What is Softphone

Softphone is a phone that allows you to talk using VoIP without necessarily having a physical phone set. It is a piece of software, or part of a VoIP software, that acts as an edge allowing you to dial numbers and bring out other phone functions using a screen (that of your computer or PDA) and your mouse, keyboard or keypad. Commonly, the interface resembles a phone with buttons in place of the keys, which you press using the mouse to dial. An example is the Skype Softphone. For speaking and listening, a headset and microphone be sufficient. You can also use an IP phone or handset.

What is Softphone

Softphone is a phone that allows you to talk using VoIP without necessarily having a physical phone set. It is a piece of software, or part of a VoIP software, that acts as an edge allowing you to dial numbers and bring out other phone functions using a screen (that of your computer or PDA) and your mouse, keyboard or keypad. Commonly, the interface resembles a phone with buttons in place of the keys, which you press using the mouse to dial. An example is the Skype Softphone. For speaking and listening, a headset and microphone be sufficient. You can also use an IP phone or handset.

H.323 Protocl

H.323 is a protocol standard for multimedia communications. H.323 was designed to support real-time transfer of audio and video data over packet networks. The standard involve several different protocols outside specific aspects of Internet telephony. The International Telecommunication Union (ITU-T) maintains H.323 and these related standards. Most voice over IP (VoIP) applications utilizes H.323.
H.323 supports call setup, rip down and forward/transfer. Architectural basics of a H.323 based system are Terminals, Multipoint Control Units (MCUs), Gateways, an optional Gatekeeper and Border Elements. Different functions of H.323 run over either TCP or UDP. H.323 competes with the newer Session Initialization Protocol (SIP) another established standard often found in VoIP systems. A key feature of H.323 is Quality of Service. H.323 Technology allows real-time prioritization and traffic management constraint to be placed on the “best effort” packet release system like TCP/IP over Ethernet.

Session Initiation Protocol

SIP, the session initiation protocol, is the IETF protocol for VOIP and other text and multimedia sessions, like instant messaging, video, online games and other services. SIP is very much like HTTP, the Web protocol, or SMTP. Messages consist of headers and a message body. SIP message bodies for phone calls are defined in SDP (Session description protocol)

SIP is a text-based protocol that uses UTF-8 encoding
SIP uses port 5060 both for UDP and TCP. SIP may use other transports

SIP offers all potentialities of the common Internet Telephony features like:

call or media transfer
call conference
call hold

Since SIP is a flexible protocol, it is possible to add more features and keep downward interoperability. SIP also does suffer from NAT or firewall restrictions. (Refer to NAT and VOIP). SIP can be regarded as the enabler protocol for telephony and voice over IP (VoIP) services. The following features of SIP play a major role in the enablement of IP telephony and VoIP:

Name Translation and User Location: Ensuring that the call reaches the called party wherever they are located. Carrying out any mapping of descriptive information to location information. Ensuring that details of the nature of the call (Session) are supported.
Feature Negotiation: This allows the group involved in a call (this may be a multi-party call) to agree on the features supported recognizing that not all the parties can support the same level of features.
Call Participant Management: During a call a participant can bring other users onto the call or cancel connections to other users. In addition, users could be transferred or placed on hold.
Call feature changes: A user should be able to change the call characteristics during the course of the call.
Media negotiation: The inherent SIP mechanisms that enable negotiation of the media used in a call, enable selection of the appropriate codec for establishing a call between the various devices. This way, less advanced devices can participate in the call, provided the appropriate codec is selected.

What is WiMax

WiMAX is a wireless digital communications system, also known as IEEE 802.16, that is intended for wireless "metropolitan area networks". WiMAX can provide broadband wireless access (BWA) up to 30 miles (50 km) for fixed stations, and 3 - 10 miles (5 - 15 km) for mobile stations. The WiFi 802.11 wireless local area network standard is limited in most cases to only 100 - 300 feet (30 - 100m). With WiMAX, WiFi-like data rates are easily supported, but the issue of interference is lessened. WiMAX operates on both licensed and non-licensed frequencies, providing a regulated environment and viable economic model for wireless carriers.WiMAX can be used for wireless networking in much the same way as the more common WiFi protocol. WiMAX is a second-generation protocol that allows for more efficient bandwidth use, interference avoidance, and is intended to allow higher data rates over longer distances.

NGN (Next Generation Networks)

Strong competition is expected in the information networking ground over the next 5-10 years. As the competition increases, it will be essential for companies to position themselves suitably to take gain of their core competencies and to practice for the emerging telecommunications environment. In this competitive environment, mergers, alliances, and the onslaught of new entrants into the market have service providers struggling to find pioneering ways to retain and/or attract the most lucrative subscribers. Today’s service providers are striving to differentiate themselves within this going up competitive landscape by searching for ways to brand and bundle new services, achieve operational cost reductions, and strategically position themselves in relation to their competition. As Figure 1 illustrate, the top 15% of today’s housing subscribers in the US are said to account for about 95% of carrier income, Thus, many service providers are looking to Next Generation Network (NGN) services as a means to attract and/or retain the most lucrative customers.

SIGTRAN Protocol

Common Channel Signaling System #7 ( SS7) transports SS7 messages over an SS7 network (normally a DS0A connection) and then there is SIGTRAN. SIGTRAN is a set of protocols defined to transport SS7 messages over IP networks. SIGTRAN allows IP networks to inter-work with the Public Switched Telephone Network (PSTN) and vice versa.
ZYTRAX now offers SIGTRAN and SS7 training.
Notes: There are two other IETF standards which define transport of PSTN signaling (non-SS7) over IP networks: ISDN User Adaption (IUA) which provides adaption at the the Q.921 layer to allow transparent transport of Q.931 messages to, say, a VoIP gateway. IUA is defined by RFC4233. V5.2 User Adaption (V5UA) which provides for the transport of ETSI standard V5.2 messages in an IP network by adaption at the LAPV5 level. V5UA is defined by RFC3807.

The SIGTRAN Protocol Stack

The SIGTRAN protocol stack consists of 3 components:
A standard IP layer.
A common signaling transport protocol, Stream Control Transmission Protocol (SCTP): SCTP provides connection oriented reliable transfer of user messages between peer SCTP users (Adaptation layer protocols). Note: The SCTP layer replaces a normal TCP/UDP layer.
An Adaptation layer: Protocols defined for this layer are M2PA, M2UA, M3UA, and SUA.

Introduction to NGN

As a Soft Switch product, SoftX3000 is located at the core control layer of NGN scheme.

SoftX3000 acts as a generic Call Controller in the packet-switched network, supports the inter working between PSTN, H.323, SIP, and MGCP domains.

SoftX3000 provides H.248 and MGCP based bearer control.

In the NGN solution of Huawei, SoftX3000 acts as the core of NGN, inter working with other

NGN components through the open network adopting distributed standard protocols