trunk. Else if hold is denied, 403 is sent back to hold-initiates.

Music on hold is an one way RTP mostly from media server.

For a bridged scenarios , separate Music on hold bridges are kept on Media Managers. When an UA has to hold , it is removed from original bridge and place on music on hold bridge. To be unhold/ resume it is placed back into the orignal bridge from music on hold bridge.

Conference

user initiates conference, the conference feature can execute on the zone where the user was logged on, irrespective of zones where the other conference attendees join from . The Call processing framework of originators zone completes the SDP exchange to establish two-way speech path among all the parties.

Incases there are multiple connections from a zone , a local MM conference bridge can be created for them which would connect back to originators MM conf bridge . this two part conf bridge will be transparent to the sip line sand users .

For provisioning inputs and settings setup a Diagnostics , Administration and Configuration platform which can process APIs for data services , licences , alarms or do remote device control such as using SNMP.

Session Border Controllers (SBC) role for PBX

At network level SBC operations include

  • bridging multiple interfaces in different networks even between the IPv4 and IPv6 networks
  • auto NAT discovery and STUN
  • protocol conversion such as TLS to UDP etc
  • Flood detection and IP filtering

For SIP specific functionalities, SBC does

  • SIP validation involving checks on syntax and message contents also consistency checks are performed.
  • stateful and call aware. tracing, monitoring and checking for validitya and health of all the SIP messages
  • Topology hiding
  • Traffic filtering
  • Codec filtering , reordering , media pinning, transcoding, or call recording
  • Data replication brings High Availability (HA) with hot backups or even Active-Active solutions.

Traffic sharing and routing roles of SBC can include

  • IP-based and Digest-based authentication
  • limiting traffic by number of concurrent calls or calling rate.
  • Dialplan and/or Custom routing
  • Dispatching/Load-balancing to a backend cluster of servers

SBC’s can be physical hardware boxes or software based applications, as the name suggests their purpose is to control the session at border between the enterprise and external service provider. They can be used for various roles such as

  • SIP to PSTN – SIP is an IP protocol whereas PSTN is a TDM one , achieving interoperability is also the KRA of an SBC
  • SIP trunking – SBC provide a secure sip connectivity to connect calls to sip trunks which provide bulk calls functionality at a flat pricing.
  • support for various fixed or mobile endpoints – SBC ensure they are RFC compliant and can extend SIP to any kind of telecom endpoint like PSTN , GSM, fax , Skype , sipphone , IP phones etc.
  • NAT (Network address translator) – To meet the packet routing challenges across a firewall or even during private -public mapping. A combo of DHCP servers and NAT provider comes very handy to reroute or perform hole punching such that signalling and media packets are not dropped and meet the required endpoint. More about NAT here – NAT traversal using STUN and TURN.
  • Load balancing – Reverse proxies and Load balancers is a much adopted industry practise to mask the inner IPs of the VoIP platform and also route traffic appropriately between control and media server .
  • Security, QoS and Regulatory compliance – since SBCs are required to typically support a large array of clients they adhere to regulatory and industry accepted standards ,which also involves security features like AAA, TLS/SSL and other means for quality of assurance like logging and fault detection, preventing DDoS etc . In many cases SBC can also encrypt / decrypt RTP streams for probing , tapping or lawful inspection .

Terminating at carriers, PSTN and IP gateways

There are 2 ways to integrate IP calls to telecom provider endpoints such as GSM or LTE phones.

  • PRI lines
  • SIP trunks
convergence

Additional SBC features

Inaddition to above it is good to have if an SBC provides extra features like forking , emergency number dialing ( 911 ) or active directory integration . Real Time Analysis and monitoring of call and metrics are also expected from a SBC since they reside on edge of the network and are more vulnerable to threats . For example Dialogic Mediant SBC’s and gateways , Audio Codes SBCs

With the shift from on premise PBXs to cloud based VM or microservice architecture , SBC vendors adopt a lager umbrella of services also including automation scripts for checks , reporting tools / consoles , developer friendly APIs to manage sessions via SBC and even WebRTC gateways to connect browser endpoints.

PBX connection to IMS systems

Typical VOIP/SIP scenario without IMS

A basic enterprise VoIP/SIP solution is illustrated in Figure. The key element is a soft switch (SIP PBX) which might be implemented as a combination of several SIP entities, such as SIP registrar, proxy server, redirect server, forking server, Back-To-Back User Agent (B2BUA) etc. SIP clients can be SIP hard-phones or soft-phones on PCs, PDAs etc. A PSTN gateway links the enterprise SIP PBX to the public PSTN. Enterprise applications, media servers, presence servers, and the VoIP/SIP PBX are interconnected through a company intranet.

VoIP System with IMS : With IMS, applications will be able to establish sessions across different access networks, with guaranteed QoS, flexible charging & AAA support. Call control, user’s database and services, which are the typical functions of softswitch, are controlled by separate units in IMS. CSCF (Call Session Control Function) handles session establishment, modification and release of IP multimedia sessions using the SIP/SDP protocol suite. Services features are separated from call control and handled by application servers. Subscriber’s database function is separated from service logic function and handled by HSS using open subscriber directory interface.

Link registration using subscribe-notify can be handled via Enterprise App server in PBX.

Forking proxy Setup of PBX : The enterprises SIP PBX can work as a forking proxy during call setup to redirect the calls.

Other usecases can involve presence sharing between different enterprise PBX with both domains interconnect their presence servers.

Telecommunications convergence – VoIP, PBX and IMS

UseCase Scenarios

Any VOIP dependant system which deals with bulksome voice / video traffic from external endpoints is a usages scenarios. Listing few

  • provision of pre-defined enterprise based SIP URI.
  • Contact Call centres
  • Remote work / offsite monitoring
  • CRM solution for sales/marketing
  • Connecting webrtc click to dial from webpage to enterprise representatives
  • connecting enterprise UCC clients to PSTN endpoints

The There are many more features and usecases for an IP-PBX solution for an enterprise. The features of modern IP PBX systems are a big addon to internal secure telecom channel in an company and accross its various office.

Future of IP PBX

There has been a significant shift in replacing hard PBX systems with software-based IP PBX such as using Freeswitch, Asterisk or other commercial-grade SIP servers which seamlessly integrate into other business software such as CRM systems, task force management systems.
In recent times cloud telephony providers, particularly CPaaS platforms have revolutionized the IP telecommunication landscape with lightweight and feature-rich communication agents( web, native platform) and services such as programmable API to control call logic and services such as recording, IVR announcements, call parking, Automatic Queueing so on.

Kamailio

Kamailio as Inbound/Outbound proxy or Session Border Controller (SBC)

Asterisk

Asterisk – installation and dial plans for WebRTC supported PJSIP clients

SIP/VOIP transformation towards IMS (Total IP)

The telecommunications industry has been going through a significant transformation over the past few years. At the outset incumbent operators used to focus on mainly basic voice services and still remained profitable due to the limited number of players in the space and requirement of huge amounts as initial investment.

However, with the advent of competitive vendors, rise in consumer base, and introduction of cost effective IP based technologies a major revolution has come about. This has enabled operators to come out of their traditional business models to maintain and enhance subscriber base by providing better and cheaper voice, multimedia and data services in order to grab the biggest possible share in this multi- billion dollar industry.

The evolution in Telecom industry has been accelerating all the time. The Next-Generation Operators wants to keep pace with the rapidly changing technology by, adapting to market needs and looking at the system and business process from multiple perspectives concurrently. Communication Service Providers (CSPs) need to consider several factors in mind before proposing any solution. They need to deploy solutions which are highly automated, highly flexible, caters to customer needs coupled with ultra low operating costs.

Upgrading a softswitrch solutions to IMS

The Softswitch is decomposed into two logical components of a subscriber-facing unit and a PSTN-facing unit. 

  • Subscriber facing unit in Softswitch is upgraded to AGCF (Access Gateway Control Function) 
  • PSTN facing unit is upgraded to MGCF (Media Gateway Controller Function) to interwork with IMS as shown.

By separating the Softswitch into these components, the network can be more easily scaled for better overall network efficiencies. More AGCFs can be added as required, allowing the network to scale with an increase in subscribers. Similarly, More PSTN trunks can be added as traffic increases. Once PSTN and subscriber control functions are separated, the IMS elements, CSCF and BGCF functions can be introduced. BGCF is the interface for interconnecting IMS with legacy PSTN networks.

New SIP-based services can now be rapidly rolled out by deploying new Application Servers (AS) and its integrations to other SBC for UCC( unified communication and colloboartion ) systems. IMS has 3GPP specified ISC interface, which is a SIP-based interface for interfacing-to-application servers. Using these constructs, multiple application servers from multiple vendors can be interconnected over the IMS ISC interface.

Intelligent Networks( IN)

Telecom networks (2014) are made up of integrated service digital network (ISDN), the public switched telephone network (PSTN) ,the Public Land Mobility Network (PLMN) and many others. Intelligent networks (IN) ensures that call control is handed over to a control platform. The control platform determines how the establishment of this call shall continue. Applying IN to any of these networks has in common that call establishment is intercepted at a designated node in the network

By hosting new services on the new platform and combining new and old services CSP‟s aim to provide service bundles that would generate new revenue streams. This process is largely dependant on IMS ( IP Multimedia Subsystem ) architecture .

Transformation towards IMS (Total IP)
Transformation towards IMS (Total IP)

Optimization in operator landscape evolve as result of synergistic technologies that come together to address the innovation and cost optimization needs of operator for better user experience. In following sections different technological evolutions that are affecting overall operator ecosystems have been discussed with focus towards Service Layer.

Fixed/mobile convergence(FMC) with IMS

“Fixed Mobile Convergence is a transition point in the telecommunications industry that will finally remove the distinctions between fixed and mobile networks, providing a superior experience to customers by creating seamless services using a combination of fixed broadband and local access wireless technologies to meet their needs in homes, offices, other buildings and on the go.”

 Fixed-Mobile Convergence Alliance (FMCA)  2004

System can communicate over the cellular network, or act as a new endpoint on the IP network. Home Subscriber Server (HSS) manages subscriber data uniformly between the cellular and IP worlds. The Handoff Server runs on top of the ISC interface, and provides a seamless experience when subscribers move from the cellular network to a Wi-Fi network. The AGCF remains the functional centre of the network, but with the introduction of the HSS, has added the Cx and Sh interfaces defined by the IMS.

Legacy to IP transformation

This section broadly covered the aspects of migration from legacy IN solution to new age JAINSLEE framework based one. Applies to Legacy IN hosting voice based services mostly  such as VPN, Access Screening ,Number Portability, SIP-Trunking,Call Gapping.

Most operator environments have seen a rise in the number of service delivery platforms. Also complexity of telecom networks have increased manifold hence CSPs are facing multiple challenges. Increased efforts and costs are required for maintaining all the SDP platforms. These platforms are generally of different vendors and cater to different technologies thereby greatly increase chances of limiting the scalability and flexibility of the operator landscape. More effort required for sustaining the life cycle of the platform and challenges in integrating non compatible SDPs due to proprietary design have been stumbling blocks in the progress of CSPs across the world.

To overcome these challenges there is trend in the market to move towards SDP consolidation wherein instead of maintaining several SDPs with their proprietary design CSPs prefer maintaining a single or less number of SDPs having standardized interfaces.

SDP consolidation
SDP consolidation (1)
SDP consolidation (2)

As illustrated in the above figure there is a transition that is taking place in the industry towards consolidation of service delivery session control. This would provide a cost effective sustenance of existing applications and the rapid creation and deployment of new services leading to increased revenue recognition by CSPs.

  • Agile Development
  • Innovative services
  • open SOA based architectures
  • IN/NGN Platform and Services
  • Reuse of existing investments in legacy service platforms
  • low cost of new service development
  • faster time to market
  • Monetize investment in Network Infrastructure uplift – SIP trunking, VoLTE etc.

Services that should be covered  in the Scope of Migration from fixed line to IP telephony are:

  • Virtual Private Network (VPN) : An Intelligent Network (IN) service, which offers the functions of a private telephone network. The basic idea behind this service is that business customers are offered the benefits of a (physical) private network, but spared from owning and maintaining it.
  • Access Screening(ASC): An IN service, which gives the operators the possibility to screen (allow/barring) the incoming traffic and decide the call routing, especially when the subscribers choose an alternate route/carrier/access network (also called Equal Access) for long distance calls on a call by call basis or pre-selected.
  • Number Portability(NP) : An IN service allows subscribers to retain their subscriber number while changing their service provider, location, equipment or type of subscribed telephony service. Both geographic numbers and non-geographic numbers are supported by the NP service.

 

WebRTC based Unified Communication platform

Using WebRTC Solution for Delivering In Context Voice which provides new monetizing benefits to the Enterprise customers of Service Providers. This includes following components:

  • WebRTC Gateway for implementation for inter-connect with SIP Legacy
  • Enhancement of WebRTC Client with new features like Cloud Address Book, Conferencing & Social Networking hooks.
  • Cloud based solutions
INtoJAISNLEE

Challenges in Migration to IMS  (Total IP )

Since long I have been advocating the benefits of migration to IMS  from a current fixed line / legacy/ proprietary VOIP / SS7 based system . However I decided to write this post on the challenges in migration to IMS system from a telecom provider’s view.  Though I could think of many , I have jot down the major 4 . they are as follows :

Data Migration challenges

  • Establishing a common data model definition
  • Data migration seamlessly
  • Configuration management
  • Extracting data from multiple sources and vendors , that includes legacy systems
  • Extracting data due to its large scale and volume

Training

  • Creating an effective knowledge share and transfer for live operations
  • Training in fallback plans, standards and policies .

Customer impact

  • Minimized customer outage
  • Enhance customer experience by delivering quality services on schedule
  • Ensuring security of customer’s confidential data
  • Transfer of customer services without any impact.

Testing in replicated environment

  • Physical pre-transfer test
  • Reducing cycle time
  • Verification and validation at every change in data environment
  • Detect production issues early in the test -lifecycle

Fallback plans

  • Pilot program and real network simulation for ensuring preparedness
  • Tracking changes in new network

 

IP Multimedia Subsystem (IMS)
IP Multimedia Subsystem (IMS) – detailed part2
Service Broker Architecture for IN and IMS
5G and IMS

Internet Telephony Convergence- JAINSLEE Platform

Convergence : Telephone networks and computer networks converging into single digital network using Internet standards.

Components in a Network

  • Client computer
  • Server computer
  • Network interfaces (NICs)
  • Connection medium
  • Network operating system
  • Hub or switch
  • Routers- Device used to route packets of data through different networks, ensuring that data sent gets to the correct address

1

Figure :simple computer network, consisting of computers, a network operating system residing on a dedicated server computer, cable (wiring) connecting the devices, network interface cards (NICs), switches, and a router.

2

Figure of a Enterprise Network : local area networks (LANs) linked to enterprise level / corporate network . It consists of :

  • Powerful servers
  • Web site
  • Corporate intranet, extranet
  • Backend systems
  • Mobile wireless LANs (Wi-Fi networks)
  • Videoconferencing system
  • Telephone network
  • Wireless cell phones etc

The convergence of Internet and Telephony opens up new revenue streams for the Communication Service Providers by delivering new innovation based convergent applications.

Before discussing digitization of Communication and telecom we need to first understand packet switching .

What is Packet switching ?
It is a method of slicing digital messages into packets, sending them along different communication paths as they become available. Then reassembling these received packets at destination. It is a more efficient use of network’s communications capacity.
Previous circuit-switched networks required assembly of complete point-to-point circuit.

What triggered this Technology development?

The Internet, IPTV and Social Media networking is evolving dynamically in the end user space of Communication Service Provider. This opens door for delivering new innovative services to end user through these converged applications.

A SP( Service provider ) has to work with multiple Communication Providers globally and based on the experience with the customers, has to conceptualize and implemented new innovative use cases on open platform to reduce the cost and  migrate from legacy to Next Generation Networks.

What does convergence mean to

  • Equipment Vendors / EV
    • Femto / FMC
    • Challenges in System Integration
    • Box to Solution Sales
    • Services software based
  • Enterprises
    • Low Capex – Hosted Models
    • Enterprise Mobility
    • IP Enabled Services
    • UC to UC
    • Web Integration with Mobility
  • Telco
    • Enterprise communication will be a big focus Area
    • Push to EVs for CAPEX
    • Wish to leverage Legacy as well
    • Challenges in Vertical Solutions
    • Will face challenges by OTT players

 

What will it do, how and in which situation ?

The underlying technology of Internet Telephony Convergent Platform is JAIN SLEE Framework which is open standard for developing core network based applications. It enables development of network agnostic applications , implemented through resource adapters for deploying same applications over different networks like SIP/IN etc.

JAINSLEE framework provides capability to form new complex services through reusable service building block in much easier way then traditional methods. This reduces cost for launching new services and bundled different services into the new convergent service in network agnostic way. 

It also bring benefits in term of reducing the dependency on Vendor proprietary platform and eventually bringing down cost involved and Time to market in launching new service.

The OpenCloud Service Layer OpenCloud

picture courtesy : Opencloud.com

What problem does this technology seek to solve?

Today communication service provider are facing vendor locking situation where most of services deployed are platform dependent which requires huge cost of investment for launching new services. Traditional service development platforms are major roadblock for operators to launch new collaborative services which involves both voice and data channels as they are not based on open standards and are tied to the vendor specific technologies. Also in a fast changing technology the operators need to switch their focus on new innovative services through which operator can monetize services and provide the value added experience to their end customers. To enable it we proposed and implemented framework which not only act as the new Internet Telephony convergent platform but also in sync with their future network transformation strategy as it is based on open standards. Through this platform same applications can be targeted to different segment of users with minimal cost impact. Some of the application which we have developed are detailed below.

a) Parental Control is an application through which parents can have control over their children’s Internet video on demand request. Once a child requests for any video, preview of the same(short clip of video) at the same instance is send to parents’ smart phones. Parents can see preview and can decide there and then weather it is adequate for his/her kids or not, and can either allow or deny through his mobile.

b) IPTV/VOD session mobility is a service which allow user to transfer their ongoing voice call/video-on-demand session from their smartphone to desktop/computing device/smart-device and vice-versa seamlessly.

c) Converged application like unified communication platform for trader community take advantage of both voice and data services and help trader community in terms of analytics and decision making process.

What is the specific breakthrough of this technology?

Internet and Telephony are two major drivers in Telecom domain. Hence the concept of convergence of Internet and Telephony is of great interest for the Telcos. Internet telephony, also known as voice-over-IP or IP telephony is the real-time delivery of voice between two or more parties, across networks using the Internet protocols, and the exchange of information required to control this delivery. New innovative use case scenarios  have been conceptualized and implemented considering new user behavior changes. These bring in value addition to CSPs in order to bring more revenue streams. Solutions like Secure VOIP bring another dimension of innovation as it provides a secured voice communication over the internet using open source software like Asterisk. This solution helps business reduce their operational communication costs using encrypted standard security algorithms.

Asterisk- Applications (1)

pic courtesy : asterisk.com

How does this technology compare with other technologies? 

Internet telephony convergent platform has the unique value proposition based on new innovative use case scenarios using multiple underlying technologies. These scenarios are implemented using Open Standards. Though many other vendors’ platform also provides some of the facilities of platform in part and pieces but none of them give complete end to end solutions suits to operators as our Internet Telephony convergent platform provides.

How does it help in achieving the goals?

We consider it as solution which can act as foundation block to build a long term partnership with operators especially in area of services landscape. This solution enables operator to monetize different voice and data convergent services and in sync with the operator’s next generation transformation initiative. The services acts as catalyst to increase the data usage of end-users. Strong business case can be built with these services by operators as they meet the future demands of tech savvy end users. These services not only fill the void between communication service provider and social media/internet/video-on-internet but also take advantage of reach of social media/internet and eventually enable operator to add new revenue stream. These services can also help operator to increase their brand visibility with added advantage of social media and internet application bundled with their core services. Operator can charge it on per application basis or can be just carrier and charge for data usage. Convergent services which involves both the voice and data, enable operator to charge on voice services , data services and application usage. With our rich experience in convergent platform domain we believe we can convert significant opportunities in this space.

Explain your journey of Technology development ?

After seeding of concept of Internet Telephony convergent platform SP should explore partner product Software centric platforms like Open cloud, Oracle, Mobicient etc which offers the capability to deliver convergent applications at a low cost and using the open standards. Standards like JAIN SLEE provide capability for developing and delivering such applications across different type of underlying network. 

Mobicents Platform

pic courtesy : Mobicents.com

One can develop the complete solution using such open, standard platforms as a base . The complete solution takes care of the real-network issues and solutions for the same. There were many hurdles and roadblock at first. Adaptation to open standards like JAIN SLEE requires fast ramp up as it is quite complex technology. In a small stipulated time a core team should have developed competency through Partner Training inputs and Brain Storming sessions. To test framework at lab, there would be dependency on many open source software and strategic partner products. There would be many incompatibility issues. Its important that such issues be  sorted out by exhaustive explorations of products and by bug fixes .

Benefits expected if this Technology is implemented / commercialized 

a) Communication service providers are able to realize appreciable cost saving through Internet Telephony convergent platform Operators deployed in their network. This is so legacy platform were costly and difficult to manage. This platform brings innovative and cost effective way of launching new collaborative services which brings new revenue stream.

b) Improved Time to market

c) Extensible architecture for the service helps in extending the service for multiple markets.

Social Benefits

Unified communications, where voice, video, email, text and other messaging technologies are combined to provide greater flexibility for users by enabling new ways to transfer information and manage connectivity. Integration of collaborative services with the social media platform like Facebook , Linkedin , Twitter etc, increases the connectivity and value experience of end users. Through social media based convergent applications operator can further increase their reach to end users by utilizing underlying the Internet Telephony convergent platform.

My Insights 

Based on my personal experience while implementing this technology/platform, I think this solution act as catalyst for enabling the transition from network eccentricity to customer eccentricity. This movement is further supplemented through the reduced dependence on legacy vendors and increased adoption of open standard based platforms. Through the converged application layer for Telcos I envisage a platform which is agnostic to underlying network layer. Unified platform allows carriers, mobile operators, and cable operators to rapidly create, manage, and deliver converged video, voice, and data service bundles across multiple networks and devices. It enhance end user experience and enable Telcos to add new revenue stream by offering value added services to their customer. 

IMSSF and RIMSSF

This post particularly describes the gateways in IMS which communication back and forth with a legacy endpoints.To get a overview of IMS itself click here  and to get a detailed description of IMS and its architecture click here .

What is IM-SSF  ?

IP Multimedia Service Switching Function is a  gateway to provide IN service such s legacy VPN ( Virtual Private Network ).

IMSSFaltanai
IMSSF

 

What is R-IM-SSF  ?

Reverse IP Multimedia Service Switching Function Works on reverse principle to connect IN network  to IMS services using IMS services such as FMFM ( find me follow me ) .

RIMSSFaltanai

More link on telecom transformation , migration and inter-opereability :

Transformation towards IMS 

 

IP Multimedia Subsystem (IMS) – detailed part2

This is a folow up on my previous post on IMS which described what is IMS and why it came into existing. Also how IMA can benifits with its rich feature set and huge OPEX savings. To read the previous post click here .

This post described IMS architecture and core concepts in detail.

ims

IMS Home Network

SIP call between 2 SIP  User agents or from SIP UA to PSTN endpoint.

ims2
  • HSS ( Home subcriber subsystem ) contains all of the subscriber information.
  • AS ( Application Servers ) conatin applications for example, be originating services or terminating services.
  • Filtering for applications for users is loaded into the S-CSCF and activated when the subscriber registers with the network.
  • DNS is used to identify elements use in the session set up.
  • The CSCFs manage the session control: registration, set up, tear down, feature activation.
    • The P-CSCF is first point of interaction with the User Agent. It also manages Quality of Service and other conditions specific to a UA.
    • The I-CSCF is used in network to network signaling. The I-CSCF hides the network topology from an external network.
    • The S-CSCF is the primary signal processing engine in IMS. It manages registration, checks for triggers for services and performs routing .
  • Media Resources may be conference services, IVRs or other network services.
  • If a call must egress to the PSTN the BGCF selects the appropriate Media Gateway that can be used.
  • Media Gateways control the conversion from IP to PSTN TDM signaling. Media Gateway Control Functions control the signaling between IMS and the PSTN (e.g. IP to SS7).

IMS Vistited Network <–> Home Network

Home Network to visited Network connectivity

ims3
  • If call orignates from Visited network the P-CSCF and S-CSCF in visited network itself , process the origination of the call and select the destination network.
  • Since call receiver of the call is in Home network the I-CSCF receives the call signaling from the Visited network, chooses the appropriate S-CSCF to process the call and the call is completed with RTP flow, depicted with blue line in diagram.

UE Registration

IMS registration is where the subscriber requests authorization to use the IMS services in the IMS network. The CSCF and HSS in core IMS network authenticates and authorizes the user .

ims4
  • The UA/UE initiates the registration process .
  • The SIP registration is passed to the S-CSCF.
    • For a user in Home network , the registration request is passed via P-CSCF to S-CSCF.
    • If the user is roaming in visiting network then the P-CSCF in the Visited network would pass the registration to the S-CSCF in the Home network through a I-CSCF.
    • Users are always registered in the Home network.
  • The S-CSCF forwards the request to the HSS via the Multimedia Auth Request (MAR) message to 1) download authentication data via the Multimedia Auth Answer (MAA) message and 2) inform the HSS that this S-CSCF is in control and any other queries to the HSS should be returned to this S-CSCF.
  • The S-CSCF creates a SIP 401 Unauthorized response that includes a challenge that the IMS terminal should answer.
  • The IMS terminal sends a new Register that contains the response to the challenge.
  • The S-CSCF validates the user and sends a Session Auth Request (SAR) message to the HSS informing it that the user is now registered and requesting the user profile, including services, that come in a Session Auth Answer message (SAA).

Subcription Changes

A registered user requires to be notified of his state changes. For example,

  • registration may be valid for a fixed period of time and then the network requires the user to register.
  • user or network element may go out of service and need to inform the other of some state change.

UA/UE subscribes to the registration state also P-CSCF serving the UA/UE subscribes so it can be informed.

ims5
  1. When the IMS terminal has completed registration the P-CSCF sends a Subscribe request for the registration event. The request is directed at the S-CSCF (which is in the Home network).
  2. The S-CSCF receives the request and installs that subscription, i.e. the S-CSCF takes the role of a notifier. The S-CSCF sends a Notify request to the P-CSCF. This request includes Public User Identities and the registration state.
  3. When the IMS terminal has completed registration it sends a Subscribe request for the registration event. The request is directed at the S-CSCF (which is in the Home network).
  4. The S-CSCF receives the request and installs that subscription, i.e. the S-CSCF takes the role of a notifier. The S-CSCF sends a Notify request to the user. This request includes Public User Identities and the registration state.

In case the S-CSCF has to shutdown or there is some other stimulus the S-CSCF will inform the user (and the P-CSCF) of the event.

Call

IMS multimedia calls sample –

  1. voice call originates from user A and enters the IMS network X at the P-CSCF
  2. P-CSCF passes the call to the S-CSCF
  3. S-CSCF interrogates the Application Server for originating services
  4. S-CSCF forwards the call to the I-CSCF of network Y.
  5. I-CSCF interrogates the HSS to determine the S-CSCF and passes the call to it.
  6. S-CSCF interrogates the Application Server for terminating services.
  7. S-CSCF passes the call to the P-CSCF assigned for the user and the voice call is completed.
  8. A video call is set up from User B to User A and the signaling path is reversed.

Finally, User A sets up a data call to User B using the same signaling path.

ims6

IP Multimedia Subsystem (IMS)

  • Why IMS ?
  • What benefits does IMS bring ?
  • Features of IMS Network
  • IMS Layers
    1. Transport / Media Endpoint Layer
      • Backhaul network
      • Border Gateways
    2. Session & Control Layer
      • HSS (Home Subscriber Server)
      • SCF (Call Session Control Function)
      • MGw (Media Gateway Control Function)
    3. Application Services Layer
      • TAS (Telephony Application Server)
      • IM-SSF ( IP Multimedia Services Switching Function)
      • OSA-GW (Open Service Access Gateway)
  • IMS-architecture
    • IMS standalone architecture
    • Interoperable IMS core for heterogeneous access networks

IMS is a an architectural framework for IP based multimedia rich communications. It was standardized by a group called 3GPP formed in 1999. It started as an enabler for 3rd generation mobile networks in European market and later spread to wirelne networks too. IMS became the key to Fixed Mobile Convergence (FMC).

Based on IETF Protocols (such as SIP, RTP, RTSP, COPS, DIAMETER, etc), IMS is now crucial for controlling conmmunication in a IP based Next Genration Network (NGN).

Communication service providers and telecom operators are migrating from circuit-switched networks to IMS technology with the increasing bandwidth (5G) and user expectations.

Why IMS ?

Early days TDM networks were not robust enough to support emerging technologies and data networking. There was a need to migrate from voic eonly network to Triple play network ( voice , video and data ). Other factors included :

  • rapid service development
  • service availiability in both home and roaming network
  • wireline and wireless convergence

Due to these above mentioned reasons TDM was outdated and IMS gained support .

What benefits does IMS bring ?

It offers counteless applications around rich multimedia services on wireless , packet swtched and even tradional circuit switched networks.

Easier to Create and Deploy New Applications and Services

  • (+)Enhanced applications are easier to develop due to open APIs and common network services.
  • (+) Third-party developers can offer their own applications and use common network services, sharing profits with minimal risk
    New services involving concurrent sessions of multimedia (voice, video, and data) during the same call are now possible.
  • (+) Reduced time-to-market for new services is possible because service providers are not tied to the timescales and functions of their primary NEPs

Capture New Subscribers,Retain Current Subscribers

  • (+) Better voice quality for business applications, such as conferencing, is possible
  • (+) Wireless applications (like SMS, and so on) can be offered to wire line or broadband subscribers.
  • (+) Service providers can more easily offer bundled services.

Lower Operating and Capital Costs

  • (+) Cost-effective implementation of services across multiple transports, such as Push-To-Talk (PTT), presence and Location-Based Services (LBS), Fixed-Mobile Convergence (FMC), mobile video services, and so on.
  • (+) Common provisioning, management and billing systems are supported for all networks.
  • (+) Significantly lower transport costs result when moving from time-switched to packet-switched channels.
  • (+) Service providers can take advantage of competitive offerings from multiple NEPs for most network elements.
  • (+) Reduced expenses for delivering licensed content to subscribers of different types of devices, encodings, or networks.

The strongest argument for adoption of IMS is that it follows established standards and open interfaces from 3GPP and ETSI. This makes it suited for interoperability, policy control accross networks, streamlined OSS/BSS, Value Added Services etc.

ims

Features of IMS network

  1. Abstraction from Underlying Network : IMS is essentially leading towards an open and standardized network and interface,irrespective of underlay network.
  1. Fixed /Mobile Convergence : Inter operability with Circuit Switched (CS) Mobile application Part (MAP)
  1. Roaming : Location awareness between home and visiting network.
  1. Application layer Call Control : IMS application layer has the provision for defining proxy or B2BUA based call flow completion . This leads to operator being able to introduce business logic into call sessions.

IMS is supplemented by SIP (IETF), Diameter (IETF) and H248(ITU-T).The release cycle of IMS is as follows

  • 2002-03-14 Rel-5  : IMS was introduced with SIP. Qos voice over MGW.
  • 2004-12-16 Rel-6 : Services like emergency , voice call continuity , IPCAN ( IP connectivity Access Network )
  • 2005-09-28 Rel-7 : Single Radio Voice Call Continuity , multimedia telephony,eCall ,ICS
  • 2008-12-11 Rel-8 : IMS centralized services , supplementary services and internetworking between  IMS and  Circuit Switched Networks,charging , QoS
  • 2009-12-10 Rel-9 : IMS emergency numbers on GPRS , EPS(Enhanced packet system) , Custom alert tone , MM broadcast/Multicast
  • 2011-3-23 Rel-10 : home NodeB, M2M, Roaming and Inter UE transfer
  • 2012-09-12 Rel-11 :-tbd
  • 2014-09-17 Rel-12 :- tbd
  • 2015-12-11 Rel-13 :- tbd

IMS Layers

Majorly IMS is divided into 3 horizontal layers given below :

2014-05-24_0015

Transport / Media Endpoint Layer

Unifies transports and media from analog, digital, or broadband formats to Real-time Transport Protocol (RTP) and SIP protocols. This is accomplished by media gateways and signaling gateways.

It also includes media servers with media processing elements to allow for announcements, in-band signaling, and conferencing. These media servers are shared across all applications (voicemail, interactive response systems, push-to-talk, and so on), maximizing statistical use of the equipment and creating a common base of media services without “hard-coding” these services into the applications.

Session & Control Layer

This layer arranges logical connections between various other network elements. It provides registration of end-points, routing of SIP messages, and overall coordination of media and signaling resources.

IMS core which is part of this layer primarily contains 2 important elements Call Session Control Function (CSCF) and Home Subscriber Server (HSS) database. These are explained below 

HSS ( Home Subscriber Server)

It is a database of user profiles and location information . It is responsible for name/address resolution and also authorization/authentication .

CSCF ( Call Session Control Function)

Handles most routing, session and security related operation for SIP messages . It is further divided into 3 parts :

  • Proxy CSCF: P_CSCF is the first point of contact from any SIP UA. It proxies UE requests to subsystem.
  • Serving CSCF: S-CSCF is a powerful part of IMS Core as it decides how UE request will be forwarded to the application servers.
  • Interrogating CSCF: I-CSCF initiates the assignment of a user to an S-CSCF (by querying the HSS) during registration.

Application Services Layer

The Application Services Layer contains multiple Application Servers (AS), such as:

  • Telephony Application Server (TAS) – for defining custom call flow logic
  • IP Multimedia Services Switching Function (IM-SSF)
  • Open Service Access Gateway (OSA-GW), and so on.

IMS Architecture

The IMS standalone architecture is suited for an all IP network

The IMS standalone archietcture is suited for an all IP network

Interoperable IMS core for heterogeneous access networks

Interoperable IMS core for heterogeneous access networks

References

  • IMS service Switching function and reverse service Switching function read here.
IMSSF and RIMSSF
IP Multimedia Subsystem (IMS) – detailed part2
Internet Telephony Convergence- JAINSLEE Platform
IMS in EPC ( Evolved Packet Core )

Update on IMS :

IMS has been mandated as the control architecture for Voice over LTE (VoLTE) networks. Also IMS is being widely adopted to mange traffic for Voice over WiFi (VoWiFi) systems.

IMS in EPC ( Evolved Packet Core )

Packet Switched and/or Circuit swicthed Communication

The earlier models were distributed between legacy circuit switched networks and evolving packet switched networks

Sources : NEC eNB SGW E-UTRAN PDG MME

With the massive improvents in quality of network srevices packet switched comunication protocls became more resilent and replaced the circuit swicthed protcols for realtime communication.

LTE ( Long Term Evolution )

LTE evolved its precursor Universal Mobile Telecommunication System (UMTS), which in turn evolved from the Global System for Mobile Communications (GSM).

It defines the access layer of Telecom architecture. EPC is the core of LTE system . LTE is often linked with evolved UMTS terrestrial radio access (E-UTRA) and evolved UMTS terrestrial radio access network (E-UTRAN). With this eco-system Evolved NodeB (eNodeB) is the base station for LTE radio.

Read more on Long Term Evolution (LTE), VOLTE and VOWifi 

What is EPC (Evolved Packet Core )  ?

EPC is a core network architecture framework by 3GPP. It conncets the E-UTRAN to server-PDNs and is largely responsible in controlling the application flow.

 Primarily EPC has 4 sub parts

  1. HSS
  2. Serving Gateway ( S-Gw)
  3. PDN gateway ( P-Gw)
  4. MME (Mobility Management Entity)

It is most often an IMS environment contgainings CSCF as Gateways with various roles .

IP Multimedia SubSystem was originally meant for evolved UMTS network to provider IP communication to mobile UAs. Today IMS is gaining plenty of attention due to oncoming

More information on IMS and IP Communication go here 

What is IMS or IP Multimedia System ?

A standardized IP-based architecture that allows the convergence of fixed and mobile communication devices and multimedia applications.  

ims arc1

Using IMS, applications can combine voice, text, pictures, and video in call sessions, offering significant ease-of-use to subscribers and allowing service providers to drive branding through a common interface  

Defined by the Third Generation Partnership Project  (3GPP) and supported by major Network Equipment Providers and service providers.  

The standard supports multiple access types – including GSM/GPRS/EDGE, WCDMA, CDMA2000, wireline broadband access and wireless LAN.

Unified Service Delivery Stack

The Unified Communication Solution leads to Network Agnostic, Agile, Cost Effective  & Customer Experience Centric Services Platform.

unified ccommunication

 

The Way from Copper -> Fiber -> 2nd Generation -> 3rd Generation -> LTE , depicts evolution of Telecommunications over the decades , in the Network layer Infrastructure area

The Sevice Layer Infrastructure is built  using techniques as Switching , Home Location Register (HLR) ,  Authetication (  AuC) etc  . The Services vary over ranges such as IN , Voice , SMS , VOIP , IM , IPTV , IMS  , Presence , MMS etc .

Top of this lies the Harmonization layer that performs the inter networking between different platforms and protocols .

The Application Layer consists of various usecases as Enhanced Screening , Social Networking Integration , Education Trade etc .

IMS , the revolution ahead

vision :
To make a model that separates the services offered by
fixed-line (traditional telcos),                mobile (traditional cellular),            and            converged service providers (cable companies and others who provide triple-play — voice,  video, and data — services) from the access networks used to receive those services.
———————————————
Layers :
 IMS architecture is broken into distinct layers:
Screenshot from 2013-05-16 18:37:09
———————————————————
Drivers :
Revenue streams for plain vanilla voice services are sharply falling and the need of the hour is to propose smart intuitive and creative service to kep up the Telecom market alive .

Telecommunications convergence – VoIP, PBX and IMS

There has been rapid evolution of telecom platform over the last few decades. Starting from the the mobile phone network-enabled universal communication agnostic to actual location to present day high bandwidth high data rate entertainment/ streaming like applications. The affordable, personal communication system has converged to enterperise level secure communication systems that cater to low latency and highly secure end to end encrypted scenarios.

IP communication

Some of the positive aspects of using IP communication over traditional communication systems are :

Higher ROI( Return of investment)

ROI is a big factor for SME before making the switch to IP telephony inplace of traditional established system like landline phone and cables. However it is for a fact that once the VOIP comm system is setup , it most certainly reduces call costs by 70%.

Third party Interations

It is often a necessaity to integrate communication system with CRM (content realationship management ) systems or Sales management systems or other lead gtracking systesm which are driven from communications with possible clients or investors ( called leads). Since most web portals are on IP protocl as HTTP, VOIP fits very well, with the click to call on webpage itself among other features such as directory integration , notofoication , call scripts etc.

VAS ( Value Added Service)

Value Added Services , refer to services build on top of existing underlying mobile communication call and sms. These could be innovation usecases build using -IVR / DTMF such as cricket score, astrology updates or call recoring , find-me-follow-me applicatoion for multiple devices , voicemail/ visual voice mail , re-routing to home phone or assiatnt phone, called ID etc. In short it can add intelligence to the way calls are managed .

As bandwidth has increased, so has the proliferation of VoIP systems. From the user’s perspective, modern mobile devices deliver the converged, multi-media communication and entertainment experience.

VOIP

VOIP , short for Voice over IP , is called so beacuse it not only converts your voice calls in analog voice into digital packets but also channels voice data through IP networks such as LAN , WAN , Internet etc using the Internet Protocol (IP) .

  • VOIP system on LAN ( Local Area Network ) can use it as its backbone system to establish communication between endpoints . For example : Office communication system within the same enterprise/building.
  • Similarity  VOIP over WAN ( Wide Area Network ) use the help  of IP PBX and VoIP service provider to enable communication across Internet . For example : OTT providers and internet calls.
  • By using the services of telecom providers in support with above plan it is also possible to land a VOIP call onto a real phone over GSM / PSTN via gateways.

For a provider of IP telephony system, number of factors come into picture such as :

  1. Bandwidth : Low bandwidth has always been a big concern for IP calls especially due to packet loss and thus high noise. While a LAN connection ensures good experience, calls over internet or VOIP PBX are not necessarily as neat. Network switching between different Internet service providers causes congestion and lags too.
  2. Inter-operability : Connecting remote works / employees to the VOIP network requires interoperablity between their hand held device like android , ios , tablets , smart watch or other types od communication devices such as hardphone, desktop-systems , kiosk , surveillance cams etc is a challenging considering the underlying OS and networking support.
  3. Traffic: Maximum simultaneous call or peak traffic rate can create bottlenecks in communication channel or worse still result in high bandwidth usage. For example as p2p conf call between 5 parties will create a mesh network between each participant resulting in 4 outgoing and 4 incoming channels.
  4. QoS (Quality of service ) :
    • Call drops ,
    • prioritization of important calls ,
    • Security preventing the attacks and hacks ,
    • keeping information secure by encryption end to end
  5. AAA : managing Authentication, Authorization and accounting
  6. Reuse existing Hardware :
    • Replacing old hardware or installing softphone apps on mobiles etc .
    • Reuse old servers . Manage setup between datacentres and cloud deployments
    • Administravtive hurdles between different counteries and geographies for using hardware
  7. Scaling
    • How quickly can it scale up or scale down ?
    • Will the communication system grow horizontly or vertically ?
    • How to ensure that the growing system can accommodate new users , physical office location , remote centers , call centres etc ?
  8. Codecs : Under low bandwidth condition it is a good idea to switch to low resolution ( in case of video ) and low bandwidth codec ( in case of audio ) .

Other factors such as privacy , accounability , Lawful interception ( legal requirnments in many enterprises ) , Auditing , SLA ( Service Level Agreements) to ensure the system is up 99.99 % of time and agrreeing to pay compensation if system is down for longer duration than 0.01 % of time so on.

Hosting the PBX

Unified communication Solutions as SaaS or IaaS refer to on-premise or cloud-hosted IP PBX Solutions. Comparison of both is as follows

On -premiseCloud Based
The solution is usually of the SaaS nature ( software as a service ) which is hosted by the consumer / business unit itself . The service provider offers his infrastructure to the consumer as a service and bills monthly / yearly etc .
Hosting the solution system on premise and setting up the infrastructure means more customization and flexibility but it also means more investment and maintenance . On the other hand hosting the solution on cloud is often a quick setup with relatively lower upfront payment. The billing is either carried out per per user basis or based on consumption . The data is synced to cloud servers for storage and can be fetched from there when required such as cloud synced Call-logs or contact-book .

Convergence Vision 

We already know some of the latest trends of industry with respect to telecom convergence such as :

FMC

Fixed Mobile Convergence (FMC) stands for integrating user’s fixed desk phone with his mobile phone. Call continuity is a VAS( Value added service ) which lets him to switch calls between different call devices even softphones , mid call also. It has multi-faced advantages such as not missing any call on account of being out of office , having the same call preferences on each device such as blocked numbers , IVR settings etc .

UC

Unified Communication(UC) refers to the accessibility of all communication and collaboration services from the users call agent ( phone / soft-phone ) . These services can include file transfer , chat , conference , call settings , blocking , white-listing , fax , cloud sync , call logs , called ID , favorites , recording .
Read more about Unified communication and collaboration here .

BYOD

Bring your own device (BYOD) is one of the hottest trends in industry almost across all domains where user is expected or is given to option to bring his personal laptop for official use . It is the responsibility of enterprise comm system to seamlessly integrate it with in-office communication system and provide the same privileges and security to business critical applications as preset in configuration settings . It increases the flexibility and productivity while keeping the infrastructure cost down.

IMS provided Network Interoperability and Access Independence

ims-access-network-independence
Image Source unknown. Represents the convergence of IMS subsystem with various access types

IMS based tele-coommunication convergence described in figure below

  • clients get direct connectivity to IP PBX in offices or hotels
  • home users connect through cable wires or Wifi/WiMax
  • non SIP based legacy endpoints connect via signalling and media gateways

The access endpoints connecte to a single managed core IP network which intercoonectes with IMS core . The back end system not only manages calls and sessions but also registration  ,  billing , operations and adminstartion.

IMS convergence vision
picture courtesy – unknown

 Intelligent Network   —>    Next Generation IMS System 

The signalling protocols migration like from signalling system 7 (SS7) to session initial protocol (SIP) have been taking place in Telco-Industry. Similarly nodes of legacy network like signal transfer point (STP) of legacy network are being migrated to call session control function (CSCF) of IMS  that allows the rapid development and deployment of enhanced, revenue-generating multimedia services for fixed, mobile and cable operators.

IMS architecture enables operators to seamlessly run a plethora of next-generation converged services over their fixed, mobile and cable networks, achieve a faster time-to-market for new services and have fewer performance bottlenecks.

converged telecommunications

Business benefits of IMS 

  1. Delivering Services: Delivering services and applications on a “wherever, however, whenever” basis.
  2. Multimedia services: Enabling service providers to offer multimedia services across both next-gen, packet-switched networks and traditional circuit-switched networks.
  3. Protocol stack: IMS architecture provides pipes and protocols onto which service providers can attach no. of applications very conveniently.
  4. Open Source standard: IMS architecture is based on open standard which makes it possible for different vendors of hardware and software to integrate with each other seamlessly.

As a subscriber, one of the main benefits of the IMS architecture is the capacity of the network to deliver the same set of services whatever the access network used.

convergence

This is made possible thanks to the centralization of the service execution process. A specific call server of the control plan (called Serving Call Session Control Function, S-CSCF) is responsible for invoking the application servers based on criteria provisioned in the central database. The S-CSCF gets these criteria (called Initial Filter Criteria) during the user’s registration in the IMS network.

Circuit Switched Voice –> Packet based VOIP 

Voice over IP revolutionized in the Telecommunication space.It also makes your communication experience much richer and nicer with a series of enhanced features and extended possibilities. The no. of user migrating from traditional circuit switched network to IP has been quite substantial in recent years. CSP are embracing VOIP technology as a potential revenue generator and investing huge chunk of money to create value propositions for themselves in VOIP.

In conclusion here are the top business benefits of adopting a converged and unified IP telephony solution such as IMS and SIP are

  • Cost Savings : Saving money is the number-one reason most businesses and households make the switch to a VoIP system, VoIP systems don’t require a phone cabinet or on-site routing equipment- just phones.
  • Features: VoIP also allows users to take advantage of advanced features only available on internet-based phone systems. Features like online call monitoring, and online phone system access to add or configure extensions are also available with VoIP systems.
  • Flexibility: VoIP allows people to go mobile and call directly from their cell phone and be charged at low VoIP rates
  • Tracking Options: Since VoIP is an internet-based system, user can track and manage their system from their computer. Most VoIP systems allow user to track call volume and call time fairly easily- a feature that can be especially helpful for businesses that bill clients hourly or for time spent on the phone.