Monthly Archives: January 2018

Business and Telecommunication

Business requirement of telephony system, perceived from three dimensions, namely hardware, software and as a service.

Telecom sector business trends and changes happening in each generation of telephony. Analysis of next generation telecom business solutions and possible market areas for small and middle business companies.

In its life journey, telephony systems have traveled from plain old legacy systems to an era of advanced stage of Voice over IP communications supported by high-speed internet connection. Telecommunication feature requirement or demand has now moved to such a diversified stage that the business demand-supply chain keeps changing at a very fast rate. Let me to explain this.

As an example: A requirement or demand of a telephony feature like video conferencing is now no longer dependent on a particular way of implementation i.e. not only limited to hardware solution within an enterprise but have also available on mobile as an application software. It means that business solution of ‘video conferencing’ can be achieved via many dimensions like free soft-phone based, license soft-phone based, hardware based, Mobile/laptop based, or even home decor Television based.

This implies that telephony business must have dynamic business model and not sticking to particular solution. Business model must be capable to switch to alternative solutions at any moment. For big players this is not an issue of any concern, but small business players, must pick wisely what kind of telephony system they will use and what their sustainability is. They must think of moving their on-going business with telecom generations of high-speed internet and digitization.

In legacy telephony, business was oriented towards cost of each audio phone call, number of calls, and Local vs STD based calls, as this ware the core services available. Videos call at that time ware among major earning business solutions.

As a generation of telephony moves on, until 2.5 G to 3G, audio and video phone call is like default services and internet based, location based, on-demand or live streaming was on a high demand. At this stage, new business players who forecast this as an era of stating point of next generation telephony service were able to make a lot of money. Many entrepreneur and many small telecom businesses have grown up to support this demand both with hardware, software and as service.

Many software solutions and telephony applications were developed, protocols were defined and universal standards were placed. Businesses grew with software application around audio call, video call, conference, 3-way calling feature, announcement, advertisement service during telephony services. Hardware based solutions also grew in parallel and High-Definition quality were integrated with above listed telephony services. Small business which were software based have grown up and earned much, whereas big players have taken up this as a complete hardware and software solution, by building up of their own networks. Some business also grew targeting services of telephony application, whereas some targeted the support of telephony service, like Analysis and Analytical tools, testing tools etc. Legacy business was on backward compatibility support of these new telephony service.

At present telephony demand has changed a lot in its demand and implementation. From fixed hardware to mobile based app solutions. Business players have also re-shuffled and realigned themselves. Telecom operators are moving ahead to meet the demand with high speed internet solutions with legacy hardly contributing to a small percentage of profit. On other hand telecom R&D organisations / MNC have started working on next generation technology with high skilled human resources. Business players oriented towards software based adhoc service providers are also growing. Businesses are investing more on mobile based telephony solution like whatsapp, facebook, etc . A bulk of small business earning are also centralised around information on public platform and on advertisement. Telecoms are now moving towards IOT (internet of things) and wifi solutions. Hardware independent, time independent, resource independent fast and easy communication is needed with rich functions and high definition quality.

Small business players interested in telecom business must consider these changes in trends of telephony application demand and supply. Changes from fixed dialling system to mobile based automatic voice-recognition based high-speed internet service, must be analysed for any business investments. Legacy telephony in future is not going to last for couple of more years. SS7/ISDN based telephony system is going to vanish from market being replaced by high-speed and wifi solution.

Eventually telecom market is moving towards virtualization system capable fast and easy expansion of resource with change in demand, Big data and their analysis, single click software application supporting telephony functionality, hardware and location independent application and location-time applications. Those small business working on core legacy technology should try an attempt to move towards high-speed internet based telephony feature. Those small business working on hardware based legacy telephony should try an attempt to move towards digitization and integration within next generation hardware systems. Those small business working on services of legacy telephony should try an attempt to move towards adhoc or license free solution with telecom operator. Small business players interested in telecom business as a solution can work on providing mobile based soft-phone solution as a client solution. As side service they can provide VoIP data analytics, new VoIP features or Wifi solutions for middle business players. IOT – internet of thing is another major evolving in telecom sector, which small application and sensor based device communication are in trial, small and middle business sectors can also think towards this dimension of telecom solution. Telecom sector is vast and it depend so each business player’s vision and mission to work upon.

VOIP explained for enthusiasts

VoIP in depth:

 

  1. Definition & Overview
  2. Protocols & procedure
  3. Technical Details
  4. Features & Functionality
  5. Future Prospects

 

Definition & Overview:

The term VoIP represents Voice communication over/via IP-network. There are many advantages of having voice communication over IP network, termed as PS ( packet switching), over traditionally voice communication, over CS( Circuit switching), like fast, easy and efficient resource utilization with enhanced features. People who wish to communicate with each other with VoIP must support TCP/IP packing, called IP-nodes/Host. VoIP is also used to establish sessions/calls to different networks like circuit switches PSTN or GSM/UMTS users.

It is a technique of sending voice/media packets over IP-network. This solution of establishing session between two parties is adopted by next generation mobile communication like 4G. VoIP is incorporated with underline protocol stack like SIP, H.323, SDP, TCP/IP suit, etc. VoIP solutions are mainly driven with the help of supporting APPLICATION layer protocol, namely SIP (Session Initiation protocol) or H.323.

Protocols & procedure:

Predominantly SIP is a versatile text based protocol mostly used in enhance technology. SIP defines request and response mechanism like INVITE, Register, Option, Publish etc and 1xx, 2xx, 3xx, 4xx, 5xx, 6xx responses respectively.  It also defines routing mechanisms with the help of SIP-proxy server to discover called user details. A session is established between two nodes (IP-based) who wish to communicate by routing request to proxy server and querying database termed location server. Once two nodes (calling party and called party) learn each other IP-addresses, voice RTP packets are sent between them. SIP stacks maintains and control the established the voip session until it is closed. Session is released or cleared by sending BYE request to other party. SIP protocol standard is defined by IETF standards of 3261rfc and is being adopted by many new upcoming technologies. H.323 protocol on other hand is a protocol suit used earlier for VoIP solution. It is still used within certain old technology and enterprise solutions.

Technical Details:

Architecture:

This section describes VoIP architecture and solution in details based on SIP protocol. VoIP architecture is defined based on three tier model.
• User-plan or Application layer mainly consists of IP-nodes who are either Calling User/Nodes or Called User/Nodes. SIP based IP-nodes are termed as UA (User Agent). SIP based IP-Node that initiates session request is termed as UAC (User Agent Client) and SIP based IP-nodes who respond to the request is termed as UAS (User Agent Server).
• Middle tier is a Server-plan or a Routing plane which help in routing SIP based request to the destination and fetching relevant data. SIP based server like SIP-proxy server, Promedia Server, Gatekeeper (for H.323 protocol) is grouped in this tier. Promedia servers are used for establishing session between SIP and H.323 based UAs.
• Database tier of the VoIP architecture consists of servers who maintains data regarding User agents, session billing and session authentication and other. Location server, Registrar server, Authentication server, Billing Server, IVR server are such example servers of this layer. The protocol which is used between the SIP proxy server and the database layer server is RADIUS, which is mainly transaction based protocol.

Procedure of VoIP Session:

SIP session/signalling are done using either TCP or UDP as a transport layer, with TCP as a recommendation to achieve fast session establishment.
• SIP based IP-node which is either a calling party or called party, is configured with IP-address and can be located anywhere physically and connected through internet.
• Called Party must be registered to the VoIP network via preconfigured REGISTRAR Server IP-address and Proxy server
• Calling Party has to be VoIP registered in order to initiate a call to the called user, it must also be configured with a Proxy server
• Called Party registers itself by sending REGISTER request to the Proxy Server. The Proxy server in turn updates the REGISTRAR server with Called Party IP-address, and/or domain and/or MSIDN/sip-URI. SIP-URI is nothing but a Unique Resource Identifier in sip format like ‘sip:[email protected]’. This association created in REGISTRAR server is uploaded into Location Server which is called as AOR – Address of Record. In most cases both REGISTRAR server and LOCATION server can be colocated into one box.
• Calling Party that wish to establish a communication with called user, sends INVITE request to the Proxy or Out-bound Proxy server with called party MSISDN or SIP-URI. Outbound Proxy server is nothing but a proxy server, configured separately for handling calls/session, which is again a deployment specific. Here Calling User will act as a UAC- User agent client and Proxy server acts as UAS – User agent server.
• Out-bound Proxy server receives INVITE request, response with 100 Trying response to the Calling user/UAC acknowledging Calling user that INVITE request is received and is being processed so it does not send multiple INIVITE request.  The proxy server fetches called user AOR from the location server and receives the called user IP-address.
• Proxy server routes the INVITE request to the called user IP-address received in AOR from the location server.
• Called Party upon receiving INVITE request, will to start ring with tone, notifying the user about the incoming call, acknowledging back with 180 ringing response. In the 180 Ringing, called party adds its own IP-address which is traversed backed to the calling party.
• When called party picks the calls, it response with 200 OK.
• Calling party upon receiving 200 OK response send another SIP request ACK, which indicates the flow of voice RTP packet.
• Session can be broken by either party, by sending a BYE request.

RTP and Voice Packets:

Voice packets are being transmitted between calling party and called party directly end to end in particular codec format.  SDP – session description protocol suit is also deployed along with SIP stack which maintains all voice packets stream and their measurement reports. Both RTP and RTCP packets are sent over UDP as a transport protocol to achieve real time streaming. RTCP provides feedback about RTP packets sends in reverse direction.

Features & Functionality:

SIP based protocol leverage to add call features like call hold, Call Park, call conference, both audio and video call, multi user support and many other call functionality. It is easy to deploy and flexible enough to twist to achieve desired result of Voice over IP.

Future Prospects and Deployment:

VoIP solution is key to fourth generation mobile communication supporting multimedia services like audio, video, chat, conference, announcement services. This technology is commonly known as IMS – IP Multimedia Subsystem, a standards of 3GPP project. SIP session createс high voice quality via dedicated bearer reserved for each SIP session. If the access type is LTE, then the voip session is termed as VoLTE( Voice over LTE) and if access type is Wifi, the voip session is termed as VoWIFI.