Regulatory and Legal Considerations with WebRTC development

This post is deals with some less known real world implication of developing and integrating WebRTC with telecom service providers network and bring the solution in action .The  regulatory and legal constrains are bought to light after the product is in action and are mostly result of short nearsightedness .  The following is a list of factors that must be kept in mind while webRTC solution development .

  • WebRTC services from telecom provider depend on the access technology, which may differ if the user accessing the network through a third party Wi-Fi hotspot.
  • —User/network type may also dictate if decryption of the media is possible/required.
  • —For Peer-to-Peer paths, media could be extracted through the use of network probes or other methodology

—Then there are Other Considerations such as specific services, for example if WebRTC is used to create softphones software permitting users to receive or originate calls to the PSTN, the current view is to treat this as a fully interconnected VoIP service subject to all the rules that apply to the PSTN – regardless of technologies employed.

CALEA

Communications Assistance for Law Enforcement Act (CALEA) , a  United States wiretapping law passed in 1994, during the presidency of Bill Clinton.

  • —CALEA requirement for an LTE user may be very different than the CALEA requirements for a user accessing the network through a third party Wi-Fi hotspot.
  • For media going through the SBC, CALEA may use a design similar to existing CALEA designs.
calea intercept infrstructure

calea intercept infrstructure

SIP Presence

We have already learned about Sip user agent and sip network server. SIP clients initiates a call and SIP server routes the call . Registrar is responsible for name resolution and user location. Sip proxy receives calls and send it to its destination or next hop.

Presence is user’s reachability and willingness to communicate its current status information . User subscribe to an event and receive notification . The components in presence are :

Presence user agentpresence components
Presence agent
Presence server
Watcher

Image source  : http://msdn.microsoft.com/en-us/library/bb896003.aspx

Sip was initially introduced as a signaling protocol but there were Lack of method to emulate constant communication and update status between entity
Three more method was introduced namely – Publish , Subscribe and Notify

Subscribe request should be send by watchers to presence server
Presence agent should authenticate and send acknowledgement
State changes should be notified to subscriber
Agents should be able to allow or terminate subscription

presence flow

Image source http://download.oracle.com/docs/cd/B32110_01/ocms.1013/b31497/about_sdp.htm#BABDHHCJ

Traces of various SIP requetss and response in presence are are follows :

subscribe request

SUBSCRIBE sip:presentity@example.com SIP/2.0
      Via: SIP/2.0/UDP host.example.com;branch=z9hG4bKnashds7
      To: <sip:presentity@example.com>
      From: <sip:watcher@example.com>;tag=12341234
      Call-ID: 12345678@host.example.com
      CSeq: 1 SUBSCRIBE
      Max-Forwards: 70
      Expires: 3600
      Event: presence
      Contact: sip:user@host.example.com
      Content-Length: 0
 

200 OK to subscribe request

SIP/2.0 200 OK
      Via: SIP/2.0/UDP host.example.com;branch=z9hG4bKnashds7
       ;received=192.0.2.1
      To: <sip:presentity@example.com>;tag=abcd1234
      From: <sip:watcher@example.com>;tag=12341234
      Call-ID: 12345678@host.example.com
      CSeq: 1 SUBSCRIBE
      Contact: sip:pa.example.com
      Expires: 3600
      Content-Length: 0
 

Notify Request

NOTIFY sip:user@host.example.com SIP/2.0
      Via: SIP/2.0/UDP pa.example.com;branch=z9hG4bK8sdf2
      To: <sip:watcher@example.com>;tag=12341234
      From: <sip:presentity@example.com>;tag=abcd1234
      Call-ID: 12345678@host.example.com
      CSeq: 1 NOTIFY
      Max-Forwards: 70
      Event: presence
      Subscription-State: active; expires=3599
      Contact: sip:pa.example.com
      Content-Type: application/pidf+xml
      Content-Length: …
 

200 OK success response to notify

SIP/2.0 200 OK
      Via: SIP/2.0/UDP pa.example.com;branch=z9hG4bK8sdf2
       ;received=192.0.2.2
      To: <sip:watcher@example.com>;tag=12341234
      From: <sip:presentity@example.com>;tag=abcd1234
      Call-ID: 12345678@host.example.com
      CSeq: 1 NOTIFY
 

PUBLISH Request

PUBLISH sip:presentity@example.com SIP/2.0
Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
To: <sip:presentity@example.com>
From: <sip:presentity@example.com>;tag=1234wxyz
Call-ID: 81818181@pua.example.com
CSeq: 1 PUBLISH
Max-Forwards: 70
Expires: 3600
Event: presence
Content-Type: application/pidf+xml
Content-Length: …

200 OK success response to PUBLISH

SIP/2.0 200 OK
Via: SIP/2.0/UDP pua.example.com;branch=z9hG4bK652hsge
;received=192.0.2.3
To: <sip:presentity@example.com>;tag=1a2b3c4d
From: <sip:presentity@example.com>;tag=1234wxyz
Call-ID: 81818181@pua.example.com
CSeq: 1 PUBLISH
SIP-ETag: dx200xyz
Expires: 1800

A call flow depicting presence in action is as given below :

presence subscribe notify

Image source http://www.cisco.com/en/US/i/100001-200000/190001-200000/190001-191000/190463.jpg

security considerations for Presence service include:

  • Access control.
  • Notifier privacy mechanism.
  • Denial of service attacks.
  • Replay Attacks.
  • Man-in-the-middle attacks.
  • Confidentiality.

some solutions for security implementation are

  • Sip registration
    TLS
    Digest Authentication
    S/MIME

References :

Rfc 3856 http://www.ietf.org/rfc/rfc3856.txt
Rfc 3265 http://www.ietf.org/rfc/rfc3265.txt
Rfc 2778 http://www.ietf.org/rfc/rfc2778.txt
Rfc 3261 http://www.ietf.org/rfc/rfc3261.txt
Rfc 3903 http://www.ietf.org/rfc/rfc3903.txt

http://en.wikipedia.org/wiki/Session_Initiation_Protocol

Summary :

Presence is a way to have sustained stateful communication. The SIP User agents can use presence service to know about others user’s online status . Presnece deployment must confirm to security standards .

A legacy telecom network

I use the term legacy telecom system many a times , but have not really described what a legacy system actually is . In my conferences too I am asked to just exactly define a legacy system . Often my clients are surprised to hear what they have in current operation is actually fitted in our own version of definition of ” Legacy system ” . 

This write up is an attempt to describe the legacy landscape . It also describes its characteristics , elements and transformation .

1. Legacy system have ATM / Frame Relay transmission . 

This  is basically Hardware  Specific and results in High Expenses.

2. Legacy systems have POTS / PSTN / ISDN as their access layer technology . 

Access layer is the first layer of telecom architecture which is responsible for interacting directly with the end use / subscriber . Legacy system technologies are again Hardware  Specific , bear High Expenses and offer Low stability.

3. Legacy system use Traditional Switches / ISDN in their Core Layer .

Core layer is the main control hub of the entire telecom architecture . Using old fashioned switches render high CAPEX ( capital Expenditure ) and OPEX ( Operational Expenses ) . 

4. In the service delivery front legacy system employ Traditional IN switches

These are very Hardware Centric.

……………………………………………

 

 

 

 

Challenges in Migration to IMS

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

SIP/VOIP transformation towards IMS (Total IP)

IN to IMs

Migration to all IP telecom

ip transformation in access layer

ip transformation in access layer

 

ip transformation in transport layer

ip transformation in transport layer

 

ip transformation in session layer

ip transformation in session layer

Service Broker Architecture for IN and IMS

We know that Service broker is a service abstraction layer between the network and application layer in  telecom environment.SB( Service Broker ) enables us to make use of existing applications and services from Intelligent Network’s SCP ( Service control Point ) , IMS’s Application Server as well as other sources  in a harmonized manner .

service broker

 

The service provider can  combine the services from various sources written in various languages in numerous permutations and combinations .  This saves the time , energy and rework required to launch a new services. 

I have written couple of posts before on Service Broker .Post on What is Service Broker . It definitions and application can be found here  : http://altanaitelecom.wordpress.com/2013/03/19/service-broker/. This also defines service orchestration and harmonization . 

Another post on Service Borker’s role and function can be found here : http://altanaitelecom.wordpress.com/2013/08/07/service-broker-2/. This mentions the service brokering role in network environment. But ofcourse it was a mere introduction  . The following post clarifies the concept in greater light . 

I believe and it truly is a wonderful thing to make use of Service Broker while network migration from IN to IMS .The following architecture model depict the placement of Service Broker component in IN and IMS integrated environment . 

 

sb1

The figure above portrays how a  service provider acts as a central Node for Services invocation and services composition. SB is responsible for Services Orchestration / Interaction , service development, third party integration and acts like a protocol gateway .

Let us discuss service broker in a full fleshed network’s structure . It includes the access network components and detailed core network components with the name of interfaces between all nodes.

 

 

sb2

 

The Applications as described by the above figure could be majorly of 4 types :

1. applications developed on a SIP application Server and invoked via SIP/ ISC

2. Applications developed over SIP servlets or JAINSLEE platform such as mobicents , Opencloud Rhino etc

3. Application developed on a SCP ( Service Control Point ) of a IN ( Intelligent Network ) . This is invoked via INAP CS1/CS1+ or CAP

4. Application developed on a J2EE server Invocated via http REST API like GSMA OneAPI such as 

  • Call Control API for voice.
    Messaging API for SMS, MMS.
    localisation API.

Provisioning via fixed/mobile brands & « service profile» in SB

Provisioning via fixed/mobile brands & « service profile» in Service Broker

Provisioning via fixed/mobile brands & « service profile» in Service Broker

BDD « Services » in SB

BDD « Services » in ServiceBroker

BDD « Services » in Service Broker

Architecture of SDP / Service Broker

Architecture of SDP / Service Broker

Architecture of SDP / Service Broker

WebRTC compatible android client

This post describes the requirement of creating a SIP phone application on android over the same codecs as WebRTC ( PCMA , PCMU , VP8) . In my project concerning the demonstration of WebRTC inter operability ( presence , audio / video call , message )  with a native android client , I had to develop a lightweight Android SIP application , customized for the look and feel of the webrtc web application . This also enables the added services to WebRTC client such as geolocation , visual voice mail , phonebook , call control options be set from android application as well .

Aim :

Android webrtc- sip client development , using sipml5 stack implemented through web services and native android programming .  

Software Used:

⦁ Eclipse IDE
⦁ Java SE Development Kit 7.0
⦁ Android SDK

Tasks :

⦁ Authorization of a user, based on his/her credentials (Database local to the application).

webrtc_android_2
⦁ Navigation Drawer on the home page which shows a menu giving the user various options like:
⦁ View Home Page
⦁ View Contact List
⦁ View/Edit My Profile
⦁ View My Location
⦁ Sign Out

⦁ Phonebook sync : Importing contact list of the Android Phone into the application. Editing user profile with values like  User Name ,  Password ,  Domain. 

webrtc_android_1
⦁ Inclusion of a Web View in the application which currently opens the desired webpage(http://sipml5.org/call.htm).

⦁ Geolocation: Showing marker for the current location of user in Google Maps.Displaying the address of the user in a Toast Message.

webrtc_android_4

⦁ Audio / Video call capability 

android_webrtc

figure 1 : Login page , figure 2 : Call page , Figure 3 : Menu bar 

Future Roadmap:

⦁ Connecting the application to a database which sits on the cloud.
⦁ Based on the entries in the database the user will be able to:
⦁ Login to the application.
⦁ View or edit his/her details in the My Profile Section.
⦁ Understanding codes of sample applications for making SIP calls from Android OS like:
⦁ SipDroid
⦁ SipDemo
⦁ IMSDroid
⦁ Modifying the existing application to be able to make SIP calls like one of the apps listed above.

Modules :

Development Done:
  1. Development of an authorization page connecting the application to a local database from where values are inserted and retrieved.
  2. Development of navigation drawer where additional options for the application will be displayed making it a user friendly application.
Development Planned:

1.Connectivity to a cloud database.  

2. App engine on cloud.

3. Importing contacts from phone address book .

4. Offine storage of profile details and few call logs .  

Architecture:

webrtc_android_enviornment

……………………………………………………………………………………..

2nd and 3rd generation of telecommunication

Although the history of telecom evolution begins with PSTN and switches we shall oit them as they are truly legacy now .  We have seen the evolution of second to third generation of telecom most recently .  Where 2 G is referred to as the GSM era  , 2.5 G as the GPRS with GSM era . The following two diagram denote the service operators architecture nodes in both these times .

Note that in pure 2G there was only circuit switched communication services .

gsm

The advent 2.5 G bought packet switching for data access along with existing circuit switching for voice network .

gsm_gprs

Note that the processes such as billing etc had begun merging for both the circuit switched and packet switched networks .

However as the mobile became smarted and hungry for faster internet , it bbecame necessary to bring in faster speed and hence was born 3G. . Now 3G was further succeeded by 3.5G ( HSPA – High Speed Downlink Packet Access ) eventually 4G ( LTE Long Term Evolution ) as we can see now but that is another story .

Difference between WebRTC and plugin based communication

A lot of service providers ie telecom operators had deduced their own ways to provide Web based communication even before WebRTC was born . With time , as WebRTC has become stronger , more secure , resilient to failure they have come around to migrate their existing system from previous closed box native APIs to opensource WebRTC APIs.

The first figure ( given below ) depicts a communication platform build over plugins and proprietary APIs using HTTP REST based signaling .

2014-07-22_1212

Web Communication Service Architecture over HTTP/ REST API

As the migration took place the proprietary API components were replaced by Open standard based entities such as plugins were replaced by WebRTC APIs, HTTP REST based signalling was replaced by SIP ( Session Initiation Protocol ) .

Web Communication Service Architecture over WebRTC SIP

Web Communication Service Architecture over WebRTC SIP

Note telecom operator network did not had to face transformation by integration of WebRTC elements .

Interoperability between WebRTC , SIP phones and others

WebRTC SIP clients

What is the role of SIP server ?

 SIP Server convert the SIP transport from WebSocket protocol to UDP, TCP or TLS which are supported by all legacy networks. It also facilitates the use of rich serves such as phonebook synchronisation , file sharing , oauth in client .

How does WebRTC Solution traverse through FireWalls ?

NAT traversal across Firewalls is achieved via TURN/STUN through ICE candidates gathering .Current ice_servers are : stun:stun.l.google.com:19302 and  turn:user@numb.viagenie.ca

What audio and video codecs are supported by WebRTC client side alone ?

Without the role of Media Server WebRTC solution supports Opus , PCMA , PCMU for audio and VP8 for video call.

RTCBreaker if enabled provides a third party B2BUA agent that performs certain level of codec conversion to H.264, H.263, Theora or MP4V-ES for non WebRTC supported agents.

What video resolution is supported by WebRTC solution ?

The browser will try to find the best video size between max and min based on the camera capabilities.

Options are : sqcif | qcif | qvga | cif | hvga | vga | 4cif | svga | 480p | 720p | 16cif | 1080p

We can also predefine the video size such as minWidth, minHeight, maxWidth, maxHeight.

What bandwidth is required to run WebRTC solution ?

We can set maximum audio and video bandwidth to use or use the browser’s ability to set it hy default at runtime . This will change the outgoing SDP to include a “b:AS=” attribute. Browser negotiates the right value using RTCP-REMB and congestion control.

SIPML5 client by dubango

calltakenoffhold

Telestax WebRTC client

2014-06-11_2215

SIPJS with flash network support

windows_IE_1

JSSIP – MIT license 2014-02-09_1444

SIP phones in Ubuntu ( Linux system)

SFL phone

linux sfl 2 linux sfl 1

Yate SIP phone

linux yate 2 linux yate 1

Linphone

ubuntulinphon4 linuxlinphone2

 

Windows Operating system SIP softwares

Xlite is well known SIP softphone for windows dessktop

xlite 1

Xlite new version

windows_xlite_7 windows_xlite_6_001 windows_xlite_6 windows_xlite_3

 

Kapanga SIP softphone . It is also runnable on Linux desktop through windows compatibility softwares like wine

windows_kapanga_3 windows_kapanga_2

FreeSwitch Communicator , comes along with the Freeswitch Media Server .

windows_freeswitchcomm__2 windows_freeswitch_comm_3

Boghe SIP RCS client

windows_boghe_5 windows_boghe_4 windows_boghe_2 windows_boghe_1

Jitsi SIP phone

jitsi 2 jitsi 1

 

MAC SIP software

idoubs desktop SIP RCS client for Mac

Screen shot 2014-06-13 at 4.03.27 PM

iOS SIP phone applications

Linphone

IMG-20140703-WA0003  IMG-20140703-WA0006 IMG-20140703-WA0007  IMG-20140710-WA0001 IMG-20140710-WA0002

Android SIP applications

Sipdroid , opensource

Screenshot_2014-07-01-19-36-47 Screenshot_2014-07-01-19-37-00 Screenshot_2014-07-01-19-37-44 Screenshot_2014-07-01-19-37-54 Screenshot_2014-07-01-19-38-46

 

………………………………………………………………………………………………………………………….

Geolocation with SIP

Aim : 

The application is aimed to optimise the assignments of tasks to field works and engineers , with features such as user location reporting and location tracking of inventory at real-time .

Description :
To map the users on a map ie geo-locate their devices so as to make a visual interpretation of their presence , calls and accessibility .  It will third party Map API’s to provide the underlying map service and use ip address , presence service of SIP and  surrounding access point to track the current location.
Geolocation Services in SIP application Server

Geolocation Services in SIP application Server

Tools Used :
1. Openstreet Maps / Google Map API webservices to display the coordinates on a Map.
2. SIP Location Servers / Registrars
3.HTML5 geolocation : automated location detection using cellular positioning, Wi-Fi positioning, and GPS.
How does it work :
It gathers information from nearby wireless access points and your computer’s IP address,  to get an estimate of your location. That location estimate is then shared with the requesting website.
Usecase : Field Force Management
Tracking the employees/ managers and workers location carrying important cargo .Direct communication in case of navigation or instruction .
sip geolocation and Field force management

sip geolocation and Field force management

Accuracy :
May be able to provide a location to within a few meters. However, in other areas it might be much more than that. All locations returned by  are estimates only.
Direct Implications :
1. The thin html client performs faster .It is applicable to mobile devices, as well as traditional desktop browser applications.
Future :
The ideas could be enhanced to track user’s location history and make informed guess about the services they might be interested in
Image

PSTN/2G/3G/4G to IMS – Internet Telephony Converged Platform

 

3g 4g ims

3g 4g ims

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

 

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 has to work with multiple Communication Service 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.

SIP Application in IMS

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. This JAIN SLEE framework enables development of network agnostic applications. This is implemented through resource adaptors for deploying same applications over different networks like SIP/IN etc. This 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.

Open cloud architecture

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)

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 this new technology 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

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 centricity to customer centricity. 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. 

……………………………………………………………………………………………………………

Business Challenges for a telecom service provider

With the fast pace of telecom evolution both towards the access network front ( ie GSM , UMTS , 3G , 4G , LTE , VOLTE ) to core network side ( ie application servers , registrar , proxies , gateway , media server etc ) a CSP ( copntent service provider ) is trying hard to keep up with the user expectation . The user expects a plethora of services , reduced cost and high speed bandwidth . If this was not enough a CSP also has competition  OTT (   Over The Top ) Players who provide communication and messaging for FREE .

Technology Evolution Challenges

  • • The increased data speeds and further more increasing hunger for the data overwhelms the existing network infrastructures.
  • • Ensure uniform service experience across the network technologies to check the customer churn.
  • • Access / Radio Technology independent delivery of services.
  • • Enhance Reuse for exiting investments.

Multiple Service Platform Challenges

  • • Typical network constitutes of Multiple Service Platforms increasing network complexity and integration challenges many fold.
  • • Heterogeneous multiple SDP Solutions typically deployed to cater to Multiple Types of Networks/ Standards/Variants
  • • Service Islands makes introduction of seamless services a challenging task for the CSP

Transport Upgrade and Convergence of Wireless Wireline

  • • Retain investments in copper wire systems while migrating towards next generation Fiber Optic systems.
  • • Severe competition among wire-line and wireless operators to provide latest services to retain subscriber base.
  • • Fixed Mobile Convergence leading to a diminishing gap among the revenue shares of various operators in the space, and leading to losses for wire-line only players.

IMSSF and RIMSSF

 

  • 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 RIMSSF  ?

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