Report status

From Dr. Mashiur Rahman :: ICT expert :: VoIP & Nanotechnology

Jump to: navigation, search

Contents

IGW Working Process and Their Over-all Network Structure

IGW

IGW stands for International Gateway.In Bangladesh, the term IGW is used as the government approved international incoming/outgoing VoIP communication gateway.Which implies every legal VoIP call to or from Bangladesh should be transferred through the IGWs.

Background

The VoIP business started in this country about 5-10 years before. It became popular among the end users very quickly, because the expense of international calls over the government provided PSTN network was huge. Comparing to that international calls through VoIP were a lot cheaper.So, the government started loosing their revenue.During that time the government had no particular laws for the VoIP service providers.Eventually, VoIP service providing were declared as banned in Bangladesh.

Later on 18th of February 2008, BTRC invited 38 selected Bangladeshi companies for the bid or auction for the International Gateway (IGW) Services License. Later, BTRC declared three companies,

  • Novotel Limited
  • Bangla Trac Communications Ltd.
  • Mir Telecom

to be the winner of the the bid.All these companies successfully won with 51.75% of revenue sharing.

What is VoIP?

Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks. Other terms frequently encountered and synonymous with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone.

Internet telephony refers to communications services — voice, facsimile, and/or voice-messaging applications — that are transported via the Internet, rather than the public switched telephone network (PSTN). The basic steps involved in originating an Internet telephone call are conversion of the analog voice signal to digital format and compression/translation of the signal into Internet protocol (IP) packets for transmission over the Internet; the process is reversed at the receiving end.[1]

VoIP Technologies and Implementation

Basic VoIP call flow

In VoIP calls signaling between two end points is first established. If signal establishment is successfully done voice transfer between two ends starts.Different protocols are used to establish this communication.The most popular and used protocols for VoIP communications are

  • H.323
  • IP Multimedia Subsystem (IMS)
  • Session Initiation Protocol (SIP)
  • Real-time Transport Protocol (RTP)

We will discuss this protocols in our next section.

Signaling Protocol For IP Calls

Signaling protocols are used to establish the initial inter-connection between two end-points.The voice or media transmission starts only when signaling is successful. Among the signaling protocols the followings are most prominent

  • SIP
  • H323

Session Initiation Protocol (SIP)

SIP was originally designed by Henning Schulzrinne and Mark Handley starting in 1996. The latest version of the specification is RFC 3261 from the IETF Network Working Group.In November 2000, SIP was accepted as a 3GPP signaling protocol and permanent element of the IP Multimedia Subsystem (IMS) architecture for IP-based streaming multimedia services in cellular systems.The SIP protocol is a TCP/IP-based Application Layer protocol. SIP is designed to be independent of the underlying transport layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol (UDP), or Stream Control Transmission Protocol (SCTP).It is a text-based protocol, incorporating many elements of the Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP),allowing for direct inspection by administrators.[2]The following picture describes a basic signaling flow of a SIP call.

Sip call flow.png

When a caller ( Phone A) calls through SIP signaling, a SIP INVITE message is sent to the server (or Soft-Switch ).The SIP header contains necessary information about both the called party and calling party.The server will forward the invite message to the calling party.In the mean time the SIP server will send a Status: 100 Trying message to the calling party.After recieving the INVITE message the called party will send the calling party Status: 183 Session progress message via the server.If the calling party accept the call it will send a Status: 200 OK message.The calling party will send a SIP ACK acknowledgement  message to the called party.And threfore, the signaling is established .Then, the media transfering starts.For media transffering typically the Real-time Transffer Protocol(RTP) is used.After finishing the session SIP: BYE message is used to teardown the session.The following picture shows the basic stracture of a SIP header.In the later parts of this paper we will go more deeper about the SIP header.

 
http://ntrg.cs.tcd.ie/undergrad/4ba2/ipng/ian.sip.header.html

H.323

H.323 Call Signaling is based on the ITU-T Recommendation Q.931 protocol and is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG over ISDN. A call model, similar to the ISDN call model, eases the introduction of IP telephony into existing networks of ISDN-based PBX systems, including transitions to IP-based Private Branch eXchanges (PBXs).

Within the context of H.323, an IP-based PBX might be an H.323 Gatekeeper or other call control element that provides service to telephones or videophones. Such a device may provide or facilitate both basic services and supplementary services, such as call transfer, park, pick-up, and hold.

While H.323 excels at providing basic telephony functionality and interoperability, H.323’s strength lies in multimedia communication functionality designed specifically for IP networks.
H.323 is a system specification that describes the use of several ITU-T and IETF protocols. The protocols that comprise the core of almost any H.323 system are:[6]

  • H.225.0 Registration, Admission and Status (RAS), which is used between an H.323 endpoint and a Gatekeeper to provide address resolution and admission control services.
  • H.225.0 Call Signaling, which is used between any two H.323 entities in order to establish communication.
  • H.245 control protocol for multimedia communication, which describes the messages and procedures used for capability exchange, opening and closing logical channels for audio, video and data, control and indications.
  • Real-time Transport Protocol (RTP), which is used for sending or receiving multimedia information (voice, video, or text) between any two entities.

Many H.323 systems also implement other protocols that are defined in various ITU-T Recommendations to provide supplementary services support or deliver other functionality to the user. Some of those Recommendations are:[citation needed]

  • H.235 series describes security within H.323, including security for both signaling and media.
  • H.239 describes dual stream use in videoconferencing, usually one for live video, the other for still images.
  • H.450 series describes various supplementary services.
  • H.460 series defines optional extensions that might be implemented by an endpoint or a Gatekeeper, including ITU-T Recommendations H.460.17,H.460.18, and H.460.19 for Network address translation (NAT) / Firewall (FW) traversal.

In addition to those ITU-T Recommendations, H.323 utilizes various IETF Request for Comments (RFCs) for media transport and media packetization, including the Real-time Transport Protocol (RTP).[5]

Signaling Protocol For TDM Calls

  • ISUP

Media Transfer Protocol

For media transmission generally Real Time Transfer Protocol(RTP)is used.The reason of using RTP in VoIP communication because it allows real-time media transport.The Real-time Transport Protocol (RTP) defines a standardized packet format for delivering audio and video over the Internet. It was developed by the Audio-Video Transport Working Group of the IETF and first published in 1996 as RFC 1889, and superseded by RFC 3550 in 2003.RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications and web-based push to talk features. For these it carries media streams controlled by H.323, MGCP, Megaco, SCCP, or Session Initiation Protocol (SIP) signaling protocols, making it one of the technical foundations of the Voice over IP industry.[6]

Transport Layer Protocols

For transport layer protocol we use User Datagram Protocol (UDP).TCP/IP is not proffered because of its simple transmission model without implicit hand-shaking dialogues for guaranteeing reliability, ordering, or data integrity.UDP assumes that error checking and correction is either not necessary or performed in the application, avoiding the overhead of such processing at the network interface level. Time-sensitive applications often use UDP because dropping packets is preferable to waiting for delayed packets, which may not be an option in a real-time system.[6]

Previous Network Architecture

ARC IMAGE.JPG

The Technologies/ Protocol That are Used

Basic call flow settings

  • IP Call->MUX->Router->SBC->Soft-Switch->Media Gateway->ICX
  • TDM Call->MUX->Media Gateway->ICX


Real Time Data Analyzing

Technical Limitations

Proposed Solutions

Conclusion

Reference

Retrieved from "http://wiki.biggani.org/Report_status"
Personal tools